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diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..8b51f1ba2 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,49582 @@ +2017-03-23 20:33 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.15.0-rc1 Released. + +2017-03-23 14:03 +0000 [f1b34e6eb4] Kevin Harwell <kharwell@digium.com> + + * AMI: Updated version + + Updated the AMI version for the following reason (see CHANGES for more details): + + The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now + contains a new optional parameter, 'MatchHeader'. + + Change-Id: I9aeac4decc89f9b464b3f026e97c7ef1acc79242 + +2017-03-14 16:45 +0000 [398e5ec16c] Richard Begg <asterisk@meric.id.au> + + * res_pjsip_session: Enable RFC3578 overlap dialing support. + + Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched + destinations) as currently provided by chan_sip is missing from res_pjsip. + This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] + which when set to yes enables 484 responses to partial destination + matches rather than the current 404. + + ASTERISK-26864 + + Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6 + +2017-03-21 06:59 +0000 [218f618095] Sean Bright <sean.bright@gmail.com> + + * res_hep: Capture actual transport type in use + + Rather than hard-coding UDP, allow consumers of the HEP API to specify + which protocol is in use. Update the PJSIP provider to pass in the + current protocol type. + + ASTERISK-26850 #close + + Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978 + +2017-03-21 09:57 +0000 [1c8b81a2a4] Sean Bright <sean.bright@gmail.com> + + * Revert "app_queue: Handle the caller being redirected out of a queue bridge" + + This reverts commit 163e9e53dc7d84dd42721e733b7706c8147bdd27. + + Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b + +2017-03-21 08:26 +0000 [b3cc20799b] Sean Bright <sean.bright@gmail.com> + + * res_pjsip_messaging: Check URI type before dereferencing + + We aren't validating that the URI we just parsed is a SIP/SIPS one before + trying to access the user, host, and port members of a possibly uninitialized + structure. + + Also update the MessageSend documentation to indicate what 'from' formats are + accepted. + + ASTERISK-26484 #close + Reported by: Vinod Dharashive + + Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30 + +2017-03-13 15:21 +0000 [91c97b5da5] Joshua Elson <joshelson@gmail.com> + + * pjsip: prevent memory corruption on creation of xml bodies + + ASTERISK-26776 #close + + Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2 + +2017-03-20 16:27 +0000 [7f34c11b6a] Sean Bright <sean.bright@gmail.com> + + * bridge_softmix: Ignore non-voice frames from translator + + Some codecs - codec_speex specifically - take voice frames and return + other types of frames, like CNG. If we subsequently treat those as + voice frames, we'll run into trouble when destroying the frame because + of the requirement that each voice frame have an associated format. + + ASTERISK-26880 #close + Reported by: Kirsty Tyerman + + Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c + +2017-03-18 12:30 +0000 [38cebc73a3] Sean Bright <sean.bright@gmail.com> + + * thread safety: Don't use getprotobyname() + + POSIX does not require getprotobyname() to be thread safe and some + implementations use static memory which causes issues when multiple + threads are used. + + Further, our usage of it today is just to ultimately get IPPROTO_TCP + for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. + + Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48 + +2017-03-19 13:26 +0000 [265455bc2d] Sean Bright <sean.bright@gmail.com> + + * res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret + + We are currently passing in the capacity of the read buffer instead of the + number of bytes that we actually read off the wire. + + Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36 + +2017-03-14 09:27 +0000 [76afb9e18a] Robert Mordec <r.mordec@slican.pl> + + * app_queue: Member stuck as pending after forwarding previous call from queue + + Queue member will get stuck in pending_members if queue calls a device + that is different from the one observed for state changes. + + This patch removes members from pending_members as a result of channel stasis + events such as blind or attended transfers and hangup. + + ASTERISK-26862 #close + + Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727 + +2017-02-22 23:26 +0000 [60b372a883] Richard Mudgett <rmudgett@digium.com> + + * CHANNEL(callid): Give dialplan access to the callid. + + * Added CHANNEL(callid) to retrieve the call identifier log tag associated + with the channel. Dialplan now has access to the call log search key + associated with the channel so it can be saved in case there is a problem + with the call. + + ASTERISK-26878 + + Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f + +2017-03-16 08:42 +0000 [9a57b24e17] Sean Bright <sean.bright@gmail.com> + + * app_queue: Fix locking behavior in stasis message handlers + + The queue_stasis_data structure contains various mutable fields that require + appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and + 'caller_uniqueid' fields need to be locked when read from or written to. + + Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088 + +2017-03-07 19:28 +0000 [8721d0bf1b] Sean Bright <sean.bright@gmail.com> + + * chan_sip: Add rtcp-mux support + + ASTERISK-26846 #close + + Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639 + +2017-03-16 16:50 +0000 [792171ea9e] Richard Mudgett <rmudgett@digium.com> + + * app_confbridge: Fix ConfbridgeTalking AMI event description. + + Thanks to Chris Howard for pointing this out on the wiki. + + Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705 + +2017-03-16 16:37 +0000 [047fb7f11e] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed. + + struct ast_rtcp does not define the dtls member if SRTP is not enabled. + + ASTERISK-26732 + + Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e + +2017-03-16 15:45 +0000 [a75f02c089] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_sdp_rtp.c: Fix cut-n-paste error + + We were inadvertenly referencing the cos_video option to determine if we + should set the tos_audio and cos_audio value on the RTP instance. + + Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0 + +2017-03-16 10:39 +0000 [776ffd7724] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_session: Only check localnet if it is defined + + If local_net is not defined on a transport, transport_state->localnet + will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in + this case, causing the external_media_address, if set, to be skipped. + + This patch causes us to only check if we are sending within a network if + local_net is defined. + + ASTERISK-26879 #close + + Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb + +2017-03-14 16:22 +0000 [139bc3495f] Richard Begg <asterisk@meric.id.au> + + * res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport + + Currently a wildcard address is used for the local RTP socket, which + will not always result in the same address as used by the SIP socket + (e.g. if explicit transport addresses are configured). + Use the transport's host address when binding new local RTP sockets if + available. + + ASTERISK-26851 + + Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a + +2017-03-16 09:07 +0000 [7ea7797e12] Joshua Colp <jcolp@digium.com> + + * res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped. + + This change removes an assumption that when DTLS is stopped + an RTCP session will be present on the RTP session. This is not + always the case. + + ASTERISK-26732 + + Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611 + +2017-03-07 08:33 +0000 [9b756662a8] gtjoseph <gjoseph@digium.com> + + * res_pjsip: Symmetric transports + + A new transport parameter 'symmetric_transport' has been added. + + When a request from a dynamic contact comes in on a transport with + this option set to 'yes', the transport name will be saved and used + for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. + It's saved as a contact uri parameter named 'x-ast-txp' and will + display with the contact uri in CLI, AMI, and ARI output. On the + outgoing request, if a transport wasn't explicitly set on the + endpoint AND the request URI is not a hostname, the saved transport + will be used and the 'x-ast-txp' parameter stripped from the + outgoing packet. + + * config_transport was modified to accept and store the new parameter. + + * config_transport/transport_apply was updated to store the transport + name in the pjsip_transport->info field using the pjsip_transport->pool + on UDP transports. + + * A 'multihomed_on_rx_message' function was added to + pjsip_message_ip_updater that, for incoming requests, retrieves the + transport name from pjsip_transport->info and retrieves the transport. + If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter + containing the transport name is added to the incoming Contact header. + + * An 'ast_sip_get_transport_name' function was added to res_pjsip. + It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a + transport name if endpoint->transport is set or if there's an + 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or + ipv6 address. Otherwise it returns NULL. + + * An 'ast_sip_dlg_set_transport' function was added to res_pjsip + which takes an ast_sip_endpoint, a pjsip_dialog, and an optional + pjsip_tpselector. It calls ast_sip_get_transport_name() and if + a non-NULL is returned, sets the selector and sets the transport + on the dialog. If a selector was passed in, it's updated. + + * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas + were modified to call ast_sip_dlg_set_transport() instead of their + original logic. + + * res_pjsip/create_out_of_dialog_request was modified to call + ast_sip_get_transport_name() and pjsip_tx_data_set_transport() + instead of its original logic. + + * Existing transport logic was removed from endpt_send_request + since that can only be called after a create_out_of_dialog_request. + + * res_pjsip/ast_sip_create_rdata was converted to a wrapper around + a new 'ast_sip_create_rdata_with_contact' function which allows + a contact_uri to be specified in addition to the existing + parameters. (See below) + + * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated + since all it did was transport selection and that is now done in + ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. + + * 'contact_uri' was added to subscription_persistence. This was + necessary because although the parsed rdata contact header has the + x-ast-txp parameter added (if appropriate), + subscription_persistence_update stores the raw packet which + doesn't have it. subscription_persistence_recreate was then + updated to call ast_sip_create_rdata_with_contact with the + persisted contact_uri so the recreated subscription has the + correct transport info to send the NOTIFYs. + + * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since + all it did was transport selection and that is now done in + ast_sip_create_dialog_uac. + + * pjsip_message_ip_updater/multihomed_on_tx_message was updated + to remove all traces of the x-ast-txp parameter from the + outgoing headers. + + NOTE: This change does NOT modify the behavior of permanent + contacts specified on an aor. To do so would require that the + permanent contact's contact uri be updated with the x-ast-txp + parameter and the aor sorcery object updated. If we need to + persue this, we need to think about cloning permanent contacts into + the same store as the dynamic ones on an aor load so they can be + updated without disturbing the originally configured value. + + You CAN add the x-ast-txp parameter to a permanent contact's uri + but it would be much simpler to just set endpoint->transport. + + Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f + +2017-03-15 13:24 +0000 [adad6020be] Richard Mudgett <rmudgett@digium.com> + + * autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. + + Dereferencing struct ast_autochan.chan without first calling + ast_autochan_channel_lock() is unsafe because the pointer could change at + any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() + itself uses struct ast_autochan.chan unsafely and can result in a deadlock + if the original channel happens to get destroyed after a masquerade in + addition to the pointer getting changed. + + The problem is more likely to happen with v11 and earlier because + masquerades are used to optimize out local channels on those versions. + However, it could still happen on newer versions if the channel is + executing a dialplan application when the channel is transferred or + redirected. In this situation a masquerade still must be used. + + * Added a lock to struct ast_autochan to safely be able to use + ast_autochan.chan while trying to get the channel lock in + ast_autochan_channel_lock(). The locking order is the channel lock then + the autochan lock. Locking in the other direction requires deadlock + avoidance. + + * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. + + * Fix unsafe ast_autochan.chan usages in app_chanspy.c. + + * app_chanspy.c: Removed unused autochan parameter from next_channel(). + + ASTERISK-26867 + + Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592 + +2017-03-07 14:13 +0000 [7bc69753bc] Mark Michelson <mmichelson@digium.com> + + * Add rtcp-mux support + + This commit adds support for RFC 5761: Multiplexing RTP Data and Control + Packets on a Single Port. Specifically, it enables the feature when + using chan_pjsip. + + A new option, "rtcp_mux" has been added to endpoint configuration in + pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with + whatever it communicates with. Asterisk follows the rules set forth in + RFC 5761 with regards to falling back to standard RTCP behavior if the + far end does not indicate support for rtcp-mux. + + The lion's share of the changes in this commit are in + res_rtp_asterisk.c. This is because it was pretty much hard wired to + have an RTP and an RTCP transport. The strategy used here is that when + rtcp-mux is enabled, the current RTCP transport and its trappings (such + as DTLS SSL session) are freed, and the RTCP session instead just + mooches off the RTP session. This leads to a lot of specialized if + statements throughout. + + ASTERISK-26732 #close + Reported by Dan Jenkins + + Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5 + +2017-03-09 11:05 +0000 [163e9e53dc] Sean Bright <sean.bright@gmail.com> + + * app_queue: Handle the caller being redirected out of a queue bridge + + A caller can leave the Queue() application after being bridged with a + member in a few ways: + + * Caller or member hangup + * Caller is transferred somewhere else (blind or atx) + * Caller is externally redirected elsewhere + + The first 2 scenarios are currently handled by subscribing to stasis + messages, but the 3rd is not explicitly covered. If a caller is + redirected away from the Queue() application, the member who was last + bridged with that caller will remain in an "In use" state until the + caller hangs up. + + This patch adds handling of the caller leaving the queue via + redirection. We monitor the caller-member bridge, and if the caller is + the one that leaves, we treat it the same as we would a caller hangup. + + ASTERISK-26400 #close + Reported by: Etienne Lessard + + Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334 + +2017-03-15 08:44 +0000 [7612601964] Joshua Colp <jcolp@digium.com> + + * res_pjsip_endpoint_identifier_ip: Don't output error if no header_match. + + This change ensures that if no header_match option is set on an + identify an error message is not output stating the option is set + to an invalid value. + + ASTERISK-26863 + + Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a + +2017-03-14 08:49 +0000 [48447313b6] Torrey Searle <torrey@voxbone.com> + + * res/res_pjsip_refer: call xfer w/o extension + + When transfering to a URI without an extension, ensure that the + s extension of the dialplan is entered + + ASTERISK-26869 #close + + Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525 + +2017-03-14 16:16 +0000 [9fd9b39e8b] Richard Mudgett <rmudgett@digium.com> + + * pbx.c: Fix crash from malformed exten pattern. + + Forgetting to indicate an exten is a pattern can cause a crash if the + "pattern" has a character set range. e.g., "9999[3-5]" The crash is due + to a buffer overwrite because the '-' exten eye-candy wasn't removed as + expected and overran the allocated space. + + The buffer overwrite is fixed two ways in this patch. + + 1) Fix ext_strncpy() to distinguish between pattern and non-pattern + extens. Now '-' characters are removed when they are eye-candy and not + when they are part of a pattern character set. Since the function is + private to pbx.c, the return value now returns the number of bytes written + to the destination buffer instead of the strlen() of the final buffer so + the callers that care don't need to add one. + + 2) Fix callers to ext_strncpy() to supply the correct available buffer + size of the destination buffer. + + ASTERISK-26668 + + Change-Id: I555d97411140e47e0522684062d174fbe32aa84a + +2017-03-14 16:51 +0000 [5389666d6f] Richard Begg <asterisk@meric.id.au> + + * chan_iax2: Reload of iax peer results in loss of host address/port + + When using a non-dynamic peer address, build_peer() invalidates the + peer address structure by setting the address family to unspecified. + However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() + will not amend the peer address if the cache is still valid, resulting + in peer connectivity failures. + To fix this, we call ast_dnsmgr_refresh() instead. + + ASTERISK-26865 + + Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082 + +2017-03-14 15:12 +0000 [658d59c683] Matt Jordan <mjordan@digium.com> + + * configure: Don't use the progress bar with curl when downloading to stdout + + In some scenarios, such as when there may not be a terminal (such as + inside a Docker container), curl will apparently direct the progress bar + to stdout. This can cause extra data to be appended to a file curl'd + down to stdout, resulting in md5 verification failures. + + This patch removes the progress bar, and tells curl to download the file + silently. + + ASTERISK-26872 #close + + Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c + +2017-03-14 07:50 +0000 [b3c2c996f1] Matt Jordan <mjordan@digium.com> + + * res_pjsip_endpoint_identifier_ip: Add an option to match requests by header + + This patch adds a new features to the endpoint identifier module, + 'match_header'. When set, inbound requests are matched by a provided SIP + header: value pair. This option works in conjunction with the existing + 'match' configuration option, such that if any 'match*' attribute + matches an inbound request, the request is associated with the specified + endpoint. + + Since this module now identifies by more than just IP address, + appropriate renaming of the module and/or variables can be done in a + non-release branch. + + ASTERISK-26863 #close + + Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 + (cherry picked from commit 30f52d79d7fc9ab0b628bef2b61ea515413795a2) + +2017-03-14 09:55 +0000 [51985565ef] Matt Jordan <mjordan@digium.com> + + * configs/samples/hep.conf.sample: Clarify how the HEP stack works + + This patch updates the documenation in hep.conf.sample to better specify + how the various HEP modules interact. + + ASTERISK-26717 #close + + Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124 + +2017-03-14 09:59 +0000 [f9b791debe] Roman Bedros (License 6842) + + * funcs/func_devstate: Remove new line in Device field of during module load + + During module loading of func_devstate, Asterisk emits the current + device state of all Custom device states currently stored in the AstDB. + This was erroneously including a new line character ('\n') to the end of + the device state, causing two new lines to be emitted in + DeviceStateChange AMI events. + + Note that this only happened for those device state changes that + occurred during startup. Regular device state changes for Custom device + states are handled elsewhere, and did not have the newline. + + ASTERISK-26643 #close + Reported by: Roman Bedros + Tested by: Matt Jordan + patches: + ami_devstate.diff uploaded by Roman Bedros (License 6842) + + Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93 + +2017-03-14 09:37 +0000 [216e28aa95] Matt Jordan <mjordan@digium.com> + + * main/stasis_cache: Demote the ERROR message when removing a nonexistent item + + This patch demotes the ERROR message that is displayed when a + nonexistent item is removed from the Stasis cache. The genesis of this + demotion is due to chan_sip's realtime peers and their interaction with + Asterisk's core ast_endpoint code, but ostensibly it could happen from + other channel drivers as well. + + Since Mark Michelson already did an excellent job of explaining on this + issue, it is quoted here for posterity: + + "Internally, when a realtime peer is retrieved, Asterisk creates an + ast_endpoint structure. When that peer is destroyed, the ast_endpoint is + destroyed as well. Part of the destruction of the ast_endpoint involves + clearing the Stasis cache of all information about that endpoint. The + problem here is that the act of creating the ast_endpoint is not enough + to actually put any information in the Stasis cache. Instead, something + has to happen, such as a state change, in order for the Stasis cache to + have any information about that endpoint. When a device registers, + chan_sip creates an ast_endpoint structure, processes the REGISTER, and + then destroys the ast_endpoint. When the ast_endpoint is destroyed, + there is nothing to destroy in the Stasis cache, so an error message is + emitted. When you use rtcachefriends, ast_endpoint structures persist + for the lifetime of the module and so you do not see this error + message." + + ASTERISK-25237 #close + + Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70 + +2017-03-12 09:21 +0000 [c8d1b915d7] Joshua Colp <jcolp@digium.com> + + * chan_pjsip: Don't assume a session will have a channel. + + When querying for PJSIP specific information using the dialplan + function CHANNEL() it is possible that the underlying session + will no longer have a channel associated with it. This is + most likely to occur when the RTCP HEP module attempts to get + the channel name. If this happens then a crash will occur. + + This change just adds a check that the channel exists on the + session before querying it. + + ASTERISK-26857 + + Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01 + +2017-03-13 10:45 +0000 [6d1eb880c2] gtjoseph <gjoseph@digium.com> + + * menuselect: Add a new 'options' support type + + The Binaural Rendering patches in the master branch required + menuselect to be updated with a new support type called 'option'. + This allows binaural rendering to be turned on or off when + bridge_softmix is built. This patch backports the 'option' + functionality to the 13 and 14 branches. + + Here's what it looks like in menuselect: + + [*] bridge_simple + [*] bridge_softmix + --- Module Options --- + [ ] binaural_rendering_in_bridge_softmix + + To create an option for a module, you can create (or update) the + menuselect-tree xml snippet in the directory where the module + resides and add a member element with an 'option' support_level. + + Example (abbreviated) from bridges/bridges.xml: + + <member name="binaural_rendering_in_bridge_softmix" + displayname="Enable binaural rendering in bridge_softmix" + remove_on_change="bridges/bridge_softmix.o bridges/bridge_softmix.so"> + <support_level>option</support_level> + <depend>bridge_softmix</depend> + <depend>fftw3</depend> + <defaultenabled>no</defaultenabled> + </member> + + The 'name' will be added or removed from the MENUSELECT_<dir> + make variable following the standard module "missing means yes" + rules. + + Example (abbreviated) from bridges/Makefile: + + ifeq ($(findstring binaural_rendering,$(MENUSELECT_BRIDGES)),) + bridge_softmix.o: _ASTCFLAGS+=-DBINAURAL_RENDERING + bridge_softmix.so: LIBS+=$(FFTW3_LIB) + endif + + Change-Id: I66d23755ed6e81f8d439cad410f2ffa7c30f25ad + +2017-03-10 20:29 +0000 [523de8eb8e] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Reduce the need for rebuilds + + Bundled pjproject should now only rebuild if one of the menuselect + "Compiler Flags" options changes. + + Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43 + +2017-03-07 08:12 +0000 [d3ef833b3b] Jean Aunis <jean.aunis@prescom.fr> + + * chan_sip: Call not cancelled after receiving a 422 response + + When receiving a 422 response, the invitestate variable must be reset to + INV_CALLING. + + ASTERISK-26841 + + Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099 + +2017-03-05 15:26 +0000 [67c989ce78] Daniel Journo <dan@keshercommunications.com> + + * pjsip/cli_commands: pjsip show channelstats shows wrong codec + + * cli_commands.c Fixed CLI output + + ASTERISK-26822 #close + + Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01 + +2017-03-07 07:37 +0000 [2a85888262] Joshua Colp <jcolp@digium.com> + + * res_pjsip_transport_websocket: Add support for IPv6. + + This change adds a PJSIP patch (which has been contributed upstream) + to allow the registration of IPv6 transport types. + + Using this the res_pjsip_transport_websocket module now registers + an IPv6 Websocket transport and uses it for the corresponding + traffic. + + ASTERISK-26685 + + Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647 + +2017-03-08 08:16 +0000 [bc6eeab822] Daniel Journo <dan@keshercommunications.com> + + * app_voicemail: Cannot set fromstring on a per-mailbox basis + + * apps/app_voicemail.c fromstring field added to mailbox which will + override the global fromstring if set. + + ASTERISK-24562 #close + + Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe + +2017-03-06 15:54 +0000 [d9972423d1] Daniel Journo <dan@keshercommunications.com> + + * Saynumber is trying to get "and" from "digits/" subfolder + + * say.c Changed 'digits/and' to 'vm-and' for en_GB + + ASTERISK-26598 #close + + Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe + +2017-03-06 13:15 +0000 [77901a58ca] Sean Bright <sean.bright@gmail.com> + + * pbx_spool: Gracefully handle long lines in call files + + Per the linked issue, we aren't checking the buffer filled by fgets() + to determine if it contains a newline, so we will fail to correctly + parse the trailing portion of a long line. + + This patch increases the buffer size from 256 to 1024, and skips any + line that exceeds that length, logging a warning in the process. + + ASTERISK-17067 #close + Reported by: Dave Olszewski + + Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0 + +2017-03-02 21:27 +0000 [4271c700f7] Richard Mudgett <rmudgett@digium.com> + + * core: Cleanup ast_get_hint() usage. + + * manager.c:manager_state_cb() Fix potential use of uninitialized hint[] + if a hint does not exist for the requested extension. Ran into this when + developing a testsuite test. The AMI event ExtensionStatus came out with + the hint header value containing garbage. The AMI event PresenceStatus + also had the same issue. + + * manager.c:action_extensionstate() no need to completely initialize the + hint[]. Only initialize the first element. + + * pbx.c:ast_add_hint() Remove unnecessary assignment. + + * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care + about the return value of ast_get_hint() there. + + Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b + +2017-02-16 04:22 +0000 [e510595c86] Jørgen H <asterisk.org@hovland.cx> + + * res_pjsip WebRTC/websockets: Fix usage of WS vs WSS. + + According to the RFC[1] WSS should only be used in the Via header + for secure Websockets. + + * Use WSS in Via for secure transport. + + * Only register one transport with the WS name because it would be + ambiguous. Outgoing requests may try to find the transport by name and + pjproject only finds the first one registered. This may mess up unsecure + websockets but the impact should be minimal. Firefox and Chrome do not + support anything other than secure websockets anymore. + + * Added and updated some debug messages concerning websockets. + + * security_events.c: Relax case restriction when determining security + transport type. + + * The res_pjsip_nat module has been updated to not touch the transport + on Websocket originating messages. + + [1] https://tools.ietf.org/html/rfc7118 + + ASTERISK-26796 #close + + Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12 + +2017-03-01 07:23 +0000 [76971d4c4a] Sean Bright <sean.bright@gmail.com> + + * res_config_pgsql: Make 'require' return consistent with other backends + + res_config_pgsql should match the behavior of other realtime backend + drivers so that queue_log can disable adaptive logging. + + ASTERISK-25628 #close + Reported by: Dmitry Wagin + + Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372 + +2017-02-28 09:41 +0000 [fa8f6c2fc4] Sean Bright <sean.bright@gmail.com> + + * res_config_pgsql: Release table locks where appropriate + + The find_table() functions NULL or a locked table pointer. We are + not consistently calling release_table() in failure paths. + + Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544 + +2017-02-28 05:41 +0000 [5b34b751a0] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * pjsip.conf.sample: user_agent: not a specific version + + Use the description of useragent from sip.conf here. + + ASTERISK-26825 #close + + Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755 + +2017-02-27 20:07 +0000 [8e6ecdade2] gtjoseph <gjoseph@digium.com> + + * res_pjsip_pubsub: Remove unneeded endpoint unref + + When a subscription was being recreated and the endpoint wasn't + found, we were trying to unref the endpoint. This was causing + FRACKs. Removed the unref. + + ASTERISK-26823 #close + + Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164 + +2017-02-16 04:16 +0000 [0595c31da7] Jørgen H <asterisk.org@hovland.cx> + + * res_pjsip: Fix crash when contact has no status + + This change fixes an assumption in res_pjsip that a contact will + always have a status. There is a race condition where this is + not true and would crash. The status will now be unknown when + this situation occurs. + + ASTERISK-26623 #close + + Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5 + +2017-02-21 18:06 +0000 [c07bcca87e] gtjoseph <gjoseph@digium.com> + + * res_pjsip_outbound_registration: Subscribe to network change events + + Outbound registration now subscribes to network change events + published by res_stun_monitor and refreshes all registrations + when an event happens. + + The 'pjsip send (un)register' CLI commands were updated to accept + '*all' as an argument to operate on all registrations. + + The 'PJSIP(Un)Register' AMI commands were also updated to + accept '*all'. + + ASTERISK-26808 #close + + Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25 + +2017-02-26 10:09 +0000 [d91f61f0b5] Vitezslav Novy <a1@vnovy.net> + + * chan_sip: Allow DTLS to be disabled when reloading. + + This change fixes a problem where removing the DTLS configuration + options and reloading would not disable DTLS. This occurred + because the DTLS configuration was not reset to an unconfigured + state on reload. + + ASTERISK-26313 + + Change-Id: I10952709cc4a7727fb50534b042bce9d64894b39 + +2017-02-27 12:25 +0000 [3d2c119778] gtjoseph <gjoseph@digium.com> + + * build: Warn if asterisk is installed in both 32 and 64 bit sys dirs + + ... and clean them both up on uninstall. + + We've fixed the issue where 'make install' was installing to + /usr/lib on 64-bit systems that use /usr/lib64. Now we need + to clean up the remnants in /usr/lib. + + * 'make install' now prints a warning if DESTDIR/ASTLIBDIR + contains 'lib64' and libasterisk* shared libraries or modules + are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed + to 'lib'. + + * 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and + DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. + + ASTERISK-26705 + + Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f + +2017-02-27 07:02 +0000 [eac818801b] Joshua Colp <jcolp@digium.com> + + * bridge_native_rtp: Handle case where channel joins already suspended. + + The bridge_native_rtp module did not properly handle the case where + a smart bridge operation occurs while a channel is suspended. In this + scenario the module would incorrectly set up local or remote RTP + bridging despite the media having to flow through Asterisk. The remote + endpoint would see two media streams and experience wonky audio. + + The module has been changed so that it ensures both channels are + not suspended when performing the native RTP bridging and this + requirement has been documented in the bridge technology. + + ASTERISK-26781 + + Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c + +2017-02-24 11:49 +0000 [d49af061bc] Joshua Colp <jcolp@digium.com> + + * config: Improve documentation and behavior of outbound_proxy option. + + This change updates the documentation for the outbound_proxy option + to ensure it is consistently stated that a full SIP URI must be + provided for the option. + + The res_pjsip_outbound_registration module has also been changed so + that the provided outbound_proxy value is checked to ensure it is a + URI and if not an error is output stating so. + + ASTERISK-26782 + + Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593 + +2017-02-09 18:05 +0000 [9c05ddbddd] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Update for pjproject 2.6 + + * Removed all 2.5.5 functional patches. + * Updated usages of pj_release_pool to be "safe". + * Updated configure options to disable webrtc. + * Updated config_site.h to disable webrtc in pjmedia. + * Added Richard Mudgett's recent resolver patches. + + Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7 + +2017-02-23 15:49 +0000 [bee55aaf2c] gtjoseph <gjoseph@digium.com> + + * build: Execute ldconfig to build cache. (take two) + + On some platforms a multiarch approach is used for libraries. + The build system does not take this into account and still + places libraries into the lib directory if no --libdir is + specified to configure. On initial startup this results in + libasteriskssl.so not being found, as it is not in the multiarch + lib directory. To make matters worse, options were being passed + to ldconfig on both Linux and FreeBSD that actually prevented + the rebuild of the cache. + + * Fedora has a /usr/share/config.site that automatically tells + autoconf to use /usr/lib64 but CentOS does not. This logic was + copied to configure.ac and modified so systems like Ubuntu, + which still use /usr/lib for 64-bit systems, aren't affected. + + Now that we have them in the correct directory... + + In order for the system loader to find libasteriskssl and + libasteriskpj, one of 3 things has to happen... + + - The linker cache must be rebuilt including the directory + where the libasterisk* libraries were installed. Only root + can rebuild the cache. This was busted. + - We have to link the asterisk binary with an rpath pointing + to the directrory where the libasterisk* libraries were + installed. This makes things very complicated and will happen + over the collective dead bodies of everyone who's had to + package a distribution with an rpath. + - Finally, you can start asterisk with LD_LIBRARY_PATH set to the + directrory where the libasterisk* libraries were installed. + + There are no other options. So... + + * The invokation of ldconfig has been moved from main/Makefile + to ASTTOPDIR/Makefile, the options have been removed, and + DESTDIR/ASTLIBDIR appended. If you aren't root, you will be + warned after the "Asterisk Installation Compete" banner that + you must re-run 'make install' as root, manually run + 'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with + LD_LIBRARY_PATH. + + ASTERISK-26705 + + Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982 + +2017-02-23 14:48 +0000 [da0cadd100] Sean Bright <sean.bright@gmail.com> + + * res_config_pgsql: Fix thread safety problems + + * A missing AST_LIST_UNLOCK() in find_table() + + * The ESCAPE_STRING() macro uses pgsqlConn under the hood and we were + not consistently locking before calling it. + + * There were a handful of other places where pgsqlConn was accessed + directly without appropriate locking. + + Change-Id: Iea63f0728f76985a01e95b9912c3c5c6065836ed + +2017-02-22 08:53 +0000 [f1963c5b8d] Sean Bright <sean.bright@gmail.com> + + * res_config_ldap: Various code improvements + + The initial motivation for this patch was to properly handle memory + allocation failures - we weren't checking the return values from the + various LDAP library allocation functions. + + In the process, because update_ldap() and update2_ldap() were + substantially the same code, they've been consolidated. + + Change-Id: Iebcfe404177cc6860ee5087976fe97812221b822 + +2017-02-22 13:08 +0000 [1ec796ce18] Michael L. Young <elgueromexicano@gmail.com> + + * build_tools: Fix download_externals to allow the use of curl or wget + + Not sure if this is really a bug versus an improvement. I can see it being + viewed as a bug though by some. + + The current build_tools/download_externals file depends on wget in order to + download external modules. The current build system is able to discover + which tool to use for fetching remote files - either wget or curl. + + This patch takes advantage of this capability by modifying the two calls to + the wget binary to instead use what was discovered by the build system. + + ASTERISK-26812 #close + + Change-Id: If9411a2554f009274d377445613ae91192d948a1 + +2017-02-22 11:13 +0000 [5c9c097d17] Joshua Colp <jcolp@digium.com> + + * Revert "build: Execute ldconfig to build cache." + + This reverts commit d90430953c508670a67de68de400fef44f5e9fba. + + Change-Id: I758fe7ea0408f83a6df8e1774310d69f482700f6 + +2017-02-21 10:47 +0000 [ca6d001144] Sean Bright <sean.bright@gmail.com> + + * pbx_realtime: Prevent premature extension matching + + The patterns provided by pbx_realtime were checked in the order in + which they were returned from the realtime backend. If there was + overlap between multiple patterns, the first one to correctly match was + chosen even though it may not have been the best match. + + We now sort the patterns descending by their length and compare in that + order. There may be cases where this still results in a sub-optimal + match, but this patch should improve the overall behavior. + + ASTERISK-18271 #close + Reported by: Charlie Smurthwaite + + Change-Id: I56d9ac15810eb1775966b669c3028e32cc7bd809 + +2017-02-21 15:09 +0000 [0654bf637c] Peter Racz + + * pbx_dundi: DUNDi weight parameter not processed correctly + + The DUNDi weight field is not always converted from network byte order + to host byte order. This can result in incorrect weight values and + incorrect selection of DUNDi destinations. + + ASTERISK-18731 #close + Reported by: Peter Racz + Patches: + dundi_weight.patch (license #6290) patch uploaded by Peter Racz + + Change-Id: Iba3e1a700ff539db57211a7bbc26f7b22ea9a1be + +2017-02-21 10:47 +0000 [d5522de597] Sean Bright <sean.bright@gmail.com> + + * realtime: Fix ast_load_realtime_multientry handling + + ast_load_realtime_multientry() returns an ast_config structure whose + ast_categorys are keyed with the empty strings. Several modules were + giving semantic meaning to the category names causing problems at + runtime. + + * app_directory: Treated the category name as the mailbox name, and + would fail to direct calls to the appropriate extension after an + entry was chosen. + + * app_queue: Queues, queue members, and queue rules were all affected + and needed to be updated. + + * pbx_realtime: Pattern matching would never succeed because the + extension entered by the user was always compared to the empty + string. + + Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7 + +2017-02-21 08:56 +0000 [5eb7875243] Sean Bright <sean.bright@gmail.com> + + * realtime: Centralize some common realtime backend code + + All of the realtime backends create artificial ast_categorys to pass + back into the core as query results. These categories have no filename + or line number information associated with them and the backends differ + slightly on how they create them. So create a couple helper macros to + help make things more consistent. + + Also updated the call sites to remove redundant error messages about + memory allocation failure. + + Note that res_config_ldap sets the category filename to the 'table name' + but that is not read by anything in the core, so I've dropped it. + + Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897 + +2017-02-16 10:30 +0000 [d90430953c] Joshua Colp <jcolp@digium.com> + + * build: Execute ldconfig to build cache. + + On some platforms a multiarch approach is used for libraries. + The build system does not take this into account and still + places libraries into the lib directory if no --libdir is + specified to configure. On initial startup this results in + libasteriskssl.so not being found, as it is not in the multiarch + lib directory. + + This change does the minimally invasive thing and executes + ldconfig so that the libraries in the lib directory are found + and their location cached. By doing so Asterisk starts up fine. + + If DESTDIR is specified, however, the old logic is executed as + the install process may not have permission to alter the ldconfig + cache. + + ASTERISK-26705 + + Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4 + +2017-01-08 20:32 +0000 [3b606093d3] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation. + + The inbound authentication object is supposed to be immutable when it is + stored in sorcery. However, the immutable property is violated if the + authentication object does not have a realm set. + + The immutable contract violation has a different effect depending upon + what sorcery back end is used. If it is the config file back end you + would get the same object back until res_pjsip is reloaded. If it is the + real-time or AstDB back end you would get a new object on each query. If + it is cached you would get the same object back until it is refreshed from + the database. + + Once an inbound authentication object has its realm set it may or may not + get updated again if the default_realm changes. + + If the same authentication object is used for inbound and outbound + authentication then the immutable violation can make it very hard to + determine why the outbound authentication now fails. The only diagnostic + message is a complaint about no realms matching when it had worked + earlier. It fails because of the difference in behaviour for an empty + realm setting between inbound and outbound authentication objects. + + * Fixed the sorcery object immutable violation by creating a new object + and setting the default_realm on it instead. The new object is a shallow + copy for speed. + + * The auth_store thread storage no longer holds an auth ref. It + interferes with the shallow copy and never needed a ref anyway. + + ASTERISK-26799 #close + + Change-Id: I2328a52f61b78ed5fbba38180b7f183ee7e08956 + +2017-02-04 20:17 +0000 [6208962b00] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Update artificial auth whenever default_realm changes. + + There was code attempting to update the artificial authentication object + whenever the default_realm changed. However, once the artificial + authentication object was created it would never get updated. The + artificial authentication object would require a system restart for a + change to the default_realm to take effect. + + ASTERISK-26799 + + Change-Id: Id59036e9529c2d3ed728af2ed904dc36e7094802 + +2017-01-01 08:02 +0000 [9f11da85a2] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Update authentication realm documentation. + + Using the same auth section for inbound and outbound authentication is not + recommended. There is a difference in meaning for an empty realm setting + between inbound and outbound authentication uses. + + An empty inbound auth realm represents the global section's default_realm + value when the authentication object is used to challenge an incoming + request. An empty outgoing auth realm is treated as a don't care wildcard + when the authentication object is used to respond to an incoming + authentication challenge. + + ASTERISK-26799 + + Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce + +2017-02-13 17:11 +0000 [473813311b] Richard Mudgett <rmudgett@digium.com> + + * pjproject: Fixes to resolve DNS SRV crashes. + + * Re #1945 (misc): Don't trigger SRV complete callback when there is a + parse error. + + * srv_resolver.c: Don't try to send query if already considered resolved. + + ** In resolve_hostnames() don't try to resolve a query that is already + considered resolved. + + ** In resolve_hostnames() fix DNS typo in comments. + + ** In build_server_entries() move a common expression assigning to cnt + earlier. + + * sip_transport.c: Fix tdata object name to actually contain the pointer. + + It helps if the logs referencing a tdata object buffer actually have a + name that includes the correct pointer as part of the name. Also since + the tdata has its own pool it helps if any logs referencing the pool have + the same name as the tdata object. This change brings tdata logging in + line with how tsx objects are named. + + ASTERISK-26669 #close + ASTERISK-26738 #close + + Change-Id: I56af2ded25476b3e870ca586ee69ed6954ef75af + +2017-02-06 14:26 +0000 [d58fdae811] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Update some debug messages to get transaction name. + + * Removed overloaded unmatched response ignore. We obviously sent the + request so we shouldn't ignore it because it isn't new work. + + ASTERISK-26669 + ASTERISK-26738 + + Change-Id: I55fb5cadc83a8e6699b347c6dc7fa32c5a617d37 + +2017-02-04 16:00 +0000 [eb9ae4f7cb] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Record the serializer earlier on the tdata. + + When PJPROJECT needs to do a DNS resolution and there is not a cached + entry available, the SIP request message goes out on the PJSIP monitor + thread instead of the original serializer thread. Thus when the response + comes back it does not get processed by the original sending serializer. + + This patch records the serializer on tdata before passing a request + message to PJPROJECT where it can in Asterisk code. There are several + places in PJPROJECT for outbound registration and publishing support that + would need to record the serializer. Unfortunately, without replacing the + PJPROJECT DNS resolver as was done in v14 we cannot fix those without + modifying PJPROJECT. + + Even if we backported the DNS resolver from v14, the outbound registration + refresh timer does not go out on a serializer thread but the PJSIP monitor + thread. Fortunately, Asterisk's outbound publish support doesn't use the + auto refresh timer that would also not go out under the serializer thread. + + This patch is v13 only. + + ASTERISK-26669 + ASTERISK-26738 + + Change-Id: I9997b9ed6dbcebd2c37d6a67dc6dcee9c78914a4 + +2017-02-20 13:38 +0000 [57f19d6efb] Richard Mudgett <rmudgett@digium.com> + + * pjproject: Increase SENDER_WIDTH column size for 64-bit system logs. + + ASTERISK-26669 + ASTERISK-26738 + + Change-Id: Ibae6fc8cae69a1f04df0c577c4c11200499d6fe0 + +2017-02-20 06:28 +0000 [47daca8a2b] Sean Bright <sean.bright@gmail.com> + + * app_voicemail: vm_authenticate accesses uninitialized memory + + vm_authenticate doesn't always set the passed ast_vm_user argument, so + we initialize to 0 before passing it in. + + ASTERISK-25893 #close + Reported by: Filip Jenicek + + Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a + +2017-02-20 11:19 +0000 [06214173a9] Joshua Colp <jcolp@digium.com> + + * Revert "build: Execute ldconfig to build cache." + + This reverts commit e910dbab90ef3d628955c49f441b2c9dda1f222c. + + Change-Id: I242aa0a965a79738dc898299959c6d2e020c86bd + +2017-02-20 08:04 +0000 [c9ea98f9bf] gtjoseph <gjoseph@digium.com> + + * pjproject cli: Add object count after object lists + + When listing a container, we now print the number of objects + in the container at the end of the list. + + Change-Id: I791cbc3ee9da9a2af9adc655164b5d32953df812 + +2017-02-20 05:53 +0000 [d8972f50f4] Sean Bright <sean.bright@gmail.com> + + * res_config_ldap: Don't try to delete non-existent attributes + + OpenLDAP will raise an error when we try to delete an LDAP attribute + that doesn't exist. We need to filter out LDAP_MOD_DELETE requests + based on which attributes the current LDAP entry actually has. There + is of course a small window of opportunity for this to still fail, + but it is much less likely now. + + Change-Id: I3fe1b04472733e43151563aaf9f8b49980273e6b + +2017-02-20 05:49 +0000 [b980cae1f7] Sean Bright <sean.bright@gmail.com> + + * res_config_ldap: Remove extraneous line numbers from log messages + + Extraneous line numbers were being output in many log messages. These + have been removed. + + Change-Id: Ice9efa3d252ee87f37fa8f5ea852fda482675431 + +2017-02-20 05:45 +0000 [011b7be62a] Sean Bright <sean.bright@gmail.com> + + * res_config_ldap: Make memory allocation more consistent + + The code in update_ldap() and update2_ldap() was using both Asterisk's + memory allocation routines as well as OpenLDAP's. I've changed it so + that everything that is passed to OpenLDAP's functions are allocated + with their routines. + + Change-Id: Iafec9c1fd8ea49ccc496d6316769a6a426daa804 + +2017-02-20 05:30 +0000 [b2836dde7e] Sean Bright <sean.bright@gmail.com> + + * res_config_ldap: Fix configuration inheritance from _general + + The "_general" configuration section allows administrators to provide + both general configuration options (host, port, url, etc.) as well as a + global realtime-to-LDAP-attribute mapping that is a fallback if one of + the later sections do not override it. This neglected to exclude the + general configuration options from the mapping. As an example, during + my testing, chan_sip requested 'port' from realtime, and because I did + not have it defined, it pulled in the 'port' configuration option from + "_general." We now filter those out explicitly. + + Change-Id: I1fc61560bf96b8ba623063cfb7e0a49c4690d778 + +2017-02-20 05:27 +0000 [6d5e9993b2] Sean Bright <sean.bright@gmail.com> + + * res_config_ldap: Fix erroneous LDAP_MOD_REPLACE in LDAP modify + + We always treat the first change of our modification batch as a + replacement when it sometimes is actually a delete. So we have to pass + the correct arguments to the OpenLDAP library. + + ASTERISK-26580 #close + Reported by: Nicholas John Koch + Patches: + res_config_ldap.c-11.24.1.patch (license #6833) patch uploaded + by Nicholas John Koch + + Change-Id: I0741d25de07c9539f1edc6eff3696165dfb64fbe + +2017-02-15 11:55 +0000 [5b7c6678ae] Sean Bright <sean.bright@gmail.com> + + * res_config_sqlite3: Fix crash when loading with invalid config + + When ast_config_load() fails with CONFIG_STATUS_FILEINVALID, it has + already destroyed the ast_config struct for us. Trying to do it again + results in a crash. + + Change-Id: If6a5c0ca718ad428e01a1fb25beb209a9ac18bc6 + +2017-02-17 16:57 +0000 [096496e13e] Richard Mudgett <rmudgett@digium.com> + + * tcptls.c: Add some missing allocation failure checks. + + Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb + +2017-02-17 17:06 +0000 [047a1e7dcc] Sean Bright <sean.bright@gmail.com> + + * pjproject-bundled: Fix checksum verification when using cURL + + ASTERISK-26802 #close + Reported by: Michael L. Young + + Change-Id: Iad293080f55d4d69ab615717a15211d916eed613 + +2017-02-16 08:38 +0000 [2cd75fe311] Sean Bright <sean.bright@gmail.com> + + * realtime: Fix LIKE escaping in SQL backends + + The realtime framework allows for components to look up values using a + LIKE clause with similar syntax to SQL's. pbx_realtime uses this + functionality to search for pattern matching extensions that start with + an underscore (_). + + When passing an underscore to SQL's LIKE clause, it will be interpreted + as a wildcard matching a single character and therefore needs to be + escaped. It is (for better or for worse) the responsibility of the + component that is querying realtime to escape it with a backslash before + passing it in. Some RDBMs support escape characters by default, but the + SQL92 standard explicitly says that there are no escape characters + unless they are specified with an ESCAPE clause, e.g. + + SELECT * FROM table WHERE column LIKE '\_%' ESCAPE '\' + + This patch instructs 3 backends - res_config_mysql, res_config_pgsql, + and res_config_sqlite3 - to use the ESCAPE clause where appropriate. + + Looking through documentation and source tarballs, I was able to + determine that the ESCAPE clause is supported in: + + MySQL 5.0.15 (released 2005-10-22 - earliest version available from + archives) + PostgreSQL 7.1 (released 2001-04-13) + SQLite 3.1.0 (released 2005-01-21) + + The versions of the relevant libraries that we depend on to access MySQL + and PostgreSQL will not work on versions that old, and I've added an + explicit check in res_config_sqlite3 to only use the ESCAPE clause when + we have a sufficiently new version of SQLite3. + + res_config_odbc already handles the escape characters appropriately, so + no changes were required there. + + ASTERISK-15858 #close + Reported by: Humberto Figuera + + ASTERISK-26057 #close + Reported by: Stepan + + Change-Id: I93117fbb874189ae819f4a31222df7c82cd20efa + +2017-02-16 10:30 +0000 [e910dbab90] Joshua Colp <jcolp@digium.com> + + * build: Execute ldconfig to build cache. + + On some platforms a multiarch approach is used for libraries. + The build system does not take this into account and still + places libraries into the lib directory if no --libdir is + specified to configure. On initial startup this results in + libasteriskssl.so not being found, as it is not in the multiarch + lib directory. + + This change does the minimally invasive thing and executes + ldconfig so that the libraries in the lib directory are found + and their location cached. By doing so Asterisk starts up fine. + + ASTERISK-26705 + + Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519 + +2017-02-16 05:46 +0000 [9b02bbfa88] Sean Bright <sean.bright@gmail.com> + + * res_config_sqlite3: Properly create missing columns when necessary + + There were two specific issues resolved here: + + 1) The code that iterated over the required fields + (via ast_realtime_require) was broken for the RQ_INTEGER1 field + type. Iteration would stop when the first RQ_INTEGER1 (0) field + was encountered. + + 2) sqlite3_changes() was used to try and count the number of rows + returned by a SELECT statement. sqlite3_changes() only counts + affected rows, so this was always returning the value from the + most recent data modification statement. We now separate read-only + queries from data modification queries and count rows appropriately + in both cases. + + ASTERISK-23457 #close + Reported by: Scott Griepentrog + + Change-Id: I91ed20494efc3fcfbc2a96ac7646999a49814884 + +2017-02-15 14:44 +0000 [0fc27fa364] Joshua Elson <joshelson@gmail.com> + + * http: Ensure capath is defined on all http creations + + ASTERISK-26794 #close + + Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1 + +2017-02-15 23:09 +0000 [7aa731c1c7] Igor Goncharovsky <igor.goncharovsky@gmail.com> + + * chan_unistim: fix char type to have consistent behavior on ARM + + There is difference exists in behaviour of char type on x86 and ARM. + On x86 by default char variable type means signed char, but in ARM + unsigned char used. This make binary calculations and negative values + works wrong on ARM. + + This patch change type of char variables used for store negative + values and binary calculations to signed char. + + ASTERISK-26714 + + Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab + +2017-02-07 13:17 +0000 [be77b845d9] gtjoseph <gjoseph@digium.com> + + * res_pjsip_pubsub: Correctly implement persisted subscriptions + + This patch fixes 2 original issues and more that those 2 exposed. + + * When we send a NOTIFY, and the client either doesn't respond or + responds with a non OK, pjproject only calls our + pubsub_on_evsub_state callback, no others. Since + pubsub_on_evsub_state (which does the sub_tree cleanup) does not + expect to be called back without the other callbacks being called + first, it just returns leaving the sub_tree orphaned. Now + pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE + which is what pjproject will set to tell us that it was the + transaction that timed out or failed and not the subscription + itself timing our or being terminated by the client. If is + TSX_STATE, pubsub_on_evsub_state now does the proper cleanup + regardless of the state of the subscription. + + * When a client renews a subscription, we don't update the + persisted subscription with the new expires timestamp. This causes + subscription_persistence_recreate to prune the subscription if/when + asterisk restarts. Now, pubsub_on_rx_refresh calls + subscription_persistence_update to apply the new expires timestamp. + This exposed other issues however... + + * When creating a dialog from rdata (which sub_persistence_recreate + does from the packet buffer) there must NOT be a tag on the To + header (which there will be when a client refreshes a + subscription). If there is one, pjsip_dlg_create_uas will fail. + To address this, subscription_persistence_update now accepts a flag + that indicates that the original packet buffer must not be updated. + New subscribes don't set the flag and renews do. This makes sure + that when the rdata is recreated on asterisk startup, it's done + from the original subscribe packet which won't have the tag on To. + + * When creating a dialog from rdata, we were setting the dialog's + remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. + When the client tried to resubscribe after a restart with the + correct cseq, we'd reject the request with an Invalid CSeq error. + + * The acts of creating a dialog and evsub by themselves when + recreating a subscription does NOT restart pjproject's subscription + timer. The result was that even if we did correctly recreate the + subscription, we never removed it if the client happened to go away + or send a non-OK response to a NOTIFY. However, there is no + pjproject function exposed to just set the timer on an evsub that + wasn't created by an incoming subscribe request. To address this, + we create our own timer using ast_sip_schedule_task. This timer is + used only for re-establishing subscriptions after a restart. + + An earlier approach was to add support for setting pjproject's + timer (via a pjproject patch) and while that patch is still included + here, we don't use that call at the moment. + + While addressing these issues, additional debugging was added and + some existing messages made more useful. A few formatting changes + were also made to 'pjsip show scheduled tasks' to make displaying + the subscription timers a little more friendly. + + ASTERISK-26696 + ASTERISK-26756 + + Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e + +2017-02-15 11:03 +0000 [73133d5354] Sean Bright <sean.bright@gmail.com> + + * res_rtp_asterisk: Use PJ_ICE_MAX_CAND instead of hard-coding 16 + + pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND, + which is a compile-time constant. Instead of hard-coding 16 when we + enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can + potentially collect more interfaces if the compile time options are + changed. + + Tangentially related to ASTERISK~24464 + + Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd + +2017-02-03 02:25 +0000 [99b40e72ae] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * libasteriskssl: do nothing with OpenSSL >= 1.1 + + OpenSSL 1.1 requires no explicit initialization. The hacks in the + library are not needed. They also happen to fail running Asterisk. + + ASTERISK-26109 #close + + Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100 + +2017-02-13 16:50 +0000 [4c31e03e80] Sean Bright <sean.bright@gmail.com> + + * app_voicemail: Allow 'Comedian Mail' branding to be overriden + + Original patch by John Covert, slight modifications by me. + + ASTERISK-17428 #close + Reported by: John Covert + Patches: + app_voicemail.c.patch (license #5512) patch uploaded by + John Covert + + Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6 + +2017-01-20 23:59 +0000 [e97e50b68b] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * tcptls: use TLS_client_method with OpenSSL 1.1 + + OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous + version-specific methods (such as TLSv1_client_method(). Other than + being simpler to use and more correct (gain support for TLS newer that + TLS1, in our case), the older ones produce a deprecation warning that + fails the build in dev-mode. + + ASTERISK-26109 #close + + Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07 + +2017-01-20 23:57 +0000 [0d555f0d81] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * openssl 1.1 support: use OPENSSL_VERSION_NUMBER + + Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect + the openssl 1.1 API. + + ASTERISK-26109 #close + + Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458 + +2017-01-25 16:25 +0000 [9d34df9a5e] Ryan Rittgarn <rrittgarn@techpro.com> + + * app_voicemail: VoiceMailPlayMsg did not play database stored messages + + When attempting to use VoiceMailPlayMsg with a realtime data backend + the message is located, but never retrieved. This patch adds the + required RETRIEVE and DISPOSE calls that will fetch the message from + the database (and IMAP storage as well for that matter). + + Also, removed extraneous make_file call. + + ASTERISK-26723 #close + + Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c + +2017-02-14 08:12 +0000 [f99e5f4de4] var <cbrunker@appenbutlerhill.com> + + * app_record: Add option to prevent silence from being truncated + + When using Record() with the silence detection feature, the stream is + written out to the given file. However, if only 'silence' is detected, + this file is then truncated to the first second of the recording. + + This patch adds the 'u' option to Record() to override that behavior. + + ASTERISK-18286 #close + Reported by: var + Patches: + app_record-1.8.7.1.diff (license #6184) patch uploaded by var + + Change-Id: Ia1cd163483235efe2db05e52f39054288553b957 + +2017-02-11 09:57 +0000 [ea8a610776] Sean Bright <sean.bright@gmail.com> + + * cli: Fix various CLI documentation and completion issues + + * app_minivm: Use built-in completion facilities to complete optional + arguments. + + * app_voicemail: Use built-in completion facilities to complete + optional arguments. + + * app_confbridge: Add missing colons after 'Usage' text. + + * chan_alsa: Use built-in completion facilities to complete optional + arguments. + + * chan_sip: Use built-in completion facilities to complete optional + arguments. Add completions for 'load' for 'sip show user', 'sip show + peer', and 'sip qualify peer.' + + * chan_skinny: Correct and extend completions for 'skinny reset' and + 'skinny show line.' + + * func_odbc: Correct completions for 'odbc read' and 'odbc write' + + * main/asterisk: Correct and extend completions for 'core show file + version.' + + * main/astmm: Use built-in completion facilities to complete arguments + for 'memory' commands. + + * main/bridge: Correct completions for 'bridge kick.' + + * main/ccss: Use built-in completion facilities to complete arguments + for 'cc cancel' command. + + * main/cli: Add 'all' completion for 'channel request hangup.' Correct + completions for 'core set debug channel.' Correct completions for 'core + show calls.' + + * main/pbx_app: Remove redundant completions for 'core show + applications.' + + * main/pbx_hangup_handler: Remove unused completions for 'core show + hanguphandlers all.' + + * res_sorcery_memory_cache: Add completion for 'reload' argument of + 'sorcery memory cache stale' and properly implement. + + Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca + +2017-01-13 11:21 +0000 [17030100ca] Norbert Varga <vnorbix@gmail.com> + + * chan_pjsip: Multidomain endpoint finding on call + + When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com), + the user part is stripped down as it would be a trunk with a specified user, + and only the host part is called as a PJSIP endpoint and can't be found. + This is not correct in the case of a multidomain SIP account, so the stripping + after the @ sign is done only if the whole endpoint (in multidomain case + 1000@test.com) can't be found. + + ASTERISK-26248 + + Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6 + +2017-02-13 05:05 +0000 [18f1b52601] Joshua Colp <jcolp@digium.com> + + * channel: Protect flags in ast_waitfor_nandfds operation. + + The ast_waitfor_nandfds operation will manipulate the flags + of channels passed in. This was previously done without + the channel lock being held. This could result in incorrect + values existing for the flags if another thread manipulated + the flags at the same time. + + This change locks the channel during flag manipulation. + + ASTERISK-26788 + + Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed + +2017-02-11 11:25 +0000 [a46a21642e] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip.c: Fix inconsistency between warning and action. + + The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE + but we have no authenticator registered to create the challenge. + + Change-Id: I62368180d774b497411b80fbaabd0c80841f8512 + +2017-02-11 11:26 +0000 [67b21dc63a] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Fix off-nominal tdata ref leak. + + Change-Id: I571f371d0956a8039b197b4dbd8af6b18843598d + +2017-02-09 10:01 +0000 [8936568515] Sean Bright <sean.bright@gmail.com> + + * manager: Restore Originate failure behavior from Asterisk 11 + + In Asterisk 11, if the 'Originate' AMI command failed to connect the provided + Channel while in extension mode, a 'failed' extension would be looked up and + run. This was, I believe, unintentionally removed in 51b6c49. This patch + restores that behavior. + + This also adds an enum for the various 'synchronous' modes in an attempt to + make them meaningful. + + ASTERISK-26115 #close + Reported by: Nasir Iqbal + + Change-Id: I8afbd06725e99610e02adb529137d4800c05345d + +2017-02-08 14:27 +0000 [2817f87d27] Richard Mudgett <rmudgett@digium.com> + + * core: Cleanup some channel snapshot staging anomalies. + + We shouldn't unlock the channel after starting a snapshot staging because + another thread may interfere and do its own snapshot staging. + + * app_dial.c:dial_exec_full() made hold the channel lock while setting up + the outgoing channel staging. Made hold the channel lock after the called + party answers while updating the caller channel staging. + + * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. + Also we need to use ast_hangup() instead of ast_channel_unref() at that + location. + + * channel.c:__ast_channel_alloc_ap() added a comment about not needing to + complete the channel snapshot staging on off-nominal exit paths. + + * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel + locks while staging the channels for the stats channel variables. + + Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a + +2017-02-10 09:35 +0000 [c7fcc4468f] gtjoseph <gjoseph@digium.com> + + * configs/samples: Fix placement of 'identify' entry in sorcery.conf + + The entry for 'identify' was incorrectly placed in the + res_pjsip section when it should be in + res_pjsip_endpoint_identifier_ip. + + ASTERISK-26785 #close + + Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a + +2017-02-08 11:50 +0000 [cbc23c31cf] Mark Michelson <mmichelson@digium.com> + + * Revert "Update qualifies when AOR configuration changes." + + This reverts commit 6492e91392b8fd394193e411c6eb64b45486093f. + + The change in question was intended to prevent the need to reload in + order to update qualifies on contacts when an AOR changes. However, this + ended up causing a deadlock instead. + + Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e + +2017-02-07 12:01 +0000 [7e14e086cf] nappsoft <infos@nappsoft.com> (license 6822) + + * srv: Fix crash when ast_srv_lookup is used and 0 records are returned. + + When performing an SRV lookup using the ast_srv_lookup function it + did not properly handle the situation where 0 records are returned. + If this happened it would wrongly assume that at least one record + was present. + + This change fixes the code so it will exit early if an error occurs + or if 0 records are returned. + + ASTERISK-26772 + patches: + srv_lookup.patch submitted by nappsoft (license 6822) + + Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351 + +2017-02-06 11:40 +0000 [7b39d6901a] Joshua Colp <jcolp@digium.com> + + * res_stasis_device_state: Protect the adding/removing of subscriptions. + + The adding and removing of device state subscriptions did not protect + fully against simultaneous manipulation. In particular the subscribe + case allowed a small window where two subscriptions could be added for + the same device state instead of just one. + + This change makes the code hold the subscriptions lock for the entirety + of each operation to ensure that two are not occurring at the same time. + + ASTERISK-26770 + + Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b + +2017-02-01 17:56 +0000 [c384dfd6b0] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Fix some off nominal tdata leaks. + + Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d + +2017-02-06 16:27 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.14.0-rc1 Released. + +2017-02-02 11:26 +0000 [70aff89e5d] Sean Bright <sean.bright@gmail.com> + + * res_odbc: Remove deprecated settings from sample configuration file + + ASTERISK-26704 #close + Reported by: Anthony Messina + + Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f + +2017-02-01 15:56 +0000 [3aee199913] Sean Bright <sean.bright@gmail.com> + + * audiohooks: Muting a hook can mute underlying frames + + If an audiohook is placed on a channel that does not require transcoding, + muting that hook will cause the underlying frames to be muted as well. + + The original patch is from David Woolley but I have modified slightly. + + ASTERISK-21094 #close + Reported by: David Woolley + Patches: + ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded + by David Woolley + + Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed + +2017-02-01 13:54 +0000 [6492e91392] Mark Michelson <mmichelson@digium.com> + + * Update qualifies when AOR configuration changes. + + Prior to this change, qualifies would only update in the following + cases: + * A reload of res_pjsip.so was issued. + * A dynamic contact was re-registered after its AOR's qualify_frequency + had been changed + This does not work well if you are using realtime for your AORs. You can + update your database to have a new qualify_frequency, but the permanent + contacts on that AOR will not have their qualifies updated. And the + dynamic contacts on that AOR will not have their qualifies updated until + the next registration, which could be a long time. + + This change seeks to fix this problem by making it so that whenever AOR + configuration is applied, the contacts pertaining to that AOR have their + qualifies updated. + + Additions from this patch: + * AOR sorcery objects now have an apply handler that calls into a newly + added function in the OPTIONS code. This causes all contacts + associated with that AOR to re-schedule qualifies. + * When it is time to qualify a contact, the OPTIONS code checks to see + if the AOR can still be retrieved. If not, then qualification is + canceled on the contact. + + Alterations from this patch: + * The registrar code no longer updates contact's qualify_frequence and + qualify_timeout. There is no point to this since those values already + get updated when the AOR changes. + * Reloading res_pjsip.so no longer calls the OPTIONS initialization + function. Reloading res_pjsip.so results in re-loading AORs, which + results in re-scheduling qualifies. + + Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121 + +2017-01-31 18:28 +0000 [43f0ff4b69] Richard Mudgett <rmudgett@digium.com> + + * channel.c: Fix unbalanced read queue deadlocking local channels. + + Using the timerfd timing module can cause channel freezing, lingering, or + deadlock issues. The problem is because this is the only timing module + that uses an associated alert-pipe. When the alert-pipe becomes + unbalanced with respect to the number of frames in the read queue bad + things can happen. If the alert-pipe has fewer alerts queued than the + read queue then nothing might wake up the thread to handle received frames + from the channel driver. For local channels this is the only way to wake + up the thread to handle received frames. Being unbalanced in the other + direction is less of an issue as it will cause unnecessary reads into the + channel driver. + + ASTERISK-26716 is an example of this deadlock which was indirectly fixed + by the change that found the need for this patch. + + * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue + did not add the same number of alerts to the alert-pipe. Correspondingly, + when there is an exceptionally long queue event, any removed frames did + not also remove the corresponding number of alerts from the alert-pipe. + + ASTERISK-26632 #close + + Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6 + +2017-01-31 16:38 +0000 [a199f94908] Richard Mudgett <rmudgett@digium.com> + + * res_agi: Prevent an AGI from eating frames it should not. (Re-do) + + A dialplan intercept routine is equivalent to an interrupt routine. As + such, the routine must be done quickly and you do not have access to the + media stream. These restrictions are necessary because the media stream + is the responsibility of some other code and interfering with or delaying + that processing is bad. A possible future dialplan processing + architecture change may allow the interception routine to run in a + different thread from the main thread handling the media and remove the + execution time restriction. + + * Made res_agi.c:run_agi() running an AGI in an interception routine run + in DeadAGI mode. No touchy channel frames. + + ASTERISK-25951 + + ASTERISK-26343 + + ASTERISK-26716 + + Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43 + +2017-01-31 16:32 +0000 [6bed318a66] Richard Mudgett <rmudgett@digium.com> + + * Frame deferral: Revert API refactoring. + + There are several issues with deferring frames that are caused by the + refactoring. + + 1) The code deferring frames mishandles adding a deferred frame to the + deferred queue. As a result the deferred queue can only be one frame + long. + + 2) Deferrable frames can come directly from the channel driver as well as + the read queue. These frames need to be added to the deferred queue. + + 3) Whoever is deferring frames is really only doing the __ast_read() to + collect deferred frames and doesn't care about the returned frames except + to detect a hangup event. When frame deferral is completed we must make + the normal frame processing see the hangup as a frame anyway. As such, + there is no need to have varying hangup frame deferral methods. We also + need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. + That fake hangup is to cause the PBX thread to break out of loops to go + execute a new dialplan location. + + 4) To properly deal with deferrable frames from the channel driver as + pointed out by (2) above, means that it is possible to process a dialplan + interception routine while frames are deferred because of the + AST_CONTROL_READ_ACTION control frame. Deferring frames is not + implemented as a re-entrant operation so you could have the unsupported + case of two sections of code thinking they have control of the media + stream. + + A worse problem is because of the bad implementation of the AMI PlayDTMF + action. It can cause two threads to be deferring frames on the same + channel at the same time. (ASTERISK_25940) + + * Rather than fix all these problems simply revert the API refactoring as + there is going to be only autoservice and safe_sleep deferring frames + anyway. + + ASTERISK-26343 + + ASTERISK-26716 #close + + Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496 + +2017-01-31 11:17 +0000 [e371e13b9e] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Handle invocation of callback on outgoing request when error occurs. + + There are some error cases in PJSIP when sending a request that will + result in the callback for the request being invoked. The code did not + handle this case and assumed on every error case that the callback was not + invoked. + + The code has been changed to check whether the callback has been invoked + and if so to absorb the error and treat it as a success. + + ASTERISK-26679 + ASTERISK-26699 + + Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91 + +2017-01-30 09:02 +0000 [339c30f2b6] Sean Bright <sean.bright@gmail.com> + + * res_rtp_asterisk: Swap byte-order when sending signed linear + + Before Asterisk 13, signed linear was converted into network byte order by a + smoother before being sent over the network. We restore this behavior by + forcing the creation of a smoother when slinear is in use and setting the + appropriate flags so that the byte order conversion is always done. + + ASTERISK-24858 #close + Reported-by: Frankie Chin + + Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9 + +2017-01-31 12:46 +0000 [7fd28cefdb] gtjoseph <gjoseph@digium.com> + + * debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts + + Forgot to install it with the original patch + + Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c + +2017-01-25 06:50 +0000 [456bc3c704] gtjoseph <gjoseph@digium.com> + + * debug_utilities: Add ast_logescalator + + The escalator works by creating a set of startup commands in cli.conf + that set up logger channels and issue the debug commands for the + subsystems specified. If asterisk is running when it is executed, + the same commands will be issued to the running instance. The original + cli.conf is saved before any changes are made and can be restored by + executing '$prog --reset'. + + The log output will be stored in... + $astlogdir/message.$uniqueid + $astlogdir/debug.$uniqueid + $astlogdir/dtmf.$uniqueid + $astlogdir/fax.$uniqueid + $astlogdir/security.$uniqueid + $astlogdir/pjsip_history.$uniqueid + $astlogdir/sip_history.$uniqueid + + Some minor tweaks were made to chan_sip, and res_pjsip_history + so their history output could be send to a log channel as packets + are captured. + + A minor tweak was also made to manager so events are output to verbose + when "manager set debug on" is issued. + + Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543 + +2017-01-23 09:35 +0000 [54b027916a] Torrey Searle <torrey@voxbone.com> + + * libastssl/pj: libastssl/pj should have an so_version + + Issue introduced in b59956a87. In the non-darwin case libastssl/pj + should be versioned. This causes the symbol file for this lib + to not be generated. + + Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c + +2017-01-24 19:51 +0000 [3c8f84786e] Kirill Katsnelson <kkm@smartaction.com> + + * make_build_h: handle backslashes in external strings + + LikewiseOpen creates user names with a backslash in them. A gentle + massage with sed(1) allows such strings to be inserted into build.h + properly quoted. I am also adding the same for host name and other + strings used in the script that are more or less user-controlled. + + ASTERISK-26754 + + Change-Id: Iac5ef2b67a68ee58f35ddbf86bb818ba6eabecae + +2017-01-17 20:46 +0000 [555e8cd2ba] Kirill Katsnelson <kkm@smartaction.com> + + * ast_careful_fwrite to support EPIPE gracefully + + When a reading end of the network socket is closed by an AMI manager, + the EPIPE is signaled when writing to our end, resulting in the + spurious log error message + + ast_careful_fwrite: fwrite() returned error: Broken pipe + + Previously EPIPE was handled in ast_carefulwrite() a few lines above, + but not in this function. + + ASTERISK-26753 + + Change-Id: I6a67335cd6526608bb9b78f796c626b1677664b8 + +2017-01-24 22:31 +0000 [be92f10a16] Kirill Katsnelson <kkm@smartaction.com> + + * app_queue: Fix queues randomly disappearing on reload + + With 500+ queues and a reload every minute, a random queue disappears + upon reload. The cause is mususe of the 'dead' flag. Namely, all queues + were marked dead up front, and then "resurrected" by dropping this flag + for those found in the configuration. But a queue marked dead can be + removed also when control leaves the app entry point on a PBX thread. + + With this change, the queue is marked only not found, and at the end of + reload only the queues that are still not found are actually marked as + dead, so the dead flag is never reset, and set only on positively dead + queues. + + ASTERISK-26755 + + Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf + +2017-01-26 07:57 +0000 [aae9df0643] Joshua Colp <jcolp@digium.com> + + * res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving. + + This change adds a missing unreference of the hostname when resolving and + also cleans up the iterator. + + ASTERISK-26735 + + Change-Id: Ic012ebaf3d89e714eec340b7b0c5e63c66af857a + +2017-01-25 15:26 +0000 [9e3150b98d] Mark Michelson <mmichelson@digium.com> + + * Add reload options to CLI/AMI stale object commands. + + Marking an object as stale in a memory cache is supposed to prime the + cache so that the next time the item is retrieved, the stale item is + deleted from the cache and a background task is run to re-populate the + cache with a fresh version of the object. + + The problem is, there are some object types out there for which there is + no natural reason that they would be retrieved from the backend with any + regularity. Outbound PJSIP registrations are a good example of this. At + startup, they are read, and an object-specific state is created that + refers to the initially-retrieved object for all time. + + Adding the "reload" option to the CLI/AMI commands gives the cache the + opportunity to manually re-retrieve the object from the backend, both + storing the new object in the cache and applying the new object's + configuration to the module that uses that object. + + Change-Id: Ieb1fe7270ceed491f057ec5cbf0e097bde96c5c8 + +2017-01-10 17:39 +0000 [c54f9d2bf0] Richard Mudgett <rmudgett@digium.com> + + * T.140: Fix format ref and memory leaks. + + * channel.c:ast_sendtext(): Fix T.140 SendText memory leak. + + * format_compatibility.c: T.140 RED and T.140 were swapped. + + * res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak. + + * res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic + scheduled red_write(). + + * res_rtp_asterisk.c: Some other minor misc tweaks. + + Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb + +2017-01-24 15:39 +0000 [a2f0adccbd] Joshua Colp <jcolp@digium.com> + + * res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0. + + When configuring a match using a netmask the error variable was + not defaulting to 0. For some people this would cause the code + to think an error occurred when adding the match when in reality + it added perfectly fine. + + ASTERISK-26693 + + Change-Id: I850c250813742bddde65c84e739093c9e01dfe56 + +2017-01-10 17:37 +0000 [607b3ac736] Richard Mudgett <rmudgett@digium.com> + + * astobj2.c: Add excessive ref count trap. + + Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a + +2017-01-10 13:11 +0000 [ab8cb5a7ce] Richard Mudgett <rmudgett@digium.com> + + * main/app.c: Memory corruption from early format destruction. + + * make_silence() created a malloced silence slin frame without adding a + slin format ref. When the frame is destroyed it will unref the slin + format that never had a ref added. Memory corruption is expected to + follow. + + * Simplified and fixed counting the number of samples in a frame list for + make_silence(). + + * Eliminated an unnecessary RAII_VAR associated with the make_silence() + frame. + + Change-Id: I47de3f9b92635b7f8b4d72309444d6c0aee6f747 + +2017-01-11 14:59 +0000 [dcd8e4b1a0] Richard Mudgett <rmudgett@digium.com> + + * frame.c: Fix off-nominal format ref leaks. + + * ast_frisolate() could leak frame format refs on allocation + failures. + + * Similified code in ast_frisolate() and code used by + ast_frisolate(). + + Change-Id: I79566d4d36b3d7801bf0c8294fcd3e9a86a2ed6d + +2017-01-13 19:08 +0000 [00a227e93d] Richard Mudgett <rmudgett@digium.com> + + * stasis_bridge.c: Fix off-nominal stasis control ref leak. + + Change-Id: Ib17218343a6596832060180e19386da9df150ac8 + +2017-01-10 12:30 +0000 [38a2021c68] Richard Mudgett <rmudgett@digium.com> + + * res_musiconhold.c: Fix format ref leak when parsing MOH config class. + + Change-Id: Ica8e8e2ce7604c2c61ec55bef07dc675361d2ea5 + +2017-01-10 14:03 +0000 [ab7a9fc5b2] Richard Mudgett <rmudgett@digium.com> + + * chan_oss.c: Fix format ref leak in oss_read(). + + Change-Id: I0a5d56c7dcf327d60f86a4c25a23571733709fd0 + +2017-01-10 17:48 +0000 [1484a991e1] Richard Mudgett <rmudgett@digium.com> + + * Add notes about embedded ast_frame structs holding a format ref. + + mod_format.h: Note ast_filestream.fr holds a format ref. + + translate.h: Note ast_trans_pvt.f holds a format ref. + + Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749 + +2017-01-19 09:05 +0000 [17f4989d49] gtjoseph <gjoseph@digium.com> + + * ari: Implement 'debug all' and request/response logging + + The 'ari set debug' command has been enhanced to accept 'all' as an + application name. This allows dumping of all apps even if an app + hasn't registered yet. To accomplish this, a new global_debug global + variable was added to res/stasis/app.c and new APIs were added to + set and query the value. + + 'ari set debug' now displays requests and responses as well as events. + This required refactoring the existing debug code. + + * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} + to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. + * In order to print the body of incoming requests even if a request + failed, the consumption of the body was moved from the ari stubs + to ast_ari_callback in res_ari.c and the moustache templates were + then regenerated. The body is now passed to ast_ari_invoke and then + on to the handlers. This results in code savings since that template + was inserted multiple times into all the stubs. + + An additional change was made to the ao2_str_container implementation + to add partial key searching and a sort function. The existing cli + code assumed it was already there when it wasn't so the tab completion + was never working. + + Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf + +2017-01-20 21:13 +0000 [30cb4eb57f] Richard Mudgett <rmudgett@digium.com> + + * PJPROJECT logging: Fix detection of max supported log level. + + The mechanism used for detecting the maximum log level compiled into the + linked pjproject did not work. The API call simply stores the requested + level into an integer and does no range checking. Asterisk was assuming + that there was range checking and limited the new value to the allowable + range. To get the actual maximum log level compiled into the linked + pjproject we need to get and save off the initial set log level from + pjproject. This is the maximum log level supported. + + * Get and save off the initial log level setting before altering it to the + desired level on startup. This has to be done by a macro rather than + calling a core function to avoid incorrectly linking pjproject. + + * Split the initial log level warning messages to warn if the linked + pjproject cannot support the requested startup level and if it is too low + to get the pjproject buildopts for "pjproject show buildopts". + + * Adjust the CLI "pjproject set log level" to check the saved max log + level and to generate normal output messages instead of a warning message. + + ASTERISK-26743 #close + + Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4 + +2017-01-21 14:43 +0000 [cd2677f966] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * tests: use datadir for sound files + + Some (voicemail-related) tests API symlinks beep.gsm and other files + from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR. + + ASTERISK-26740 #close + + Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89 + +2017-01-20 23:41 +0000 [b62f84bfb1] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * test_voicemail_api: order of params to VERIFY macros + + Fix order of parameters in calls to VM_API_INT_VERIFY and + VM_API_STRING_VERIFY + + ASTERISK-26739 #close + + Change-Id: I30dc6b36893aadad6012be3f16f93aa5720870d6 + Note: status: builds. Not tested any further. + +2017-01-05 13:21 +0000 [e3dcb9ddd9] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands. + + ASTERISK-23828 #close + + Change-Id: Ifb8a3b61f447aedc58a8e6b36a810f7566018567 + +2017-01-23 16:18 +0000 [75497c33ea] Mark Michelson <mmichelson@digium.com> + + * Free endpoint ACLs when destroying PJSIP endpoints. + + If endpoint ACLs were specified, they were not being freed + when endpoints were destroyed. On systems with realtime endpoints, this + could add up quickly since each DB lookup would allocate the ACL without + freeing it. + + ASTERISK-26731 #close + Reported by Ustinov Artem + + Change-Id: Ie1f8bf5b7a0de628c975beba01e69c56893331ad + +2017-01-23 09:10 +0000 [177e81ee47] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fix setting max log level + + An earlier attempt to prevent pjsua from spitting out an extra 6795 + lines of debug output every time the testsuite called it was also + turning off the ability for asterisk to output debug info when it + needed to. This patch reverts the earlier fix and instead adds + a pjproject patch that sets the startup log level to 1 for pjsua + pjsystest and the pjsua python binding. This is an asterisk-only + patch that does not affect pjproject functionality and will not be + submitted upstream. + + Change-Id: I347a8b58b2626f2906ccfc1d339e907627a0c9e8 + +2017-01-23 10:08 +0000 [6d23b2e360] Joshua Colp <jcolp@digium.com> + + * res_pjsip_endpoint_identifier_ip: Read settings before resolving. + + An option has been added, srv_lookups, which controls whether + SRV lookups are performed on the provided match hosts or not. + It was possible for this option to be applied after resolution + had already happened. + + This change makes it so hosts are stored away, settings are read + and applied, and then resolution is done. This ensures that no + matter the ordering the srv_lookups option is in effect. + + ASTERISK-26735 + + Change-Id: I750378cb277be0140f8c5539450270afbfc43388 + +2017-01-22 17:25 +0000 [a969bf3577] Richard Mudgett <rmudgett@digium.com> + + * LISTFILTER: Remove outdated ERROR message. + + Feeding LISTFILTER an empty variable results in an invalid ERROR message. + Earlier changes made the message useless because we can no longer tell if + the variable is empty or does not exist. It is valid to try to remove a + value from an empty list just as it is valid to try to remove a value that + is not in a non-empty list. + + * Removed the outdated ERROR message. + + * Added more test cases to the LISTFILTER unit test. + + Change-Id: Ided9040e6359c44a335ef54e02ef5950a1863134 + +2017-01-05 15:11 +0000 [3890337e7a] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Fix AMI event list counts. + + Fix the AMI PJSIPShowSubscriptionsInbound, PJSIPShowSubscriptionsOutbound, + and PJSIPShowResourceLists actions event counts. The reported counts may + not necessarily be accurate depending on what happens. + + The subscriptions count would be wrong if Asterisk ever has outbound + subscriptions. + + The resource list count could be wrong if a list were added or removed + during the AMI action being processed. + + Change-Id: I4344301827523fa174960a42c413fd19abe4aed5 + +2017-01-05 13:02 +0000 [fe4801c4f9] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Fix incorrect message string wrapping. + + Change-Id: Id771e6fe56d89ce365ddcbb423f820af97211120 + +2017-01-05 13:01 +0000 [46484b8730] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Eliminate trivial SCOPED_LOCK usage. + + Change-Id: Ie0b69a830385452042fa19e7d267c6790ec6b6be + +2017-01-05 12:58 +0000 [8160474d7d] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: alloca can never fail. + + Change-Id: Ia2a6158e5fdf311bc2a1c0c43417978de504b1f1 + +2017-01-13 11:03 +0000 [c628a7acac] gtjoseph <gjoseph@digium.com> + + * debug_utilities: Create ast_loggrabber + + ast_loggrabber gathers log files from customizable search patterns, + optionally converts POSIX timestamps to a readable format and + tarballs the results. + + Also a few tweaks were made to ast_coredumper. + + Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 + (cherry picked from commit 5fa1c56d7e76999aa14f133a33f6b168e7c3b99c) + +2017-01-01 03:47 +0000 [e335b706ee] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_authenticator_digest.c: Fix spacing in warning messages. + + Change-Id: I573f0343c0c63a785cd4da60d57cc9f8b9ce7f49 + +2017-01-12 15:58 +0000 [883e7fde31] Kevin Harwell <kharwell@digium.com> + + * abstract/fixed/adpative jitter buffer: disallow frame re-inserts + + It was possible for a frame to be re-inserted into a jitter buffer after it + had been removed from it. A case when this happened was if a frame was read + out of the jitterbuffer, passed to the translation core, and then multiple + frames were returned from said translation core. Upon multiple frames being + returned the first is passed on, but sebsequently "chained" frames are put + back into the read queue. Thus it was possible for a frame to go back into + the jitter buffer where this would cause problems. + + This patch adds a flag to frames that are inserted into the channel's read + queue after translation. The abstract jitter buffer code then checks for this + flag and ignores any frames marked as such. + + Change-Id: I276c44edc9dcff61e606242f71274265c7779587 + +2017-01-13 21:23 +0000 [473330983b] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Change when high water warning logged. + + The task processor queue reached X scheduled tasks message was originally + intended to get logged only once per task processor to prevent spamming + the log. This is no longer necessary since high and low water thresholds + can better control when the message is logged. + + It is beneficial to generate the warning each time a task processor + reaches the high water level because PJSIP stops processing new requests + while any high water alert is active. Without this change you would have + to enable at least debug level 3 logging to know about a repeated alert + trigger. + + * Made generate the warning message whenever a task is pushed into the + task processor that triggers the high water alert. + + * Appended 'again' to the warning for a repeated high water alert trigger. + + Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999 + +2017-01-10 05:54 +0000 [0047b1bc49] Aaron An <anjb@ti-net.com.cn> + + * res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt) + + Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter) + always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and + "AST_RTP_STAT_STRCPY". + It should compare "combined" with "stat" not "current_stat". + + ASTERISK-26710 #close + Reported-by: Aaron An + Tested-by: AaronAn + + Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15 + +2017-01-10 18:10 +0000 [47474cfd54] gtjoseph <gjoseph@digium.com> + + * debug_utilities: Create the ast_coredumper utility + + This utility allows easy manipulation of asterisk coredumps. + + * Configurable search paths and patterns for existing coredumps + * Can generate a consistent coredump from the running instance + * Can dump the lock_infos table from a coredump + * Dumps backtraces to separate files... + - thread apply 1 bt full -> <coredump>.thread1.txt + - thread apply all bt -> <coredump>.brief.txt + - thread apply all bt full -> <coredump>.full.txt + - lock_infos table -> <coredump>.locks.txt + * Can tarball corefiles and optionally delete them after processing + * Can tarball results files and optionally delete them after processing + * Converts ':' in coredump and results file names '-' to facilitate + uploading. Jira for instance, won't accept file names with colons + in them. + + Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. + + [1] For *BSDs, the "devel/gdb" package might have to be installed to + get a recent gdb. The utility will check all instances of gdb + it finds in $PATH and if one isn't found that can run python, it + prints a friendly error. + + Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd + +2017-01-08 10:29 +0000 [f8cd73ec3c] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fix compilation with MALLOC_DEBUG + + When MALLOC_DEBUG was specified, make was failing. Immediately + remaking would work. The issues was in the ordering of the make + dependencies. + + Change-Id: If6030b54fc693f3179f32bfd20c6b5d5f1b3f7cd + +2017-01-05 06:11 +0000 [37aaaa2da2] Joshua Colp <jcolp@digium.com> + + * res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. + + This change implements SRV support for the IP based endpoint + identifier module. All possible addresses through SRV are looked + up and added as matches. If no SRV records are available a + fallback to normal host resolution is done. If an IP address + is provided then no SRV lookup occurs. + + This is configured using the "srv_lookups" option on the + identify section and defaults to "yes". + + ASTERISK-26693 + + Change-Id: I6b641e275bf96629320efa8b479737062aed82ac + +2016-12-22 09:13 +0000 [569dac8e50] Alexander Traud <pabstraud@compuserve.com> + + * res_pjsip_session: Access SIPDOMAIN via Dialplan. + + This feature was available in the SIP channel driver chan_sip. For example, + Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not + local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect + and dial remote SIP-URIs. This change here sets the SIP destination domain of + an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well. + + ASTERISK-26670 #close + + Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243 + +2017-01-04 05:50 +0000 [367128e70b] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND. + + After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats + but remember the joint format. Cached formats contain default parameters, + often create an empty fmtp line. However, a joint format might have passed + format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and + contain the resulting format parameters from a SDP negotiation. + + ASTERISK-26691 #close + + Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc + +2017-01-03 15:14 +0000 [d7e5a747c3] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Compile pjsua with max log level = 2 + + A while back, we changed config_site.h to set PJ_LOG_MAX_LEVEL = 6. + This allowed us to control the log level better from inside Asterisk. + An unfortunate side effect of this was that the pjsua binary and + python bindings were also compiled with log level set to 6 so whenever + a testsuite test that uses pjsua runs, it spits out 6795 lines of + debug in an instant even before the test starts. I believe this + overruns the Jenkins capture buffer and prevents the test from + properly terminating. In turn, this results in the testsuite just + hanging until the job is killed. It's more frequent on the higher + end agents because they can spit out the messages faster. + + Unfortunately, the messages are all spit out before we have control + of the python pj.Lib instance where we can set logging levels so the + only alternative was to actually compile pjsua and _pjsua.so with an + overridden PJ_LOG_MAX_LEVEL. Although defining a lower max level was + done in the Makefile, the define in config_site.h had to be wrapped + with "#ifndef" so the change would take effect. + + Change-Id: I2af9e7d48dde1927279c586c9c725d868fe6f3ff + +2016-12-22 16:00 +0000 [34e728cfb9] Joshua Colp <jcolp@digium.com> + + * chan_pjsip: Use session for retrieving CHANNEL() information. + + The CHANNEL() dialplan function implementation for PJSIP allows + querying of PJSIP specific information. This used the channel + passed in to get the PJSIP session and associated information. + It is possible for this channel to be masqueraded and end + up as a different channel type by the time the information + request is actually acted upon. + + This change retrieves the PJSIP session safely and accesses + data from it (including channel). This provides a guarantee + that the session and channel will not be altered when the + request is being acted upon. + + ASTERISK-26673 + + Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6 + +2016-12-31 19:56 +0000 [a398f98b08] Joshua Elson <joshelson@gmail.com> + + * res_pjsip: Fix known compact header issues + + ASTERISK-26684 #close + + Change-Id: Ifd7e401c45015119dd5e8421dbfe3afa6381744a + +2016-12-30 09:10 +0000 [0ab9d103f6] JoshE (license 6075) + + * res_pjsip_refer: Handle compact Refer-To header. + + refer_incoming_refer_request needed to look for the "r" header as well + as the "Refer-To" header. + + ASTERISK-26655 #close + patches: + refer_compact_fix.diff submitted by JoshE (license 6075) + + Change-Id: I610410a99b02427ea5db887aeb454d5f12c2259f + +2016-12-23 12:11 +0000 [21151408f7] Richard Mudgett <rmudgett@digium.com> + + * bridge_native_rtp.c: Minor code cleanups. + + In native_rtp_bridge_compatible_check() + + * Made one variable declaration per line. + + * Extracted if test assignment to make the test easier to see. + + * Made long if tests easier to see the combinatorial logic. + + * Added bridge id to a couple debug messages. + + Change-Id: I65bc5732aa7c9a2537f062f106fbea711cf2daad + +2016-12-23 12:10 +0000 [9dcf9e9cea] Richard Mudgett <rmudgett@digium.com> + + * bridge_native_rtp.c: Fix native rtp bridge data race. + + native_rtp_bridge_compatible() didn't lock the bridge channels before + checking the channels for native bridging ability. As a result, one of + the channel's native format capabilities structure got replaced out from + under the native bridge check. Use of a stale pointer to freed memory + causes bad things to happen. + + MALLOC_DEBUG, DO_CRASH, and the + tests/channels/pjsip/transfers/blind_transfer/caller_direct_media + testsuite test caught this. + + * Add missing channel locking in native_rtp_bridge_compatible(). + + Change-Id: If25fdb3ac8e85563c4857fb8216b3d9dc3d0fa53 + +2016-12-21 16:28 +0000 [a9e459f8ac] Richard Mudgett <rmudgett@digium.com> + + * res_rtp_asterisk.c: Fix uninitialized memory crash. + + ast_rtp_remote_address_set() could pass an uninitialized 'us' parameter to + ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' + parameter may not get initialized. Thus when the code tries to save the + 'us' parameter to the local address we could try to copy a ridiculous + sized memory buffer and segfault. + + * Made pass an initialized 'us' parameter to ast_ouraddrfor(). + + * Optimized out the 'us' struct variable. + + ASTERISK-26672 #close + + Change-Id: I4acea5dcdf0813da2c7d3e11c2d6067d160d17dc + +2016-12-21 17:55 +0000 [bcdd282ada] Richard Mudgett <rmudgett@digium.com> + + * res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip(). + + We access uninitialized memory when the 'ourip' parameter does not + have an initial guess to our IP address. + + ASTERISK-26672 + + Change-Id: I35507ea1ad7455d2be188f6ccdd4add7bd150e15 + +2016-12-21 16:25 +0000 [ac31233dbe] Richard Mudgett <rmudgett@digium.com> + + * acl.c: Improve ast_ouraddrfor() diagnostic messages. + + * Made not generate strings unless they will actually be used. + + ASTERISK-26672 + + Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3 + +2016-12-21 17:54 +0000 [0aa5db4b38] Richard Mudgett <rmudgett@digium.com> + + * chan_rtp.c: Fix uninitialized memory crash. + + unicast_rtp_request() could pass an uninitialized 'us' parameter to + ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' + parameter may not get initialized. Thus when the code tries to save the + 'us' parameter to the local address we could try to copy a ridiculous + sized memory buffer and segfault. + + * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort + the UnicastRTP channel request if it fails. + + ASTERISK-26672 + + Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0 + +2016-12-07 15:23 +0000 [e2fa3c7eda] Richard Mudgett <rmudgett@digium.com> + + * res_rtp_asterisk.c: Fix off nominal memory leak. + + Change-Id: I95b1088d11244a2edae6607c12fbf33b38658a75 + +2016-12-09 12:23 +0000 [d13be4eff6] Martin Tomec <tomec@ipex.cz> + + * app_queue: Ensure member is removed from pending when hanging up. + + In some cases member is added to pending_members, and the channel + is hung up before any extension state change. So the member would + stay in pending_members forever. So when we call do_hang, we + should also remove member from pending. + + ASTERISK-26621 #close + + Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54 + +2016-12-18 15:23 +0000 [815f755155] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Make build single threaded + + There were just too many issues in various environments with + multi threaded building of pjproject. It doesn't really speed + things up anyway since asterisk is already being compiled in + parallel. + + Change-Id: Ie5648fb91bb89b4224b6bf43a0daa1af793c4ce1 + +2016-12-08 20:00 +0000 [493849dcd7] Corey Farrell <git@cfware.com> + + * chan_sip: Reorder unload_module to deal with stuck TCP threads. + + In some situations TCP threads may become frozen. This creates the + possibility that Asterisk could segfault if they become unfrozen after + chan_sip has been dlclose'd. This reorders the unload_module process to + allow abort if threads do not exit within 5 seconds. + + High level order as follows: + 1) Unregister from the core to stop new requests. + 2) Signal threads to stop + 3) Clear config based tables (but do not free the table itself). + 4) Verify that threads have shutdown, cancel unload if not. + 5) Clean all remaining resources. + + ASTERISK-26586 + + Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882 + +2016-12-16 01:32 +0000 [ab447f8a6a] David M. Lee <dlee@respoke.io> + + * configure: fix with-pjproject-bundled + + The AC_ARG_WITH macro's shell variable is withval; not enableval. Purely + coincidentally, the option would work when --enable-dev-mode is given. + + Also fixed a portability problem with bootstrap.sh, since -printf is not + a portable option for find. + + Change-Id: I0f0e5b1a934b5af5737713834361e9c95b96b376 + +2016-12-15 13:25 +0000 [35736d419a] Richard Mudgett <rmudgett@digium.com> + + * autosupport: Add 'pjproject show buildopts' + + Change-Id: I8aa55a7c3fb175235ddc7f85e9457d5102d06fa7 + +2016-12-14 14:21 +0000 [4b285d226d] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi.c: Fix bounds check regression. + + Caused by ASTERISK-25494 + + Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb + +2016-12-13 14:34 +0000 [9114574188] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Add/update ERROR msg if invalid URI. + + ASTERISK-24499 + + Change-Id: Ie305153e47e922233b2ff24715e0e326e5fa3a6c + +2016-12-12 18:38 +0000 [75a6afbec5] Richard Mudgett <rmudgett@digium.com> + + * MESSAGE: Flush Message/ast_msg_queue channel alert pipe. + + ASTERISK-25083 + + Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2 + +2016-12-13 14:06 +0000 [91485734a4] gtjoseph <gjoseph@digium.com> + + * res_sorcery_memory_cache: Change an error to a debug message + + When a sorcery user calls ast_sorcery_delete on an object that + may have already expired from the cache, res_sorcery_memory_cache + spits out an ERROR. Since this can happen frequently and validly when + an inbound registration expires after the cache entry expired, the + errors are unnecessary and misleading. Changed to a debug/1. + + Change-Id: Idf3a67038c16e3da814cf612ff4d6d18ad29ecd7 + +2016-12-09 08:14 +0000 [cd46e86491] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Retry download if previously saved tarball is bad + + If a tarball is corrupted during download, the makefile will attempt to + download it again. If the tarball somehow gets corrupted after it's + downloaded however, the makefile was just failing. We now + retry the download. + + ASTERISK-26653 #close + + Change-Id: I1b24d454852d80186f60c5a65dc4624ea8a1c359 + +2016-12-08 12:54 +0000 [22820e10fe] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * chan_sip: Delete unneeded check + + P is always true. We check it before + + Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb + +2016-12-08 12:58 +0000 [6aa2c5e5f9] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * Small code cleanup in chan_sip + + The conditional expressions of the 'if' operators situated + alongside each other are identical. + + Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a + +2016-12-08 12:43 +0000 [b596fac838] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * Fix typo in chan_sip + + The conditional expressions of the 'if' operators + situated alongside each other are identical. + + Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb + +2016-12-08 12:30 +0000 [483ed9f1aa] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * res_pjsip: Fix 'A = B != C' kind. + + Consider reviewing the expression of the 'A = B != C' kind. + The expression is calculated as following: 'A = (B != C)' + + Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d + +2016-11-30 09:31 +0000 [41c6319c4e] Walter Doekes <walter+asterisk@wjd.nu> + + * chan_sip: Do not allow non-SP/HTAB between header key and colon. + + RFC says SIP headers look like: + + HCOLON = *( SP / HTAB ) ":" SWS + SWS = [LWS] ; sep whitespace + LWS = [*WSP CRLF] 1*WSP ; linear whitespace + WSP = SP / HTAB ; from rfc2234 + + chan_sip implemented this: + + HCOLON = *( LOWCTL / SP ) ":" SWS + LOWCTL = %x00-1F ; CTL without DEL + + This discrepancy meant that SIP proxies in front of Asterisk with + chan_sip could pass on unknown headers with \x00-\x1F in them, which + would be treated by Asterisk as a different (known) header. For + example, the "To\x01:" header would gladly be forwarded by some proxies + as irrelevant, but chan_sip would treat it as the relevant "To:" header. + + Those relying on a SIP proxy to scrub certain headers could mistakenly + get unexpected and unvalidated data fed to Asterisk. + + This change fixes so chan_sip only considers SP/HTAB as valid tokens + before the colon, making it agree on the headers with other speakers of + SIP. + + ASTERISK-26433 #close + AST-2016-009 + + Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b + +2016-11-14 18:18 +0000 [888142e891] Joshua Colp <jcolp@digium.com> + + * res_format_attr_opus: Fix crash when fmtp contains spaces. + + When an opus offer or answer was received that contained an + fmtp line with spaces between the attributes the module would + fail to properly parse it and crash due to recursion. + + This change makes the module handle the space properly and + also removes the recursion requirement. + + ASTERISK-26579 + + Change-Id: I01f53e5d9fa9f1925a7365f8d25071b5b3ac2dc3 + +2016-12-06 14:54 +0000 [ebc67d3053] gtjoseph <gjoseph@digium.com> + + * res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command + + The PJSIPShowRegistrationsInbound AMI command was just dumping out + all AORs which was pretty useless and resource heavy since it had + to get all endpoints, then all aors for each endpoint, then all + contacts for each aor. + + PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail + events which meets the intended purpose of the other command and has + significantly less overhead. Also, some additional fields that were + added to Contact since the original creation of the ContactStatusDetail + event have been added to the end of the event. + + For compatibility purposes, PJSIPShowRegistrationsInbound is left + intact. + + ASTERISK-26644 #close + + Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a + +2016-12-06 16:45 +0000 [d506874477] Richard Mudgett <rmudgett@digium.com> + + * Bundled pjproject: Fix finding SIP transactions. + + Occasionally SIP message transactions are not found when they should be. + In the particular case an incoming INVITE transaction is CANCELed but the + INVITE transaction cannot be found so a 481 response is returned for the + CANCEL. The problematic calls have a '_' character in the Via branch + parameter. + + The problem is in the pjproject PJ_HASH_USE_OWN_TOLOWER feature's code. + The problem with the "own tolower" code is that it does not calculate the + same hash value as when the pj_tolower() function is used. The "own + tolower" code will erroneously modify the ASCII characters '@', '[', '\\', + ']', '^', and '_'. Calls to pj_hash_calc_tolower() can use the + PJ_HASH_USE_OWN_TOLOWER substitute algorithm when enabled. Calls to + pj_hash_get_lower(), pj_hash_set_lower(), and pj_hash_set_np_lower() call + find_entry() which never uses the PJ_HASH_USE_OWN_TOLOWER algorithm. As a + result you may not be able to find a hash tabled entry because the + calculated hash values would differ. + + * Simply disable PJ_HASH_USE_OWN_TOLOWER. + + ASTERISK-26490 #close + + Change-Id: If89bfdb5f301b8b685881a9a2a6e0c3c5af32253 + +2016-12-06 12:06 +0000 [4b233675d8] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fix missing inclusion of symbols + + Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to + the CFLAGS. Not sure how they went missing. + + Also fixed an uninstall problem where we weren't removing the + symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was + there, I fixed it for libasteriskssl as well. + + Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556 + +2016-12-02 12:04 +0000 [580f83dac7] Richard Mudgett <rmudgett@digium.com> + + * Remove files that got merged in error somehow to the 13 branch. + + Change-Id: Id79e2226c31084f9252d5aede9050d3cf13322c8 + +2016-11-30 18:25 +0000 [61ba2a014a] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Filter redundant statsd reporting. + + Increasing the testsuite shutdown timeout before forcibly killing + Asterisk allowed more events to be sent out. Some tests failed as + a result. The tests/channels/pjsip/statsd/registrations failed + because we now get the statsd events that a comment in the test + configuration stated couldn't be intercepted. Unfortunately, we + get a variable number of events because of internal status state + transition races generating redundant statsd events. + + We were reporting redundant statsd PJSIP.registrations.state changes + for internal state changes that equated to the same thing publicly. + + * Made update_client_state_status() filter out redundant statsd + updates. + + ASTERISK-26527 + + Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646 + +2016-11-22 11:20 +0000 [2ceb609edb] Guido Falsi <mad@madpilot.net> + + * res_rtp: Fix regression when IPv6 is not available. + + The latest Release candidate fails to create RTP streams when IPv6 + is not available. Due to the changes made in September the ast_sockaddr + structure passed around to create these streams is always of AF_INET6 + type, causing failure when used for IPv4. This patch adds a utility + function to check for availability of IPv6 and applies such check + at startup to determine how to create the ast_sockaddr structures. + + ASTERISK-26617 #close + + Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e + +2016-11-28 19:43 +0000 [53459cdaa9] Eduardo S. Libardi <eslibardi@gmail.com> + + * res_calendar_caldav: Add support reading gmail calendar + + The response from gmail calendar includes the string name + "caldav:calendar-data". res_calendar_caldav implements + the example included in RFC 4791: string "C:calendar-data". + When reading the calendar, res_calendar_caldav compare the + string and if does not match just discards the event. + This commit compares the response to both strings, + successfully loading gmail calendar events. + Writing to gmail calendar is working prior to this fix. + + ASTERISK-26624 + Reported by: Eduardo S. Libardi + + Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a + +2016-11-23 18:27 +0000 [44fe4a5769] Richard Mudgett <rmudgett@digium.com> + + * PJPROJECT logging: Made easier to get available logging levels. + + Use of the new logging is as simple as issuing the new CLI command or + setting the new pjproject.conf option. + + Other options that can affect the logging are how you have the pjproject + log levels mapped to Asterisk log types in pjproject.conf and if you have + configured Asterisk to log the DEBUG type messages. Altering the + pjproject.conf level mapping shouldn't be necessary for most installations + as the default mapping is sensible. Configuring Asterisk to log the DEBUG + message type is standard practice for collecting debug information. + + * Added CLI "pjproject set log level" command to dynamically adjust the + maximum pjproject log message level. + + * Added CLI "pjproject show log level" command to see the currently set + maximum pjproject log message level. + + * Added pjproject.conf startup section "log_level" option to set the + initial maximum pjproject log message level so all messages could be + captured from initialization. + + * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into + bundled pjproject. Pjproject will use the currently set run time log + level to determine if a log message is generated just like Asterisk + verbose and debug logging levels. + + * In log_forwarder(), made always log enabled and mapped pjproject log + messages. DEBUG mapped log messages are no longer gated by the current + Asterisk debug logging level. + + * Removed RAII_VAR() from res_pjproject.c:get_log_level(). + + ASTERISK-26630 #close + + Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389 + +2016-11-30 10:48 +0000 [17b0b91afa] Mark Michelson <mmichelson@digium.com> + + * Frame deferral: Re-queue deferred frames one-at-a-time. + + The recent change that made frame deferral into an API had a behavior + change to it. When frame deferral was completed, we would take all of + the deferred frames and queue them all onto the channel in one call to + ast_queue_frame_head(). Before frame deferral was API-ized, places that + performed manual frame deferral would actually take each deferred frame + and queue them onto the channel. + + This change in behavior caused the confbridge_recording test to start + failing consistently. Without going too crazily deep into the details, + a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect + was attempting to break it out of the sleep, but because there were more + frames in the channel read queue than expected, the channel ended up + being unable to break from its sleep loop. + + By restoring the behavior of individual frame queuing after deferral, + the test starts passing again. + + Note, this points to a potential underlying issue pointing to an + "unbalance" that can occur when queuing multiple frames at once, + and so a follow-up issue is being created to investigate that + possibility. + + Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d + +2016-06-28 16:26 +0000 [b0c9f07f04] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * OpenSSL 1.1.0 support + + OpenSSL 1.1.0 includes some major changes in the interface. See + https://wiki.openssl.org/index.php/1.1_API_Changes . + + Status: Right now there are still a few deprecation notes with OpenSSL + 1.1.0. But it's a start. + + Changes: + * CRYPTO_LOCK is no longer available. Replace it with its value for now. + I don't completely understand what it is used for there. + * Remove several functions from libasteriskssl that seem to no longer be + needed. + * Structures have become opaque and are accesses with accessors. + * ERR_remove_thread_state() no longer needed. + * SSLv2 code now could no longer be used in 1.1. + + ASTERISK-26109 #close + + Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b + +2016-11-28 15:12 +0000 [a33ed3327a] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip: Fix documentation whitespace issues + + Tabs > Spaces. + + Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0 + +2016-11-22 10:27 +0000 [09c36a6535] Matt Jordan <mjordan@digium.com> + + * res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter + + Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise + 'ws' when WebSockets are to be used as the transport. This applies to + both secure and insecure WebSockets. + + There were two bugs in Asterisk with respect to this: + + (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for + insecure websockets and 'wss' for secure websockets. While this + would seem to make sense - since 'WS' and 'WSS' are used for the Via + Transport parameter - this is not the case for the SIP URI. This + patch corrects that by registering the secure websockets with + pjproject using the shorthand 'WS', and by returning 'ws' when asked + for the transport parameter. Note that in pjproject, it is perfectly + valid to have multiple transports use the same shorthand. + + (2) In chan_sip, we return an upper-case version of the transport 'WS' + instead of 'ws'. Since we should be strict in what we send and + liberal in what we accept (within reason), this patch lower-cases + the transport before appending it to the parameter. + + ASTERISK-24330 #close + Reported by: cervajs, Inaki Baz Castillo + + Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42 + +2016-11-28 11:03 +0000 [29e887e9e1] gtjoseph <gjoseph@digium.com> + + * build_tools: Fix download_externals to handle certified branches + + download_externals wasn't handling the "certified/13.x" version + correctly. + + Change-Id: I124d195bb117ca36fd7bf1150c630f3b474a9d9a + +2016-11-02 05:05 +0000 [bfb8c962c4] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * autoconf: more variants for OSARCH linux-gnu + + There are quite a few odd GNU/Linux platforms. Just call all of them + linux-gnu. + + Specifically this fixes building the Debian platforms mips64el and x32. + And maybe also others. + + ASTERISK-26546 #close + + Change-Id: I06ec4bd7f0ee1c84b6b24d81538223b07c4174b1 + +2016-11-17 08:25 +0000 [a1fa909033] Timo Teräs <timo.teras@iki.fi> + + * codec_dahdi: Fix poll.h include. + + POSIX defines poll.h. sys/poll.h should not be used as it is c-library + internal header which may or may not exist. Notably in musl including + sys/poll.h generates warning of being incorrect. + + Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252 + +2016-11-26 10:57 +0000 [0cc8351484] Michael Kuron <m.kuron@gmx.de> + + * chan_sip: Fix segfault during module unload + + If a TCP/TLS connection was pending (not accepted and not timed out) during + unload of chan_sip, Asterisk would segfault when trying to send a signal to + a thread whose thread ID hadn't been recorded yet. This commit fixes that by + recording the thread ID before calling the blocking connect() syscall. + This was a regression introduced by 776a14386a55b5425c7e9617eff8af8b45427144. + + The above wasn't enough to fix the segfault, which was now delayed to the + point where connect() timed out. Therefore, it was necessary to also remove + the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be + used to interruput the connect() syscall. + This was a regression introduced by 5d313f51b982a18f7321adcf7c7a4e822d8b2714. + + ASTERISK-26586 #close + + Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b + +2016-11-11 08:16 +0000 [8756ce64b7] gestoip2 <gestoip2@ull.edu.es> + + * res_rtp_asterisk: RTT miscalculation in RTCP + + When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. + RTT calculation is correct, but the data representation isn't. RTT is + represented by a 32-bit fixed-point number with the integer part in the + first 16 bits and the fractional part in the last 16 bits. In order to + get the RTT value, the fractional part is miscalculated, there is an + unnecessary 16 bit shift that causes overflow. Besides this there is + another mistake, when transforming the integer value to the fixed point + fractional part via bitwise operation, that loses precision. + + * RTT fractional part is no longer shifted, avoiding overflow. + + * RTT fractional part is transformed to its fixed-point value more + precisely. + + * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. + + * Fixed NTP timestamp report logging. The usec was inexplicably + multiplied by 4096. + + ASTERISK-26566 #close + Reported by Hector Royo Concepcion + + Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f + +2016-11-15 13:44 +0000 [8e77d6f520] Michael Kuron <m.kuron@gmx.de> + + * tcptls: Use new certificate upon sip reload + + Previously, a TLS server socket would only be restarted upon sip reload if the + bind address had changed. This commit adds checking for changes to TLS + parameters like certificate, ciphers, etc. so they get picked up without + requiring a reload of the entire chan_sip module. This does not affect open + connections in any way, but new connections will use the new TLS parameters. + The changes also apply to HTTP and Manager. + + ASTERISK-26604 #close + + Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6 +2016-11-11 00:29 +0000 [86d824b7ff] Timo Teräs <timo.teras@iki.fi> + + * addons/chan_mobile: do not use strerror_r + + The two reasons why it might be used are that some systems do not + implement strerror in thread safe manner, and that strerror_r returns + the error code in the string in case there's no error message. + + However, all of asterisk elsewhere uses strerror() and assumes it + to be thread safe. And in chan_mobile the errno is also explicitly + printed so neither of the above reasons are valid. + + The reasoning to remove usage is that there are actually two versions + of strerror_r: XSI and GNU. They are incompatible in their return + value, and there's no easy way to figure out which one is being + used. glibc gives you the GNU version if _GNU_SOURCE is defined, + but the same feature test macro is needed for other symbols. On + all other systems you assumedly get XSI symbol, and compilation warnings + as well as non-working error printing. + + Thus the easiest solution is to just remove strerror_r and use + strerror as rest of the code. Alternative is to introduce ast_strerror + in separate translation unit so it can request the XSI symbol in + glibc case, and replace all usage of strerror. + + Change-Id: I84d35225b5642d85d48bc35fdf399afbae28a91d + +2016-11-21 09:40 +0000 [425da14927] gtjoseph <gjoseph@digium.com> + + * build: Backport addition of librt check to configure.ac + + A while back, a master-only change was made to check for librt which + should probably have been cherry-picked to 13 at that time. Sometime + between then and now, part of that change did make it into 13 but it + was incomplete and non-functional. This patch backports the rest + of the librt check and allows the link of libasteriskpj to use the + results. + + Change-Id: I1424008fd8c90f389dda53162ec4a340b253a3c1 + +2016-11-16 12:05 +0000 [2a40c3a867] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Improve reliability of pjproject download + + The download process now has a timeout which will cause wget to retry + if it stops retrieving data for 5 seconds and fetch and curl to timeout + if the whole retrieval take smore than 30 seconds. + + If the tarball retrieval works, the MD5SUM file is retrieved from + the downloads site and the md5 checksum is verified. + + If either the tarball retrieval or MD5SUM retrieval fails, or the + checksums don't match, the entire process is retried once. If it + fails again, any incomplete tarball is deleted. + + .DELETE_ON_ERROR: was also added to the Makefile. Not only does + this delete the tarball on failure, it till also delete corrupted + library files from the pjproject source directory should they + fail to build correctly. + + Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and + Ubuntu 14. + + Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1 + +2016-11-11 07:13 +0000 [12c4e664bc] Mikheili Dautashvili <mishadaut@gmail.com> + + * main/app.c: Transmit Silence on ControlPlayback pause + + ASTERISK-26562 #close + + Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8 + +2016-11-15 15:01 +0000 [cf6d13180e] Alexei Gradinari <alex2grad@gmail.com> + + * chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no + + The sending codec is switched to the receiving codec and then + is switched back to the best native codec on EVERY receiving RTP packets. + This is because after call of ast_channel_set_rawwriteformat there is call + of ast_set_write_format which calls set_format which sets rawwriteformat + to the best native format. + + This patch adds a new function ast_set_write_format_path which set + specific write path on channel and uses this function to switch + the sending codec. + + ASTERISK-26603 #close + + Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d + +2016-11-10 13:34 +0000 [ee73af1d88] gtjoseph <gjoseph@digium.com> + + * Update for 13.12.2 + +2016-11-04 10:57 +0000 [a3614d75f6] Kevin Harwell <kharwell@digium.com> + + * Revert "chan_sip: Fix lastrtprx always updated" + + This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc. + + Unfortunately, the aforementioned commit caused a regression (incoming calls + would eventually disconnect). Thus it is being removed. + + ASTERISK-26523 #close + ASTERISK-25270 + + Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d + +2016-10-27 13:48 +0000 [7d7b52c434] Mark Michelson <mmichelson@digium.com> + + * Update for 13.12.1 + +2016-10-26 07:51 +0000 [9c761b8f45] Joshua Colp <jcolp@digium.com> + + * app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. + + When executing the MailboxExists dialplan application and + MAILBOX_EXISTS dialplan function the passed in temporary voice + mailbox was not cleared, causing it to try to free garbage. + + ASTERISK-26503 #close + + Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3 + +2016-10-25 14:13 +0000 [226a7e36c5] Mark Michelson <mmichelson@digium.com> + + * Update for 13.12.0 + +2016-10-17 14:08 +0000 [df75b647da] Mark Michelson <mmichelson@digium.com> + + * Update for 13.12.0-rc1 + +2016-11-18 18:59 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.13.0-rc1 Released. + +2016-11-18 09:45 +0000 [cb624b10ae] Mark Michelson <mmichelson@digium.com> + + * Bump ARI version to 1.10.0 + + The video-related bridge changes mean that the version needs to be + bumped. + + Change-Id: I41c4495068562bef03aa76728f188b8ac4bd393d + +2016-11-17 10:50 +0000 [bde3d022a3] Mark Michelson <mmichelson@digium.com> + + * manager: update minor version + + Based on bridge video AMI event changes, bump the minor version of AMI. + + Change-Id: I02586bd6cafc0baa33ea98c2f75356c0f5e03435 + +2016-11-16 20:24 +0000 [b213045fe4] gtjoseph <gjoseph@digium.com> + + * build: Various OpenBSD issues + + OpenBSD's 'find' doesn't take the -delete argument so you have to pipe + through 'xargs rm -rf'. + + 'echo -e' doesn't like \t starting a line. It just prints 't' which + causes the libasteriskpj.exports file to be garbage. They were just + cosmetic so they were removed. + + librt doesn't exist so the link of libasteriskpj.so fails. It's not + actually needed for linux anyway so -lrt was removed from the link. + + res_rtp_asterisk was failing to load because of an undefined + DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if + so DTLSv1_method is used instead. + + ASTERISK-26608 + + Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c + +2016-11-14 18:45 +0000 [404596b790] gtjoseph <gjoseph@digium.com> + + * channel: Fix issues in hangup scenarios caused by frame deferral + + ASTERISK-26343 + + Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 + +2016-11-16 15:42 +0000 [2c031b67d3] Mark Michelson <mmichelson@digium.com> + + * res_format_attr_opus: Fix fmtp generation. + + res_format_attr_opus assumed that the string being passed into it was + empty. It tried to determine if the only thing it had written was + + a=fmtp:<num> + + And if it had, it would reset the string. Its calculation was off when + working with chan_sip, though. chan_sip passes the entire built SDP + rather than an empty string. This resulted in always putting an empty + fmtp line in the SDP. + + ASTERISK-26520 #close + Reported by scgm11 + + Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5 + +2016-11-15 16:23 +0000 [ed0f1afc8c] Richard Mudgett <rmudgett@digium.com> + + * codec_opus: Fix warning when Opus negotiated but codec_opus not loaded. + + When Opus is negotiated but not loaded, the log is spammed with messages + because the system does not know how to calculate the number of samples in + a frame. + + * Suppress the warning by supplying a function that assumes 20ms of + samples in the frame. For pass through support it doesn't really seem to + matter what number of samples is returned anyway. + + ASTERISK-26605 #close + + Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f + +2016-11-14 14:36 +0000 [e632222bc4] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak. + + Responding to authentication challenges leaks PJSIP memory pools. + + The leak was introduced with a pjproject 2.5.5 API change. + https://trac.pjsip.org/repos/ticket/1929 changed the API usage of + pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to + clean up cached authentication allocations that get allocated with + pjsip_auth_clt_reinit_req(). + + ASTERISK-26516 #close + + Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8 + +2016-11-15 12:01 +0000 [c92dcc76da] gtjoseph <gjoseph@digium.com> + + * file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type + + One of the code paths in __ast_file_read_dirs will only get executed if + the OS doesn't support dirent->d_type OR if the filesystem the + particular file is on doesn't support it. So, while standard Linux + systems support the field, some filesystems like XFS do not. In this + case, we need to call stat() to determine whether the directory entry + is a file or directory so we append the filename to the supplied + directory path and call stat. We forgot to truncate path back to just + the directory afterwards though so we were passing a complete file name + to the callback in the dir_name parameter instead of just the directory + name. + + The logic has been re-written to only create a full_path if we need to + call stat() or if we need to descend into another directory. + + Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba + +2015-05-14 17:12 +0000 [7b96e8cc3d] Maciej Szmigiero <mail@maciej.szmigiero.name> + + * Add X.509 subject alternative name support to TLS certificate + verification. + + This way one X.509 certificate can be used for hosts that + can be reached under multiple DNS names or for multiple hosts. + + Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name> + + ASTERISK-25063 #close + + Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f + +2016-11-14 15:57 +0000 [0790aa528a] Matt Jordan <mjordan@digium.com> + + * pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS + + The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how + many pairs of local/remote candidates will be made. If for some reason + we reach this upper bound, ICE will generally fail and no media will + flow between the browser and Asterisk. + + This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of + pairs of candidates we'd theoretically allow, which is + PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied + PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame + Docker), this is far too low to allow WebRTC calls to succeed. + + Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed + even when the system Asterisk was running on had quite a few virtual + interfaces. + + Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55 + +2016-11-14 15:32 +0000 [993a6f96c7] Matt Jordan <mjordan@digium.com> + + * apps/app_echo: Only relay a single video source change frame + + In 9785e8d0, app_echo was updated to relay video source updates to the + channel for the purposes of displaying video in WebRTC tests. + Unfortunately, this can cause a Kafkaesque nightmare if two or more + Local channels are in a bridge together where their ends are in + app_echo. When this situation occurs, a video update sent into app_echo + will cause the video update to be relayed to the other Local channels, + causing another round of video updates, etc. In not much time at all, + the channel length queues will be overwhelmed, channel alert pipes will + fail, and all hell will break loose as Asterisk merrily continues to + throw more video update requests onto the channels. + + This patch updates app_echo to *only* relay a single video update. Once + a video update has been made, all further video updates are dropped. + This meets the intended purpose of the original patch: if we get a video + update and we're in app_echo, go ahead and ask the sender to update + themselves. However, once we've got that video stream sync'd up, don't + keep spamming the world. + + Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74 + +2016-11-08 10:11 +0000 [d23b4af477] Matt Jordan <mjordan@digium.com> + + * res/ari/resource_bridges: Add the ability to manipulate the video source + + In multi-party bridges, Asterisk currently supports two video modes: + * Follow the talker, in which the speaker with the most energy is shown + to all participants but the speaker, and the speaker sees the + previous video source + * Explicitly set video sources, in which all participants see a locked + video source + + Prior to this patch, ARI had no ability to manipulate the video source. + This isn't important for two-party bridges, in which Asterisk merely + relays the video between the participants. However, in a multi-party + bridge, it can be advantageous to allow an external application to + manipulate the video source. + + This patch provides two new routes to accomplish this: + (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} + Sets a video source to an explicit channel + (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource + Removes any explicit video source, and sets the video mode to talk + detection + + ASTERISK-26595 #close + + Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621 + +2016-11-14 14:22 +0000 [404a62eeee] gtjoseph <gjoseph@digium.com> + + * Revert "Revert "channel: Use frame deferral API for safe sleep."" + + This reverts commit 58c88cfbaa80cb43419cde9186d643d1c5d24baf. + + Change-Id: I72692e2b2e83ef6da9390075ff20b138b2c374b6 + +2016-11-14 14:22 +0000 [09d8febc91] gtjoseph <gjoseph@digium.com> + + * Revert "Revert "autoservice: Use frame deferral API"" + + This reverts commit 1df434e2b4bd7cc34b9b4addf405a3caa7ac16b8. + + Change-Id: Id2b8a8bccbb4bbdd82b792275d4cd6f32563e401 + +2016-11-14 14:21 +0000 [ffad2b44df] gtjoseph <gjoseph@digium.com> + + * Revert "Revert "AGI: Only defer frames when in an interception routine."" + + This reverts commit 6be5d8de0da7e804544507f70382425af9a07b3f. + + Change-Id: I4b548137f52ae0686d8f09e21496b778d1c6a797 + +2016-11-14 14:21 +0000 [2fefb6187f] gtjoseph <gjoseph@digium.com> + + * Revert "Revert "Add API for channel frame deferral."" + + This reverts commit 6b5a7ced136b7178ae0b2ba39221eba1cd2e37c9. + + Change-Id: I61d1dbb2e69e1977f684b7dfc8e98211024e1cd1 + +2016-11-14 12:16 +0000 [5e0c224043] gtjoseph <gjoseph@digium.com> + + * cli: Fix ast_el_read_char to work with libedit >= 3.1 + + Libedit 3.1 is not build with unicode on as a default and so the + prototype for the el_gets callback changed from expecting a char buffer + to accepting a wchar buffer. If ast_el_read_char isn't changed, + the cli reads garbage from teh terminal. + + Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and + updated ast_el_read_char to use the HAVE_ define to detemrine whether + to use char or wchar. + + ASTERISK-26592 #close + + Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a + +2016-11-11 02:41 +0000 [3faca1d4ff] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * Fix closing rtp ports after call finished in chan_unistim. + + Fix ASTERISK-26565 by adding ast_rtp_instance_stop before + rtp instance destroy for chan_unistim. Also several fixes + for displayed text translation. + + Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc + +2016-09-23 17:54 +0000 [412d43fa21] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip.c: Rework endpt_send_request() req_wrapper code. + + * Don't hold the req_wrapper lock too long in endpt_send_request(). We + could block the PJSIP monitor thread if the timeout timer expires. + sip_get_tpselector_from_endpoint() does a sorcery access that could take + awhile accessing a database. pjsip_endpt_send_request() might take awhile + if selecting a transport. + + * Shorten the time that the req_wrapper lock is held in the callback + functions. + + * Simplify endpt_send_request() req_wrapper->timeout code. + + * Removed some redundant req_wrapper->timeout_timer->id assignments. + + Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9 + +2016-09-21 15:10 +0000 [2e7fc56d3c] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Fix tdata leaks in off nominal paths. + + Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b + +2016-10-24 12:41 +0000 [da68b185b3] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. + + Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94 + +2016-11-10 10:57 +0000 [b70eb07c53] Joshua Colp <jcolp@digium.com> + + * res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp. + + When optimistic SRTP was on it was possible for us to still + set up a call without an audio stream if an offer was received + with required SRTP. + + This change makes it so this scenario will now fail with a 488 + response. + + ASTERISK-26575 + + Change-Id: I7d14187037681f48879bd20319ac79d0877318f3 + +2016-11-10 08:33 +0000 [71dc333565] Joshua Colp <jcolp@digium.com> + + * app_queue: Add mention of 'ABANDON' variable to CHANGES. + + ASTERISK-26558 + + Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e + +2016-11-10 07:41 +0000 [6b5a7ced13] gtjoseph <gjoseph@digium.com> + + * Revert "Add API for channel frame deferral." + + This reverts commit 9231a56cf3d6f5eca1bf2d37d827453400690773. + Multiple testsuite failures were detected after the fact. + + Change-Id: I3bac8d7c3ddb69a4ddf6c5d6de0ffa5ff7ff3af7 + +2016-11-10 07:41 +0000 [6be5d8de0d] gtjoseph <gjoseph@digium.com> + + * Revert "AGI: Only defer frames when in an interception routine." + + This reverts commit 5c10091f3d1430c6fc04015226f8c3e3aa9d8282. + Multiple testsuite failures were detected after the fact. + + Change-Id: I397a841acc17ae230c512449cd6bed89d2ef3b73 + +2016-11-10 07:41 +0000 [1df434e2b4] gtjoseph <gjoseph@digium.com> + + * Revert "autoservice: Use frame deferral API" + + This reverts commit 2e3a3545754749de21873bfdc6d1a40ec7d8893f. + Multiple testsuite failures were detected after the fact. + + Change-Id: Ia45fa4633fae74dca345b24bb6722737c63035de + +2016-11-10 07:40 +0000 [58c88cfbaa] gtjoseph <gjoseph@digium.com> + + * Revert "channel: Use frame deferral API for safe sleep." + + This reverts commit 44f7e252397fd87420b3374df26941d7436401b3. + Multiple testsuite failures were detected after the fact. + + Change-Id: I56299087da22128a95f0c8f3955f740890d7ca65 + +2016-11-09 18:18 +0000 [a562fbe618] gtjoseph <gjoseph@digium.com> + + * build: Fix default values for some SANITIZER options + + 2 of the sanitizers didn't have default values so in systems that + don't support sanitizers menuselect would spit out warnings. They + were harmless but confusing. They've now been set to "0". + + Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58 + +2016-11-06 06:04 +0000 [7fd5031c1c] Sebastian Gutierrez <sgutierrez@integraccs.com> + + * app_queue: new variable set when abandoned + + sets the variable ABANDONED to TRUE if the call was not answered. + + ASTERISK-26558 + + Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3 + +2016-11-08 10:48 +0000 [e043d1a55c] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_session: Do not call session supplements when it's too late. + + res_pjsip_sesssion was hooking into transaction and invite state + changes. One of the reasons for doing so was due to the + PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the + message sending process, and so we should call session supplements to + alter the outgoing message. + + In reality, this event was meant to indicate that the message either + a) had already been sent, or + b) required a DNS lookup and would be sent when the DNS query + completed. + + In case (a), this meant we were altering an already-sent + request/response for no reason. In case (b), this potentially meant we + could be trying to alter a request/response at the same time that the + DNS resolution completed. In this case, it meant we might be stomping on + memory being used by the thread actually sending the message. This + caused potential crashes and memory corruption. + + This patch removes the calls to session supplements from the case where + the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to + alter the message at this point is too late, and it can cause nothing + but harm to try to do it. Because there were no longer any calls to the + handle_outgoing() function, it has been removed. + + Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92 + +2016-11-03 16:46 +0000 [44f7e25239] Mark Michelson <mmichelson@digium.com> + + * channel: Use frame deferral API for safe sleep. + + This is another case where manual frame deferral can be replaced with + centralized routines instead. + + Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e + +2016-11-03 16:46 +0000 [2e3a354575] Mark Michelson <mmichelson@digium.com> + + * autoservice: Use frame deferral API + + Rather than use manual frame deferral, just let the channel API do it + for us. + + ASTERISK-26343 + + Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49 + +2016-11-03 16:42 +0000 [5c10091f3d] Mark Michelson <mmichelson@digium.com> + + * AGI: Only defer frames when in an interception routine. + + AGI recently was modified to defer important frames. This was because + when AGI was used in a connected line interception routine, the + resulting connected line frame would end up getting discarded by the + AGI. + + However, this caused bad behavior in other cases. Specifically, during a + transfer, if someone attempted to manually set the Caller ID on a + channel in an AGI, the deferred connected line frame would end up + overwriting what had been manually set in the AGI. + + Since the initial issue was specific to interception routines, this + change removes the manual frame deferral from AGI and instead uses the + new frame deferral API in interception routines. + + ASTERISK-26343 #close + Reported by Morton Tryfoss + + Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208 + +2016-11-03 16:36 +0000 [9231a56cf3] Mark Michelson <mmichelson@digium.com> + + * Add API for channel frame deferral. + + There are several places in Asterisk that have duplicated logic + for deferring important frames until later. + + This commit adds a couple of API calls to facilitate this automatically. + + ast_channel_start_defer_frames(): Future reads of deferrable frames on + this channel will be deferred until later. + + ast_channel_stop_defer_frames(): Any frames that have been deferred get + requeued onto the channel. + + ASTERISK-26343 + + Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641 + +2016-11-03 07:42 +0000 [a9ac1f5de4] Alexander Anikin <may213@yandex.ru> + + * chan_ooh323: Fixes to work right with Cisco devices + + Changed output packets queue processing algo to one read-one write + instead of all read-all send + + Remove h.245 tunneling parameter from ReleaseComplete packet + + ASTERISK-24400 #close + Reported by: Dmitry Melekhov + Tested by: Dmitry Melekhov + + Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6 + +2016-11-03 13:10 +0000 [0ee249075a] Alexander Anikin <may213@yandex.ru> + + * chan_ooh323: reset rrq count on gk registration + + reset registration attempts count on success registration on gatekeeper + + Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336 + +2016-11-06 03:46 +0000 [59c23e1768] Michael Kuron <m.kuron@gmx.de> + + * automon: restore mixing of the both channels after recording stops + + This is a regression over Asterisk 11, introduced by + 2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via + the automon DTMF code would automatically be mixed together using sox because + app_monitor would be called with the m option. This commit restores this + behavior. + + Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759 + +2016-11-04 15:42 +0000 [e79acaeb75] Matt Jordan <mjordan@digium.com> + + * res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems + + Not surprisingly, using Respoke (and possibly other systems) it is + possible to blow past the 16k limit for a WebSocket packet size. This + patch bumps it up to 32k, which, at least for Respoke, is sufficient. + For now. + + Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that + matter), this patch adds a LOW_MEMORY directive that sets the buffer to + 8k for systems who have asked for their reduced memory availability to + be considered. + + Change-Id: Id235902537091b58608196844dc4b045e383cd2e + +2016-11-04 15:40 +0000 [7a83196985] Matt Jordan <mjordan@digium.com> + + * res_stasis: Set a video source mode on Stasis created bridges + + When a bridge is created via ARI (through res_stasis), no video source + mode is set by default. As a result, any endpoint sending video media + won't ever see any video reflected back to it. + + This patch defaults a bridge to a 'follow the talker' video mode. + Further work can be done to add routes that allow for the video mode to + be controlled through the /bridges resource. + + Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866 + +2016-11-04 15:37 +0000 [e7dc536b7a] Matt Jordan <mjordan@digium.com> + + * main/bridge_channel: Fix channel reference leak on video source + + When a channel is made the video source, the bridge holds a reference to + it. Whenever the video source changes, that reference is released. + However, a ref leak does occur if the channel leaves the bridge (such as + being hung up) while it is the video source, as the bridge never + releases the ref in such a case. + + This patch adds a line to the bridge_channel_internal_join routine such + that, when a channel finishes its time in the bridge, it notifies the + bridge via ast_bridge_remove_video_src that if it is a video source its + reference should be released. + + ASTERISK-26555 #close + + Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a + +2016-11-04 15:36 +0000 [7c824b955d] Matt Jordan <mjordan@digium.com> + + * main/bridge: Add some verbose logging for video source changes + + It's actually quite useful to see the source of a video stream change. + This doesn't happen terribly often, even with talk detection - but when + it does, it's nice to know which channel is now providing your video + stream. + + As a verbose 5 level message, it shouldn't be terribly spammy or costly + to have, and is 'lower level' then most other verbose messages that the + bridge system emits. + + ASTERISK-26555 + + Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c + +2016-11-04 15:33 +0000 [fd6af2dee8] Matt Jordan <mjordan@digium.com> + + * bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source + + WebRTC clients really, really want to know the SSRC of the media they're + getting. Changing the SSRC is generally not a good thing. + + bridge_softmix, starting in Asterisk 12, started changing the SSRC of + parties as they joined or left the bridge. With most phones, this isn't + a problem: phones just play back the stream they're getting. With WebRTC + clients, however, the SSRC is tied to a media stream that may be + negotiated. When a new SSRC just shows up, the media can be dropped. + + As it turns out, the SSRC change shouldn't even be necessary. From the + perspective of the client, it's still talking to Asterisk with the same + media stream: why indicate that the far party has suddenly changed to a + different source of media? + + This patch opts to just remove the SSRC changes. With this patch, video + clients that join/leave a softmix bridge actually get the video stream + instead of freaking out. + + ASTERISK-26555 + + Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf + +2016-10-28 15:11 +0000 [bd4d7d8ad0] Kevin Harwell <kharwell@digium.com> + + * stasis_recording/stored: remove calls to deprecated readdir_r function. + + The readdir_r function has been deprecated and should no longer be used. This + patch removes the readdir_r dependency (replaced it with readdir) and also moves + the directory search code to a more centralized spot (file.c) + + Also removed a strict dependency on the dirent structure's d_type field as it + is not portable. The code now checks to see if the value is available. If so, + it tries to use it, but defaults back to using the stats function if necessary. + + Lastly, for most implementations of readdir it *should* be thread-safe to make + concurrent calls to it as long as different directory streams are specified. + glibc falls into this category. However, since it is possible that there exist + some implementations that are not safe, locking has been added for those other + than glibc. + + ASTERISK-26412 + ASTERISK-26509 #close + + Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba + +2016-11-04 10:57 +0000 [cb30963d22] Kevin Harwell <kharwell@digium.com> + + * Revert "chan_sip: Fix lastrtprx always updated" + + This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc. + + Unfortunately, the aforementioned commit caused a regression (incoming calls + would eventually disconnect). Thus it is being removed. + + ASTERISK-26523 #close + ASTERISK-25270 + + Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d + +2016-11-02 10:52 +0000 [3a1f9c5dab] Joshua Colp <jcolp@digium.com> + + * res_stasis: Don't unsubscribe from a NULL bridge. + + A NULL bridge has special meaning in res_stasis for + unsubscribing. It means that a subscription to ALL + bridges should be removed. This should not be done + as part of the normal subscription management in + the res_stasis channel loop. + + ASTERISK-26468 + + Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0 + +2016-11-03 13:45 +0000 [eceab15f33] Alexander Anikin <may213@yandex.ru> + + * chan_ooh323: Fix infinite loop on read second part of H.225 packet + + Fix logic on read second part of H.225 packet. There was infinite loop on + wrong connections due to read before poll. + + Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff + +2016-11-03 11:55 +0000 [a9992da4aa] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fix issue with libasteriskpj needing libresample + + libresample is only needed by pjproject if we're building pjsua, which + we only do if TEST_FRAMEWORK is selected. It's required by pjsua to + process audio which is needed by some testsuite tests. Unfortunately, + pjproject relies on a newer version of libresample than the version + that ships by most distros so we need to compile the version that's + bundled with pjproject. Since we only need it for pjsua, we DON'T want + it's symbols exposed when we actually build asterisk. + + There was a problem however... TEST_FRAMEWORK is only known AFTER we've + already run ./configure on both asterisk and pjproject but pjproject's + ./configure needs to test it to know whether to set up to build + libresample or not. The previous way of figuring this out was to + always tell ./configure "yes" but not actually build the library. This + caused an issue where building libasteriskpj was being told to include + libresample but it wasn't actually there. + + The solution is to still do a default pjproject configure during an + asterisk ./configure but if makeopts or menuselect.makeopts changes + subsequently, we now reconfigure pjproject, taking into account the + current state of TEST_FRAMEWORK. Previously, if makeopts or + menuselect.makeopts changed, only a recompile of pjproject was done. + + Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685 + +2016-11-01 19:48 +0000 [714412f6c4] Sebastian Gutierrez <sgutierrez@integraccs.com> + + * chan_sip: add missing account code + + Added missing account to AMI event of sip show peers + + ASTERISK-26176 #close + + Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482 + +2016-09-13 04:08 +0000 [0cf1778eed] Alexander Traud <pabstraud@compuserve.com> + + * rtp_engine: Allow more than 32 dynamic payload types. + + The dynamic range (96-127) allows 32 RTP Payload Types. RFC 3551 section 3 + allows to reassign other ranges. Consequently, when the dynamic range is + exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This + enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload + types. + + ASTERISK-26311 #close + + Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964 + (cherry picked from commit 9ac53877f688c06acaa7c377f15da8770e4ee88b) + +2016-11-02 09:15 +0000 [d971647949] Joshua Colp <jcolp@digium.com> + + * app_dial: Fix incorrect device state when channel is picked up. + + Given the scenario where multiple channels are dialed using Dial() + but the caller is picked up using PickupChan() all outgoing channels + except the channel specified to PickupChan() would be marked + as ringing until the call had been hung up. + + When using the PickupChan application the channel executing the + application is swapped into place of another channel. As part + of this process the channel is answered. The Dial application + has explicit logic which checks if the channel is answered, + cancels all other outgoing channels, and bridges. This logic is + different than the normal logic that is executed when an outgoing + channel is answered. This different logic failed to publish dial + events stating that the other outgoing channels had been canceled. + As a result references to the outgoing channels were held onto by + the dial masquerade process until the call had been ended and + the channels had gone away. This would result in the channels + appearing in the "core show channels" list despite not being present + anymore and would also result in incorrect device state. + + This change makes it so that this logic also publishes + dial events stating that the other outgoing channels have been + canceled. + + ASTERISK-26549 + + Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f + +2016-11-01 13:13 +0000 [afecb2cfc0] Richard Mudgett <rmudgett@digium.com> + + * bundled pjproject: Fix DNS write to freed memory. + + PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS + patch. + + The patch below fixes a write to freed memory under cartain DNS lookup + conditions. + + 0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch + + ASTERISK-26516 + Reported by: Richard Mudgett + + Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5 + +2016-11-01 06:56 +0000 [5f188bb7a8] Joshua Colp <jcolp@digium.com> + + * res_pjsip_sdp_rtp: Limit number of formats to defined maximum. + + The res_pjsip_sdp_rtp module did not restrict the number of + formats added to a media stream in the SDP to the defined + limit. If allow=all was used with additional loaded codecs this + could result in the next media stream being overwritten some. + + This change restricts the module to limit it to the defined + maximum and also increases the maximum in our bundled pjproject. + + ASTERISK-26541 #close + + Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8 + +2016-11-01 04:18 +0000 [94c9496ed5] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * netsock.c: fix includes for HURD + + ASTERISK-25070 + + Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814 + +2016-11-01 04:00 +0000 [c1c9487375] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * define PATH_MAX for HURD + + PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD + define it to a constant. It is indeed not safe to assume there won't be + longer paths and Asterisk generally does err safely on such cases. + + So even for HURD we'll just pretend PATH_MAX is 4096. + + ASTERISK-25070 #close + + Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3 + +2016-10-31 17:35 +0000 [50fa868ab8] Kevin Harwell <kharwell@digium.com> + + * codecs.conf.sample: Add sample and option descriptions for codec_opus + + codecs.conf.sample was missing codec opus's configuration options, descriptions, + and examples. This patch adds the configuration options and examples to + codecs.conf.sample that can be used with codec_opus. + + ASTERISK-26538 #close + + Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b + +2016-11-01 08:32 +0000 [b3f10b7b94] Grachev Sergey <grachev@mcn.ru> + + * chan_sip: Incorrect display option Outbound reg. retry 403 + + If in sip.conf (general section) set option register_retry_403=no, + the command "sip show settings" return value: + Outbound reg. retry 403:0 + If in sip.conf (general section) set option register_retry_403=yes, + the command "sip show settings" return value: + Outbound reg. retry 403:-1 + + * In static char "sip show settings" for "Outbound.reg. retry 403" + option use AST_CLI_YESNO + + ASTERISK-26476 #close + + Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9 + +2016-10-20 07:27 +0000 [29692d4aa4] Matt Jordan <mjordan@digium.com> + + * res/stasis: Add CLI commands for displaying/debugging ARI apps + + This patch adds three new CLI commands: + - ari show apps: list the registered ARI applications + - ari show app: show detailed information about an ARI application + - ari set debug: dump events being sent to an ARI application + + Note that while these CLI commands live in the res_stasis module, we use + the 'ari' family for these commands. This was done as most users of + Asterisk aren't aware of the semantic differences between ARI and + res_stasis, and some 'ari' CLI commands already exist. + + ASTERISK-26488 #close + + Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5 + +2016-10-31 16:12 +0000 [a36a7d0cf4] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fix compile of pjsua so it handles audio + + In order for pjsua and its python binding to actually negotiate + audio for the testsuite tests, it needs g711 and resample. The + pj* libraries themselves do not. Unfortunately, pjproject relies + on a brand new libresample that most distros don't ship so we need + to use the libresample already bundled with pjproject. Only the pjsua + executable and the _pjsua.so python library are linked with it so it + shouldn't interfere with asterisk itself. + + Also it was pointed out that apply_patches couldn't handle multiple + patches that depended on each other during the dry-run, so the + dry-run was removed. + + Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098 + +2016-10-31 13:46 +0000 [42bd70b29f] Etienne Lessard <elessard@proformatique.com> + + * manager: Add documentation for NewConnectedLine event. + + The NewConnectedLine event has been added by commit fe7671f, but the + documentation was missing. + + ASTERISK-26537 #close + + Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6 + +2016-10-30 13:33 +0000 [30b1bc77d2] Corey Farrell <git@cfware.com> + + * vector: Prevent NULL argument to memcpy. + + Headers declare that memcpy does not accept NULL argument for the first + two parameters. Add a conditional block to prevent memcpy and ast_free + from running on vectors with NULL element array. + + ASTERISK-26526 #close + + Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71 + +2016-10-29 10:31 +0000 [b96f18560b] Corey Farrell <git@cfware.com> + + * astobj2: Declare private variable data_size for AO2_DEBUG only. + + Every ao2 object contains storage for a private variable data_size, + though the value is never read if AO2_DEBUG is disabled. This change + makes the variable conditional, reducing memory usage. + + ASTERISK-26524 #close + + Change-Id: If859929e507676ebc58b0f84247a4231e11da07f + +2016-10-28 16:59 +0000 [6b1c55dc9b] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fix issue where "/version.mak" wasn't found + + main/Makefile includes third-party/pjproject/build.mak but + doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak" + evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile + before the include. + + Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604 + +2016-10-28 14:55 +0000 [d7f457e4c1] Richard Mudgett <rmudgett@digium.com> + + * bundled pjproject: Crashes while resolving DNS names. + + PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS + patch. + + The patches below fix the DNS lookup race condition crash caused by + attempting to send the same message twice for the single DNS lookup. + + 0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch + 0006-r5473-svn-backport-Fix-pending-query.patch + + The patch below removes a cached DNS response from the hash table when + another thread is referencing the old entry. The table still contained + the entry when it was destroyed which can result in inexplicable crashes. + + 0006-r5475-svn-backport-Remove-DNS-cache-entry.patch + + ASTERISK-26344 #close + Reported by: Ian Gilmour + + ASTERISK-26387 #close + Reported by: Harley Peters + + Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4 + +2016-10-28 09:50 +0000 [87903a6848] Rusty Newton <rnewton@digium.com> + + * SAC documentation: don't specify transports for endpoints and registrations + + Removing explicit transport definition for endpoints and registrations. It + isn't necessary and isn't generally advised. + + ASTERISK-26514 #close + + Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb + +2016-10-27 21:49 +0000 [f373de3020] Corey Farrell <git@cfware.com> + + * Fix shutdown crash caused by modules being left open. + + It is only safe to run ast_register_cleanup callbacks when all modules + have been unloaded. Previously these callbacks were run during graceful + shutdown, making it possible to crash during shutdown. + + ASTERISK-26513 #close + + Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21 + +2016-10-26 18:48 +0000 [61a5c3460e] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Remove usage of tar's --strip-components option + + Older versions of tar don't support the --strip-components option so + instead of doing 'tar --strip-components=1 -C source', we now just + untar to the tarball's root directory (pjproject-<version>) and + rename that directory to 'source'. + + Also fixed an issue where the pjproject source directory is a hard + coded absolute pathname. + + ASTERISK-26510 #close + ASTERISK-22480 #close + + Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0 + +2016-10-27 08:07 +0000 [675c71ae8c] Joshua Colp <jcolp@digium.com> + + * res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls. + + The res_pjsip_caller_id module wrongly assumed that a + saved From header would always exist on sessions. This + is true until an inbound call is received and a session + timer causes an UPDATE to be sent. In this case there will + be no saved From header and a crash will occur. This change + makes it fall back to the From header of the outgoing request + if no saved From header is present. + + ASTERISK-26307 #close + + Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa + +2016-10-26 07:51 +0000 [14496ce1e5] Joshua Colp <jcolp@digium.com> + + * app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. + + When executing the MailboxExists dialplan application and + MAILBOX_EXISTS dialplan function the passed in temporary voice + mailbox was not cleared, causing it to try to free garbage. + + ASTERISK-26503 #close + + Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3 + +2016-10-23 07:38 +0000 [e0bc17edff] Joshua Colp <jcolp@digium.com> + + * pjsip: Fix a few media bugs with reinvites and asymmetric payloads. + + When channel format changes occurred as a result of an RTP + re-negotiation the bridge was not informed this had happened. + As a result the bridge technology was not re-evaluated and the + channel may have been in a bridge technology that was incompatible + with its formats. The bridge is now unbridged and the technology + re-evaluated when this occurs. + + The chan_pjsip module also allowed asymmetric codecs for sending + and receiving. This did not work with all devices and caused one + way audio problems. The default has been changed to NOT do this + but to match the sending codec to the receiving codec. For users + who want asymmetric codecs an option has been added, asymmetric_rtp_codec, + which will return chan_pjsip to the previous behavior. + + The codecs returned by the chan_pjsip module when queried by + the bridge_native_rtp module were also not reflective of the + actual negotiated codecs. The nativeformats are now returned as + they reflect the actual negotiated codecs. + + ASTERISK-26423 #close + + Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc + +2016-10-26 06:32 +0000 [f534f67f52] Joshua Colp <jcolp@digium.com> + + * res_pjsip_sdp_rtp: Fix address family of explicit media_address. + + When an explicit media_address is provided the address family + in the SDP needs to be set to reflect it. + + ASTERISK-26309 + + Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79 + +2016-10-25 11:20 +0000 [3a2092b722] gtjoseph <gjoseph@digium.com> + + * test_astobj2_thrash: Fix multithreaded issues + + The test uses 4 threads to grow, count, lookup and shrink 15K objects + in a container. If there's only 1 execution engine available, the test + will complete in <50ms. If each threads gets its own execution engine, + the test may timeout after 60 seconds because the count thread does a + locked ao2_callback on the whole container in a tight loop with only + a sched_yield to give up time. The lock contention makes the test + execution times wildly variable and mostly timeout. 2 execution + engines are OK, 3 results in about 33% failure rate and >=4 causes + a 80% failure rate. + + To fix, the sched_yield was changed to a usleep(500). + + Also, the number of buckets specified for the container was an even + number so that was changed to the next prime number greater than + (MAX_HASH_ENTRIES / 100). That's 151 currently. + + Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77 + +2016-10-24 14:13 +0000 [640203802e] Pascal Cadotte Michaud <pcadotte@proformatique.com> + + * typo: s/paranthesis/parenthesis/ in a comment + + Change-Id: I7c1f4eb051177ee22cbe97e063d4a3effe29be30 + +2016-10-24 10:55 +0000 [9b3557e054] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Fixed various build issues + + * CFLAGS is now properly set when using older gcc. + * All third-party pjproject targets have been removed. This fixes + an issue with older libsrtp in some distros. + * Manually removing the source directory now causes a rebuild. + * EXTERNALS_CACHE_DIR is now properly checked. + * Whitespace fixes. + + Change-Id: I98fec6847efc5602a9f41cb95096fd660a49fa60 + +2016-09-19 06:13 +0000 [bb982480d8] Joshua Colp <jcolp@digium.com> + + * pjsip: Support dual stack automatically. + + This change adds support for dual stack automatically. No + configuration is required and the IP address and version + in the SIP messages and SDP will be automatically changed + based on the transport over which the message is being + sent. RTP usage has also been changed to listen on both + IPv4 and IPv6 simultaneously to allow media to flow, and + to allow ICE support on both simultaneously. This also + allows failover between IPv6 and IPv4 to work as expected. + + ASTERISK-26309 #close + + Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d + +2016-10-17 14:18 +0000 [eff97808fb] Mark Michelson <mmichelson@digium.com> + + * ARI: Detect duplicate channel IDs + + ARI and AMI allow for an explicit channel ID to be specified + when originating channels. Unfortunately, there is nothing in + place to prevent someone from using the same ID for multiple + channels. Further complicating things, adding ID validation to channel + allocation makes it impossible for ARI to discern why channel allocation + failed, resulting in a vague error code being returned. + + The fix for this is to institute a new method for channel errors to be + discerned. The method mirrors errno, in that when an error occurs, the + caller can consult the channel errno value to determine what the error + was. This initial iteration of the feature only introduces "unknown" and + "channel ID exists" errors. However, it's possible to add more errors as + needed. + + ARI uses this feature to determine why channel allocation failed and can + return a 409 error during origination to show that a channel with the + given ID already exists. + + ASTERISK-26421 + + Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06 + +2016-10-19 17:53 +0000 [c2036c827c] snuffy <snuffy22@gmail.com> + + * Fix issue with CLI not returning to prompt after running "features show" + + ASTERISK-26444 #close + + Change-Id: I91d645b7e6e5dba35f8c410df2be77a8c0e3acb8 + +2016-10-04 18:24 +0000 [3c62b60e56] Michael Walton <mike@farsouthnet.com> + + * res_rtp_asterisk: Add ice_blacklist option + + Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the + form ice_blacklist = <subnet spec>, e.g. ice_blacklist = + 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay + discovery. This is useful for optimizing the ICE process where a system + has multiple host address ranges and/or physical interfaces and certain + of them are not expected to be used for RTP. Multiple ice_blacklist + configuration lines may be used. If left unconfigured, all discovered + host addresses are used, as per previous behavior. + + Documention in rtp.conf.sample. + + ASTERISK-26418 #close + + Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9 + +2016-10-18 16:30 +0000 [012fda29d2] Mark Michelson <mmichelson@digium.com> + + * CDR: Alter destruction pattern for CDR chains. + + CDRs form chains. When the root of the chain is destroyed, it then + unreferences the next CDR in the chain. That CDR is destroyed, and it + then unreferences the next CDR in the chain. This repeats until the end + of the chain is reached. While this typically does not cause any sort of + problems, it is possible in strange scenarios for the CDR chain to grow + way longer than expected. In such a scenario, the destruction pattern + can result in a stack overflow. + + This patch fixes the problem by switching from a recursive pattern to an + iterative pattern for destruction. When the root CDR is destroyed, it is + responsible for iterating over the rest of the CDRs and unreferencing + each one. Other CDRs in the chain, since they are not the root, will + simply destroy themselves and be done. This causes the stack depth not + to increase. + + ASTERISK-26421 #close + Reported by Andrew Nagy + + Change-Id: I3ca90c2b8051f3b7ead2e0e43f60d2c18fb204b8 + +2016-10-18 09:04 +0000 [6d462b9eaf] Alexei Gradinari <alex2grad@gmail.com> + + * chan_pjsip: segfault on already disconnected session + + On heavy loaded system the TCP/TLS incoming calls could be + disconnected by pjproject while these calls are being + processed by asterisk. + + This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref + to inform pjproject that an INVITE session is in use. + + ASTERISK-26482 #close + + Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33 + +2016-10-18 03:01 +0000 [662b560c35] Alexander Traud <pabstraud@compuserve.com> + + * cli: Auto-complete File not Module for core set debug. + + Since Asterisk 1.8, the command "core set debug" on the command-line interface + asks not for a file (.c) but a module name. This change shows modules (.so) on + the auto-completion via a tabulator or the question mark. Now, when you + partially type a module name, TAB or ?, you get the correct candidiates. + + ASTERISK-26480 + + Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0 + +2016-09-11 10:13 +0000 [6f5880913f] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * menuselect: invalid test for GTK2 + + configuire.ac was only checking for the existence of pkg-config + and not the gtk2 package itself. Now it calls AST_PKG_CONFIG_CHECK + for gtk+-2.0. + + ASTERISK-26356 #close + + Change-Id: I8079d515d6ea99f9ab320a7eaa71c2aaa101ccd5 + +2016-10-13 02:06 +0000 [644fad7477] Moises Silva <moises.silva@gmail.com> + + * chan_rtp: Set a sane default rtp engine for unicast. + + ASTERISK-26439 + + Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011 + +2016-10-15 20:05 +0000 [42cfdcd1b7] Matt Jordan <mjordan@digium.com> + + * res/ari: Add the Asterisk EID field to outgoing events + + This patch adds the Asterisk EID field to all outgoing ARI events. + Because this field should be added to all events as they are + transmitted, it is appended to the JSON message just prior to it being + handed off to the application message handler. This makes it somewhat + resilient to both new events being added to ARI, as well as other + potential event transport mechanisms. + + ASTERISK-26470 #close + + Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d + +2016-10-16 17:25 +0000 [74d9385273] gtjoseph <gjoseph@digium.com> + + * utils.c: Fix ast_set_default_eid for multiple platforms + + ast_set_default_eid was searching for ethX, emX, enoX, ensX and even + pciD#U interface names. While this was a good attempt, it wasn't + inclusive enough to capture interfaces like enp6s0 or ens6d1, etc. + + Rather than relying on interface names, we now simply find the first + interface returned by the OS that has a hardware address and that + address isn't all 0x00 or all 0xff. The code IS different for BSD, + Solaris and Linux based on what method is available for enumerating + interfaces. + + Tested on: + FreeBSD9 + CentOS6 + Ubuntu14 + Fedora24 + + I was unable to test on Solaris at this time but the code for Solaris + is used elsewhere at Digium. + + Change-Id: Iaa6db87ca78a9a375e47d70e043ae08c1448cb72 + +2016-10-13 14:09 +0000 [0306869399] Leandro Dardini <ldardini@gmail.com> + + * app_queue: Added initialization for "context" parameter + + When using Asterisk Realtime Architecture, empty fields are skipped and the + default values are used. If the "context" parameter in queue was set and then + cleared from the database, the old value remains in memory and it continues + to be used. This change initialize the "context" parameter with an empty value, + allowing clearing the parameter. + + ASTERISK-26462 #close + + Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905 + +2016-10-11 06:55 +0000 [a859bcb49c] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia. + + In the SIP channel driver chan_sip, auto_comedia was expected to be used in + tandem with auto_force_rport. Or stated differently: Only when auto_force_rport + was chosen (the default), auto_comedia worked. This change allows auto_comedia + to be set independently of the state of (auto_)force_rport. For example, + nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments + when IPv6 clients are behind a Firewall. + + ASTERISK-26457 #close + + Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2 + +2016-10-17 19:08 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.12.0-rc1 Released. + +2016-10-17 11:39 +0000 [546ec4b038] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Add patch to address SSL crash + + Addresses crashes when an attempt is made to operate on an SSL socket + after the socket has been closed. + + ASTERISK-26477 #close + + Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002 + +2016-10-15 04:58 +0000 [f1fd873df0] Michael Kuron <m.kuron@gmx.de> + + * chan_sip: Only send video on outgoing channel if incoming channel supports it + + Previously, the settings videosupport=always and videosupport=yes behaved + identically and unconditionally caused a video offer to be sent in the SDP on + an outgoing call. This was a regression introduced with commit + 5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 in Asterisk 1.6.1. + + This commit restores correct behavior: videosupport=always causes a video offer + to be sent unconditionally, while videosupport=yes will only offer video on an + outbound channel if the incoming channel it is bridged to also supports video. + That way, the device receiving the outgoing call can display the correct user + interface elements for audio or video and will not unnecessarily show a blank + video window on an audio-only call. + + ASTERISK-17470 #close + + Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae + +2016-10-14 00:18 +0000 [ce4cfd2eca] Corey Farrell <git@cfware.com> + + * Fix issues with bundled pjproject cached download. + + Previously when testing I had a preexisting makeopts in ASTTOPDIR. The + ordering of configure.ac causes --with-externals-cache to be processed + after third-party configure. In cases where the Asterisk clone is + cleaned it would cause pjproject to be downloaded to /tmp. This + moves processing of the externals cache and sounds cache to happen + before third-party configure. + + This also addresses a possible issue with the third-party Makefile. If + TMPDIR is set by the environment it would override the path given to + --with-externals-cache. + + ASTERISK-26416 + + Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d + +2016-10-12 16:24 +0000 [3c54328c57] Richard Mudgett <rmudgett@digium.com> + + * Audit ast_json_pack() calls for needed UTF-8 checks. + + Added needed UTF-8 checks before constructing json objects in various + files for strings obtained outside the system. In this case string values + from a channel driver's peer and not from the user setting channel + variables. + + * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json + object construction. + + ASTERISK-26466 + Reported by: Richard Mudgett + + Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096 + +2016-10-12 16:20 +0000 [7f8f125738] Richard Mudgett <rmudgett@digium.com> + + * json: Check party id name, number, subaddresses for UTF-8. + + * Updated unit test as ast_json_name_number() is now NULL tolerant. + + ASTERISK-26466 #close + Reported by: Richard Mudgett + + Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6 + +2016-10-11 18:14 +0000 [9621c9bcbc] Richard Mudgett <rmudgett@digium.com> + + * json: Add UTF-8 check call. + + Since the json library does not make the check function public we + recreate/copy the function in our interface module. + + ASTERISK-26466 + Reported by: Richard Mudgett + + Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99 + +2016-10-12 17:42 +0000 [e4bb9f9a37] Richard Mudgett <rmudgett@digium.com> + + * aoc.c: Whitespace cleanup + + * In s_to_json() removed unnecessary ast_json_ref() to ast_json_null() + when creating the type json object. The ref is a noop. + + Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a + +2016-10-12 17:27 +0000 [bcac905bd3] Richard Mudgett <rmudgett@digium.com> + + * app_queue.c: Fix clearing of pause reason string. + + The pause reason is not always cleared when it should be cleared. + + * Made set_queue_member_pause() always clear pause reason if not pausing + with a reason string. + + Change-Id: I993dad19626ec017478a230e980989438b778c53 + +2016-10-12 16:22 +0000 [ee4ae2b648] Richard Mudgett <rmudgett@digium.com> + + * app_minivm.c: Fix malformed ast_json_pack() call. + + Change-Id: I082b239022fac462666e52a14a44304748908dc0 + +2016-10-12 16:30 +0000 [90ae4e4337] gtjoseph <gjoseph@digium.com> + + * res_config_mysql: Fix several issues related to recent table changes + + Unlike any of the other database drivers, res_config_mysql checks that + the table definition matches the requirements for every insert and + update statement. Since all requirements are forced to 'char', any + column that isn't a char, like ps_contacts' expiration_time, + qualify_timeout, etc., will throw a warning. It's kinda harmless but + very misleading. Since no other driver does those checks on insert + or update, they've been removed from res_config_mysql. Also, all + the logic that actually attempted to ALTER the table to fix the issue + has been removed. With the move to alembic, the auto-alter + functionality is not only unnecessary, it's also dangerous. + + The other issue is that res_config_mysql calls the mysql_insert_id + function inside store_mysql. Presumably the intention was to return + the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE + IS NON_PORTABLE AND MAY CHANGE. That value is then returned to + config realtime as the number of rows inserted. Guess what? The value + changed. It now only returns the number of rows inserted if there's an + auto increment column on the table, which ps_contacts doesn't have. + Otherwise it returns 0. So now, the insert worked but we tell config + realtime and sorcery that no rows were inserted. That call to + mysql_insert_id was removed and we now always return 1 if the insert + succeeded. We're only inserting 1 row at a time anyway. If the insert + fails, we still return -1. + + ASTERISK-26362 #close + Reported-by: Carlos Chavez + + Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4 + +2016-09-29 13:08 +0000 [86c15db6a1] Torrey Searle <torrey@voxbone.com> + + * res_fax: Fix a tight race condition causing fax to crash in audio fallback + + When T.38 gets rejected and G711 failback occurs there is a period of + time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set, + leading to a crash. + + Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982 + +2016-09-30 16:29 +0000 [29b7a5b00f] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com> + + * Add text of cdr directory into README.md for ast-db-manage + + Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636 + +2016-10-06 09:58 +0000 [f919edc4e2] gtjoseph <gjoseph@digium.com> + + * app_dial: Add the "Q" option to set the cause on unanswered channels + + The "Q" option will set the cause on the unanswered channels when + another channel answers. It overrides the default of + ANSWERED_ELSEWHERE. + + NOTE: chan_sip does not support setting the cause on a CANCEL to + anything other than ANSWERED_ELSEWHERE. + + ASTERISK-26446 #close + + Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47 + +2016-10-10 16:59 +0000 [a884b26392] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * vector: After remove element recheck index + + Small fix. It is necessary to double-check + the index that we just removed because there + is a new element. + + ASTERISK-26453 #close + + Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7 + +2016-09-29 12:52 +0000 [349c34f72a] Torrey Searle <torrey@voxbone.com> + + * res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge + + If a bridge switched to P2P when a DTMF was in progress it + was possible for the DTMF to continue being sent indefinitely. + + Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29 + +2016-10-10 10:59 +0000 [9da3489d24] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * res_pjsip_config_wizard: Memory leak in module_unload + + Fixed a memory leak. It removes only the first element. + Added a useful feature in vector.h to remove all items + under the CMP through a callback function / macro. + + ASTERISK-26453 #close + + Change-Id: I84508353463456d2495678f125738e20052da950 + +2016-10-09 21:53 +0000 [fa2885b3ff] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * cel_odbc: Fix memory leak on module unload + + Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715 + +2016-10-03 11:30 +0000 [e6b0053d75] gtjoseph <gjoseph@digium.com> + + * bundled_pjproject: Add tests for programs used by the Makefile, et al. + + Added tests for bzip2, tar, patch, sed and nm to configure.ac. + + Set DOWNLOAD_TO_STDOUT to a working command line regardless of + whether the download program is wget, curl or fetch. + + Added a 'configure.m4' file to the third-party directory which takes + care of calling any third-party project setup. Had to move some + pjproject_bundled stuff up in configure.ac so it was called before + the third-party configure macro. + + The pjproject tarball is now downloaded to the externals_cache_dir if + it was specified on the ./configure command line + + Removed regeneration of the pjproject aconfigure file. It was only + needed for an old patch that no longer applies. + + Converted the tests for symbols to explicit tests since we know that + they're now available in the bundled version. Saves a little time + during configure. + + ASTERISK-26416 #close + Reported-by: Corey Farrell + + Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b + +2016-10-05 14:53 +0000 [0dc0356e39] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Add MALLOC_DEBUG capability + + pjproject_bundled will now use the asterisk memory debugging APIs + if MALLOC_DEBUG is turned on in menuselect. + + Because this required stubs for the executable programs and the python + bindings, some Makefile reorganization was needed to properly handle + the dependencies. As a result, the makefile now individually makes + each of the pjproject libraries separately instead of making them all + in 1 shot. The only visible change is that there are separate status + lines printed for each library instead oif 1 for all libs. Also, the + making of the pjproject dependency files was eliminated. They're not + needed for building unless you're actively modifying pjproject source + files and it makes the build process faster. Finally, any issues with + parallel builds should be resolved again making the build faster. + + Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0 + +2016-10-07 17:32 +0000 [dd873bcada] Corey Farrell <git@cfware.com> + + * astobj2: Add backtrace to log_bad_ao2. + + * Compile __ast_assert_failed unconditionally. + * Use __ast_assert_failed to log messages from log_bad_ao2 + * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run. + + Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751 + +2016-10-04 16:59 +0000 [86550f9c17] gtjoseph <gjoseph@digium.com> + + * alembic: Allow cdr, config and voicemail to exist in the same schema + + cdr, config and voicemail are all separate alembic trees. Because + alembic's default is to use a table named 'alembic_version' to store + the current tree revision, the 3 trees can't exist in the same schema + without stepping on each other. + + Now each tree uses 'alembic_version_<tree_name>' as the version table. + Each tree's env.py script now first checks for 'alembic_version'. If + it finds it AND its revision is in the tree's history, the script + renames it to 'alembic_version_<tree_name>'. Regardless, the script + then continues with the migration using 'alembic_version_<tree_name>' + and creates that table if it's not found. The result is that if an + existing 'alembic_version' table was found but it didn't belong to this + tree, it's left alone and 'alembic_version_<tree_name>' is used or + created. + + WARNING: If multiple trees are using the same schema, they MUST NOT + CRU or D any objects with names that might exist in the other trees. + An example would be 'yesno_values' type. If two trees perform + operations on it, one tree could pull it out from under the other. + Thankfully we currently don't share any names among cdr, config and + voicemail. + + NOTE: Since the env.py scripts in each tree were identical, a common + env.py has been placed in the ast-db-manage directory and a symlink + to it has been placed in each tree directory. + + ASTERISK-24311 #close + Reported-by: Dafi Ni + + Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898 + +2016-10-05 04:25 +0000 [f166681c12] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Honor support of Symmetric Response (rport) for SIP requests. + + In the SIP channel driver chan_sip, the default is "auto_force_rport". When no + NAT was detected, for example in case of IPv6, Asterisk uses the IP address + from the headers within the SIP-REGISTER for subsequent SIP signaling. When + the remote party specifies support for Symmetric Response (RFC 3581) via the + parameter "rport", Asterisk should not extract the port from the SIP headers + but reuse the port of the transport. This did not happen because of a typo. + + ASTERISK-26438 #close + + Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6 + +2016-10-04 20:46 +0000 [430f6e5388] Michael Walton <mike@farsouthnet.com> + + * audiohooks: Remove redundant codec translations when using audiohooks + + The main frame read and write handlers in main/channel.c don't use the + optimum placement in the processing flow for calling audiohooks + callbacks, as far as codec translation is concerned. This change places + the audiohooks callback code: + * After the channel read translation if the frame is not linear before + the translation, thereby increasing the chance that the frame is linear + as required by audiohooks + * Before the channel write translation if the frame is linear at this + point + This prevents the audiohooks code from instantiating additional + translation paths to/from linear where a linear frame format is already + available, saving valuable CPU cycles + + ASTERISK-26419 + + Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f + +2016-09-29 14:02 +0000 [2449d2877c] Kevin Harwell <kharwell@digium.com> + + * Remove "format_ogg_opus: New format" + + This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c. + + ASTERISK-26426 #close + + Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5 + +2016-09-27 16:10 +0000 [f0a2e628d6] gtjoseph <gjoseph@digium.com> + + * download_externals: Fix issue with re-install + + Needed to ignore an xmlstarlet return code for optional element. + + Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873 + Reported-by: Dan Jenkins + +2016-09-22 09:49 +0000 [5258c067ae] gtjoseph <gjoseph@digium.com> + + * codec_opus: Add download ability to menuselect + + Updated codecs/codecs.xml to add codec_opus to the external + download list. + + ASTERISK-26409 + + Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4 + +2016-07-23 14:50 +0000 [a5af8709c8] gtjoseph <gjoseph@digium.com> + + * codec_opus: Replace res_format_attr_opus with the one from codec_opus + + Preparation + + ASTERISK-26409 + + Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 + (cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9) + +2016-07-23 15:56 +0000 [44c0c51cf1] gtjoseph <gjoseph@digium.com> + + * format_ogg_opus: New format + + Add Ogg/Opus playback support. + + This uses libopusfile in order to be able to read .opus files and play + them back. + + Writing/recording support is not present at this time. + + ASTERISK-26409 + + Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955 + +2016-09-24 19:05 +0000 [0ab443007b] gtjoseph <gjoseph@digium.com> + + * build_tools: Add ability to download variants to download_externals + + Some external packages have multiple variants that apply to different + builds of asterisk. The DPMA for instance has a "bundled" variant that + needs to be downloaded if asterisk was configured with + --with-pjproject-bundled. + + There are 2 ways to specify variants: + + If you need the user to make the decision about which variant to + download, simply create multiple menuselect "member" entries like so... + + <member name="res_digium_phone" displayname="..snipped.."> + <support_level>external</support_level> + <depend>xmlstarlet</depend> + <depend>bash</depend> + <defaultenabled>no</defaultenabled> + </member> + + <member name="res_digium_phone-bundled" displayname="..snipped.."> + <support_level>external</support_level> + <depend>xmlstarlet</depend> + <depend>bash</depend> + <defaultenabled>no</defaultenabled> + </member> + + Note that the second entry has "-<variant>" appended to the name. + You can then use the existing menuselect facilities to restrict which + members to enable or disable. Youy probably don't want the user to + enable multiple at the same time. + + If you want to hide the details of the variants, the better way to + do it is to create 1 member with "variant" elements. + + <member name="res_digium_phone" displayname="..snipped.."> + <support_level>external</support_level> + <depend>xmlstarlet</depend> + <depend>bash</depend> + <defaultenabled>no</defaultenabled> + <member_data> + <downloader> + <variants> + <variant tag="bundled" + condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/> + </variants> + </downloader> + </member_data> + </member> + + The condition must be a bash expression suitable for use with an "if" + statement. Any environment variable can be used plus those available + in makeopts. + + In this case, if asterisk was configured with --with-pjproject-bundled + the bundled variant will be automatically downloaded. Otherwise the + normal version will be downloaded. + + Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e + +2016-09-22 01:40 +0000 [a0a17a8c6f] Aaron An <anjb@ti-net.com.cn> + + * channels/chan_pjsip: fix HANGUPCAUSE function bug. + + HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. + This patch change the call order of ast_queue_control_data + and ast_queue_control in chan_pjsip_incoming_response. + + ASTERISK-26396 #close + Reported by: AaronAn + Tested by: AaronAn + + Change-Id: Ide2d31723d8d425961e985de7de625694580be61 + +2016-09-23 09:54 +0000 [0502675e5c] Alessandro Crespi + + * chan_sip: Resolve externhost not to IPv6; instead go for IPv4. + + For the channel driver chan_sip, you specify externhost=example.com in sip.conf + when your Asterisk is behind a NAT and your IP address is assigned dynamically. + Or stated differently: You do not have a static IP address to use "externaddr" + directly. This NAT support is quite handy but just about IPv4. Previously, + Asterisk resolved "externhost" to any IP version. When the first DNS answer + resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and + connection (c=). This happened in outgoing SIP-REGISTER and while answering + SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an + IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". + + ASTERISK-18232 #close + Reported by: Jacek Kowalski + Tested by: Alexander Traud + patches: + changes.patch submitted by Alessandro Crespi + + Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac + +2016-09-20 09:42 +0000 [0056bcaebd] gtjoseph <gjoseph@digium.com> + + * chan_sip: Address runaway when realtime peers subscribe to mailboxes + + Users upgrading from asterisk 13.5 to a later version and who use + realtime with peers that have mailboxes were experiencing runaway + situations that manifested as a continuous stream of taskprocessor + congestion errors, memory leaks and an unresponsive chan_sip. + + A related issue was that setting rtcachefriends=no NEVER worked in + asterisk 13 (since the move to stasis). In 13.5 and earlier, when a + peer tried to register, all of the stasis threads would block and + chan_sip would again become unresponsive. After 13.5, the runaway + would happen. + + There were a number of causes... + * mwi_event_cb was (indirectly) calling build_peer even though calls to + mwi_event_cb are often caused by build_peer. + * In an effort to prevent chan_sip from being unloaded while messages + were still in flight, destroy_mailboxes was calling + stasis_unsubscribe_and_join but in some cases waited forever for the + final message. + * add_peer_mailboxes wasn't properly marking the existing mailboxes + on a peer as "keep" so build_peer would always delete them all. + * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions + then just creating them again. + + All of this was causing a flood of subscribes and unsubscribes on + multiple threads all for the same peer and mailbox. + + Fixes... + * add_peer_mailboxes now marks mailboxes correctly and build_peer only + deletes the ones that really are no longer needed by the peer. + * add_peer_mwi_subs now only adds subscriptions marked as "new" instead + of unsubscribing and resubscribing everything. It also adds the peer + object's address to the mailbox instead of its name to the subscription + userdata so mwi_event_cb doesn't have to call build_peer. + + With these changes, with rtcachefriends=yes (the most common setting), + there are no leaks, locks, loops or crashes at shutdown. + + rtcachefriends=no still causes leaks but at least it doesn't lock, loop + or crash. Since making rtcachefriends=no work wasnt in scope for this + issue, further work will have to be deferred to a separate patch. + + Side fixes... + * The ast_lock_track structure had a member named "thread" which gdb + doesn't like since it conflicts with it's "thread" command. That + member was renamed to "thread_id". + + ASTERISK-25468 #close + + Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0 + +2016-09-21 15:03 +0000 [323aff3a09] Joshua Colp <jcolp@digium.com> + + * core: Ensure presencestate subtype and message are NULL. + + When retrieving presence state information there is no + guarantee that the subtype and message passed in are + set to NULL. This change ensures they are. + + ASTERISK-26397 #close + + Change-Id: I61f8187972d5d8bbd7d6b7f4daa4f4f7e8237b23 + +2016-09-21 10:48 +0000 [10c180760c] Joshua Colp <jcolp@digium.com> + + * res_odbc: Make pooling option deprecation notice more useful. + + This changes the notice for the deprecation of the old + pooling options to point to the new option for doing + pooling. This gives a clearer direction as to what to + look into. + + ASTERISK-26389 #close + + Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10 + +2016-09-12 07:37 +0000 [42cc267016] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * cdr_mysql: fix UTC support + + * Make 'cdrzone=UTC' work properly. + * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone + + ASTERISK-26359 #close + + Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778 + +2016-09-21 08:46 +0000 [f16ab19292] Joshua Colp <jcolp@digium.com> + + * odbc: Remove options that are no longer applicable. + + The pooling, shared_connection, limit, and idlecheck options + are no longer used in res_odbc. + + ASTERISK-26389 + + Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6 + +2016-09-20 15:17 +0000 [c9ce299b64] Corey Farrell <git@cfware.com> + + * core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. + + Move the function outside the conditional block that excludes + LOW_MEMORY. + + ASTERISK-26273 #close + + Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4 + +2016-09-20 10:05 +0000 [610eb4c189] Corey Farrell <git@cfware.com> + + * logger: Fix default console settings. + + When logger.conf is missing or invalid we should be printing notices, + warnings and errors to the console. The logmask was incorrectly + calculated. + + Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3 + +2016-06-27 14:26 +0000 [36092ee3a0] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * sd_notify (systemd status notifications) support + + sd_notify() is used to notify systemd of changes to the status of the + process. This allows the systemd daemon to know when the process + finished loading (and thus only start another program after Asterisk has + finished loading). + + To use this, use a systemd unit with 'Type=notify' for Asterisk. + + This commit also adds the function ast_sd_notify(), a wrapper around + sd_notify that does nothing if not built with systemd support. + + Also adds support for libsystemd detection in the configure script. + + Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 + (cherry picked from commit 07b95f7c65b7c083724f1af2b26f93cc22cad58c) + +2016-09-19 14:21 +0000 [9372d32100] Walter Doekes <walter+github@wjd.nu> + + * asterisk.c: Non-root users also get the astcanary after core restart. + + Without this change, a 'core restart' would kill the astcanary forever + if you're not running as root. Both with and without this patch, the + scheduling priority was still SCHED_RR after restart. + + Additionally, the astcanary is now spawned if you start with high + priority and Asterisk doesn't get a chance to lower it. For example + through: `chrt -r 10 sudo -u asterisk asterisk -c` + + Also reap killed astcanary processes on core restart. + + ASTERISK-26352 #close + + Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55 + +2016-09-19 09:40 +0000 [e96448e991] Walter Doekes <walter+github@wjd.nu> + + * asterisk.c: When astcanary dies on linux, reset priority on all threads. + + Previously only the canary checking thread itself had its priority set + to SCHED_OTHER. Now all threads are traversed and adjusted. + + ASTERISK-19867 #close + Reported by: Xavier Hienne + + Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39 + +2016-09-09 06:35 +0000 [01884a7af6] Timo Teräs <timo.teras@iki.fi> + + * Fix showing of swap details when sysinfo() is available + + If sysinfo() is available, but not sysctl() or swapctl() the + printing code for swap buffer sizes is incorrectly omitted. + The above condition happens with musl c-library. + + Fix #if rule to consider defined(HAVE_SYSINFO). And also + remove the redundant || defined(HAVE_SYSCTL) which was + incorrectly there to start with. Now swap information is + displayed only if an actual libc function to get it is + available. + + This also fixes warnings previously seen with musl libc: + + [CC] asterisk.c -> asterisk.o + asterisk.c: In function 'handle_show_sysinfo': + asterisk.c:773:6: warning: variable 'totalswap' set but not used + [-Wunused-but-set-variable] + int totalswap = 0; + ^~~~~~~~~ + asterisk.c:770:11: warning: variable 'freeswap' set but not used + [-Wunused-but-set-variable] + uint64_t freeswap = 0; + ^~~~~~~~ + + Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca + +2016-09-12 18:00 +0000 [cdbad152c7] Richard Mudgett <rmudgett@digium.com> + + * res_config_odbc.c: Fix buffer size limitation creating invalid SQL. + + Creating ODBC SQL queries resulted in queries too large to fit into the + supplied buffer. The resulting truncated buffer contained an invalid SQL + query. + + * Made SQL query generation code use a thread storage buffer that can + increase in size as needed. + + * Fixed bad multi-line warning messages. + + ASTERISK-26263 #close + Reported by: Jeppe Ryskov Larsen + + Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae + +2016-09-14 08:42 +0000 [449719be00] Joshua Colp <jcolp@digium.com> + + * res_pjsip_multihomed: Change Contact port to listening port. + + The res_pjsip_multihomed module determines what interface and transport + a request is going out on and updates the SIP message accordingly with + the address information. This currently incorrectly updates the Contact + header for connectionful protocols to the ephemeral connection port, + instead of the bound address for the listening socket which can actually + accept the connection back. If the remote side attempts to connect back on + the epehemeral port it will fail. + + This change makes it so the port is updated to the bound port on + connectionful protocols and is maintained on UDP (as there can be + multiple of those). + + ASTERISK-26374 #close + + Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab + +2016-09-07 14:48 +0000 [4d64b176eb] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Prevent SERVFAIL from marking name server bad + + A name server that returns "Server Failure" is indicating only that + the server couldn't process that particular request. We should NOT + assume that the name server is incapable of serving other requests. + + Here's the scenario we've been encountering... + + * 2 local name servers configured in resolv.conf. + * An OPTIONS request causes a request for A and AAAA records to go out + to both nameservers. + * The A responses both come back successfully resolved. + * Because of an issue at some upstream nameserver, the AAAA responses + for that particular query come back as "SERVFAIL" from both local + name servers. + * Both local servers are marked as bad and no further queries can be + sent until the 60 second ttl expires. Only previously cached results + can be used. + * In this case, 60 seconds is just enough time for another OPTIONS + request to go out to the same host so the cycle repeats. + + We could set the bad ttl really low but that also affects REFUSED and + NOTAUTH which probably DO signal a real server issue. Besides, even + a really low bad ttl would be an issue on a pbx. + + Although we use our own resolver in 14 and master and don't have this + issue there, Teluu has merged this patch upstream so it's appropriate + to cherry-pick to 14 and master to keep pjproject consistent. + + + Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0 + +2016-09-14 07:59 +0000 [1cac856e17] Joshua Colp <jcolp@digium.com> + + * rtp: Preserve timestamps on video frames. + + Currently when receiving video over RTP we store only + a calculated samples on the frame. When starting the video + it can take some time for this calculation to actually yield + a value as it requires constant changing timestamps. As well + if a video frame passes over multiple RTP packets this calculation + will fail as the timestamp is the same as the previous RTP + packet and the number of samples calculated will be 0. + + This change preserves the timestamp on the frame and allows + it to pass through the core. When sending the video this timestamp + is used instead of a new one being calculated. + + ASTERISK-26367 #close + + Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd + +2016-09-14 09:51 +0000 [9df4056d70] Joshua Colp <jcolp@digium.com> + + * res_pjsip_transport_management: Convert time in log message to seconds. + + ASTERISK-26375 #close + + Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc + +2016-09-13 05:34 +0000 [98e42cc662] Steve Davies <steve@one47.co.uk> + + * chan_sip: Fix session timeout on retransmit of non-UDP packets + + Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for + SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP + connections, allowing the TCP layer to handle the retransmits. Unfortunately, + this caused sessions to be terminated with a retransmit timeout becasue it + stopped at the point of the first retrans call. + + This patch waits for the 64*T1 timer to expire instead. + + ASTERISK-19968 + + Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204 + +2016-09-12 12:25 +0000 [0388882cdb] Richard Mudgett <rmudgett@digium.com> + + * app_queue: Fix CLI "queue show" and AMI Queues action output truncation. + + The output of CLI "queue show" and AMI Queues action is truncated and + "failed to extend from 240 to 327" messages are generated if the queue + member and interface names are lengthy. + + * Increase the string buffer size from 240 to 512 in order to accommodate + for more information fields added to the output since v1.8. + + ASTERISK-26360 #close + Reported by: Richard Mudgett + + Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d + +2016-09-12 03:28 +0000 [da8ba990d1] Walter Doekes <walter+github@wjd.nu> + + * chan_sip: Allow target refresh (Contact update) on re-INVITE. + + Previously, the Contact was stored only on initial INVITE and on any + 18X and 200. That meant that after re-INVITEs from *us* the Contact + could get updated, but after re-INVITEs from the *peer*, it did not. + + This changeset fixes this inconsistency, properly allowing target + refreshes through re-INVITES (RFC3261, 12.2). + + If your strictrtp setting allows it, this change allows you to switch + the source IP of a connected/calling device mid-call with a simple + re-INVITE from the new IP. + + ASTERISK-26358 #close + + Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435 + +2016-08-31 15:22 +0000 [e9ddab4685] Richard Mudgett <rmudgett@digium.com> + + * sip_to_pjsip.py: Map legacy_useroption_parsing. + + Map the sip.conf general section legacy_useroption_parsing to the + new pjsip.conf global ignore_uri_user_options. + + ASTERISK-26316 + Reported by: Kevin Harwell + + Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc + +2016-08-29 18:08 +0000 [30af92e78d] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Add ignore_uri_user_options option. + + This implements the chan_sip legacy_useroption_parsing option but with a + better name. + + * Made the caller-id number and redirecting number strings obtained from + incoming SIP URI user fields always truncated at the first semicolon. + People don't care about anything after the semicolon showing up on their + displays even though the RFC allows the semicolon. + + ASTERISK-26316 #close + Reported by: Kevin Harwell + + Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62 + +2016-09-09 06:26 +0000 [7ed5dc2c58] Walter Doekes <walter+github@wjd.nu> + + * contrib: Let safe_asterisk script continue without /dev/tty9. + + If you use the safe_asterisk script, it uses hardcoded defaults before + running configurable values from /etc/asterisk/startup.d. The hardcoded + default has TTY=9. Some containerized environments don't have such a + TTY, and safe_asterisk would stop. + + The custom configuration from /etc/asterisk/startup.d/* isn't read until + after it stopped, so changing TTY in a custom config did not help. + + This changeset changes safe_asterisk to continue if the TTY setting was + untouched and /dev/tty9 and /dev/vc/9 aren't found. + + Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc + +2016-09-09 05:39 +0000 [7580a736bb] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Only invoke unidentified endpoint logic when unidentified. + + The code was incorrectly invoking the unidentified logic when + an endpoint had actually been identified, causing log messages + to be output. + + ASTERISK-26349 #close + + Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f + +2016-08-23 06:35 +0000 [efcfc4c1ee] Corey Farrell <git@cfware.com> (license 5909) + + * chan_sip: Don't allocate new RTP instances on top of old ones. + + In some scenarios dialog_initialize_rtp can be called multiple times on + the same dialog. This can cause RTP instances to be leaked along with + multiple file descriptors for each instance. + + This change makes it so the existing RTP instances are destroyed and + not overwritten, stopping the memory leak. + + ASTERISK-26272 #close + patches: + ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) + + Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 + +2016-08-16 15:34 +0000 [f1ffc22933] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Do not crash on ACKs from unknown endpoints. + + The endpoint identification PJSIP module is intended to identify which + endpoint an incoming request is from. If an endpoint is not identified, + then an artificial endpoint is used in its place when proceeding. + + The problem is that the ACK request type is an exception to the rule. + The artificial endpoint is not used when processing an ACK. This results + in the possibility of having a NULL endpoint being used further on. + + The reason ACK is an exception is an attempt not to spam security logs + with unidentified requests. Presumably, you've already logged the + unidentified request on the preceeding INVITE. + + Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion + didn't cause an issue. A new change in 13.10 added endpoint ACL checking + shortly after endpoint identification. Because we are accessing a NULL + endpoint, this ACL check resulted in a crash. + + The fix here is to be sure to retrieve the artificial endpoint for all + request types. ACKs still do not generate unidentified request security + events. + + ASTERISK-26264 #close + Reported by nappsoft + + AST-2016-006 + + Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703 + +2016-09-06 11:46 +0000 [23d6ec7417] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_messaging.c: Misc cleanups and fixes. + + * Eliminated RAII_VAR in get_outbound_endpoint(). + + * Simplify update_to() coding. However, this function can only be a NoOp + because the To string can only be a URI and not a name-address formatted + string. + + * Simplify update_from() coding. Also fixed a code path modifying the + from string when the caller could still want to use the original string. + + * Fixed msg_data_create() incompletely removing the "pjsip:" to then add + back the "sip:" string if needed. The code didn't handle the "pjsip:sip:" + case because it left the colon after pjsip in the string. + + Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db + +2016-09-07 16:00 +0000 [5f19657710] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Allow global headers to be overridden. + + Currently when you add global headers from the dialplan both + the header in the dialplan and the globally configured header + are added to the resulting SIP INVITE. This change makes it + so the headers in the dialplan take precedence and are the + only ones added. + + Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad + +2016-08-11 12:10 +0000 [206d4f57dc] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * followme: initialize all config items on reload + + Some configuration directives were not initialized on reload, and hence + were not reset to default if they were removed from followme.conf. + + ASTERISK-26288 #close + + Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150 + +2016-08-01 20:55 +0000 [117a7741c8] gtjoseph <gjoseph@digium.com> + + * build: Add download capability for external packages + + The DPMA and g729a, silk, siren7 and siren14 codecs hosted at + http://downloads.digium.com/pub/telephony/ are now listed in the + "External" sections of the "Resource Modules" and "Codec Translators" + pages in menuselect. Any that are selected will automatically be + downloaded and installed when "make install" is run. Their LICENSE and + README (if avaialble) files will be installed to + ASTVARLIBDIR/documentation/thirdparty/<product_name>. + + Example use with codecs: + + The codecs/codecs.xml file is a menuselect style xml file that lists + the codecs to be included. Their support levels are 'external', which + triggers the download and install, and defaultenabled is no. Also + because codec_g729a is actually in a directory named codec_g729 on the + download server, the newly added 'member_data' element is used to + override the default of the directory name being the package name. You + can use the 'directory_name' attribute to keep default base URL + (http://downloads.digium.com/pub/telephony/) but use the new directory, + or you use the 'remote_url' attribute to specify a full URL to the + download directory. In this case, you must still follow the same + subdirectory naming conventions as that used for the packages located + at 'http://downloads.digium.com/pub/telephony'. + + A new configure option '--with-externals-cache' was added and like + '--with-sounds-cache' it allows the installer to cache tarballs so + they're not downloaded every time. + + To assist with the download and install process, each external package + now has a manifest.xml file that, among other things, contains a package + version and checksums for each file in the tarball. The manifest is + saved to both the cache directory and ASTMODDIR and together with the + manifest.xml on the downloads site, tells the install scripts whether + a download and/or update is needed. + + bash and xmlstarlet are required for downloader operation. If they're + not installed, the external items in menuselect will be unavailable. + + Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a + +2016-09-06 02:41 +0000 [d04ae7d1d8] Walter Doekes <walter+github@wjd.nu> + + * chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. + + Certain SNOM phones send so-called "optional crypto" in their SDP body. + Regular SRTP setup looks like this: + + m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 + a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... + + SNOM-style "optional crypto" looks like this: + + m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 + a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... + + A crypto line is supplied, but the m-line does not have SAVP. + + When res_srtp.so is *not* loaded, then chan_sip.so treats the optional + crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the + incoming call with the following message: + + WARNING: process_sdp: Failed to receive SDP offer/answer with + required SRTP crypto attributes for audio + + For platforms that want to start providing SRTP this presents a + compatibility problem. + + This changeset lets chan_sip handle the SDP as if no crypto-line was + supplied: i.e. accept the call as regular RTP, just like it did before + res_srtp was loaded. + + Now you'll get this informative warning instead: + + WARNING: Ignoring crypto attribute in SDP because RTP transport is + insecure + + ASTERISK-23989 #close + Reported by: Olle Johansson + + Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2 + +2016-09-03 16:04 +0000 [df3d0188e4] Matt Jordan <mjordan@digium.com> + + * apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option + + In any scenario in which the callee is not connected to the caller, the + current code in app_dial will crash due to raising a Dial End Stasis + Message after the callee channel has been hung up. This patch corrects + the error by simply moving the explicit hangup of the callee (peer) + channel until after the dial end message. + + ASTERISK-25691 #close + + Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d + +2016-09-03 16:02 +0000 [a64063cc97] Matt Jordan <mjordan@digium.com> + + * apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 + + If the callee selects option '5' using the Dial application's privacy + (P) option, the DIALSTATUS is erroneously set to ANSWER. This option + reflects the callee sending the caller to VoiceMail one time; the call + is definitely *not* ANSWERed in such a scenario. With this patch, the + DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that + is set when the 'send to VoiceMail every time' option is set. + + ASTERISK-25691 + + Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358 + +2016-08-30 16:40 +0000 [03fc438f6e] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). + + Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09 + +2016-08-29 18:06 +0000 [b5e753227d] Richard Mudgett <rmudgett@digium.com> + + * pjsip_configuration.c: Ignore repeated identify by methods. + + Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838 + +2016-08-30 17:26 +0000 [9b7501b6ad] Richard Mudgett <rmudgett@digium.com> + + * config_global.c: Comments and a default expression adjustment. + + Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3 + +2016-08-31 15:14 +0000 [3314e1cec2] Richard Mudgett <rmudgett@digium.com> + + * sip_to_pjsip.py: Map canreinvite as directmedia alias. + + Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2 + +2016-08-31 15:37 +0000 [6372f40ba0] Richard Mudgett <rmudgett@digium.com> + + * sip_to_pjsip.py: Fix typo converting outboundproxy registration. + + Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15 + +2016-08-31 15:13 +0000 [11eb1afd2d] Richard Mudgett <rmudgett@digium.com> + + * sip_to_pjsip.py: Fix comment typo and tabs. + + Change-Id: If35174614545727817d329c60ba4456c028941b5 + +2016-08-31 15:56 +0000 [0f9b144c1a] Richard Mudgett <rmudgett@digium.com> + + * Sample configs: Eliminate false multiline comment block starts. + + Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6 + +2016-09-02 11:36 +0000 [8d1c535bd6] Richard Mudgett <rmudgett@digium.com> + + * format_cap.c: Fix CLI "core show channeltype Surrogate" crash. + + * Make ast_format_cap_get_names() NULL tolerant. + + ASTERISK-26331 #close + Reported by: CGI.NET + + Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3 + +2016-08-18 14:45 +0000 [9bca895469] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_session: segfault on already disconnected session + + On heavy loaded system the TCP/TLS incoming calls could be + disconnected by pjproject while these calls are being + processed by asterisk which could use the session's memory pools. + If the session in the disconnected state then the session memory + pools were already freed, so we get segfault. + + This patch adds a lifetime control on an INVITE session to pjproject. + The lifetime of the session is manipulated by calling + pjsip_inv_add_ref/pjsip_inv_dec_ref. + This patch uses these functions to inform pjproject that the + session is in use. + + This patch adds check if the session state is not disconnected + and also checks if the memory pool is not NULL. + + This patch also places tasks 'session_end' and 'session_end_completion' + into session's serializer to avoid race condition. + + ASTERISK-26291 #close + + Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7 + +2016-08-10 15:14 +0000 [63feffa126] Mark Michelson <mmichelson@digium.com> + + * ConfBridge: Make some announcements asynchronous. + + Confbridge announcements tend to block a channel while they are being + played. In some circumstances, this is warranted since you want that + particular channel not to hear the announcement (Example: "John Doe has + entered the conference"). For others it makes less sense. + + This change first introduces methods for playing sounds asynchronously + into the conference. This is very similar to how synchronous sounds are + played, except the channel initiating the playback does not wait for the + sound to complete before moving on. + + Asynchronous announcements are used for two circumstances: + * Sounds played for a user after they have left the bridge + * Sounds that play first to a single user and then the rest of the + conference (if the channel and conference use the same language) + + ASTERISK-26289 #close + Reported by Mark Michelson + + Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a + +2016-08-31 12:23 +0000 [a002a4d2db] Michael Kuron <m.kuron@gmx.de> + + * app_mp3: Use correct buffer size and the same sample rate as the channel + + Previously, the buffer used for MP3 streamed from HTTP servers had a size of + 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 + minute. Only when the buffer is full does audio start to play. + For MP3 files streamed from a server, that is usually not a big deal as long as + the connection to the server is fast enough to supply that much data within a + second or two. For MP3 live streams however, it takes 1 minute to download 1 + minute of audio, so without this change, app_mp3 wasn't really usable for MP3 + live streams. + This commit changes the buffer size so that it covers 6 seconds of an MP3 file + streamed from a server and 0.5 seconds of an MP3 live stream. The latter is + identified by the use of a .m3u file extension. + + app_mp3 so far only supported 8 kHz audio. + Now it always runs at the sample rate of the channel. + + ASTERISK-26085 #close + + Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0 + +2016-08-26 10:39 +0000 [308a65fe6c] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: qualify/unqualify added/deleted realtime endpoints + + If the PJSIP endpoint's AOR with the permanent contact + was deleted from the realtime storage the res_pjsip module + continues trying to qualify this contact. + The error 'Unable to find an endpoint to qualify contact' + appeares every 'qualify_frequency' seconds. + This patch deletes this contact in this case. + + The PJSIP endpoint's AOR with the permanent contact + is never qualified if it is added to realtime storage + after asterisk started. + This patch adds qualifying for the AOR's permanent contacts + on the first handling of this AOR. + + ASTERISK-26319 #close + + Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe + +2016-08-17 02:51 +0000 [2fa168348e] chris de rock <chris@derock.de> + + * app_macro: Consider '~~s~~' as a macro start extension. + + As described in issue ASTERISK-26282 the AEL parser creates macros with + extension '~~s~~'. app_macro searches only for extension 's' so the + created extension cannot be found. with this patch app_macro searches for + both extensions and performs the right extension. + + ASTERISK-26282 #close + + Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb + +2016-08-29 07:10 +0000 [27951792c4] Etienne Lessard <elessard@proformatique.com> + + * pbx.c: Prevent infinite recursion in manager_show_dialplan_helper. + + Previously, if context A was including context B and context B was including + context A, i.e. if there was a circular dependency between contexts, then + calling manager_show_dialplan_helper could lead to an infinite recursion, + resulting in a crash. + + This commit applies the same solution as the one implemented in the + show_dialplan_helper function. The manager_show_dialplan_helper and + show_dialplan_helper functions contain lots of code in common, but the former + was missing the "infinite recursion avoidance" code. + + ASTERISK-26226 #close + + Change-Id: I1aea85133c21787226f4f8442253a93000aa0897 + +2016-08-26 14:34 +0000 [fb82fdb013] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Disable srtp use by pjmedia + + The reason for the disable is that while Asterisk works fine with older + libsrtp versions, newer versions of pjproject won't compile with them. + Debian 6 for instance, has libsrtp 1.4.4 which is older than what + pjproject is expecting. + + We don't use most of pjmedia but we DO use it for SDP negotiation. + Luckily disabling srtp in pjmedia doesn't interfere with it's ability + to negitiate a secure channel. The proper crypto attributes are + negotiated in both directions. + + ASTERISK-26279 #close + + Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2 + +2016-08-26 08:41 +0000 [847bd47ff0] Alexander Traud <pabstraud@compuserve.com> + + * channel: No hung-up on failing security requirements. + + In your Diaplan, if you specify + same => n,Set(CHANNEL(secure_bridge_media)=1) + same => n,Set(CHANNEL(secure_bridge_signaling)=1) + only the SIP channel driver chan_sip supports this. All other channels drivers + like res_pjsip fail. In case of failure, the original sRTP source code released + the whole channel, even if not hung-up, yet. This change does not release the + channel but instead hangs-up the channel. + + ASTERISK-26306 + + Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db + +2016-08-20 09:04 +0000 [b59d3b48d0] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. + + When using the migration script sip_to_pjsip.py, and your sip.conf is + configured with bindaddr=::, two transports are written to pjsip.conf, one for + 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 + and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface + like in chan_sip. + + Furthermore, the script internal functions "build_host" and "split_hostport" + did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change + makes sure, even such addresses are parsed correctly. + + ASTERISK-26309 + + Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48 + +2016-08-25 07:06 +0000 [f69f5cd3c4] Joshua Colp <jcolp@digium.com> + + * app_queue: Ensure member is removed from pending when hanging up. + + When dialing channels it is possible that they may not ever + leave the not in use state (Local channels in particular) by + the time we cancel them. If this occurs but we know they were + dialed we explicitly remove them from the pending members + container so that subsequent call attempts occur. + + ASTERISK-26299 #close + + Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65 + +2016-08-04 20:11 +0000 [5cd583d7a2] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Cache global config options. + + We may check a global config option hundreds of times a second or more. + Asking sorcery for the global configuration from the config files backend + involves several allocations and container traversals. Using realtime + without a memory cache is a lot worse because you have to lookup in the + realtime database each time to reconstitute the sorcery object. With a + memory cache for realtime, there is about the same amount of overhead as + for config files. Either way, it is still fairly expensive to access the + sorcery object that much. + + * Cache the global config options so we can access them faster. You must + now always perform a res_pjsip reload to change the global options. + + Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7 + +2016-08-23 11:02 +0000 [8b4b2500ee] Richard Mudgett <rmudgett@digium.com> + + * res_fax: Fix deadlock in ast_channel_get_t38_state(). + + ast_channel_get_t38_state() calls ast_channel_queryoption() with + AST_OPTION_T38_STATE. If the passed in channel is a local channel then a + deadlock can happen if a channel lock is held when called. + + * Made ast_channel_get_t38_state() callers not hold a channel lock before + calling. + + * Update ast_channel_get_t38_state() doxygen to note that no channel locks + can be held when calling the function. + + ASTERISK-26203 #close + Reported by: Etienne Lessard + + ASTERISK-24822 #close + Reported by: David Brillert + + ASTERISK-22732 #close + Reported by: Richard Mudgett + + Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214 + +2016-08-23 10:39 +0000 [e8d4f40022] Richard Mudgett <rmudgett@digium.com> + + * res_fax: Fix deadlock setting FAXMODE channel variable. + + ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. + Unfortunately, it also introduced a deadlock potential because + set_channel_variables() which sets FAXMODE can be called during a + masquerade. The ast_channel_get_t38_state() which gets the value used to + set FAXMODE cannot be called with the channel locked. As a result, local + channels can deadlock because of how they must acquire the locks necessary + to operate. + + The intent of FAXMODE is for dialplan to know how a fax was transferred + after the fax completes. However, the previous patch sets FAXMODE to the + channel's current T.38 state AFTER the fax has completed and where T.38 + may have already disconnected. + + * Set FAXMODE based upon T.38 negotiations exchanged either with the fax + applications or the fax framehooks. + + ASTERISK-26203 + Reported by: Etienne Lessard + + ASTERISK-24822 + Reported by: David Brillert + + ASTERISK-22732 + Reported by: Richard Mudgett + + Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1 + +2016-08-22 12:31 +0000 [35cf6c7702] Richard Mudgett <rmudgett@digium.com> + + * res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). + + fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be + called with any channel locks already held. A deadlock can happen if the + function is operating on a local channel. + + * Made fax_gateway_indicate_t38() unlock the channel before calling + ast_indicate_data() since fax_gateway_indicate_t38() is always called with + the channel locked. + + * Made fax_gateway_indicate_t38() return void since nothing cared about + its return value. + + ASTERISK-26203 + Reported by: Etienne Lessard + + ASTERISK-24822 + Reported by: David Brillert + + ASTERISK-22732 + Reported by: Richard Mudgett + + Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407 + +2016-08-23 11:16 +0000 [50b2aa506f] Richard Mudgett <rmudgett@digium.com> + + * res_fax.c: Add chan locked precondition comments. + + Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7 + +2016-08-23 10:42 +0000 [038cbc0215] Richard Mudgett <rmudgett@digium.com> + + * ast_framehook_detach() must be called with the channel locked. + + The framehook container could become corrupted if the channel lock is not + held before calling. + + Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584 + +2016-08-22 15:01 +0000 [88e9d05ef7] Richard Mudgett <rmudgett@digium.com> + + * ast_framehook_attach() must be called with the channel locked. + + The framehook container could become corrupted if the channel lock is not + held before calling. + + Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438 + +2016-08-24 14:42 +0000 [c9e83f6d0b] gtjoseph <gjoseph@digium.com> + + * res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options + + ast_multicast_rtp_create_options now checks for NULL or empty options + + Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362 + +2016-08-19 18:19 +0000 [cb8fd610e2] Corey Farrell <git@cfware.com> + + * Fix checks for allocation debugging. + + MALLOC_DEBUG should not be used to check if debugging is actually + enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only + indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it + is active. + + Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53 + +2016-08-10 15:14 +0000 [b8b5d52b5e] Mark Michelson <mmichelson@digium.com> + + * ConfBridge: Rework announcer channel methodology + + NOTE: This patch was submitted earlier and reverted because of a failing + test. The test has been patched so that it adjusts for the changes here, + so this is being resubmitted for review. + + One feature that confbridge has is the ability to play sounds to all + participants in the conference. Prior to this commit, the algorithm for + this was as follows: + + * Grab the playback lock + * Push the conference announcer channel into the bridge + * Play back the sound + * Pull the conference announcer channel from the bridge + * Release the playback lock + + The issue here is that the act of adding the playback channel to the + bridge and removing it for each announcement is expensive. Amongst the + expenses: + + * The announcer channel is imparted into the bridge, meaning a new + thread is spun up for each playback. + * When the announcer is added or removed from the bridge, it results + in the BRIDGEPEER channel variable being set on all channels in the + bridge. This requires keeping the bridge locked and locking each + individual channel in order to set it. + * There's also just the general overhead of adding the channel and + removing it from the bridge. The bridge potentially has to reconfigure + every single time + + With this commit, the paradigm for playing back announcements has + shifted. + + * The announcer channel is now added to the bridge when the conference + is allocated, and it is hung up when the conference is destroyed. + * A taskprocessor is used to queue playbacks onto the announcer channel. + This keeps the behavior from before where playbacks do not overlap. + * The announcer channel is no longer placed into the bridge as + departable. Since we are not constantly removing the channel from + the bridge, it is safe to add the channel using an independent thread + and simply hang the channel up when it is time for the conference to + be destroyed. + + The use of the taskprocessor for playbacks opens up the interesting + possibility of having asynchronous announcements played. In this commit, + however, the behavior is still exactly the same as it previously was. + + ASTERISK-26289 + Reported by Mark Michelson + + Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0 + +2016-08-23 05:54 +0000 [d5d7cbfcfb] Joshua Colp <jcolp@digium.com> + + * Revert "ConfBridge: Rework announcer channel methodology" + + This reverts commit 0cdeb2bfb0f4203384c08858951af3c77be8b9b3. + + Change-Id: I18ba73b6d4dc0b994f4ffb01ae0b6cfad36ac636 + +2016-08-22 17:08 +0000 [c16ef02318] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Default endpoints to the "offline" status. + + A recent change attempted to optimize startup by not updating contact + status. Instead, code responsible for qualifying contacts updates the + status as it becomes known. The code even accounts for contacts/AORs + that are not set to be qualified. + + The problem, though, is when there are no contacts associated with an + endpoint. A common case is when an endpoint is set to register its + contacts but has not done so yet. In this case, prior to registration, + the endpoint's device state will appear to be "not in use" and hints + associated with that device will appear to be "idle". In actuality, the + device state and hint should both appear as "unavailable". The reason + for the failure is that the optimization change made all persistent + endpoint states set to "unknown". + + The fix here is to change the hard-coded "unknown" to be "offline" + instead. The default state will be offline until the qualifying code + determines that the contact is actually online. This way, if there are + no contacts at all, then the state stays as offline, and device state + and hints appear correctly. + + ASTERISK-26269 #close + Reported by nappsoft + + Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a + +2016-08-20 14:51 +0000 [e54dcf4fd5] David M. Lee <dlee@respoke.io> + + * res_odbc_transaction: add dep on generic_odbc + + When res_odbc_transaction depended on res_odbc, it got the generic_odbc + headers and libs implicitly. Now that it no longer depends on res_odbc, + its dependency on generic_odbc must be explicit. + + Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911 + +2016-08-20 11:18 +0000 [be38c95def] Alexander Traud <pabstraud@compuserve.com> + + * pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations. + + PJProject supports a lot of platforms even Windows, some with different defaults + when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS, + "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows. + + Because of this, if configured with just an IPv6 address/transport, PJProject + listens to both IPv4 and IPv6. However, this is not supported by the PJProject + team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP, + incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only + server which accepts incoming connections on IPv4. + + If you try to configure two transports, one with IPv4 and one with IPv6 on the + same interface, as expected by the PJProject team, the IPv4 transport is not + able to bind because the IPv6 transport listens to both already. + + One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide. + Then, you are able to configure two transports, one for each IP version on the + same interface. That way, you get a server which works with IPv4 clients and + IPv6 clients at the same time over the same interface. + + Here, this change sets this parameter directly within PJProject to match the + expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack + servers out of the box like in chan_sip. This change was accepted by the + PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected + to arrive in the next version, PJProject 2.6.0. Until then, this change is + incorporated in the bundled PJProject of Asterisk. + + ASTERISK-26309 + + Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae + +2016-08-10 15:14 +0000 [0cdeb2bfb0] Mark Michelson <mmichelson@digium.com> + + * ConfBridge: Rework announcer channel methodology + + One feature that confbridge has is the ability to play sounds to all + participants in the conference. Prior to this commit, the algorithm for + this was as follows: + + * Grab the playback lock + * Push the conference announcer channel into the bridge + * Play back the sound + * Pull the conference announcer channel from the bridge + * Release the playback lock + + The issue here is that the act of adding the playback channel to the + bridge and removing it for each announcement is expensive. Amongst the + expenses: + + * The announcer channel is imparted into the bridge, meaning a new + thread is spun up for each playback. + * When the announcer is added or removed from the bridge, it results + in the BRIDGEPEER channel variable being set on all channels in the + bridge. This requires keeping the bridge locked and locking each + individual channel in order to set it. + * There's also just the general overhead of adding the channel and + removing it from the bridge. The bridge potentially has to reconfigure + every single time + + With this commit, the paradigm for playing back announcements has + shifted. + + * The announcer channel is now added to the bridge when the conference + is allocated, and it is hung up when the conference is destroyed. + * A taskprocessor is used to queue playbacks onto the announcer channel. + This keeps the behavior from before where playbacks do not overlap. + * The announcer channel is no longer placed into the bridge as + departable. Since we are not constantly removing the channel from + the bridge, it is safe to add the channel using an independent thread + and simply hang the channel up when it is time for the conference to + be destroyed. + + The use of the taskprocessor for playbacks opens up the interesting + possibility of having asynchronous announcements played. In this commit, + however, the behavior is still exactly the same as it previously was. + + ASTERISK-26289 + Reported by Mark Michelson + + Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5 + +2016-08-19 10:21 +0000 [b494b9f88c] Alexei Gradinari <alex2grad@gmail.com> + + * compilation failed with -Werror=maybe-uninitialized + + The compilation failed for devmode + --enable DONT_OPTIMIZE + --enable BETTER_BACKTRACES + --enable DO_CRASH + --enable TEST_FRAMEWORK + + res_pjsip/pjsip_configuration.c: In function dtls_handler: + res_pjsip/pjsip_configuration.c:974:20: error: + back may be used uninitialized in this function [-Werror=maybe-uninitialized] + int size = strlen(front); + ^ + cc1: all warnings being treated as errors + + Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580 + +2016-08-19 03:59 +0000 [a628009eb9] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Add cert_file. + + When using the migration script sip_to_pjsip.py, cert_file was not migrated to + pjsip.conf. A previous change regarding this contained a copy/paste error. + + ASTERISK-22374 + + Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b + +2016-08-18 09:21 +0000 [b1fe070d0b] Alexander Traud <pabstraud@compuserve.com> + + * sip.conf: tlsclientmethod is using sslv23 as default. + + When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL + SSLv23_method. This was documented incorrectly in the file sip.conf.sample. + + SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method + enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that + function should have been called 'secure_method' or 'automatic_method' back in + the 90s. + + Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if + you face a server which has problems like not falling back to TLSv1.0 + automatically. + + ASTERISK-24425 + + Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3 + +2016-08-18 17:16 +0000 [ff2378c735] Kevin Harwell <kharwell@digium.com> + + * rest-api: Swagger scripts were not replacing format variable in file brief + + Given resource paths did not have 'json' substituted in for the '{format}'. For + some auto generated documentation/comment strings it resulted in something like + the following: + + "... REST handler for /api-docs/sounds.{format}" + + This patch makes sure the resource api's path is properly substituted. + + ASTERISK-25472 #close + + Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23 + +2016-08-16 15:57 +0000 [43f400ef95] Jason Parker (license 4993) + + * res_format_attr_g729: Add annexb=no format parameter to SDPs + + Historically, Asterisk has always specified annexb=no for the g729 format. + However, when using res_pjsip no format attribute was specified. This patch + makes it so the SDP now contains a format attribute line with annexb=no. + + Note, that this means only g729a is negotiated. Even for pass through support. + According to rfc7261 the type of annex used (a or b) is dependent upon the + answerer. However, Asterisk being a back to back user agent makes this tricky + to support at this time, thus we only allow annex 'a' for now. + + ASTERISK-26228 #close + patches: + res_format_attr_g729.c submitted by Jason Parker (license 4993) + + Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0 + +2016-08-18 15:15 +0000 [4c1ae07d51] gtjoseph <gjoseph@digium.com> + + * res_odbc: Correct the dependency relationship with res_odbc_transaction + + The MODULEINFO dependencies between these 2 modules was reversed. + res_odbc should depend on res_odbc_transaction, not the other way + around. + + ASTERISK-25984 #close + + Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f + +2016-08-18 12:04 +0000 [cab6975b02] Kevin Harwell <kharwell@digium.com> + + * sip_to_pjsip: Set correct tls transport method + + A recent update had a copy/paste error where the unused variable 'val' was + being passed to the set_value function instead of the 'method' value itself. + + This patch passes in the right variable. + + ASTERISK-22374 + + Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06 + +2016-08-18 08:19 +0000 [2381ddde63] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Map the TLS method correctly. + + When using the migration script sip_to_pjsip.py and tlsclientmethod is not set + in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to + overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is + offering/using not just TLSv1.0 but TLSv1.2 as well. + + ASTERISK-22374 + + Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f + +2016-08-18 08:17 +0000 [6500f5e138] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent. + + When using the migration script sip_to_pjsip.py, no section of type=system or + type=general were created. Therefore the keys compactheaders, timerb, timert1, + and useragent were not migrated to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1 + +2016-08-18 08:16 +0000 [21e9c69e56] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Map (session-)timers correctly. + + When using the migration script sip_to_pjsip.py, session-timers=accept and + session-timers=refuse were mapped to wrong values. + + ASTERISK-22374 + + Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092 + +2016-08-18 08:15 +0000 [c9a97398f7] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Write username even without authname. + + When using the migration script sip_to_pjsip.py, now the (mandatory) username is + written to pjsip.conf, even if there was no (optional) authname in the register + string in sip.conf. + + ASTERISK-22374 + + Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f + +2016-08-18 08:14 +0000 [60275359bc] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Parse register even with transport. + + When using the migration script sip_to_pjsip.py and the register string + started with a transport in sip.conf - like tls://... - register was not parsed + correctly and therefore not migrated correctly to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2 + +2016-08-18 08:13 +0000 [0d479232eb] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit. + + When using the migration script sip_to_pjsip.py, those keys got missing. These + keys might appear several times and the function "merge_value" tried to collect + those. However, because these keys have different names in sip.conf and + pjsip.conf, "merge_value" was not able to find the new key name in sip.conf. + This change lets "merge_value" search with the old key name in sip.conf and + write with the new key name in pjsip.conf. + + ASTERISK-22374 + + Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2 + +2016-08-18 08:11 +0000 [cbc1b2d020] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Map externhost/ip to Transports. + + When using the migration script sip_to_pjsip.py, the externhost or externip of + sip.conf were erroneously written to Endpoints instead to Transports. + + ASTERISK-22374 + + Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4 + +2016-08-18 08:04 +0000 [5f33e99534] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry. + + When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and + minexpiry were not migrated to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b + +2016-08-18 08:03 +0000 [231ea0350d] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Write media_encryption. + + When using the migration script sip_to_pjsip.py, encryption=yes got missing and + media_encryption=sdes was not written to pjsip.conf, because of a typo. + + ASTERISK-22374 + + Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05 + +2016-08-18 08:02 +0000 [23eb065121] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Write cos and tos. + + When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got + missed, because of a typo. Therefore, cos and tos were not written to + pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused + by a copy-and-paste error. + + ASTERISK-22374 + + Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2 + +2016-08-18 07:55 +0000 [0b675a208b] Alexander Traud <pabstraud@compuserve.com> + + * sip_to_pjsip: Add cert_file and ca_list_path. + + When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were + not migrated to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825 + +2016-08-17 14:13 +0000 [1cd12d73a6] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Fix unbound srv failover tests. + + Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover + functionality if a TCP connection gets disconnected. Under these + conditions, session_inv_on_state_changed() gets a + PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new + transport. Unfortunately, session_inv_on_tsx_state_changed() also gets + the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates + the session. + + * Made session_inv_on_tsx_state_changed() complete terminating the session + on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still + PJSIP_INV_STATE_DISCONNECTED. + + ASTERISK-26305 #close + Reported by: Richard Mudgett + + Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d + +2016-08-16 15:36 +0000 [329507fe20] gtjoseph <gjoseph@digium.com> + + * res_pjsip: Add contact_user to endpoint + + contact_user, when specified on an endpoint, will override the user + portion of the Contact header on outgoing requests. + + Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4 + +2016-08-17 08:10 +0000 [6f448f32fe] Torrey Searle <torrey@voxbone.com> + + * res_ari: Add http prefix to generated docs + + updated the uri handler to include the url prefix of the http server + this enables res_ari to add it to the uris when generating docs + + Change-Id: I279335a2625261a8492206c37219698f42591c2e + +2016-08-17 06:12 +0000 [56e0aed177] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Detect ca_list_path capabilities in external PJProject. + + Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the + bundled PJProject but not in an external PJProject. Therefore, ca_list_path + could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2 + is detected again to enable ca_list_path again. + + ASTERISK-26303 #close + + Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0 + +2016-08-16 12:24 +0000 [2edcfcf1eb] gtjoseph <gjoseph@digium.com> + + * ari: Add documentation that path parameters are case-sensitive + + Added to api.wiki.mustache so that the generated object pages + have the notation in the table header as well as under each method + that has path parameters. + + ASTERISK-25492 #close + + Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf + +2016-08-15 15:29 +0000 [f4e28b3a09] Corey Farrell <git@cfware.com> + + * Refactor usage pattern of xmldoc info tag. + + This updates func_channel.c and main/message.c to use a generic xpointer + include instead of including info from each channel driver. Now the + name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in + documentation for func_channel. Setting the name attribute of info to + MessageToInfo or MessageFromInfo causes it to be included in the + MessageSend application and AMI action. + + Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea + +2016-08-04 20:00 +0000 [a8d9a53bae] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_config.c: Cleanup ao2 container usage idioms. + + Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a + +2016-08-04 15:57 +0000 [74a91b9ee5] Richard Mudgett <rmudgett@digium.com> + + * sorcery.c: Minor optimizations. + + * Remove some unused parameters from internal functions: + sorcery_wizard_create() + sorcery_wizard_update() + sorcery_wizard_delete() + + * Created the struct sorcery_observer_invocation ao2 object without a lock + since it is not needed in sorcery_observer_invocation_alloc(). + + * Cleanup generic ao2 container sorcery object id hash, sort, and cmp + functions. + + Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e + +2016-08-01 11:04 +0000 [29beb2890c] Richard Mudgett <rmudgett@digium.com> + + * sorcery.c: Tweak some container declaration formatting. + + * Tweak sorcery_object_type_alloc() formatting. + * Tweak ast_sorcery_init() formatting. + + Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944 + +2016-08-11 23:30 +0000 [9b822293bd] Corey Farrell <git@cfware.com> + + * pbx.c: Additional fixes to ast_context_remove_extension_callerid2. + + Do not check registrar of the first extension head. We should only check + the registrar when we match the priority. + + Additionally fix a couple calls to strcmp which used the input callerid + instead of the clean version ex.cidmatch. + + ASTERISK-26233 + + Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1 + +2016-08-10 14:41 +0000 [403c794684] Alexei Gradinari <alex2grad@gmail.com> + + * core: Entity ID is not set or invalid + + The Exchanging Device and Mailbox States could not working + if the Entity ID (EID) is not set manually and can't be obtained + from ethernet interface. + + This patch replaces debug message to warning + and addes missing description about option 'entityid' to + asterisk.conf.sample. + + With this patch the asterisk also: + (1) decline loading the modules which won't work without EID: + res_corosync and res_pjsip_publish_asterisk. + (2) warn if EID is empty on loading next modules: + pbx_dundi, res_xmpp + + Starting with v197 systemd/udev will automatically assign "predictable" + names for all local Ethernet interfaces. + This patch also addes some new ethernet prefixes "eno" and "ens". + + ASTERISK-26164 #close + + Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6 + +2016-06-15 17:10 +0000 [93332cb1d0] Evgeniy Tsybra <cjack@yandex.ru> + + * chan_sip: Fix lastrtprx always updated + + Packets are read regulary, when there is no data in buffer fr->frametype + is AST_FRAME_NULL. There was no check of frametype and lastrtprx always + updated and, therefore, rtptimeout did not work at all. + + ASTERISK-25270 #close + + Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d + +2016-08-15 07:17 +0000 [2735ec899a] Joshua Colp <jcolp@digium.com> + + * manager: Clarify that dialplan manipulation actions are under system class. + + ASTERISK-26246 #close + + Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657 + +2016-08-13 22:02 +0000 [f59bd47ed3] Matt Jordan <mjordan@digium.com> + + * app_dial: Improve documentation + + * Add some helpful <literal> and other embedded paragraph tags + + * Document some of the lesser known channel variables set by Dial + + * Add examples for some common Dial uses, along with some more + challenging but useful options + + Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1 + +2016-08-13 20:16 +0000 [4facaac408] Matt Jordan <mjordan@digium.com> + + * manager: Add <see-also> tags to relate interrelated events/actions together + + Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4 + +2016-08-13 20:15 +0000 [232d4fe24f] Matt Jordan <mjordan@digium.com> + + * manager: Add <see-also> tags to relate Bridge related events,actions, and apps + + Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85 + +2016-08-13 20:14 +0000 [63c0b2f7c9] Matt Jordan <mjordan@digium.com> + + * manager: Add <see-also> tags to relate AoC events and actions + + Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e + +2016-08-13 20:13 +0000 [0422667d6c] Matt Jordan <mjordan@digium.com> + + * manager: Add <see-also> tags to relate UserEvent actions/apps/events + + Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4 + +2016-08-12 15:53 +0000 [f9e734974b] Matt Jordan <mjordan@digium.com> + + * res_agi: Improve documentation + + * Groups of AGI commands that have similar functionality now reference + each other, and all reference the AGI application for ease of wiki + reference. + + * The documentation for the AGI application has been improved, in + particular noting the various AGI types and how they are invoked. + + * A warning message has been added to DeadAGI, noting that it is + deprecated. + + Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d + +2016-08-12 13:53 +0000 [781bb410d0] Matt Jordan <mjordan@digium.com> + + * manager: Add <see-also> links between related events + + This patch adds some see-also references between related AMI events. It + focuses primarily on those events that are guaranteed to come in pairs, + such as DTMFBegin/DTMFEnd, as well as those that occur during the life + cycle of an Asterisk channel, such as Newchannel/Hangup. + + Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3 + +2016-08-12 11:15 +0000 [cfd6852d39] Matt Jordan <mjordan@digium.com> + + * func_channel: Reorganize documentation + + * Following the example of the PJSIP channel driver, the channel + technology specific documentation has been moved to the respective + channel drivers that provide that functionality. This has the benefit + of locating the documentation of items with those modules that provide + it. + + * Examples of using the CHANNEL function for both standard items as well + as for PJSIP have been added. + + * The 'max_forwards' standard item has been documented. + + Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b + +2016-08-11 22:11 +0000 [cb043249b6] Corey Farrell <git@cfware.com> + + * Run mandatory cleanup when startup fails. + + Errors during startup result in an exit. These error branches should be + calling ast_run_atexit(0) to ensure mandatory cleanup is run. + + ASTERISK-26267 #close + + Change-Id: If226f2326ae2df7add20040696132214cf2bb680 + +2016-08-11 11:24 +0000 [4d5e96ab53] gtjoseph <gjoseph@digium.com> + + * res_pjsip_caller_id: Copy header name to short header name + + When compact_headers was set, we were sending a zero-length header name + for PAI and RPID because we always forced the short header name length + to 0. We did this because we cloned the header from "From" and wanted + to clear "f" from the sname. By cloning however, we bypass pjproject's + automatic logic that sets sname to name if there's no compact form of + the header, which there isn't for PAI and RPID. So now we force sname + to be the same as name right after we set name. + + res_pjsip_diversion needed the same treatment for the Diversion header. + + ASTERISK-26241 #close + + Change-Id: I633ec139630cd83809aae00336cee4a10077e467 + +2016-08-11 12:18 +0000 [143df33110] gtjoseph <gjoseph@digium.com> + + * res_pjsip: Fail global load if debug or default_from_user are empty + + If debug was specified in the global configuration but left blank, + the logger would treat it as a wildcard and log all hosts. If + default_from_user was empty, a crash would result. + + The global apply handler now checks for empty strings. + + ASTERISK-26239 #close + ASTERISK-26238 #close + + Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336 + +2016-08-01 15:07 +0000 [1fc5c90014] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip res_pjsip_mwi: Misc fixes and cleanups. + + * Eliminated RAII_VAR() usage in + ast_sip_persistent_endpoint_update_state(). + + * Added a missing allocation failure check to + persistent_endpoint_find_or_create(). + + * Made persistent_endpoint_find_or_create() create the new object without + a lock as it isn't needed. + + * Cleaned up some ao2 container allocation idioms. + + * Reordered res_pjsip_mwi.c load_module() and unload_module() + + Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8 + +2016-08-04 18:03 +0000 [73052e5732] Richard Mudgett <rmudgett@digium.com> + + * location.c: Misc fixes and cleanups. + + * Eliminated most RAII_VAR() usage. + + * Added several missing allocation failure checks. + + * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without + a lock as it is not needed. + + Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc + +2016-08-02 13:53 +0000 [9d4bd3d763] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Tweak high water checks. + + * The high water check in ast_taskprocessor_alert_set_levels() would + trigger immediately if the new high water level is zero and the queue was + empty. + + * The high water check in taskprocessor_push() was off by one. + + Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d + +2016-08-03 16:24 +0000 [e1248c3075] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Make aor named lock a mutex. + + The named aor lock was always being locked for writes so a rwlock adds no + benefit and may be slower because rwlocks are biased toward read locking. + + Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28 + +2016-07-29 17:41 +0000 [6e40334d89] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Add missing allocation failure check. + + Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28 + +2016-08-11 11:13 +0000 [a3c5488ff4] Matt Jordan <mjordan@digium.com> + + * app_queue: Prevent crash when a call is forwarded to an invalid location + + When a call forward attempt is made from a Queue member, the current + code will hang up the forwarding channel in an off-nominal condition + prior to raising the Stasis events informing the rest of Asterisk that + the call was forwarded. This will result in a slew of dreaded FRACKs, + most likely leading to a crash. + + This patch modifies the code such that we don't hang up the forwarding + channel even in an off-nominal condition until we've safely raised the + Stasis messages. + + ASTERISK-25797 #close + + Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38 + +2016-08-11 10:50 +0000 [5913929d31] Kevin Harwell <kharwell@digium.com> + + * alembic: add auth_username to endpoint's identify_by enum + + A new identify_by option was added recently, auth_username. However, this + setting was not added as an allowable choice in the database enumeration + value. + + This patch updates the current enumeration, adding in the new setting. + + ASTERISK-26268 #close + + Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8 + +2016-08-06 10:57 +0000 [1589452fdc] Alexei Gradinari <alex2grad@gmail.com> + + * pjsip: Fix deadlock with suspend taskprocessor on masquerade + + If both channels which should be masqueraded + are in the same serializer: + 1st channel will be locked waiting condition 'complete' + 2nd channel will be locked waiting condition 'suspended' + + On heavy load system a chance that both channels will be in + the same serializer 'pjsip/distibutor' is very high. + + To reproduce compile res_pjsip/pjsip_distributor.c with + DISTRIBUTOR_POOL_SIZE=1 + + Steps to reproduce: + 1. Party A calls Party B (bridged call 'AB') + 2. Party B places Party A on hold + 3. Party B calls Voicemail app (non-bridged call 'BV') + 4. Party B attended transfers Party A to voicemail using REFER. + 5. When asterisk masquerades calls 'AB' and 'BV', + a deadlock is happened. + + This patch adds a suspension indicator to the taskprocessor. + When a session suspends/unsuspends the serializer + it sets the indicator to the appropriate state. + The session checks the suspension indicator before + suspend the serializer. + + ASTERISK-26145 #close + + Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b + +2016-08-09 12:07 +0000 [f6ec94cca6] Kevin Harwell <kharwell@digium.com> + + * alembic/sqlalchemy: auto increment only allowed on a single column + + The extensions table defined two columns (id and priority) as primary key + autoincrement columns. However only one is allowed when defining the primary + key. + + This patch removes the autoincrement attribute from the priority column since + it does not need to be as such and really should not have been on there in the + first place. + + This patch also removes 'context', 'exten', and 'priority' from the primary key + index and creates a new combined unique contraint index on them. + + ASTERISK-26183 #close + + Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b + +2016-08-07 09:58 +0000 [5f815f9dba] Matt Jordan <mjordan@digium.com> + + * channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH + + This patch adds a new PJSIP specific dialplan function, + PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media + session will be refreshed via either an UPDATE or re-INVITE request. + When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, + the formats in use on a PJSIP channel can be re-negotiated and changed + dynamically after call setup. + + ASTERISK-26277 #close + + Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b + +2016-08-09 16:19 +0000 [a119bab6a6] Mark Michelson <mmichelson@digium.com> + + * res_rtp_asterisk: Cache local RTCP address. + + When an RTCP packet is sent or received, res_rtp_asterisk generates a + Stasis event that contains the RTCP report as well as the local and + remote addresses that the report pertains to. + + The addresses are determined using ast_find_ourip(). For the local + address, this will typically result in a lookup of the hostname of the + server, and then a DNS lookup of that hostname. If you do not have the + host in /etc/hosts, then this results in a full DNS lookup, which can + potentially block for some time. + + This is especially problematic when performing RTCP reads, since those + are done on the same thread responsible for reading and writing media. + + This patch addresses the issue by performing a lookup of the local + address when RTCP is allocated. We then use this cached local address + for the Stasis events when necessary. + + ASTERISK-26280 #close + Reported by Mark Michelson + + Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556 + +2016-08-08 12:53 +0000 [a06a1af0eb] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack + + The PJSIP taskprocessors could be overflowed on startup + if there are many (thousands) realtime endpoints + configured with unsolicited mwi. + The PJSIP stack could be totally unresponsive for a few minutes + after boot completed. + + This patch creates a separate PJSIP serializers pool for mwi + and makes unsolicited mwi use serializers from this pool. + This patch also adds 2 new global options to tune taskprocessor + alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. + + This patch also adds new global option 'mwi_disable_initial_unsolicited' + to disable sending unsolicited mwi to all endpoints on startup. + If disabled then unsolicited mwi will start processing + on next endpoint's contact update. + + ASTERISK-26230 #close + + Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a + +2016-08-04 10:16 +0000 [485fd27f7c] Joshua Colp <jcolp@digium.com> + + * res_pjsip_outbound_publish: Use a serializer shutdown group for unload. + + This change replaces the custom unload process for the outbound + publish module with the common serializer shutdown group. + + ASTERISK-25217 #close + + Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6 + +2016-08-03 15:39 +0000 [805f105f88] Corey Farrell <git@cfware.com> + + * Add missing checks during startup. + + This ensures startup is canceled due to allocation failures from the + following initializations. + * channel.c: ast_channels_init + * config_options.c: aco_init + + ASTERISK-26265 #close + + Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611 + +2016-07-22 16:37 +0000 [ea71bd6e3e] Alexei Gradinari <alex2grad@gmail.com> + + * app_voicemail: Add taskprocessor alert level options. + + On heavy loaded system with IMAP or DB storage, + 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. + It could happen when the IMAP or DB server dies or is unreachable. + It could happen on startup when there are many (thousands) + realtime endpoints configured with unsolicited mwi. + If the taskprocessor queue reaches the high water level + then the alert is triggered and pjsip stops processing new requests + until the queue reaches the low water level to clear the alert. + + This patch adds 2 new 'general' configuration options + to tune taskprocessor alert levels: + 'tps_queue_high' - Taskprocessor high water alert trigger level. + 'tps_queue_low' - Taskprocessor low water clear alert level + + ASTERISK-26229 #close + + Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8 + +2016-08-03 09:47 +0000 [9dc8cfabd5] Joshua Colp <jcolp@digium.com> + + * astconfigparser: Really handle case where line is simply a comment. + + The regular expression would match causing the code that handled + the line if it was merely a comment to never get executed. + + Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819 + +2016-07-23 08:51 +0000 [ad3e65433c] gtjoseph <gjoseph@digium.com> + + * asterisk.c: Add auto generation and persistence of UUID + + Upcoming features will require the generation and persistence + of a UUID. + + Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d + +2016-08-02 12:55 +0000 [efc4034d72] Kevin Harwell <kharwell@digium.com> + + * rest-api: Code out of sync with the model + + Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006 + +2016-07-29 13:13 +0000 [f6821fbaec] Mark Michelson <mmichelson@digium.com> + + * Remove SILK payload mappings from Asterisk core. + + SILK is a bit of a hog when it comes to using up our limited number of + dynamic payload types in the RTP engine. By freeing up four slots, it + allows for other codecs to potentially take the place. + + Now, codec_silk.so will dynamically use the payload slots in the RTP + engine when it loads. + + A better fix would be make RTP dynamic payload types actually + dynamic. However, at this stage of Asterisk 14 development, this is a + risky move that would be imprudent. + + Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612 + (cherry picked from commit d50895c7b04036aeaad58990089399e46db4c817) + +2016-08-01 11:08 +0000 [102d28c11a] Joshua Colp <jcolp@digium.com> + + * sorcery: Use more compatible regex for local expressions. + + This changes the use of an empty regex for both res_sorcery_config + and res_sorcery_memory to "." instead. This is a more compatible + regular expression which also works on FreeBSD. + + ASTERISK-26206 #close + + Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388 + +2016-08-02 03:08 +0000 [b78d10a2df] Alexander Traud <pabstraud@compuserve.com> + + * res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. + + ASTERISK-26256 #close + + Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058 + +2016-08-01 16:13 +0000 [1f95c011c7] gtjoseph <gjoseph@digium.com> + + * menuselect: Add an opaque "member_data" string to the acceptable xml + + Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe + +2016-07-27 09:56 +0000 [df42f64d62] David M. Lee <dlee@respoke.io> + + * Replace strdupa with more portable ast_strdupa + + The strdupa function is a GNU extension, and not widely portable. We + have an ast_strdupa function used within Asterisk which is preferred. + I pulled the definition up from menuselect.c into the menuselect.h + header file so it can be shared across menuselect. + + Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e + +2016-07-24 18:27 +0000 [56a07fbab9] gtjoseph <gjoseph@digium.com> + + * menuselect: Various menuselect enhancements + + * Add 'external' as a support level. + * Add ability for module directories to add entries to the menu + by adding members to the <module_prefix>/<module_prefix>.xml file. + * Expand the description field to 3 lines in the ncurses implementation. + * Allow the description field to wrap in the newt implementation. + * Add description field to the gtk implementation. + + Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 + (cherry picked from commit 90f445729d5d86050d9d379485ff0a99f4a006c1) + +2016-07-29 04:48 +0000 [7f9369c1b6] Joshua Colp <jcolp@digium.com> + + * astconfigparser: Handle case where line is simply a comment. + + Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5 + +2016-07-28 14:10 +0000 [57e9c66819] Corey Farrell <git@cfware.com> + + * pbx.c: Fix handling of '-' in extension name and callerid + + This adds a two strings to ast_exten. name to go with exten and + cidmatch_display to go with cidmatch. The new fields contain input used + to add the extension in the first place. The existing fields now + contain stripped input that excludes insignificant spaces and dashes. + These stripped fields should always be used for comparisons. The + unstripped fields should normally be used for display, but displaying + stripped values will not cause runtime errors. + + Note the actual string is only stored twice if it contains dashes. If + no dashes are found then both 'char *' fields point to the same memory. + So this change has a minimum effect on memory usage. + + The existing functions ast_get_extension_name and + ast_get_extension_cidmatch return unstripped values as they did before + this change. Other similar bugs likely still exist where unstripped + extensions are saved outside pbx.c then passed back in. + + ASTERISK-26233 #close + + Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f + +2016-07-27 17:17 +0000 [873fc0fda5] Richard Mudgett <rmudgett@digium.com> + + * pbx.c: Allow dangerous functions when adding a hint to dialplan. + + We can allow dangerous functions when adding a hint since altering + dialplan is itself a privileged activity. Otherwise, we could never + execute dangerous functions. + + ASTERISK-25996 #close + Reported by: Andrew Nagy + + Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba + +2016-07-21 10:36 +0000 [f00525a6f6] Alexei Gradinari <alex2grad@gmail.com> + + * pjproject: fixed a few bugs + + This patch fixes the issue in pjsip_tx_data_dec_ref() + when tx_data_destroy can be called more than once, + and checks if invalid value (e.g. NULL) is passed to. + + This patch updates array limit checks and docs + in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability(). + + Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40 + +2016-07-17 18:28 +0000 [972cee2e4c] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Update for pjproject 2.5.5 + + Add more --disable-* switches to Makefile.rules including + --disable-opus which was causing bundled pjproject to fail with + "undefined reference" errors in libasteriskpj. + + Changed PJ_ENABLE_EXTRA_CHECK to 1. + + Removed 2 obsolete patches and added a new one. + The new one was merged by Teluu on 6/27/2016. + + ASTERISK-26148 #close + + Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 + +2016-07-27 10:33 +0000 [8902a51d59] David M. Lee <dlee@respoke.io> + + * Portably sscanf tv_usec + + In a timeval, tv_usec is defined as a suseconds_t, which could be + different underlying types on different platforms. Instead of trying to + scanf directly into the timeval, scanf into a long int, then copy that + into the timeval. + + Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95 + +2016-07-27 12:36 +0000 [852e763571] Kevin Harwell <kharwell@digium.com> + + * rtp_engine: Failed assertion and wrong name given for codec + + Fixed an assert check that would trigger when the passed in value was negative. + The negative value was being cast to an unsigned value. This resulted in the + check failing. + + Also fixed another problem when loading formats in the engine. When setting the + mime type the format's name was being passed in instead of the codec's name. + + Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c + +2016-07-21 22:44 +0000 [e8c34680ca] Richard Mudgett <rmudgett@digium.com> + + * dsp.c: Add fax and DTMF detection unit tests. + + * Add fax amplitude and frequency sweep tests. + * Add DTMF amplitude and twist unit tests. + + Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7 + +2016-07-21 11:56 +0000 [c1f240b818] Richard Mudgett <rmudgett@digium.com> + + * dsp.c: Added descriptive comments to Goertzel calculations. + + * Added doxygen to describe some struct members and what is going on in + the code. + + Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d + +2016-07-13 13:48 +0000 [003a52fd62] Richard Mudgett <rmudgett@digium.com> + + * dsp.c: Fix incorrect format reference typo. + + Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896 + +2016-07-25 21:18 +0000 [4c0a0cbe02] Richard Mudgett <rmudgett@digium.com> + + * dsp.c: Correct DTMF twist dsp.conf documentation. + + Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae + +2016-07-22 04:43 +0000 [87433c2566] Joshua Colp <jcolp@digium.com> + + * astconfigparser.py: Update with realtime fixes. + + When configuring SIP URIs in the pjsip.conf file it is + necessary to escape the semicolon so the parser does not + treat it as a comment. This change allows this to work in + the astconfigparser implementation. + + A secondary bug where some data was lost if a configuration + option included a "=" in its value was also fixed. + + A bug where sections would be considered equal despite + being different has also been fixed. + + Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8 + +2016-07-28 14:32 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.11.0-rc1 Released. + +2016-07-28 09:29 +0000 [3bfaf6b172] Mark Michelson <mmichelson@digium.com> + + * Release summaries: Add summaries for 13.11.0-rc1 + +2016-07-28 09:27 +0000 [ca145e1807] Mark Michelson <mmichelson@digium.com> + + * .version: Update for 13.11.0-rc1 + +2016-07-28 09:27 +0000 [918ebf79ff] Mark Michelson <mmichelson@digium.com> + + * .lastclean: Update for 13.11.0-rc1 + +2016-07-28 09:27 +0000 [d7afc1cf9d] Mark Michelson <mmichelson@digium.com> + + * realtime: Add database scripts for 13.11.0-rc1 + +2016-07-21 22:28 +0000 [159e437e5a] Richard Mudgett <rmudgett@digium.com> + + * dsp.c: Fix erroneous fax tone detection. + + The Goertzel calculations get less accurate the lower the signal level + being worked with becomes because there is less resolution remaining. + If it is too low we can erroneously detect a tone where none really + exists. The searched for fax frequencies not only need to be so much + stronger than the background noise they must also be a minimum strength. + + * Add needed minimum threshold test to tone_detect(). + + * Set TONE_THRESHOLD to allow low volume frequency spread detection. + + ASTERISK-26237 #close + Reported by: Richard Mudgett + + Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc + +2016-07-22 14:44 +0000 [eda95236d1] Mark Michelson <mmichelson@digium.com> + + * Fix sqlalchemy error regarding identifier length. + + sqlalchemy was complaining: + + sqlalchemy.exc.IdentifierError: Identifier + 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 + characters + + This fixes the problem by changing the index name to be + "ps_contacts_qualifyfreq_exp" instead. + + ASTERISK-26227 #close + Reported by Mark Michelson + + Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9 + +2016-07-22 05:46 +0000 [66c9dfb272] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Enable Session-Timers for SIP over TCP (and TLS). + + Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that + scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables + Session-Timers for SIP over TCP (and for SIP over TLS). + + However with longer international calls via TCP, the SIP channel might break, + because all hops on the Internet route must stay online (have not a single power + outage, for example). Therefore with Session-Timers enabled (which are enabled + at default), you might see dropped calls. Consequently even with this change, + you might be better-off going for session-timers=refuse in your sip.conf. + + ASTERISK-19968 #close + + Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 + +2016-07-15 16:16 +0000 [33716106e0] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Whitespace and comment cleanup. + + Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38 + +2016-07-21 09:05 +0000 [52ab0bf258] gtjoseph <gjoseph@digium.com> + + * chan_sip: Prevent deadlock when issuing "sip show channels" + + sip_show_channels locks the dialogs container first then locks each + sip_pvt so it can spit out the details. The rest of sip dialog + processing locks the sip_pvt first then locks the dialogs container + if it needs to. Both lock in the order they need but deadlocks can + result. To fix, sip_show_channels and sip_show_channelstats have + been converted to use an iterator rather than ao2_callback. This way + the container is locked only while getting the next entry and is + unlocked when the callback is called. + + ASTERISK-23013 #close + + Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 + +2016-07-19 15:22 +0000 [5997ec7c9e] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. + + This patch removed call of pjsip_tx_data_dec_ref in send_notify + if send_request failed. + The pjsip_dlg_send_request deletes the message on error by itself. + + It seems this patch fixes next issues: + ASTERISK-26199 + ASTERISK-26166 + ASTERISK-26174 + + Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a + +2016-07-18 22:46 +0000 [7fdf7c3d4c] Corey Farrell <git@cfware.com> + + * Add conditional support for noreturn functions. + + This adds support for tagging functions with the noreturn attribute. + If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE + and DO_CRASH are enabled then failed assertions never return. This can + resolve a large number of false positives with static analyzers. + + ASTERISK-26220 #close + + Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753 + +2016-07-19 13:18 +0000 [dcb8aa8c1c] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi.c: Fix deadlock potential in fax redirection. + + The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to + deadlock if an incoming fax happens during the Playback or similar + application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + ASTERISK-26216 #close + Reported by: Richard Mudgett + + Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa + +2016-07-13 18:49 +0000 [fa91cf3eec] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix deadlock potential in fax redirection. + + The sip_read() has the potential to deadlock if an incoming fax happens + during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e + +2016-07-13 18:48 +0000 [2e1bdc3775] Richard Mudgett <rmudgett@digium.com> + + * chan_pjsip.c: Fix deadlock potential in fax redirection. + + The chan_pjsip_cng_tone_detected() has the potential to deadlock if an + incoming fax happens during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5 + +2016-07-12 17:33 +0000 [628e8c91d5] Richard Mudgett <rmudgett@digium.com> + + * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. + + The fax_detect_framehook() has the potential to deadlock if an incoming + fax happens during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + * Made only detach the framehook if we detected a fax and not on other + possible frames. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d + +2016-07-12 17:24 +0000 [676aeede36] Richard Mudgett <rmudgett@digium.com> + + * res_fax: Fix FAXOPT(faxdetect) timeout option. + + The fax detection timeout option did not work because basically the wrong + variable was checked in fax_detect_framehook(). As a result, the timer + would timeout immediately and disable fax detection. + + * Fixed ignoring negative timeout values. We'd complain and then go right + on using the negative value. + + * Fixed destroy_faxdetect() in the off-nominal case of an incomplete + object creation. + + * Added more range checking to FAXOPT(gateway) timeout parameter. + + ASTERISK-26214 #close + Reported by: Richard Mudgett + + Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 + +2016-07-18 16:16 +0000 [652130feb2] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi: Add faxdetect_timeout option. + + The new option allows the channel driver's faxdetect option to timeout on + a call after the specified number of seconds into a call. The new feature + is disabled if the timeout is set to zero. The option is disabled by + default. + + * Don't clear dsp_features after passing them to the dsp code in + my_pri_ss7_open_media(). We should still remember them especially for the + new faxdetect_timeout option. + + ASTERISK-26214 + Reported by: Richard Mudgett + + Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 + +2016-07-15 20:44 +0000 [851b1c3a17] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Add fax_detect_timeout endpoint option. + + The new endpoint option allows the PJSIP channel driver's fax_detect + endpoint option to timeout on a call after the specified number of + seconds into a call. The new feature is disabled if the timeout is set + to zero. The option is disabled by default. + + ASTERISK-26214 + Reported by: Richard Mudgett + + Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d + +2016-07-19 04:48 +0000 [021d4892cd] Alexander Traud <pabstraud@compuserve.com> + + * Makefile: Retain XML Declaration and DTD in docs. + + Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo, + the XML Declaration and DTD were overwritten by this. + + ASTERISK-26212 #close + + Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd + +2016-07-18 18:39 +0000 [c8e41d14a1] Corey Farrell <git@cfware.com> + + * Unit tests: Use AST_TEST_DEFINE in conditional code only. + + If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead + code. This places all existing unit tests into a conditional block if + they weren't already. + + ASTERISK-26211 #close + + Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686 + +2016-07-18 05:13 +0000 [e404f51b42] Alexander Traud <pabstraud@compuserve.com> + + * res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets. + + With this change, the initial RTP sequence number is randomly chosen not between + 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over + counter (ROC) synchronization is not lost for sRTP, when the very first RTP + packets get lost; see http://srtp.sourceforge.net/faq.html#Q6 + + ASTERISK-26207 #close + + Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464 + +2016-07-18 04:14 +0000 [5f24874ebb] Alexander Traud <pabstraud@compuserve.com> + + * Makefile: Suppress echoing of target 'config' again. + + ASTERISK-26038 #close + + Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f + +2016-07-14 03:25 +0000 [76d4983c15] Corey Farrell <git@cfware.com> + + * features.c: Remove unneeded adsi.h include. + + adsi.h is no longer used by features.c since parking was moved to a + module. + + Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59 + +2016-07-14 18:06 +0000 [cb58f853e1] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_mwi: remove unneeded check on endpoint's contacts. + + The function create_mwi_subscriptions_for_endpoint checks + if there is active contacts by retrieving aors and contacts. + + This function is used to create all unsolicited mwi subscriptions + on startup and is used when contact added. + + In both cases it's not necessary to check if there are contacts. + The contacts are needed when asterisk sends mwi. + + ASTERISK-26200 #close + + Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa + +2016-06-30 15:58 +0000 [28501051b4] Mark Michelson <mmichelson@digium.com> + + * Update support for SILK format. + + This commit adds scaffolding in order to support the SILK audio format + on calls. Roughly, this is what is added: + + * Cached silk formats. One for each possible sample rate. + * ast_codec structures for each possible sample rate. + * RTP payload mappings for "SILK". + + In addition, this change overhauls the res_format_attr_silk file in the + following ways: + + * The "samplerate" attribute is scrapped. That's native to the format. + * There are far more checks to ensure that attributes have been + allocated before attempting to reference them. + * We do not SDP fmtp lines for attributes set to 0. + + These changes make way to be able to install a codec_silk module and + have it actually work. It also should allow for passthrough silk calls + in Asterisk. + + Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e + +2016-07-14 07:45 +0000 [43b5f8d57b] Richard Miller (license 5685) + + * app_queue: Only remove queue member from pending when state changes. + + It is possible for a not in use state change to occur multiple + times causing a queue member to be removed from the pending call + container prematurely. + + The first not in use state change will remove the queue member + from the container. At this moment the member may be called and + placed in the pending container. After this another not in use + state change can be received which will remove it from the + container. Despite being called at this point the code will + incorrectly see that there are no pending calls to it. + + This change only removes it from the pending container if the + state has actually changed. + + ASTERISK-26133 #close + patches: + app_queue.diff submitted by Richard Miller (license 5685) + + Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0 + +2016-07-14 02:40 +0000 [a17b071e36] Corey Farrell <git@cfware.com> + + * pbx: Fix leak of timezone for time based includes. + + Create include_free to run ast_destroy_timing and ast_free, use that in + all places that freed an ast_include structure. This fixes a couple of + paths that previously did not run ast_destroy_timing. + + ASTERISK-26196 #close + + Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838 + +2016-07-13 17:45 +0000 [8cef8f35e7] Kevin Harwell <kharwell@digium.com> + + * translate: explicit format destination not properly set + + If the destination format's name differed from the codec name then the + translator's explict_dst field would be improperly set. In some circumstances + it would end up setting it to a newly created format that has the same name + as the codec when it actually needed to be the given destination codec. + + This could cause the translation path to use the wrong format. For instance, + if an endpoint had specified 'myulaw' as a format the translator could end up + using a 'ulaw' format (with whatever/default settings) instead. If the format + attribute settings differed between the two then there may unexpected results + during processing. + + This patch removes the name check when building the translation path. This + should make it always set the translator's explicit_dst to the given destination + format as long as the sample rate and types match. + + Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5 + +2016-07-08 11:46 +0000 [afbd10b0c5] Richard Mudgett <rmudgett@digium.com> + + * stasis_endpoint.c: Fix contactstatus_to_json(). + + The roundtrip_usec json member is optional. If it isn't present then + don't put it into the converted json structure where ast_json_pack() + will choke on it. + + Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0 + +2016-07-13 13:45 +0000 [2be13d62fd] Corey Farrell <git@cfware.com> + + * chan_sip: Fix reference leak in mwi_event_cb + + Cleanup the peer reference when stasis_subscription_final_message is + true. Also free peer_name even if peer exists, after reload a new + peer_name will be allocated. + + ASTERISK-26193 #close + + Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69 + +2016-06-22 07:13 +0000 [332beb27d8] Eugene Voityuk <eugene@thirdlane.com>,Alexander Traud <pabstraud@compuserve.com> + + * res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. + + Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) + support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added + for DTLS. The source code from main/tcptls.c should have been re-used to ease + security audits. Therefore, this change rolls back the change from July 2015 and + re-uses the code from July 2014. This has the additional benefits to work under + CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. + + ASTERISK-25659 #close + Reported by: StefanEng86, urbaniak, pay123 + Tested by: sarumjanuch, traud + patches: + res_rtp_asterisk.patch submitted by sarumjanuch + dtls_centos_step_1.patch submitted by traud + dtls_centos_step_2.patch submitted by traud + + Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c + +2016-07-13 11:30 +0000 [672a64bda3] Corey Farrell <git@cfware.com> + + * threadpool: Fix leak in ast_threadpool_serializer_group error path. + + ast_threadpool_serializer_group leaks a reference to ser when listener + is allocated but tps is not. Although listener takes the reference to + ser cleanup functions are not run without tps. + + ASTERISK-26191 #close + + Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585 + +2016-07-11 10:22 +0000 [fea201f7e6] Richard Mudgett <rmudgett@digium.com> + + * pjsip_options.c: Fix container operation. + + aor_observer_deleted() needs to operate on all contacts found for the + deleted AOR instead of only the first one found. This is really only a + problem if there is more than one contact for the AOR. + + Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1 + +2016-07-11 10:21 +0000 [02877b4b4f] Richard Mudgett <rmudgett@digium.com> + + * pjsip_configuration.c: Misc cleanups. + + * Fix some whitespace in various routines. + + * Rename i to iter in persistent_endpoint_update_state(). + + * Fix off-nominal copy/paste message wording in + persistent_endpoint_contact_deleted_observer() + + Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8 + +2016-07-13 08:57 +0000 [148cd1b319] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf. + + Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. + + ASTERISK-26046 #close + + Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7 + +2016-07-11 10:25 +0000 [97b4c7a5b4] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Fix statsd regression. + + The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f + patch introduced several regressions when the newly created "Updated" + state goes out for each endpoint registration refresh. + + 1) It restarted any OPTIONS RTT ping cycle. + + 2) It would interfere with a currently active ping and throw off that + ping's resulting RTT calculation. + + 3) It cleared the RTT time each time the endpoint was refreshed. + + 4) The cleared RTT time was sent out as a statsd update each time. + + 5) It created two AMI events for each update. + + * Revert the original patch and reimplement it. Now the current contact + status state is re-sent instead of the state being momentarily toggled + every time the endpoint refreshes its registration. The statsd events are + not created for the re-sent refresh because they are sent after every + OPTIONS ping. + + ASTERISK-26160 #close + Reported by: Matt Jordan + + Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1 + +2016-07-12 03:50 +0000 [3be6fa1e4b] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Allow own CFLAGS on ./configure. + + Before this change, make failed with the error + Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH + when CFLAGS were supplied to the configure script. This was introduced with + <https://reviewboard.asterisk.org/r/1852/> which disabled BUILD_NATIVE when + CFLAGS were supplied. Those who need different -march= values, please, go for + ./configure + make menuselect.makeopts or make menuselect + ./menuselect/menuselect --disable BUILD_NATIVE + + ASTERISK-25289 #close + + Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc + +2016-07-11 13:42 +0000 [5ee205d8bb] Richard Mudgett <rmudgett@digium.com> + + * ast_expr2: Fix off-nominal memory leak. + + Thanks to ibercom for pointing out a memory leak that was missed + in the earlier patch for the issue. + + ASTERISK-26119 + Reported by: Alexei Gradinari + + Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71 + +2016-07-11 10:17 +0000 [f5e9872016] Alexander Traud <pabstraud@compuserve.com> + + * install_prereq: Checkout of libSRTP 1.5.x. + + Since 5th November 2014, the master branch of libSRTP changed the prefix of + several member names and is not compatible with the source code in Asterisk + anymore. Therefore instead, this change checks out the latest version of the + libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as + backend. This makes AES-GCM and AES-IN possible. + + ASTERISK-22131 #close + + Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6 + +2016-07-10 19:08 +0000 [17efed6cf7] Joshua Colp <jcolp@digium.com> + + * func_odbc: Fix connection deadlock. + + The func_odbc module was modified to ensure that the + previous behavior of using a single database connection + was maintained. This was done by getting a single database + connection and holding on to it. With the new multiple + connection support in res_odbc this will actually starve + every other thread from getting access to the database as + it also maintains the previous behavior of having only + a single database connection. + + This change disables the func_odbc specific behavior if + the res_odbc module is running with only a single database + connection active. The connection is only kept for the + duration of the request. + + ASTERISK-26177 #close + + Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f + +2016-07-09 13:32 +0000 [06ba533bc7] Corey Farrell <git@cfware.com> + + * chan_sip: Fix reference leaks in error paths. + + * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. + * build_peer leaks peer on failure to allocate the endpoint. + + This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed + with an unref in the appropriate place. + + ASTERISK-26184 #close + + Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12 + +2016-07-07 12:41 +0000 [9d4e664f62] Corey Farrell <git@cfware.com> + + * REF_DEBUG: Prevent logging of container node objects. + + Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being + recorded to the refs log for the node being replaced. This prevents + logging of those unrefs since they would produce errors in + refcounter.py. + + ASTERISK-26181 #close + + Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4 + +2016-07-07 10:55 +0000 [e26bd15e7a] Scott Griepentrog <scott@griepentrog.com> + + * PJSIP: provide valid tcp nodelay option for reuse + + When using TCP transport with chan_pjsip, the TCP_NODELAY + option value was allocated on the stack, then passed as a + pointer to the tcp transport configuration structure, and + later re-used on subsequently created sockets when it was + no longer valid. This patch changes the allocation to be + a static. + + ASTERISK-26180 #close + Reported by: Scott Griepentrog + + Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0 + +2016-07-07 10:38 +0000 [77b0145a25] Joshua Colp <jcolp@digium.com> + + * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. + + Some T.38 implementations may send another re-invite after the initial + one which adds additional negotiation details (such as the max bitrate). + Currently this will fail when passthrough is being done in chan_sip as we + do nothing if T.38 is already active. + + Other handlers of T.38 inside of Asterisk (such as res_fax) handle this + scenario so this change adds support for it to chan_sip and res_pjsip_t38. + If a request to negotiate is received while T.38 is already enabled a + new re-INVITE is sent and negotiation is done again. + + ASTERISK-26179 #close + + Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c + +2016-07-04 16:38 +0000 [b4a9fa2c9e] Alexei Gradinari <alex2grad@gmail.com> + + * res_sorcery_realtime: fix bug when successful UPDATE is treated as failed + + If the SQL UPDATE statement changes nothing then SQLRowCount returns 0. + This value should be treated as success. + But the function sorcery_realtime_update treats it as failed. + + This bug was found using stress tests on PJSIP. + If there are 2 consecutive SIP REGISTER requests with the same contact data + during 1 second then res_pjsip_registrar adds contact location on 1st request + and tries to update contact location on 2nd. + The update fails and res_pjsip_registrar even removes correct contact location. + + The test "object_update_uncreated" was removed from test_sorcery_realtime.c + because it's now a valid situation. + + This patch also adds missing debug of extra SQL parameter. + + ASTERISK-26172 #close + + Change-Id: I05a7f3051455336c9dda29efc229decf86071303 + +2016-06-24 19:55 +0000 [1dfd3fc995] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_session: Check for presence of an active negotiator + + It is possible in a hypothetical situation for a session refresh to be + invoked on a PJSIP when the negotiatior on the INVITE session has not + yet been established. While this shouldn't occur with existing uses of + ast_sip_session_refresh, the crashes that occur due to improperly + calling PJSIP functions that expect a non-NULL negotiatior are + avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this + means that simply checking for the presence of the negotiator before + passing it to other PJSIP functions that use it is allowable. As such, + this patch adds checks for the presence of the negotiator before calling + PJSIP functions that assume it is non-NULL. + + Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d + +2015-10-19 18:55 +0000 [9dd0aeeb44] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_pubsub: Add additional debug statements + + When something very sad and wrong occurs, it's challenging sometimes to + figure out why. This patch adds some additional debug statements on + off-nominal paths to try and make debugging easier. + + Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640 + +2015-10-19 18:55 +0000 [1ec4f8dd00] Matt Jordan <mjordan@digium.com> + + * res/res_corosync: Raise a Stasis message on node join/leave events + + When res_corosync detects that a node leaves or joins, it currently is + informed of this via Corosync callbacks. However, there are a few + limitations with the information presented: + (1) While we have information that Corosync is aware of - such as the + Corosync nodeid - that information is really only useful inside of + Corosync or res_corosync. There's no way to translate a Corosync + nodeid to some other internally useful unique identifier for the + Asterisk instance that just joined or left the cluster. + (2) While res_corosync is notified of the instance joining or leaving + the cluster, it has no mechanism to inform the Asterisk core or + other modules of this event. This limits the usefulness of res_corosync + as a heartbeat mechanism for other modules. + + This patch addresses both issues. + + First, it adds the notion of a cluster discovery message both within the + Stasis message bus, as well as the binary event messages that + res_corosync uses to transmit data back and forth within the cluster. + When Asterisk joins the cluster, it sends a discovery message to the other + nodes in the cluster, which correlates the Corosync nodeid along with + the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids + to Asterisk EIDs, such that it can map changes in cluster state with the + Asterisk instance that has that nodeid. Likewise, when an Asterisk + instance receives a discovery message from a node in the cluster, it now + sends its own discovery message back to the originating node with the + local Asterisk EID. This lets Asterisk instances within the cluster + build a complete picture of the other Asterisk instances within the + cluster. + + Second, it publishes the discovery messages onto the Stasis message bus. + Said messages are published whenever a node joins or leaves the cluster. + Interested modules can subscribe for the ast_cluster_discovery_type() + message under the ast_system_topic() and be notified when changes in + cluster state occur. + + Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465 + +2016-07-04 13:54 +0000 [2c16a81dd5] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: Added "subscribe_context" to endpoint + + If specified, incoming SUBSCRIBE requests will be searched for the matching + extension in the indicated context. If no "subscribe_context" is specified, + then the "context" setting is used. + + ASTERISK-25471 #close + + Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514 + +2016-07-04 05:58 +0000 [a1bd57884d] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf. + + Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This + avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is + using AS_HELP_STRING everywhere else already. + + ASTERISK-26046 + + Change-Id: I8299faf504ceaeee3e39930c59293809e116c631 + +2016-06-30 15:17 +0000 [640fbbbe28] Richard Mudgett <rmudgett@digium.com> + + * features: Fix channel datastore access. + + Found as a result of the testsuite tests/callparking test crashing. + + Several calls to ast_get_chan_featuremap_config() and + ast_get_chan_features_xfer_config() did not lock the channel before + calling so the channel's datastore list was accessed without the lock's + protection. Apparently another thread deleted a datastore on the + channel's list while the crashing thread was walking the list. Crash at + 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result. + + * Add missing channel locks to calls that were not already protected + as the doxygen for those calls indicates. + + Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1 + +2016-06-22 17:26 +0000 [359134c8d3] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Don't send extra BYE if SDP invalid. + + When an answer SDP is invalid we were disconnecting the outgoing call and + sending two BYE requests. The first BYE was sent by PJPROJECT because of + the invalid SDP answer. The second BYE was sent by Asterisk because it + thought the canceled call was the result of the RFC5407 section 3.1.2 race + condition. + + * Made not send the BYE on a canceled session if the SDP negotiation is + incomplete because PJPROJECT has already sent a BYE for the failed + negotiation. + + ASTERISK-25772 #close + Reported by: Dmitriy Serov + + Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836 + +2016-06-27 17:19 +0000 [5fabcf2ca1] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: End call on initial invalid SDP negotiation. + + When an incoming call defers SDP negotiation and then sends us an invalid + SDP in the ACK, we need to send a BYE to disconnect the call. In this + case SDP negotiation has failed and we don't have valid media streams + negotiated. + + ASTERISK-25772 + + Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8 + +2016-06-23 15:13 +0000 [38a4e983dc] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip.c: Register PJMEDIA error code decoder. + + Registering the PJMEDIA error codes allows errors found when parsing an + incoming SDP to be easier to figure out. + + "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" + is much easier to understand than "Unknown error 220030". + + ASTERISK-25772 + + Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0 + +2016-06-27 16:56 +0000 [1952434df5] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Remove unused parameter from handle_incoming(). + + Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa + +2016-06-22 18:02 +0000 [28928ba5c4] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Add missing NULL checks when using pjsip_inv_end_session(). + + pjsip_inv_end_session() is documented as being able to return the + passed in tdata parameter set to NULL on success. + + Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047 + +2016-06-30 08:25 +0000 [43a78100c0] gtjoseph <gjoseph@digium.com> + + * configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject + + There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK + from getting set when using an external pjproject. + + ASTERISK-26099 #close + Reported-by: Ross Beer + + Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae + +2016-06-29 15:31 +0000 [99eff80e76] Matt Jordan <mjordan@digium.com> + + * hep.conf.sample: Default 'enabled' to 'no' + + Following the principle of least surprise, we should not be sending + massive numbers of PJSIP and RTCP HEP packets out into the ether to some + only-slightly-random IP address. Having 'enabled' set to 'no' in the + sample configuration file should prevent this from happening for those + who run 'make samples'. + + ASTERISK-26159 #close + + Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1 + +2016-06-29 15:09 +0000 [78960975f2] Matt Jordan <mjordan@digium.com> + + * pjproject/patches/config_site: Increase the max number of ICE candidates + + When negotiating ICE candidates with WebRTC capable endpoints, many + networks will result in a browser offering ICE candidates that exceeds + the default number of max candidates, 16. This patch bumps the max + candidates to 32, with the max checks at twice the number of candidates. + In practice, this has shown to be sufficient for browser/WebRTC + negotiation. + + Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5 + +2016-06-28 09:00 +0000 [d07c8a0504] gtjoseph <gjoseph@digium.com> + + * codecs: Fix ABI incompatibility created by adding format_name to ast_codec + + Adding format_name even to the end of ast_codec caused issued with + binary codec modules because the pointer would be garbage in asterisk + when they registered. So, the ast_codec structure was reverted and an + internal_ast_codec structure was created just for use in codec.c. A new + internal-only API was also added (__ast_codec_register_with_format) so + that codec_builtin could register codecs with the format_name in a + separate parameter rather than in the ast_codec structure. + + ASTERISK-26144 #close + Reported-by: Alexei Gradinari + + Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba + +2016-06-28 08:22 +0000 [f3d236ca7f] gtjoseph <gjoseph@digium.com> + + * BuildSystem: Fix a few issues hightlighted by gcc 6.x + + gcc 6.1.1 caught a few more issues. + Made sure the unit tests still pass for the func_env and stdtime + issues. + + ASTERISK-26157 #close + + Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e + +2016-06-28 10:33 +0000 [9d5b0934d9] Matt Jordan <mjordan@digium.com> + + * configs/basic-pbx/modules.conf: Remove 'bad' modules + + This patch removes the following modules: + - pbx_functions: It never existed. + - res_pjsip_log_forwarder: It no longer exists. + - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs + aren't going to be installing HOMER + - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't + loaded, and we aren't configured to make use of the + module + + Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5 + +2016-06-22 11:19 +0000 [1dfc286418] Joshua Colp <jcolp@digium.com> + + * siren: Add format attribute modules for Siren7 and Siren14. + + This change removes hardcoded SDP parsing and generation for + Siren7 and Siren14 from chan_sip and moves it to format attribute + modules so it can also be used by chan_pjsip. + + With this the fmtp lines for both are added with the bitrate + information. + + ASTERISK-26021 + + Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037 + +2016-06-23 04:33 +0000 [5f0a098243] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf. + + Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C + but requires ANSI C anyway. + + ASTERISK-26046 + + Change-Id: I914c014385e1862102d90fe7650621def78db02e + +2016-06-02 17:26 +0000 [b3c787d1dd] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: improve realtime performance #2 + + The patch removes updating all Endpoints' status on startup. + Instead, only non-qualified aors with static contact + and non-qualified non-expired contacts are retrieved from the realtime to + update the endpoint status to ONLINE. + The endpoint name was added to the contact object to simply find the endpoint + that created this contact. + + The status of endpoints with qualified aors will be updated by 'qualify' + functions. + + ASTERISK-26061 #close + + Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df + +2016-06-23 13:47 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.10.0-rc1 Released. + +2016-06-23 08:42 +0000 [62349ac1b4] Mark Michelson <mmichelson@digium.com> + + * Release summaries: Add summaries for 13.10.0-rc1 + +2016-06-23 08:38 +0000 [8da6ba4328] Mark Michelson <mmichelson@digium.com> + + * .version: Update for 13.10.0-rc1 + +2016-06-23 08:38 +0000 [170b85e3ae] Mark Michelson <mmichelson@digium.com> + + * .lastclean: Update for 13.10.0-rc1 + +2016-06-23 08:38 +0000 [4af7049b8f] Mark Michelson <mmichelson@digium.com> + + * realtime: Add database scripts for 13.10.0-rc1 + +2016-06-22 15:04 +0000 [3d904659ec] Corey Farrell <git@cfware.com> + + * res_fax: Fix reference leak in fax_v21_session_new. + + fax_v21_session_new created a session details object but only released + the allocation reference during error conditions. fax_session_new adds + it's own reference to details if needed so the caller is always + responsible for cleaning it's own reference. + + ASTERISK-26141 #close + + Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88 + +2016-06-22 13:41 +0000 [48db4c2159] gtjoseph <gjoseph@digium.com> + + * res_rtp_asterisk: Fix a self-comparison identified by gcc 6 + + gcc 6 caught a previously unidentified self-comparison in + ice_candidate_cmp. Fixed it and re-ordered the predicates for better + short-circuiting. + + ASTERISK-26140 #close + + Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7 + +2016-06-22 10:37 +0000 [bc69b03316] gtjoseph <gjoseph@digium.com> + + * chan_unistim: Fix memcpy in get_to_address + + A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD) + was using a pointer to a pointer as the destination of a memcpy and a + '&' instead of '*' in the sizeof. + + ASTERISK-26138 #close + + Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708 + +2016-06-20 13:18 +0000 [1b79e2deff] Mark Michelson <mmichelson@digium.com> + + * Fix Alembic upgrades. + + A non-existent constraint was being referenced in the upgrade script. + This patch corrects the problem by removing the reference. + + This patch fixes another realtime problem as well. Our Alembic scripts + store booleans as yes or no values. However, Sorcery tries to insert + "true" or "false" instead. This patch updates Sorcery to use "yes" and + "no" + + ASTERISK-26128 #close + + Change-Id: I366dbbf91418a9cb160b3ca74b0e59b5ac284bec + +2016-06-22 10:55 +0000 [e30602587c] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf. + + Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not + support the platform SVR2 from the year 1987 anymore. + + ASTERISK-26046 + + Change-Id: I28161b037feb2d29ab46ed20e785928460226c22 + +2016-06-22 10:51 +0000 [77da168e58] gtjoseph <gjoseph@digium.com> + + * test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO + + Since the file was missing the depends on pjproject, it wasn't + picking up the pjproject related include path. If there was no + system installed pjproject and pjproject-bundled was used, a compile + would fail because pjsip.h wasn't found. + + ASTERISK-26139 #close + + Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757 + +2016-06-21 06:52 +0000 [dfcd466bf0] Torrey Searle <torrey@voxbone.com> + + * res_rtp_asterisk: fix memory leak in dtls + + ensure that cert bios get freed after creating the fingerprint + + ASTERISK-26129 #close + + Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451 + +2016-06-21 17:42 +0000 [c982da0641] Richard Mudgett <rmudgett@digium.com> + + * res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro. + + Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c + +2016-06-12 11:19 +0000 [6a568bcc66] gtjoseph <gjoseph@digium.com> + + * res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription + + Occasionally under load we'll attempt to send a final NOTIFY on a + subscription that's already been terminated and a SEGV will occur + down in pjproject's evsub_destroy function. This is a result of a + race condition between all the paths that can generate a notify + and/or destroy the underlying pjproject evsub object: + + * The client can send a SUBSCRIBE with Expires: 0. + * The client can send a SUBSCRIBE/refresh. + * The subscription timer can expire. + * An extension state can change. + * An MWI event can be generated. + * The pjproject transaction timer (timer_b) can expire. + + Normally when our pubsub_on_evsub_state is called with a terminate, + we push a task to the serializer and return at which point the dialog + is unlocked. This is usually not a problem because the task runs + immediately and locks the dialog again. When the system is heavily + loaded though, there may be a delay between the unlock and relock + during which another event may occur such as the subscription timer + or timer_b expiring, an extension state change, etc. These may also + cause a terminate to be processed and if so, we could cause pjproject + to try to destroy the evsub structure twice. There's no way for us to + tell that the evsub was already destroyed and the evsub's group lock + can't tolerate this and SEGVs. + + The remedy is twofold. + + * A patch has been submitted to Teluu and added to the bundled + pjproject which adds add/decrement operations on evsub's group lock. + + * In res_pjsip_pubsub: + * configure.ac and pjproject-bundled's configure.m4 were updated + to check for the new evsub group lock APIs. + * We now add a reference to the evsub group lock when we create + the subscription and remove the reference when we clean up the + subscription. This prevents evsub from being destroyed before + we're done with it. + * A state has been added to the subscription tree structure so + termination progress can be tracked through the asyncronous tasks. + * The pubsub_on_evsub_state callback has been split so it's not doing + double duty. It now only handles the final cleanup of the + subscription tree. pubsub_on_rx_refresh now handles both client + refreshes and client terminates. It was always being called for + both anyway. + * The serialized_on_server_timeout task was removed since + serialized_pubsub_on_rx_refresh was almost identical. + * Missing state checks and ao2_cleanups were added. + * Some debug levels were adjusted to make seeing only off-nominal + things at level 1 and nominal or progress things at level 2+. + + ASTERISK-26099 #close + Reported-by: Ross Beer. + + Change-Id: I779d11802cf672a51392e62a74a1216596075ba1 + +2016-06-21 07:05 +0000 [ef97911a1c] Alexander Traud <pabstraud@compuserve.com> + + * res_rtp_asterisk: Use latest DTLS version available by underlying platform. + + Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the + underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for + WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based + cipher-suites. + + ASTERISK-26130 #close + + Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0 + +2016-06-21 10:53 +0000 [69d58a1e37] Scott Griepentrog <scott@griepentrog.com> + + * PJSIP: provide transport type with received messages + + The receipt of a SIP MESSAGE may occur over any transport including TCP + and TLS. When the message is received, the original URI is added to the + message in the field PJSIP_RECVADDR, but this is insufficient to ensure + a reply message can reach the originating endpoint. This patch adds the + PJSIP_TRANSPORT field populated with the transport type. + + ASTERISK-26132 #close + + Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e + +2016-06-21 08:01 +0000 [cbfa9f771e] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf. + + Some configure scripts used both AC_HELP_STRING and its replacement + AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were + changed to AS_HELP_STRING. + + ASTERISK-26046 + + Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f + +2016-06-20 10:29 +0000 [ba0d9e7f7a] Joshua Colp <jcolp@digium.com> + + * res_pjsip_session: Handle race condition at shutdown with timer. + + When shutting down res_pjsip_session will get unloaded before res_pjsip. + The act of unloading unregisters all the PJSIP services and sets + their module IDs to -1. In some cases it is possible for a timer to + occur after this happens which calls into res_pjsip_session. The + res_pjsip_session module can then try to get the session from the + INVITE session using the module ID. Since the module ID is now -1 + this fails. + + This change stores a copy of the module ID and uses it for the timer + callback scenario. If the module ID is -1 the callback immediately + returns but if the module ID is valid then it continues as normal. + + This works as the original ID of the module is guaranteed to still + be valid when used with the INVITE session. + + ASTERISK-26127 #close + + Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573 + +2016-06-20 12:13 +0000 [c1512f4108] Richard Mudgett <rmudgett@digium.com> + + * app_voicemail.c: Fix IMAP compile error. + + Fix compile error introduced by the patch for + ASTERISK-26045 + + Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3 + +2016-06-16 15:56 +0000 [5134a8043a] Alexei Gradinari <alex2grad@gmail.com> + + * fix: memory leaks, resource leaks, out of bounds and bugs + + ASTERISK-26119 #close + + Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c + +2016-06-13 17:40 +0000 [cfebe3b94a] Mark Michelson <mmichelson@digium.com> + + * ARI: Ensure announcer channels are destroyed. + + Announcer channels were not being destroyed because the + stasis_app_control structure that referenced them was not being + destroyed. The control structure was not being destroyed because it was + not being unlinked from its container. It was not being unlinked from + its container because the after bridge callback for the announcer + channel was not being run. The after bridge callback was not being run + because the after bridge datastore was not being removed from the + channel on destruction. The channel was not being destroyed because the + hangup that used to destroy the channel was now only reducing the + reference count to one. The reference count of the channel was only + being reduced to one because the stasis_app_control structure was + holding the final reference... + + The control structure used to not keep a reference to the channel, so + that loop described above did not happen. + + The solution is to manually remove the control structure from its + container when the playback on a bridge is complete. + + ASTERISK-26083 #close + Reported by Joshua Colp + + Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4 + +2016-06-20 08:05 +0000 [76516bd79d] Alexander Traud <pabstraud@compuserve.com> + + * http: leverage 'bindaddr' for TLS in http.conf + + The internal HTTP/WebSocket server supports both TCP and TLS, which can be + activated separately via the file http.conf. The source code intends to re-use + the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified + explicitly. This did not work because of a typo. This change resolves this typo. + + ASTERISK-26126 #close + + Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f + +2016-05-31 09:10 +0000 [89cc86fc38] Vasil Kolev <vasil.kolev@securax.org> + + * chan_sip: bigger buffers for headers, better failure mode + + Currently chan_sip can give weird messages if the contacts don't + fit in the From: or To: headers. This fix changes the from,to and + invite variables to use ast_str, allocates and deallocates them and + resizes them if needed. + + ASTERISK-26069 #close + + Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3 + +2016-05-18 17:37 +0000 [d53a36ff33] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_transport_management.c: Misc cleanups to survive shutdown. + + * In unload_module(), reordered destroying things to minimize the window + that the global transports container could be used by other threads on + shutdown. When shutting down you need to stop things in the opposite + order of creation. + + * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to + eliminate the crash potential by other threads using the container on + shutdown. + + * Made struct monitored_transport.sip_received not use + ast_atomic_fetchadd_int() since it is used as a boolean value that is only + set TRUE. It was previously incremented for every received SIP message + and could theoretically overflow. + + * In monitored_transport_state_callback(), allocated the monitored + transport object without a lock since the lock was unused. + + * In keepalive_global_loaded(), removed releasing the transports container + if the keepalive_thread could not be started. I set it up to be tried + again if the user reloads the configuration. + + Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff + +2016-01-05 19:08 +0000 [03953d8034] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip.c: Add check that timer actually got scheduled. + + Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1 + +2016-06-13 13:33 +0000 [32ab98116e] Richard Mudgett <rmudgett@digium.com> + + * res_rtp_multicast.c: Fix warning message typo. + + Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3 + +2016-02-11 18:15 +0000 [0429c53368] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Reorganize ast_sip_session_terminate(). + + Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b + +2016-06-10 12:35 +0000 [5823f279f3] Richard Mudgett <rmudgett@digium.com> + + * chan_rtp: Backport changes from master. + + * Deprecate chan_multicast_rtp. + + Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e + +2016-06-10 16:13 +0000 [dde58df318] Richard Mudgett <rmudgett@digium.com> + + * chan_rtp.c: Copy file from chan_multicast_rtp.c + + Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef + +2016-06-08 06:15 +0000 [ca38a3cbb4] Alexander Traud <pabstraud@compuserve.com> + + * core: Not the configured but granted number of possible file descriptors. + + With CLI "core show settings", simply the parameter maxfiles of the file + asterisk.conf was shown. If that parameter was not set, nothing was displayed + although the environment might have set a default number itself. Or if maxfiles + were not granted (completely), still maxfiles was shown. Now, the maximum number + of possible file descriptors in the environment is shown. + + ASTERISK-26097 + + Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b + +2016-06-07 18:45 +0000 [caf6cccc5c] Joshua Colp <jcolp@digium.com> + + * cel: Ensure only one dial status per channel exists. + + CEL wrongly assumed that a channel would only have a single dial + event on it. This is incorrect. Particularly in a queue each + call attempt to a member will result in a dial event, adding + a new dial status in CEL without removing the old one. This + would cause the container to grow with only one dial status + being removed when the channel went away. The other dial status + entries would remain leaking memory. + + This change fixes the memory leak by ensuring that only one dial + status will only ever exist for each channel. + + The behavior during the scenario where multiple events are received + has also been improved. For failure cases the first failure will + be the dial status. If an answer dial status is received, though, + it will take priority and the dial status for the channel will be + answer. + + Memory usage has also been decreased by storing the minimal + amount of information and the code has been cleaned up slightly. + + ASTERISK-25262 #close + + Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe + +2016-06-09 10:37 +0000 [715ef071a1] Mark Michelson <mmichelson@digium.com> + + * chan_pjsip: Lock channel when checking for RTP changes. + + bridge_native_rtp can call into an RTP-capable channel driver in order + for the driver to update information about who the channel is + communicating with. For SIP channel drivers, this means deactivating + RTCP and sending a reinvite so that the endpoints can communicate + directly. + + bridge_native_rtp does the right thing and has the channel locked when + calling into the channel driver. chan_pjsip can't alter session + properties in this thread, though. chan_pjsip queues a task on the + session serializer in order to update properties there. + + The problem is that this queued task was not locking the channel. This + meant that the queued task could attempt to deactivate RTCP at the same + time that the channel thread was attempting to process an incoming RTCP + packet. This could lead to a crash. + + This patch fixes the issue by locking the channel in the queued task + when altering RTP properties. + + ASTERISK-26092 #close + Reported by Niklas Larsson + + Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159 + +2016-06-09 09:20 +0000 [a99ddc6a0d] gtjoseph <gjoseph@digium.com> + + * build: Fix ast_sockaddr initialization to be more portable + + A change to glibc 2.22 changed the order of the sockadddr_storage + members which caused the places where we do an initialization of + ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those + initializers (which we shouldn't have been using anyway) have been + replaced with memsets. + + Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4 + +2016-06-08 12:26 +0000 [eabb398d71] Matt Jordan <mjordan@digium.com> + + * res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded + + A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not + loaded and does not have a configuration file. Previously when this + occurred, checks were put in to see if the configuration was loaded + successfully. While this is a good idea - and has been added to the + offending function in res_hep - the reality is res_hep_pjsip and + res_hep_rtcp have no business running if res_hep isn't also running. + + As such, this patch also adds a function to res_hep that returns whether + or not it successfully loaded. Oddly enough, ast_module_check returns + "everything is peachy" even if a module declined its load - so it cannot + be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this + function to see if they should continue to load; if it fails, they + decline their load as well. + + ASTERISK-26096 #close + + Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea + +2016-06-08 05:58 +0000 [0d84421f93] Alexander Traud <pabstraud@compuserve.com> + + * astfd: Not maximum size of a single file but maximum file descriptors. + + With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", the maximum size of a + single file was shown. Now, the maximum number of possible file descriptors is + shown. + + ASTERISK-26097 + + Change-Id: Icf98d145774b38cac144ca76d19eaef42ce659a3 + +2016-06-02 14:53 +0000 [9c5a0b814b] Timo Teräs <timo.teras@iki.fi> + + * Fix #include poll.h and sys/cdefs.h + + POSIX defines poll.h, sys/poll.h should not be used at is c-library + internal header which may or may not exist. Notable in musl it + generates warning of being incorrect. And add explict include of + sys/cdefs.h where needed. + + Change-Id: I142930df53fe7585a06b854b6faddc5301e024be + +2016-06-03 22:44 +0000 [9c35f34301] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. + + This patch fixes a race condition processing received REGISTER requests + and their retransmissions caused by REGISTER requests being processed by + two threads. The "sip_transaction Unable to register REGISTER transaction + (key exists)" message is a notable symptom of this issue. + + This issue was more likely to happen before the pjsip/distributor + serializers were created. Instead of steps one and two below placing the + REGISTER messages into the same pjsip/distributor they were placed in + random pjsip/default serializers. + + 1) REGISTER requests come in and get placed on the pjsip/distributor + serializer. + + 2) Before the first request is processed a retransmission comes in and is + placed on the same pjsip/distributor serializer. + + 3) The first request goes up the pjsip stack and is then shunted off to + the pjsip/aor/<aor> serializer. + + 4) Before the first request is completed processing in the pjsip/aor/<aor> + serializer, the second request goes up the pjsip stack and is also shunted + off to the pjsip/aor/<aor> serializer. + + 5) The first request completes processing and sends out its response. + + 6) The second request completes processing and tries to send out its + response but pjlib complains that the REGISTER transaction key already + exists. + + 7) Sadness ensues. + + * The race is eliminated by removing the pjsip/aor/<aor> serializer and + continuing the processing in the pjsip/distributor serializer. Now any + retransmissions queued in the pjsip/distributor serializer will be + processed after the first message is completely processed. + + ASTERISK-26088 #close + Reported by: Richard Mudgett + + Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a + +2016-06-03 11:35 +0000 [557333ea4c] Richard Mudgett <rmudgett@digium.com> + + * stasis: Add setting subscription congestion levels. + + Stasis subscriptions and message routers create taskprocessors to process + the event messages. API calls are needed to be able to set the congestion + levels of these taskprocessors for selected subscriptions and message + routers. + + * Updated CDR, CEL, and manager's stasis subscription congestion levels + based upon stress testing. Increased the congestion levels to reduce the + potential for bursty call setup/teardown activity from triggering the + taskprocessor overload alert. CDRs in particular need an extra high + congestion level because they can take awhile to process the stasis + messages. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: Id0a716394b4eee746dd158acc63d703902450244 + +2016-06-02 18:19 +0000 [110d772467] Richard Mudgett <rmudgett@digium.com> + + * sorcery: Add setting object type congestion levels. + + Sorcery creates taskprocessors for object types to process object observer + callbacks. An API call is needed to be able to set the congestion levels + of these taskprocessors for selected object types. + + * Updated PJSIP's contact and contact_status sorcery object type observer + default congestion levels based upon stress testing. Increased the + congestion levels to reduce the potential for bursty register/unregister + and subscribe/unsubscribe activity from triggering the taskprocessor + overload alert. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6 + +2016-06-02 16:08 +0000 [610eee2a36] Richard Mudgett <rmudgett@digium.com> + + * taskprocessors: Implement high/low water mark alerts. + + When taskprocessors get backed up, there is a good chance that we are + being overloaded and need to defer adding new work to the system. + + * Implemented a high/low water alert mechanism for modules to check if the + system is being overloaded and take appropriate action. When a + taskprocessor is created it has default congestion levels set. A + taskprocessor can later have those congestion levels altered for specific + needs if stress testing shows that the taskprocessor is a symptom of + overloading or needs to handle bursty activity without triggering an + overload alert. + + * Add CLI "core show taskprocessor" low/high water columns. + + * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was + never a good thing to use when creating a taskprocessor because of the + nature of how its references needed to be cleaned up on a partial + creation. + + * Made res_pjsip's distributor check if the taskprocessor overload alert + is active before placing a message representing brand new work onto a + distributor serializer. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I182f1be603529cd665958661c4c05ff9901825fa + +2016-05-27 17:31 +0000 [26e3492246] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session: Use distributor serializer for incoming calls. + + We must continue using the serializer that the original INVITE came in on + for the dialog. There may be retransmissions already enqueued in the + original serializer that can result in reentrancy and message sequencing + problems. + + Outgoing call legs create the pjsip/outsess/<endpoint> serializers for + their dialogs. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc + +2016-05-27 16:28 +0000 [ceb1007ed7] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. + + * Resolves potential reentrancy problems if system restarted in the middle + of subscription message transactions. + + * Fixes memory leak recreating persistent subscriptions when the + subscription resource tree could not be created. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be + +2016-05-27 12:50 +0000 [27bafc3a8b] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. + + We must continue using the serializer that the original SUBSCRIBE came in + on for the dialog. There may be retransmissions already enqueued in the + original serializer that can result in reentrancy and message sequencing + problems. The "sip_transaction Unable to register SUBSCRIBE transaction + (key exists)" message is a notable symptom of this issue. + + Outgoing subscriptions still create the pjsip/pubsub/<endpoint> + serializers for their dialogs. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0 + +2016-05-26 17:35 +0000 [16b08444da] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Consistently pick a serializer for messages. + + Incoming messages that are not part of a dialog or a recognized response + to one of our requests need to be sent to a consistent serializer. Under + load we may be queueing retransmissions before we can process the original + message. We don't need to throw these messages onto random serializers + and cause reentrancy and message sequencing problems. + + * Created a pool of pjsip/distributor serializers that get picked by + hashing the call-id and remote tag strings of the received messages. + + * Made ast_sip_destroy_distributor() destroy items in the reverse order of + creation. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I2ce769389fc060d9f379977f559026fbcb632407 + +2016-06-02 12:51 +0000 [993b769524] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Ignore messages until fully booted. + + We should not be processing any incoming messages until we are fully + booted. We may not have dialplan or other needed configuration loaded + yet. + + ASTERISK-26089 #close + Reported by: Scott Griepentrog + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264 + +2016-06-02 12:04 +0000 [321a9b128f] Joshua Colp <jcolp@digium.com> + + * res_odbc: Implement a connection pool. + + Testing has shown that our usage of UnixODBC is problematic + due to bugs within UnixODBC itself as well as the heavy weight + cost of connecting and disconnecting database connections, even + when pooling is enabled. + + For users of UnixODBC 2.3.1 and earlier crashes would occur due + to insufficient protection of the disconnect operation. This was + fixed in UnixODBC 2.3.2 and above. + + For users of UnixODBC 2.3.3 and higher a slow-down would occur + under heavy database use due to repeated connection establishment. + A regression is present where on each connection the database + configuration is cached again, with the cache growing out of + control. + + The connection pool implementation present in this change helps + to mitigate these issues by reducing how much we connect and + disconnect database connections. We also solve the issue of + crashes under UnixODBC 2.3.1 by defaulting the maximum number of + connections to 1, returning us to the previous working behavior. + For users who may have a fixed version the maximum concurrent + connection limit can be increased helping with performance. + + The connection pool works by keeping a list of active connections. + If the connection limit has not been reached a new connection is + established. If the connection limit has been reached then the + request waits until a connection becomes available before + continuing. + + ASTERISK-26074 #close + ASTERISK-26054 #close + + Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff + +2016-06-07 05:45 +0000 [c6ee4a0f44] Alexander Traud <pabstraud@compuserve.com> + + * res_srtp: Instead of libSRTP use OpenSSL as random source. + + Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. + Therefore, the symbol RAND_bytes is used instead of crypto_get_random. + + ASTERISK-24436 #close + + Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96 + +2016-06-07 02:16 +0000 [d38b8e6399] Alexander Traud <pabstraud@compuserve.com> + + * BuildSystem: Avoid 'ar cru' and use 'ar cr' instead. + + In several internal library projects, the files are archived with the help of + 'ar cr'. Only the projects editline and the Objective Open H.323 stack + implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms + changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier + ignored since `D' is the default (see `U')". For consistency and to avoid this + message all projects use 'ar cr' now. + + ASTERISK-26091 #close + + Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40 + +2016-05-27 14:49 +0000 [c27c232057] gtjoseph <gjoseph@digium.com> + + * ari/resource_channels: Add 'formats' to channel create/originate + + If you create a local channel and don't specify an originator channel + to take capabilities from, we automatically add all audio formats to + the new channel's capabilities. When we try to make the channel + compatible with another, the "best format" functions pick the best + format available, which in this case will be slin192. While this is + great for preserving quality, it's the worst for performance and + overkill for the vast majority of applications. + + In the absense of any other information, adding all formats is the + correct thing to do and it's not always possible to supply an + originator so a new parameter 'formats' has been added to the channel + create/originate functions. It's just a comma separated list of formats + to make availalble for the channel. Example: "ulaw,slin,slin16". + 'formats' and 'originator' are mutually exclusive. + + To facilitate determination of format names, the format name has been + added to "core show codecs". + + ASTERISK-26070 #close + + Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b + +2016-06-02 04:59 +0000 [cda3385409] Joshua Colp <jcolp@digium.com> + + * alembic: Fix migration. + + The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting + to use UniqueConstraint and failing. It was not imported and after + importing it also continued to fail. + + I've changed the script to use the explicit name of the constraint + instead. + + Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9 + +2016-06-01 13:57 +0000 [e2132dd358] Richard Mudgett <rmudgett@digium.com> + + * logging,cdr,cel: Fix stringfield memory leak. + + The stringfields refactor to allow adding stringfields to the end of a + structure (f6f4cf459f43f072604927209b39646f84aaa2e2) exposed some + incomplete cleanup code by some stringfield users. + + The most noticeable leaker is the logging system where there is a leak for + every log message generated. + + ASTERISK-26078 #close + Reported by: Etienne Lessard + Patches: + jira_asterisk_26078_v13.patch (license #5621) patch uploaded + by Richard Mudgett + + Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782 + +2016-05-25 10:34 +0000 [2de58c6d01] Alexei Gradinari <alex2grad@gmail.com> + + * core/dial: New channel variable FORWARDERNAME + + Added a new channel variable FORWARDERNAME which indicates which + channel was responsible for a forwarding requests received on dial attempt. + + Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. + + ASTERISK-26059 #close + + Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2 + +2016-05-31 13:02 +0000 [b2ce0e354b] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Use correct rdata info access method (Part 2). + + The pjproject doxygen for rdata->msg_info.info says to call + pjsip_rx_data_get_info() instead of accessing the struct member directly. + You need to call the function mostly because the function will generate + the struct member value if it is not already setup. + + Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799 + +2016-05-30 19:27 +0000 [fe305ccf01] gtjoseph <gjoseph@digium.com> + + * res_pjsip_mwi_body_generator: Re-order the body items + + Re-ordered the body items so Message-Account is second. + + Messages-Waiting: no + Message-Account: sip:1571@<IP Removed>:5060 + Voice-Message: 0/0 (0/0) + + ASTERISK-26065 #close + Reported-by: Ross Beer + + Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3 + +2016-05-30 10:58 +0000 [e8abfdcdc5] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Move to pjproject 2.5 + + Although all the patches we had against 2.4.5 were applied by Teluu, + a new bug was introduced preventing re-use of tcp and tls transports + This patch removes all the previous patches against 2.4.5, updates + the version to 2.5, and adds a new patch to correct the transport + re-use problem. + + Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068 + +2016-05-27 12:25 +0000 [37d039fdf3] Rusty Newton <rnewton@digium.com> + + * res_pjsip: Add clarifying documentation to PJSIP_HEADER help text + + Added notes about when you can read or write headers. Specifically + about being able to read on the inbound channel and write on an + outbound channel. + + ASTERISK-26063 #close + Reported by: Private Name + Tested by: Rusty Newton + + Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5 + +2016-05-25 18:30 +0000 [03d5b3ce5c] Richard Mudgett <rmudgett@digium.com> + + * pjsip_distributor.c: Use correct rdata info access method. + + The pjproject doxygen for rdata->msg_info.info says to call + pjsip_rx_data_get_info() instead of accessing the struct member directly. + You need to call the function mostly because the function will generate + the struct member value if it is not already setup. + + Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2 + +2016-05-20 13:56 +0000 [859bbec09b] Alexei Gradinari <alex2grad@gmail.com> + + * app_voicemail: fix bugs, imap mm_status log change to debug + + Fixed some bugs: + - create dirpath when save downloading message from IMAP storage. + - create IMAP folder if not exists when saving to IMAP storage + - check if file successfully opened before write to it + - some IMAP checks + - remove non-standard flag 'Unseen' + etc + + Change to debug IMAP mm_status log instead of verbose. + + Remove unused X-Asterisk-VM-Caller-channel message header + for security reason. The clients should not know name of peer/endpoint. + + ASTERISK-26045 #close + + Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b + +2016-05-19 14:56 +0000 [230686f4ec] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: add "via_addr", "via_port", "call_id" to contact + + As res_pjsip_nat rewrites contact's address, only the last Via header + can contain the source address of registered endpoint. + Also Call-Id header may contain the source address of registered + endpoint. + + Added "via_addr", "via_port", "call_id" to contact. + Added new fields ViaAddress, CallID to AMI event ContactStatus. + + ASTERISK-26011 + + Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576 + +2016-05-24 16:56 +0000 [04c12561a7] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: chatty verbose messages + + There are a lot of verbose messages about Endpoint and Contact status + changes if there are many dynamic endpoints. + The patch sets verbose level 2 for Endpoint status changes + and verbose level 3 for Contact status changes. + + ASTERISK-26055 #close + + Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7 + +2016-02-12 09:59 +0000 [a42bea3314] Corey Farrell <git@cfware.com> + + * threadpool: Fix potential data race. + + worker_start checked for ZOMBIE status without holding a lock. All + other read/write of worker status are performed with a lock, so this + check should do the same. + + ASTERISK-25777 #close + + Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781 + +2016-05-18 10:58 +0000 [a32616d60c] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * Makefile: remove OSARCH check for init install + + There are more specific checks for the platform. + + Specifically this allows installing OS/X init scripts. + + ASTERISK-26038 #close + + Change-Id: If08933621145b10362a0cfe73c079301d9c13f50 + Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> + +2016-05-21 05:42 +0000 [9ddaab789e] Jesper (License 5518) + + * func_curl: Don't trim response text on non-ASCII characters + + The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of + a signed comparison. + + ASTERISK-25669 #close + Reported by: Jesper + patches: + strings.curl.trim.patch submitted by Jesper (License 5518) + + Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a + +2016-05-20 16:59 +0000 [9453d1187a] Richard Mudgett <rmudgett@digium.com> + + * parking.h: Update ast_parking_park_call() doxygen to reality. + + ASTERISK-26029 + + Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260 + +2016-05-10 14:30 +0000 [cd89501d48] Alexei Gradinari <alex2grad@gmail.com> + + * func_odbc: single database connection should be optional + + func_odbc was changed in Asterisk 13.9.0 + to make func_odbc use a single database connection per DSN + because of reported bug ASTERISK-25938 + with MySQL/MariaDB LAST_INSERT_ID(). + + This is drawback in performance when func_odbc is used + very often in dialplan. + + Single database connection should be optional. + + ASTERISK-26010 + + Change-Id: I57d990616c957dabf7597dea5d5c3148f459dfb6 + +2016-05-20 09:39 +0000 [c0b190dd9a] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Match dialogs on responses better. + + When receiving an incoming response to a dialog-starting INVITE, we were + not matching the response to the INVITE dialog. Since we had not + recorded the to-tag to the dialog structure, the PJSIP-provided method + to find the dialog did not match. + + Most of the time, this was not a problem, because there is a fall-back + that makes the response get routed to the same serializer that the + request was sent on. However, in cases where an asynchronous DNS lookup + occurs in the PJSIP core, the thread that sends the INVITE is not + actually a threadpool serializer thread. This means we are unable to + record a serializer to handle the incoming response. + + Now, imagine what happens when an INVITE is sent on a non-serialized + thread, and an error response (such as a 486) arrives. The 486 ends up + getting put on some random threadpool thread. Eventually, a hangup task + gets queued on the INVITE dialog serializer. Since the 486 is being + handled on a different thread, the hangup task can execute at the same + time that the 486 is being handled. The hangup task assumes that it is + the sole owner of the INVITE session and channel, so it ends up + potentially freeing the channel and NULLing the session's channel + pointer. The thread handling the 486 can crash as a result. + + This change has the incoming response match the INVITE transaction, and + then get the dialog from that transaction. It's the same method we had + been using for matching incoming CANCEL requests. By doing this, we get + the INVITE dialog and can ensure that the 486 response ends up being + handled by the same thread as the hangup, ensuring that the hangup runs + after the 486 has been completely handled. + + ASTERISK-25941 #close + Reported by Javier Riveros + + Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0 + +2016-05-19 11:41 +0000 [ddcf983e39] Joshua Colp <jcolp@digium.com> + + * res_sorcery_astdb: Filter fields to only the registered ones. + + This change introduces the same filtering that is done in res_sorcery_realtime + to the res_sorcery_astdb module. This allows persisted sorcery objects + that may contain unknown fields to still be read in from the AstDB + and used. This is particularly useful when switching between different + versions of Asterisk that may have introduced additional fields. + + ASTERISK-26014 #close + + Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2 + +2016-05-09 21:40 +0000 [39fedfa423] snuffy <snuffy22@gmail.com> + + * res_pjsip_empty_info: Respond to empty SIP INFO packets + + Some SBCs require responses to empty SIP INFO packets + after establishing call via INVITE, if not responded to + they may drop your call after unspecified timeout of X minutes. + + They are identified by having no Content-Type, check for this + and respond with 200 - OK message. + + ASTERISK-24986 #close + Reported-by: Ilya Trikoz, Federico Santulli + + Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0 + +2016-05-18 07:54 +0000 [935e0496c4] gtjoseph <gjoseph@digium.com> + + * udptl: Don't eat sequence numbers until OK is received + + Scenario: + Local fax -> Asterisk w/ firewall -> Provider -> Remote fax + + * Local fax starts rtp call to remote fax + * Remote fax starts t38 call back to local fax. + * Local fax sends t38 no-signal to Asterisk before sending an OK. + * udptl processes the frame and increments the expected sequence number. + * chan_sip drops the frame because the call isn't up so nothing goes out + the external interface to open the port for incoming packets. + * Local fax sends OK and Asterisk sends OK to the remote fax. + * Remote fax sends t38 packets which are dropped by the firewall. + * Local fax re-sends t38 no-signal with the same sequence number. + * udptl drops the frame because it thinks it's a dup. + * Still no outgoing packets to open the firewall. + * t38 negotiation fails. + + The patch drops frames t38 received before udptl sequence processing + when the call hasn't been answered yet. The second no-signal frame + is then seen as new and is relayed out the external interface which + opens the port and allows negotiation to continue. + + ASTERISK-26034 #close + + Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 + +2016-05-17 11:14 +0000 [77e8ec162b] gtjoseph <gjoseph@digium.com> + + * chan_sip: Prevent extra Session-Expires headers from being added + + When chan_sip does a re-INVITE to refresh a session and authentication + is required, the INVITE with the Authorization header containes a + second Session-Expires header without the ";refersher=" parameter. + This is causing some proxies to return a 400. Also, when Asterisk is + the uas and the refresher, it is including the Session-Expires and + Min-SE headers in OPTIONS messages which is not allowed per RFC4028. + + This patch (based on the reporter's) Checks to see if a Session-Expires + header is already in the message before adding another one. It also + checks that the method is INVITE or UPDATE. + + ASTERISK-26030 #close + + Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9 + +2016-05-16 15:29 +0000 [3f6ef63099] gtjoseph <gjoseph@digium.com> + + * res_pjsip_outbound_registration: Clean up state when registration is deleted + + Nothing was cleaning up the registration state object when ast_sorcery_delete + was called on a registration. So, the registration was deleted from sorcery + but the state object went right on refreshing the registration (or failing + to refresh the registration) with the peer. + + * Added a 'deleted' observer on registration that removes the state object. + + ASTERISK-25964 #close + Reported-by Matt Jordan + + Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23 + +2016-05-15 19:05 +0000 [b6f9392a12] gtjoseph <gjoseph@digium.com> + + * res_pjsip: Set TCP_NODELAY on TCP transports + + Although it's perfectly legal to place multiple SIP messages in the same packet, + it can cause problems because the Linux default is to enable Path MTU Discovery + which sets the Don't Fragment bit on the packets. If adding a second message to + the packet causes the MTU to be exceeded, and the destination isn't equipped to + send a FRAGMENTATION NEEDED response to a large packet, the packet will just be + dropped. + + We can't specifically tell the stack to send only 1 message per packet, but we + can turn on TCP_NODELAY when we create the transport. This will at least tell + the stack to send packets as soon as possible. + + ASTERISK-26005 #close + Reported-by: Ross Beer + + Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd + +2016-05-14 21:48 +0000 [361a16f316] Matt Jordan <mjordan@digium.com> + + * configs/samples/pjsip.conf.sample: Fix typo + + A ':' is not a valid token for starting a comment. + + Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad + +2016-05-12 07:08 +0000 [f91a7dc993] Matt Jordan <mjordan@digium.com> + + * res/res_hep_pjsip: Fix reported local IP address when bound to 'any' + + When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its + local address the 'any' address, as opposed to the IP address we + actually received the packet on. This can cause some confusion in Homer, + as it will dutifully report what we send it. + + This patch uses the PJSIP inspection routines to determine which IP + address we probably received the packet on based on the remote party's + IP address. In the event that this fails, it falls back to the IP + address natively reported by the transport. + + Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3 + +2016-05-14 12:29 +0000 [9de5cd209e] Sean Bright <sean.bright@gmail.com> + + * res_ari: Correct Location headers returned by some ARI resources + + The Location headers returned by: + + * /bridges/{bridgeId}/play + * /bridges/{bridgeId}/record + * /channels/{channelId}/play + * /channels/{channelId}/record + + Did not have the '/ari' prefix, and in the case of the 'play' resources, were + using 'playback' instead of 'playbacks.' + + Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c + +2016-05-13 11:38 +0000 [524a302974] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: Endpoint IP Access Controls + + With the old SIP module we can use IP access controls per peer. + PJSIP module missing this feature. + + This patch added next configuration Endpoint options: + "acl" - list of IP ACL section names in acl.conf + "deny" - List of IP addresses to deny access from + "permit" - List of IP addresses to permit access from + "contact_acl" - List of Contact ACL section names in acl.conf + "contact_deny" - List of Contact header addresses to deny + "contact_permit" - List of Contact header addresses to permit + + This patch also better logging failed request: + add custom message instead of "No matching endpoint found" + add SIP method to logging + + ASTERISK-25900 + + Change-Id: I456dea3909d929d413864fb347d28578415ebf02 + +2016-05-11 20:17 +0000 [89ae4466ea] Matt Jordan <mjordan@digium.com> + + * res_hep: Provide an option to pick the UUID type + + At one point in time, it seemed like a good idea to use the Asterisk + channel name as the HEP correlation UUID. In particular, it felt like + this would be a useful identifier to tie PJSIP messages and RTCP + messages together, along with whatever other data we may eventually send + to Homer. This also had the benefit of keeping the correlation UUID + channel technology agnostic. + + In practice, it isn't as useful as hoped, for two reasons: + 1) The first INVITE request received doesn't have a channel. As a + result, there is always an 'odd message out', leading it to be + potentially uncorrelated in Homer. + 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. + This causes RTCP information to be uncorrelated to the SIP message + traffic seen by those capture nodes. + + In order to support both (in case someone is trying to use res_hep_rtcp + with a non-PJSIP channel), this patch adds a new option, uuid_type, with + two valid values - 'call-id' and 'channel'. The uuid_type option is used + by a module to determine the preferred UUID type. When available, that + source of a correlation UUID is used; when not, the more readily available + source is used. + + For res_hep_pjsip: + - uuid_type = call-id: the module uses the SIP Call-ID header value + - uuid_type = channel: the module uses the channel name if available, + falling back to SIP Call-ID if not + For res_hep_rtcp: + - uuid_type = call-id: the module uses the SIP Call-ID header if the + channel type is PJSIP and we have a channel, + falling back to the Stasis event provided + channel name if not + - uuid_type = channel: the module uses the channel name + + ASTERISK-25352 #close + + Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c + +2016-05-10 02:56 +0000 [a73d79c22f] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * basic-cfg: asterisk.conf: remove [directories] + + A minimal configuration does not need to explicitly spell out the + directories. The built-in defaults will do just fine. In many cases + they are wrong. + + Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c + Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> + +2016-05-10 03:06 +0000 [1c56de9453] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * basic-cfg: asterisk.conf: defaults of options + + Note the default of remmed-out options. To clarify that those values are + not the defaults. + + Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738 + Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> + +2016-05-10 03:08 +0000 [d7af591c59] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * basic-cfg: asterisk.conf: debug level 5 spams + + Don't suggest users to use debug level 5, which spews (usually + non-useful) debug information. Reduce the suggestion to (an + arbitrarily-selected) level 2. + + Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60 + Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> + +2016-05-10 03:10 +0000 [9b7db18fc1] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * basic-cfg: asterisk.conf: don't set languages + + * No need to set language in a miniml configuration. 'en' will do just + fine. + * It would be useful to have an example of setting it to a different + language. + * Setting the documentation language explicitly is likewise not + required. Setting it to a different value is not common. At least + until there is a set of translated documentation. + + Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7 + Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> + +2016-05-10 08:17 +0000 [eec539a46e] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * followme: delete the right recorded name file + + FollowMe with the option a records the name of the caller and plays it + to the callee. However it has failed to clean up that recorded file + as it tried to delete the file name without the '.sln' extension. + + ASTERISK-26008 #close + + Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec + Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> + +2016-05-12 14:36 +0000 [02d30e171e] Mark Michelson <mmichelson@digium.com> + + * Use doubles instead of floats for conversions when comparing strings. + + In 13.9.0, there was an issue where PJSIP contacts added to an AOR would + be deleted at seemingly random times. + + One reason this was happening was because of an operation to retrieve + the contacts whose expiration time was less than or equal to the current + time. When retrieving existing contacts, the contact's expiration time + and the current time were converted from a string to a float, and those + two floats were compared. + + On some systems, including mine, this conversion was horribly off. For + instance, I could regularly see the string "1463079214" get converted + into 1463079168.000000. When switching from using a float to using a + double, the conversion was as expected. + + Why was the conversion to float off? My best guess is that the + conversion to float was attempting to store the entire value in the 23 + bit significand of the IEEE-754 floating point number. In particular, if + you take only the 23 most significant bits of 1463079214, you get the + messed up 1463079168 that we were seeing in the conversion. It likely + was possible to get a more precise value by composing the number using + an exponent, but the conversion did not work that way. With a double, + you have a 52 bit significand, allowing the entire value to fit there, + and thereby allowing an accurate conversion. + + ASTERISK-26007 #close + Reported by Greg Siemon + + Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070 + +2016-05-12 09:13 +0000 [e2df15bae9] gtjoseph <gjoseph@digium.com> + + * pjsip_distributor: Add missing newline to NOTICE + + There was a newline missing from the end of the "no matching endpoint" notice. + + Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181 + +2016-05-10 10:19 +0000 [a94a12bbf7] Sebastian Damm <damm@sipgate.de> + + * res_pjsip_outbound_registration: generate correct Contact URI for TLS + + There are two types of SIP URIs indicating a secure transport: + * sips:user@example.org + * sip:user@example.org;transport=tls + + When using a sips URI, Asterisk checks incoming INVITEs and answers from + the other side for sips URIs, and rejects the packet if there are only + sip URIs. So Asterisk should only generate a sips Contact URI if the + other side supports it. + + This patch makes Asterisk generate either a sip or sips Contact URI + depending on the format of the server URI. + + If you want a sip URI, use: + server_uri=sip:example.org\;transport=tls + + If you want a sips URI, use: + server_uri=sips:example.org + + ASTERISK-25990 #close + Reported-by: Sebastian Damm + + Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2 + +2016-05-05 16:41 +0000 [36d66a23e0] Alexei Gradinari <alex2grad@gmail.com> + + * logger: Add PID to syslog messages. + + During refactoring of this support the addition of + the PID to messages was removed. This change adds it + back in. + + ASTERISK-25538 #close + + Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36 + +2016-05-11 14:07 +0000 [37214b0bdf] Matt Jordan <mjordan@digium.com> + + * configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER + + When running on a system that does not support or use AST_UNDEFINED_SANITIZER + or AST_LEAK_SANITIZER, the configure script would incorrectly set those + constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would + cause menuselect to error out, complaining that a blank value is not a + valid option. This patch corrects the issue by setting the value to 0 if + the options that those constants enable/disable is not found. + + Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba + +2016-05-03 15:43 +0000 [49b25a0956] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches + + When reloading, or fetching realtime data, if the "apply" failed for any + numerous reasons the current state object would not be maintained. This + potentially resulted in publishes being stopped for some states/clients when + they should not have been. + + This patch makes it so the current state object is kept upon any type of reload/ + fetch failures. + + Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30 + +2016-05-03 15:31 +0000 [1b5c91b7be] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_outbound_publish: Potential crash due to off nominal path + + It was possible for the explicit publish destroy function to be called without + the pjsip client ever being initialized. This fix checks to make sure there is + a client to destroy before attempting. + + Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c + +2016-05-03 15:35 +0000 [10de553c9d] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_outbound_publishing: After unloading the library won't load again + + The same thing was happening in res_pjsip_publish_asterisk. When the library + was unloaded it did not unregister the object type from sorcery. Subsequent + loads resulted in a failed load due to the sorcery type already existing. + + Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9 + +2016-05-03 14:59 +0000 [1a833b9739] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_outbound_publish: Ref leak in off nominal callback paths + + There were a few spots where the client object's reference was being leaked in + sip_outbound_publish_callback. This patch cleans up those leaks. + + Change-Id: I485d0bc9335090f373026f77c548042e258461df + +2016-05-03 15:39 +0000 [4752ef02e0] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_outbound_publish: Won't unload if condition wait times out + + When res_pjsip_outbound_publish unloads it has to wait for all current + publishing objects to get done. However if the wait condition times out + then it does not fail the unload. This sometimes results in an infinite + loop check while unloading. This patch now fails the unload operation if + the condition times out. + + Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec + +2016-05-05 11:37 +0000 [4d063814ba] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_authenticator_digest: Don't use source port in nonce verification + + From the issue reporter: + "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of + the timestamp, the source address, the source port, a server UUID that is + calculated at startup, and the authentication realm. + + Rather than caching nonces that we create, we instead attempt to re-calculate + the nonce when receiving an incoming request with authentication. We then + compare the re-calculated nonce to the incoming nonce, and if they don't match, + then authentication has failed early. + + The problem is that it is possible, especially when using TCP, to receive two + requests from the same endpoint but have differing source ports for those + requests. Asterisk itself commonly will use different source ports for + outbound TCP requests." + + This patch removes the source port dependency when building the nonce. + + ASTERISK-25978 #close + + Change-Id: I871b5f4adce102df1c4988066283095ec509dffe + +2016-05-07 14:39 +0000 [fb6227a372] gtjoseph <gjoseph@digium.com> + + * config_transport: Tell pjproject to allow all SSL/TLS protocols + + The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. + SSL is not allowed. So, even if you specify "sslv3" for a transport method, + it's silently ignored and one of the TLS protocols is used. This was a new + behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that + we never caught. + + Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). + This tells pjproject to set the socket protocol to match the method. + + ASTERISK-26004 #close + + Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078 + +2016-05-04 02:40 +0000 [2db17a793c] Jaco Kroon <jaco@uls.co.za> + + * app_confbridge: Add a regcontext option for confbridge bridge profiles. + + This patch allows for having app_confbridge register the name of the + conference as an extension into a specific context, similar to + regcontext for chan_sip. This variant is not quite as involved as the + one in chan_sip and doesn't allow for multiple contexts or custom + extensions, you can only specify the context and the conference name + will always be used as the extension to register. + + ASTERISK-25989 #close + + Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f + +2016-05-08 20:19 +0000 [2a7130b8b0] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Check for python-dev and TEST_FRAMEWORK + + The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set. + The python bindings are now built only if TEST_FRAMEWORK is set and a + python development package is installed. + + libresample was also disabled. + + ASTERISK-25993 #close + Reported-by: Joshua Colp + + Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03 + +2016-05-04 15:16 +0000 [72eb7c8301] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: module load priority + + The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_* + and res_pjsip_registrar modules should load ASAP + to avoid "No matching endpoint found" for legitimate endpoint. + + ASTERISK-25994 + + Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b + +2016-05-04 03:17 +0000 [dd00c71aae] Chris Trobridge <christ.trobridge@ultra-aep.com> + + * config_options.c: Expand #ifdef to contain whole if statement. + + ASTERISK-25956 #close + + Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38 + +2016-05-05 15:16 +0000 [e6eb17efd9] Alexei Gradinari <alex2grad@gmail.com> + + * stasis_endpoints: Add new Status and Headers to ContactStatus + + ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. + ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. + These additions should be also in stasis_endpoints + to include in command "manager show event ContactStatus" + + Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a + +2016-05-05 05:07 +0000 [fa11f4c920] Joshua Colp <jcolp@digium.com> + + * file: Ensure nativeformats remains valid for lifetime of use. + + It is possible for the nativeformats of a channel to change + throughout its lifetime. As a result a user of it needs to either + ensure the channel is locked when accessing the formats or keep + a reference to the nativeformats themselves. + + This change fixes the file playback support so it keeps a + reference to the nativeformats when accessing things. + + ASTERISK-25998 #close + + Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 + +2016-04-15 09:32 +0000 [9c2032240e] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: improve realtime performance + + This patch modified pjsip_options to retrieve only + permament contacts for aor if the qualify_frequency is > 0 + and persisted contacts if the qualify_frequency is > 0. + + This patch also fixed a bug in res_sorcery_astdb. + res_sorcery_astdb doesn't save object data retrived from astdb. + + ASTERISK-25826 + + Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05 + +2016-05-04 16:11 +0000 [fe38d21c2a] Alexei Gradinari <alex2grad@gmail.com> + + * pjsip: Added "reg_server" to contacts (fixed alembic) + + ASTERISK-25931 + + Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549 + +2016-04-07 16:33 +0000 [7a14e669f0] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip/AMI: add contact.updated event + + With the old SIP module AMI sends PeerStatus event on every + successfully REGISTER requests, ie, on start registration, + update registration and stop registration. + + With PJSIP AMI sends ContactStatus only when status is changed. + Regarding registration: + on start registration - Created + on stop registration - Removed + but on update registration nothing + + This patch added contact.updated event. + + ASTERISK-25904 + + Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f + +2016-05-02 16:08 +0000 [06d4ac0355] Alexei Gradinari <alex2grad@gmail.com> + + * res_fax: add FAXMODE variable + + The app_fax set FAXMODE variable, but res_fax missing this feature. + This patch add FAXMODE variable which is set to either "audio" or "T38". + + ASTERISK-25980 + + Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b + +2016-05-02 16:52 +0000 [2d17fe06c5] Alexei Gradinari <alex2grad@gmail.com> + + * res_fax/t38_gateway: Peer V.21 session is created on wrong channel + + The channel and peer V.21 sessions are created on the same channel now. + The peer V.21 session should be created only on peer channel + when one of channel can handle T.38. + + Also this patch enable debug for T.38 gateway session + if global fax debug enabled. + + ASTERISK-25982 + + Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e + +2016-05-01 02:21 +0000 [a2f19d82a8] Diederik de Groot <dkgroot@talon.nl> + + * configs/basic-pbx/asterisk.conf: contains incorrect path separator + + Note: When packagers use these files (as an example) the paths are never + really used when they are split using '='. + + Note: Thirdparty applications will also have trouble parsing the file when + expecting '=>'. + + Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004 + +2016-04-30 17:52 +0000 [f39089f17c] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Various fixes discovered during testing of OSes + + For all OSes: + * Disabled third-party codecs in pjproject and added + '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the + configure options since we don't use the pjsip codec capability. + + FreeBSD: + * Added FreeBSD support to install_prereq. + * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". + * Added __progname and environ to asterisk.exports.in. + * Reverted the use of ldconfig to create shared library symlinks to ln. + * Only enable epoll in pjproject if `uname -s` is Linux. + * Added a patch to pjproject to take the name of the 'make' command from + an environment variable if supplied. This is needed for the python bindings. + (merged by Teluu into pjproject trunk 5/3/2016) + FreeBSD support isn't complete. Still some general issues regarding + make/gmake having nothing to do with pjproject. With some handholding it DOES + build successfully. + + CentOS: + Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. + CentOS 6/7 32/64 build and run the pjsip testsuite successfully. + + Ubuntu: + No changes required. + Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. + + Debian: + No changes required. + Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. + + There will utimately be a follow-up patch to create an install_prereq for + the testsuite as I've discovered a few missing requirements. + + ASTERISK-25968 #close + + Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c + +2016-03-17 14:29 +0000 [8028fc7585] Andrew Nagy <andrew.nagy@the159.com> + + * app_voicemail: always copy dynamic struct to avoid race condition + + Voicemail email addresses can be corrupt or voicemail + emails can end up being sent to the wrong email address if asterisk is + reading voicemail.conf during a reload and processing an email at the + same time. This patch always copies the struct that would otherwise only + be copied once. + + ASTERISK-24463 #close + Reported by: John Campbell + Tested by: Etienne Lessard + Tested by: Andrew Nagy + Change-Id: I3a0643813116da84e2617291903d0d489b7425fb + +2016-04-15 14:26 +0000 [3cb8934de0] Alexei Gradinari <alex2grad@gmail.com> + + * pjsip: Added "reg_server" to contacts. + + If the Asterisk system name is set in asterisk.conf, it will be stored + into the "reg_server" field in the ps_contacts table to facilitate + multi-server setups. + + ASTERISK-25931 + + Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8 + +2016-04-28 11:35 +0000 [7992923c70] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Start body generator users after suppliers. + + Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb + +2016-04-28 16:06 +0000 [5dc0e082b2] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Add useful information to some messages. + + Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a + +2016-04-26 15:58 +0000 [f9e416f053] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Fix body generator registration race. + + Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67 + +2016-04-26 15:13 +0000 [b1b2019046] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.h: Fix doxygen association. + + Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632 + +2016-04-25 16:00 +0000 [b7f07fdff5] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_publish.c: Remove redundant flag check. + + Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353 + +2016-04-28 16:54 +0000 [719ece5659] gtjoseph <gjoseph@digium.com> + + * pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE + + When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled, + the input uri string will become corrupted if it contains escape sequences. + It's not possible to automatically strdup or strdupa the input string because + the output uri pj_str_t's will have pointers to chunks of the input string. + Getting around this would require more memory management code and wouldn't + be worth the savings of doing the unescape in place. + + ASTERISK-25970 #close + Reported-by: Dmitriy Serov + + Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88 + +2016-03-07 18:34 +0000 [38bed4515d] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Add ability to identify by Authorization username + + A feature of chan_sip that service providers relied upon was the ability to + identify by the Authorization username. This is most often used when customers + have a PBX that needs to register rather than identify by IP address. From my + own experiance, this is pretty common with small businesses who otherwise + don't need a static IP. + + In this scenario, a register from the customer's PBX may succeed because From + will usually contain the PBXs account id but an INVITE will contain the caller + id. With nothing recognizable in From, the service provider's Asterisk can + never match to an endpoint and the INVITE just stays unauthorized. + + The fixes: + + A new value "auth_username" has been added to endpoint/identify_by that + will use the username and digest fields in the Authorization header + instead of username and domain in the the From header to match an endpoint, + or the To header to match an aor. This code as added to + res_pjsip_endpoint_identifier_user rather than creating a new module. + + Although identify_by was always a comma-separated list, there was only + 1 choice so order wasn't preserved. So to keep the order, a vector was added + to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar + to find the aor. The res_pjsip_endpoint_identifier_* modules are called in + globals/endpoint_identifier_order. + + Along the way, the logic in res_pjsip_registrar was corrected to match + most-specific to least-specific as res_pjsip_endpoint_identifier_user does. + + The order is: + + username@domain + username@domain_alias + username + + Auth by username does present 1 problem however, the first INVITE won't have + an Authorization header so the distributor, not finding a match on anything, + sends a securty_alert. It still sends a 401 with a challenge so the next + INVITE will have the Authorization header and presumably succeed. As a result + though, that first security alert is actually a false alarm. + + To address this, a new feature has been added to pjsip_distributor that keeps + track of unidentified requests and only sends the security alert if a + configurable number of unidentified requests come from the same IP in a + configurable amout of time. Those configuration options have been added to + the global config object. This feature is only used when auth_username + is enabled. + + Finally, default_realm was added to the globals object to replace the hard + coded "asterisk" used when an endpoint is not yet identified. + + The testsuite tests all pass but new tests are forthcoming for this new + feature. + + ASTERISK-25835 #close + Reported-by: Ross Beer + + Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d + +2016-04-27 13:23 +0000 [677d5b5151] Mark Michelson <mmichelson@digium.com> + + * func_odbc: Check connection status before executing queries. + + A recent change to func_odbc made it so that a single connection was + maintained per DSN. The problem was that the code was optimistic about + the health of the connection after initially opening it and did nothing + to re-connect in case the connection had died. + + This change adds a check before executing a query to ensure that the + connection to the database is still up and running. + + ASTERISK-25963 #close + Reported by Ross Beer + + Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d + +2016-04-15 11:59 +0000 [df3639700a] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: disable multi domain to improve realtime performace + + This patch added new global pjsip option 'disable_multi_domain'. + Disabling Multi Domain can improve Realtime performance by reducing + number of database requests. + + ASTERISK-25930 #close + + Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7 + +2016-04-26 11:13 +0000 [949bf6b282] Joshua Colp <jcolp@digium.com> + + * chan_sip: Give more time for TCP/TLS threads to stop. + + The unload process currently tells each TCP/TLS to terminate but + does not wait for them to do so. This introduces a race condition + where the container holding the threads may be destroyed before + the threads are able to remove themselves from it. When they + finally do the container is invalid and can't be used causing a + crash. + + A previous change existed which waited a bit to wait for any + stranglers to finish. This change extends this and waits longer. + + ASTERISK-25961 #close + + Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6 + +2016-04-26 05:48 +0000 [6959f5484b] Joshua Colp <jcolp@digium.com> + + * app_queue: Fix crash when unloading module. + + When unloading the app_queue module the members in each queue are + destroyed and as part of this they are removed from the pending + members container. Unfortunately a crash would occur as the container + was destroyed before the members were removed. + + This change tweaks ordering so the container destruction occurs + after the members are destroyed. + + ASTERISK-16115 + + Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b + +2016-04-24 22:51 +0000 [b38f1146e5] gtjoseph <gjoseph@digium.com> + + * config: Fix ast_config_text_file_save2 writability check for missing files + + A patch I did back in 2014 modified ast_config_text_file_save2 to check the + writability of the main file and include files before truncating and re-writing + them. An unintended side-effect of this was that if a file doesn't exist, + the check fails and the write is aborted. + + This patch causes ast_config_text_file_save2 to check the writability of the + parent directory of missing files instead of checking the file itself. This + allows missing files to be created again. A unit test was also added to + test_config to test saving of config files. + + The regression was discovered when app_voicemail's passwordlocation=spooldir + feature stopped working. + + ASTERISK-25917 #close + Reported-by: Jonathan Rose + + Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80 + +2016-04-21 23:53 +0000 [29bab0d1a4] Kirill Katsnelson <kkm@smartaction.com> + + * chan_sip: Make autocreated peers send PeerStatus events + + Since Stasis has been introduced, an attempt to send AMI messages by an + autocreated peer caused a crash, and all events from autocreated peers were + semi-inadvertently disabled altogether in 0b83761. This change restores the + disabled functionality. + + ASTERISK-25950 + + Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974 + +2016-04-21 14:23 +0000 [c345e530f4] Kevin Harwell <kharwell@digium.com> + + * app_queue: queue members can receive multiple calls + + It was possible for a queue member that is a member of at least 2 or more + queues to receive mulitiple calls at the same time. This happened because + of a race between when a member was being rung and when the device state + notified the other queue(s) member object of the state change. + + This patch makes it so when a queue member is being rung it gets added to + a global pool of queue members. If that same member is tried again, e.g. + from another queue, and it is found to already exist in the pending member + container then it will not ring that member. + + ASTERISK-16115 #close + + Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48 + +2016-04-25 08:11 +0000 [c0688a6398] Javier Acosta <javier.acosta@beeonline.es> + + * Fix case sensitive actions in AMI QueueSummary and QueueStatus + + ASTERISK-25954 #close + Reported by: Javier Acosta + + Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256 + +2016-04-22 15:25 +0000 [ebf0724a83] Richard Mudgett <rmudgett@digium.com> + + * test_message.c: Wait longer in case dialplan also processes the test message. + + Bumped the wait from 1 second to 5 seconds. The test message was hitting my + default call handler and failing the test because it took longer. + + Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba + +2016-04-12 15:29 +0000 [ba63aa7c9e] Richard Mudgett <rmudgett@digium.com> + + * Manager: Short circuit AMI message processing. + + Improve AMI message processing performance if there are no consumers + listening for the messages. We now skip creating the AMI event message + text strings. + + Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3 + +2016-04-13 17:54 +0000 [d5ee6acf28] Richard Mudgett <rmudgett@digium.com> + + * manager.c: Eliminate most RAII_VAR usage. + + * Made ast_manager_event_blob_create() not allocate the ao2 event object + with a lock as it is not needed. + + Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c + +2016-04-13 17:09 +0000 [7303e3dc96] Richard Mudgett <rmudgett@digium.com> + + * manager_channels.c: Fix allocation failure crash. + + An earlier allocation failure failed to create a channel snapshot for the + AMI HangupRequest/SoftHangupRequest event which resulted in a crash in + channel_hangup_request_cb(). Where the stasis message gets generated + cannot tell if the NULL snapshot returned was because of an allocation + failure or the channel was a dummy channel. + + * Made channel_hangup_request_cb() check if the channel blob has a + snapshot and exit if it doesn't. + + * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). + + Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24 + +2016-04-13 13:50 +0000 [1e93f3d723] Richard Mudgett <rmudgett@digium.com> + + * Bridge system: Fix memory leaks and double frees on impart failure. + + You cannot reference the passed in features struct after calling + ast_bridge_impart(). Even if the call fails. + + Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 + +2016-04-13 13:20 +0000 [5e388d4188] Richard Mudgett <rmudgett@digium.com> + + * bridge_softmix.c: Fix crash if channel fails to join mixing tech. + + softmix_bridge_join() failed because of an allocation failure. To address + this, the softmix bridge technology now checks if the channel failed to + join softmix successfully. In addition, the bridge now begins the process + of kicking the channel out of the bridge so we don't have channels + partially in the bridge for very long. + + * Fix the test_channel_feature_hooks.c unit tests. The test channel must + have a valid codec to join the simple_bridge technology. This patch makes + joining a bridge more strict by not allowing partially joined channels to + remain in the bridge. + + Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b + +2016-03-17 12:28 +0000 [9740277713] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) + + There are several places that do scheduled tasks or periodic housecleaning, + each with its own implementation: + + * res_pjsip_keepalive has a thread that sends keepalives. + * pjsip_distributor has a thread that cleans up expired unidentified requests. + * res_pjsip_registrar_expire has a thread that cleans up expired contacts. + * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. + * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. + + There are also places where we should be doing scheduled work but aren't. + A good example are the places we have sorcery observers to start registration + or qualify. These don't work when changes are made to a backend database + without a pjsip reload. We need to check periodically. + + As a first step to solving these issues, a new ast_sip_sched facility has + been created. + + ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. + When a task is ready to run, ast_sip_task_pusk is called for it. This ensures + that the task is executed in a PJLIB registered thread and doesn't hold up the + ast_sched thread so it can immediately continue processing the queue. The + serializer used by ast_sip_sched is one of your choosing or a random one from + the res_pjsip pool if you don't choose one. + + Another feature is the ability to automatically clean up the task_data when the + task expires (if ever). If it's an ao2 object, it will be dereferenced, if + it's a malloc'd object it will be freed. This is selectable when the task is + scheduled. Even if you choose to not auto dereference an ao2 task data object, + the scheduler itself maintains a reference to it while the task is under it's + control. This prevents the data from disappearing out from under the task. + + There are two scheduling models. + + AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at + the specific interval. That is, every "interval" milliseconds, regardless of + how long the task takes. If the task takes longer than the interval, it will + be scheduled at the next available multiple of interval. For exmaple: If the + task has an interval of 60 secs and the task takes 70 secs (it better not), + the next invocation will happen at 120 seconds. + + AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should + start "interval" milliseconds after the current invocation has finished. + + Also, the same ast_sched facility for fixed or variable intervals exists. The + task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or + AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. + + One res_pjsip.h housekeeping change was made. The pjsip header files were + added to the top. There have been a few cases lately where I've needed + res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because + I didn't add the pjsip header files to my source even though I never referenced + any pjsip calls. + + Finally, a few new convenience APIs were added to astobj2 to make things a + little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and + ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros + were also copied from res_phoneprov because I got tired of having to duplicate + the same hash, sort and compare functions over and over again. The + AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for + aor_container_alloc into your source. + + This facility can be used immediately for the situations where we already have + a thread that wakes up periodically or do some scheduled work. For the + registration and qualify issues, additional sorcery and schema changes would + need to be made so that we can easily detect changed objects on a periodic + basis without having to pull the entire database back to check. I'm thinking + of a last-updated timestamp on the rows but more on this later. + + Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c + +2016-04-25 21:43 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.9.0-rc1 Released. + +2016-04-25 16:42 +0000 [5237b7cd47] Joshua Colp <jcolp@digium.com> + + * Release summaries: Add summaries for 13.9.0-rc1 + +2016-04-25 16:40 +0000 [5a3850ecba] Joshua Colp <jcolp@digium.com> + + * .version: Update for 13.9.0-rc1 + +2016-04-25 16:40 +0000 [60b39040e4] Joshua Colp <jcolp@digium.com> + + * .lastclean: Update for 13.9.0-rc1 + +2016-04-25 16:40 +0000 [5078454464] Joshua Colp <jcolp@digium.com> + + * realtime: Add database scripts for 13.9.0-rc1 + +2016-04-22 17:53 +0000 [eb7c581806] gtjoseph <gjoseph@digium.com> + + * res_agi: Prevent run_agi from eating frames it shouldn't + + The run_agi function is eating control frames when it shouldn't be. This is + causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond + transfer. + + Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie + answers. + + Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE + and is left thinking he's connected to Bob. + + In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls + an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on + Charlie's channel. + + The fix was to accumulate deferrable frames in the "forever" loop instead of + dropping them, and re-queue them just before running the actual agi command + or exiting. + + ASTERISK-25951 #close + + Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645 + +2016-04-22 13:49 +0000 [068ae54c76] Mark Michelson <mmichelson@digium.com> + + * func_odbc: Use one connection per DSN. + + res_odbc was changed in Asterisk 13.8.0 to remove connection management, + opting instead to let unixodbc maintain open connections and return + those to Asterisk as requested. + + This was a boon for realtime, since it meant that multiple threads could + potentially run parallel queries since they could each be using their + own database connections. + + However, on the user-facing side, func_odbc, there were some inherent + behaviors being relied on that no longer hold true after the change. + One such reported behavior was that MySQL's LAST_INSERTED_ID() works + per-connection. This means that if Asterisk uses separate connections + for every database operation, whereas before it used one connection for + everything, we have broken expectations and functionality. + + The fix provided in this patch is to make func_odbc use a single + database connection per DSN. This way, user-facing database usage will + have the same behavior as it did pre-13.8.0. However, realtime, which is + the real workhorse of database interaction, will continue to let + unixodbc manage connections. + + ASTERISK-25938 #close + Reported by Edwin Vandamme + + Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc + +2016-04-22 13:02 +0000 [6aeefa89bc] Leif Madsen <leif@leifmadsen.com> + + * Remove reference to non-existent sip.conf option + + Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f + + ASTERISK-25927 #close + + Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8 + +2016-04-21 08:26 +0000 [e750ea9b5b] Diederik de Groot <dkgroot@talon.nl> + + * lock.c: Check *lt before dereferencing it + + *lt is NULL if t->tracking == 0 + + ASTERISK-25948 #close + + Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba + +2016-04-15 14:36 +0000 [a036c35903] Richard Mudgett <rmudgett@digium.com> + + * res_stasis: Handle re-enter stasis bridge with swap channel. + + We lose the fact that there is a swap channel if there is one. We + currently wind up rejoining the stasis bridge as a normal join after the + swap channel has already been kicked from the bridge. + + This patch preserves the swap channel so the AMI/ARI events can note that + the channel joining the bridge is swapping with another channel. Another + benefit to swaqpping in one operation is if there are any channels that + get lonely (MOH, bridge playback, and bridge record channels). The lonely + channels won't leave before the joining channel has a chance to come back + in under stasis if the swap channel is the only reason the lonely channels + are staying in the bridge. + + ASTERISK-25947 #close + Reported by: Richard Mudgett + + ASTERISK-24649 + Reported by: John Bigelow + + ASTERISK-24782 + Reported by: John Bigelow + + Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee + +2016-04-19 16:58 +0000 [9942d50aa5] Richard Mudgett <rmudgett@digium.com> + + * bridge: Hold off more than one imparting channel at a time. + + An earlier patch blocked the ast_bridge_impart() call until the channel + either entered the target bridge or it failed. Unfortuantely, if the + target bridge is stasis and the imprted channel is not a stasis channel, + stasis bounces the channel out of the bridge to come back into the bridge + as a proper stasis channel. When the channel is bounced out, that + released the block on ast_bridge_impart() to continue. If the impart was + a result of a transfer, then it became a race to see if the swap channel + would get hung up before the imparted channel could come back into the + stasis bridge. If the imparted channel won then everything is fine. If + the swap channel gets hung up first then the transfer will fail because + the swap channel is leaving the bridge. + + * Allow a chain of ast_bridge_impart()'s to happen before any are + unblocked to prevent the race condition described above. When the channel + finally joins the bridge or completely fails to join the bridge then the + ast_bridge_impart() instances are unblocked. + + ASTERISK-25947 + Reported by: Richard Mudgett + + ASTERISK-24649 + Reported by: John Bigelow + + ASTERISK-24782 + Reported by: John Bigelow + + Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 + +2016-04-19 17:52 +0000 [516c626a7d] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_callerid: Clear out display name if id->name is not valid + + When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning + the From header, then it overwrites the display name and uri from the channel's + connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was + leaving the display name from the From header in the new RPID or PAI header. + On an attended transfer where the originator had a caller id number set but not + a display name, the re-INVITE to the final transferee had the number of the + originator but the display name of the transferer. + + Added a check to clear out the display name in the new header if + connected.id.name was invalid. + + ASTERISK-25942 #close + + Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b + +2016-04-19 13:02 +0000 [ded3794fc6] Joshua Colp <jcolp@digium.com> + + * app_talkdetect: Make the module core supported. + + This module is used as part of testsuite tests to confirm + stuff works. I'm accordingly marking it as core as it is + required by those tests. + + Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88 + +2016-04-18 12:12 +0000 [efae187217] Mark Michelson <mmichelson@digium.com> + + * PJSIP: Remove PJSIP parsing functions from uri length validation. + + The PJSIP parsing functions provide a nice concise way to check the + length of a hostname in a SIP URI. The problem is that in order to use + those parsing functions, it's required to use them from a thread that + has registered with PJLib. + + On startup, when parsing AOR configuration, the permanent URI handler + may not be run from a PJLib-registered thread. Specifically, this could + happen when Asterisk was started in daemon mode rather than + console-mode. If PJProject were compiled with assertions enabled, then + this would cause Asterisk to crash on startup. + + The solution presented here is to do our own parsing of the contact URI + in order to ensure that the hostname in the URI is not too long. The + parsing does not attempt to perform a full SIP URI parse/validation, + since the hostname in the URI is what is important. + + ASTERISK-25928 #close + Reported by Joshua Colp + + Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60 + +2016-04-18 17:00 +0000 [f436b9ab11] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_registrar: Fix bad memory-ness with user_agent. + + Recent changes to the PJSIP registrar resulted in tests failing due to + missing AOR_CONTACT_ADDED test events. The reason for this was that the + user_agent string had junk values in it, resulting in being unable to + generate the event. + + I'm going to be honest here, I have no idea why this was happening. Here + are the steps needed for the user_agent variable to get messed up: + * REGISTER is received + * First contact in the REGISTER results in a contact being removed + * Second contact in the REGISTER results in a contact being added + * The contact, AOR, expiration, and user agent all have to be passed as + format parameters to the creation of a string. Any subset of those + parameters would not be enough to cause the problem. + + Looking into what was happening, the thing that struck me as odd was + that the user_agent variable was meant to be set to the value of the + User-Agent SIP header in the incoming REGISTER. However, when removing a + contact, the user_agent variable would be set (via ast_strdupa inside a + loop) to the stored contact's user_agent. This means that the + user_agent's value would be incorrect when attempting to process further + contacts in the incoming REGISTER. + + The fix here is to use a different variable for the stored user agent + when removing a contact. Correcting the behavior to be correct also + means the memory usage is less weird, and the issue no longer occurs. + + ASTERISK-25929 #close + Reported by Joshua Colp + + Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08 + +2016-04-18 13:41 +0000 [49bfdc9ac0] Joshua Colp <jcolp@digium.com> + + * res_pjsip_transport_management: Allow unload to occur. + + At shutdown it is possible for modules to be unloaded that wouldn't + normally be unloaded. This allows the environment to be cleaned up. + + The res_pjsip_transport_management module did not have the unload + logic in it to clean itself up causing the res_pjsip module to not + get unloaded. As a result the res_pjsip monitor thread kept going + processing traffic and timers when it shouldn't. + + Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a + +2016-04-15 11:41 +0000 [f4693d1897] Richard Mudgett <rmudgett@digium.com> + + * bridge_channel.c: Ignore role setup failure in channel push. + + We have to setup the channel roles after the bridge class push is called + because the bridge class push callback may have set roles on the incoming + channel. Since we have already partially pushed the channel into the + bridge and reversing what we have already done could be problematic, the + only thing we can do is press on to complete pushing the channel into the + bridge. + + * Ignore any channel role setup errors after pushing the channel into a + bridge. The channel may behave incorrectly in the bridge but we can no + longer abort the push at this time. + + Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00 + +2016-04-17 15:37 +0000 [22335fe18a] Jaco Kroon <jaco@uls.co.za> + + * chan_sip: Don't verify table if rtupdate=no + + If rtupdate=no do not verify sipregs/peers table has updatable fields. + + ASTERISK-25934 #close + + Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d + +2016-04-18 04:53 +0000 [3b9d8b60b2] ibercom <ibercom123@gmail.com> + + * app_queue: Frequent segfaults in function can_ring_entry() + + ASTERISK-25888 #close + + Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117 + +2016-04-15 16:51 +0000 [724acb6ce7] Richard Mudgett <rmudgett@digium.com> + + * stasis_bridge.c: Update stasis bridge push diagnostic messages. + + Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a + +2016-04-14 13:49 +0000 [5f78801859] Mark Michelson <mmichelson@digium.com> + + * transport management: Register thread with PJProject. + + The scheduler thread that kills idle TCP connections was not registering + with PJProject properly and causing assertions if PJProject was built in + debug mode. + + This change registers the thread with PJProject the first time that the + scheduler callback executes. + + AST-2016-005 + + Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 + +2016-03-08 12:12 +0000 [7fb3724a77] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_transport_management: Kill idle TCP connections. + + "Idle" here means that someone connects to us and does not send a SIP + request. PJProject will not automatically time out such connections, so + it's up to Asterisk to do it instead. + + When we receive an incoming TCP connection, we will start a timer + (equivalent to transaction timer D) waiting to receive an incoming + request. If we do not receive a request in that timeframe, then we will + shut down the TCP connection. + + ASTERISK-25796 #close + Reported by George Joseph + + AST-2016-005 + + Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 + +2016-03-08 10:52 +0000 [707fd4dcd0] Mark Michelson <mmichelson@digium.com> + + * Rename res_pjsip_keepalive res_pjsip_transport_management + + ASTERISK-25796 + Reported by George Joseph + + AST-2016-005 + + Change-Id: Id322a05f927392293570599730050bc677d99433 + +2016-04-14 07:15 +0000 [0b4bb19e0b] Mark Michelson <mmichelson@digium.com> + + * AST-2016-004: Fix crash on REGISTER with long URI. + + Due to some ignored return values, Asterisk could crash if processing an + incoming REGISTER whose contact URI was above a certain length. + + ASTERISK-25707 #close + Reported by George Joseph + + Patches: + 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch + + AST-2016-004 + + Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55 + +2016-04-12 13:10 +0000 [f6e080c6a4] Richard Mudgett <rmudgett@digium.com> + + * bridge_softmix.c: Fix crash if could not allocate the dsp. + + Fix off nominal crash where we could not setup the channel to process + frames for the softmix bridge technology because of allocation failure. + + Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372 + +2016-04-12 15:41 +0000 [cf15a2f2d3] gtjoseph <george.joseph@fairview5.com> + + * pjproject: Add patch for removing strip of '[]' from header params + + From the patch submitted to Teluu on 4/12/2016 + <<<<<<<<< + The wholesale stripping of '[]' from header parameters causes issues if + something (like a port) occurs after the final ']'. + + '[2001:a::b]' will correctly parse to '2001:a::b' + '[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left + with ':8080' and parsing stops with a syntax error. + + I can't even find a case where stripping the '[]' is a good thing anyway. Even + if you continued to parse and resulted in a string that looks like this... + '2001:a::b:8080', it's not valid. + + This came up in Asterisk because Kamailio sends us a Contact with an alias + URI parameter that has an IPv6 address in it like this: + Contact: <sip:1171@127.0.0.1:5080;alias=[2001:1:2::3]~43691~6> + which should be legal but causes a syntax error because of the characters + after the final ']'. Even if it didn't, the '[]' should still not be stripped. + + I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6 + enabled. No issues were caused by removing the code that strips the '[]'. + >>>>>>>>>>> + + ASTERISK-25123 #close + Reported-by: Anthony Messina + + Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a + +2016-04-12 09:10 +0000 [daa086fae4] Joshua Colp <jcolp@digium.com> + + * app_voicemail: Fix test_voicemail_notify_endl test. + + The test_voicemail_notify_endl test checks the end-of-line + characters of an email message to confirm that they are consistent. + The test wrongfully assumed that reading from the email message + into a buffer will always result in more than 1 character being + read. This is incorrect. If only 1 character was read the test + would go outside of the buffer and access other memory causing + a crash. + + The test now checks to ensure that 2 or more characters are read + in ensuring the test stays within the buffer. + + ASTERISK-25874 #close + + Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710 + +2016-04-07 12:02 +0000 [f896136460] Alexei Gradinari <alex2grad@gmail.com> + + * app_voicemail/IMAP: function 'save_to_folder' creates wrong folder + + If try to move message to Cust1 (number 5) + the function 'save_to_folder' tries to create Greeting folder instead of Cust1. + + This patch fixed it by setting GREETINGS_FOLDER = -1 + + ASTERISK-24927 #close + + Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51 + +2016-04-07 16:18 +0000 [70b7673f09] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip: Add headers to AMI Event ContactStatusDetail + + * Added Useragent and RegExpire headers to AMI Event + ContactStatusDetail with associated documentation. + + ASTERISK-25903 #close + + Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239 + +2016-04-11 16:20 +0000 [64ecd41c8f] Alexei Gradinari <alex2grad@gmail.com> + + * Codecs: strip codec name while parsing allow/disallow options + + Failed registration using PJSIP/Realtime if one of the codec name + in allow/disallow option is wrong or contains space. + + This patch strip codec name. + + ASTERISK-25914 + + Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d + +2016-04-11 14:26 +0000 [3f6c4667b8] Jaco Kroon <jaco@uls.co.za> + + * core_unreal: Fix hangupcauses not getting set on Local channels + + ASTERISK-25912 #close + + Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa + +2016-04-01 13:30 +0000 [fe7e48db03] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip contact: Lock expiration/addition of contacts + + Contact expiration can occur in several places: res_pjsip_registrar, + res_pjsip_registrar_expire, and automatically when anyone calls + ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar + may also be attempting to renew or add a contact. Since none of this was locked + it was possible for one thread to be renewing a contact and another thread to + expire it immediately because it was working off of stale data. This was the + casue of intermittent registration/inbound/nominal/multiple_contacts test + failures. + + Now, the new named lock functionality is used to lock the aor during contact + expire and add operations and res_pjsip_registrar_expire now checks the + expiration with the lock held before deleting the contact. + + ASTERISK-25885 #close + Reported-by: Josh Colp + + Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059 + +2016-04-10 14:16 +0000 [0c414eaf35] gtjoseph <george.joseph@fairview5.com> + + * pjproject: Add patch to fix Via IPv6 parsing + + There's a bug in pjproject's sip_parser where the ":" wasn't correctly + interpreted. This is causing IPv6 addresses in the "received" parameter of the + Via header to cause a syntax check failure. + + This patch was submitted to Teluu on 4/10/2016. + + ASTERISK-25910 #close + Reported-by: Anthony Messina + + Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922 + +2016-03-31 20:04 +0000 [772ff3048f] gtjoseph <george.joseph@fairview5.com> + + * lock: Add named lock capability + + Locking some objects like sorcery objects can be tricky because the underlying + ao2 object may not be the same for all callers. For instance, two threads that + call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 + different ao2 objects if the underlying wizard had to rehydrate the aor from a + database. Locking one ao2 object doesn't have any effect on the other even if + those objects had locks in the first place. + + Named locks allow access control by keyspace and key strings. Now an "aor" + named "1000" can be locked and any other thread attempting to lock "aor" "1000" + will wait regardless of whether the underlying ao2 object is the same or not. + Mutex and rwlocks are supported. + + This capability will initially be used to lock an aor when multiple threads may + be attempting to prune expired contacts from it. + + Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 + +2016-04-05 16:56 +0000 [fd601f26f7] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_outbound_publish: Add transport for outbound PUBLISH + + The first available transport of the appropriate type is used now. + This patch adds new config option 'transport' for outbound-publish. + If transport is set then outbound PUBLISH requests will use this transport. + + ASTERISK-25901 #close + + Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151 + +2016-04-07 16:39 +0000 [5f768d2a9c] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event + + BLF pickup isn't working on Cisco SPA and Snom phones + if the direction="recipient" attribute is missing in 'dialog' tag. + + This patch adds direction="recipient" if extension state is + Ringing. + + ASTERISK-24601 #close + + Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c + +2016-04-07 10:59 +0000 [82638fb0c7] Richard Mudgett <rmudgett@digium.com> + + * pbx.c: Minor code rearangements. + + * Pull out a loop invariant. + + * Convert an else-if ladder to a switch statement. + + Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95 + +2016-04-07 11:37 +0000 [bc320df173] Alexei Gradinari <alex2grad@gmail.com> + + * app_voicemail/IMAP: IMAP access FATAL error: Out of memory + + Sometimes uw-imap function 'mail_fetchbody' returns huge len + which then pass to uw-imap function 'rfc822_base64'. + uw-imap tries to allocate huge memory and abort() on fail. + + This patch check the len. + If the len more than max size (128 Mbytes) log error. + This patch also set variables len, newlen to avoid uninizialezed len. + This patch also check pointer returned by rfc822_base64. + + ASTERISK-25899 #close + + Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca + +2016-04-07 12:26 +0000 [2ef8a954b3] Richard Mudgett <rmudgett@digium.com> + + * pbx: Update doxygen for extension state watchers. + + Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e + +2016-04-07 11:49 +0000 [d312fdeb1b] gtjoseph <george.joseph@fairview5.com> + + * alembic: Remove batch operations (and sqlite support) + + Because SQLite doesn't support full ALTER capabilities, alembic scripts + require batch operations. However, that capability wasn't available until + 0.7.0 which some distributions haven't reached yet. Therefore, the batch + operations introduced in commit 86d6e44cc (review 2319) have been reverted + and SQLite is unsupported again, for now anyway. + + Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql. + + ASTERISK-25890 #close + Reported-by: Harley Peters + + Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80 + +2016-04-07 11:05 +0000 [901e8d78c4] Joshua Colp <jcolp@digium.com> + + * res_pjsip_registrar_expire: Fix race condition at shutdown. + + When shutting down, the PJSIP sorcery is destroyed. The registrar + expiration module queries the PJSIP sorcery to determine what + to expire. As there was no synchronization between termination + of the expiration thread and the unloading of the module it was + possible for the thread to try to access the PJSIP sorcery after + it had been destroyed. + + This change ensures that the thread is shut down before allowing + the module to be considered unloaded. + + Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b + +2016-04-06 16:28 +0000 [8207372e66] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Fix configuration setting of "regcontext". + + Due to a merge problem two options were swapped causing the + regcontext setting to not get set. + + Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1 + +2016-04-06 08:01 +0000 [0735a4d6d7] Jacek Konieczny <jkonieczny@eggsoft.pl> + + * frame.c: Copy the whole subclass in ast_frdup(). + + The problem is ast_frdup() does not copy whole frame.subclass for voice, + video and image frames, only the format is copied. For video frames, the + subclass structure contains the .frame_ending flag used to put the RTP + marker where it needs to be. + + ASTERISK-25894 #close + + Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33 + +2016-04-05 14:23 +0000 [c61dca6419] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Handle deferred SDP hold/unhold properly. + + Some SIP devices indicate hold/unhold using deferred SDP reinvites. In + other words, they provide no SDP in the reinvite. + + A typical transaction that starts hold might look something like this: + + * Device sends reinvite with no SDP + * Asterisk sends 200 OK with SDP indicating sendrecv on streams. + * Device sends ACK with SDP indicating sendonly on streams. + + At this point, PJMedia's SDP negotiator saves Asterisk's local state as + being recvonly. + + Now, when the device attempts to unhold, it again uses a deferred SDP + reinvite, so we end up doing the following: + + * Device sends reinvite with no SDP + * Asterisk sends 200 OK with SDP indicating recvonly on streams + * Device sends ACK with SDP indicating sendonly on streams + + The problem here is that Asterisk offered recvonly, and by RFC 3264's + rules, if an offer is recvonly, the answer has to be sendonly. The + result is that the device is not taken off hold. + + What is supposed to happen is that Asterisk should indicate sendrecv in + the 200 OK that it sends. This way, the device has the freedom to + indicate sendrecv if it wants the stream taken off hold, or it can + continue to respond with sendonly if the purpose of the reinvite was + something else (like a session timer refresher). + + The fix here is to alter the SDP negotiator's state when we receive a + reinvite with no SDP. If the negotiator's state is currently in the + recvonly or inactive state, then we alter our local state to be + sendrecv. This way, we allow the device to indicate the stream state as + desired. + + ASTERISK-25854 #close + Reported by Robert McGilvray + + Change-Id: I7615737276165eef3a593038413d936247dcc6ed + +2016-03-27 23:33 +0000 [50b0922a22] gtjoseph <george.joseph@fairview5.com> + + * config: Allow filters when appending to a category + + In sorcery based config files where there are multiple categories with the same + name, you can't use the (+) operator to reliably append to a category because + config.c stops looking when it finds the first one with the same name. + + Example: + + [1000] + type = endpoint + + [1000] + type = aor + + [1000](+) + authenticate_qualify = yes + + This config will fail because config.c appends authenticate_qualify to the + first category it finds, the endpoint, and that's not valid for endpoint. + + Solution: + + The capability to find a category that contains a certain variable already + exists so the only real change was to parse anything after the '+' that's not a + comma, as a filter string. + + [1000] + type = endpoint + + [1000] + type = aor + + [1000](+type=aor) + authenticate_qualify = yes + + This now works as expected. + + Although the following example doesn't make any sense for pjsip, you can even + specify multiple filters: + + [1000](+type=aor&qualify_frequency=10) + + ASTERISK-25868 #close + Reported-by: Nick Repin + + Change-Id: I10773da4c79db36fbf1993961992af63d3441580 + +2016-04-05 10:21 +0000 [cb56ef8069] Joshua Colp <jcolp@digium.com> + + * res_http_websocket: Make core supported. + + Websockets are a core part of ARI support and as such this + module should also be core supported. + + Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c + +2016-03-25 23:22 +0000 [f6f4cf459f] gtjoseph <george.joseph@fairview5.com> + + * stringfields: Refactor to allow fields to be added to the end of structures + + String fields are great, except that you can't add new ones without breaking + ABI compatibility because it shifts down everything else in the structure. + The only alternative is to add your own char * field to the end of the + structure and manage the memory yourself which isn't ideal, especially since + you then can't use the OPT_STRINGFIELD_T type. + + Background: + + The reason string fields had to be declared inside the + AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared + fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS + declared the pool, then the fields, then the manager, you could use the offsets + of the pool and manager and iterate over the sequential addresses in between to + access the fields. The actual pool, field allocation and field set operations + don't actually care where the field is. It's just iteration over the fields + that was the problem. + + Solution: Extended String Fields + + An extended string field is one that is declared outside the + AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent + structure. Other than using AST_STRING_FIELD_EXTENDED instead of + AST_STRING_FIELD, it looks the same as other string fields. It's storage comes + from the pool and it participates in string field compare and copy operations + peformed on the parent structure. It's also a valid target for the + OPT_STRINGFIELD_T aco option type. + + Implementation: + + To keep track of the extended fields and make sure that ABI isn't broken, the + existing embedded_pool pointer in the manager structure was repurposed to be a + pointer to a separate header structure that contains the embedded_pool pointer + plus a vector of fields. The length of the manager structure didn't change and + the embedded_pool pointer isn't used in the macros, only the stringfields C + code. A side benefit of this is that changing the header structure in the + future won't break ABI. + + ast_string_fields_init initializes the normal string fields and appends them to + the vector, and subsequent calls to ast_string_field_init_extended initialize + and append the extended fields. Cleanup, ast_string_fields_cmp, and + ast_string_fields_copy can now work on the vector instead of sequentially + traversing the addresses between the pool and manager. + + The total size of a structure using string fields didn't change, whether using + extended fields or not, nor have the offsets of any structure members, either + inside the original block or outside. Adding an extended field to the end of a + structure is the same as adding a char *. + + Details: + + The stringfield C code was pulled out from utils.c and into stringfields.c. + It just made sense. + + Additional work was done in ast_string_field_init and + ast_calloc_with_stringfields to handle the allocation of the new header + structure and the vector, and the associated cleanup. In the process some + additional NULL pointer checking was added. + + A lot of work was done in stringfields.h since the logic for compare and copy + is there. Documentation was added as well as somne additional NULL checking. + + The ability to call ast_calloc_with_stringfields with a number of structures + greater than 1 never really worked. Well, the calloc worked but there was no + way to access the additional structures or clean them up. It was agreed that + there was no use case for requesting more than 1 structure so an ast_assert + was added to prevent it and the iteration code removed. + + Testing: + + The stringfield unit tests were updated to test both normal and extended + fields. Tests for ast_string_field_ptr_set_by_fields and + ast_calloc_with_stringfields were also added. + + As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except + res_pjsip itself, saved off. The patch was then added and a full compile and + install was performed. Then the older res_pjsip_* moduled were copied over the + installed versions so res_pjsip was new and the rest were old. No issues. + + contact->aor, which is a char * at the end of contact, was then changed to an + extended string field and a recompile and reinstall was performed, again + leaving stock versions of the the res_pjsip_* modules. Again, no issues with + the res_pjsip_* modules using the old stringfield implementation and with + contact->aor as a char *, and res_pjsip itself using the new stringfield + implementation and contact->aor being an extended string field. + + Finally, several existing string fields were converted to extended string + fields to test OPT_STRINGFIELD_T. Again, no issues. + + Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61 + +2016-04-04 18:02 +0000 [fe448ac8a7] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_mwi: Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7 + + I forgot the new voicemail_extension wasn't a stringfield and didn't check + for NULL where I should have. + + Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb + +2016-04-03 11:47 +0000 [9d4318f798] gtjoseph <george.joseph@fairview5.com> + + * install_prereq: Fix check_installed_debs remove subversion + + check_installed_debs wasn't handling virtual packages like libsrtp-dev and + libresample-dev and on multiarch systems it was accidentally filtering out all + packages if any :i386 packages were found instead of just filtering out the + :i386 packages themselves. + + Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda + +2016-04-01 13:09 +0000 [566601837e] gtjoseph <george.joseph@fairview5.com> + + * utils.c: Fix typo in handle_show_locks + + ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e). + + Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf + +2016-03-30 18:34 +0000 [964f54bd5d] gtjoseph <george.joseph@fairview5.com> + + * pjproject_bundled: Fix use of LDCONFIG for shared library link creation + + LDCONFIG apparently isn't set to something sane on all systems so the creation + of the shared library links fails. Instead of just testing for non-blank, + main/Makefile now checks that LDCONFIG is actually executable and reverts to + LN if it isn't. + + This applies to both libasteriskpj and libasteriskssl. + + Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG. + + ASTERISK-25873 #close + Reported-by: Hans van Eijsden + + Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729 + +2016-03-30 13:31 +0000 [5f73c2ef0a] Richard Mudgett <rmudgett@digium.com> + + * res_stasis.c: Protect channel datastore list from stasis end. + + Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95 + +2016-03-29 18:06 +0000 [74d63f56ee] Richard Mudgett <rmudgett@digium.com> + + * res_ari: Cannot get control also means channel is unavailable. + + The only caller of ari_bridges_play_found() has this note: + + If ari_bridges_play_found fails because the channel is unavailable for + playback, The channel will be removed from the playback list soon. We can + keep trying to get channels from the list until we either get one that + will work or else there isn't a channel for this bridge anymore, in which + case we'll revert to ari_bridges_play_new. + + Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6 + +2016-03-29 14:29 +0000 [cf49b44090] Richard Mudgett <rmudgett@digium.com> + + * res_stasis_recording.c: Cleanup stasis_app_recording_find_by_name(). + + Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7 + +2016-03-28 14:23 +0000 [7f53f1d89e] Richard Mudgett <rmudgett@digium.com> + + * core_unreal.c: Add clarification comment about channel ref. + + Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3 + +2016-03-29 13:47 +0000 [ecf4102d02] Richard Mudgett <rmudgett@digium.com> + + * res_stasis: Add control ref to playback and recording structs. + + The stasis_app_playback and stasis_app_recording structs need to have a + struct stasis_app_control ref. Other threads can get a reference to the + playback and recording structs from their respective global container. + These other threads can then use the control pointer they contain after + the control struct has gone. + + * Add control ref to stasis_app_playback and stasis_app_recording structs. + + With the refs added, the control command queue can now have a circular + control reference which will cause the control struct to never get + released if the control's command queue is not flushed when the channel + leaves the Stasis application. Also the command queue needs better + protection from adding commands if the control->is_done flag is set. + + * Flush the control command queue on exit. + + ASTERISK-25882 #close + + Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d + +2016-03-28 18:10 +0000 [a179aba65e] Richard Mudgett <rmudgett@digium.com> + + * res_stasis: Fix crash on a hanging up channel. + + * Give the struct stasis_app_control ao2 object a ref to the channel held + in the object. Now the channel will still be around if a thread needs to + post a stasis message instead of crash because the topic was destroyed. + + * Moved stopping any lingering silence generator out of the struct + stasis_app_control destructor and made it a part of exiting the Stasis + application. Who knows which thread the destructor will be called under + so it cannot affect the channel's silence generator. Not only was the + channel unprotected when the silence generator was stopped, stasis may no + longer even control the channel. + + ASTERISK-25882 + + Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4 + +2016-03-30 12:38 +0000 [16c7d8e74a] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_mwi: Allow subscribe to vm access extension as an alias + + Background: + + If your extension is 1000 and the voicemail access extension is 1571 and you + dial 1571, usually a dialplan rule calls voicemailmain with your extension and + you are placed directly in your mailbox. Therefore most admins program the + voicemail (or other speed dial) button on their phones to the access extension. + Some phones (Snom at least) use whatever is programmed there to also subscribe + for MWI and so can't dial one number and subscribe to another. This works fine + in chan_sip because chan_sip completely ignores the user portion of the + SUBSCRIBE message request URI. If it can match the peer, is subscribes to the + peer's mailbox. The user could be set to anything or nothing and you'd still + get subscribed to your mailbox. + + Issue: + + chan_pjsip actually uses the user portion of the URI to find an aor and its + mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can + create an aor for 1571 but you certainly can't add your entire voicemail + system's mailboxes to it and everyone would get notified of every MWI. + + Solution: + + When an MWI subscribe comes in and an aor can't be found that matches the + resource directly, check the resource against the endpoint's aors. If an aor + is found that has a voicemail_extension that matches the resource, use it. + + ASTERISK-25865 + Reported-by: Ross Beer + + Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e + +2016-03-24 22:55 +0000 [d8f0bc3572] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited + + res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds + the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes + on endpoints for unsolicited mwi and on aors for subscriptions required + that the admin know in advance which the client wanted. If you specified + mailboxes on the endpoint, subscriptions were rejected even if you also + specified mailboxes on the aor. + + Voicemail extension: + * Added a global default_voicemail_extension which defaults to "". + * Added voicemail_extension to both endpoint and aor. + * Added ast_sip_subscription_get_dialog for support. + * Added ast_sip_subscription_get_sip_uri for support. + + When an unsolicited NOTIFY is constructed, the From header is parsed, the + voicemail extension from the endpoint is substituted for the user, and the + result placed in the Message-Account field in the body. + + When a subscribed NOTIFY is constructed, the subscription dialog local uri + is parsed, the voicemail_extension from the aor (looked up from the + subscription resource name) is substituted for the user, and the result + placed in the Message-Account field in the body. + + If no voicemail extension was defined, the Message-Account field is not added + to the NOTIFY body. + + mwi_subscribe_replaces_unsolicited: + * Added mwi_subscribe_replaces_unsolicited to endpoint. + + The previous behavior was to reject a subscribe if a previous internal + subscription for unsolicited MWI was found for the mailbox. That remains the + default. However, if there are mailboxes also set on the aor and the client + subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal + subscription is removed and replaced with the external subscription. This + allows an admin to configure mailboxes on both the endpoint and aor and allows + the client to select which to use. + + ASTERISK-25865 #close + Reported-by: Ross Beer + + Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea + +2016-03-30 09:46 +0000 [8dc8d6ceb8] gtjoseph <george.joseph@fairview5.com> + + * res_rtp_asterisk: Fix placement of txcount increment + + Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount + for rtcp packets as well as rtp packets and that was causing sender reports + to be generated instead of receiver reports in cases where no rtp was actually + being sent. + + Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp, + to rtp_sento which only handles rtp packets. + + Discovered by the hep/rtcp-receiver test. + + Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5 + +2016-03-26 22:33 +0000 [c7eb18d865] gtjoseph <george.joseph@fairview5.com> + + * chan_pjsip: Add 'pjsip show channelstats' + + Added the ability to show channel statistics to chan_pjsip (cli_functions.c) + + Moved the existing 'pjsip show channel(s)' functionality from + pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's + private header so it made sense to move the existing channel commands as well. + + Now using stasis_cache_dump to get the channel snapshots rather than retrieving + all endpoints, then getting each one's channel snapshots. Much more efficient. + + Change-Id: I03b114522126d27434030b285bf6d531ddd79869 + +2016-03-10 19:52 +0000 [1583559a06] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS + + No one seemed to notice but every time an OPTIONS goes out, it goes + out with a From of "asterisk" (or whatever the default from_user is set to), + even if you specify an endpoint. + + The issue had several causes... + qualify_contact is only called with an endpoint if called from the CLI. + If the endpoint is NULL, qualify_contact only looks up the endpoint if + authenticate_qualify=yes. Even then, it never passes it on to + ast_sip_create_request where the From header is set. Therefore From + is always "asterisk" (or whatever the default from_user is set to). + Even if ast_sip_create_request were to get an endpoint, it only sets + the From if endpoint->from_user is set. + + The fix is 4 parts... + + First, create_out_of_dialog_request was modified to use the endpoint id + if endpoint was specified and from_user is not set. + + Second, qualify_contact was modified to always look up an endpoint if + one wasn't specified regardless of authenticate_qualify. It then passes + the endpoint on to create_out_of_dialog_request. + + Third (and most importantly), find_an_endpoint was modified to find + an endpoint by using an "aors LIKE %contact->aor%" predicate with + ast_sorcery_retrieve_by_fields. As such, this patch will only work + if the sorcery realtime optimizations patch goes in. Otherwise we'd + be pulling the entire endpoints database every time we send an OPTIONS. + Since we already know the contact's aor, the on_endpoint callback was also + modified to just check if the contact->aor is an exact match to one of + the endpoint's. + + Finally, since we now have an endpoint for every OPTIONS request, + res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was + updated to get the transport from the endpoint and set it on tdata. + Now the correct transport is used. + + Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af +2016-03-25 10:59 +0000 [0cfab30b28] Jacek Konieczny <jkonieczny@eggsoft.pl> + + * res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS + + Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 + explicitly states: + + There MUST be a separate DTLS-SRTP session for each distinct pair of + source and destination ports used by a media session + + This means RTP keying material cannot be used for DTLS RTCP, which was + the reason why RTCP encryption would fail. + + ASTERISK-25642 + + Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a + +2016-03-25 10:42 +0000 [6a9c18fb59] Jacek Konieczny <jkonieczny@eggsoft.pl> + + * app_echo: forward and generate VIDUPDATE frames + + When using app_echo via WebRTC with VP8 video the video would appear + only after a few minutes, because there would be nothing to request + a full reference frame. + + This fixes the problem in both ways: + - echos any VIDUPDATE frames received on the channel + - sends one such frame when first video frame is to be forwarded + + This makes the echo work with Firefox and Chrome WebRTC implementation. + + ASTERISK-25867 #close + + Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e + +2016-03-27 12:53 +0000 [1bce690ccb] gtjoseph <george.joseph@fairview5.com> + + * res_rtp_asterisk: Fix packet stats on bridged connection + + rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated + for bridged streams because the calulations were being done after the + bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated. + + Moved the calculations so they occur for all valid received packets and + all transmitted packets. Also added rxoctetcount and txoctetcount to + ast_rtp_instance_stat. + + Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb + +2016-03-25 23:19 +0000 [50f90d4099] Philip Correia + + * res_parking: Fix blind transfer dynamic lots creation. + + Blind transfers to a recognized parking extension need to use the parker's + channel variable values to create the dynamic parking lot. This is + because there is always only one parker while the parkee may actually be a + multi-party bridge. A multi-party bridge can never supply the needed + channel variables to create the dynamic parking lot. In the multi-party + bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and + channel variables are inherited by the local channel used to park the + bridge. + + * In park_common_setup(), make use the parker instead of the parkee to + supply the dynamic parking lot channel variable values. In all but one + case, the parkee is the same as the parker. However, in the recognized + parking extension blind transfer scenario for a two party bridge they are + different channels. For consistency, we need to use the parker channel. + + * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the + local channel when blind transferring a multi-party bridge to a recognized + parking extension. + + * When a local channel starts a call, the Local;2 side needs to inherit + the CHANNEL(parkinglot) value from Local;1. + + The DTMF one-touch parking case wasn't even trying to create dynamic + parking lots before it aborted the attempt. + + * In parking_park_call(), add missing code to create a dynamic parking + lot. + + A DTMF bridge hook is documented as returning -1 to remove the hook. + Though the hook caller is really coded to accept non-zero. See the + ast_bridge_hook_callback typedef. + + * In feature_park_call(), don't remove the DTMF one-touch parking hook + because of an error. + + ASTERISK-24605 #close + Reported by: Philip Correia + Patches: + call_park.patch (license #6672) patch uploaded by Philip Correia + + Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9 + +2016-03-08 15:55 +0000 [5aa5c49413] gtjoseph <george.joseph@fairview5.com> + + * sorcery/res_pjsip: Refactor for realtime performance + + There were a number of places in the res_pjsip stack that were getting + all endpoints or all aors, and then filtering them locally. + + A good example is pjsip_options which, on startup, retrieves all + endpoints, then the aors for those endpoints, then tests the aors to see + if the qualify_frequency is > 0. One issue was that it never did + anything with the endpoints other than retrieve the aors so we probably + could have skipped a step and just retrieved all aors. But nevermind. + + This worked reasonably well with local config files but with a realtime + backend and thousands of objects, this was a nightmare. The issue + really boiled down to the fact that while realtime supports predicates + that are passed to the database engine, the non-realtime sorcery + backends didn't. + + They do now. + + The realtime engines have a scheme for doing simple comparisons. They + take in an ast_variable (or list) for matching, and the name of each + variable can contain an operator. For instance, a name of + "qualify_frequency >" and a value of "0" would create a SQL predicate + that looks like "where qualify_frequency > '0'". If there's no operator + after the name, the engines add an '=' so a simple name of + "qualify_frequency" and a value of "10" would return exact matches. + + The non-realtime backends decide whether to include an object in a + result set by calling ast_sorcery_changeset_create on every object in + the internal container. However, ast_sorcery_changeset_create only does + exact string matches though so a name of "qualify_frequency >" and a + value of "0" returns nothing because the literal "qualify_frequency >" + doesn't match any name in the objset set. + + So, the real task was to create a generic string matcher that can take a + left value, operator and a right value and perform the match. To that + end, strings.c has a new ast_strings_match(left, operator, right) + function. Left and right are the strings to operate on and the operator + can be a string containing any of the following: = (or NULL or ""), !=, + >, >=, <, <=, like or regex. If the operator is like or regex, the + right string should be a %-pattern or a regex expression. If both left + and right can be converted to float, then a numeric comparison is + performed, otherwise a string comparison is performed. + + To use this new function on ast_variables, 2 new functions were added to + config.c. One that compares 2 ast_variables, and one that compares 2 + ast_variable lists. The former is useful when you want to compare 2 + ast_variables that happen to be in a list but don't want to traverse the + list. The latter will traverse the right list and return true if all + the variables in it match the left list. + + Now, the backends' fields_cmp functions call ast_variable_lists_match + instead of ast_sorcery_changeset_create and they can now process the + same syntax as the realtime engines. The realtime backend just passes + the variable list unaltered to the engine. The only gotcha is that + there's no common realtime engine support for regex so that's been noted + in the api docs for ast_sorcery_retrieve_by_fields. + + Only one more change to sorcery was done... A new config flag + "allow_unqualified_fetch" was added to reg_sorcery_realtime. + "no": ignore fetches if no predicate fields were supplied. + "error": same as no but emit an error. (good for testing) + "yes": allow (the default); + "warn": allow but emit a warning. (good for testing) + + Now on to res_pjsip... + + pjsip_options was modified to retrieve aors with qualify_frequency > 0 + rather than all endpoints then all aors. Not only was this a big + improvement in realtime retrieval but even for config files there's an + improvement because we're not going through endpoints anymore. + + res_pjsip_mwi was modified to retieve only endpoints with something in + the mailboxes field instead of all endpoints then testing mailboxes. + + res_pjsip_registrar_expire was completely refactored. It was retrieving + all contacts then setting up scheduler entries to check for expiration. + Now, it's a single thread (like keepalive) that periodically retrieves + only contacts whose expiration time is < now and deletes them. A new + contact_expiration_check_interval was added to global with a default of + 30 seconds. + + Ross Beer reports that with this patch, his Asterisk startup time dropped + from around an hour to under 30 seconds. + + There are still objects that can't be filtered at the database like + identifies, transports, and registrations. These are not going to be + anywhere near as numerous as endpoints, aors, auths, contacts however. + + Back to allow_unqualified_fetch. If this is set to yes and you have a + very large number of objects in the database, the pjsip CLI commands + will attempt to retrive ALL of them if not qualified with a LIKE. + Worse, if you type "pjsip show endpoint <tab>" guess what's going to + happen? :) Having a cache helps but all the objects will have to be + retrieved at least once to fill the cache. Setting + allow_unqualified_fetch=no prevents the mass retrieve and should be used + on endpoints, auths, aors, and contacts. It should NOT be used for + identifies, registrations and transports since these MUST be + retrieved in bulk. + + Example sorcery.conf: + + [res_pjsip] + endpoint=config,pjsip.conf,criteria=type=endpoint + endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error + + ASTERISK-25826 #close + Reported-by: Ross Beer + Tested-by: Ross Beer + + Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67 + +2016-03-23 14:24 +0000 [05fc3a96d1] Richard Mudgett <rmudgett@digium.com> + + * res_parking: Cleanup find_channel_parking_lot_name() usage. + + Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02 + +2016-03-18 14:01 +0000 [a4189763ab] Richard Mudgett <rmudgett@digium.com> + + * res_parking: Misc fixes. + + res/parking/parking_applications.c: + + * Add malloc fail checks in setup_park_common_datastore(). + + * Fix playing parking failed announcement to only happen on non-blind + transfers in park_app_exec(). It could never go out before because a test + was provedly always false. + + res/parking/parking_bridge.c: + + * Fix NULL tolerance in generate_parked_user() because + bridge_parking_push() can theoretically pass a NULL parker channel if the + parker channel went away for some reason. + + * Clarify some weird code dealing with blind_transfer in + bridge_parking_push(). + + res/parking/parking_bridge_features.c: + + * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel + which will be bulk copied to the Local;2 channel on the subsequent + ast_call(). The additional advantage is if the parker channel has the + BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed + to be overridden. + + res/parking/parking_manager.c: + + * Fix AMI Park action input range checking of the Timeout header in + manager_park(). + + * Reduced locking scope to where needed in manager_park(). + + res/res_parking.c: + + * Fix some off nominal missing unlocks by eliminating the returns. + + Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca + +2014-12-15 05:23 +0000 [6f95b5eda1] Philip Correia + + * res_parking: Update parking documentation for dynamic parking lots. + + * Remove duplicate res_parking.conf courtesytone config option + documentation. + + ASTERISK-24596 #close + Reported by: Philip Correia + + ASTERISK-24605 + Reported by: Philip Correia + Patches: + call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia + + Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5 + +2016-03-24 14:08 +0000 [81ce60f6d4] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. + + Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those + codecs, which the caller did not request/support. That fix was not complete + because on the second Session Timer all codecs were sent again. Some VoIP/SIP + clients interpreted that complete codec-list as a change in the SIP session. + Because of that, Asterisk did not send the RTP audio via NAT anymore which + created a non-audio scenario after the second Session Timer fired. + + ASTERISK-24543 #close + + Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66 + +2016-03-19 07:34 +0000 [c6e4c48e67] Gianluca Merlo <gianluca.merlo@gmail.com> + + * config: fix flags in uint option handler + + The configuration unsigned integer option handler sets flags for the + parser as if the option should be a signed integer (PARSE_INT32), + leading to errors on "out of range" values. Fix flags (PARSE_UINT32). + + A fix to res_pjsip is also present which stops invalid flags from + being passed when registering sorcery object fields for qualify + status. + + ASTERISK-25612 #close + + Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e + +2016-03-10 16:58 +0000 [59c8e189fd] Mark Michelson <mmichelson@digium.com> + + * Restrict CLI/AMI commands on shutdown. + + During stress testing, we have frequently seen crashes occur because a + CLI or AMI command attempts to access information that is in the process + of being destroyed. + + When addressing how to fix this issue, we initially considered fixing + individual crashes we observed. However, the changes required to fix + those problems would introduce considerable overhead to the nominal + case. This is not reasonable in order to prevent a crash from occurring + while Asterisk is already shutting down. + + Instead, this change makes it so AMI and CLI commands cannot be executed + if Asterisk is being shut down. For AMI, this is absolute. For CLI, + though, certain commands can be registered so that they may be run + during Asterisk shutdown. + + ASTERISK-25825 #close + + Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990 +2016-03-24 07:45 +0000 [ff3eebf454] Walter Doekes <walter+asterisk@wjd.nu> + + * musiconhold: Only warn if music class is not found in memory and database. + + The log message when a MusicOnHold music class was not found was changed + from debug level to WARNING level in Asterisk 11.19 and 13.5. For those + using realtime musiconhold, this message is wrong because it warns + before checking the database. + + This changeset delays the warning until after the database has been + checked. + + Reported-by: Conrad de Wet + ASTERISK-25444 #close + + Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf + +2016-03-24 05:38 +0000 [82e55e4883] Walter Doekes <walter+asterisk@wjd.nu> + + * core/logging: Fix broken syslog levels on older glibc. + + The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However + this macro is broken in older glibc (< 2.17); it would left-shift the + facility a second time, causing the resultant priority to become + invalid. + + The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: + + The priority argument is formed by ORing the facility and the level + values [...]. + + ASTERISK-25510 #close + Reported by: Michael Newton + + Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03 + +2016-03-23 08:59 +0000 [d963a33749] gtjoseph <george.joseph@fairview5.com> + + * pjproject-bundled: Cleanups for reported issues + + PortAudio should no longer be required + PJSIP_MAX_PKT_LEN is now 6000 + Older autoconf issue fixed. (CentOS 6) + + Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd + +2015-11-20 08:02 +0000 [c5170677e7] Francesco Castellano <francesco.castellano@messagenet.it> + + * chan_sip.c: Space after port causes unnecessary resolution attempt + + check_via() already skips leading blanks where the sent-by address (with the + optional port) should be placed. + + Since RFC 3261 allows for blanks between the port ant the Via parameters: + > https://tools.ietf.org/html/rfc3261#section-20.42 + (actually it allows a lot of blanks more ;-)). I just switched from + ast_skip_blanks() to ast_strip() on the local copy of the string. + + ASTERISK-21301 #close + + Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06 + +2016-03-19 17:49 +0000 [51deadee38] gtjoseph <george.joseph@fairview5.com> + + * progdocs: Exclude ./third-party from documentation generation + + We don't need pjproject's documentation embedded in Asterisk's. + + Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17 + +2016-03-18 20:32 +0000 [aa2fcd244e] Gianluca Merlo <gianluca.merlo@gmail.com> + + * func_aes: fix misuse of strlen on binary data + + The encryption code for AES_ENCRYPT evaluates the length of the data to + be encoded in base64 using strlen. The data is binary, thus the length + of it can be underestimated at the first NULL character. + Reuse the write pointer offset to evaluate it, instead. + + ASTERISK-25857 #close + + Change-Id: If686b5d570473eb926693c73461177b35b13b186 + +2016-04-20 10:46 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.8.2 Released. + +2016-04-20 05:45 +0000 [26d67ce885] Joshua Colp <jcolp@digium.com> + + * Release summaries: Remove previous versions + +2016-04-20 05:45 +0000 [d9909232ed] Joshua Colp <jcolp@digium.com> + + * .version: Update for 13.8.2 + +2016-04-20 05:45 +0000 [fc57bb9b15] Joshua Colp <jcolp@digium.com> + + * .lastclean: Update for 13.8.2 + +2016-04-20 05:45 +0000 [ac04474f38] Joshua Colp <jcolp@digium.com> + + * realtime: Add database scripts for 13.8.2 + +2016-04-18 17:00 +0000 [91a3e1184f] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_registrar: Fix bad memory-ness with user_agent. + + Recent changes to the PJSIP registrar resulted in tests failing due to + missing AOR_CONTACT_ADDED test events. The reason for this was that the + user_agent string had junk values in it, resulting in being unable to + generate the event. + + I'm going to be honest here, I have no idea why this was happening. Here + are the steps needed for the user_agent variable to get messed up: + * REGISTER is received + * First contact in the REGISTER results in a contact being removed + * Second contact in the REGISTER results in a contact being added + * The contact, AOR, expiration, and user agent all have to be passed as + format parameters to the creation of a string. Any subset of those + parameters would not be enough to cause the problem. + + Looking into what was happening, the thing that struck me as odd was + that the user_agent variable was meant to be set to the value of the + User-Agent SIP header in the incoming REGISTER. However, when removing a + contact, the user_agent variable would be set (via ast_strdupa inside a + loop) to the stored contact's user_agent. This means that the + user_agent's value would be incorrect when attempting to process further + contacts in the incoming REGISTER. + + The fix here is to use a different variable for the stored user agent + when removing a contact. Correcting the behavior to be correct also + means the memory usage is less weird, and the issue no longer occurs. + + ASTERISK-25929 #close + Reported by Joshua Colp + + Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08 + (cherry picked from commit f436b9ab111f1ff57c6dd3970051f123b42c1103) + +2016-04-18 13:41 +0000 [70e25ced60] Joshua Colp <jcolp@digium.com> + + * res_pjsip_transport_management: Allow unload to occur. + + At shutdown it is possible for modules to be unloaded that wouldn't + normally be unloaded. This allows the environment to be cleaned up. + + The res_pjsip_transport_management module did not have the unload + logic in it to clean itself up causing the res_pjsip module to not + get unloaded. As a result the res_pjsip monitor thread kept going + processing traffic and timers when it shouldn't. + + Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a + (cherry picked from commit 49bfdc9ac029e0ef17cd8e85d8d7b7731387a34e) + +2016-04-18 12:12 +0000 [856931edc2] Mark Michelson <mmichelson@digium.com> + + * PJSIP: Remove PJSIP parsing functions from uri length validation. + + The PJSIP parsing functions provide a nice concise way to check the + length of a hostname in a SIP URI. The problem is that in order to use + those parsing functions, it's required to use them from a thread that + has registered with PJLib. + + On startup, when parsing AOR configuration, the permanent URI handler + may not be run from a PJLib-registered thread. Specifically, this could + happen when Asterisk was started in daemon mode rather than + console-mode. If PJProject were compiled with assertions enabled, then + this would cause Asterisk to crash on startup. + + The solution presented here is to do our own parsing of the contact URI + in order to ensure that the hostname in the URI is not too long. The + parsing does not attempt to perform a full SIP URI parse/validation, + since the hostname in the URI is what is important. + + ASTERISK-25928 #close + Reported by Joshua Colp + + Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60 + (cherry picked from commit efae187217750e322cd6167705a33f888b631427) + +2016-04-14 20:26 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.8.1 Released. + +2016-04-14 15:26 +0000 [18e6f12e83] Kevin Harwell <kharwell@digium.com> + + * Release summaries: Remove previous versions + +2016-04-14 15:26 +0000 [625c07711a] Kevin Harwell <kharwell@digium.com> + + * .version: Update for 13.8.1 + +2016-04-14 15:26 +0000 [584f1fb3c7] Kevin Harwell <kharwell@digium.com> + + * .lastclean: Update for 13.8.1 + +2016-04-14 15:26 +0000 [1e37a63379] Kevin Harwell <kharwell@digium.com> + + * realtime: Add database scripts for 13.8.1 + +2016-04-14 13:49 +0000 [dcf1b3c098] Mark Michelson <mmichelson@digium.com> + + * transport management: Register thread with PJProject. + + The scheduler thread that kills idle TCP connections was not registering + with PJProject properly and causing assertions if PJProject was built in + debug mode. + + This change registers the thread with PJProject the first time that the + scheduler callback executes. + + AST-2016-005 + + Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 + +2016-03-08 12:12 +0000 [efafbb1319] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_transport_management: Kill idle TCP connections. + + "Idle" here means that someone connects to us and does not send a SIP + request. PJProject will not automatically time out such connections, so + it's up to Asterisk to do it instead. + + When we receive an incoming TCP connection, we will start a timer + (equivalent to transaction timer D) waiting to receive an incoming + request. If we do not receive a request in that timeframe, then we will + shut down the TCP connection. + + ASTERISK-25796 #close + Reported by George Joseph + + AST-2016-005 + + Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 + +2016-03-08 10:52 +0000 [159f639770] Mark Michelson <mmichelson@digium.com> + + * Rename res_pjsip_keepalive res_pjsip_transport_management + + ASTERISK-25796 + Reported by George Joseph + + AST-2016-005 + + Change-Id: Id322a05f927392293570599730050bc677d99433 + +2016-04-14 07:15 +0000 [c164ff004d] Mark Michelson <mmichelson@digium.com> + + * AST-2016-004: Fix crash on REGISTER with long URI. + + Due to some ignored return values, Asterisk could crash if processing an + incoming REGISTER whose contact URI was above a certain length. + + ASTERISK-25707 #close + Reported by George Joseph + + Patches: + 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch + + AST-2016-004 + + Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55 + +2016-03-29 19:39 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.8.0 Released. + +2016-03-29 14:39 +0000 [0f885f0076] Mark Michelson <mmichelson@digium.com> + + * Release summaries: Add summaries for 13.8.0 + +2016-03-29 14:34 +0000 [a1fa37aebd] Mark Michelson <mmichelson@digium.com> + + * Release summaries: Remove previous versions + +2016-03-29 14:34 +0000 [e7de5fd439] Mark Michelson <mmichelson@digium.com> + + * .version: Update for 13.8.0 + +2016-03-29 14:34 +0000 [8baf813848] Mark Michelson <mmichelson@digium.com> + + * .lastclean: Update for 13.8.0 + +2016-03-29 14:34 +0000 [42469df205] Mark Michelson <mmichelson@digium.com> + + * realtime: Add database scripts for 13.8.0 + +2016-03-22 18:31 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.8.0-rc1 Released. + +2016-03-22 13:26 +0000 [a698424678] Mark Michelson <mmichelson@lunkwill> + + * Release summaries: Add summaries for 13.8.0-rc1 + +2016-03-22 13:21 +0000 [e395a0b973] Mark Michelson <mmichelson@lunkwill> + + * .version: Update for 13.8.0-rc1 + +2016-03-22 13:21 +0000 [38a86b2dbf] Mark Michelson <mmichelson@lunkwill> + + * .lastclean: Update for 13.8.0-rc1 + +2016-03-22 13:21 +0000 [e0c8c8bf4a] Mark Michelson <mmichelson@lunkwill> + + * realtime: Add database scripts for 13.8.0-rc1 + +2016-03-18 14:31 +0000 [6a40520fe9] Kevin Harwell <kharwell@digium.com> + + * chan_pjsip: ref leak when checking direct_media_glare + + Fix the reference leak introduced in the following commit: + + 9444ddadf8525d1ce66a1faf1db97f9f6c265ca4 + + ASTERISK-25849 + + Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85 +2016-03-16 12:37 +0000 [9444ddadf8] Kevin Harwell <kharwell@digium.com> + + * chan_pjsip: transfers with direct media reinvite has wrong address/port + + During a transfer involving direct media a race occurs between when the + transferer channel is swapped out, initiating rtp changes/updates, and the + subsequent reinvites. + + When Alice, after speaking with Charlie (Bob is on hold), connects Bob and + Charlie invites are sent to each in order to establish the call between them. + Bob is taken off hold and Charlie is told to have his media flow through + Asterisk. However, if before those invites go out the bridge updates Bob's + and/or Charlie's rtp information with direct media data (i.e. address, port) + then the invite(s) will contain the remote data in the SDP instead of the + Asterisk data. + + The race occurs in the native bridge glue code when updating the peer. The + direct_media_address can get set twice before sending out the first invite + during call connection. This can happen because the checking/setting of the + direct_media_address happened in one thread while the sending of the invite(s) + happened in another thread. + + This fix removes the race condition by moving the checking/setting of the + direct_media_address to be in the same thread as the sending of the invites(s). + This serializes the checking/setting and sending so they can no longer happen + out of order. + + ASTERISK-25849 #close + + Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873 + +2015-10-19 07:11 +0000 [88240f98d9] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com> + + * install_prereq: Update repositories before install on Debian systems + + When to install packages the indexed local is more old of the + version of software on the repository they have been upgraded by security + update then get the package will give 404 not found. + + The patch prevent by update local index to repository for aptitude before + install. + + ASTERISK-25495 #close + + Reporte by: Rodrigo Ramírez Norambuena + + Change-Id: I645959e553aac542805ced394cac2dca964051fa + (cherry picked from commit 88f3dbaec9509bfba8bc1de7799aa0dc65304bb5) + +2015-06-03 20:12 +0000 [efcf9a96db] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * install_prereq: Check if is installed aptitude otherwise to install. + + If in Debian or system based, dont have aptitude installed the script do + nothing. This patch checked if aptitude installed, if not installed. + + Also, if execute script with all packages installed yet, the script not show + nothing and return exit 1 because the command 'grep' get nothing from pipe from + 'awk'. + + ASTERISK-25113 #close + Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f + (cherry picked from commit 6737ded0581a9e1256bdfe30c1d747e7ca93f8b3) + +2016-03-03 04:43 +0000 [2b1b8e382a] Sergio Medina Toledo <lumasepa@gmail.com> + + * res_pjsip_refer.c: Fix seg fault in process of Refer-to header. + + The "Refer-to" header of an incoming REFER request is parsed by + pjsip_parse_uri(). That function requires the URI parameter to be NULL + terminated. Unfortunately, the previous code added the NULL terminator by + overwriting memory that may not be safe. The overwritten memory results + could be benign, memory corruption, or a segmentation fault. Now the URI + is NULL terminated safely by copying the URI to a new chunk of memory with + the correct size to be NULL terminated. + + ASTERISK-25814 #close + + Change-Id: I32565496684a5a49c3278fce06474b8c94b37342 + +2016-03-11 12:22 +0000 [de04308ae4] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix mwi resub deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 #close + + Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6 + +2016-03-10 17:01 +0000 [5f6627a8a4] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix registration timeout and expire deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508 + +2016-03-10 12:17 +0000 [32bd7a64f9] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix t38id deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f + +2016-03-09 16:34 +0000 [43556b800b] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix reinviteid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1 + +2016-03-09 16:32 +0000 [38c1cdab2c] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix packet retransid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + * Fix retrans_pkt() to call check_pendings() with both the owner channel + and the private objects locked as required. + + * Refactor dialog retransmission packet list to safely remove packet + nodes. The list nodes are now ao2 objects. The list has a ref and the + scheduled entry has a ref. + + ASTERISK-25023 + + Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641 + +2016-03-09 16:26 +0000 [e4ad55c888] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix waitid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + * Made always run check_pendings() under the scheduler thread so scheduler + ids can be checked safely. + + ASTERISK-25023 + + Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52 + +2016-03-08 15:08 +0000 [98d5669c28] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix session timers deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900 + +2016-03-07 13:21 +0000 [9cb8f73226] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix autokillid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + * Fix clearing autokillid in __sip_autodestruct() even though we could + reschedule. + + ASTERISK-25023 + + Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f + +2016-03-07 18:28 +0000 [c5c7f48a15] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix provisional_keepalive_sched_id deadlock. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48 + +2016-03-09 11:22 +0000 [f959d84dfd] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + * Make dialog_unlink_all() unschedule all items at once in the sched + thread. + + ASTERISK-25023 + + Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4 + +2016-03-10 21:54 +0000 [5f3225ddcc] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Clear scheduled immediate events on unload. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + The reordering of chan_sip's shutdown is to handle any immediate events + that get put onto the scheduler so resources aren't leaked. The typical + immediate events at this time are going to be concerned with stopping + other scheduled events. + + ASTERISK-25023 + + Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20 + +2016-03-15 14:51 +0000 [7a74971771] Richard Mudgett <rmudgett@digium.com> + + * sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Delaying destruction of the chan_sip sip_pvt structures caused the + /channels/chan_sip/test_sip_rtpqos unit test to crash. That test + registers a special test ast_rtp_engine with the rtp engine module. When + the unit test completes it cleans up by unregistering the test + ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt + happens after the unit test returns, the destructor tries to call the rtp + engine destroy callback of the test ast_rtp_engine auto variable which no + longer exists on the stack. + + * Change the test ast_rtp_engine auto variable to a static variable. Now + the variable can still exist after the unit test exits so the delayed + sip_pvt destruction can complete successfully. + + ASTERISK-25023 + + Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13 + +2016-03-15 13:31 +0000 [d2c09ed73b] Andrew Nagy <andrew.nagy@the159.com> + + * app_stasis: Don't hang up if app is not registered + + This prevents pbx_core from hanging up the channel if the app isn't + registered. + + ASTERISK-25846 #close + + Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce +2016-03-07 15:50 +0000 [b2d2906445] Richard Mudgett <rmudgett@digium.com> + + * sched.c: Ensure oldest expiring entry runs first. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + * Updated sched unit test to check new behavior. + + ASTERISK-25023 + + Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3 + +2016-03-04 18:25 +0000 [9ae21b510f] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full(). + + Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12 + +2016-03-07 18:56 +0000 [56bcb97a3c] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Simplify sip_pvt destructor call levels. + + Remove destructor calling destroy_it calling really_destroy_it + for no benefit. Just make the destructor the really_destroy_it + function. + + Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a + +2016-03-14 08:59 +0000 [677a65fcbb] Joshua Colp <jcolp@digium.com> + + * build: Add configure check for proto field of PJSIP TLS transport setting. + + Older versions of PJSIP do not have the proto field on the TLS transport + setting structure. This change adds a configure check so even if it is + not present we will still be able to build. + + Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9 + +2016-03-12 16:02 +0000 [32f0a3d52a] gtjoseph <george.joseph@fairview5.com> + + * build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE + + I can't ever recall actually needing the intermediate files or the checking + that a double compile produces. What I CAN remember is every DONT_OPTIMIZE + build needing 3 invocations of gcc instead of 1 just to do the checks and + produce those intermediate files. + + Having said that, Richard pointed out that the reason for the double compile + was that there were cases in the past where a submitted patch failed to compile + because the submitter never tried it with the optimizations turned on. + + To get the best of both worlds, COMPILE_DOUBLE has been split into its own + option. If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected + BUT you can then turn it off if all you need are the debugging symbols. This + way you have to make an informed decision about disabling COMPILE_DOUBLE. + + To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature + was added to menuselect. The <use> element can now contain an "autoselect" + attribute which will turn the used member on but not create a hard dependency. + The cflags.xml implementation for COMPILE_DOUBLE looks like this... + + <member name="DONT_OPTIMIZE" displayname="Disable Optimizations ..."> + <use autoselect="yes">COMPILE_DOUBLE</use> + <support_level>core</support_level> + </member> + <member name="COMPILE_DOUBLE" displayname="Pre-compile with ...> + <depend>DONT_OPTIMIZE</depend> + <support_level>core</support_level> + </member> + + When DONT_OPTIMIZE is turned on, COMPILE_DOUBLE is turned on because + of the use. + When DONT_OPTIMIZE is turned off, COMPILE_DOUBLE is turned off because + of the depend. + When COMPILE_DOUBLE is turned on, DONT_OPTIMIZE is turned on because + of the depend. + When COMPILE_DOUBLE is turned off, DONT_OPTIMIZE is left as is because + it only uses COMPILE_DOUBLE, it doesn't depend on it. + + I also made a few tweaks to the ncurses implementation to move things + left a bit to allow longer descriptions. + + Change-Id: Id49ca930ac4b5ec4fc2d8141979ad888da7b1611 + +2016-03-10 13:09 +0000 [38499e7125] gtjoseph <george.joseph@fairview5.com> + + * pjproject: Pass (dont_)optimize flags to pjproject and fix pjsua + + The pjproject Makefile now uses the Asterisk optimization flags which + are determined by the setting of the DONT_OPTMIZE menuselect flag. + The Makefile was also restructured so a change to the top level + menuselect.makeopts will result in a rebuild of pjproject. + + Also, "--disable-resample" was removed from the pjproject configure + options. Without resample, pjsua (which is used by the testsuite) + can't make audio calls. When it can't, it segfaults. + + Change-Id: I24b0a4d0872acef00ed89b3c527a713ee4c2ccd4 + +2016-03-11 16:03 +0000 [336cae73cc] Walter Doekes <walter+asterisk@wjd.nu> + + * app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. + + Channel masquerading had a conflict with autochannel locking. + + When locking autochannel->channel, the channel is fetched from the + autochannel and then locked. During the fetch, the autochannel -- which + has no locks itself -- can be modified by someone who owns the channel + lock. That means that the value of autochan->channel cannot be trusted + until you hold the lock. + + In practice, this caused problems with Local channels getting + masqueraded away while the ChanSpy attempted to get info from that + channel. The old channel which was about to get removed got locked, but + the new (replaced) channel got unlocked (no-op). Because the replaced + channel was now locked (and would never get unlocked), it couldn't get + removed from the channel list in a timely manner, and would now cause + deadlocks when iterating over the channel list. + + This change checks the autochannel after locking the channel for changes + to the autochannel. If the channel had been changed, the lock is + reobtained on the new channel. + + In theory it seems possible that after this fix, the lock attempt on the + old (wrong) channel can be on an already destroyed lock, maybe causing + a crash. But that hasn't been observed in the wild and is harder induce + than the current deadlock. + + Thanks go to Filip Frank for suggesting a fix similar to this and + especially to IRC user hexanol for pointing out why this deadlock was + possible and testing this fix. And to Richard for catching my rookie + while loop mistake ;) + + ASTERISK-25321 #close + + Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def + +2016-03-07 21:34 +0000 [875d5e9872] gtjoseph <george.joseph@fairview5.com> + + * pjproject_bundled: Remove --with-external-pa from configure options. + + Not sure why it was there in the first place as we already specify + --disable-sound. + + Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9 + +2016-03-06 14:38 +0000 [530cff5f5f] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Strip spaces from items parsed from comma-separated lists + + Configurations like "aors = a, b, c" were either ignoring everything after "a" + or trying to look up " b". Same for mailboxes, ciphers, contacts and a few + others. + + To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To + facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were + updated to handle null pointers. + + In some cases, an ast_strlen_zero() test was added to skip consecutive commas. + + There was also an attempt to ast_free an ast_strdupa'd string in + ast_sip_for_each_aor which was causing a SEGV. I removed it. + + Although this issue was reported for realtime, the issue was in the res_pjsip + modules so all config mechanisms were affected. + + ASTERISK-25829 #close + Reported-by: Mateusz Kowalski + + Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2 + +2016-03-04 20:37 +0000 [3c8076a83b] gtjoseph <george.joseph@fairview5.com> + + * install_prereq: Add packages for bundled pjproject + + RedHat/CentOS needs python-devel + Debian/Ubuntu needs automake, libsrtp-dev and python-dev + + Ubuntu also needed libncurses5-dev for cmenuselect so while not + needed for pjproject, I adedd it anyway. + + Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089 + +2016-02-24 17:25 +0000 [27f32cd0a6] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited + + Per RFC3325, the 'From' header is now anonymized on outgoing calls when + caller id presentation is prohibited. + + TID = trust_id_outbound + PRO = Set(CALLERID(pres)=prohib) + USR = endpoint/from_user + DOM = endpoint/from_domain + PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) + + Conditions |Result + --------------------|---------------------------------------------------- + TID PRO USR DOM |PAI FROM + --------------------|---------------------------------------------------- + Y Y abc def.ghi |PRI "Anonymous" <sip:abc@def.ghi> + Y Y abc |PRI "Anonymous" <sip:abc@anonymous.invalid> + Y Y def.ghi |PRI "Anonymous" <sip:anonymous@def.ghi> + Y Y |PRI "Anonymous" <sip:anonymous@anonymous.invalid> + + Y N abc def.ghi |YES <sip:abc@def.ghi> + Y N abc |YES <sip:abc@<ip_address>> + Y N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi> + Y N |YES "Caller Name" <sip:<caller_exten>@<ip_address>> + + N Y abc def.ghi |NO "Anonymous" <sip:abc@def.ghi> + N Y abc |NO "Anonymous" <sip:abc@anonymous.invalid> + N Y def.ghi |NO "Anonymous" <sip:anonymous@def.ghi> + N Y |NO "Anonymous" <sip:anonymous@anonymous.invalid> + + N N abc def.ghi |YES <sip:abc@def.ghi> + N N abc |YES <sip:abc@<ip_address>> + N N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi> + N N |YES "Caller Name" <sip:<caller_exten>@<ip_address>> + + ASTERISK-25791 #close + Reported-by: Anthony Messina + + Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9 + +2016-03-03 17:34 +0000 [7cf7b0a4f9] gtjoseph <george.joseph@fairview5.com> + + * third_party/Makefile.rules: Replace unsupported != operator with $(shell ...) + + Apparently the != operator is fairly new so I've replaced it with + the old $(shell ...) syntax. + + Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479 + Reported-by: Richard Mudgett +2016-01-23 15:50 +0000 [53f57001f2] gtjoseph <george.joseph@fairview5.com> + + * loader: Retry dlopen when loading fails + + Although we use the RTLD_LAZY flag when calling dlopen + the first time on a module, this only defers resolution + for function calls. Pointer references to functions are + determined at link time so dlopen expects them to be there. + Since we don't cross-module link, pointers to functions + in other modules won't be available and dlopen will fail. + + Doing a "hardened" build also causes problems because it + typically sets "-z now" on the ld command line which + overrides RTLD_LAZY at run time. + + If the failing module isn't a GLOBAL_SYMBOLS module, then + dlopen will be called again after all the GLOBAL_SYMBOLS + modules have been loaded and they'll eventually resolve. + + If the calling module IS a GLOBAL_SYMBOLS module itself + and a third module depends on it, then there's an issue + because the second time through the dlopen loop, + GLOBAL_SYMBOLS modules aren't given any special treatment + and since the order in which dlopen is called isn't + deterministic, the dependent may again be tried before the + module it needs is loaded. + + Simple solution: Save modules that fail load_resource + because of a dlopen error in a list and retry them + immediately after the first pass. Keep retrying until + the failed list is empty or we reach a #defined max + retries. Error messages are suppressed until the final + pass which also gets rid of those confusing error messages + about module failures that are later corrected. + + Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb + +2016-03-01 16:18 +0000 [40d9e9e238] Kevin Harwell <kharwell@digium.com> + + * bridge.c: Crash during attended transfer when missing a local channel half + + It's possible for the transferer channel to get hung up early during the + attended transfer process. For instance, a phone may send a "bye" immediately + upon receiving a sip notify that contains a sip frag 100 (I'm looking at you + Jitsi). When this occurs a race begins between the transferer being hung up + and completion of the transfer code. + + If the channel hangs up too early during a transfer involving stasis bridging + for instance, then when the created local channel goes to look up its swap + channel (and associated datastore) it can't find it (since it is no longer in + the bridge) thus it fails to enter the stasis application. Consequently, the + created local channel(s) hang up as well. If the timing is just right then the + bridging code attempts to add the message link with missing local channel(s). + Hence the crash. + + Unfortunately, there is no great way to solve the problem of the unexpected + "bye". While we can't guarantee we won't receive an early hangup, and in this + case still fail to enter the stasis application, we can make it so asterisk + does not crash. + + This patch does just that by locking the local channel structure, checking + that the local channel's peer has not been lost, and then continuing. This + keeps the local channel's peer from being ripped out from underneath it by + the local/unreal hangup code while attempting to set the stasis message link. + + ASTERISK-25771 + + Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880 + +2016-03-01 18:08 +0000 [ff3da61c35] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 + + During the transfer process, some phones (okay it was the Jitsi softphone, + but maybe others are out there) send a "bye" immediately after receiving a + SIP Notify. When a "bye" is received early for some types of transfers the + transferer channel may no longer be available during late stage transfer + processing. + + For instance, during an attended transfer involving stasis bridging at one + point the created local channel looks for an associated swap channel in + order to retrieve the stasis application name. If the transferer has hung + up then the local channel will fail to find it. The local channel then has + no way to know which stasis app to enter, so it fails and hangs up as well. + Thus the transfer does not complete as expected. + + This patch delays the sending of the initial notify in order to give the + transfer process enough time to gather the necessary data for a successful + transfer. + + ASTERISK-25771 + + Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16 + +2016-03-03 08:26 +0000 [26b8f2692e] Joshua Colp <jcolp@digium.com> + + * res_pjsip_dtmf_info: NULL terminate the message body. + + PJSIP does not ensure that when printing the message body the + buffer will be NULL terminated. This is problematic when searching + for the signal and duration values of the DTMF. + + This change ensures the buffer is always NULL terminated. + + Change-Id: I52653a1a60c93092d06af31a27408d569cc98968 + +2016-03-01 20:03 +0000 [86d6e44cc1] gtjoseph <george.joseph@fairview5.com> + + * alembic: Fix downgrade and tweak for sqlite + + Downgrade had a few issues. First there was an errant 'update' statement in + add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we + weren't cleaning up the ENUMs so subsequent upgrades on postgres failed + because the types already existed. + + For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly. + Fortunately alembic batch_operations takes care of this for us if we + use it so the alter and drops were converted to use batch operations. + + Here's an example downgrade: + + with op.batch_alter_table('ps_endpoints') as batch_op: + batch_op.drop_column('tos_audio') + batch_op.drop_column('tos_video') + batch_op.add_column(sa.Column('tos_audio', yesno_values)) + batch_op.add_column(sa.Column('tos_video', yesno_values)) + batch_op.drop_column('cos_audio') + batch_op.drop_column('cos_video') + batch_op.add_column(sa.Column('cos_audio', yesno_values)) + batch_op.add_column(sa.Column('cos_video', yesno_values)) + + with op.batch_alter_table('ps_transports') as batch_op: + batch_op.drop_column('tos') + batch_op.add_column(sa.Column('tos', yesno_values)) + # Can't cast integers to YESNO_VALUES, so dropping and adding is required + batch_op.drop_column('cos') + batch_op.add_column(sa.Column('cos', yesno_values)) + + Upgrades from base to head and downgrades from head to base were tested + repeatedly for postgresql, mysql/mariadb, and sqlite3. + + Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8 + +2016-03-02 15:55 +0000 [6f0d7ce9db] gtjoseph <george.joseph@fairview5.com> + + * config_transport: Fix objects returned by ast_sip_get_transport_states + + ast_sip_get_transport_states was returning a container of internal_state + objects instead of ast_sip_transport_state objects. This was causing + transport lookups to fail, most noticably in res_pjsip_nat, which + couldn't find the correct external addresses. This was causing contacts + to go out with internal ip addresses. + + ASTERISK-25830 #close + Reported-by: Sean Bright + + Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e + +2016-03-02 11:17 +0000 [1ea7a5a774] Scott Griepentrog <scott@griepentrog.com> + + * CHAOS: cleanup possible null vars on msg alloc failure + + In message.c, if msg_alloc fails to init the string field, + vars may be null, so use a null tolerant cleanup. + + In res_pjsip_messaging.c, if msg_data_create fails, mdata + will be null, so use a null tolerant cleanup. + + ASTERISK-25323 + + Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56 + +2016-03-02 09:34 +0000 [3c37c7071f] Scott Griepentrog <scott@griepentrog.com> + + * CHAOS: prevent crash on failed strdup + + This patch avoids crashing on a null pointer + if the strdup() allocation fails. + + ASTERISK-25323 + + Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5 + +2016-02-29 18:11 +0000 [9633be9d25] Richard Mudgett <rmudgett@digium.com> + + * func_callerid.c: Update REDIRECTING reason documentation. + + Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386 + +2016-02-26 18:57 +0000 [4165ea7778] Richard Mudgett <rmudgett@digium.com> + + * SIP diversion: Fix REDIRECTING(reason) value inconsistencies. + + Previous chan_sip behavior: + + Before this patch chan_sip would always strip any quotes from an incoming + reason and pass that value up as the REDIRECTING(reason). For an outgoing + reason value, chan_sip would check the value against known values and + quote any it didn't recognize. Incoming 480 response message reason text + was just assigned to the REDIRECTING(reason). + + Previous chan_pjsip behavior: + + Before this patch chan_pjsip would always pass the incoming reason value + up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip + would send the reason value as passed down. + + With this patch: + + Both channel drivers match incoming reason values with values documented + by REDIRECTING(reason) and values documented by RFC5806 regardless of + whether they are quoted or not. RFC5806 values are mapped to the + equivalent REDIRECTING(reason) documented value and is set in + REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a + quoted string version ('"unconditional"') is converted to + REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal + with 'cfu' instead of any of the aliases. + + The incoming 480 response reason text supported by chan_sip checks for + known reason values and if not matched then puts quotes around the reason + string and assigns that to REDIRECTING(reason). + + Both channel drivers send outgoing known REDIRECTING(reason) values as the + unquoted RFC5806 equivalent. User custom values are either sent as is or + with added quotes if SIP doesn't allow a character within the value as + part of a RFC3261 Section 25.1 token. Note that there are still + limitations on what characters can be put in a custom user value. e.g., + embedding quotes in the middle of the reason string is silly and just + going to cause you grief. + + * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. + e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the + 'cfu' value. + + * Added missing malloc() NULL return check in res_pjsip_diversion.c + set_redirecting_reason(). + + * Fixed potential read from a stale pointer in res_pjsip_diversion.c + add_diversion_header(). The reason string needed to be copied into the + tdata memory pool to ensure that the string would always be available. + Otherwise, if the reason string returned by reason_code_to_str() was a + user's reason string then the string could be freed later by another + thread. + + Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87 + +2016-02-26 18:54 +0000 [41f4af4ce5] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. + + Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd + +2016-02-29 20:41 +0000 [4c5998ff55] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. + + * Fix double unref of other_party channel in off nominal path. + + * This is unlikely to be a real problem. However, for safety, + in handle_incoming_request() keep the datastore ref with the + other_party channel ref until we are finished with the other_party + channel. + + Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821 + +2016-01-18 21:54 +0000 [b59956a875] gtjoseph <george.joseph@fairview5.com> + + * build-system: Allow building with static pjproject + + Background here: + http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html + + From CHANGES: + * To help insure that Asterisk is compiled and run with the same known + version of pjproject, a new option (--with-pjproject-bundled) has been + added to ./configure. When specified, the version of pjproject specified + in third-party/versions.mak will be downloaded and configured. When you + make Asterisk, the build process will also automatically build pjproject + and Asterisk will be statically linked to it. Once a particular version + of pjproject is configured and built, it won't be configured or built + again unless you run a 'make distclean'. + + To facilitate testing, when 'make install' is run, the pjsua and pjsystest + utilities and the pjproject python bindings will be installed in + ASTDATADIR/third-party/pjproject. + + The default behavior remains building with the shared pjproject + installation, if any. + + Building: + + All you have to do is include the --with-pjproject-bundled option on + the ./configure command line (and remove any existing --with-pjproject + option if specified). Everything else is automatic. + + Behind the scenes: + + The top-level Makefile was modified to include 'third-party' in the + list of MOD_SUBDIRS. + + The third-party directory was created to contain any third party + packages that may be needed in the future. Its Makefile automatically + iterates over any subdirectories passing on targets. + + The third-party/pjproject directory was created to house the pjproject + source distribution. Its Makefile contains targets to download, patch + configure, generate dependencies, compile libs, apps and python bindings, + sanitized build.mak and generate a symbols list. + + When bootstrap.sh is run, it automatically includes the configure.m4 + file in third-party/pjproject. This file has a macro to download and + conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR + and PJPROJECT_BUNDLED. It also tests for the capabilities like + PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to + trying to compile. Of course, bootstrap.sh is only run once and the + configure file is incldued in the patch. + + When configure is run with the new options, the macro in configure.m4 + triggers the download, patch, conifgure and tests. No compilation is + performed at this time. The downloaded tarball is cached in /tmp so + it doesn't get downloaded again on a distclean. + + When make is run in the top-level Asterisk source directory, it will + automatically descend all the subdirectories in third_party just as it + does for addons, apps, etc. The top-level Makefile makes sure that + the 'third-party' is built before 'main' so that dependencies from the + other directories are built first. + + When main does build, a new shared library (libasteriskpj) is created that + links statically to the pjproject .a files and exports all their symbols. + The asterisk binary links to that, just as it does with libasteriskssl. + + When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject + python bindings are installed in ASTDATADIR/third-party/pjproject. This + will facilitate testing, including running the testsuite which will be + updated to check that directory for the pjsua module ahead of the system + python library. + + Modules should continue to depend on pjproject if they use pjproject APIs + directly. They should not care about the implementation. No changes to any + res_pjsip modules were made. + + Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103 + +2016-02-22 16:59 +0000 [18a323e542] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Fix T.38 issues caused by leaving a bridge. + + chan_sip could not handle AST_T38_TERMINATED frames being sent to it when + the channel left the bridge. The action resulted in overlapping outgoing + reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not + happy. + + * Force T.38 to be remembered as locally bridged. Now when the channel + leaves the native RTP bridge after T.38, the channel remembers that it has + already reINVITEed the media back to Asterisk. It just needs to terminate + T.38 when the AST_T38_TERMINATED arrives. + + * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant + AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if + they happen before the T.38 state changes to disabled. Now the T.38 state + is set to disabled before the reINVITE is sent. + + ASTERISK-25582 #close + + Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce + +2016-02-18 18:27 +0000 [263a39f2cc] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_t38.c: Back out part of an earlier fix attempt. + + This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d + commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame + that it puts into the bridge may or may not be processed by the time the + bridged peer is kicked out of the bridge. If it is processed then all is + well. However, if it is not processed then that channel is stuck in fax + mode until it hangs up or maybe if it joins another bridge for T.38 + faxing. + + ASTERISK-25582 + + Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7 + +2016-02-22 13:54 +0000 [221422be50] Richard Mudgett <rmudgett@digium.com> + + * bridge core: Add owed T.38 terminate when channel leaves a bridge. + + The channel is now going to get T.38 terminated when it leaves the + bridging system and the bridged peers are going to get T.38 terminated as + well. + + ASTERISK-25582 + + Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7 + +2016-02-19 16:01 +0000 [0a5bc64491] Richard Mudgett <rmudgett@digium.com> + + * channel api: Create is_t38_active accessor functions. + + ASTERISK-25582 + + Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b + +2016-02-19 19:06 +0000 [513638a5f4] Richard Mudgett <rmudgett@digium.com> + + * bridge_channel: Don't settle owed events on an optimization. + + Local channel optimization could cause DTMF digits to be duplicated. + Pending DTMF end events would be posted to a bridge when the local channel + optimizes out and is replaced by the channel further down the chain. When + the real digit ends, the channel would get another DTMF end posted to the + bridge. + + A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B + + 1) LocalA has the /n flag to prevent optimization. + 2) B is sending DTMF to A through the local channel chain. + 3) When LocalB optimizes out it can move B to the position of LocalB;1 + 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would + settle an owed DTMF end to the bridge toward LocalA;2. + 5) When B finally ends its DTMF it sends the DTMF end down the chain. + 6) Without this patch, A would hear the DTMF digit end when LocalB + optimizes out and when B ends the original digit. + + ASTERISK-25582 + + Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251 + +2016-02-22 12:15 +0000 [7c4495cb70] Richard Mudgett <rmudgett@digium.com> + + * channel.c: Route all control frames to a channel through the same code. + + Frame hooks can conceivably return a control frame in exchange for an + audio frame inside ast_write(). Those returned control frames were not + handled quite the same as if they were sent to ast_indicate(). Now it + doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a + channel or ast_indicate(). + + ASTERISK-25582 + + Change-Id: I5775f41421aca2b510128198e9b827bf9169629b + +2016-02-25 15:13 +0000 [48d713a832] gtjoseph <george.joseph@fairview5.com> + + * sorcery: Refactor create, update and delete to better deal with caches + + The ast_sorcery_create, update and delete function have been refactored + to better deal with caches and errors. + + The action is now called on all non-caching wizards first. If ANY succeed, + the action is called on all caching wizards and the observers are notified. + This way we don't put something in the cache (or update or delete) before + knowing the action was performed in at least 1 backend and we only call the + observers once even if there were multiple writable backends. + + ast_sorcery_create was never adding to caches in the first place which + was preventing contacts from getting added to a memory_cache when they + were created. In turn this was causing memory_cache to emit errors if + the contact was deleted before being retrieved (which would have + populated the cache). + + ASTERISK-25811 #close + Reported-by: Ross Beer + + Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46 +2016-02-25 15:39 +0000 [ee947d4a7a] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s. + + There are a few cases where we're emitting notices or warnings + for things that really need neither, like a client retrying to subscribe + to mwi when they're not conifgured for it. They get a 404 so there's no + need for non-debug messages. + + Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f +2016-02-25 14:17 +0000 [6e70e8ccdb] gtjoseph <george.joseph@fairview5.com> + + * res_sorcery_memory_cache: Fix SEGV in some CLI commands + + A few of the CLI commands weren't checking for enough arguments + and were SEGVing. + + Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413 + +2016-02-25 10:29 +0000 [4417f64d83] Leif Madsen <leif@leifmadsen.com> + + * Add initial support to build Docker images + + This work-in-progress is the first step to being able to reliably + build Asterisk containers from the Asterisk source. I'm submitting + this based on feedback gained at AstriDevCon 2015. + + Information about how to use this is provided in contrib/docker/README.md + and will result in a local Asterisk container being built right from + your source. I believe this can eventually be automated via + hub.docker.com. + + Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1 + +2016-02-22 19:31 +0000 [e7a6abbbd3] Richard Mudgett <rmudgett@digium.com> + + * rtp_engine.h: Remove extraneous semicolons. + + Change-Id: Ib462633d396fa941379dfef648dcd2245e350084 + +2016-02-23 14:57 +0000 [6656afffa0] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Suppress T.38 SDP c= line if addr is the same. + + Use the correct comparison function since we only care if the address + without the port is the same. + + Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0 + +2016-02-16 08:14 +0000 [ea9deff996] Christof Lauber <christof.lauber@annax.ch> + + * res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables + + Introduced realloaction of ast_str buf in sqlite3_escape functions in case + the returned buffer from threadstorage was actually too small. + + Change-Id: I3c5eb43aaade93ee457943daddc651781954c445 + +2016-02-11 11:01 +0000 [d2a1457e0b] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip/config_transport: Allow reloading transports. + + The 'reload' mechanism actually involves closing the underlying + socket and calling the appropriate udp, tcp or tls start functions + again. Only outbound_registration, pubsub and session needed work + to reset the transport before sending requests to insure that the + pjsip transport didn't get pulled out from under them. + + In my testing, no calls were dropped when a transport was changed + for any of the 3 transport types even if ip addresses or ports were + changed. To be on the safe side however, a new transport option was + added (allow_reload) which defaults to 'no'. Unless it's explicitly + set to 'yes' for a transport, changes to that transport will be ignored + on a reload of res_pjsip. This should preserve the current behavior. + + Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf + +2016-02-07 17:34 +0000 [6b921f706d] gtjoseph <george.joseph@fairview5.com> + + * res_pjproject: Add ability to map pjproject log levels to Asterisk log levels + + Warnings and errors in the pjproject libraries are generally handled by + Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings + or errors so the messages emitted by pjproject directly are either superfluous + or misleading. A good exampe of this are the level-0 errors pjproject emits + when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't + consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS + client be treated any differently? + + A config file for res_pjproject has bene added (pjproject.conf) and a new + log_mappings object allows mapping pjproject levels to Asterisk levels + (or nothing). The defaults if no pjproject.conf file is found are the same + as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR, + 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level> + + Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898 + +2016-02-18 10:55 +0000 [f295088764] Alexei Gradinari <alex2grad@gmail.com> + + * res_pjsip_outbound_publish: Fix processing 412 response + + When Asterisk receives a 412 (Conditional Request Failed) response + it has to recreate publish session. + There is bug in res_pjsip_outbound_publish.c + The function sip_outbound_publish_client_alloc is called with wrong object + while processing 412 (Conditional Request Failed) response. + This patch fixes it. + + ASTERISK-25229 #close + + Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359 +2016-02-18 11:15 +0000 [f1f79812c1] Mark Michelson <mmichelson@digium.com> + + * Fix failing threadpool_auto_increment test. + + The threadpool_auto_increment test fails infrequently for a couple of + reasons + * The threadpool listener was notified of fewer tasks being pushed than + were actually pushed + * The "was_empty" flag was set to an unexpected value. + + The problem is that the test pushes three tasks into the threadpool. + Test expects the threadpool to essentially gather those three tasks, and + then distribute those to the threadpool threads. It also expects that as + the tasks are pushed in, the threadpool listener is alerted immediately + that the tasks have been pushed. In reality, a task can be distributed + to the threadpool threads quicker than expected, meaning that the + threadpool has already emptied by the time each subsequent task is + pushed. In addition, the internal threadpool queue can be delayed so + that the threadpool listener is not alerted that a task has been pushed + even after the task has been executed. + + From the test's point of view, there's no way to be able to predict + exactly the order that task execution/listener notifications will occur, + and there is no way to know which listener notifications will indicate + that the threadpool was previously empty. + + For this reason, the test has been updated to only check the things it + can check. It ensures that all tasks get executed, that the threads go + idle after the tasks are executed, and that the listener is told the + proper number of tasks that were pushed. + + Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c + +2016-02-16 23:37 +0000 [79dc5e2f00] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com> + + * app_queue: fix Calculate talktime when is first call answered + + Fix calculate of average time for talktime is wrong when is completed the + first call beacuse the time for talked would be that call. + + ASTERISK-25800 #close + + Change-Id: I94f79028935913cd9174b090b52bb300b91b9492 + +2016-02-17 13:30 +0000 [5a3a857dd6] Richard Mudgett <rmudgett@digium.com> + + * cel.c: Fix mismatch in ast_cel_track_event() return type. + + The return type of ast_cel_track_event() is not large enough to return all + 64 potential bits of the event enable mask. Fortunately, the defined CEL + events do not really need all 64 bits and the return value is only used to + determine if the requested CEL event is enabled. + + * Made the ast_cel_track_event() return 0 or 1 only so the return value + can fit inside an int type instead of zero or a truncated 64 bit non-zero + value. + + Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c + +2016-02-16 16:37 +0000 [87ab65c557] gtjoseph <george.joseph@fairview5.com> + + * res_odbc: Fix exports.in for missing symbols + + res_odbc.exports.in was missing a few symbols. + Changed to wildcards. + + Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c + +2016-02-16 12:20 +0000 [c0f3062031] gtjoseph <george.joseph@fairview5.com> + + * res_statsd: Fix exports.in for missing symbols + + res_statsd.export.in was missing the _va variations of the log + functions causing Asterisk to crash in res_pjsip if OPTIONAL_API + wasn't enabled. + + ASTERISK-25727 #close + Reported-by: Gergely Dömsödi + + Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b + +2016-02-15 21:31 +0000 [5e848dae7b] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_config_wizard: Add command to export primitive objects + + A new command (pjsip export config_wizard primitives) has been added that + will export all the pjsip objects it created to the console or a file + suitable for reuse in a pjsip.conf file. + + ASTERISK-24919 #close + Reported-by: Ray Crumrine + + Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b + +2016-02-15 15:37 +0000 [34c64707d1] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip_caller_id: Fix segfault when replacing rpid or pai header + + If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid + or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify + the header added by the dialplan function. Since the header added by the + dialplan function is generic string, there are no virtual functions to parse + the uri and we get a segfault when we try. Since the modify, was really only + an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER + and recreate it. + + This raises a question for another time though: What should happen with + duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups + so if it's session supplement is loaded after res_pjsip_caller_id's (or any + other module that adds headers), there'll be dups in the message. + + ASTERISK-25337 #close + + Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa + +2016-02-15 13:08 +0000 [ebe167f792] Mark Michelson <mmichelson@digium.com> + + * Fix creation race of contact_status structures. + + It is possible when processing a SIP REGISTER request to have two + threads end up creating contact_status structures in sorcery. + contact_status is created using a "find or create" function. If two + threads call into this at the same time, each thread will fail to find + an existing contact_status, and so both will end up creating a new + contact status. + + During testing, we would see sporadic failures because the + PJSIP_CONTACT() dialplan function would operate on a different + contact_status than what had been updated by res_pjsip/pjsip_options. + + The fix here is two-fold: + 1) The "find or create" function for contact_status now has a lock + around the entire operation. This way, if two threads attempt the + operation simultaneously, the first to get there will create the object, + and the second will find the object created by the first thread. + + 2) res_sorcery_memory has had its create callback updated so that it + will not allow for objects with duplicate IDs to be created. + + Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97 + +2016-02-15 12:52 +0000 [1c4f2a920d] Joshua Colp <jcolp@digium.com> + + * res_pjsip_pubsub: Move where the subscription is stored to after initialized. + + A problem arose when testing the AMI subscription listing actions where it + was possible for a subscription that had not been fully initialized to be + listed. This was problematic as the underlying listing code would crash. + + This change makes it so the subscription tree is fully set up before it is + added to the list of subscriptions. This ensures that when the listing actions + get the subscription it is valid. + + ASTERISK-25738 #close + + Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48 + +2015-02-20 20:51 +0000 [ac00c6bc2d] Corey Farrell <git@cfware.com> + + * main/asterisk.c: Reverse #if statement in listener() to fix code folding. + + listener() opens the same code block in two places (#if and #else). This + confuses some folding editors causing it to think that an extra code block + was opened. Folding in 'geany' causes all code after listener() to be + folded as if it were part of that procedure. + + ASTERISK-24813 #close + + Change-Id: I4b8c766e6c91e327dd445e8c18f8a6f268acd961 + +2016-02-09 17:34 +0000 [b1b797e0e7] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Refactor load_module/unload_module + + load_module was just too hairy with every step having to clean up all + previous steps on failure. + + Some of the pjproject init calls have now been moved to a separate + load_pjsip function and the unload_pjsip function was enhanced to clean + up everything if an error happened at any stage of the load process. + + In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns + and ast_threadpool_shutdowns were also corrected. + + Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302 + +2016-02-09 22:42 +0000 [20e9792fbc] Badalyan Vyacheslav <slavon.net@gmail.com> + + * Resources/res_phoneprov: fix memory leak and heap-use-after-free + + * heap-use-after-free happens when we free "cfg" + but then use "value" which refers to it + + * A memory leak occurs because in some cases + it is not released "defaults" + + ASTERISK-25721 #close + Reported by: Badalyan Vyacheslav + Tested by: Badalyan Vyacheslav + + Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469 + +2016-02-11 11:21 +0000 [962a9d61f8] Etienne Lessard (license #6394) + + * func_iconv: Ensure output strings are properly terminated. + + ASTERISK-25272 #close + Reported by: Etienne Lessard + patches: + AST-25272.patch submitted by Etienne Lessard (license #6394) + + Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17 + +2016-02-10 16:16 +0000 [c1bf014ea0] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Handle pjsip_dlg_create_uas deprecation + + Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with + pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically + increments the lock on the returned dialog. To account for this, configure.ac + now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c + has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use + the original call or the new one. If the new one was used, the ref count is + decremented before returning. + + ASTERISK-25751 #close + Reported-by Josh Colp + + Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8 + +2016-02-09 23:40 +0000 [bd07b6f0dd] Badalyan Vyacheslav <slavon.net@gmail.com> + + * Build: Added testing compiler to support the system sanitizes + + In older versions of the compiler was not sanitizes. + Compilers other than GCC can not support the Usan and TSAN + or have other options for *FLAGS. + + ASTERISK-25767 #close + Reported by: Badalyan Vyacheslav + Tested by: Badalyan Vyacheslav + + Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916 + +2016-02-09 20:57 +0000 [e9e896abd1] Badalyan Vyacheslav <v.badalyan@open-bs.ru> + + * Build: Fix menuselect USAN conflicts + + USAN can be used together with other sanitizers. + + Reported by: Badalyan Vyacheslav + Tested by: Badalyan Vyacheslav + + Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f + +2016-02-09 14:21 +0000 [93e8ed0154] Corey Farrell <git@cfware.com> + + * Simplify and fix conditional in FD_SET. + + FD_SET contains a conditional statement to protect against buffer + overruns. The statement was overly complicated and prevented use + of the last array element of ast_fdset. We now just verify the fd + is less than ast_FDMAX. + + Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40 + +2016-02-09 07:11 +0000 [a7c8d4cd6b] Joshua Colp <jcolp@digium.com> + + * tests/test_sorcery_memory_cache_thrash: Improve termination process. + + When terminating the threads thrashing a sorcery memory cache each + would be told to stop and then we would wait on them. During at + least one thrashing test this was problematic due to the specific + usage pattern in use. It would take some time for termination of the + thread to occur. + + This would occur due to contention between the threads retrieving + and the threads updating the cache. As the retrieving threads are + given priority it may be some time before the updating threads + are able to proceed. + + This change makes it so all threads are told to stop and then each + are joined to ensure they stop. This way all the threads should + stop at around the same time instead of waiting for one to stop, + the next to stop, then the next, and so on. As a result of this + the execution time for each thrash test is much closer to their + expected value than previously seen as well. + + Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827 +2016-01-29 17:56 +0000 [2451d4e455] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Fix infinite recursion when loading transports from realtime + + Attempting to load a transport from realtime was forcing asterisk into an + infinite recursion loop. The first thing transport_apply did was to do a + sorcery retrieve by id for an existing transport of the same name. For files, + this just returns the previous object from res_sorcery_config's internal + container, if any. For realtime, the res_sourcery_realtime driver looks in the + database and finds the existing row but now it has to rehydrate it into a + sorcery object which means calling... transport_apply. And so it goes. + + The main issue with loading from realtime (apart from the loop) was that + transport stores structures and pointers directly in the ast_sip_transport + structure instead of the separate ast_transport_state structure. This patch + separates those items into the ast_sip_transport_state structure. The pattern + is roughly the same as res_pjsip_outbound_registration. + + Although all current usages of ast_sip_transport and ast_sip_transport_state + were modified to use the new ast_sip_get_transport_state API, the original + items are left in ast_sip_transport and kept updated to maintain ABI + compatability for third-party modules. They are marked as deprecated and + noted that they're now in ast_sip_transport_state. + + ASTERISK-25606 #close + Reported-by: Martin Moučka + + Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19 + +2016-01-25 17:36 +0000 [6f978fbfe5] Richard Mudgett <rmudgett@digium.com> + + * app_confbridge: Only use b_profile options from the conference. + + A user cannot set new bridge options after the conference is created by + the first user. Attempting to do so is documented as undefined behavior. + + This patch ensures that the bridge profile options used are from the + conference and not what a subsequent user may have tried to set. + + Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266 + +2016-02-05 10:29 +0000 [ec8fd6714d] gtjoseph <george.joseph@fairview5.com> + + * chan_misdn: Fix a few issues causing compile errors + + Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98 + +2016-02-04 16:17 +0000 [6a799cd78f] Mark Michelson <mmichelson@digium.com> + + * Check for OpenSSL defines before trying to use them. + + The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior + to OpenSSL version 1.0.1. A recent commit attempts to, by default, set + these options, which can cause problems on systems with older OpenSSL + installations. + + This commit adds a configure script check for those defines and will not + attempt to make use of those if they do not exist. We will print a + warning urging the user to upgrade their OpenSSL installation if those + defines are not present. + + Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d +2016-02-03 14:25 +0000 [953d1cc11a] gtjoseph <george.joseph@fairview5.com> + + * pjsip/alembic: Add missing columns to system and registration + + ps_systems needed disable_tcp_switch + ps_registrations needed line and endpoint + + ASTERISK-25737 #close + + Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19 + +2016-02-04 11:39 +0000 [23829b3253] Mark Michelson <mmichelson@digium.com> + + * res_stasis_device_state: Fix refcounting error. + + Device state subscription lifetimes were governed by when the + subscription was established and unsubscribed from. However, it is + possible that at the time of unsubscription, there could be device state + events still in flight. When those device state events occur, the device + state callback could attempt to dereference a freed pointer. Crash. + + This change ensures that the lifetime of the device state subscription + does not end until the underlying stasis subscription has confirmed that + its final message has been sent. + + Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2 + +2016-01-27 10:44 +0000 [4e8e6d3922] Sean Bright <sean.bright@gmail.com> + + * res_rtp_asterisk: Allow ICE host candidates to be overriden + + During ICE negotiation the IPs of the local interfaces are sent to the remote + peer as host candidates. In many cases Asterisk is behind a static one-to-one + NAT, so these host addresses will be internal IP addresses. + + To help in hiding the topology of the internal network, this patch adds the + ability to override the host candidates by matching them against a + user-defined list of replacements. + + Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f + +2015-12-07 12:46 +0000 [c6b1b2b1c8] Richard Mudgett <rmudgett@digium.com> + + * AST-2016-003 udptl.c: Fix uninitialized values. + + Sending UDPTL packets to Asterisk with the right amount of missing + sequence numbers and enough redundant 0-length IFP packets, can make + Asterisk crash. + + ASTERISK-25603 #close + Reported by: Walter Doekes + + ASTERISK-25742 #close + Reported by: Torrey Searle + + Change-Id: I97df8375041be986f3f266ac1946a538023a5255 +2016-02-03 12:05 +0000 [f8acadde2c] Joshua Colp <jcolp@digium.com> + + * AST-2016-001 http: Provide greater control of TLS and set modern defaults. + + This change exposes the configuration of various aspects of the TLS + support and sets the default to the modern standards. + + The TLS cipher is now set to the best values according to the + Mozilla OpSec team, different TLS versions can now be disabled, and + the cipher order can be forced to be that of the server instead of + the client. + + ASTERISK-24972 #close + + Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8 +2015-09-28 17:07 +0000 [3c81a052c8] Richard Mudgett <rmudgett@digium.com> + + * AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow. + + Setting the sip.conf timert1 value to a value higher than 1245 can cause + an integer overflow and result in large retransmit timeout times. These + large timeout times hold system file descriptors hostage and can cause the + system to run out of file descriptors. + + NOTE: The default sip.conf timert1 value is 500 which does not expose the + vulnerability. + + * The overflow is now detected and the previous timeout time is + calculated. + + ASTERISK-25397 #close + Reported by: Alexander Traud + + Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290 +2016-02-03 14:07 +0000 [2a6ee8caeb] gtjoseph <george.joseph@fairview5.com> + + * logging: Remove/fix some message annoyances + + test_dlinklists doesn't need to NOTICE everyone that every macro worked. + + res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or + provider was registered. + + res_odbc was missing a newline at the end of one message. + + Change-Id: I6c06361518ef3711821795e535acd439782a995e + +2016-02-02 10:52 +0000 [32fc784284] Alexei Gradinari License #5691 + + * res_sorcery_realtime: Fix regex regression. + + A regression was introduced where searching for realtime PJSIP objects + by regex by starting the regex with a leading "^" would cause no items + to be returned. + + This was due to a change which attempted to drop the requirement for a + leading "^" to be present due to how some CLI commands formulate their + regexes. However, the change, rather than simply eliminating the + requirement, caused any regexes that did begin with "^" to end up not + returning the expected results. + + This change fixes the problem by inspecting the regex and formulating + the realtime query differently depending on if it begins with "^". + + ASTERISK-25702 #close + Reported by Nic Colledge + + Patches: + realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691 + + Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693 + +2016-02-02 04:05 +0000 [0405c31756] Karsten Wemheuer <kwe-digium@iptam.com> + + * res_xmpp: Does not connect in component mode + + The module res_xmpp does not accept usernames in the form used in component + mode (XEP-0114). In component mode there is no @something in the name. + In component mode the connection is now not dropped anymore. + + If the xmpp server sends out a "stream" tag before handshake is finished, + the connection gets dropped in res_xmpp. Now this tag will be ignored and + the connection will be established. + + After connecting there will be an exchange of presence states. This does + not work as expected in component mode. The responsible function + "xmpp_pak_presence" is left before the states get sent out. Sending + presence states in component mode is now moved to the top of the function. + + ASTERISK-25735 #close + + Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c +2016-02-01 13:04 +0000 [8804d0973c] gtjoseph <george.joseph@fairview5.com> + + * build_system: Fix some warnings highlighted by clang + + Fix some warnings found with clang. + + Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd + +2016-02-01 13:16 +0000 [109b0aff6b] gtjoseph <george.joseph@fairview5.com> + + * res/Makefile: Fix bug in "clean" target for ari + + The "clean" target was attempting to clean res/ari from inside + the res directory which doesn't remove anything. Removed the res/ + prefix. + + Change-Id: Ib1a518d54efa81b9fd5a42742d43cc3767435bf6 + +2016-01-31 20:13 +0000 [a85fab7c44] gtjoseph <george.joseph@fairview5.com> + + * pjsip/alembic: Fix definition of qualify_timeout + + A recent commit set qualify_timeout to Decimal which isn't supported. + This path corrects it to Float. + + Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf + +2016-01-29 07:39 +0000 [aa9348ab9a] Stefan Engström <stefanen@kth.se> + + * chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip. + + When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a) + AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect + asterisk to include the same value for its own ip in both cases a) and b), + but it seems a) produces a contact header like Contact: + <sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like + <sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf + + My guess is that manager_sipnotify should call + ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does, + because after applying this patch, both cases a) and b) produce + the contact header that I expect: <sip:asterisk@192.168.1.227:8060> + + Reported by: Stefan Engström + Tested by: Stefan Engström + + Change-Id: I86af5e209db64aab82c25417de6c768fb645f476 +2015-12-23 15:07 +0000 [65bd4fcc3f] Mark Michelson <mmichelson@digium.com> + + * res_odbc: Remove connection management + + Asterisk by default will create a single database connection and share + it among all threads that attempt to access the database. In previous + versions of Asterisk, this was tolerable, because the most used channel + driver, chan_sip, mostly accessed the database from a single thread. + With PJSIP, however, many threads may be attempting to perform database + operations, and there is the potential for many more database accesses, + meaning the concurrency is a horrible bottleneck if only one connection + is shared. + + Asterisk has a connection pooling facility built into it, but the + implementation has flaws. For one, there is a strict limit on the number + of simultaneous connections that could be made to the database. Anything + beyond the maximum would result in a failed operation. Attempting to + predict what the maximum should be is nearly impossible even for someone + intimately familiar with Asterisk's threading model. In addition, use of + transactions in the dialplan can cause some severe bugs if connection + pooling is enabled. + + This commit seeks to fix the concurrency problem by removing all + connection management code from Asterisk and leaving that to the + underlying unixODBC code instead. Now, Asterisk does not share a single + connection, nor does it try to maintain a connection pool. Instead, all + Asterisk ever does is request a connection from unixODBC and allow + unixODBC to either allocate those connections or retrieve them from a + pool. + + Doing this has a bit of a ripple effect. For one, since connections are + not long-lived objects, several of the safeguards that previously + existed have been removed. We don't have to worry about trying to use a + connection that has gone stale. In every case, when we request a + connection, it has just been made and we don't need to perform any + sanity checks to be sure it's still active. + + Another major player affected by this change is transactions. + Transactions and their respective connections were so tightly coupled + that it was almost pornographic. This code change moves + transaction-related code to its own file separate from the core ODBC + functionality. This way, the core of ODBC does not even have to know + that transactions exist. + + In making this large change, I had to look at a lot of code and + understand it. When making this change, I discovered several places + where the behavior is definitely not ideal, but it seemed outside the + scope of this change to be fixing it. Instead, any place where I saw + some sort of room for improvement has had a XXX comment added explaining + what could be altered to improve it. + + Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf + +2016-01-28 12:44 +0000 [2a9e623ff9] Richard Mudgett <rmudgett@digium.com> + + * config_options.c: Fix warning message wording. + + Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4 + +2016-01-25 17:34 +0000 [ed3c9c1512] Richard Mudgett <rmudgett@digium.com> + + * app_confbridge.c: Replace inlined code with existing function. + + Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51 + +2016-01-25 16:05 +0000 [1d0abf86e7] Richard Mudgett <rmudgett@digium.com> + + * app_confbridge: Add ability to get the muted conference state. + + * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. + + * Added Muted header to AMI ConfbridgeListRooms action response list + events to indicate the muted conference state. + + * Added Muted column to CLI "confbridge list" output to indicate the muted + conference state and made the locked column a yes/no value instead of a + locked/unlocked value. + + ASTERISK-20987 + Reported by: hristo + + Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1 + +2016-01-26 17:59 +0000 [f0d40afa69] Richard Mudgett <rmudgett@digium.com> + + * app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. + + Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7 + +2016-01-25 15:48 +0000 [3e51e5c7fd] Richard Mudgett <rmudgett@digium.com> + + * app_confbridge: Make non-admin users join a muted conference muted. + + ASTERISK-20987 #close + Reported by: hristo + + Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38 + +2016-01-27 13:02 +0000 [9da18af992] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Add res_pjproject dependency to UPGRADE.txt and samples + + Since res_pjsip now depends on res_pjproject, this is now mentioned + in UPGRADE.txt and the basic-pbx modules.conf has been updated. + + Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d +2016-01-27 10:29 +0000 [aee8448bc2] gtjoseph <george.joseph@fairview5.com> + + * build_system: Prevent goals needing makeopts from running when it's missing + + The Makefile only optionally includes makeopts so when goals like uninstall that + dont depend on anything else are run after a distclean, rules like + 'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts + to remove everything in the root directory. + + Although there's a rule defined for makeopts which prints a message and does + an 'exit 1', since '-include makepopts' was specified (with the -), the exit + was ignored letting the rest of the rules run. + + This patch makes makeopts required unless the goal has the string 'clean' in it. + + ASTERISK-25730 #close + Reported-by: George Joseph + + Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7 + +2016-01-25 09:35 +0000 [f22074e5d9] Joshua Colp <jcolp@digium.com> + + * config: Allow options to register when documentation is unavailable. + + The config options framework is strict in that configuration options must + be documented unless XML documentation support is not available. In + practice this is useful as it ensures documentation exists however in + off-nominal cases this can cause strange problems. + + If it is expected that a config option has a non-zero or non-empty + default value but the config option documentation is unavailable + this reasonable expectation will not be met. This can cause obscure + crashes and weirdness depending on how the code handles it. + + This change tweaks the behavior to ensure that the config option + is still allowed to register, apply default values, and be set when + devmode is not enabled. If devmode is enabled then the option can + NOT be set. + + This also does not remove the initial documentation error message that + is output on load when registering the configuration option. + + ASTERISK-25725 #close + + Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8 + +2016-01-25 10:23 +0000 [4a3275abb9] Mark Michelson <mmichelson@digium.com> + + * Stasis: Use custom structure when setting variables. + + A recent change to queue channel variable setting to the Stasis control + queue caused a regression. When setting channel variables, it is + possible to give a NULL channel variable value in order to unset the + variable (i.e. remove it from the channel variable list). The change + introduced a call to ast_variable_new(), which is not tolerant of NULL + channel variable values. + + This new change switches from using ast_variable to using a custom + channel variable struct that is lighter weight and NULL value-tolerant. + + Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d + +2016-01-25 16:56 +0000 [b2c8a99f9e] Rusty Newton <rnewton@digium.com> + + * sounds/Makefile: Incremented core and extra sounds versions to 1.5 + + Core and extra sounds 1.5 was recently released! The tarballs contain + change descriptions however I figure more people will see this one so + I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra + to Core for en, en_GB, fr and added for languages that didn't already + have Extra sound sets (it,ja,ru). + + In addition all of the English and Russian sounds have been completely + re-recorded. + + Sounds moved and added: + activated,added,all-circuits-busy-now,astcc-followed-by-pound + at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy + ,call-fwd-unconditional,calling,call-waiting,cancelled, + cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated + ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist + ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello + ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to + ,location,number,number-not-answering,num-was-successfully,one-moment-please + ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option + ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached + ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial + ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension + ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered + ,your + + There were also a few random fixes here and there to file names for a few + of the languages. + + ASTERISK-25068 #close + + Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3 +2016-01-25 16:51 +0000 [8261bda1bf] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Prevent crash from AMI command on freed subscription. + + A test recently uncovered that running an ill-timed AMI command to show + inbound subscriptions could cause a crash since Asterisk will try to + operate on a freed subscription. + + The fix for this is to remove the subscription tree from the list of + subscriptions at the time that we are sending our final NOTIFY request + out. This way, as the subscription is in the process of dying, it is + inaccessible from AMI. + + Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23 + +2016-01-25 11:03 +0000 [a6823bb0c4] Corey Farrell <git@cfware.com> + + * chan_sip: Fix buffer overrun in sip_sipredirect. + + sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer + of 256 characters. This patch reduces the copy to 255 characters to leave + room for the string null terminator. + + ASTERISK-25722 #close + + Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab + +2016-01-22 15:08 +0000 [1003c2eb05] Mark Michelson <mmichelson@digium.com> + + * Stasis: Fix potential memory leak of control data. + + When queuing tasks onto the Stasis control queue, you can pass an + arbitrary data pointer and a function to free that data. All ARI + commands that use the Stasis control queue made the assumption that the + destructor function would be called in all paths, whether the task was + queued successfully or not. However, this was not correct. If a task was + queued onto a control structure that was already completed, the + allocated data would not be freed properly. + + This patch corrects this by making sure that all return paths call the + data destructor. + + Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb + +2016-01-21 10:58 +0000 [eedd77fda0] Mark Michelson <mmichelson@digium.com> + + * Stasis: Use control queue to prevent crash. + + A crash occurred when attempting to set a channel variable on a channel + that had already been hung up. This is because there is a small window + between when a control is grabbed and when the channel variable is set + that the channel can be hung up. + + The fix here is to queue the setting of the channel variable onto the + control queue. This way, the manipulation of the channel happens in a + thread where it is safe to be done. + + In this change, I also noticed that the setting of bridge roles on + channels was being done outside of the control queue, so I also changed + those operations to be done in the control queue. + + ASTERISK-25709 #close + Reported by Mark Michelson + + Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78 + +2016-01-22 11:48 +0000 [1c95b211a0] Richard Mudgett <rmudgett@digium.com> + + * logger.c: Fix buffer overrun found by address sanitizer. + + The null terminator of the tail struct member was not being allocated + when no logger.conf config file is installed. + + ASTERISK-25714 #close + Reported by: Badalian Vyacheslav + + Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30 + +2016-01-21 16:40 +0000 [6ff945ab87] Corey Farrell <git@cfware.com> + + * Build System: Add support for checking alembic branches. + + * Add 'check-alembic' target to root Makefile. + * Create build_tools/make_check_alembic to do the actual checks. + + ASTERISK-25685 + + Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004 + +2016-01-19 18:20 +0000 [02035212de] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case. + + ASTERISK-25712 #close + Reported by: Richard Mudgett + + Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f + +2016-01-18 03:49 +0000 [c68c66c61f] Diederik de Groot <ddegroot@talon.nl> + + * main/asterisk.c: ast_el_read_char + + Make sure buf[res] is not accessed at res=-1 (buffer underrun). + Address Sanitizer will complain about this quite loudly. + + ASTERISK-24801 #close + + Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9 + +2016-01-13 16:49 +0000 [f87c3275cc] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Add CLI "pjsip dump endpt [details]" + + Dump the res_pjsip endpt internals. + + In non-developer mode we will not document or make easily accessible the + "details" option even though it is still available. The user has to know + it exists to use it. Presumably they would also be aware of the potential + crash warning below. + + Warning: PJPROJECT documents that the function used by this CLI command + may cause a crash when asking for details because it tries to access all + active memory pools. + + Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb + +2016-01-18 17:16 +0000 [46b2de55f9] Matt Jordan <mjordan@digium.com> + + * funcs/func_cdr: Correctly report high precision values for duration and billsec + + When CDRs were refactored, func_cdr's ability to report high precision values + for duration and billsec (the 'f' option) was broken. This was due to func_cdr + incorrectly interpreting the duration/billsec values provided by the CDR engine + in milliseconds, as opposed to seconds. Since the CDR engine only provides + duration and billsec in seconds, and does not expose either attribute with + sufficient precision to merely pass back the underlying value, this patch fixes + the bug by re-calculating duration and billsec with microsecond precision based + on the start/answer/end times on the CDR. + + ASTERISK-25179 #close + + Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841 + +2016-01-18 19:20 +0000 [137fe5ae01] gtjoseph <george.joseph@fairview5.com> + + * res_pjproject: Add module providing pjproject logging and utils + + res_pjsip_log_forwarder has been renamed to res_pjproject + and enhanced as follows: + + As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch, + a new ast_pjproject_get_buildopt function has been added. It + allows the caller to get the value of one of the buildopts. + + The initial use case is retrieving the runtime value of + PJ_MAX_HOSTNAME to insure we don't send a hostname greater + than pjproject can handle. Since it can differ between + the version of pjproject that Asterisk was compiled against + and the version of pjproject that Asterisk is running against, + we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk + source code. + + Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e + +2016-01-19 17:15 +0000 [b5c13c1545] Joshua Colp <jcolp@digium.com> + + * test_threadpool: Wait for each task to complete and fix memory leak. + + This change makes the thread_timeout_thrash unit test wait for + each task to complete. This fixes the problem where the test would + prematurely end when all threads were gone and a new one had to be + started to handle the last task. It also increases the thrasing as + it is now more likely for each task to encounter the above scenario. + + This also fixes a memory leak where the data for each task was not + being freed. + + ASTERISK-25611 #close + + Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6 + +2016-01-18 19:44 +0000 [0ab89182d9] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Increase CLI "core ping taskprocessor" timeout. + + Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029 + +2016-01-18 19:43 +0000 [a2a8ea3330] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Fix some taskprocessor unrefs. + + You have to call ast_taskprocessor_unref() outside of the taskprocessor + implementation code. Taskprocessor use since v12 has become more + transient than just the singleton uses in earlier versions. + + Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb + +2016-01-19 13:44 +0000 [d604a9afc8] Richard Mudgett <rmudgett@digium.com> + + * Fix alembic branches on v13. + + Change-Id: I313449b609ede18ad1e1763a655dd23b9210a8e0 + +2016-01-18 18:45 +0000 [a0c79f3a4f] gtjoseph <george.joseph@fairview5.com> + + * pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject + + Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b + +2016-01-14 09:26 +0000 [018ccf680b] Rusty Newton <rnewton@digium.com> + + * func_channel: Add help text for undocumented CHANNEL function arguments + + Adding help text documentation for: + * hangupsource + * appname + * appdata + * exten + * context + * channame + * uniqueid + * linkedid + + ASTERISK-24097 #close + Reported by: Steven T. Wheeler + Tested by: Rusty Newton + + Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d + +2016-01-16 13:18 +0000 [5644bca9f9] Daniel Journo <dan@keshercommunications.com> + + * Update version number in features.conf.sample + + Update the version number in the comments from Asterisk 12 to Asterisk 12+ + + Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b + +2016-01-15 19:52 +0000 [3f5f30cf82] Corey Farrell <git@cfware.com> + + * main/config: Clean config maps on shutdown. + + ASTERISK-25700 #close + + Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808 + +2016-01-14 14:42 +0000 [660fedecb7] Kevin Harwell <kharwell@digium.com> + + * bridge_basic: don't cache xferfailsound during an attended transfer + + The xferfailsound was read from the channel at the beginning of the transfer, + and that value is "cached" for the duration of the transfer. Therefore, changing + the xferfailsound on the channel using the FEATURE() dialplan function does + nothing once the transfer is under way. + + This makes it so the transfer code instead gets the xferfailsound configuration + options from the channel when it is actually going to be used. + + This patch also fixes a potential memory leak of the props object as well as + making sure the condition variable gets initialized before being destroyed. + + ASTERISK-25696 #close + + Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4 + +2015-07-10 10:37 +0000 [9cda1de34d] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Simplify ast_taskprocessor_get() return code. + + Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1 + +2016-01-13 18:20 +0000 [a79af2b312] Richard Mudgett <rmudgett@digium.com> + + * astmm.c: Add more stats to CLI "memory show" commands. + + * Add freed regions totals to allocations and summary. + + * Add totals for all allocations and not just the selected allocations. + + Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a + +2016-01-14 16:00 +0000 [83feb7db3b] Kevin Harwell <kharwell@digium.com> + + * bridge_basic: don't play an attended transfer fail sound after target hangs up + + If the attended transfer destination answers (picks call up or goes to + voicemail) and then hangs up on the transferer then transferer hears the + fail sound. + + This patch makes it so the fail sound is not played when the transfer + destination/target hangs up after answering. + + ASTERISK-25697 #close + + Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded + +2016-01-14 13:22 +0000 [935d641f3b] Mark Michelson <mmichelson@digium.com> + + * Remove res/ari/* content during 'make clean'. + + 'make clean' and 'make distclean' can leave behind .o files in the + res/ari/ directory. One observed consequence of this is that running + Asterisk with MALLOC_DEBUG can cause Asterisk to crash immediately on + startup sometimes. + + By ensuring that we are making a clean build, we can be sure that stale + files are not being included in the build and causing problems when + build options should have caused files to be re-built. + + ASTERISK-25683 #close + Reported by yaron nahum + + Change-Id: I1f48baa904d2468eddeefb42ee68a56af7adc7b7 + +2016-01-13 15:58 +0000 [46f21df302] Daniel Journo <dan@keshercommunications.com> + + * pjsip/alembic: Fix qualify_timeout column definition + + Corrects the qualify_timeout column type from Integer to Decimal + + ASTERISK-25686 #close + Reported-by: Marcelo Terres + + Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8 + +2016-01-12 11:14 +0000 [32b29d7b02] Joshua Colp <jcolp@digium.com> + + * app: Queue hangup if channel is hung up during sub or macro execution. + + This issue was exposed when executing a connected line subroutine. + When connected or redirected subroutines or macros are executed it is + expected that the underlying applications and logic invoked are fast + and do not consume frames. In practice this constraint is not enforced + and if not adhered to will cause channels to continue when they shouldn't. + This is because each caller of the connected or redirected logic does not + check whether the channel has been hung up on return. As a result the + the hung up channel continues. + + This change makes it so when the API to execute a subroutine or + macro is invoked the channel is checked to determine if it has hung up. + If it has then a hangup is queued again so the caller will see it + and stop. + + ASTERISK-25690 #close + + Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea + +2016-01-13 07:20 +0000 [e7cfda0b38] Sean Bright <sean.bright@gmail.com> + + * res_musiconhold: Prevent multiple simultaneous reloads. + + There are two ways in which the reload() function in res_musiconhold can be + called from the CLI: + + * module reload res_musiconhold.so + * moh reload + + In the former case, the module loader holds a lock that prevents multiple + concurrent calls, but in the latter there is no such protection. + + This patch changes the 'moh reload' CLI command to invoke the module loader + directly, rather than call reload() explicitly. + + ASTERISK-25687 #close + + Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c +2016-01-12 14:25 +0000 [5586abc957] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts". + + PJPROJECT has a function available to dump the compile time + options used when building the library. + + * Add CLI "pjsip show buildopts" command. + + * Update contrib/scripts/autosupport to get pjproject information. + + Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748 + +2016-01-12 10:36 +0000 [4cd58c3b20] Mark Michelson <mmichelson@digium.com> + + * res_sorcery_realtime: Remove leading ^ requirement. + + res_sorcery_realtime's search-by-regex callback performed a check to + ensure that the passed-in regex began with a caret (^). If it did not, + then no results would be returned. + + This callback only started to become used when "like" support was added + to PJSIP CLI commands. The CLI command for listing objects would pass an + empty regex ("") to the sorcery backend if no "like" statement was + present. For most sorcery backends, this resulted in returning all + objects. However, for realtime, this resulted in returning no objects. + + This commit seeks to fix the regression by removing the requirement from + res_sorcery_realtime for the passed-in-regex to begin with a caret. + + ASTERISK-25689 #close + Reported by Marcelo Terres + + Change-Id: I22b4dc5d7f3f11bb29ac2e42ef94682e9bab3b20 + +2016-01-07 11:57 +0000 [219c204a41] gtjoseph <george.joseph@fairview5.com> + + * pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address + + On a system with multiple ip addresses in the same subnet, if a + transport is bound to a specific ip address and endpoint/media_address + is set, the SIP/SDP will have the correct address in all fields but + the rtp stream MAY still originate from one of the other ip addresses, + most probably the "primary" ip address. This happens because + res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with + the "all" ip address (0.0.0.0 or ::). + + The new option causes res_pjsip_sdp_rtp/create_rtp to call + ast_rtp_instance_new with the endpoint's media_address (if specified) + instead of the "all" address. This causes the packets to originate from + the specified address. + + ASTERISK-25632 + ASTERISK-25637 + Reported-by: Olivier Krief + Reported-by: Dan Journo + + Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88 + +2016-01-10 16:22 +0000 [22801a06ee] Daniel Journo <dan@keshercommunications.com> + + * pjsip: Add option global/regcontext + + Added new global option (regcontext) to pjsip. When set, Asterisk will + dynamically create and destroy a NoOp priority 1 extension + for a given endpoint who registers or unregisters with us. + + ASTERISK-25670 #close + Reported-by: Daniel Journo + + Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62 + +2016-01-08 15:22 +0000 [1600ebca7d] Kevin Harwell <kharwell@digium.com> + + * pbx: Deadlock between contexts container and context_merge locks + + Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5) + introduced the possibility of a deadlock. Due to the mentioned modifications + ast_change_hints now needs to keep both merge/delete and state callbacks from + occurring while it executes. Unfortunately, sometimes ast_change_hints can be + called with the contexts container locked. When this happens it's possible for + another thread to grab the context_merge_lock before the thread calling into + ast_change_hints does and then try to obtain the contexts container lock. This + of course causes a deadlock between the two threads. The thread calling into + ast_change_hints waits for the other thread to release context_merge_lock and + the other thread is waiting on that one to release the contexts container lock. + + Unfortunately, there is not a great way to fix this problem. When hints change, + the subsequent state callbacks cannot run at the same time as a merge/delete, + nor when the usual state callbacks do. This patch alleviates the problem by + having those particular callbacks (the ones run after a hint change) occur in a + serialized task. By moving the context_merge_lock to a task it can now safely be + attempted or held without a deadlock occurring. + + ASTERISK-25640 #close + Reported by: Krzysztof Trempala + + Change-Id: If2210ea241afd1585dc2594c16faff84579bf302 + +2016-01-10 17:08 +0000 [0fc3dad965] Corey Farrell <git@cfware.com> + + * devicestate: Cleanup engine thread during graceful shutdown. + + ASTERISK-25681 #close + + Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1 + +2016-01-10 13:51 +0000 [f34dd10495] Corey Farrell <git@cfware.com> + + * manager: Cleanup manager_channelvars during shutdown. + + ASTERISK-25680 #close + + Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446 + +2016-01-10 13:27 +0000 [1d3a1167fc] Corey Farrell <git@cfware.com> + + * res_calendar: Cleanup scheduler context at unload. + + ASTERISK-25679 #close + + Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f + +2016-01-08 11:49 +0000 [3a160cdbf6] Joshua Colp <jcolp@digium.com> + + * res_rtp_asterisk: Revert DTLS negotiation changes. + + Due to locking issues within pjnath these changes are being + reverted until pjnath can be changed. + + ASTERISK-25645 + + Revert "res_rtp_asterisk.c: Fix DTLS negotiation delays." + + This reverts commit 24ae124e4f7310cfa64c187b944b2ffc060da28d. + + Change-Id: I2986cfb2c43dc14455c1bcaf92c3804f9da49705 + + Revert "res_rtp_asterisk: Resolve further timing issues with DTLS negotiation" + + This reverts commit 965a0eee46d24321f74c244e23c5a5f45e67e12b. + + Change-Id: Ie68fafde27dad4b03cb7a1e27ce2a8502c3f7bbe + +2016-01-09 17:57 +0000 [4b10fc9173] gtjoseph <george.joseph@fairview5.com> + + * Revert "pjsip_location: Delete contact_status object when contact is deleted" + + This reverts commit 0a9941de9d24093b5ff44096d1d7406f29d11e45. + + Matt, + + This patch causes another problem and should not have been needed. + Before this patch, persistent_endpoint_contact_deleted_observer WAS + deleting the contact_status when ast_sip_location_delete_contact was + called. By deleting it yourself in ast_sip_location_delete_contact + it was gone before the observer could run and the observer therefore + was throwing an error and not sending stasis/AMI/statsd messages. + + So, I don't think this was the cause of your original issue. I also + had verified the contact AMI and statsd lifecycle and it was working. + I'll double check now though. + + ASTERISK-25675 + Reported-by: Daniel Journo + + Change-Id: Ib586a6b7f90acb641b0c410f659743ab90e84f1a + +2016-01-09 18:04 +0000 [79b4309881] Corey Farrell <git@cfware.com> + + * pbx_dundi: Run cleanup on failed load. + + During failed startup of pbx_dundi no cleanup was performed. Add a call + to unload_module before returning AST_MODULE_LOAD_DECLINE. + + ASTERISK-25677 #close + + Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29 + +2016-01-09 13:28 +0000 [a5406b1f9e] Corey Farrell <git@cfware.com> + + * res_crypto: Perform cleanup at shutdown. + + This change causes res_crypto to unregister CLI at shutdown while still + preventing the module from being unloaded. + + ASTERISK-25673 #close + + Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc + +2016-01-06 19:10 +0000 [cf8e7a580b] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Create human friendly serializer names. + + PJSIP name formats: + pjsip/aor/<aor>-<seq> -- registrar thread pool serializer + pjsip/default-<seq> -- default thread pool serializer + pjsip/messaging -- messaging thread pool serializer + pjsip/outreg/<registration>-<seq> -- outbound registration thread pool + serializer + pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer + pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer + pjsip/session/<endpoint>-<seq> -- session thread pool serializer + pjsip/websocket-<seq> -- websocket thread pool serializer + + Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084 + +2016-01-06 19:09 +0000 [4276f185f0] Richard Mudgett <rmudgett@digium.com> + + * Sorcery: Create human friendly serializer names. + + Sorcery name formats: + sorcery/<type>-<seq> -- Sorcery thread pool serializer + + Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47 + +2016-01-06 19:09 +0000 [f02ac1b7f9] Richard Mudgett <rmudgett@digium.com> + + * Stasis: Create human friendly taskprocessor/serializer names. + + Stasis name formats: + subm:<topic>-<seq> -- Stasis subscription mailbox task processor + subp:<topic>-<seq> -- Stasis subscription thread pool serializer + + Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd + +2016-01-07 16:15 +0000 [ec1f1c6742] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: New API for human friendly taskprocessor names. + + * Add new API call to get a sequence number for use in human friendly + taskprocessor names. + + * Add new API call to create a taskprocessor name in a given buffer and + append a sequence number. + + Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9 + +2016-01-06 17:19 +0000 [d8bc3e0c8b] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Fix CLI "core show taskprocessors" output format. + + Update the CLI "core show taskprocessors" output format to not be + distorted because UUID names are longer than previously used taskprocessor + names. + + Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601 + +2016-01-07 21:07 +0000 [2c4b7502de] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Fix CLI "core show taskprocessors" unref. + + Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5 + +2016-01-07 20:44 +0000 [3b33ac7a46] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Sort CLI "core show taskprocessors" output. + + Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e + +2016-01-06 19:00 +0000 [0fc32c4dd3] Richard Mudgett <rmudgett@digium.com> + + * ccss.c: Replace space in taskprocessor name. + + The CLI "core ping taskprocessor" command does not work very + well with taskprocessor names that have spaces in them. You + have to put quotes around the name so using tab completion + becomes awkward. + + Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0 + +2016-01-05 16:54 +0000 [0e0c24ad78] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock. + + Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b + +2016-01-07 03:33 +0000 [0f79c8839b] Diederik de Groot <ddegroot@talon.nl> + + * main: Use ast_strdup instead of strdup + + Fix compile error in main/utils.c because strdup was used in dummy_start + + Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793 + +2016-01-07 03:21 +0000 [4285dee778] Diederik de Groot <ddegroot@talon.nl> + + * include/asterisk/time.h: Renamed global declaration:tv + + Renamed global declaration:tv to dummy_tv_var_for_types, + which would oltherwise cause 'shadow' warnings when 'tv' + was declared as a local variable elsewhere. + + Added comment to note that dummy_tv_var_for_types is never + really exported and only used as a place holder. + + ASTERISK-25627 #close + + Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28 + +2016-01-07 15:37 +0000 [96094feab6] Mark Michelson <mmichelson@digium.com> + + * PJSIP: Prevent deadlock due to dialog/transaction lock inversion. + + A deadlock was observed where the monitor thread was stuck, therefore + resulting in no incoming SIP traffic being processed. + + The problem occurred when two 200 OK responses arrived in response to a + terminating NOTIFY request sent from Asterisk. The first 200 OK was + dispatched to a threadpool worker, who locked the corresponding + transaction. The second 200 OK arrived, resulting in the monitor thread + locking the dialog. At this point, the two threads are at odds, because + the monitor thread attempts to lock the transaction, and the threadpool + thread loops attempting to try to lock the dialog. + + In this case, the fix is to not have the monitor thread attempt to hold + both the dialog and transaction locks at the same time. Instead, we + release the dialog lock before attempting to lock the transaction. + + There have also been some debug messages added to the process in an + attempt to make it more clear what is going on in the process. + + ASTERISK-25668 #close + Reported by Mark Michelson + + Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a + +2016-01-07 09:39 +0000 [52e9de0016] Corey Farrell <git@cfware.com> + + * ast_format_cap_append_by_type: Resolve codec reference leak. + + This resolves a reference leak caused by ASTERISK-25535. The pointer + returned by ast_format_get_codec is saved so it can be released. + + ASTERISK-25664 #close + + Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec + +2016-01-04 04:26 +0000 [86eae38d7e] Aaron An <anjb@ti-net.com.cn> + + * cel/cel_radius: Fix wrong pointer. + + The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter + y not the address of y. + + I capture the radius UDP packet via tcpdump, and the AV pairs are not correct, + then i review the source code and compare it with cdr/cdr_radius.c. Fix it and + it works. + + ASTERISK-25647 #close + Reported by: Aaron An + Tested by: Aaron An + + Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0 + +2016-01-05 14:52 +0000 [881dc862e0] gtjoseph <george.joseph@fairview5.com> + + * asterisk.h: Add ASTERISK_REGISTER_FILE macro + + The 11/13 branches and master use 2 different file version macros. 11/13 + uses ASTERISK_FILE_VERSION but master uses ASTERISK_REGISTER_FILE. This + means a new file added to 11/13 can't just be cherry-picked to master + because the macro has to be changed. + + To make cherry-picking possible, ASTERISK_REGISTER_FILE was added + to asterisk.h as a simple alias for ASTERISK_FILE_VERSION(__FILE__, NULL) + The "$Revision$" tag doesn't do anything since Asterisk moved to git so + just passing NULL as the verison works fine. asterisk.h was also + annotated to deprecate ASTERISK_FILE_VERSION and suggest using + ASTERISK_REGISTER_FILE for all new files. + + Finally, 2 recent file additions, pbx_builtins.c and pbx_functions.c, + were modified to use the new macro to make sure it actually worked. + 'core show file version' showed the correct output. + + Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5 + +2016-01-05 11:06 +0000 [d228b62fd4] gtjoseph <george.joseph@fairview5.com> + + * stasis_cache_pattern: Backport to 13 + + Somehow stasis_cache_pattern got out of sync between 13 and master + and it was causing duplicate channel message issues in 13 when + related to a specific endpoint. I.E. from statsd, + 'endpoints.PJSIP.1174.channels 0|g' was being emitted twice. + + Backporting stasis_cache_pattern from master to 13 solved + the issue and running the unit and testsuite tests confirmed + that no new ones were created. + + ASTERISK-25317 #close + + Change-Id: Ia8707462f62d15eed14541c37f332a7bbbceb548 +2016-01-04 20:23 +0000 [e462f0063f] Corey Farrell <git@cfware.com> + + * main/pbx: Move hangup handler routines to pbx_hangup_handler.c. + + This is the sixth patch in a series meant to reduce the bulk of pbx.c. + This moves hangup handler management functions to their own source. + + Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104 + +2016-01-04 19:46 +0000 [ab191d124c] Corey Farrell <git@cfware.com> + + * main/pbx: Move dialplan application management routines to pbx_app.c. + + This is the sixth patch in a series meant to reduce the bulk of pbx.c. + This moves dialplan application management functions to their own source. + + Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c + +2016-01-04 18:20 +0000 [09a9b93896] Corey Farrell <git@cfware.com> + + * main/pbx: Move switch routines to pbx_switch.c. + + This is the fifth patch in a series meant to reduce the bulk of pbx.c. + This moves ast_switch functions to their own source. + + Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e + +2016-01-04 18:00 +0000 [c608274a39] Corey Farrell <git@cfware.com> + + * main/pbx: Move timing routines to pbx_timing.c. + + This is the fourth patch in a series meant to reduce the bulk of pbx.c. + This moves pbx timing functions to their own source. + + Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6 + +2015-12-29 04:31 +0000 [338a8ffed6] Martin Tomec <tomec.martin@gmail.com> + + * app_queue: Add member flag "in_call" to prevent reading wrong lastcall time + + Member lastcall time is updated later than member status. There was chance to + check wrapuptime for available member with wrong (old) lastcall time. + New boolean flag "in_call" is set to true right before connecting call, and + reset to false after update of lastcall time. Members with "in_call" set to true + are treat as unavailable. + + ASTERISK-19820 #close + + Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500 + +2015-12-28 17:23 +0000 [e13719bff1] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com> + + * app_queue: Added reason pause of member + + In app_queue added value Paused Reason on QueueMemberStatus when a member + on queue is paused and the reason was set. + + ASTERISK-25480 #close + Reporte by: Rodrigo Ramírez Norambuena + + Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e + +2015-12-30 10:49 +0000 [4ec85a9f07] gtjoseph <george.joseph@fairview5.com> + + * voicemail: Move app_voicemail / res_mwi_external conflict to runtime + + The menuselect conflict between app_voicemail and res_mwi_external + makes it hard to package 1 version of Asterisk. There no actual + build dependencies between the 2 so moving this check to runtime + seems like a better solution. + + The ast_vm_register and ast_vm_greeter_register functions in app.c + were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there + is already a voicemail module registered. The modules' load_module + functions were then modified to return DECLINE instead of -1 to the + loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, + the modules were incorrectly causing Asterisk to stop so this needed + to be cleaned up anyway. + + Now you can build both and use modules.conf to decide which voicemail + implementation to load. + + The default menuselect options still build app_voicemail and not + res_mwi_external but if both ARE built, res_mwi_external will load + first and become the voicemail provider unless modules.conf rules + prevent it. This is noted in CHANGES. + + Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247 + +2016-01-04 16:22 +0000 [7fdcfd7724] Corey Farrell <git@cfware.com> + + * main/pbx: Move variable routines to pbx_variables.c. + + This is the third patch in a series meant to reduce the bulk of pbx.c. + This moves channel and global variable routines to their own source. + + Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6 + +2015-12-04 17:22 +0000 [80a8b2a4cd] Richard Mudgett <rmudgett@digium.com> + + * app_dial: Immediately exit dial if the caller is already hung up. + + If a caller hangs up before dial is executed within an AGI then the AGI + has likely eaten all queued frames before executing the dial in DeadAGI + mode. With the caller hung up and no pending frames from the caller's + read queue, dial would not know that the call has hung up until a called + channel answers. It is rather annoying to whoever just answered the + non-existent call. + + Dial should not continue execution in DeadAGI mode, hangup handlers, or + the h exten. + + * Added a check early in dial to abort dialing if the caller has hungup. + + ASTERISK-25307 #close + Reported by: David Cunningham + + Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418 + +2016-01-02 10:26 +0000 [1087b0c6ed] Matt Jordan <mjordan@digium.com> + + * main/cdr: Allow setting properties on a finalized CDR if it is the last one + + Prior to this patch, we explicitly disallowed setting any properties on a + finalized CDR. This seemed like a good idea at the time; in practice, it was + more restrictive. + + There are weird and strange scenarios where setting a property on a finalized + CDR is definitely wrong. For example, we may Fork a CDR, finalizing the + previous one, then change a property. In said case, the old CDR is supposed + to now be 'immutable' (so to speak), and should not be updated. From the + perspective of the code, a forked CDR that is finalized is just finalized. + Hence why we decided these should not be updated. + + In practice, it is much more common to want to set a property on a CDR in + the h extension or in a hangup handler. Disallowing a common scenario to make + an esoteric behaviour work isn't good. This patch fixes this by allowing + callers to set a property IF we are the last CDR in the chain. This preserves + the finalized CDR if it was forked, while allowing the more common case to + function. + + ASTERISK-25458 #close + + Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9 + +2016-01-02 10:23 +0000 [1f23e65b89] Matt Jordan <mjordan@digium.com> + + * main/cdr: Set the end time on a CDR if endbeforehexten is Yes + + Prior to this patch, the CDR engine attempted to set the end time on a CDR + that was executing hangup logic and with endbeforehexten set to Yes by + calling a function that inspects the properties on the Party A snapshot to + determine if we are ready to set the end time. That always failed. This is + because a Party A snapshot is not updated for CDRs that are executing hangup + logic with endbeforehexten=Yes. + + Instead of calling a function that looks at the Party A snapshot, we just + simply set the end time on the CDR. This is safe to call multiple times, and is + safe to call at this point as we know that (a) we are executing hangup logic, + and (b) we are supposed to set the end time at this point. + + ASTERISK-25458 + + Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3 + +2015-12-30 20:51 +0000 [2ffade4574] Corey Farrell <git@cfware.com> + + * main/pbx: Move custom function routines to pbx_functions.c. + + This is the second patch in a series meant to reduce the bulk of pbx.c. + This moves custom function management routines to their own source. + + Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177 + +2015-12-28 19:18 +0000 [20b8474f20] gtjoseph <george.joseph@fairview5.com> + + * main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c + + We joked about splitting pbx.c into multiple files but this first step was + fairly easy. All of the pbx_builtin dialplan applications have been moved + into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() + is called by asterisk.c just after load_pbx(). + + A few functions were renamed and are cross-exposed between the 2 source files. + + Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a + +2015-12-24 20:26 +0000 [e4a566918a] Matt Jordan <mjordan@digium.com> + + * tests/test_stasis_endpoints: Remove expected duplicate events + + The cache_clear test was written to expect duplicate Stasis messages + sent from the technology endpoint to the all caching topic. This patch + fixes the test to no longer expect these duplicate messages. + + ASTERISK-25137 + + Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981 + +2015-12-28 14:02 +0000 [a280400758] Joshua Colp <jcolp@digium.com> + + * test_time: Provide a timeout when waiting. + + The test_timezone_watch unit test is written to expect a + condition to be signaled when the inotify daemon thread runs. + There exists a small window where the test_timezone_watch + thread can signal the inotify daemon thread while it is not + reading on the underlying file descriptor. If this occurs + the test_timezone_watch thread will wait indefinitely for a + signal that will never arrive. + + This change adds a timeout to the condition so it will return + regardless after a period of time. + + Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390 + +2015-05-27 13:22 +0000 [3a1c4885be] gtjoseph <george.joseph@fairview5.com> + + * endpoint/stasis: Eliminate duplicate events on endpoint status change + + When an endpoint is created, its messages are forwarded to both the tech + endpoint topic and the all endpoints topic. This is done so that various + parties interested in endpoint messages can subscribe to just the tech + endpoint and receive all messages associated with that particular technology, + as opposed to subscribing to the all endpoints topic. Unfortunately, when the + tech endpoint is created, it also forwards all of its messages to the all + topic. This results in duplicate messages whenever an endpoint publishes its + messages. + + This patch resolves the duplicate message issue by creating a new function + for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts + as a normal caching topic, save that it no longer forwards messages it receives + to the all endpoints topic. This allows it to act as an aggregation "sink", + while preserving the necessary caching behaviour. + + ASTERISK-25137 #close + Reported-by: Vitezslav Novy + + ASTERISK-25116 #close + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> + + Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b +2015-12-24 22:19 +0000 [136c537695] Dade Brandon <dade@xencall.com> + + * res_http_websocket.c: prevent avoidable disconnections caused by write errors + + Updated ast_websocket_write to encode the entire frame in to one + write operation, to ensure that we don't end up with a situation + where the websocket header has been sent, while the body can not + be written. + + Previous to August's patch in commit b9bd3c14, certain network + conditions could cause the header to be written, and then the + sub-sequent body to fail - which would cause the next successful + write to contain a new header, and a new body (resulting in + the peer receiving two headers - the second of which would be + read as part of the body for the first header). + + This was patched to have both write operations individually fail + by closing the websocket. + + In a case available to the submitter of this patch, the same + body which would consistently fail to write, would succeed + if written at the same time as the header. + + This update merges the two operations in to one, adds debug messages + indicating the reason for a websocket connection being closed during + a write operation, and clarifies some variable names for code legibility. + + Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598 + +2015-12-27 22:38 +0000 [f2efbb5d75] Corey Farrell <git@cfware.com> + + * Remove res_jabber file that was left behind. + + Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b + +2015-12-13 13:09 +0000 [dde7f3c1c4] Matt Jordan <mjordan@digium.com> + + * res_pjsip_history: Add a module that provides PJSIP history for debugging + + This patch adds a new module, res_pjsip_history, that provides a slightly + better way of debugging SIP message traffic on a busy Asterisk system. The + existing mechanisms all rely on passively dumping a SIP message to the CLI. + While this is perfectly fine for logging purposes and well controlled + environments, on many installations, the amount of SIP messages Asterisk + receives will quickly swamp the CLI. This makes it difficult to view/capture + those messages that you want to diagnose in real time. + + This patch provides another way of handling this. When enabled, the module + will store SIP message traffic in memory. This traffic can then be queried + at leisure. + + In order to make the querying useful, a CLI command has been implemented, + 'pjsip show history', that supports a basic expression syntax similar to + SQL or other query languages. A small number of useful fields have been + added in this initial patch; additional fields can easily be added in + later improvements. Those fields are: + - number: The entry index in the history + - timestamp: The time the message was recieved + - addr: The source/destination address of the message + - sip.msg.request.method: The request method + - sip.msg.call-id: The Call-ID header + + Note - this is a resurrection of the module initially proposed on Review Board + here: https://reviewboard.asterisk.org/r/4053/ + + Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36 + +2015-12-25 09:56 +0000 [be050f2638] Dade Brandon <dade@xencall.com> + + * chan_sip.c: fix websocket_write_timeout default value + + websocket_write_timeout was not being set to its default value + during sip config reload, which meant that prior to this commit, + 1) the default value of 100 was not used, unless an invalid value + (or 1) was specified in sip.conf for websocket_write_timeout, and + 2) if the websocket_write_timeout directive was removed from sip.conf + without a full restart of asterisk, then the previous value would + continue to be used indefinitely. + + This essentially lead to a 0ms write timeout (the first write attempt + in ast_careful_fwrite must have succeeded) in websocket write requests + from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf. + + Changes to websocket_write_timeout still only apply to new websocket + sessions, after the sip reload -- timeouts on existing sessions are + not adjusted during sip reload. + + Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953 + +2015-12-23 17:40 +0000 [b3024cad10] Richard Mudgett <rmudgett@digium.com> + + * bridge_basic.c: Fix GOTO_ON_BLINDXFR + + Use of GOTO_ON_BLINDXFR would not work at all. The target location would + never be executed by the transferring channel. + + * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with + valid context, exten, and priority parameters from the transferring + channel. + + * Renamed some feature_blind_transfer() local variables for clarity. + + ASTERISK-25641 #close + Reported by Dmitry Melekhov + + Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac + +2015-12-24 12:19 +0000 [0a9941de9d] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_location: Delete contact_status object when contact is deleted + + In 450579e908, a change was made that removed the deletion of the + 'contact_status' object when a 'contact' object is deleted in sorcery. + This unfortunately means that the 'contact_status' object persists, even when + something has explicitly removed a contact. The result is that the state of + the contact will not be regenerated if that contact is re-created, and the + stale state will be reported/used for that contact. It also results in + no ContactStatusChanged events being generated for either ARI or AMI. + + This patch restores the deletion logic that was removed. Doing so now + results in the expected events being generated again. + + Change-Id: I28789a112e845072308b5b34522690e3faf58f07 + +2015-12-24 10:18 +0000 [1e24a0ca8a] Kevin Harwell <kharwell@digium.com> + + * res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef + + Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151 + +2015-12-20 21:33 +0000 [1d3d20dd68] Dade Brandon <dade@xencall.com> + + * app_amd: Correct documentation to reflect functionality + + Update documentation to reflect that maximum_number_of_words + has functionality inconsistent with the variable name (and inconsistent + with prior documentation.) + + Update documentation for silence_threshold, which previously implied + that it was measuring time, rather than noise averages in the sample. + + Update the comments in amd.conf.sample. + + ASTERISK-25639 #close + Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093 + +2015-12-17 19:05 +0000 [965a0eee46] Dade Brandon <dade@xencall.com> + + * res_rtp_asterisk: Resolve further timing issues with DTLS negotiation + + Resolves an edge case dtls negotiation delay for certain networks which + somehow manage to drop the rtcp side's packet when these are both sent + ast_rtp_remote_address_set, causing it to have to time-out and restart + the handshake. + + Move dtls pending bio flush in to it's own function, and call it from + ast_rtp_on_ice_complete, when we're rtp->ice, rather than when + ast_rtp_remote_address_set. + + Keep the existing flush from the recent change to res_rtp_remote_address_set + if ice is not being used. + + ASTERISK-25614 #close + Reported-by: XenCALL + Tested by: XenCALL + + Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5 + +2015-12-18 09:54 +0000 [ae428d8460] Carlos Oliva <carlos.oliva@invoxcontact.com> + + * app_queue: update RT members when the 1st call joins a queue with no agents + + If a call enters on a queue and the members on that queue are updated in + realtime (ex: using mysql inserting a new agent) the queue members are + never refreshed and the call will stay in the queue until other event occurs. + This happens only if this is the first call of the queue and there is no + agents servicing. + This patch prevent this issue, ensuring realtime members are updated if + there is one call in the queue and no available agents + + ASTERISK-25442 #close + + Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682 + +2015-12-05 10:01 +0000 [59d5bb0613] Joshua Colp <jcolp@digium.com> + + * res_sorcery_memory_cache: Add support for a full backend cache. + + This change introduces the configuration option 'full_backend_cache' + which changes the cache to be a full mirror of the backend instead + of a per-object cache. This allows all sorcery retrieval operations + to be carried out against it and is useful for object types which + are used in a "retrieve all" or "retrieve some" pattern. + + ASTERISK-25625 #close + + Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5 + +2015-12-17 10:25 +0000 [0cefcabd58] Joshua Colp <jcolp@digium.com> + + * rtp_engine: Ignore empty filenames in DTLS configuration. + + When applying an empty DTLS configuration the filenames in the + configuration will be empty. This is actually valid to do and + each filename should simply be ignored. + + Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539 + +2015-12-17 08:10 +0000 [158a0a5422] Joshua Colp <jcolp@digium.com> + + * chan_sip: Enable WebSocket support by default. + + Per the documentation the WebSocket support in chan_sip is + supposed to be enabled by default but is not. This change + corrects that. + + Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423 + +2015-12-14 12:04 +0000 [a9d6fc571d] Joshua Colp <jcolp@digium.com> + + * json: Audit ast_json_* usage for thread safety. + + The JSON library Asterisk uses, jansson, is not thread + safe for us in a few ways. To help with this wrappers for JSON + object reference count increasing and decreasing were added + which use a global lock to ensure they don't clobber over + each other. This does not extend to reference count manipulation + within the jansson library itself. This means you can't safely + use the object borrowing specifier (O) in ast_json_pack and + you can't share JSON instances between objects. + + This change removes uses of the O specifier and replaces them + with the o specifier and an explicit ast_json_ref. Some cases + of instance sharing have also been removed. + + ASTERISK-25601 #close + + Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1 + +2015-12-16 11:28 +0000 [53bd5a539a] Mark Michelson <mmichelson@digium.com> + + * Alembic: Increase column size of PJSIP AOR "contact". + + When running the PJSIP AMI "show_endpoint" test with automatic + conversion to realtime, the test would fail. This was because the AOR + "contact" column was sized at 40, and the configured contact was larger + than that. + + This commit increases the size of the contact column to 255 characters. + + Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1 + +2015-12-16 11:25 +0000 [da17dc4d75] Mark Michelson <mmichelson@digium.com> + + * Alembic: Add PJSIP global keep_alive_interval. + + The keep_alive_interval option was added about a year ago, but no + alembic revision was created to add the appropriate column to the + database. + + This commit fixes the problem and adds the column. This was discovered + by running the testsuite with automatic conversion to realtime enabled. + + Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a + +2015-12-08 13:04 +0000 [fe8011cc50] sungtae kim <pchero21@gmail.com> + + * AMI: Fixed OriginateResponse message + + When the asterisk sending OriginateResponse message, + it doesn't set the "Uniqueid". + And it didn't support correct response message for + Application originate. + + ASTERISK-25624 #close + + Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d + +2015-12-15 18:01 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.7.0-rc1 Released. + +2015-12-15 11:57 +0000 [0370acecfc] Kevin Harwell <kharwell@lunkwill> + + * Release summaries: Add summaries for 13.7.0-rc1 + +2015-12-15 11:54 +0000 [d1bb33fe0b] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.7.0-rc1 + +2015-12-15 11:54 +0000 [d06a65de01] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.7.0-rc1 + +2015-12-15 11:54 +0000 [fb37b44660] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.7.0-rc1 + +2015-12-15 11:48 +0000 [20b7164b8c] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.7.0-rc1 + +2015-12-15 11:48 +0000 [6cbf2414c3] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.7.0-rc1 + +2015-12-15 11:48 +0000 [ba1794464d] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.7.0-rc1 + +2015-12-15 11:39 +0000 [b3e9753a23] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.7.0-rc1 + +2015-12-15 11:39 +0000 [b0df64b5f0] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.7.0-rc1 + +2015-12-15 11:39 +0000 [ce9a59faf6] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.7.0-rc1 + +2015-12-15 11:28 +0000 [2e26bef5bb] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.7.0-rc1 + +2015-12-15 11:28 +0000 [5e9b47516d] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.7.0-rc1 + +2015-12-15 11:28 +0000 [034112c574] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.7.0-rc1 + +2015-12-15 11:19 +0000 [d1f8ff1789] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.7.0-rc1 + +2015-12-15 11:19 +0000 [9376488bef] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.7.0-rc1 + +2015-12-15 11:19 +0000 [a894c9e7a9] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.7.0-rc1 + +2015-12-15 11:12 +0000 [52afb0f112] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.7.0-rc1 + +2015-12-15 11:12 +0000 [2de343eb85] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.7.0-rc1 + +2015-12-15 11:12 +0000 [184de2a160] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.7.0-rc1 + +2015-12-14 13:53 +0000 [24ae124e4f] server-pandora <server-pandora@xencall.com> + + * res_rtp_asterisk.c: Fix DTLS negotiation delays. + + - Trigger pending DTLS packets to send out, once the RTP instance's remote + address is set. + - Avoids locking the DTLS structure unnecessarily by only doing this if + DTLS is passive. + - Add DTLS locks around the structurally sensitive calls in the SSL + portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock + inside of itself, and we're dealing with the SSL BIO in at least two + threads. + + WebRTC channels may receive a DTLS handshake before + ast_rtp_remote_address_set is called, which causes there to be a pending + response to send out. Previous to 1ad827, this was handled by calling + dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP + packet could trigger the pending handshake response. Since that was + rightfully removed, whenever the DTLS handshake is received before the + remote address is set, we would have to wait until another SSL packet + arrives. + + As of Chrome M47's optimizations to their handshake process, WebRTC + conversations between Chrome M47+ and Asterisk, where Asterisk is passive, + experience a 1 second delay without this patch, because the SSL handshake + is received before ICE negotation stores the remote_address, and the next + SSL packet isn't received until after a 1 second timeout in Chrome, which + causes a new handshake request. + + ASTERISK-25614 #close + + Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908 + +2015-12-14 15:25 +0000 [36097a185d] Richard Mudgett <rmudgett@digium.com> + + * Fix sscanf() format string type mismatch. + + ASTERISK-25615 + Reported by: George Joseph + + Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b + +2015-12-13 13:13 +0000 [94f9927784] Matt Jordan <mjordan@digium.com> + + * main/utils: Don't emit an ERROR message if the read end of a pipe closes + + An ERROR or WARNING message should generally indicate that something has gone + wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not + in control of when the far end closes its reading on a file descriptor. If the + far end does close the file descriptor in an unclean fashion, this isn't a bug + or error in Asterisk, particularly when the situation can be gracefully + handled in Asterisk. + + Currently, when this happens, a user would see the following somewhat cryptic + ERROR message: + + "utils.c: write() returned error: Broken pipe" + + There's a few problems with this: + (1) It doesn't provide any context, other than 'something broke a pipe' + (2) As noted, it isn't actually an error in Asterisk + (3) It can get rather spammy if the thing breaking the pipe occurs often, such + as a FastAGI server + (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even + mask legitimate issues + + This patch changes ast_carefulwrite to only log an ERROR if we actually had one + that was reasonably under our control. For debugging purposes, we still emit + a debug message if we detect that the far side has stopped reading. + + Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566 + +2015-12-12 11:08 +0000 [5b867fa904] gtjoseph <george.joseph@fairview5.com> + + * pjsip/config_transport: Check pjproject version at runtime for async ops + + pjproject < 2.5.0 will segfault on a tls transport if async_operations + is greater than 1. A runtime version check has been added to throw + an error if the version is < 2.5.0 and async_operations > 1. + + To assist in the check, a new api "ast_compare_versions" was added + to utils which compares 2 major.minor.patch.extra version strings. + + ASTERISK-25615 #close + + Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-12-10 11:44 +0000 [14b41115e3] Jonathan Rose <jrose@digium.com> + + * chan_sip: Add TCP/TLS keepalive to TCP/TLS server + + Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously + this option was only being set on session sockets. + http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ + According to the link above, the SO_KEEPALIVE option is useful for knowing + when a TCP connected endpoint has severed communication without indicating + it or has become unreachable for some reason. Without this patch, keep + alive is not set on the socket listening for incoming TCP sessions and + in Komatsu's report this resulted in the thread listening for TCP becoming + stuck in a waiting state. + + ASTERISK-25364 #close + Reported by: Hiroaki Komatsu + + Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36 +2015-12-09 09:48 +0000 [cd119ed4a2] Tyler Cambron <tcambron@digium.com> + + * res_chan_stats: Fix bug to send correct statistics to StatsD + + Fixed a bug that originally would show a negative number of + active calls occuring in Asterisk. A gauge is persistent so + incrementing and decrementing it results in a more consistent + performance. Also changed to the call to StatsD to use + ast_statsd_log_string() so that a "+" could be sent to StatsD. + + ASTERISK-25619 #close + + Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7 + +2015-12-07 13:07 +0000 [ddf4dddf4f] Corey Farrell <git@cfware.com> + + * app_meetme: Set default value for audio_buffers. + + The default value was never set for audio_buffers, causing bad + audio quality. This ensures the default is always set. + + ASTERISK-25569 #close + + Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44 +2015-12-08 01:57 +0000 [142d4fefb8] Filip Jenicek <phill@janevim.cz> + + * chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) + + Asterisk may crash when calling ast_channel_get_t38_state(c) + on a locked channel which is being hung up. + + ASTERISK-25609 #close + + Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b + +2015-12-08 17:49 +0000 [21962dad93] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Add existence and readablity checks for tls related files + + Both transport and endpoint now check for the existence and readability + of tls certificate and key files before passing them on to pjproject. + This will cause the object to not load rather than waiting for pjproject + to discover that there's a problem when a session is attempted. + + NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located + in build_peer which is gigantic and I didn't want to disturb it. + Error messages will emit but it won't interrupt chan_sip loading. + + ASTERISK-25618 #close + + Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-12-02 12:42 +0000 [28d9243079] Eugene Voityuk <eugene@thirdlane.com> + + * chan_sip.c: Start ICE negotiation when response is sent or received. + + The current logic for ICE negotiation starts it + when receiving an SDP with ICE candidates. This is + incorrect as ICE negotiation can only start when each + call party have at least one pair of local and remote + candidate. Starting ICE negotiation early would result + in negotiation failure and ultimately no audio. + + This change makes it so ICE negotiation is only started + when a response with SDP is received or when a response + with SDP is sent. + + ASTERISK-24146 + + Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca +2015-12-08 11:03 +0000 [e03582a1c2] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls + + See ASTERISK-25615. + If the transport protocol is tls and async_operations > 1, pjproject + will segfault if more than one operation is attempted on the same socket. + Until this is fixed upstream, a check has been added to throw an error + if a tls transport config has async_operations set to > 1. + + ASTERISK-25615 + + Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-12-08 08:39 +0000 [876600ce6e] Alexander Traud <pabstraud@compuserve.com> + + * codec_resample: Increase buffer for Opus Codec with FEC. + + ASTERISK-25599 #close + + Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e + +2015-12-08 03:46 +0000 [69e3d40ad7] Alexander Traud <pabstraud@compuserve.com> + + * translate: Avoid a warning message when doing FEC within Opus Codec. + + ASTERISK-25616 #close + + Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319 + +2015-12-04 15:36 +0000 [2b992014dc] Richard Mudgett <rmudgett@digium.com> + + * chan_sip: Fix crash involving the bogus peer during sip reload. + + A crash happens sometimes when performing a CLI "sip reload". The bogus + peer gets refreshed while it is in use by a new call which can cause the + crash. + + * Protected the global bogus peer object with an ao2 global object + container. + + ASTERISK-25610 #close + + Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed + +2015-12-06 16:32 +0000 [529535f0c2] Matt Jordan <mjordan@digium.com> + + * Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" + + This reverts commit 6614babea27fbafbe11820ea03737dd5c4f9ecec. + + Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks + in core_unreal/chan_local. Local channels attempt to reach across both their + peer and the peer's bridge to inspect T.38 state. Given the propensity of + Local channel chains, managing the locking situation in such a scenario is + practically infeasible. + + Change-Id: Ic687397ffea08dfb899345a443bd990ec3d0416a + +2015-12-04 16:23 +0000 [450579e908] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip/contacts/statsd: Make contact lifecycle events more consistent + + It will never be perfect or even pretty, mostly because of the differences + between static and dynamic contacts. + + Created: + + Can't use the contact or contact_status alloc functions + because the objects come and go regardless of the actual state. + + Can't use the contact_apply_handler, ast_sip_location_add_contact or + a sorcery created handler because they only get called for dynamic + contacts. Similarly, permanent_uri_handler only gets called for + static contacts. + + So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is + the only place it can go and not have duplicated code. Both + permanent_uri_handler and contact_apply_handler call find_or_create. + + Removed: + + Can't use the destructors for the same reason as above. The only + place to put this is in persistent_endpoint_contact_deleted_observer + which I believe is the "correct" place but even that will handle only + dynamic contacts. This doesn't called on shutdown however. There is + no hook to use for static contacts that may be removed because of a + config change while asterisk is in operation. + + I moved the cleanup of contact_status from ast_sip_location_delete_contact + to the handler as well. + + Status Change and RTT: + + Although they worked fine where they were (in update_contact_status) I + moved them to persistent_endpoint_contact_status_observer to make it + more consistent with removed. There was logic there already to detect + a state change. + + Finally, fixed a nit in permanent_uri_handler rmudgett reported + eralier. + + ASTERISK-25608 #close + + Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d + Reported-by: George Joseph + Tested-by: George Joseph + +2015-11-21 06:02 +0000 [5a18193dc0] Alexander Traud <pabstraud@compuserve.com> + + * res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8. + + ASTERISK-25584 #close + + Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91 + +2015-11-21 05:21 +0000 [3e2178c05e] Alexander Traud <pabstraud@compuserve.com> + + * res_format_attr_opus: Update to latest RFC 7587. + + Beside that, the format-attribute module sends only non-default values in the + line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore, + previously the parameter stereo was not parsed when being the first parameter. + + ASTERISK-25583 #close + + Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73 +2015-12-02 14:11 +0000 [072d94183c] Jonathan Rose <jrose@digium.com> + + * Fix crash in audiohook translate to slin + + This patch fixes a crash which would occur when an audiohook was + applied to a channel using an audio codec that could not be translated + to signed linear (such as when using pass-through codecs like OPUS or + when the codec translator module for the format in use is not loaded). + + ASTERISK-25498 #close + Reported by: Ben Langfeld + + Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384 + +2015-12-03 12:07 +0000 [9184fbeb34] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Use a MD5 hash for static Contact IDs + + When 90d9a70789 was merged, it mostly tested dynamic contacts created as + a result of registering a PJSIP endpoint. Contacts generated in this + fashion typically have a long alphanumeric string as their object identifier, + which maps reasonably well for StatsD. Unfortunately, this doesn't work in the + general case. StatsD treats both '.' and ':' characters as special characters. + In particular, having a ':' appear in the middle of a StatsD metric will + result in the metric being rejected. + + This causes some obvious issues with SIP URIs. + + The StatsD API should not be responsible for escaping the metric name passed + to it. The metric is treated as a single long string, and it would be + challenging to know what to escape in the string passed to the function. + Likewise, we don't want to escape the metric in PJSIP, as that involves + overhead that is wasted when either res_statsd isn't loaded or enabled. + + This patch takes an alternative approach. The Contact ID has been changed + to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the + aforementioned special characters, (b) can be done on Contact creation, + which has minimal impact on run-time performance, and (c) also conforms to an + earlier commit that changed the ID for dynamic contacts. + + The downside of this is that StatsD users will have to map SHA1 hashes back to + the Contacts that are emitting the statistics. To that end, the CLI commands + have been updated to include the first 10 characters of the MD5 hash, which + should be enough to match what is shown in Graphite (or some other StatsD + backend). + + ASTERISK-25595 #close + + Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2 + Reported-by: Matt Jordan + Tested-by: George Joseph + +2015-11-30 22:19 +0000 [ed9134282e] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Update logging to show contact->uri in messages + + An earlier commit changed the id of dynamic contacts to contain + a hash instead of the uri. This patch updates status change + logging to show the aor/uri instead of the id. This required + adding the aor id to contact and contact_status and adding + uri to contact_status. The aor id gets added to contact and + contact_status in their allocators and the uri gets added to + contact_status in pjsip_options when the contact_status is + created or updated. + + ASTERISK-25598 #close + + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511 + +2015-12-01 16:11 +0000 [eadad24b59] Jonathan Rose <jrose@digium.com> + + * Unset BRIDGEPEER when leaving a bridge + + Currently if a channel is transferred out of a bridge, the BRIDGEPEER + variable (also BRIDGEPVTCALLID) remain set even once the channel is + out of the bridge. This patch removes these variables when leaving + the bridge. + + ASTERISK-25600 #close + Reported by: Mark Michelson + + Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da + +2015-11-30 14:22 +0000 [bb0b60619d] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_memory_cache.c: Fix off nominal ref leak. + + Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49 + +2015-11-30 16:42 +0000 [e7c88e11aa] Richard Mudgett <rmudgett@digium.com> + + * sched.c: Make not return a sched id of 0. + + According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0 + was never returned historically and several users incorrectly coded usage + of the returned sched ID assuming that 0 was invalid. + + ASTERISK-25476 + + Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20 + +2015-11-25 12:23 +0000 [4aed349a7b] Richard Mudgett <rmudgett@digium.com> + + * Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions) + + chan_sip.c: + * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to + ao2 conversion. + + * Initialize register scheduler ids earlier because of ASTOBJ to ao2 + conversion. + + chan_skinny.c: + * Fix more scheduler usage for the valid 0 id value. + + ASTERISK-25476 + + Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95 + +2015-11-24 12:44 +0000 [6d9156d10f] Richard Mudgett <rmudgett@digium.com> + + * Audit improper usage of scheduler exposed by 5c713fdf18f. + + channels/chan_iax2.c: + * Initialize struct chan_iax2_pvt scheduler ids earlier because of + iax2_destroy_helper(). + + channels/chan_sip.c: + channels/sip/config_parser.c: + * Fix initialization of scheduler id struct members. Some off nominal + paths had 0 as a scheduler id to be destroyed when it was never started. + + chan_skinny.c: + * Fix some scheduler id comparisons that excluded the valid 0 id. + + channel.c: + * Fix channel initialization of the video stream scheduler id. + + pbx_dundi.c: + * Fix channel initialization of the packet retransmission scheduler id. + + ASTERISK-25476 + + Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8 + +2015-12-01 07:55 +0000 [b76c196e13] Alexander Traud <pabstraud@compuserve.com> + + * codec_resample: Increase buffer for Opus Codec. + + ASTERISK-25599 #close + + Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10 + +2015-11-28 08:46 +0000 [6614babea2] Matt Jordan <mjordan@digium.com> + + * bridges/bridge_t38: Add a bridging module for managing T.38 state + + When 4875e5ac32 was merged, it fixed several issues with a direct media bridge + transitioning to handling a T.38 fax. However, it uncovered a race condition + caused by the bridging core. When a channel involved in a T.38 fax leaves a + bridge, the frame queued by the channel driver that should inform the far side + that it is no longer in a T.38 fax may not make it across the bridge. The + bridging framework is *extremely* aggressive in tearing down the bridge, and + control frames that are currently in flight *may* get dropped. + + This patch adds a new module to the bridging framework, bridge_t38. This module + maintains some notion of the T.38 state for the two channels in a bridge. When + the bridge detects that it is being torn down or when one of the two channels + leaves, it informs the respective channel(s) that they should stop faxing. This + ensures that channels switch back to audio if they survive and are ejected out + of a bridge while faxing. + + ASTERISK-25582 + + Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0 + +2015-11-27 07:39 +0000 [3fcf160fae] Niklas Larsson <niklas@tese.se> + + * CHANGES: Fix a typo + + Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7 +2015-11-25 15:26 +0000 [45efbf8503] Kevin Harwell <kharwell@digium.com> + + * fastagi: record file closed after sending result + + The fastagi record-file testsuite test sometimes fails reporting an empty + recorded file. This was happening because Asterisk was sending the agi result + notification prior to actually closing the file and the data, being buffered, + had not been written to the file yet when the test attempts to check the file + size. + + This patch makes it so the record file stream is closed prior to sending the + agi result notification. + + ASTERISK-25593 #close + + Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde + +2015-11-25 13:29 +0000 [b2787876d6] Walter Doekes <walter+asterisk@wjd.nu> + + * main: Slight refactor of main. Improve color situation. + + Several issues are addressed here: + - main() is large, and half of it is only used if we're not rasterisk; + fixed by spliting up the daemon part into a separate function. + - Call ast_term_init from rasterisk as well. + - Remove duplicate code reading/writing asterisk history file. + - Attempt to tackle background color issues and color changes that + occur. Tested by starting asterisk -c until the colors stopped + changing at odd locations. + + ASTERISK-25585 #close + + Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f + +2015-11-24 13:54 +0000 [59881fbb99] David M. Lee <dlee@respoke.io> + + * Fixed some typos + + Fixes some minor typos in the CHANGES file, plus an embarrasing typo in + the StatsD API. + + Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7 + +2015-11-24 13:07 +0000 [b75f587d15] Corey Farrell <git@cfware.com> + + * res_pjsip_notify: Fix CLI usage info + + The usage info for 'pjsip send notify' previously referenced the + chan_sip configuration sip_notify.conf. Fix this to reference + the correct configuration pjsip_notify.conf. + + ASTERISK-25590 #close + + Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea + +2015-11-23 14:27 +0000 [fc45f4040d] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_realtime.c: Fix crash from NULL sorcery object type. + + If the sorcery object type is not found a NULL is returned. + Unfortunately, sorcery_realtime_filter_objectset() will crash after + complaining about not finding the object type and saying to expect errors. + + * Use ao2_cleanup() instead of ao2_ref() to prevent the crash. + + ASTERISK-25165 + Reported by Corey Farrell + + Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97 + +2015-11-20 21:08 +0000 [4875e5ac32] Matt Jordan <mjordan@digium.com> + + * chan_pjsip: Handle T.38 faxes with direct media bridges + + When a channel is in a direct media bridge, a re-INVITE may arrive that forces + Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge + must change its technology to a simple bridge, and re-INVITE the media back + to Asterisk. + + Generally, this logic mostly already exists in Asterisk. However, prior to this + patch, there were a few bugs: + (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from + ever entering into a direct media bridge. This applies even when the only + media being passed over the channel is audio. This patch fixes this bug + by having the framehook specify that it defers caring about any frame type. + This allows the channels to enter into a direct media bridge, which will + be broken when a re-INVITE is received. + (2) When a re-INVITE is received, nothing instructed the bridging layer to + re-inspect the allowed bridging technology. This now occurs when either + a re-INVITE is received from a peer, or when a response is received from + the far end (that is, when the T.38 state changes to either + T38_PEER_REINVITE or T38_LOCAL_REINVITE). + (3) chan_pjsip needs to do a small amount of work to prevent a direct media + bridge from being chosen when a T.38 session is in progress. When a T.38 + session supplement has a t38 datastore - which is added when we detect + we should start thinking about T.38 on a channel - we now refuse a native + RTP bridge. + (4) When a BYE request is received, we don't terminate the T.38 session. If + the other side of a T.38 fax survives the hangup (due to the 'g' flag + in Dial, for example), we don't currently re-INVITE the media on the + other channel back to audio. This patch now has res_pjsip_t38 intercept + BYE requests and inform the far side that the T.38 session is terminated. + This naturally causes the correct re-INVITEs to be sent. + + ASTERISK-25582 + + Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb + +2015-11-20 21:07 +0000 [2b94d9a10d] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_t38: Add debug statements + + This patch adds some debug statements to res_pjsip_t38. These statements help + to determine which SDP negotiation callbacks are being executed, and, when + a particular callback exits, why a callback may not have applied its logic + to the local or remote SDP. + + Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77 + +2015-10-22 09:44 +0000 [af288b2d96] Matt Jordan <mjordan@digium.com> + + * main/cli: Use proper string methods to check existence of context/exten/app + + Because the context, extension, and application are stored in stringfields, + checking for them being NULL doesn't work so well. This patch uses the + appropriate string library call, ast_strlen_zero, to see if there is a value + in the context/exten/app values. + + Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23 + +2015-11-18 09:43 +0000 [d27aac0a9d] Matt Jordan <mjordan@digium.com> + + * res/res_endpoint_stats: Add module to emit endpoint StatsD statistics + + This patch adds a module that emits StatsD statistics about Asterisk + endpoints. This includes: + * A GUAGE statistic for endpoint states, tracking how many endpoints are in + a particular state. + * A GUAGE statistic for each endpoint, counting the number of channels + currently associated with an endpoint. + + ASTERISK-25572 + + Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305 + +2015-11-18 10:07 +0000 [90d9a70789] Matt Jordan <mjordan@digium.com> + + * res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts + + This patch adds the ability to send StatsD statistics related to the + state of PJSIP contacts. This includes: + * A GUAGE statistic measuring the count of contacts in a particular state. + This measures how many contacts are reachable, unreachable, etc. + * The RTT time for each contact, if those contacts are qualified. This + provides StatsD engines useful time-based data about each contact. + + ASTERISK-25571 + + Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c + +2015-11-13 10:34 +0000 [75097a0955] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_outbound_registration: Add registration statistics for StatsD + + This patch adds outbound registration statistics for StatsD. This includes + the following: + * A GUAGE metric for the overall count of outbound registrations. + * A GUAGE metric for each state an outbound registration can be in. As the + outbound registrations change state, the overall count of how many + outbound registrations are in the particular state is changed. + + These statistics are particularly useful for systems with a large number of + SIP trunks, and where measuring the change in state of the trunks is useful + for monitoring. + + ASTERISK-25571 + + Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37 + +2015-11-19 09:40 +0000 [8f71263e72] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_outbound_registration: Apply configuration on object type load + + When Asterisk is configured to use a dynamic sorcery backend (such as + res_sorcery_astdb) with 'registration' objects, it will fail to create the + internal state objects associated with the registration objects on module + load. This is due to nothing actually querying for the specific objects + and calling their sorcery apply handler during module load. + + This patch fixes that by calling get_registrations in the sorcery observer's + object_type_loaded handler. Doing this causes the sorcery backends to be + asked for the current state of all registration objects, which causes the + apply handler to be called and the internal run-time state to be created. + + ASTERISK-25575 #close + + Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23 + +2015-11-11 11:51 +0000 [0b508789ab] Alexander Traud <pabstraud@compuserve.com> + + * translate: Provide translation modules the result of SDP negotiation. + + Previously, a trancoding module did not have access to the joint but cached + format. Therefore, the module did not have access to the attributes negotiated + via SDP (line fmtp). Now, a translation module receives the joint format. + + ASTERISK-25545 #close + + Change-Id: Id6878a989b50573298dab115d3371ea369e1a718 + +2015-11-19 01:14 +0000 [1aa552b2a2] Alexander Traud <pabstraud@compuserve.com> + + * res_format_attr_h264: Do not reset string buffer. + + When no parameter is present, Asterisk does not generate the line fmtp, as + expected. However, because a buffer was reset, even rtpmap and fmtp of previous + media codecs got removed. Now, Asterisk does not reset other codecs in case of + no parameter for H.264. + + ASTERISK-25573 #close + + Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286 + +2015-11-18 10:05 +0000 [3354b325c6] Matt Jordan <mjordan@digium.com> + + * res_statsd: Add functions that support variable arguments + + Often, the metric names of statistics we are generating for StatsD have some + dynamic component to them. This can be the name of a particular resource, or + some internal status label in Asterisk. With the current set of functions, + callers of the statsd API must first build the metric name themselves, then + pass this to the API functions. This results in a large amount of boilerplate + code and usage of either fixed length static buffers or dynamic memory + allocation, neither of which is desireable. + + This patch adds two new functions to the StatsD API that support a printf + style format specifier for constructing the metric name. A dynamic string, + allocated in threadstorage, is used to build the metric name. This eases + the burden on users of the StatsD API. + + Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea + +2015-11-17 14:53 +0000 [d4a522d587] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts. + + Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d + +2015-11-17 14:53 +0000 [e44ab3816c] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Fix 423 response handling. + + Receiving a 423 Interval Too Brief response after authentication for an + outbound registration attempt results in assuming that the registrar has + rejected the registration permanently. If there are no configured retries + for fatal responses then the outbound registration is stopped for that + endpoint. + + For registrations, PJSIP/PJPROJECT intercepts the handling of 423 + responses and does not include any authentication in the updated + registration request. When the updated request is challenged then the + Asterisk code assumes that we were challenged again because the peer + rejected the authentication we sent earlier. + + * Made registration challenges keep track of the CSeq number to determine + if the received challenge response was for the request we thought we sent. + If the response's CSeq number differs from the CSeq number we last sent + with authentication then authenticate again because it is a challenge to a + different request. + + Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09 + +2015-11-03 14:36 +0000 [1e0040b88f] Tyler Cambron <tcambron@digium.com> + + * StatsD: Add res_statsd compatibility + + Added a new api to res_statsd.c to allow it to receive a + character pointer for the value argument. This allows for a + '+' and a '-' to easily be sent with the value. + + ASTERISK-25419 + Reported By: Ashley Sanders + + Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611 + +2015-11-16 13:56 +0000 [f62b642fe3] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip: Fix off nominal crash with requests that fail and have a timer + + When a request is sent using pjsip_endpt_send_request and fails, a condition + exists where the request wrapper, which is an AO2 object, may be de-ref'd + more times than it should. This occurs when the request's callback is called, + and, in the callback, the timer on the PJSIP heap is cancelled. When that + occurs, the request wrapper's lifetime is decremented. When + pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of + the request wrapper again, even though we've already cancelled the reference + associated with the timer. + + This patch checks the return result of pj_timer_heap_cancel_if_active before + removing the reference associated with the timer. We now only decrement it + in this case if a timer is cancelled as a result of the function call. + + Change-Id: I21332343a1a019c1117076f9bf2df27be2850102 + +2015-11-13 14:03 +0000 [fdd2afcd16] Mark Michelson <mmichelson@digium.com> + + * Confbridge: Add a user timeout option + + This option adds the ability to specify a timeout, in seconds, for a + participant in a ConfBridge. When the user's timeout has been reached, + the user is ejected from the conference with the CONFBRIDGE_RESULT + channel variable set to "TIMEOUT". + + The rationale for this change is that there have been times where we + have seen channels get "stuck" in ConfBridge because a network issue + results in a SIP BYE not being received by Asterisk. While these + channels can be hung up manually via CLI/AMI/ARI, adding some sort of + automatic cleanup of the channels is a nice feature to have. + + ASTERISK-25549 #close + Reported by Mark Michelson + + Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98 + +2015-11-16 04:29 +0000 [7debb986a5] Alec Davis <sivad.a@paradise.net.nz> + + * app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! + + commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) + refer ASTERISK-24958 + + above commit removed ast_channel_lock(qe->chan); + but failed to remove corresponding ast_channel_unlock(qe->chan); + + ASTERISK-25561 #close + Reported Alec Davis + + Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a + +2015-11-14 07:02 +0000 [afd9a89e5a] Joshua Colp <jcolp@digium.com> + + * hashtab: Add NULL check when destroying iterator. + + The hashtab API is pretty NULL tolerant which has resulted + in remaining callers not doing much checks themselves. + Unfortunately the function to destroy an iterator does not + do a NULL check and will result in a crash if passed NULL. + This change fixes that. + + ASTERISK-25552 #close + + Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619 + +2015-11-13 14:32 +0000 [c0f2f8de45] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_rfc3326.c: Fix crash when channel goes away. + + If an authenticated incoming caller does not respond to our 200 OK INVITE + response with an ACK then PJSIP will hangup the call. Unfortunately, + there is a chance that the session's channel will go away between one use + of the channel pointer and another when building the BYE request because + the BYE is being built by the monitor thread and not the call's serializer + thread. + + * Added a check to ensure that the thread trying to add the Reason header + is the call's serializer thread. This ensures that the channel will not + go away on us. + + Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89 + +2015-11-13 14:19 +0000 [4f43b85c92] Mark Michelson <mmichelson@digium.com> + + * Taskprocessors: Increase high-water mark + + In practical tests, we have seen certain taskprocessors, specifically + Stasis subscription taskprocessors, cross the recently-added high-water + mark and emit a warning. This high-water mark warning is only intended + to be emitted when things have tanked on the system and things are + heading south quickly. In the practical tests, the Stasis taskprocessors + sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in + any danger at all. + + As such, this ups the high-water mark to 500 tasks instead. It also + redefines the SIP threadpool request denial number to be a multiple of + the taskprocessor high-water mark. + + Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce + +2015-11-11 11:46 +0000 [d8d3991390] Alexander Traud <pabstraud@compuserve.com> + + * format: Register format-attribute module with cached formats. + + In Asterisk 13, cached formats are created before their corresponding format- + attribute module is registered. Cached formats are involved when a local + extension is called. Therefore, ast_format_generate_sdp_fmtp did not work + on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264, + and format-attribute modules provided externally. + + ASTERISK-25160 #close + + Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354 + +2015-11-12 11:17 +0000 [367972e42d] Mark Michelson <mmichelson@digium.com> + + * res_pjsip distributor: Don't send 503 response to responses. + + When the SIP threadpool is backed up with tasks, we send 503 responses + to ensure that we don't try to overload ourselves. The problem is that + we were not insuring that we were not trying to send a 503 to an + incoming SIP response. + + This change makes it so that we only send the 503 on incoming requests. + + Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404 + +2015-11-11 17:11 +0000 [2f9cb7d62b] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Deny requests when threadpool queue is backed up. + + We have observed situations where the SIP threadpool may become + deadlocked. However, because incoming traffic is still arriving, the SIP + threadpool's queue can continue to grow, eventually running the system + out of memory. + + This change makes it so that incoming traffic gets rejected with a 503 + response if the queue is backed up too much. + + Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816 + +2015-11-12 06:24 +0000 [4e5bf12b33] Joshua Colp <jcolp@digium.com> + + * format_cap: Don't append the 'none' format when appending all. + + When appending all formats of a type all the codecs are iterated + and added. This operation was incorrectly adding the ast_format_none + format which is special in that it is supposed to be used when no + format is present. It shouldn't be appended. + + ASTERISK-25535 + + Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c + +2015-11-11 04:16 +0000 [07583c2888] Steve Davies <steve@one47.co.uk> + + * Further fixes to improper usage of scheduler + + When ASTERISK-25449 was closed, a number of scheduler issues mentioned in + the comments were missed. These have since beed raised in ASTERISK-25476 + and elsewhere. + + This patch attempts to collect all of the scheduler issues discovered so + far and address them sensibly. + + ASTERISK-25476 #close + + Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b + +2015-11-11 11:04 +0000 [b818d70533] Joshua Colp <jcolp@digium.com> + + * threadpool: Handle worker thread transitioning to dead when going active. + + This change adds handling of dead worker threads when moving them + to be active. When this happens the worker thread is removed from + both the active and idle threads container. If no threads are able + to be moved to active then the pool grows as configured. + + A unit test has also been added which thrashes the idle timeout + and thread activation to exploit any race conditions between the + two. + + ASTERISK-25546 #close + + Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143 + +2015-11-10 09:27 +0000 [4bf84459c7] Alexander Traud <pabstraud@compuserve.com> + + * rtp_engine: Init a format-attribute module to its RFC defaults. + + Previously, format-attribute modules relied on an existing fmtp line in SDP + negotiation. However, fmtp is optional for several formats like the Opus Codec. + Now, the format-attribute module is called with an empty fmtp, which allows the + module to initialise itself to RFC defaults. Furthermore now, Asterisk is able + to differentiate between internally and externally created formats. + + ASTERISK-25537 #close + + Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52 + +2015-11-09 03:04 +0000 [1bff400df7] Alexander Traud <pabstraud@compuserve.com> + + * ast_format_cap_get_names: To display all formats, the buffer was increased. + + ASTERISK-25533 #close + + Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a + +2015-11-09 07:04 +0000 [f3ac4d8090] Alexander Traud <pabstraud@compuserve.com> + + * ast_format_cap: Avoid format creation on module load, use cache instead. + + Since Asterisk 13, formats are immutable and cached. However while loading a + module like chan_sip, some formats were created instead using cached ones. + + ASTERISK-25535 #close + + Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b + +2015-11-06 07:54 +0000 [6d1bdb9d3b] Walter Doekes <walter+asterisk@wjd.nu> + + * func_callerid: Document that CALLERID(pres) is available. + + CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) + and CALLERID(name-pres). But for channel driver that don't make a + distinction between the two (e.g. SIP), it makes more sense to get/set + both at once. This change reveals the availability of CALLERID(pres), + CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and + REDIRECTING(from-pres). + + ASTERISK-25373 #close + + Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a +2015-11-06 07:52 +0000 [8410336681] Walter Doekes <walter+asterisk@wjd.nu> + + * docs: Fix a few typo's in app docs (more then, resourse). + + Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7 + +2015-11-06 07:36 +0000 [0d425f2eb4] Walter Doekes <walter+asterisk@wjd.nu> + + * xmldoc: Improve xmldoc wrapping of 'core show ...' output. + + Previously, the wrapping did both lookahead and lookback, which, + together with color escape sequences, caused some lines to be wrapped + way earlier than other lines. This led to inconsistent output. + + This simplifies the wrapping code and makes it more sane: if maxcolumns + is hit, we simply jump back to the last space and wrap there. + + ASTERISK-25527 #close + + Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957 + +2015-11-06 06:57 +0000 [33752e0837] Sean Bright (license #5060) + + * res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP. + + In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual + amount of channels is negotiated in-band. Therefore now, the Opus codec and its + attribute rtpmap are registered with two channels. + + ASTERISK-24779 #close + Reported by: PowerPBX + Tested by: Alexander Traud + patches: + asterisk-24779.patch submitted by Sean Bright (license #5060) + + Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b + +2015-11-03 16:19 +0000 [6ff48319d9] Jonathan Rose <jrose@digium.com> + + * taskprocessor: Add high water mark warnings + + If a taskprocessor's queue grows large, this can indicate that there + may be a problem with tasks not leaving the processor or else that + the number of available task processors for a given type of task is + too low. This patch makes it so that if a taskprocessor's task queue + grows above 100 queued tasks that it will emit a warning message. + Warning messages are emitted only once per task processor. + + ASTERISK-25518 #close + Reported by: Jonathan Rose + + Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c + +2015-11-04 14:31 +0000 [506aea26e6] Matt Jordan <mjordan@digium.com> + + * main/dial: Protect access to the format_cap structure of the requesting channel + + When a dial attempt is made that involves a requesting channel, we previously + were not: + a) Protecting access to the native format capabilities structure on the + requesting channel. That is inherently unsafe. + b) Reference bumping the lifetime of the format capabilities structure. + + In both cases, something else could sneak in, blow away the format + capabilities, and we'd be holding onto an invalid format_cap structure. When + the newly created channel attempts to construct its format capabilities, things + go poorly. + + This patch: + a) Ensures that we get a reference to the native format capabilities while + the requesting channel is locked + b) Holds a reference to the native format capabilities during the creation + of the new channel. + + ASTERISK-25522 #close + + Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f + +2015-10-30 22:57 +0000 [d098d00424] Corey Farrell <git@cfware.com> + + * Fix cli display of build options. + + A previous commit reduced the AST_BUILDOPTS compiler define to + only include options that affected ABI. This included some options + that were previously displayed by cli "core show settings". This + change corrects the CLI display while still restricting buildopts.h + to ABI effecting options only. + + ASTERISK-25434 #close + Reported by: Rusty Newton + + Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325 + +2015-11-03 11:15 +0000 [afec1b1b64] Matt Jordan <mjordan@digium.com> + + * res_pjsip/location: Destroy contact_status objects on contact deletion + + The contact_status Sorcery objects are currently not destroyed when a contact + is deleted. This causes the contact's last known RTT/status to be 'sticky' + when the contact itself may no longer exist. This patch causes the + contact_status objects associated with both dynamic and static contacts to + be destroyed if the AoR holding those contacts is also destroyed (or via + other paths where a contact may be deleted.) + + Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e + +2015-11-03 10:58 +0000 [715f770c9f] Matt Jordan <mjordan@digium.com> + + * pjsip_configuration: On delete, remove the persistent version of an endpoint + + When an endpoint is deleted (such as through an API), the persistent endpoint + currently continues to lurk around. While this isn't harmful from a memory + consumption perspective - as all persistent endpoints are reclaimed on + shutdown - it does cause Stasis endpoint related operations to continue + to believe that the endpoint may or may not exist. + + This patch causes the persistent endpoint related to a PJSIP endpoint to be + destroyed if the PJSIP endpoint is deleted. + + Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb +2015-11-03 08:15 +0000 [f0f190af08] Matt Jordan <mjordan@digium.com> + + * main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field + + The JSON packing for the ContactStatusChange event forgot to include the + roundtrip_usec field. As a result, the field never showed up in any event, + even when the data was available. This patch corrects that error by properly + packing the JSON blob with the data. + + Change-Id: I8df80da659a44010afbd48f645967518ff5daa17 + +2015-11-02 20:24 +0000 [0393bd6bed] Corey Farrell <git@cfware.com> + + * chan_sip: Allow websockets to be disabled. + + This patch adds a new setting "websockets_enabled" to sip.conf. + Setting this to false allows chan_sip to be used without causing + conflicts with res_pjsip_transport_websocket. + + ASTERISK-24106 #close + Reported by: Andrew Nagy + + Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7 + +2015-11-02 17:19 +0000 [6fbffe42e1] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Set threadpool max size default to 50. + + During a stress test of subscriptions, a huge blast of + subscription-related traffic resulted in the threadpool expanding to a + ridiculous number of threads. The balooning of threads resulted in an + increase of memory, which led to a crash due to being out of memory. + + An easy fix for the particular test was to limit the size of the + threadpool, thus reining in the amount of memory that would be used. It + was decided that there really is no downside to having a non-infinite + default value for the maximum size of the threadpool, so this change + introduces 50 threads as the maximum threadpool size for the SIP + threadpool. + + ASTERISK-25513 #close + Reported by John Bigelow + + Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be + +2015-11-02 06:57 +0000 [11e54b1932] Matt Jordan <mjordan@digium.com> + + * pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction + + When an AoR is created or destroyed dynamically, the scheduled OPTIONS + requests that qualify the contacts on the AoR are not necessarily started + or destroyed, particularly for persistent contacts created for that AoR. + This patch adds create/update/delete sorcery observers for an AoR, which + schedule/unschedule the qualifies as expected. + + Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d + +2015-10-30 13:22 +0000 [118d628e08] Matt Jordan <mjordan@digium.com> + + * Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs + + This patch adds a rule for installing the Super Awesome Company based 'Basic + PBX' configuration files. As part of adding this rule, a bit of the content + that makes up installing the configuration files under the 'samples' target + was refactored into a make subroutine for usage by additional later config + make targets. + + Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404 +2015-10-29 08:28 +0000 [9a021a42ad] Joshua Colp <jcolp@digium.com> + + * res_pjsip_pubsub: Fix assertion when UAS dialog creation fails. + + When compiled with assertions enabled one will occur when destroying + the subscription tree when UAS dialog creation fails. This is because + the code assumes that a dialog will always exist on a subscription + tree when in reality during this specific scenario it won't. + + This change makes it so a dialog is not removed from the subscription + tree if it is not present. + + ASTERISK-25505 #close + + Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee + +2015-10-26 11:42 +0000 [1256aedf66] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Do not send all codecs on INVITE. + + Since version 13, Asterisk sent all allowed codecs as callee, even when the + caller did not request/support them. In case of dynamic RTP payloads, this led + to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the + intersection between the requested and the supported codecs is send again. + + ASTERISK-24543 #close + + Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287 + +2015-10-24 13:08 +0000 [5f593e7c38] gtjoseph <george.joseph@fairview5.com> + + * build: GCC 5.1.x catches some new const, array bounds and missing paren issues + + Fixed 1 issue in each of the affected files. + + ASTERISK-25494 #close + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77 + +2015-10-20 16:02 +0000 [162acd45f7] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Add "like" processing to pjsip list and show commands + + Add the ability to filter output from pjsip list and show commands + using the "like" predicate like chan_sip. + + For endpoints, aors, auths, registrations, identifyies and transports, + the modification was a simple change of an ast_sorcery_retrieve_by_fields + call to ast_sorcery_retrieve_by_regex. For channels and contacts a + little more work had to be done because neither of those objects are + true sorcery objects. That was just removing the non-matching object + from the final container. Of course, a little extra plumbing in the + common pjsip_cli code was needed to parse the "like" and pass the regex + to the get_container callbacks. + + Some of the get_container code in res_pjsip_endpoint_identifier was also + refactored for simplicity. + + ASTERISK-25477 #close + Reported by: Bryant Zimmerman + Tested by: George Joseph + + Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1 + +2015-10-21 11:51 +0000 [c58091737d] Kevin Harwell <kharwell@digium.com> + + * res_pjsip_outbound_registration: registration stops due to fatal 4xx response + + During outbound registration it is possible to receive a fatal (any permanent/ + non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due + to a problem with the registrar itself. Upon receiving the failure response + Asterisk terminates outbound registration for the given endpoint. + + This patch adds an option, 'fatal_retry_interval', that when set continues + outbound registration at the given interval up to 'max_retries' upon receiving + a fatal response. + + ASTERISK-25485 #close + + Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2 + +2015-10-22 17:07 +0000 [ebe69dee0d] Mark Michelson <mmichelson@digium.com> + + * format_cap: Detect vector allocation failures. + + A crash was seen on a system that ran out of memory due to Asterisk not + checking for vector allocation failures in format_cap.c. With this + change, if either of the AST_VECTOR_INIT calls fail, we will return a + value indicating failure. + + Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8 + +2015-10-02 15:32 +0000 [3b19efefef] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog. + + A certain situation can result in our attempting to send a NOTIFY on a + destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but + that subscriber has dropped off the network. We end up retransmitting + that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY + transaction. When the pjsip evsub code is told that the transaction has + been terminated, it responds in kind by alerting us that the + subscription has been terminated, destroying the subscription, and then + removing its reference to the dialog, thus destroying the dialog. + + The problem is that when we get told that the subscription is being + terminated, we detect that we have not sent a terminating NOTIFY + request, so we queue up such a NOTIFY to be sent out. By the time that + queued NOTIFY gets sent, the dialog has been destroyed, so attempting to + send that NOTIFY can result in a crash. + + The fix being introduced here is actually a reintroduction of something + the pubsub code used to employ. We hold a reference to the dialog and + wait to decrement our reference to the dialog until our subscription + tree object is destroyed. This way, we can send messages on the dialog + even if the PJSIP evsub code wants to terminate earlier than we would + like. + + In doing this, some NULL checks for subscription tree dialogs have been + removed since NULL dialogs are no longer actually possible. + + Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5 + +2015-09-29 14:53 +0000 [0a346f095f] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Ensure dialog lock balance. + + When sending a NOTIFY, we lock the dialog and then unlock the dialog + when finished. A recent change made it so that the subscription tree's + dialog pointer will be set NULL when sending the final NOTIFY request + out. This means that when we attempt to unlock the dialog, we pass a + NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog + remains locked after we think we have unlocked it. When a response to + the NOTIFY arrives, the monitor thread attempts to lock the dialog, but + it cannot because we never released the dialog lock. This results in + Asterisk being unable to process incoming SIP traffic any longer. + + The fix in this patch is to use a local pointer to save off the pointer + value of the subscription tree's dialog when locking and unlocking the + dialog. This way, if the subscription tree's dialog pointer is NULLed + out, the local pointer will still have point to the proper place and the + dialog lock will be unlocked as we expect. + + Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a + +2015-09-28 16:36 +0000 [ad39508095] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Prevent crashes on final NOTIFY. + + The SIP dialog is removed from the subscription tree when the final + NOTIFY is sent. However, after the final NOTIFY is sent, the persistence + update function still attempts to access the cseq from the dialog, + resulting in a crash. + + This fix removes the subscription persistence at the same time that the + dialog is removed from the subscription tree. This way, there is no + attempt to update persistence when the subscription is being destroyed. + + Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb + +2015-09-17 17:28 +0000 [067f408760] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Remove serializer when sending final NOTIFY. + + There have been crashes seen where a taskprocessor's listener is NULL + unexpectedly. + + Looking at backtraces, the problem was specifically seen in PJSIP + serializers. + + Subscriptions make the mistake of removing a serializer from a dialog + during subscription tree destruction. Since subscription trees are + reference-counted, guaranteeing the circumstances behind the destruction + are not possible. This makes it so that the dialog serializer can be + removed while not holding the dialog lock. This makes it possible for + the distributor to get a pointer to the dialog serializer and have that + serializer get freed out from under it. + + The fix for this is to remove the serializer from a subscription dialog + when sending the final NOTIFY. This guarantees that the serializer is + removed with the dialog lock held. By doing this, we guarantee that if + the distributor gains access to the dialog's serializer, it will not be + possible for the serializer to get freed by another thread. + + Change-Id: I21f5dac33529f65cec45679bdace60670800ff66 + +2015-09-02 09:14 +0000 [1bcc592765] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Fix crash on destruction of empty subscription tree. + + If an old persistent subscription is recreated but then immediately + destroyed because it is out of date, the subscription tree will have no + leaf subscriptions on it. This was resulting in a crash when attempting + to destroy the subscription tree. + + A simple NULL check fixes this problem. + + Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac + +2015-09-01 15:47 +0000 [b3cc2bd7df] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Solidify lifetime and ownership of objects. + + There have been crashes and general instability seen in the pubsub code, + so this patch introduces three changes to increase the stability. + + First, the ownership model for subscriptions has been modified. Due to + RLS, subscriptions are stored in memory as a tree structure. Prior to my + patch, the PJSIP subscription was the owner of the subscription tree. + When the PJSIP subscription told us that it was terminating, we started + destroying the subscription tree along with all of the individual leaf + subscriptions that belong to the tree. The problem with this model is + that the two actors in play here, the PJSIP subscription and the + individual leaf subscriptions, need to have joint ownership of the + subscription tree. So now, the PJSIP subscription and the individual + leaf subscriptions each have a reference to the subscription tree. This + way, we will not actually free memory until no players are left that + care. The PJSIP subscription is a bigger stakeholder, in that if the + PJSIP subscription's reference to the subscription tree is removed, the + subscription tree instructs the leaf subscriptions to shut down and drop + their references to the subscription tree when possible. The individual + leaf subscriptions, upon being told to shut down, can drop their stasis + subscriptions or whatever they use to learn of new state, and then drop + their reference to the subscription tree once they are ready to die. + + Second, the lifetime of a PJSIP subscription's reference to our + subscription tree has been altered. As I learned from doing a deep dive, + the PJSIP evsub code can tell Asterisk multiple times that the + subscription has been terminated, and not all of these times + are especially helpful. I have altered the message flow that we use for + SIP subscriptions such that we will always drop the PJSIP subscription's + reference to the subscription tree when we send the NOTIFY that + terminates a SIP subscription. This also means that we will now queue + NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so + that we can have predictable state changes from the PJSIP evsub code. + + Third, the synchronization of operations has been improved. PJSIP can + call into our code from a serializer thread (e.g. upon receiving an + incoming request) or from the monitor thread (e.g. when a subscription + times out). Because of this, there is the possibility of competing + threads stepping on each other. PJSIP attempts to do some + synchronization on its own by always keeping the dialog lock held when + it calls into us. However, since we end up pushing tasks into the + serializer, the result was that serialized operations were not grabbing + the dialog lock and could, as a result, step on something that was being + attempted by a different thread. Now we ensure that serialized + operations grab the dialog lock, then check for extenuating + circumstances, then proceed with their operation if they can. + + Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5 + +2015-10-19 15:28 +0000 [c8c65dfa41] Richard Mudgett <rmudgett@digium.com> + + * strings.c: Fix __ast_str_helper() to always return a terminated string. + + Users of functions which call __ast_str_helper() such as the ones listed + below are likely to not check the return value for failure so ensuring + that the string is always nil terminated is a good safety measure. + + ast_str_set_va() + ast_str_append_va() + ast_str_set() + ast_str_append() + + Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07 + +2015-10-19 15:27 +0000 [b271d4a28a] Richard Mudgett <rmudgett@digium.com> + + * Add missing failure checks to ast_str_set_va() callers. + + Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f + +2015-10-21 11:44 +0000 [f2725c8b77] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Move URI validation to use time. + + In a realtime based system with a limited number of threadpool threads + it is possible for a deadlock to occur. This happens when permanent + endpoint state is updated, which will cause database queries to be done. + These queries may result in URI validation being done which is done + synchronously using a PJSIP thread. If all PJSIP threads are in use + processing traffic they themselves may be blocked waiting to get the + permanent endpoint container lock when identifying an endpoint. + + This change moves URI validation to occur at use time instead of + configuration time. While this comes at a cost of not seeing a problem + until you use it it does solve the underlying deadlock problem. + + ASTERISK-25486 #close + + Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a + +2015-10-21 08:08 +0000 [84ff075d41] Alexander Traud <pabstraud@compuserve.com> + + * format: Update the maximum packetization time for iLBC 30. + + In September 2006, the maximum packetization time (ptime) were set to such a + low value, packetization was disabled for many codecs actually. This was fixed + for many codecs but not for iLBC 30. This enables packetization for iLBC which + can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf. + + ASTERISK-7803 + + Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12 +2015-10-21 09:51 +0000 [869ef2a8ee] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Fix autoframing=yes. + + With Asterisk 13, the structures ast_format and ast_codec changed. Because of + that, the paketization timing (framing) of the RTP channel moved away from the + formats/codecs. In the course of that change, the ptime of the callee was not + honored anymore, when the optional autoframing was enabled. + + ASTERISK-25484 #close + + Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4 + +2015-10-20 22:24 +0000 [9fd2adc204] Matt Jordan <mjordan@digium.com> + + * rest-api-templates: Wikify error code response reasons + + Error response code descriptions may contain wiki markup that need to be + escaped. Without this patch, Confluence will reject the document being sent + and the responsible script will raise an exception. + + Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5 + +2015-10-20 12:06 +0000 [72cbb6df55] Matt Jordan <mjordan@digium.com> + + * funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function + + When ab803ec342 was committed, it accidentally forgot to actually *add* the + HOLD_INTERCEPT function. This highlights two interesting points: + * Gerrit forces you to put the patch as it is going to into the repo up for + review, which Review Board did not. Yay Gerrit. + * No one apparently bothered to use this feature, or else they don't know about + it. I'm going to go with the latter explanation. + + ASTERISK-24922 + + Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396 + +2015-10-19 19:59 +0000 [9fc9777fa3] Matt Jordan <mjordan@digium.com> + + * contrib/scripts/autosupport: Update for Asterisk 13 + + This patch adds some minor tweaks for autosupport to update it for Asterisk 13. + This includes: + * Finally removing most references to Zaptel + * Adding support for some additional 'core' commands, and fixing nomenclature + that generally hasn't been used for some time + * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels + + Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 + +2015-10-14 14:15 +0000 [dc6ec661b3] mdu113 <mulitskiy@acedsl.com> + + * res_config_pgsql.c: Fix deadlock loading realtime configuration. + + On v13, loading several thousand PJSIP endpoints on Asterisk start causes + a deadlock most of the time. + + Thanks to mdu113 for discovering that there was a call to pgsql_exec() not + protected by the pgsql_lock reentrancy lock. + + {quote} + I believe a code path exists that attempts to use pgsql connection without + locking pgsql_lock. I believe what happens during that deadlock that I + see is two concurrent threads are both attempting to send query to pgsql, + one of the thread is using a code path without locking pgsql_lock. If + they managed to send queries at the same time, it seems postgres ignores + one of the queries and replies only to the one of them. If it happens so + that the thread holding the lock didn't receive the reply it will wait for + it (and hold the lock) forever (or at least for very long time), thus + completely blocking all access to db. + {quote} + + * Added missing reentrancy locking around pgsql_exec() in find_table(). + + * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion + between the psql_tables list lock and the pgsql_lock. + + ASTERISK-25455 #close + Reported by: mdu113 + Patches: + res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113 + + Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2 + +2015-10-13 14:13 +0000 [f8707ae9a5] Olle Johansson (License 5267) + + * channels/chan_sip: Set cause code to 44 on RTP timeout + + To quote Olle: + + "When issuing a hangup due to RTP timeouts the cause code is not set. I have + selected 44 based on Cisco's implementation..." + + ASTERISK-25135 #close + Reported by: Olle Johansson + patches: + rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) + + Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc + +2015-10-10 15:20 +0000 [486b172b50] Ivan Poddubny <ivan.poddubny@gmail.com> + + * Build: Add menuselect options for using compiler sanitizers + + This patch adds menuselect options for building Asterisk with + various sanitizers provided by gcc and clang. + + When one of *SANITIZER flags is set in menuselect, the appropriate + option is added to CFLAGS ad LDFLAGS for the build. + + Information on sanitizers in the project wiki: + https://github.com/google/sanitizers/wiki + + GCC Manual: + https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html + + Clang Compiler User's Manual: + http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation + + ASTERISK-24718 #close + Reported by: Badalian Vyacheslav + + Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0 + +2015-10-12 11:21 +0000 [e14023ca35] Richard Mudgett <rmudgett@digium.com> + + * config.c: Fix off-nominal memory leak. + + Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 + +2015-10-12 11:20 +0000 [a99e821520] Richard Mudgett <rmudgett@digium.com> + + * config.c: Fix potential memory corruption after [section](+). + + The memory corruption could happen if the [section](+) is the last section + in the file with trailing comments. In this case process_text_line() has + left *last_cat is set to newcat and newcat is destroyed. + + Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93 + +2015-10-12 11:21 +0000 [8d31d2526b] Richard Mudgett <rmudgett@digium.com> + + * config.c: Fix #include after [section](+). + + An #include right after a [section](+) would associate any variable + assignments before a new section in the #include with the wrong section. + + * Fix section association by setting the current section to the appended + section. + + * Fix '+' and '!' section flag interaction corner case depending upon + which flag came first. If the '!' came first then it would be ignored. + If the '!' came after then it would affect the appended section. The '!' + will now no longer be ignored. + + ASTERISK-25461 #close + Reported by: Sean Pimental + + Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3 + +2015-10-06 18:01 +0000 [3329c714f7] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Fix deadlock when sending out-of-dialog requests. + + The struct send_request_wrapper has a pjsip lock associated with it that + is created non-recursive. There is a code path for the struct + send_request_wrapper lock that will attempt to lock it recursively. The + reporter's deadlock showed that the thread calling endpt_send_request() + deadlocked itself right after the wrapper object got created. + + Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited + MWI NOTIFY messages can hit this deadlock. + + * Replaced the struct send_request_wrapper pjsip lock with the mutex lock + that can come with an ao2 object since all of Asterisk's mutexes are + recursive. Benefits include removal of code maintaining the pjsip + non-recursive lock since ao2 objects already know how to maintain their + own lock and the lock will show up in the CLI "core show locks" output. + + ASTERISK-25435 #close + Reported by: Dmitriy Serov + + Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d + +2015-10-06 11:05 +0000 [a1435aa3fa] Stefan Engström <stefanen@kth.se> + + * res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending + + In ast_rtp_read, the value of the variable 'mark' which we try to assign to a + frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate + it to 0 or 1. + + ASTERISK-25451 #close + Change-Id: I53bdf5c026041730184a6a809009c028549ce626 + +2015-10-07 01:24 +0000 [3357678b94] Ivan Poddubny <ivan.poddubny@gmail.com> + + * func_presencestate: Return "not_set" when no data is set in AstDB + + Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not + exist, i.e. when a new CustomPresence is added in the dialplan. + + ASTERISK-25400 #close + Reported by: Andrew Nagy + + Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a + +2015-10-06 20:43 +0000 [b714b2152d] Matt Jordan <mjordan@digium.com> + + * res/res_rtp_asterisk: Fix assignment after ao2 decrement + + When we decide we will no longer schedule an RTCP write, we remove the + reference to the RTP instance, then assign -1 to the stored scheduler ID + in case something else comes along and wants to see if anything is scheduled. + + That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to + fix the regression introduced by 3cf0f29310, this improper assignment on a + potentially destroyed object started getting tripped on the build agents. + + Frankly, this should have been crashing a lot more often earlier. I can only + assume that the timing was changed just enough by both changes to start + actually hitting this problem. + + As it is, simply moving the assignment prior to the ao2 deference is sufficient + to keep the RTP instance from being referenced when it is very, truly, + aboslutely dead. + + (Note that it is still good practice to assign -1 to the scheduler ID when we + know we won't be scheduling it again, as the ao2 deref *may* not always destroy + the ao2 object.) + + ASTERISK-25449 + + Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7 + +2015-10-06 12:40 +0000 [f939e2bd48] Florian Sauerteig <ffs@ccn.net> + + * chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers. + + If a Via header containes an IPv6 address and a port number is ommitted, + as it is the standard port, we now leave the port empty and to not set it + to the value after the first colon of the IPv6 address. + + ASTERISK-25443 #close + + Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70 + +2015-10-05 16:53 +0000 [426263a64d] Richard Mudgett <rmudgett@digium.com> + + * chan_pjsip: Fix crash on reINVITE before initial INVITE completes. + + Apparently some endpoints attempt to send a reINVITE before completing the + initial INVITE transaction. In this case PJSIP responds appropriately to + the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip + is using the initial INVITE transaction state to determine if an INVITE is + the initial INVITE or a reINVITE. Since the initial INVITE transaction + has not been confirmed yet chan_pjsip thinks the reINVITE is an initial + INVITE and starts another PBX thread on the channel. The extra PBX thread + ensures that hilarity ensues. + + * Fix checks for a reINVITE on incoming requests to look for the presence + of a to-tag instead of the initial INVITE transaction state. + + * Made caller_id_incoming_request() determine what to do if there is a + channel on the session or not. After a channel is created it is too late + to just store the new party id on the session because the session's party + id has already been copied to the channel's caller id. + + ASTERISK-25404 #close + Reported by: Chet Stevens + + Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be + +2015-10-05 21:34 +0000 [50fa9ff997] Matt Jordan <mjordan@digium.com> + + * Fix improper usage of scheduler exposed by 5c713fdf18f + + When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of + '0' returned. While this was valid per the documentation for the API, it was + apparently never returned previously. As a result, several users of the + scheduler API viewed the result as being invalid, causing them to reschedule + already scheduled items or otherwise fail in interesting ways. + + This patch corrects the users such that they view '0' as valid, and a returned + ID of -1 as being invalid. + + Note that the failing HEP RTCP tests now pass with this patch. These tests + failed due to a duplicate scheduling of the RTCP transmissions. + + ASTERISK-25449 #close + + Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39 +2015-08-26 16:58 +0000 [8f777ab584] Debian Amtelco <dan@amtelco.com> + + * chan_pjsip: Add Referred-By header to the PJSIP REFER packet. + + Some systems require the REFER packet to include a Referred-By header. + If the channel variable SIPREFERREDBYHDR is set, it passes that value as the + Referred-By header value. Otherwise, it adds the current dialog’s local info. + + Reported by: Dan Cropp + Tested by: Dan Cropp + + Change-Id: I3d17912ce548667edf53cb549e88a25475eda245 + +2015-10-03 06:27 +0000 [74635b5638] Ivan Poddubny <ivan.poddubny@gmail.com> + + * manager: Fix GetConfigJSON returning invalid JSON + + When GetConfigJSON was introduced back in 1.6, it returned each + section as an array of strings: ["key=value", "key2=value2"]. + Afterwards, it was changed a few times and became + ["key": "value", "key2": "value2"], which is not a correct JSON. + This patch fixes that by constructing a JSON object {} instead of + an array []. + + Also, the keys "istemplate" and "tempates" that are used to + indicate templates and their inherited categories are now wrapped in + quotes. + + ASTERISK-25391 #close + Reported by: Bojan Nemčić + + Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8 + +2015-09-30 17:28 +0000 [40c69e78f5] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_memory_cache.c: Fix deadlock with scheduler. + + A deadlock can happen when a sorcery object is being expired from the + memory cache when at the same time another object is being placed into the + memory cache. There are a couple other variations on this theme that + could cause the deadlock. Basically if an object is being expired from + the sorcery memory cache at the same time as another thread tries to + update the next object expiration timer the deadlock can happen. + + * Add a deadlock avoidance loop in expire_objects_from_cache() to check if + someone is trying to remove the scheduler callback from the scheduler. + + ASTERISK-25441 #close + + Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc + +2015-10-01 14:30 +0000 [dfeb513e85] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_memory_cache.c: Replace inline code with function. + + Make sorcery_memory_cache_close() call remove_all_from_cache() instead of + partially inlining it. + + ASTERISK-25441 + + Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c + +2015-10-01 14:27 +0000 [ced0a2d71b] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_memory_cache.c: Shutdown in a less crash potential order. + + Basically you should shutdown in the opposite order of how you setup since + later setup pieces likely depend on earlier setup pieces. e.g., + Registering your external API with the rest of the system should be the + last thing setup and the first thing unregistered during shutdown. + + Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e + +2015-09-30 17:27 +0000 [cc279eea11] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_memory_cache.c: Misc tweaks. + + Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160 + +2015-09-30 17:27 +0000 [9af3b613f6] Richard Mudgett <rmudgett@digium.com> + + * res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK. + + Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c + +2015-09-30 13:42 +0000 [56ed7b9dd5] Joshua Colp <jcolp@digium.com> + + * res_rtp_asterisk: Move "Set role" warning to be debug. + + In practice the set_role API callback can be invoked even + when no ICE is present on an RTP instance. This can occur + if ICE has not been enabled on it. + + ASTERISK-25438 #close + + Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69 + +2015-09-28 15:31 +0000 [ddebb217f0] Richard Mudgett <rmudgett@digium.com> + + * sched.c: Add warning about negative time interval request. + + Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc + +2015-09-29 21:15 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.6.0-rc1 Released. + +2015-09-29 16:12 +0000 [bba1c4066b] Kevin Harwell <kharwell@lunkwill> + + * Release summaries: Add summaries for 13.6.0-rc1 + +2015-09-29 16:08 +0000 [82c4aecdbb] Kevin Harwell <kharwell@lunkwill> + + * .version: Update for 13.6.0-rc1 + +2015-09-29 16:08 +0000 [bc18db7388] Kevin Harwell <kharwell@lunkwill> + + * .lastclean: Update for 13.6.0-rc1 + +2015-09-29 16:08 +0000 [b9c53f95e3] Kevin Harwell <kharwell@lunkwill> + + * realtime: Add database scripts for 13.6.0-rc1 + +2015-09-29 14:53 +0000 [d30939b6e8] Kevin Harwell <kharwell@digium.com> + + * ARI: Changed version from 1.8.0 to 1.9.0 + + Change-Id: I510991c60d28d171f47c4b58bba4947f7fc71b13 + +2015-09-25 18:37 +0000 [5f19c9bade] Richard Mudgett <rmudgett@digium.com> + + * res/ari/config.c: Fix user sort compare function. + + Made use the ao2 sort compare template function and OBJ_SEARCH_xxx + identifiers. + + Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c + +2015-09-25 17:26 +0000 [3a85764039] Richard Mudgett <rmudgett@digium.com> + + * res/ari/config.c: Optimize conf_alloc() object init. + + * Now conf_alloc() has more off nominal error checking. + + * Eliminated RAII_VAR() use in conf_alloc(). + + * Eliminated a dubius shortcut when destroying cfg->general in + conf_destructor() that would cause a crash if cfg->general failed to get + allocated. + + * Add some ACO registration section comments. + + Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a + +2015-09-25 16:48 +0000 [028033e5a8] Richard Mudgett <rmudgett@digium.com> + + * res/ari/config.c: Fix conf_alloc() object init. + + Need to finish initializing the string fields in the ao2 object before + putting any default strings into them. + + ASTERISK-25383 #close + Reported by: yaron nahum + + Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84 + +2015-09-27 20:45 +0000 [90165e306d] Matt Jordan <mjordan@digium.com> + + * res/res_stasis: Fix accidental subscription to 'all' bridge topic + + When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a + NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to + all bridges. Unfortunately, the res_stasis control loop did not check that + a bridge changing on a channel's control object was actually also non-NULL. + As a result, app_subscribe_bridge will be called with a NULL bridge when a + channel leaves a bridge. This causes a new subscription to be made to the + bridge. If an application has also subscribed to the bridge, the application + will now have two subscriptions: + (1) The explicit one created by the app + (2) The implicit one accidentally created by the control structure + + As a result, the 'BridgeDestroyed' event can be sent multiple times. This + patch corrects the control loop such that it only subscribes an application + to a new bridge if the bridge pointer is non-NULL. + + ASTERISK-24870 + + Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f + +2015-09-04 13:51 +0000 [e1223ff6db] Scott Griepentrog <scott@griepentrog.com> + + * Scripts: check file versions of Asterisk and dependencies + + To help in diagnosing mismatched modules and libraries, this + script scans for version, repository, and source information + and reports what is found. + + ASTERISK-25376 #close + Reported by: Ashley Sanders + + Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6 + +2015-09-24 14:56 +0000 [6b1e7583c1] Richard Mudgett <rmudgett@digium.com> + + * app_queue.c: Force COLP update if outgoing channel name changed. + + * When a call is answered and the outgoing channel name has changed then + force a connected line update because the channel is no longer the same. + The channel was masqueraded into by another channel. This is usually + because of a call pickup. + + Note: Forwarded calls are handled in a controlled manner so the original + channel name is replaced with the forwarded channel. + + ASTERISK-25423 #close + Reported by: John Hardin + + Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172 + +2015-09-24 14:20 +0000 [6bf304bf25] Richard Mudgett <rmudgett@digium.com> + + * app_queue.c: Factor out a connected line update routine. + + Replace inlined code with update_connected_line_from_peer(). + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3 + +2015-09-24 13:27 +0000 [e36b5f1e8e] Richard Mudgett <rmudgett@digium.com> + + * app_dial.c: Make 'A' option pass COLP updates. + + While the 'A' option is playing the announcement file allow the caller and + peer to exchange COLP update frames. + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9 + +2015-09-24 12:59 +0000 [747bfac895] Richard Mudgett <rmudgett@digium.com> + + * app_dial.c: Force COLP update if outgoing channel name changed. + + * When a call is answered and the outgoing channel name has changed then + force a connected line update because the channel is no longer the same. + The channel was masqueraded into by another channel. This is usually + because of a call pickup. + + Note: Forwarded calls are handled in a controlled manner so the original + channel name is replaced with the forwarded channel. + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c + +2015-09-24 12:37 +0000 [14481d9aa0] Richard Mudgett <rmudgett@digium.com> + + * app_dial.c: Factor out a connected line update routine. + + Replace inlined code with update_connected_line_from_peer(). + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091 + +2015-09-23 17:41 +0000 [bbeda190c3] Richard Mudgett <rmudgett@digium.com> + + * app_dial.c: Remove some no-op code. + + Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54 + +2015-09-23 14:02 +0000 [f050fa76eb] Mark Michelson <mmichelson@digium.com> + + * logger: Prevent duplicate dynamic channels from being added. + + There was a problem observed where the "logger add channel" CLI command + would allow for a channel with the same name to be added multiple times. + This would result in each message being written out to the same file + multiple times. + + The problem was due to the difference in how logger channel filenames + are stored versus the format they are allowed to be presented when they + are added. For instance, if adding the logger channel "foo" through the + CLI, the result would be a logger channel with the file name + /var/log/asterisk/foo being stored. So when trying to add another "foo" + channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily + add the duplicate channel. + + The fix presented here is to introduce two new methods in the logger + code: + * make_filename(): given a logger channel name, this creates the + filename for that logger channel. + * find_logchannel(): given a logger channel name, this calls + make_filename() and then traverses the list of logchannels in order + to find a match. + + This change has made use of make_filename() and find_logchannel() + throughout to more consistently behave. + + ASTERISK-25305 #close + Reported by Mark Michelson + + Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36 + +2015-09-24 14:49 +0000 [629458d349] Mark Michelson <mmichelson@digium.com> + + * Do not swallow frames on channels leaving bridges. + + When leaving a bridge, indications on a channel could be swallowed by + the internal indication logic because it appears that the channel is on + its way to be hung up anyway. One such situation where this is + detrimental is when channels on hold are redirected out of a bridge. The + AST_CONTROL_UNHOLD indication from the bridging code is swallowed, + leaving the channel in question to still appear to be on hold. + + The fix here is to modify the logic inside ast_indicate_data() to not + drop the indication if the channel is simply leaving a bridge. This way, + channels on hold redirected out of a bridge revert to their expected "in + use" state after the redirection. + + ASTERISK-25418 #close + Reported by Mark Michelson + + Change-Id: If6115204dfa0551c050974ee138fabd15f978949 + +2015-09-22 17:08 +0000 [5f15cd93f0] Richard Mudgett <rmudgett@digium.com> + + * app_page.c: Fix crash when forwarding with a predial handler. + + Page uses the async method of dialing with the dial API. When a call gets + forwarded there is no calling channel available. If the predial handler + was set then the calling channel could not be put into auto-service + for the forwarded call because it doesn't exist. A crash is the result. + + * Moved the callee predial parameter string processing to before the + string is passed to the dial API rather than having the dial API do it. + There are a few benefits do doing this. The first is the predial + parameter string processing doesn't need to be done for each channel + called by the dial API. The second is in async mode and the forwarded + channel is to have the predial handler executed on it then the + non-existent calling channel does not need to be present to process the + predial parameter string. + + * Don't start auto-service on a non-existent calling channel to execute + the predial handler when the dial API is in async mode and forwarding a + call. + + ASTERISK-25384 #close + Reported by: Chet Stevens + + Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981 + +2015-09-03 21:19 +0000 [b50e372394] Matt Jordan <mjordan@digium.com> + + * ARI: Add events for Contact and Peer Status changes + + This patch adds support for receiving events regarding Peer status changes + and Contact status changes. This is particularly useful in scenarios where + we are subscribed to all endpoints and channels, where we often want to know + more about the state of channel technology specific items than a single + endpoint's state. + + ASTERISK-24870 + + Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9 + +2015-09-04 12:24 +0000 [3502c0431d] Matt Jordan <mjordan@digium.com> + + * res/res_stasis_device_state: Allow for subscribing to 'all' device state + + This patch adds support for subscribing to all device state changes. This is + done either by subscribing to an empty device, e.g., 'eventSource=deviceState:', + or by the WebSocket connection specifying that it wants all state in the + system. + + ASTERISK-24870 + + Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32 + +2015-09-04 12:25 +0000 [4c9f613309] Matt Jordan <mjordan@digium.com> + + * ARI: Add the ability to subscribe to all events + + This patch adds the ability to subscribe to all events. There are two possible + ways to accomplish this: + (1) On initial WebSocket connection. This patch adds a new query parameter, + 'subscribeAll'. If present and True, Asterisk will subscribe the + applications to all ARI events. + (2) Via the applications resource. When subscribing in this manner, an ARI + client should merely specify a blank resource name, i.e., 'channels:' + instead of 'channels:12354'. This will subscribe the application to all + resources of the 'channels' type. + + ASTERISK-24870 #close + + Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6 + +2015-09-21 08:16 +0000 [ec514ad64d] Elazar Broad <elazar@thebroadfamily.com> + + * core/logging: Fix logging to more than one syslog channel + + Currently, Asterisk will log to the last configured syslog + channel in logger.conf. This is due to the fact that the + final call to openlog() supersedes all of the previous calls. + This commit removes the call to openlog() and passes the + facility to ast_log_vsyslog(), along with utilizing the + LOG_MAKEPRI macro to ensure that the message is routed to + the correct facility and with the correct priority. + + ASTERISK-25407 #close + Reported by: Elazar Broad + Tested by: Elazar Broad + + Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2 + +2015-09-21 18:06 +0000 [aeddee39fb] Kevin Harwell <kharwell@digium.com> + + * app_record: RECORDED_FILE variable not being populated + + The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. + This patch makes it so the variable is always set to the filename. + + ASTERISK-25410 #close + + Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653 + +2015-09-16 08:22 +0000 [2bd27d1222] Joshua Colp <jcolp@digium.com> + + * pbx: Update device and presence state when changing a hint extension. + + When changing a hint extension without removing the hint first the + device state and presence state is not updated. This causes the state + of the hint to be that of the previous extension and not the current + one. This state is kept until a state change occurs as a result of + something (presence state change, device state change). + + This change updates the hint with the current device and presence + state of the new extension when it is changed. Any state callbacks + which may have been added before the hint extension is changed are + also informed of the new device and presence state if either have + changed. + + ASTERISK-25394 #close + + Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f + +2015-09-17 16:34 +0000 [c94f46080f] Scott Griepentrog <scott@griepentrog.com> + + * CHAOS: avoid crash if string create fails + + Validate string buffer allocation before using them. + + ASTERISK-25323 + + Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0 + +2015-09-17 04:52 +0000 [b59c4d82b5] Walter Doekes <walter+asterisk@wjd.nu> + + * chan_sip: Fix From header truncation for extremely long CALLERID(name). + + The CALLERID(num) and CALLERID(name) and other info are placed into the + `char from[256]` in initreqprep. If the name was too long, the addr-spec + and params wouldn't fit. + + Code is moved around so the addr-spec with params is placed there first, + and then fitting in as much of the display-name as possible. + + ASTERISK-25396 #close + + Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260 + +2015-09-17 16:59 +0000 [4cc59533b9] Richard Mudgett <rmudgett@digium.com> + + * CHAOS: res_pjsip_diversion avoid crash if allocation fails + + Validate ast_malloc buffer returned before using it in + set_redirecting_value(). + + ASTERISK-25323 + + Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253 + +2015-09-17 16:47 +0000 [4fb95bbc4e] Kevin Harwell <kharwell@digium.com> + + * app_queue: AgentComplete event has wrong reason + + When a queued caller transfers an agent to another extension sometimes the + raised AgentComplete event has a reason of "caller" and sometimes "transfer". + Since a transfer has taken place this should always be transfer. This occurs + because sometimes the stasis hangup event arrives before the transfer event + thus writing a different reason out. + + With this patch, when a hangup event is received during a transfer it will + check to see if the channel that is hanging up is part of a transfer. If so + it will return and let the subsequently received transfer event handler take + care of the cleanup. + + ASTERISK-25399 #close + + Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d + +2015-09-17 13:09 +0000 [fb6b5c684b] Scott Griepentrog <scott@griepentrog.com> + + * PJSIP: avoid crash when getting rtp peer + + Although unlikely, if the tech private is returned as + a NULL, chan_pjsip_get_rtp_peer() would crash. + + ASTERISK-25323 + + Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a + +2015-09-17 11:31 +0000 [6409e7b11a] Kevin Harwell <kharwell@digium.com> + + * app_queue: Crash when transferring + + During some transfer scenarios involving queues Asterisk would sometimes + crash when trying to obtain a channel snapshot (could happen on caller or + member channels). This occurred because the underlying channel had already + disappeared when trying to obtain the latest snapshot. + + This patch adds a reference to both the member and caller channels that + extends to the lifetime of the queue'd call, thus making sure the channels + will always exist when retrieving the latest snapshots. + + ASTERISK-25185 #close + Reported by: Etienne Lessard + + Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128 + +2015-09-16 17:36 +0000 [fe5077b1f8] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Eliminate race during initial NOTIFY. + + There is a slim chance of a race condition occurring where two threads + can both attempt to manipulate the same area. + + Thread A can be handling an incoming initial SUBSCRIBE request. Thread A + lets the specific subscription handler know that the subscription has + been established. + + At this point, Thread B may detect a state change on the subscribed + resource and queue up a notification task on Thread C, the subscription + serializer thread. + + Now Thread A attempts to generate the initial NOTIFY request to send to + the subscriber at the same time that Thread C attempts to generate a + state change NOTIFY request to send to the subscriber. + + The result is that Threads A and C can step on the same memory area, + resulting in a crash. The crash has been observed as happening when + attempting to allocate more space to hold the body for the NOTIFY. + + The solution presented here is to queue the subscription establishment + and initial NOTIFY generation onto the subscription serializer thread + (Thread C in the above scenario). This way, there is no way that a state + change notification can occur before the initial NOTIFY is sent, and if + there is a quick succession of NOTIFYs, we can guarantee that the two + NOTIFY requests will be sent in succession. + + Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815 + +2015-08-28 15:42 +0000 [b88c54fa4b] Alexander Traud <pabstraud@compuserve.com> + + * translate: Fix transcoding while different in frame size. + + When Asterisk translates between codecs, each with a different frame size (for + example between iLBC 30 and Speex-WB), too large frames were created by + ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame + length, creating several frames when necessary. Affects all transcoding modules + which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex. + + ASTERISK-25353 #close + + Change-Id: I2e229569d73191d66a4e43fef35432db24000212 + +2015-09-10 17:19 +0000 [5c713fdf18] Mark Michelson <mmichelson@digium.com> + + * scheduler: Use queue for allocating sched IDs. + + It has been observed that on long-running busy systems, a scheduler + context can eventually hit INT_MAX for its assigned IDs and end up + overflowing into a very low negative number. When this occurs, this can + result in odd behaviors, because a negative return is interpreted by + callers as being a failure. However, the item actually was successfully + scheduled. The result may be that a freed item remains in the scheduler, + resulting in a crash at some point in the future. + + The scheduler can overflow because every time that an item is added to + the scheduler, a counter is bumped and that counter's current value is + assigned as the new item's ID. + + This patch introduces a new method for assigning scheduler IDs. Instead + of assigning from a counter, a queue of available IDs is maintained. + When assigning a new ID, an ID is pulled from the queue. When a + scheduler item is released, its ID is pushed back onto the queue. This + way, IDs may be reused when they become available, and the growth of ID + numbers is directly related to concurrent activity within a scheduler + context rather than the uptime of the system. + + Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2 + +2015-08-21 21:50 +0000 [865377fc38] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com> + + * chan_sip.c: Validation on module reload + + Change validation on reload module because now used the cli function for + reload. The sip_reload() function never fail and ever return NULL for this + reason on reload() now use the call the sip_reload() and return + AST_MODULE_LOAD_SUCCESS. + + This problem is dectected on reload by PUT method on ARI, getting always + 404 http code when the module is reloaded. + + ASTERISK-25325 #close + Reporte by: Rodrigo Ramírez Norambuena + + Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb + +2015-08-21 17:39 +0000 [e75aff53e6] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used. + + Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093 + +2015-09-09 12:24 +0000 [4d91d01df1] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Add some notification comments. + + Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20 + +2015-08-21 18:01 +0000 [f36a9d1221] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response. + + We should not try to send a SIP response message because we may be + restoring a persistent subscription where we are not responding to a SIP + request. + + Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec + +2015-08-21 17:40 +0000 [94582f8fab] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Fix off-nominal memory leak. + + Fix off-nominal visited vector leak in build_resource_tree(). + + Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c + +2015-08-21 15:26 +0000 [8b3ed52239] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Fix one byte buffer overrun error. + + ast_sip_pubsub_register_body_generator() did not account for the null + terminator set by sprintf() in the allocated output buffer. + + Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2 + +2015-08-21 15:25 +0000 [4329bd1e4c] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Use ast_alloca() instead of alloca(). + + Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3 + +2015-08-21 11:04 +0000 [a456a20ecf] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_pubsub.c: Add missing error return in load_module(). + + Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc + +2015-08-21 11:03 +0000 [f58f4c6e27] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip/location.c: Use the builtin ao2_callback() match function instead. + + Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f + +2015-09-10 09:49 +0000 [9d1f176e29] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Copy default_from_user to avoid crash. + + The default_from_user retrieval function was pulling the + default_from_user from the global configuration struct in an unsafe way. + If using a database as a backend configuration store, the global + configuration struct is short-lived, so grabbing a pointer from it + results in referencing freed memory. + + The fix here is to copy the default_from_user value out of the global + configuration struct. + + Thanks go to John Hardin for discovering this problem and proposing the + patch on which this fix is based. + + ASTERISK-25390 #close + Reported by Mark Michelson + + Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c + +2015-09-10 08:39 +0000 [1dd0e220bf] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route + + We will only rewrite the Contact header if there is no Record-Route header in + the received request. If a malfunctioning proxy places a Record-Route header + into a REGISTER request, we will decide that we shouldn't update the IP/port + in the Contact header, and we will end up storing a contact with an AoR that + contains the NAT'd IP address. + + While it is nice to have the proxy *not* send a Record-Route in a REGISTER + request, it's also a good idea to not process the header in a non-dialog + message. This patch updates the code to explicitly ignore the Record-Route + header in REGISTER requests. + + ASTERISK-25387 #close + + Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f + +2015-09-03 21:15 +0000 [4eedd9ef9d] Matt Jordan <mjordan@digium.com> + + * main/config_options: Check for existance of internal object before derefing + + Asterisk can load and register an object type while still having an invalid + sorcery mapping. This can cause an issue when a creation call is invoked. + For example, mis-configuring PJSIP's endpoint identifier by IP address mapping + in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a + subsequent ARI invocation to create the object will cause a crash, as the + internal type may not be registered as sorcery expects. + + Merely checking for a NULL pointer here solves the issue. + + Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac +2015-09-09 16:46 +0000 [71408df2b8] Alexander Anikin <may213@yandex.ru> + + * chan_ooh323: Add ProgressIndicator IE with inband info available + + Add ProgressIndicator IE with inband info present to Progress and + Alerting Q.931 message + + ASTERISK-25227 #close + Reported by: Alexandr Dranchuk + + Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203 +2015-09-08 10:35 +0000 [f72f9ceefc] Scott Griepentrog <scott@griepentrog.com> + + * pjsip: avoid possible crash req_caps allocation failure + + Make certain that the pjsip session has not failed to + allocate the format capabilities structure, which can + otherwise cause a crash when referenced. + + ASTERISK-25323 + + Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750 + +2015-09-03 14:07 +0000 [fbf720db91] Jonathan Rose <jrose@digium.com> + + * ParkAndAnnounce: Add variable inheritance + + In Asterisk 11, the announcer channel would receive channel variables + from the channel being parked by means of normal channel inheritance. + This functionality was lost during the big res_parking project in + Asterisk 12. This patch restores that functionality. + + ASTERISK-25369 #close + Review: https://gerrit.asterisk.org/#/c/1180/ + + Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e + +2015-09-04 16:33 +0000 [695f26cbb7] David M. Lee <dlee@respoke.io> + + * res_rtp_asterisk: Add more ICE debugging + + In working through a recent ICE negotiation bug, I found the debug + logging in res_rtp_asterisk to be lacking. This patch adds a number of + debug and warning statements that were helpful. + + Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80 +2015-09-01 10:16 +0000 [4ed9c9a280] Guido Falsi <madpilot@freebsd.org> + + * Core/General: Add #ifdef needed on FreeBSD. + + pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD + too. + + ASTERISK-25310 #close + Reported by: Guido Falsi + + Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4 + +2015-09-08 07:21 +0000 [5469caa9dd] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Use hash for contact object identity instead of Contact URI. + + In the wild it is possible for Contact URIs to be quite long as + parameters can exist on them. This can present a problem when storing + them in the AstDB as the URI is used as part of the object name and + there is a fixed length limit for the AstDB. This will cause + the contact to not get stored. + + This change uses the MD5 hash of the Contact URI as part of the + object name instead. This has a fixed length which is guaranteed + to not exceed the AstDB length limit. + + ASTERISK-25295 #close + + Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02 + +2015-09-07 13:19 +0000 [480c443e26] Alexander Anikin <may213@yandex.ru> + + * chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy + + Call ast_rtp_instance_stop on ooh323_destroy to free resources + allocated by rtp instance + + ASTERISK-25299 #close + Report by: Alexandr Dranchuk + + Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f + +2015-09-07 11:15 +0000 [c3e6debdb9] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip: Purge contacts when an AoR is deleted + + When an AoR is deleted by an external mechanism, such as through ARI, we + currently do not remove dynamic contacts that were created for that AoR as a + result of a received REGISTER request. As a result, re-creating the AoR will + cause the dynamic contact to be interpreted as a persistent contact, leading + to some rather strange state being created for the contacts/endpoints. + + This patch adds a sorcery observer for the 'aor' object. When a delete is + issued on the underlying sorcery object, the observer is called, and all + contacts created and persisted in sorcery for that AoR are also removed. Note + that we don't want to perform this action when an AO2 object that is an AoR is + destroyed, as the AoR can still exist in the backing storage (and we would + thus be removing valid contacts from an AoR that still "exists".) + + ASTERISK-25381 #close + + Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328 + +2015-09-05 14:58 +0000 [78d0b9d97e] Matt Jordan <mjordan@digium.com> + + * channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id + + This patch adds a new option to the CHANNEL function that allows for the + extraction of the SIP call-id. It is used in conjunction with the 'pjsip' + option, and will return the Call-ID of the INVITE request that established + the PJSIP channel. + + ASTERISK-25352 + + Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a + +2015-09-04 16:06 +0000 [61c6c6aa6c] David M. Lee <dlee@respoke.io> + + * Fix when remote candidates exceed PJ_ICE_MAX_CAND + + We were passing the wrong count into pj_ice_sess_create_check_list(), + causing the create to fail if we ever received more than PJ_ICE_MAX_CAND + candidates. + + Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378 + +2015-09-04 14:40 +0000 [ac62928d6b] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Change default from user value. + + When Asterisk sends an outbound SIP request, if there is no direct + reason to place a specific value for the username in the From header, + Asterisk would generate a UUID. For example, this would happen when + sending outbound OPTIONS requests when qualifying or when sending + outbound INVITE requests when originating (if no explicit caller ID were + provided). The issue is that some SIP providers reject these sorts of + requests with a "Name too long" error response. + + This patch aims to fix this by changing the default outbound username in + From headers to "asterisk". This value can be overridden by changing the + default_from_user option in the global options if desired. + + ASTERISK-25377 #close + Reported by Mark Michelson + + Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190 + +2015-09-04 09:26 +0000 [6002472a62] Scott Griepentrog <scott@griepentrog.com> + + * endpoint snapshot: avoid second cleanup on alloc failure + + In ast_endpoint_snapshot_create(), a failure to init the + string fields results in two attempts to ao2_cleanup the + same pointer. Removed RAII_VAR to eliminate problem. + + ASTERISK-25375 #close + Reported by: Scott Griepentrog + + Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979 + +2015-09-04 05:33 +0000 [d32e516c7c] Martin Tomec <tomec.martin@gmail.com> + + * res/pjsip: Mark WSS transport as secure + + Pjsip is refusing to use unsecure transport with "sips" in url. + WSS should be considered as secure transport. + + ASTERISK-24602 #comment Partially fixed by setting WSS as secure + + Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353 + +2015-09-02 17:26 +0000 [ad9cb6c2ce] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Fix contact refleak on stateful responses. + + When sending a stateful response, creation of the transaction can fail, + most commonly because we are trying to create a transaction from a + retransmitted request. When creation of the transaction fails, we end up + leaking a reference to a contact that was bumped when the response was + created. + + This patch adds the missing deref and fixes the reference leak. + + Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07 + +2015-09-02 12:41 +0000 [cc1363209e] Joshua Colp <jcolp@digium.com> + + * pbx: Fix crash when issuing "core show hints" with long pattern match. + + When issuing the "core show hints" CLI command a combination of both + the hint extension and context is created. This uses a fixed size + buffer expecting that the extension will not exceed maximum extension + length. When the extension is actually a pattern match this constraint + does not hold true, and the extension may exceed the maximum extension + length. In this case extra characters are written past the end of the + fixed size buffer. + + This change makes it so the construction of the combined hint extension + and context can not exceed the size of the buffer. + + ASTERISK-25367 #close + + Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499 + +2015-09-01 09:05 +0000 [d58c8d73af] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: re-re-fix persistent subscription storage. + + A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as + a means of writing an appropriate packet to persistent storage. While + this partially solved the issue, it had its own problems. + pjsip_msg_print will always add a Content-Length header to the message + it prints. Frequent restarts of Asterisk can result in persistent + subscriptions being written with five or more Content-Length headers. In + addition, sometimes some apparent corruption of individual headers could + be seen. + + This aims to fix the problem by not running a parsed message through an + interpreter but rather by taking the raw message and saving it. The + logic for what to save is going to be different depending on whether a + SUBSCRIBE was received from the wire or if it was pulled from + persistence. When receiving a packet from the wire, when using a + streaming transport, the rdata->pkt_info.packet may contain multiple SIP + messages or fragments. However, the rdata->msg_info.msg_buf will always + contain the current SIP message to be processed. When pulling from + persistence, though, the rdata->msg_info.msg_buf will be NULL since no + transport actually handled the packet. However, since we know that we + will always ever pull one SIP message from persistence, we are free to + save directly from rdata->pkt_info.packet instead. + + ASTERISK-25365 #close + Reported by Mark Michelson + + Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b + +2015-08-31 15:24 +0000 [03fe79f29e] Mark Michelson <mmichelson@digium.com> + + * Fix deadlock on presence state changes. + + A deadlock was observed where three threads were competing for different + locks: + + * One thread held the hints lock and was attempting to lock a specific + hint. + * One thread was holding the specific hint's lock and was attempting to + lock the contexts lock + * One thread was holding the contexts lock and attempting to lock the + hints lock. + + Clearly the second thread was doing the wrong thing here. The fix for + this is to make sure that the hint's lock is not held on presence state + changes. Something similar is already done (and commented about) for + device state changes. + + ASTERISK-25362 #close + Reported by Mark Michelson + + Change-Id: I15ec2416b92978a4c0c08273b2d46cb21aff97e2 + +2015-08-29 10:36 +0000 [a676ba2aad] Joshua Colp <jcolp@digium.com> + + * taskprocessor: Fix race condition between unreferencing and finding. + + When unreferencing a taskprocessor its reference count is checked + to determine if it should be unlinked from the taskprocessors + container and its listener shut down. In between the time when the + reference count is checked and unlinking it is possible for + another thread to jump in, find it, and get a reference to it. If + the thread then uses the taskprocessor it may find that it is not + in the state it expects. + + This change locks the taskprocessors container during almost the + entire unreference operation to ensure that any other thread which + may attempt to find the taskprocessor has to wait. + + ASTERISK-25295 + + Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c + +2015-08-28 20:22 +0000 [1b1561f4c8] Joshua Colp <jcolp@digium.com> + + * res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items. + + The keepalive support in res_pjsip_sdp_rtp currently assumes + that a stream will only be negotiated once. This is false. + If the stream is replaced and later added back it can be + negotiated again causing multiple keepalive scheduled items + to exist. This change explicitly deletes the existing + keepalive scheduled item before adding the new one. + + The res_pjsip_sdp_rtp module also does not stop RTP + keepalives or timeout timer if the stream has been + replaced. This change adds a callback to the session media + interface to allow a media stream to be stopped without + the resources being destroyed. This allows the scheduled + items and RTP to be stopped when the stream no longer + exists. + + ASTERISK-25356 #close + + Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de + +2015-08-28 19:57 +0000 [85e1cb51b2] Joshua Colp <jcolp@digium.com> + + * sched: ast_sched_del may return prematurely due to spurious wakeup + + When deleting a scheduled item if the item in question is currently + executing the ast_sched_del function waits until it has completed. + This is accomplished using ast_cond_wait. Unfortunately the + ast_cond_wait function can suffer from spurious wakeups so the + predicate needs to be checked after it returns to make sure it has + really woken up as a result of being signaled. + + This change adds a loop around the ast_cond_wait to make sure that + it only exits when the executing task has really completed. + + ASTERISK-25355 #close + + Change-Id: I51198270eb0b637c956c61aa409f46283432be61 + +2015-08-27 12:26 +0000 [c2c7319082] Joshua Colp <jcolp@digium.com> + + * res_pjsip_session: Don't invoke session supplements twice for BYE requests. + + When a BYE request is received the PJSIP invite session implementation + creates and sends a 200 OK response before we are aware of it. This + causes the INVITE session state callback to be called into and ultimately + the session supplements run on the BYE request. Once this response has + been sent the normal transaction state callback is invoked which + invokes the session supplements on the BYE request again. This can + be problematic in particular with res_pjsip_rfc3326 as it may + attempt to update the hangup cause code on the channel while it is + in the process of being hung up. + + This change makes it so the session supplements are only invoked + once by the INVITE session state callback. + + ASTERISK-25318 #close + + Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a + +2015-08-26 15:26 +0000 [6862c2a167] Scott Griepentrog <scott@griepentrog.com> + + * Chaos: handle failed allocation in get_media_encryption_type + + If the ast_strndup() call fails to allocate a copy of the + transport string for parsing, fail gracefully. + + ASTERISK-25323 + Reported by: Scott Griepentrog + + Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28 + +2015-08-26 14:25 +0000 [f1cd636658] Scott Griepentrog <scott@griepentrog.com> + + * Chaos: make hangup NULL tolerant + + In chan_pjsip_new, if allocation of the pvt + structure fails, ast_hangup is called. But + it was written to assume pvt was valid, and + this change corrects that. + + ASTERISK-25323 + Reported by: Scott Griepentrog + + Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87 + +2015-08-26 05:40 +0000 [c01111223f] Joshua Colp <jcolp@digium.com> + + * chan_sip: Allow call pickup to set the hangup cause. + + The call pickup implementation in chan_sip currently sets the channel + hangup cause to "normal clearing" if call pickup is successfully + performed. This action overwrites the "answered elsewhere" hangup cause + set by the call pickup code and can result in the SIP device in + question showing a missed call when it should not. + + This change sets the hangup cause to "normal clearing" as a + default initially but allows the call pickup to change it as + needed. + + ASTERISK-25346 #close + + Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff + +2015-08-25 07:17 +0000 [2a4eee0cd9] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Add common ast_sip_get_host_ip API. + + Modules commonly used the pj_gethostip function for retrieving the + IP address of the host. This function does not cache the result and may + result in a DNS lookup occurring, or additional work. If the DNS + server is unreachable or network issues arise this can cause the + pj_gethostip function to block for a period of time. + + This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string + function which does the same thing but caches the host IP address at + module load time. This results in no additional work being done each + time the local host IP address is needed. + + ASTERISK-25342 #close + + Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e + +2015-08-24 11:04 +0000 [7c4d0c3506] Joshua Colp <jcolp@digium.com> + + * res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced + + When recreating a subscription it is possible for a freed sub_tree + to be referenced when the initial NOTIFY fails to be created. + + Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788 + +2015-08-24 06:21 +0000 [6c2dab1e88] Joshua Colp <jcolp@digium.com> + + * bridge: Kick channel from bridge if hung up during action. + + When executing an action in a bridge it is possible for the + channel to be hung up without the bridge becoming aware of it. + This is most easily reproducible by hanging up when the bridge + is streaming DTMF due to a feature timeout. This change makes + it so after action execution the channel is checked to determine + if it has been hung up and if it has it is kicked from the bridge. + + ASTERISK-25341 #close + + Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062 + +2015-08-23 18:26 +0000 [bc6fe07f5c] Matt Jordan <mjordan@digium.com> + + * res_pjsip/pjsip_configuration: Disregard empty auth values + + When an endpoint is backed by a non-static conf file backend (such as + the AstDB or Realtime), the 'auth' object may be returned as being an + empty string. Currently, res_pjsip will interpret that as being a valid + auth object, and will attempt to authenticate inbound requests. This + isn't desired; is an auth value is empty (which the name of an auth + object cannot be), we should instead interpret that as being an invalid + auth object and skip it. + + ASTERISK-25339 #close + + Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7 + +2015-08-19 12:10 +0000 [0582776f7f] Richard Mudgett <rmudgett@digium.com> + + * ari/ari_websockets.c: Fix ast_debug parameter type mismatch. + + This is a type mismatch fix of the debugging commit + c63316eec10e1990a88bf4712238d6deb375bfa9 made to find out why + a testsuite test was failing only on one of the continuous + integration build agents. + + Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75 + +2015-08-19 10:30 +0000 [504213f542] Scott Griepentrog <scott@griepentrog.com> + + * contrib: script install_prereq should install sqlite3 + + Asterisk needs the sqlite 3 library, which is package + sqlite-devel in CentOS. By adding this package to the + script, a problem with configure failing is resolved. + + ASTERISK-25331 #close + Reported by: Kevin Harwell + + Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec + +2015-08-18 16:06 +0000 [77518d5434] Richard Mudgett <rmudgett@digium.com> + + * res_http_websocket.c: Fix some off nominal path cleanup. + + * Remove extraneous unlock on off-nominal path. + * Add missing HTTP error reply. + + Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b + +2015-08-18 14:46 +0000 [c61547fee6] Richard Mudgett <rmudgett@digium.com> + + * res_ari.c: Add missing off nominal unlock and remove a RAII_VAR(). + + Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf + +2015-08-17 16:41 +0000 [bd867cd078] Richard Mudgett <rmudgett@digium.com> + + * app_queue.c: Extract some functions for simpler code. + + * Extract set_queue_member_pause() from set_member_paused() for simpler + and more consistent code. + + * Extract set_queue_member_ringinuse() from + set_member_ringinuse_help_members() for simpler code. + + Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306 + +2015-08-14 12:55 +0000 [e5f5b9f384] Richard Mudgett <rmudgett@digium.com> + + * app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'. + + Setting the 'paused' and 'ringinuse' options on a queue member using the + dialplan function QUEUE_MEMBER did not behave the same way as the + equivalent dialplan applications or AMI actions. + + * Made queue_function_mem_write() call the set_member_paused() and + set_member_value() for the 'paused' and 'ringinuse' options respectively. + A beneficial side effect is that the queue name is now optional and sets + the value in all queues the interface is a member. + + * Update QUEUE_MEMBER XML documentation. + + * Fix error checking in QUEUE_MEMBER() write. + + ASTERISK-25215 #close + Reported by: Lorne Gaetz + + Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb + +2015-08-17 13:34 +0000 [ded51e3d77] Richard Mudgett <rmudgett@digium.com> + + * app_queue.c: Fix error checking in QUEUE_MEMBER() read. + + Change-Id: I7294e13d27875851c2f4ef6818adba507509d224 + +2015-08-17 11:00 +0000 [ab373f2cef] Scott Griepentrog <scott@griepentrog.com> + + * CHAOS: prevent sorcery object with null id + + When allocating a sorcery object, fail if the + id value was not allocated. + + ASTERISK-25323 + Reported by: Scott Griepentrog + + Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e + +2015-08-14 15:46 +0000 [b719f56c72] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_sdp_rtp: Restore removed NULL check. + + When sending an RTP keepalive, we need to be sure we're not dealing with + a NULL RTP instance. There had been a NULL check, but the commit that + added the rtp_timeout and rtp_hold_timeout options removed the NULL + check. + + Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64 + +2015-08-13 12:30 +0000 [cea5dc7b8a] Richard Mudgett <rmudgett@digium.com> + + * audiohook.c: Simplify variable usage in audiohook_read_frame_both(). + + Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c + +2015-08-13 12:22 +0000 [b3a56bee83] Richard Mudgett <rmudgett@digium.com> + + * audiohook.c: Fix MixMonitor crash when using the r() or t() options. + + The built frame format in audiohook_read_frame_both() is now set to a + signed linear format before the rx and tx frames are duplicated instead of + only for the mixed audio frame duplication. + + ASTERISK-25322 #close + Reported by Sean Pimental + + Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538 + +2015-08-12 12:59 +0000 [25af2d71c8] Kevin Harwell <kharwell@digium.com> + + * chan_sip.c: wrong peer searched in sip_report_security_event + + In chan_sip, after handling an incoming invite a security event is raised + describing authorization (success, failure, etc...). However, it was doing + a lookup of the peer by extension. This is fine for register messages, but + in the case of an invite it may search and find the wrong peer, or a non + existent one (for instance, in the case of call pickup). Also, if the peers + are configured through realtime this may cause an unnecessary database lookup + when caching is enabled. + + This patch makes it so that sip_report_security_event searches by IP address + when looking for a peer instead of by extension after an invite is processed. + + ASTERISK-25320 #close + + Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4 +2015-08-13 05:26 +0000 [e18c300550] Joshua Colp <jcolp@digium.com> + + * res_http_websocket: When shutting down a session don't close closed socket + + Due to the use of ast_websocket_close in session termination it is + possible for the underlying socket to already be closed when the + session is terminated. This occurs when the close frame is attempted + to be written out but fails. + + Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b +2015-08-11 05:24 +0000 [b4e9416138] Joshua Colp <jcolp@digium.com> + + * res_http_websocket: Forcefully terminate on write errors. + + The res_http_websocket module will currently attempt to close + the WebSocket connection if fatal cases occur, such as when + attempting to write out data and being unable to. When the + fatal cases occur the code attempts to write a WebSocket close + frame out to have the remote side close the connection. If + writing this fails then the connection is not terminated. + + This change forcefully terminates the connection if the + WebSocket is to be closed but is unable to send the close frame. + + ASTERISK-25312 #close + + Change-Id: I10973086671cc192a76424060d9ec8e688602845 + +2015-08-10 13:43 +0000 [256bc52b66] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. + + Pressing DTMF digits on a phone to go out on a DAHDI channel can result in + the digit not being recognized or even heard by the peer. + + Phone -> Asterisk -> DAHDI/channel + + Turns out the DAHDI behavior with DTMF generation (and any other generated + tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When + Asterisk requests to start sending DTMF then DAHDI waits until its write + buffer is empty before generating any samples for the DTMF tones. When + Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI + immediately stops generating the DTMF samples. As a result, the more + samples there are in the DAHDI write buffer the shorter the time DTMF + actually gets sent on the wire. If there are more samples in the write + buffer than the time DTMF is supposed to be sent then no DTMF gets sent on + the wire. With the "buffers=12,half" setting and each buffer representing + 20 ms of samples then the DAHDI write buffer is going to contain around + 120 ms of samples. For DTMF to be recognized by the peer the actual sent + DTMF duration needs to be a minimum of 40 ms. Therefore, the intended + duration needs to be a minimum of 160 ms for the peer to receive the + minimum DTMF digit duration to recognize it. + + A simple and effective solution to work around the DAHDI behavior is for + Asterisk to flush the DAHDI write buffer when sending DTMF so the full + duration of DTMF is actually sent on the wire. When someone is going to + send DTMF they are not likely to be talking before sending the tones so + the flushed write samples are expected to just contain silence. + + * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting + to send a DTMF digit. + + ASTERISK-25315 #close + Reported by John Hardin + + Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a + +2015-08-05 14:21 +0000 [800e0ea48d] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi.c: Lock private struct for ast_write(). + + There is a window of opportunity for DTMF to not go out if an audio frame + is in the process of being written to DAHDI while another thread starts + sending DTMF. The thread sending the audio frame could be past the + currently dialing check before being preempted by another thread starting + a DTMF generation request. When the thread sending the audio frame + resumes it will then cause DAHDI to stop the DTMF tone generation. The + result is no DTMF goes out. + + * Made dahdi_write() lock the private struct before writing to the DAHDI + file descriptor. + + ASTERISK-25315 + Reported by John Hardin + + Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb + +2015-08-10 18:23 +0000 [c126afe18f] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. + + If the saved SUBSCRIBE message is not parseable for whatever reason then + Asterisk could crash when libpjsip tries to parse the message and adds an + error message to the parse error list. + + * Made ast_sip_create_rdata() initialize the parse error rdata list. The + list is checked after parsing to see that it remains empty for the + function to return successful. + + ASTERISK-25306 + Reported by Mark Michelson + + Change-Id: Ie0677f69f707503b1a37df18723bd59418085256 + +2015-08-10 07:40 +0000 [f68c995bc9] Alexander Traud <pabstraud@compuserve.com> + + * chan_sip: Fix negotiation of iLBC 30. + + iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is + supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, + only iLBC 30 is negotiated now. + + ASTERISK-25309 #close + + Change-Id: I92d724600a183eec3114da0ac607b994b1a793da + +2015-08-09 18:42 +0000 [8e194047ac] Matt Jordan <mjordan@digium.com> + + * res/res_format_attr_silk: Expose format attributes to other modules + + This patch adds the .get callback to the format attribute module, such + that the Asterisk core or other third party modules can query for the + negotiated format attributes. + + Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c + +2015-08-09 17:56 +0000 [a0f451c35e] Matt Jordan <mjordan@digium.com> + + * main/format: Add an API call for retrieving format attributes + + Some codecs that may be a third party library to Asterisk need to have + knowledge of the format attributes that were negotiated. Unfortunately, + when the great format migration of Asterisk 13 occurred, that ability + was lost. + + This patch adds an API call, ast_format_attribute_get, to the core + format API, along with updates to the unit test to check the new API + call. A new callback is also now available for format attribute modules, + such that they can provide the format attribute values they manage. + + Note that the API returns a void *. This is done as the format attribute + modules themselves may store format attributes in any particular manner + they like. Care should be taken by consumers of the API to check the + return value before casting and dereferencing. Consumers will obviously + need to have a priori knowledge of the type of the format attribute as + well. + + Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3 + +2015-08-07 22:11 +0000 [26f0559a94] David M. Lee <dlee@respoke.io> + + * Replace htobe64 with htonll + + We don't have a compatability function to fill in a missing htobe64; but + we already have one for the identical htonll. + + Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac + +2015-08-07 14:20 +0000 [df9ce36366] Scott Emidy <jemidy@digium.com> + + * ARI: Retrieve existing log channels + + An http request can be sent to get the existing Asterisk logs. + + The command "curl -v -u user:pass -X GET 'http://localhost:8088 + /ari/asterisk/logging'" can be run in the terminal to access the + newly implemented functionality. + + * Retrieve all existing log channels + + ASTERISK-25252 + + Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808 + +2015-08-07 11:14 +0000 [e9f1bc08cb] Scott Emidy <jemidy@digium.com> + + * ARI: Creating log channels + + An http request can be sent to create a log channel + in Asterisk. + + The command "curl -v -u user:pass -X POST + 'http://localhost:088/ari/asterisk/logging/mylog? + configuration=notice,warning'" can be run in the terminal + to access the newly implemented functionality for ARI. + + * Ability to create log channels using ARI + + ASTERISK-25252 + + Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782 + +2015-08-06 15:18 +0000 [78364132ce] Scott Emidy <jemidy@digium.com> + + * ARI: Deleting log channels + + An http request can be sent to delete a log channel + in Asterisk. + + The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 + /ari/asterisk/logging/mylog'" can be run in the terminal + to access the newly implemented functionally for ARI. + + * Able to delete log channels using ARI + + ASTERISK-25252 + + Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6 + +2015-08-06 12:48 +0000 [e25569ef95] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: More accurately persist packet. + + The pjsip_rx_data structure has a pkt_info.packet field on it that is + the packet that was read from the transport. For datagram transports, + the packet read from the transport will correspond to the SIP message + that arrived. For streamed transports, however, it is possible to read + multiple SIP messages in one packet. + + In a recent case, Asterisk crashed on a system where TCP was being used. + This is because at some point, a read from the TCP socket resulted in a + 200 OK response as well as an incoming SUBSCRIBE request being stored in + rdata->pkt_info.packet. When the SUBSCRIBE was processed, the + combination 200 OK and SUBSCRIBE was saved in persistent storage. Later, + a restart of Asterisk resulted in the crash because the persistent + subscription recreation code ended up building the 200 OK response + instead of a SUBSCRIBE request, and we attempted to access + request-specific data. + + The fix here is to use the pjsip_msg_print() function in order to + persist SUBSCRIBE requests. This way, rather than using the raw socket + data, we use the parsed SIP message that PJSIP has given us. If we + receive multiple SIP messages from a single read, we will be sure only + to save off the relevant SIP message. There also is a safeguard put in + place to make sure that if we do end up reconstructing a SIP response, + it will not cause a crash. + + ASTERISK-25306 #close + Reported by Mark Michelson + + Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2 + +2015-08-04 16:12 +0000 [8521a86367] Joshua Colp <jcolp@digium.com> + + * res_pjsip: Ensure sanitized XML is NULL terminated. + + The ast_sip_sanitize_xml function is used to sanitize + a string for placement into XML. This is done by examining + an input string and then appending values to an output + buffer. The function used by its implementation, strncat, + has specific behavior that was not taken into account. + If the size of the input string exceeded the available + output buffer size it was possible for the sanitization + function to write past the output buffer itself causing + a crash. The crash would either occur because it was + writing into memory it shouldn't be or because the resulting + string was not NULL terminated. + + This change keeps count of how much remaining space is + available in the output buffer for text and only allows + strncat to use that amount. + + Since this was exposed by the res_pjsip_pidf_digium_body_supplement + module attempting to send a large message the maximum allowed + message size has also been increased in it. + + A unit test has also been added which confirms that the + ast_sip_sanitize_xml function is providing NULL terminated + output even when the input length exceeds the output + buffer size. + + ASTERISK-25304 #close + + Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302 + +2015-08-05 05:23 +0000 [9a12804e59] Joshua Colp <jcolp@digium.com> + + * res_rtp_asterisk: Don't leak temporary key when enabling PFS. + + A change recently went in which enabled perfect forward secrecy for + DTLS in res_rtp_asterisk. This was accomplished two different ways + depending on the availability of a feature in OpenSSL. The fallback + method created a temporary instance of a key but did not free it. + This change fixes that. + + ASTERISK-25265 + + Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396 +2015-08-04 09:47 +0000 [27dc2094e9] Mark Michelson <mmichelson@digium.com> + + * res_http_websocket: Debug write lengths. + + Commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee attempted to fix a + test failure observed on 32 bit test agents by ensuring that a cast from + a 32 bit unsigned integer to a 64 bit unsigned integer was happening in + a predictable place. As it turns out, this did not cause test runs to + succeed. + + This commit adds several redundant debug messages that print the payload + lengths of websocket frames. The idea here is that this commit will not + cause tests to succeed for the faulty test agent, but we might deduce + where the fault lies more easily this way by observing at what point the + expected value (537) changes to some ungangly huge number. + + If you are wondering why something like this is being committed to the + branch, keep in mind that in commit + 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee I noted that the observed test + failures only happen when automated tests are run. Attempts to run the + tests by hand manually on the test agent result in the tests passing. + + Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d + +2015-08-03 11:06 +0000 [39cc28f6ea] Mark Michelson <mmichelson@digium.com> + + * res_http_websocket: Avoid passing strlen() to ast_websocket_write(). + + We have seen a rash of test failures on a 32-bit build agent. Commit + 48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where + we were not encoding a 64-bit value correctly over the wire. This + commit, however, did not solve the test failures. + + In the failing tests, ARI is attempting to send a 537 byte text frame + over a websocket. When sending a frame this small, 16 bits are all that + is required in order to encode the payload length on the websocket + frame. However, ast_websocket_write() thinks that the payload length is + greater than 65535 and therefore writes out a 64 bit payload length. + Inspecting this payload length, the lower 32 bits are exactly what we + would expect it to be, 537 in hex. The upper 32 bits, are junk values + that are not expected to be there. + + In the failure, we are passing the result of strlen() to a function that + expects a uint64_t parameter to be passed in. strlen() returns a size_t, + which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit + unsigned value to somewhere where a 64-bit unsigned value is expected + would cause no problems. In fact, in manual runs of failing tests, this + works just fine. However, ast_websocket_write() uses the Asterisk + optional API, which means that rather than a simple function call, there + are a series of macros that are used for its declaration and + implementation. These macros may be causing some sort of error to occur + when converting from a 32 bit quantity to a 64 bit quantity. + + This commit changes the logic by making existing ast_websocket_write() + calls use ast_websocket_write_string() instead. Within + ast_websocket_write_string(), the 64-bit converted strlen is saved in a + local variable, and that variable is passed to ast_websocket_write() + instead. + + Note that this commit message is full of speculation rather than + certainty. This is because the observed test failures, while always + present in automated test runs, never occur when tests are manually + attempted on the same test agent. The idea behind this commit is to fix + a theoretical issue by performing changes that should, at the least, + cause no harm. If it turns out that this change does not fix the failing + tests, then this commit should be reverted. + + Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67 + +2015-07-28 05:33 +0000 [aed068844c] Mark Duncan <mark@syon.co.jp> + + * res/res_rtp_asterisk: Add ECDH support + + This will add ECDH support to Asterisk. It will + detect auto ECDH support in OpenSSL + (1.0.2b and above) during ./configure. If this is + available, it will use it, + otherwise it will fall back to prime256v1 (this + behavior is consistent with + other projects such as Apache and nginx). + + This fixes WebRTC being broken in Firefox 38+ due + to Firefox now only supporting + ciphers with perfect forward secrecy. + + ASTERISK-25265 #close + + Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b + +2015-07-29 14:17 +0000 [1ae762634c] Benjamin Ford <bford@digium.com> + + * ARI: Rotate log channels. + + An http request can be sent to rotate a specified log channel. + If the channel does not exist, an error response will be + returned. + + The command "curl -v -u user:pass -X PUT 'http://localhost:8088 + /ari/asterisk/logging/logChannelName/rotate'" can be run in the + terminal to access this new functionality. + + * Added the ability to rotate log files through ARI + + ASTERISK-25252 + + Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01 + +2015-07-29 13:49 +0000 [aeeb170fc4] Richard Mudgett <rmudgett@digium.com> + + * rtp_engine.c: Fix performance issue with several channel drivers that use RTP. + + ast_rtp_codecs_get_payload() gets called once or twice for every received + RTP frame so it would be nice to not allocate an ao2 object to then have + it destroyed shortly thereafter. The ao2 object gets allocated only if + the payload type is not set by the channel driver as a negotiated value. + The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. + + * Made static_RTP_PT[] an array of ao2 objects that + ast_rtp_codecs_get_payload() can return instead of an array of structs + that must be copied into a created ao2 object. + + ASTERISK-25296 #close + Reported by: Richard Mudgett + + Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0 + +2015-07-29 17:00 +0000 [84262749d2] Richard Mudgett <rmudgett@digium.com> + + * res_rtp_asterisk.c: Fix off-nominal crash potential. + + ASTERISK-25296 + Reported by: Richard Mudgett + + Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b + +2015-07-29 13:48 +0000 [1519eb44a7] Richard Mudgett <rmudgett@digium.com> + + * rtp_engine.c: Must protect mime_types_len with mime_types_lock. + + Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e + +2015-07-24 18:42 +0000 [a93b7a927c] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list. + + Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2 + +2015-07-24 18:38 +0000 [741fa0d26d] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_sdp_rtp.c: Fixup some whitespace. + + Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973 + +2015-07-27 19:10 +0000 [89b21fd9a3] Richard Mudgett <rmudgett@digium.com> + + * rtp_engine.h: No sense allowing payload types larger than RFC allows. + + * Tweaked add_static_payload() to not use magic numbers. + + Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b + +2015-07-23 14:04 +0000 [7427c7f13b] Richard Mudgett <rmudgett@digium.com> + + * rtp_engine.c: Minor tweaks. + + * Fix off nominial ref leak of new_type in + ast_rtp_codecs_payloads_set_m_type(). + + * No need to lock static_RTP_PT_lock in + ast_rtp_codecs_payloads_set_m_type() and + ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type + parameter sanity check. + + * No need to create ast_rtp_payload_type ao2 objects with a lock since the + lock is not used. + + Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4 + +2015-07-23 12:41 +0000 [e20f435b60] Richard Mudgett <rmudgett@digium.com> + + * rtp_engine.h: Misc comment fixes. + + Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43 + +2015-07-17 16:23 +0000 [bc5d7f9c37] Richard Mudgett <rmudgett@digium.com> + + * chan_sip.c: Tweak glue->update_peer() parameter nil value. + + Change glue->update_peer() parameter from 0 to NULL to better indicate it + is a pointer. + + Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd + +2015-07-30 17:05 +0000 [13eb491e35] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Fix crashes seen when call cancelled. + + Two testsuite tests crashed in the same place as a result of an INVITE + being CANCELed. + + tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified + tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp + + The session pointer is no longer in the inv->mod_data[session_module.id] + location because the INVITE transaction has reached the terminated state. + + ASTERISK-25297 #close + Reported by: Richard Mudgett + + Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427 + +2015-07-29 14:35 +0000 [48698a5e21] Mark Michelson <mmichelson@digium.com> + + * res_http_websocket: Properly encode 64 bit payload + + A test agent was continuously failing all ARI tests when run against + Asterisk 13. As it turns out, the reason for this is that on those test + runs, for some reason we decided to use the super extended 64 bit + payload length for websocket text frames instead of the extended 16 bit + payload length. For 64-bit payloads, the expected byte order over the + network is + + 7, 6, 5, 4, 3, 2, 1, 0 + + However, we were sending the payload as + + 3, 2, 1, 0, 7, 6, 5, 4 + + This meant that we were saying to expect an absolutely MASSIVE payload + to arrive. Since we did not follow through on this expected payload + size, the client would sit patiently waiting for the rest of the payload + to arrive until the test would time out. + + With this change, we use the htobe64() function instead of htonl() so + that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. + + Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a + +2015-07-29 12:23 +0000 [10ba72a927] Mark Michelson <mmichelson@digium.com> + + * Add a test event for inband ringing. + + This event is necessary for the bridge_wait_e_options test to be able to + confirm that ringing is being played on the local channel that runs the + BridgeWait() application with the e(r) option. + + ASTERISK-25292 #close + Reported by Kevin Harwell + + Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e + +2015-07-16 12:16 +0000 [8458b8d441] Jonathan Rose <jrose@digium.com> + + * holding_bridge: ensure moh participants get frames + + Currently, if a blank musiconhold.conf is used, musiconhold will fail + to start for a channel going into a holding bridge with an anticipation + of getting music on hold. That being the case, no frames will be written + to the channel and that can pose a problem for blind transfers in PJSIP + which may rely on frames being written to get past the REFER framehook. + This patch makes holding bridges start a silence generator if starting + music on hold fails and makes it so that if no music on hold functions + are installed that the ast_moh_start function will report a failure so + that consumers of that function will be able to respond appropriately. + + ASTERISK-25271 #close + + Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99 + +2015-07-24 22:20 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.5.0-rc1 Released. + +2015-07-24 17:15 +0000 [a4b527393b] Matt Jordan <mjordan@digium.com> + + * Release summaries: Add summaries for 13.5.0-rc1 + +2015-07-24 17:11 +0000 [158b0b8ebf] Matt Jordan <mjordan@digium.com> + + * .version: Update for 13.5.0-rc1 + +2015-07-24 17:11 +0000 [a0a7650e34] Matt Jordan <mjordan@digium.com> + + * .lastclean: Update for 13.5.0-rc1 + +2015-07-24 17:11 +0000 [4d238af086] Matt Jordan <mjordan@digium.com> + + * realtime: Add database scripts for 13.5.0-rc1 + +2015-07-24 12:56 +0000 [f78a4b52b8] Matt Jordan <mjordan@digium.com> + + * Bump the ARI version to 1.8.0 + + Due to backwards compatible changes, the ARI version should be bumped to + 1.8.0 prior to the release of 13.5.0. Note that a previous patch already + bumped the version of AMI for this release. + + Change-Id: I419033bfbbc0d3533a29ccb32b2981f39e0883e7 + +2015-07-18 11:16 +0000 [2749721791] Joshua Colp <jcolp@digium.com> + + * pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. + + This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' + endpoint options. These allow the channel to be hung up if RTP + is not received from the remote endpoint for a specified number of + seconds. + + ASTERISK-25259 #close + + Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9 + +2015-07-24 09:46 +0000 [b4e19e414a] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Add rtp_keepalive to sample config file. + + Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19 + +2015-07-23 13:11 +0000 [f635520527] Mark Michelson <mmichelson@digium.com> + + * Local channels: Alternate solution to ringback problem. + + Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a + specific scenario involving local channels and a native local RTP bridge + could result in ringback still being heard on a calling channel even + after the call is bridged. + + That commit caused many tests in the testsuite to fail with alarming + consequences, such as not sending DialBegin and DialEnd events, and + giving incorrect hangup causes during calls. + + This commit reverts the previous commit and implements and alternate + solution. This new solution involves only passing AST_CONTROL_RINGING + frames across local channels if the local channel is in AST_STATE_RING. + Otherwise, the frame does not traverse the local channels. By doing + this, we can ensure that a playtones generator does not get started on + the calling channel but rather is started on the local channel on which + the ringing frame was initially indicated. + + ASTERISK-25250 #close + Reported by Etienne Lessard + + Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39 + +2015-07-22 12:24 +0000 [f509730cb9] Joshua Colp <jcolp@digium.com> + + * audiohook: Use manipulated frame instead of dropping it. + + Previous changes to sample rate support in audiohooks accidentally + removed code responsible for allowing the manipulate audiohooks + to work. Without this code the manipulated frame would be dropped + and not used. This change restores it. + + ASTERISK-25253 #close + + Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13 + +2015-07-22 09:46 +0000 [54b25c80c8] Mark Michelson <mmichelson@digium.com> + + * Local channels: Do not block control -1 payloads. + + Control frames with a -1 payload are used as a special signal to stop + playtones generators on channels. This indication is sent both by + app_dial as well as by ast_answer() when a call is answered in case any + tones were being generated on a calling channel. + + This control frame type was made to stop traversing local channel pairs + as an optimization, because it was thought that it was unnecessary to + send these indications, and allowing such unnecessary control frames to + traverse the local channels would cause the local channels to optimize + away less quickly. + + As it turns out, through some special magic dialplan code, it is + possible to have a tones being played on a non-local channel, and it is + important for the local channel to convey that the tones should be + stopped. The result of having tones continue to be played on the + non-local channel is that the tones play even once the channel has been + bridged. By not blocking the -1 control frame type, we can ensure that + this situation does not happen. + + ASTERISK-25250 #close + Reported by Etienne Lessard + + Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815 + +2015-07-22 05:16 +0000 [f1493f900e] Joshua Colp <jcolp@digium.com> + + * audiohook: Read the correct number of samples based on audiohook format. + + Due to changes in audiohooks to support different sample rates the + underlying storage of samples is in the format of the audiohook + itself and not of the format being requested. This means that if a + channel is using G722 the samples stored will be at 16kHz. If + something subsequently reads from the audiohook at a format which + is not the same sample rate as the audiohook the number of samples + needs to be adjusted. + + Given the following example: + 1. Channel writing into audiohook at 16kHz (as it is using G722). + 2. Chanspy reading from audiohook at 8kHz. + + The original code would read 160 samples from the audiohook for + each 20ms of audio. This is incorrect. Since the audio in the + audiohook is at 16kHz the actual number needing to be read is 320. + Failure to read this much would cause the audiohook to reset + itself constantly as the buffer became full. + + This change adjusts the requested number of samples by determining + the duration of audio requested and then calculating how many + samples that would be in the audiohook format. + + ASTERISK-25247 #close + + Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d + +2015-07-20 12:39 +0000 [62c64c3bd1] Rusty Newton <rnewton@digium.com> + + * Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c + + * In sip.conf.sample fix sentence where we said that WS or WSS are supported + transports for use in an outbound register definition. They are not + supported in that case. + * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used + to enable CDR on a channel. + + ASTERISK-24867 #close + Reported by: Rusty Newton + + ASTERISK-24853 #close + Reported by: PSDK + + Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca + +2015-07-09 14:17 +0000 [d9094ddd73] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Add rtp_keepalive endpoint option. + + This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the + chan_sip option, this specifies an interval, in seconds, at which we + will send RTP comfort noise frames. This can be useful for keeping RTP + sessions alive as well as keeping NAT associations alive during lulls. + + ASTERISK-25242 #close + Reported by Mark Michelson + + Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b + +2015-07-16 09:13 +0000 [a23adcca3d] Michael Cargile <mikec@vicidial.com> + + * res/res_musiconhold: Add a warning when MOH does not exist + + Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b + +2015-07-19 09:11 +0000 [03064daeb2] Matt Jordan <mjordan@digium.com> + + * res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf + + Misconfiguring sorcery.conf with a 'config' wizard with no extra data + will currently crash Asterisk on startup, as the wizard requires a comma + delineated list to parse. This patch updates res_sorcery_config to check + for the presence of the data before it starts manipulating it. + + Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847 + +2015-07-16 09:46 +0000 [2c626ceb64] Joshua Colp <jcolp@digium.com> + + * chan_pjsip: Don't change formats when frame of unsupported format is received. + + Receipt of an RTP packet currently causes the formats on an PJSIP channel to + change to the format of the RTP packet. In some off-nominal cases it's possible + for this to be a format that has not been configured or negotiated. This change + makes it so only formats explicitly configured on the endpoint are allowed. + + ASTERISK-25258 #close + + Change-Id: If93d641fb6418a285928839300d7854cab8c1020 + +2015-07-17 04:59 +0000 [abb14ac5b8] Patric Marschall <patric.marschall@1und1.de> + + * sig_pri.h: force_restart_unavailable_chans in wrong scope + + In channels/sig_pri.h, struct sig_pri_span, the field + force_restart_unavailable_chans is only defined if + + #if defined(HAVE_PRI_MCID) is true. + + All other occurences of force_restart_unavailable_chans are outside of the + + #if defined(HAVE_PRI_MCID) + endif + + scope. + + ASTERISK-25257 #close + Reported by: Patric Marschall + + Change-Id: I071de89cc2cd0d85927a013036e235851f672549 +2015-07-14 16:55 +0000 [875aee4c09] Richard Mudgett <rmudgett@digium.com> + + * pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable. + + ASTERISK-25256 #close + Reported by: Richard Mudgett + + Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3 + +2015-07-08 16:39 +0000 [8bcf6d2801] Matt Jordan <mjordan@digium.com> + + * ARI: Add support for push configuration of dynamic object + + This patch adds support for push configuration of dynamic, i.e., + sorcery, objects in Asterisk. It adds three new REST API calls to the + 'asterisk' resource: + * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current + object given its ID. This returns back a list of ConfigTuples, which + define the fields and their present values that make up the object. + * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an + object. A body may be passed with the request that contains fields to + populate in the object. The same format as what is retrieved using + the GET operation is used for the body, save that we specify that the + list of fields to update are contained in the "fields" attribute. + * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic + object from its backing storage. + + Note that the success/failure of these operations is somewhat + configuration dependent, i.e., you must be using a sorcery wizard that + supports the operation in question. If a sorcery wizard does not support + the create or delete mechanisms, then the REST API call will fail with a + 403 forbidden. + + ASTERISK-25238 #close + + Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c + +2015-07-15 15:40 +0000 [e31cb6b248] Richard Mudgett <rmudgett@digium.com> + + * strings.h: Fix issues with escape string functions. + + Fixes for issues with the ASTERISK-24934 patch. + + * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is + an empty string. If it were an empty string the functions returned NULL + as if there were a memory allocation failure. This failure caused the AMI + VarSet event to not get posted if the new value was an empty string. + + * Fixed dest buffer overwrite potential in ast_escape() and + ast_escape_c(). If the dest buffer size is smaller than the space needed + by the escaped s parameter string then the dest buffer would be written + beyond the end by the nul string terminator. The num parameter was really + the dest buffer size parameter so I renamed it to size. + + * Made nul terminate the dest buffer if the source string parameter s was + an empty string in ast_escape() and ast_escape_c(). + + * Updated ast_escape() and ast_escape_c() doxygen function description + comments to reflect reality. + + * Added some more unit test cases to /main/strings/escape to cover the + empty source string issues. + + ASTERISK-25255 #close + Reported by: Richard Mudgett + + Change-Id: Id77fc704600ebcce81615c1200296f74de254104 + +2015-07-14 14:29 +0000 [243c0d1609] Richard Mudgett <rmudgett@digium.com> + + * parking_applications.c: Fix ast_verb() line terminator. + + Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775 + +2015-07-14 14:36 +0000 [c782320c68] Richard Mudgett <rmudgett@digium.com> + + * res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park. + + setup_park_common_datastore() was assuming that a non-NULL string returned + for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty + strings. Things got crashy as a result. + + * Made setup_park_common_datastore() treat the channel variable values the + same whether they are NULL or empty for ATTENDEDTRANSFER and + BLINDTRANSFER. + + ASTERISK-25254 #close + Reported by: Richard Mudgett + + Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2 + +2015-07-10 18:01 +0000 [2735dd5b2d] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). + + Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb + +2015-07-10 10:42 +0000 [3d0ca343ca] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Add some helpful comments and minor tweaks. + + Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743 + +2015-07-10 10:43 +0000 [8d08bb179c] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Fix off nominal crash potential in debug message. + + Change-Id: I09928297927ee85f7655289acee3a586816466bc + +2015-07-15 10:31 +0000 [0a1a550593] Matt Jordan <mjordan@digium.com> + + * apps/app_dictate: Fix typo in attribution + + Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian + (GameGamer43) for pointing that out. + + Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106 + +2015-07-15 10:28 +0000 [3384e64ef6] Benjamin Ford <bford@digium.com> + + * ARI: Fixed unload mode for unload module. + + Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM, + which would unload a module even if it was in use. + + * Changed unload mode to proper mode + + ASTERISK-25173 + + Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533 + +2015-07-08 16:38 +0000 [0b6ff77afb] Matt Jordan <mjordan@digium.com> + + * res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails + + Having a debug message tell us that we attempted to look up an item but + failed is nice in circumstances when it isn't clear if the wizard was + queried correctly or not. + + Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7 + +2015-07-08 16:37 +0000 [2f0d6d346c] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_outbound_registration: Fix WARNING message + + Newlines are nice. + + Change-Id: Icf0d915db02882e47cd9077ed9009f5d44140d42 + +2015-07-08 16:35 +0000 [cd2213f1ae] Matt Jordan <mjordan@digium.com> + + * res_pjsip/configuration: Fix a variety of default value problems + + This patch fixes some bad default value handling in the following + settings: + + * The 'message_context' and 'accountcode' settings are not mandatory. As + such, we can allow their stringfield values to be empty. + * The 'media_encryption' setting applies a default value of 'none' to + the setting, which it then can't parse or understand. Since the value + is documented to be 'no', this will now apply that as the default + value. + + Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83 + +2015-07-08 16:32 +0000 [2e4bdbd78a] Matt Jordan <mjordan@digium.com> + + * main/sorcery: Provide log messages when a wizard does not support an operation + + If a sorcery wizard does not support one of the 'optional' CRUD + operations (namely the CUD), log a WARNING message so we are aware of + why the operation failed. This also removes an assert in this case, as + the CUD operation may have been triggered by an external system, in + which case it is not a programming error but a configuration error. + + Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53 + +2015-07-10 18:17 +0000 [653f2087e0] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session.c: Fix crash on call disconnect. + + The crash fix for ASTERISK-25183 backported some code from master to try + to make sure that a BYE response is processed by the same serializer used + by the BYE request. The identified race condition causing that backport + was the BYE request code had not finished processing after sending the BYE + before the BYE response came in for processing under a different thread. + Unfortunately, there is still a race condition. Now the race condition is + between destroying the call session's serializer in + ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a + reference to the serializer for a BYE response. Even worse, the new race + condition is a design limitation of the taskprocessor implementation that + didn't matter in versions before v12. Back then, taskprocessors were only + destroyed when a module unloaded. Now res_pjsip can destroy them when a + call ends. + + However, as noted on the ASTERISK-25183 commit, + session_inv_on_state_changed() is disassociating the dialog from the + session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED. + This is a tad too soon because our BYE request transaction has not + completed yet. + + * Split session_end() that is called by session_inv_on_state_changed() to + hold off session destruction until the BYE transaction timeout occurs or a + failed initial INVITE transaction timeout occurs in + session_inv_on_tsx_state_changed(). + + ASTERISK-25201 #close + Reported by: Matt Jordan + + Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961 + +2015-07-14 13:12 +0000 [1aafadf814] Benjamin Ford <bford@digium.com> + + * ARI: Added new functionality to reload a single module. + + An http request can be sent to reload an Asterisk module. If the + module can not be reloaded or is not already loaded, an error + response will be returned. + + The command "curl -v -u user:pass -X PUT 'http://localhost:8088 + /ari/asterisk/modules/{moduleName}'" (or something similar, based + on configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Asterisk modules can be reloaded through http requests + + ASTERISK-25173 + + Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1 + +2015-07-14 08:55 +0000 [9dcae23cfc] Benjamin Ford <bford@digium.com> + + * ARI: Added new functionality to unload a single module. + + An http request can be sent to unload an Asterisk module. If the + module can not be unloaded or is already unloaded, an error response + will be returned. + + The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 + /ari/asterisk/modules/{moduleName}'" (or something similar, depending + on configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Asterisk modules can be unloaded through http requests + + ASTERISK-25173 + + Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57 + +2015-07-13 16:00 +0000 [c219a98d2b] Benjamin Ford <bford@digium.com> + + * ARI: Added new functionality to load a single module. + + An http request can be sent to load an Asterisk module. If the + module can not be loaded or is loaded already, an error response + will be returned. + + The command curl -v -u user:pass -X POST 'http://localhost:8088/ari + /asterisk/modules/{moduleName}'" (or something similar, depending on + configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Asterisk modules can be loaded through http requests + + ASTERISK-25173 + + Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33 + +2015-07-13 10:54 +0000 [73e35d20de] Benjamin Ford <bford@digium.com> + + * ARI: Added new functionality to get information on a single module. + + An http request can be sent to retrieve information on a single + module, including the resource name, description, use count, status, + and support level. + + The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari + /asterisk/modules/{moduleName}'" (or something similar, depending on + configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Information on a single module can now be retrieved + + ASTERISK-25173 + + Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463 + +2015-07-08 14:56 +0000 [97ee0ee6c6] Kevin Harwell <kharwell@digium.com> + + * bridge.c: Fixed race condition during attended transfer + + During an attended transfer a thread is started that handles imparting the + bridge channel. From the start of the thread to when the bridge channel is + ready exists a gap that can potentially cause problems (for instance, the + channel being swapped is hung up before the replacement channel enters the + bridge thus stopping the transfer). This patch adds a condition that waits + for the impart thread to get to a point of acceptable readiness before + allowing the initiating thread to continue. + + ASTERISK-24782 + Reported by: John Bigelow + + Change-Id: I08fe33a2560da924e676df55b181e46fca604577 + +2015-07-08 16:28 +0000 [bb76b88baf] Matt Jordan <mjordan@digium.com> + + * main/sorcery: Don't fail object set creation from JSON if field fails + + Some individual fields may fail their conversion due to their default + values being invalid for their custom handlers. In particular, + configuration values that depend on others being enabled (and thus have + an empty default value) are notorious for tripping this routine up. An + example of this are any of the DTLS options for endpoints. Any of the + DTLS options will fail to be applied (as DTLS is not enabled), causing + the entire object set to be aborted. + + This patch makes it so that we log a debug message when skipping a + field, and rumble on anyway. + + ASTERISK-25238 + + Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76 + +2015-07-08 16:21 +0000 [5f13c2226a] Matt Jordan <mjordan@digium.com> + + * main/format_cap: Parse capabilities generated by ast_format_cap_get_names + + We have a strange relationship between the parsing of format + capabilities from a string and their representation as a string. We + expect the format capabilities to be expressed as a string in the + following format: + + allow = !all,ulaw,alaw + disallow = g722 + + While we would generate the string representation of those formats as: + + allow = (ulaw|alaw) + disallow = (ulaw|alaw|g729...) + + When the configuration framework needs to store values as a string, it + generates the format capabilities using the second representation; this + representation however cannot be parsed when the entry is rehydrated. + This patch fixes that by updating + ast_format_cap_update_by_allow_disallow to parse an entry as if it were + in the generated format if it has a leading '(' and a trailing ')'. + + ASTERISK-25238 + + Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca + +2015-06-27 17:53 +0000 [2325b106fd] Matt Jordan <mjordan@digium.com> + + * tests/test_devicestate: Add additional tests for the device state API + + This patch adds more tests that exercise the device state API. This includes: + + * Tests that cover adding a device state provider, as well as deleting a + device state provider. This also verifies that you cannot add an + already added device state provider, and cannot delete an already + deleted device state provider. + * A test that covers changing device state and receiving said updates + from a device state subscriber. This also covers hitting both the + device state cache as well as a custom device state provider. + * A test that covers converting device state to channel state and device + state values to a string representation and back. + * A test that covers obtaining device state from an active channel and a + channel driver that provides its own device state. + + Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d + +2015-06-27 17:51 +0000 [328f0be806] Matt Jordan <mjordan@digium.com> + + * main/devicestate: Prevent duplicate registration of device state providers + + Currently, the device state provider API will allow you to register a + device state provider with the same case insensitive name more than + once. This could cause strange issues, as the duplicate device state + providers will not be queried when a device's state has to be polled. + This patch updates the API such that a device state provider with the + same name as one that has already registered will be rejected. + + Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2 + +2015-07-10 22:25 +0000 [bee41eec62] Matt Jordan <mjordan@digium.com> + + * res/res_sorcery_memory_cache: Fix test registration issues + + Again, tests now need to not end with a newline. This patch makes it so + the tests can register again, unit tests will actually pass, and we can + stop wasting time trying to figure out why builds are failing when they + really aren't failing. + + Change-Id: Ide519fbeba89f413c733446c5ff7b224fc4ce840 + +2015-07-10 21:42 +0000 [4d738e9026] Matt Jordan <mjordan@digium.com> + + * tests/test_sorcery_memory_cache_thrash: Fix test loading problems + + Because unit tests now want descriptions to not end with a newline, the + sorcery memory cache thrash tests failed to register. This patch + corrects their descriptions. + + Change-Id: Id004b1becfdeed8ee3c846f49beab76a5c0f68b6 + +2015-06-26 10:57 +0000 [47ea312b24] Benjamin Ford <bford@digium.com> + + * ARI: Added new functionality to get all module information. + + An http request can be sent to retrieve a list of all existing modules, + including the resource name, description, use count, status, and + support level. + + The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/ + asterisk/modules" (or something similar, depending on configuration) + can be run in the terminal to access this new functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Information on modules can now be retrieved + + Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0 + +2015-07-09 09:18 +0000 [d558b00c85] Joshua Colp <jcolp@digium.com> + + * bridge_native_rtp.c: Don't start native RTP bridging after attended transfer. + + The bridge_native_rtp module adds a frame hook to channels which are in + a native RTP bridge. This frame hook is used to intercept when a hold + or unhold frame traverses the bridge so native RTP can be stopped or + started as appropriate. This is expected but exposes a specific bug + when attended transfers are involved. + + Upon completion of an attended transfer an unhold frame is queued up + to take one of the channels involved off hold. After this is done + the channel is moved between bridges. + + When the frame hook is involved in this case for the unhold it + releases the channel lock and acquires the bridge lock. This + allows the bridge core to step in and move the channel + (potentially changing the bridging techology) from another thread. + Once completed the bridge lock is released by the bridge core. + The frame hook is then able to acquire the bridge lock and + wrongfully starts native RTP again, despite the channel no longer + being in the bridge or needing to start native RTP. In fact at + this point the frame hook is no longer attached to the channel. + + This change makes it so the native RTP bridge data is available to + the frame hook when it is invoked. Whether the frame hook has + been detached or not is stored on the native RTP bridge data and + is checked by the frame hook before starting or stopping native + RTP bridging. If the frame hook has been detached it does nothing. + + ASTERISK-25240 #close + + Change-Id: I13a73186a05f4e5a764f81e5cd0ccec1ed1891d2 + +2015-05-16 17:02 +0000 [b74b071369] Joshua Colp <jcolp@digium.com> + + * res_sorcery_memory_cache: Backport to 13 + + Gerrit is complaining of conflicts when trying to create a patch series + of all of the cherry-picked master commits, so I have instead squashed + it all into one commit. + + ASTERISK-25067 #close + Reported by: Matt Jordan + + Change-Id: I6dda90343fae24a75dc5beec84980024e8d61eb9 + +2015-07-08 04:21 +0000 [7ff1ac8797] Joshua Colp <jcolp@digium.com> + + * res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used. + + This change fixes a bug where the DTLS timeout timer would be + initialized to 0 if DTLS was not used for an RTP session. + + ASTERISK-25103 + + Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac + +2015-07-01 07:55 +0000 [05e8e14982] Joshua Colp <jcolp@digium.com> + + * res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context. + + This change moves logic for setting up the DTLS SSL contexts to + when the SDP is done being processed instead of when ICE negotiation + completes. It also stops handshakes from being initiated when we + are acting as a server. + + Manipulating the SSL context when ICE negotiation has completed + is problematic as the SSL context is not protected and if acting + as a client the remote side may have started DTLS negotiation + already. + + The retransmission timeout timer code has also been split up + and simplified some. Both RTP and RTCP now have their own timers + and the points at which the timer is stopped and started is now + more specific. When a packet is sent the timer is started. When + a response is received but before it is processed the timer is + stopped. This provides a guarantee that the timeout is not + occurring while the response is processed. + + ASTERISK-22805 #close + ASTERISK-24550 #close + ASTERISK-24651 #close + ASTERISK-24832 #close + ASTERISK-25103 #close + ASTERISK-25127 #close + + Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91 + +2015-06-26 16:10 +0000 [38bace4fbb] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_t38.c: Fix always false if test. + + Calling t38_change_state() sets the t38 state so it makes little sense to + then check the state right after the call for something else. + + * Made the code in t38_interpret_parameters() reject or exit T.38 mode as + intended but not implemented. + + Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2 + +2015-06-30 11:17 +0000 [2f7688c788] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str(). + + Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907 + +2015-06-30 11:14 +0000 [74be3a50d7] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_mwi.c: Eliminate a simple RAII_VAR. + + Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da + +2015-06-30 11:11 +0000 [589e93617a] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_mwi.c: Fix mid-line log message line breaks. + + * Add create_mwi_subscriptions_for_endpoint() doxygen comment. + + Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2 + +2015-06-26 18:48 +0000 [0d67e04359] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_mwi.c: Fix MWI subscription memory corruption crash. + + MWI subscriptions can crash or corrupt memory when using the subscription + datastore to access the MWI subscription object because the datastore is + not holding a reference to the object. + + * Give the subscription datastore a ref to the MWI subscription object. + It is unfortunate that the ref causes a circular ref chain that must be + explicitly broken to allow the memory to get released. The loop is broken + when the subscription is shutdown and if the subscription setup fails. + + ASTERISK-25168 #close + Reported by: Carl Fortin + + Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f + +2015-07-02 14:51 +0000 [0422433f47] Richard Mudgett <rmudgett@digium.com> + + * PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error. + + When res_pjsip body generator modules were generating XML or XPIDF + response bodies, there was a chance that the generated body would be the + exact size of the supplied buffer. Adding the nul string terminator would + then write beyond the end of the buffer and potentially corrupt memory. + + * Fix MALLOC_DEBUG high fence violations caused by adding a nul string + terminator on the end of a buffer for XML or XPIDF response bodies. + + * Made calls to pj_xml_print() safer if the XML prolog is requested. Due + to a bug in pjproject, the return value could be -1 _or_ + AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough. + + * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the + return value of pj_xml_print() when the supplied buffer is not large + enough. + + ASTERISK-25168 + Reported by: Carl Fortin + + Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de + +2015-06-26 10:36 +0000 [8ea214aed7] Richard Mudgett <rmudgett@digium.com> + + * PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences. + + When a caller calls a FAX number and then hangs up right after the call is + answered then the T.38 re-INVITE automatic reject timer may still be + running after the channel goes away. + + * Added session NULL channel checks on the code paths that get executed by + t38_automatic_reject() to prevent a crash when the T.38 re-INVITE + automatic reject timer expires. + + ASTERISK-25168 + Reported by: Carl Fortin + + Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403 + +2015-06-05 15:37 +0000 [ada7346792] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip: Need to use the same serializer for a pjproject SIP transaction. + + All send/receive processing for a SIP transaction needs to be done under + the same threadpool serializer to prevent reentrancy problems inside + pjproject and res_pjsip. + + * Add threadpool API call to get the current serializer associated with + the worker thread. + + * Pick a serializer from a pool of default serializers if the caller of + res_pjsip.c:ast_sip_push_task() does not provide one. + + This is a simple way to ensure that all outgoing SIP request messages are + processed under a serializer. Otherwise, any place where a pushed task is + done that would result in an outgoing out-of-dialog request would need to + be modified to supply a serializer. Serializers from the default + serializer pool are picked in a round robin sequence for simplicity. + + A side effect is that the default serializer pool will limit the growth of + the thread pool from random tasks. This is not necessarily a bad thing. + + * Made pjsip_distributor.c save the thread's serializer name on the + outgoing request tdata struct so the response can be processed under the + same serializer. + + This is a cherry-pick from master. + + **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a + + NOTE: session_inv_on_state_changed() is disassociating the dialog from the + session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED. + Unfortunately this is a tad too soon because our BYE request transaction + has not completed yet. + + ASTERISK-25183 #close + Reported by: Matt Jordan + + Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a + +2015-07-04 18:22 +0000 [55137c3d12] Joshua Colp <jcolp@digium.com> + + * res/res_http_websocket: Don't send HTTP response fragmented. + + This change makes it so that when accepting a WebSocket + connection the HTTP response is sent as one packet instead of + fragmented. Browsers don't like it when you send it fragmented. + + ASTERISK-25103 + + Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69 + +2015-06-27 18:47 +0000 [49f81ddb85] Matt Jordan <mjordan@digium.com> + + * Makefile: Remove coverage files on 'make clean' + + This patch updates a variety of Makefiles in Asterisk's build system to + remove .gcda and .gcno files when 'make clean' is executed. These files + are generated when '--enable-coverage' is passed to the Asterisk + configure script. + + Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602 + +2015-07-02 09:08 +0000 [e0f565663b] Walter Doekes <walter+asterisk@wjd.nu> + + * chan_sip: Fix early call pickup channel leak. + + When handle_invite_replaces() was called, and either ast_bridge_impart() + failed or there was no bridge (because the channel we're picking up was + still ringing), chan_sip would leak a channel. + + Thanks Matt and Corey for checking the bridge path. + + ASTERISK-25226 #close + + Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8 + +2015-07-02 06:19 +0000 [a5a262be78] Walter Doekes <walter+asterisk@wjd.nu> + + * chan_mgcp: Don't call close on fd -1. + + ASTERISK-25220 #close + + Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3 + +2015-07-02 06:10 +0000 [b835312b4c] Walter Doekes <walter+asterisk@wjd.nu> + + * rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format. + + When running valgrind on Asterisk, it complained about: + + ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304) + ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...) + ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292) + ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437) + + The code in question is a struct assignment, which may be performed by + memcpy as a compiler optimization. It is changed to only copy the struct + contents if source and destination are different. + + ASTERISK-25219 #close + + Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a + +2015-07-02 05:16 +0000 [6551e16e03] Walter Doekes <walter+asterisk@wjd.nu> + + * astfd: Fix buffer overflow in DEBUG_FD_LEAKS. + + If DEBUG_FD_LEAKS was used and more file descriptors than the default of + 1024 were available, some DEBUG_FD_LEAKS-patched functions would + overwrite memory past the fixed-size (1024) fdleaks buffer. + + This change: + - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe + - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024 + - stores pointers to constants instead of copying the contents + - reorders the fdleaks struct for possibly tighter packing + - adds a tiny bit of documentation + + ASTERISK-25212 #close + + Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5 + +2015-07-02 04:57 +0000 [f4dd9560cf] Walter Doekes <walter+asterisk@wjd.nu> + + * res_timing: Don't close FD 0 when out of open files. + + This fixes so a failure to get a timer file descriptor does not cascade + to closing FD 0. + + On error, both res_timing_kqueue and res_timing_timerfd would call the + destructor before setting the file handle. The file handle had been + initialized to 0, causing FD 0 to be closed. This in turn, resulted in + floods of "CLI>" messages and an unusable terminal. + + ASTERISK-19277 #close + Reported by: Barry Chern + + For the 13 branch, this was already fixed. This patch only ensures that + we do not attempt to close a negative file descriptor. + + Change-Id: I147d7e33726c6e5a2751928d56561494f5800350 + +2015-07-01 17:25 +0000 [78a1f4aa46] Richard Mudgett <rmudgett@digium.com> + + * chan_vpb.cc: Fix compiler warning Jenkins found. + + Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0 + +2015-07-01 13:34 +0000 [6b16fbfc22] Scott Griepentrog <scott@griepentrog.com> + + * Channel alert pipe: improve diagnostic error return + + When a frame is queued on a channel, any failure in + ast_channel_alert_write is logged along with errno. + + This change improves the diagnostic message through + aligning the errno value with actual failure cases. + + ASTERISK-25224 + Reported by: Andrey Biglari + + Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b + +2015-07-01 16:04 +0000 [8e07ab145d] Matt Jordan <mjordan@digium.com> + + * sorcery/realtime: Add a bit of debug and warning messages for bad configs + + When a mapping does not exist between a sorcery.conf defined object and + a realtime mapping in extconf, currently, the user will receive a slew + of ERROR messages that don't really tell what is happening. Some ERROR + messages may even be misleading, as they occur after the sorcery API has + already given up on the attempt to load and create the sorcery object. + + This patch adds a bit of debug and a useful WARNING message for when a + wizard's open callback fails for a particular object type. In the bad + configurations that resulted in this patch, this provided a 'root cause' + WARNING message that pointed in the right direction of the configuration + problem. + + Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b +2015-06-29 12:45 +0000 [156395e743] Mark Michelson <mmichelson@digium.com> + + * res_sorcery_realtime: Fix leak of sorcery object type. + + This prevents a leak of a sorcery object type when realtime sorcery + objects are retrieved by fields or when multiple objects are retrieved. + + The extent of this leak is that sorcery object types would be leaked. + These are allocated whenever an object type is registered with sorcery, + meaning that on module shutdown, these objects would be leaked. This + could be problematic if many reloads were performed, but it is not as + severe as if every sorcery object retrieved from realtime were being + leaked. + + ASTERISK-25165 #close + Reported by Corey Farrell + + Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8 + +2015-06-26 22:02 +0000 [a5e9c4e9b2] Matt Jordan <mjordan@digium.com> + + * res/res_corosync: Always decline module load, instead of failing + + Returns a 'failure' from the module load routine indicates to Asterisk + that it should abort loading completely. This is rarely - in fact, + really, never - a good option. Aborting load of Asterisk from a dynamic + module implies that the core, and the rest of the dynamic modules, don't + matter: we should abandon all processing. + + res_corosync is really not that important. + + This patch updates the module such that, if it fails to load, it + politely declines (emitting ERROR messages along the way), and allows + Asterisk to continue to function. + + Note that this issue was keeping Asterisk unit tests from running on + certain build agents. + + Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f + +2015-06-26 20:38 +0000 [399cd8bcd9] Matt Jordan <mjordan@digium.com> + + * main/pbx: Resolve case sensitivity regression in PBX hints + + When 8297136f was merged for ASTERISK-25040, a regression was introduced + surrounding the case sensitivity of device names within hints. + Previously, device names - such as 'sip/foo' - were compared in a case + insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After + that patch, only the case sensitive name would match, i.e., 'SIP/foo'. + As a result, some dialplan hints stopped working. + + This patch re-introduces case insensitive matching for device names in + hints. + + ASTERISK-25040 + + ASTERISK-25202 #close + + Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c + (cherry picked from commit 96bbcf495a1da9e607d9b04a44b5c4f49e83cc03) + +2015-06-26 16:12 +0000 [24eec5a10b] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_nat: Adjust when contact should be rewritten. + + A previous change made the contact only get rewritten if the dialog's + route set was not marked frozen. Unfortunately, while the intent of this + is correct, the dialog's route set actually gets marked as frozen + earlier than expected, especially for UAS dialogs. + + Instead, the idea is that the contact needs to not be rewritten if there + is a pre-existing route set on the dialog. This is now accomplished by + checking the dialog's route set list instead of checking if the route + set is frozen. + + Doing this causes some broken tests to begin passing again. + + ASTERISK-25196 + Reported by Mark Michelson + + Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e + +2015-06-19 18:27 +0000 [0ec461a637] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Add a serializer shutdown group. + + The client_state objects contain a serializer used to send the outbound + REGISTER messages. Once all those message transactions are complete then + the module can shutdown. + + ASTERISK-24907 #close + Reported by: Kevin Harwell + + Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547 + +2015-06-26 10:41 +0000 [05a2cc1293] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_refer: Prevent sending duplicate headers. + + res_pjsip_refer will attempt to add Referred-By or Replaces headers to + outbound INVITEs at times. If the INVITE gets challenged for + authentication, then we will resend the INVITE. Prior to this patch, the + Referred-By or Replaces header would be re-added to the outbound INVITE, + resulting in duplicated headers. + + ASTERISK-25204 #close + Reported by Mark Michelson + + Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d + +2015-06-23 17:43 +0000 [028fa54620] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_nat: Rewrite route set when required. + + When performing some provider testing, the rewrite_contact option was + interfering with proper construction of a route set when sending an ACK + after receiving a 200 OK response to an INVITE. + + The initial INVITE was sent to address sip:foo. The 200 OK had a Contact + header with URI sip:bar. In addition, the 200 OK had Record-Route + headers for sip:baz and sip:foo, in that order. Since the Record-Route + headers had the lr parameter, the result should have been: + + * Set R-URI of the ACK to sip:bar. + * Add Route headers for sip:foo and sip:baz, in that order. + + However, the rewrite_contact option resulted in our rewriting the + Contact header on the 200 OK to sip:foo. The result was: + + * R-URI remained sip:foo. + * We added Route headers for sip:foo and sip:baz, in that order. + + The result was that sip:bar was not indicated in the ACK at all, so the + far end never received our ACK. The call eventually dropped. + + The intention of rewrite_contact is to rewrite the most immediate + destination of our SIP request to be the same address on which we + received a request or response. In the case of processing a SIP response + with Record-Route headers, this means that instead of rewriting the + Contact header, we should instead rewrite the bottom-most Record-Route + header. In the case of processing a SIP request with Record-Route + headers, this means we rewrite the top-most Record-route header. + Like when we rewrite the Contact header, we also ensure to update + the dialog's route set if it exists. + + ASTERISK-25196 #close + Reported by Mark Michelson + + Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f +2015-06-19 16:16 +0000 [84c12f9e0c] Richard Mudgett <rmudgett@digium.com> + + * threadpool, res_pjsip: Add serializer group shutdown API calls. + + A module trying to unload needs to wait for all serializers it creates and + uses to complete processing before unloading. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059 + +2015-06-16 15:06 +0000 [602c4b74b5] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs + + * handle_client_state_destruction() must always be passed a ref to + client_state because it will always unref client_state. + handle_registration_response() was not passing a client_state ref. + + * Made the final un-REGISTER message get sent normally using the pjproject + register control structure in handle_client_state_destruction(). The + previous code attempted to short circuit the response handling for the + module to unload. That doesn't work for a couple reasons. One, + pjsip_regc_send() may call the registered callback before it returns and + unbalance the client_state ref count. Two, the registered callback + handles any authentication for the un-REGISTER message. + + * Made the distinction between internal registration state and external + registration status with sip_outbound_registration_status_str(). This is + necessary to avoid altering documented AMI messages with internal + changes. + + * Removed references to client_state->client outside of the serializer + thread. When handle_client_state_destruction() destroys the pjproject + register control structure that memory is freed and cannot be referenced + anymore. These accesses were to provide information for debug and + off-nominal warning messages. + + * In sip_outbound_registration_timer_cb() you should not access entry->id + after unrefing client_state because the passed in entry is normally + pointing to the timer entry in the client_state object. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f + +2015-06-15 15:28 +0000 [8c6a95a9ac] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API + + The sorcery pjsip 'registration' config object needs to be destroyed on + module unload. Otherwise, a reload of res_pjsip could try to use + callbacks for a previously unloaded instance of the module provided by + ast_sorcery_object_register() or one of the variants. Also, if + res_pjsip_outbound_registration were subsequently reloaded, the sorcery + config field objects would be registered in sorcery twice. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697 + +2015-06-25 06:42 +0000 [e4a2ef9e4e] Joshua Colp <jcolp@digium.com> + + * channel: Remove ignore of answer on non-outgoing channels. + + Due to the way that channels can now be moved around inside of + Asterisk it is possible for the outgoing flag of a channel to get + cleared before it has been answered. This results in the bridge + not receiving notification that the outgoing leg has been answered. + + This most easily exhibits itself with DTMF based blond transfers. + Since the answer of the outgoing leg is ignored the other party + continues to receive both a locally generated ringing and the + media stream of the outgoing leg upon its answer. This results + in no media being heard. + + This change removes the ignore of the answer and allows it + to pass through. + + ASTERISK-25171 #close + + Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e + +2015-06-15 15:28 +0000 [20f3d77ab9] Richard Mudgett <rmudgett@digium.com> + + * sorcery: Add ast_sorcery_object_unregister() API call. + + Find and unlink the specified sorcery object type to complement + ast_sorcery_object_register(). Without this function you cannot + completely unload individual modules that use sorcery for configuration. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88 + +2015-06-15 13:38 +0000 [4313f32969] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Reorder load_module() and unload_module(). + + It is best if the loading code creates and initializes the module's + infrastructure before letting the system know of its existence. The + unloading code needs to reverse the actions of the loading code and in the + reverse order. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4 + +2015-06-23 14:34 +0000 [890c923786] Richard Mudgett <rmudgett@digium.com> + + * AMI: Add Linkedid to the standard channel snapshot AMI event headers. + + * The AMI version is bumped to 2.8.0. + + ASTERISK-25189 #close + Reported by: John Hardin + + Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac + +2015-06-24 14:30 +0000 [2602a7484b] Richard Mudgett <rmudgett@digium.com> + + * test.c: Add unit test registration checks for summary and description. + + Added checks when a unit test is registered to see that the summary and + description strings do not end with a new-line '\n' for consistency. + + The check generates a warning message and will cause the + /main/test/registrations unit test to fail. + + * Updated struct ast_test_info member doxygen comments. + + Change-Id: I295909b6bc013ed9b6882e85c05287082497534d + +2015-06-24 14:39 +0000 [2b0482d699] Richard Mudgett <rmudgett@digium.com> + + * Unit tests: Fix unit test description strings. + + Analyzing the code shows that the unit test summary and description + strings should not end with a new-line character. Where these strings are + used in the code a new-line is provided for output. + + Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba + +2015-06-23 11:21 +0000 [e99e654d75] Joshua Colp <jcolp@digium.com> + + * app_dial: Hold reference to calling channel formats when dialing outbound. + + Currently when requesting a channel the native formats of the + calling channel are provided to the core for usage when dialing + the outbound channel. This occurs without holding the channel lock + or keeping a reference to the formats. This is problematic as + the channel driver may end up changing the formats during this time. + In the case of chan_sip this happens when an SDP negotiation + completes. + + This change makes it so app_dial keeps a reference to the native + formats of the calling channel which guarantees that they will + remain valid for the period of time needed. + + ASTERISK-25172 #close + + Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db +2015-06-17 05:04 +0000 [80e82dc97f] Joshua Colp <jcolp@digium.com> + + * res_pjsip_mwi: Set up unsolicited MWI upon registration. + + The res_pjsip_mwi previously required a reload to set up the proper + subscriptions to allow unsolicited MWI to work. This change + makes it so the act of registering will also cause this to occur. + This is particularly useful if realtime is involved as no reload + needs to occur within Asterisk to cause the MWI information + to get sent. + + ASTERISK-25180 #close + + Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252 + +2015-06-22 15:11 +0000 [35a99b6394] Kevin Harwell <kharwell@digium.com> + + * bridge.c: Hangup attended transfer target if bridged + + After completing an attended transfer the transfer target channel was not being + hung up after leaving the bridge. Added an explicit softhangup to hangup said + channel, but only if it was previously bridged. + + ASTERISK-24782 #close + Reported by: John Bigelow + + Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada + +2015-06-17 16:23 +0000 [036bc0012f] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Add missing line endings to CLI commands + + Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7 + +2015-06-12 14:29 +0000 [bec7435945] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage. + + Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e + +2015-06-12 13:33 +0000 [c2519fdf1c] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_outbound_registration.c: Misc code cleanups. + + * Break some long lines. + + * Fix doxygen comment. + + Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305 + +2015-06-22 09:26 +0000 [a419c69def] Alexander Traud (License 6520) + + * chan_sip: Reload peer without its old capabilities. + + On reload, previously allowed codecs were not removed. Therefore, it was not + possible to remove codecs while Asterisk was running. Furthermore, newly added + codecs got appended behind the previous codecs. Therefore, it was not possible + to add a codec with a priority of #1. This change removes the old capabilities + before the current ones are added. + + ASTERISK-25182 #close + Reported by: Alexander Traud + patches: + asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) + + Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802 + +2015-06-20 19:38 +0000 [74616ae43d] Joshua Colp <jcolp@digium.com> + + * chan_sip: Destroy peers without holding peers container lock. + + Due to the use of stasis_unsubscribe_and_join in the peer destructor + it is possible for a deadlock to occur when an event callback is + occurring at the same time. + + This happens because the peer may be destroyed while holding the + peers container lock. If this occurs the event callback will never + be able to acquire the container lock and the unsubscribe will + never complete. + + This change makes it so the peers that have been removed from the + peers container are not destroyed with the container lock held. + + ASTERISK-25163 #close + + Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33 + +2015-06-18 13:16 +0000 [9015bb4c8c] Mark Michelson <mmichelson@digium.com> + + * Resolve race conditions involving Stasis bridges. + + This resolves two observed race conditions. + + First, a bit of background on what the Stasis application does: + + 1a Creates a stasis_app_control structure. This structure is linked into + a global container and can be looked up using a channel's unique ID. + 2a Puts the channel in an event loop. The event loop can exit either + because the stasis_app_control structure has been marked done, or + because of some other factor, such as a hangup. In the event loop, the + stasis_app_control determines if any specific ARI commands need to be + run on the channel and will run them from this thread. + 3a Checks if the channel is bridged. If the channel is bridged, then + ast_bridge_depart() is called since channels that are added to Stasis + bridges are always imparted as departable. + 4a Unlink the stasis_app_control from the container. + + When an ARI command is received by Asterisk, the following occurs + 1b A thread is spawned to handle the HTTP request + 2b The stasis_app_control(s) that corresponds to the channel(s) in the + request is/are retrieved. If the stasis_app_control cannot be + retrieved, then it is assumed that the channel in question has exited + the Stasis app or perhaps was never in Stasis in the first place. + 3b A command is queued onto the stasis_app_control, and the channel's + event loop thread is signaled to run the command. + 4b While most ARI commands do nothing further, some, such as adding or + removing channels from a bridge, will block until the command they + issued has been completed by the channel's event loop. + + The first race condition that is solved by this patch involves a crash + that can occur due to faulty detection of the channel's bridged status + in step 3a. What can happen is that in step 2a, the event loop may run + the ast_bridge_impart() function to asynchronously place the channel + into a bridge, then immediately exit the event loop because the channel + has hung up. In step 3a, we would detect that the channel was not + bridged and would not call ast_bridge_depart(). The reason that the + channel did not appear to be bridged was that the depart_thread that is + spawned by ast_bridge_impart() had not yet started. That is the thread + where the channel is marked as being bridged. Since we did not call + ast_bridge_depart(), the Stasis application would exit, and then the + channel would be destroyed Then the depart_thread would start up and + try to manipulate the destroyed channel, causing a crash. + + The fix for this is to switch from using ast_channel_is_bridged() to + checking the NULLity of ast_channel_internal_bridge_channel() to + determine if ast_bridge_depart() needs to be called. The channel's + internal bridge_channel is set when ast_bridge_impart() is called and + is NULLed by the call to ast_bridge_depart(). If the channel's internal + bridge_channel is non-NULL, then the channel must have been imparted + into the bridge and needs to be departed, even if the actual bridging + operation has not yet started. By departing the channel when necessary, + the thread that is running the Stasis application will block until the + bridge gives the okay that the depart_thread has exited. + + The second race condition that is solved by this patch involves a leak + of HTTP handler threads. The problem was that step 2b would successfully + retrieve a stasis_app_control structure. Then step 2a would exit the + channel from the event loop due to a hangup. Steps 3a and 4a would + execute, and then finally steps 3b and 4b would. The problem is that at + step 4b, when attempting to add a channel to a bridge, the thread would + block forever since the channel would never execute the queued command + since it was finished with the event loop. This meant that the HTTP + handling thread would be leaked, along with any references that thread + may have owned (in my case, I was seeing bridges leaked). + + The fix for this is to hone in better on when the channel has exited the + event loop. The stasis_app_control structure has an is_done field that + is now set at each point where the channel may exit the event loop. If + step 2b retrieves a valid stasis_app_control structure but the control + is marked as done, then the attempted operation exits immediately since + there will be nothing to service the attempted command. + + ASTERISK-25091 #close + Reported by Ilya Trikoz + + Change-Id: If66265b73b4c9f8f58599124d777fedc54576628 +2015-06-16 11:13 +0000 [723a9d4225] Mark Michelson <mmichelson@digium.com> + + * Parking: Add documentation for AMI ParkedCallSwap event. + + This event was added some time ago in order to clarify when a channel + took the place of another channel in a parking lot. However, there was + no XML documentation added for the event. This patch adds the XML + documentation. + + ASTERISK-24900 #close + Reported by Rusty Newton + + Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac +2015-06-15 16:40 +0000 [79bf56c78a] Corey Farrell <git@cfware.com> + + * func_pjsip_aor: Fix leaked contact from iterator. + + ASTERISK-25162 #close + + Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e + +2015-06-12 16:58 +0000 [31c77b157b] Kevin Harwell <kharwell@digium.com> + + * res_pjsip: Add option to force G.726 to be treated as AAL2 packed. + + Some phones send g.726 audio packed for AAL2, which differs from what is + recommended by RFC 3351. If Asterisk receives audio formatted as such when + negotiating g.726 then it sounds a bit distorted. Added an option to + res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 + AAL2 packed. + + ASTERISK-25158 #close + Reported by: Steve Pitts + + Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615 + +2015-06-14 19:48 +0000 [de8c7f46ed] Matt Jordan <mjordan@digium.com> + + * main/cdr: Carry over the disable flag when 'disable all' is specified + + The CDR_PROP function (as well as the NoCDR application) set the + 'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This + flag is supposed to be applied to all CDRs that are currently in the + chain, as well as all CDRs that may be created in the future. Currently, + however, the flag is only applied to the existing CDRs in the chain; new + CDRs do not receive the 'disable all' flag. In particular, this affects + parallel dials, which generate new CDRs for each pair of channels in + the dial attempt. + + This patch carries over the 'disable all' flag when it is specified on a + CDR and a new CDR is generated for the chain. + + ASTERISK-24344 #close + + Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e +2015-06-12 14:28 +0000 [78ea356e78] Matt Jordan <mjordan@digium.com> + + * main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines + + When a parallel dial occurs, a new CDR will be created for each dial + attempt that is made. In most circumstances, the act of creating each + CDR in the chain will include a step that updates the Party A snapshot, + which causes the context/extension of the Party A to be copied onto the + CDR object. + + However, when the Party A is in a subroutine, we explicitly do *not* + copy the context/extension onto the CDR. This prevents the Macro or + GoSub routine name from blowing away the context/extension that the + channel was originally executing in. For the original CDR, this is not a + problem: the original CDR already recorded the last known 'good' state + of the channel just prior to it going into the subroutine. However, for + newly generated CDRs in a chain, there is no context/extension set on + them. Since we are in a subroutine, we will never set the Party A's + context/extension on the CDR, and we end up with a CDR with no + destination recorded on it. + + This patch updates the creation of a chained CDR such that it copies + over the original CDR's context/extension. This is the last known "good" + state of the CDR, and is a reasonable starting point for the newly + generated CDR. In the case where we are not in a subroutine, subsequent + code will update the location of the CDR from the Party A information; + in the case where we are in a subroutine, the context/extension on the + original CDR is the correct information. + + ASTERISK-24443 #close + + Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a + +2015-06-11 08:18 +0000 [3f57f3f8ec] Damian Ivereigh <damo@launtel.net.au> + + * chan_sip.c: Update dialog fromtag after request with auth + + If a client sends and INVITE which is 401 rejected, then subsequently + sends a new INVITE with the auth info and uses a different fromtag + from the first INVITE, Asterisk will accept the new INVITE as part of + the original dialog - match_req_to_dialog() specifically ignores the + fromtag. However it does not update the stored dialog with the new + fromtag. + + This results in Asterisk being unable to match future packets that are + part of this dialog (such as the ACK to the OK or the OK to the BYE), + and the call is dropped. + + This problem was originally found when using an NEC-i SV8100-GE (NEC SIP + Card). + + * After a successful match of a packet to the dialog, if the packet is + not a SIP_RESPONSE, authentication is present and the fromtags are + different, the stored fromtag is updated with the one from the recent + INVITE. + + ASTERISK-25154 #close + Reported by: Damian Ivereigh + Tested by: Damian Ivereigh + + Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e + +2015-06-11 18:52 +0000 [30a0f2d9ac] Matt Jordan <mjordan@digium.com> + + * chan_pjsip: Set the context and extension on the channel when created + + Prior to this patch, chan_pjsip was failing to pass the endpoint's + context and the desired extension to the ast_channel_alloc_* routine. + This caused a new channel snapshot to be issued without a context and + extension, which can cause some reporting issues for users of AMI, CEL, + and other APIs. The channel driver would later set the context and + extension on the channel such that the channel would start in the + correct location in the dialplan, but the information reported in the + initial event would be incorrect. + + This patch modifies the channel driver such that it now passes the + context and extension directly into the allocation routine. This + provides the information in the new channel snapshot published over + Stasis. + + ASTERISK-25156 #close + Reported by: cloos + + Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e + +2015-06-10 18:28 +0000 [dbb067279e] Joshua Colp <jcolp@digium.com> + + * bridge: When performing a blonde transfer update connected line information. + + When performing a blonde transfer the code uses the old masquerade + mechanism to move a channel around. As a result of this certain information, + such as connected line, is moved between the channels involved. Upon + completion of the move a frame is queued which is supposed to update the + connected line information on the channel. This does not occur as the + code considers it a redundant update since the masquerade operation + updated the channel (but did not inform it of the new connected line + information). The code also does not queue a connected line update + to be handled by the thread handling the channel. Without this any + other channel that may be loosely involved does not know it is + talking to a different caller. + + This change does the following to resolve this: + + 1. The indicated connected line information is cleared upon + completion of the masquerade operation when doing a blonde transfer. + This prevents the connected line update from being considered + redundant. + + 2. A connected line update frame is now queued upon the completion + of the masquerade operation so any other channel loosely involved + knows that there is a different caller. + + ASTERISK-25157 #close + Reported by: Joshua Colp + + Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20 + +2015-06-11 14:39 +0000 [a2f4d03c87] Richard Mudgett <rmudgett@digium.com> + + * app_directory: Fix crash when using the alias option 'a'. + + The voicemail.conf mailbox key/value pair is defined as: + <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]] + Where all fields in the value including the field values are optional. + + Since the parsing code for the mailbox key/value pair is sloppy, this + patch tightens the parsing for the directory information. + + * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' + respectively in search_directory_sub(). Those names make more sense. + + * Made sure that search_directory_sub() is dealing with the voicemail.conf + mailbox options field if it even exists when looking for the 'hidefromdir' + and 'alias' options. + + * Fix crash if a voicemail.conf mailbox is just + <mailbox>=<password>,<name> when the 'a' option is used. If there were no + fields after the name then the 'options' pointer was not checked for NULL. + + * Fix users.conf alias processing if the 'a' option is used. The wrong + variable was used. + + ASTERISK-25087 #close + Reported by: Chet Stevens + + Change-Id: I86052ea77307beddddba5279824d39dc0d593374 + +2015-06-09 15:31 +0000 [a2b718f4f6] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip.h: Fix some doxygen comments. + + Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976 + +2015-06-05 13:46 +0000 [32ddf6d86b] Richard Mudgett <rmudgett@digium.com> + + * taskprocessor.c: Remove extra unref from off-nominal path. + + Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60 + +2015-04-20 16:00 +0000 [cf98c744d5] Yousf Ateya <y.ateya@starkbits.com> + + * chan_iax2: Prevent deadlock between hangup and sending lagrq/ping + + channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/ + send_ping. This deadlock happens because the scheduled task send_lagrq(or + send_ping) starts execution after the call hangup procedure starts but before + it deletes the tasks in the scheduler. + + The solution is to delete scheduled lagrq (and ping) task asynchronously + (i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will + be called in a new context (doesn't have callno locked). + + This commit also cleans up the procedure of sending LAGRQ and PING. + + main/sched.c: Do not assert when deleting non existant entry from scheduler. + This assert seems to be the reason for a lot of awkward code to avoid it. + + ASTERISK-24983 #close + Reported by: Y Ateya + + Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c + +2015-05-31 12:37 +0000 [8af6c9cf6b] Ivan Poddubny <ivan.poddubny@gmail.com> + + * res_pjsip_transport_websocket: Fix use-after-free bugs. + + This patch fixes use-after-free bugs caught by AddressSanitizer. + + 1. PJSIP transport manager may decide to destroy transport on its own. + For example, when the contact registered via websocket has not renewed + its registration in time. The transport was destoyed, but the websocket + listener thread was still active until the socket closes, and then tried + to call transport_shutdown on transport that has been freed. + + Also, the transport destructor accessed wstransport->rdata.tp_info.pool + right after freeing memory that contained wstransport itself. + + This patch converts transport to an ao2 object, allowing it to be + refcounted, so that it is available until both websocket listener and + pjsip transport manager are finished with it. + + 2. The websocket listener deletes the last reference on websocket session + when the tcp connection is closed, and it gets destroyed, but + the transport manager may still use it, for example when disconnect + happens in the middle of a SIP transaction. + + A new reference to websocket session has been added that is released + with the transport to prevent this. + + ASTERISK-25096 #close + Reported by: Josh Kitchens + + ASTERISK-24963 #close + Reported by: Badalian Vyacheslav + + Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b + +2015-06-09 13:41 +0000 [3046bc17ed] ibercom <ibercom123@gmail.com> + + * weakref attribute detection broken with gcc 4.6 and higher + + GCC 4.7 Manual: + http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html + + weakref ("target") + + A weak reference is an alias that does not by itself require a definition + to be given for the target symbol. + + ASTERISK-22559 #close + Reported by: Ibercom + + Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf + +2015-06-08 10:09 +0000 [55c8daf88b] Corey Farrell <git@cfware.com> + + * Fix unsafe uses of ast_context pointers. + + Although ast_context_find, ast_context_find_or_create and + ast_context_destroy perform locking of the contexts table, + any context pointer can become invalid at any time that the + contexts table is unlocked. This change adds locking around + all complete operations involving these functions. + + Places where ast_context_find was followed by ast_context_destroy + have been replaced with calls ast_context_destroy_by_name. + + ASTERISK-25094 #close + Reported by: Corey Farrell + + Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa + +2015-06-04 07:14 +0000 [e0090216db] ibercom <ibercom123@gmail.com> + + * CLI: Cosmetic issue - core show uptime + + Show uptime information ends with an unnecessary space. + + Now NEEDCOMMA is better defined. + + Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1 + +2015-06-03 17:41 +0000 [88212ccb7f] Mark Michelson <mmichelson@digium.com> + + * res_pjsip: Prevent access of NULL channels. + + It is possible to receive incoming requests or responses after the channel + on an ast_sip_session has been destroyed and NULLed out. Handlers of these + sorts of requests or responses need to be prepared for the possibility + that the channel is NULL or else they could cause a crash. + + While several places have been amended to deal with NULL channels, there + were still a couple of places that needed updating. + + res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to + return early if there is no channel on the session. + + res_pjsip_session.c: When handling a 302 response, we need to stop the + redirecting attempt if there is no channel on the session. + + ASTERISK-25148 #close + reported by Mark Michelson + + Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9 + +2015-06-01 11:45 +0000 [f5d5aa67dc] Kevin Harwell <kharwell@digium.com> + + * AMI: Escape string values. + + So this issue is a bit complicated. Since it is possible to pass values to AMI + that contain a '\r\n' (or other similar sequences) these values need to be + escaped. One way to solve this is to escape the values and then pass the escaped + values to the AMI variable parameter string building function. However, this + puts the onus on the pre-build function to escape all string values. This + potentially requires a fair amount of changes along with a lot of string + allocations/freeing for all values. + + Surely there is a way to push this complexity down a level into the string + building function itself? This of course is possible, but ends up requiring a + way to distinguish between strings that need to be escaped and those that don't. + The best way to handle this is by introducing a new format specifier in the + format string. For instance a %s (no escape) and %S (escape). However, that is + a bit weird and unexpected. + + So faced with those possibilities this patch implements a limited version of the + first option. Instead of attempting to escape all string values this patch only + escapes those values that make sense. This approach limits the number of changes + and doesn't suffer from the odd format specifier problem. + + ASTERISK-24934 #close + Reported by: warren smith + + Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0 + +2015-06-03 13:17 +0000 [5dc9fb4198] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip/location: Fix ref leak in contact_apply_handler + + contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status + to force the creation of a contact_status object whenever a new + contact is added but it didn't unref the returned object. + + Added an ao2_cleanup(status) to plug the leak. + + ASTERISK-25141 + + Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 + Reported-by: Corey Farrell + +2015-06-02 15:07 +0000 [d908272b7e] David M. Lee <dlee@respoke.io> + + * Fixes for OS X + + * Add some type casting so tv_usec can really be a long, instead of + some strange platform specific type. + + * Add some .dylib style files to .gitignore. + + * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer + versions of GCC, when compiling the Homebrew formula for Asterisk, + are not properly passing the -Xlinker options to the linker. Given + that -Wl, does exactly the [same thing][], and does it properly, this + patch changes the -Xlinker options to use -Wl, instead. + + [reasons unknown]: http://bit.ly/1SUbEYx + [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html + + Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd + +2015-05-30 20:22 +0000 [9e7827e3ac] Corey Farrell <git@cfware.com> + + * pjsip_configuration: Fix leak in persistent_endpoint_update_state. + + The loop to find the first available contact of an endpoint grabbed + contact from the iterator, then checked for offline state. This + caused the first contact after the state was found to leak a reference. + + ASTERISK-25141 + + Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08 +2015-05-31 11:33 +0000 [888bb49618] Ivan Poddubny <ivan.poddubny@gmail.com> + + * Fix buffer overflow in slin sample frames generation. + + The length of frames retured by sample functions was twice as large as + real, what caused global buffer overflow caught by AddressSanitizer. + + ASTERISK-24717 #close + Reported by: Badalian Vyacheslav + + Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6 + +2015-05-29 16:19 +0000 [857166b5e5] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip/location: Fix memory leak in permanent_uri_handler + + When permanent_uri_handler was creating the contact status + object for each contact, it wasn't unreffing it at the + end of the loop. + + ASTERISK-25141 #close + Reported-by: Corey Farrell + + Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12 + +2015-05-29 14:52 +0000 [1558a89129] gtjoseph <george.joseph@fairview5.com> + + * Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change" + + This reverts commit 35c699086ae2fd81b2473307ccb2ae79ad32375a. + + Change-Id: Ia98c2b4820cf579a5b9bb75e9e05d7a233205fb7 + +2015-05-27 13:22 +0000 [35c699086a] gtjoseph <george.joseph@fairview5.com> + + * endpoint/stasis: Eliminate duplicate events on endpoint status change + + When an endpoint was created, it's messages were being forwarded to + both the tech endpoint topic and the all endpoints topic. Since + the tech topic was also forwarded to all, this was resulting in + duplicate messages whenever an endpoint published. This patch + causes the endpoint to only forward to the tech topic and lets + the tech topic forward to all. + + To accomplish this, the existing stasis_cp_single_create function + (which both creates and forwards) was cloned and split into 2 + functions, one that creates the topic and one that sets up the + forwarding. This allows endpoint_internal_create to create + the topic from the endpoint_all cache without forwarding it there, + then allows it to do the forward to the tech's topic. + + ASTERISK-25137 #close + Reported-by: Vitezslav Novy + ASTERISK-25116 #close + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> + + Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c + +2015-05-26 13:56 +0000 [fe21f2e52f] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_session: Fix in-dialog authentication. + + When the remote peer requires authentication for in-dialog requests then + re-INVITEs to the peer cause the call to be disconnected and other + in-dialog requests to the peer like MESSAGE just don't go through. + + * Made session_inv_on_tsx_state_changed() handle in-dialog authentication + for re-INVITEs and other methods. Initial INVITEs cannot be handled here + because the INVITE transaction must be restarted earlier. + + * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in + preparation for removing the file. The generic outbound authentication + code did not work as well as anticipated. + + * Created outbound_invite_auth() to only handle initial outbound INVITEs. + Re-INVITEs cannot be handled here. The re-INVITE transaction is still in + progress and the PJSIP library cannot handle the overlapping INVITE + transactions. Other method types should not be handled here as this code + only works on outgoing calls and we need to handle incoming and outgoing + calls. + + ASTERISK-25131 #close + Reported by: Richard Mudgett + + Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0 + +2015-05-21 17:21 +0000 [262d590819] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes + + Add a new ContactStatus AMI event. + Publish the following status/state changes: + Created + Removed + Reachable + Unreachable + Unknown + + Contact URI, new status/state, aor and endpoint names, and the + last qualify rtt result are included in the event. + + ASTERISK-25114 #close + + Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> + +2015-05-26 07:44 +0000 [5a42397018] Joshua Colp <jcolp@digium.com> + + * sorcery: Fix cache creation callback. + + The cache creation callback function expects to receive a sorcery_details + structure and not just a standalone object. + + Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450 + +2015-05-24 13:47 +0000 [97a6ce1717] Ivan Poddubny <ivan.poddubny@gmail.com> + + * Astobj2: Correctly treat hash_fn returning INT_MIN + + The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0. + However, abs(INT_MIN) = INT_MIN and is still negative, as well as + abs(INT_MIN) % num_buckets, and as a result this led to a crash. + + One way to trigger the bug is using host=::80 or 0.0.0.128 in peer + configuration section in chan_sip or chan_iax. + + This patch takes the remainder before applying abs, so that bucket + number is always in range. + + ASTERISK-25100 #close + Reported by: Mark Petersen + + Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899 +2015-05-23 04:36 +0000 [554bd1e39c] Ivan Poddubny <ivan.poddubny@gmail.com> + + * res_pjsip_transport_websocket: Fix crash on receiving large SIP packets + + Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves + truncated before passing to pjsip_tpmgr_receive_packet, but the length + was passed unaltered, thus causing memory corruption and segfault. + + ASTERISK-25122 #close + + Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab + +2015-05-22 21:50 +0000 [0d266cbe02] Corey Farrell <git@cfware.com> + + * Stasis: Fix unsafe use of stasis_unsubscribe in modules. + + Many uses of stasis_unsubscribe in modules can be reached through unload. + These have been switched to stasis_unsubscribe_and_join. + + Some subscription callbacks do nothing, for these I've created a noop + callback function in stasis.c. This is used by some modules that monitor + MWI topics in order to enable cache, since the callback does not become + invalid after dlclose it is safe to use stasis_unsubscribe on these, even + during module unload. + + ASTERISK-25121 #close + + Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c + +2015-05-22 12:22 +0000 [51ffed5e61] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS + + In addition to specifying lists of 'presence' and 'message-summary', + users can also create lists of type 'dialog'. These should be treated in + the same fashion as 'presence'. + + Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e + +2015-05-22 12:18 +0000 [7950b65e4f] Matt Jordan <mjordan@digium.com> + + * res/res_pjsip_exten_state: Fix confusing NOTICE message + + When a SUBSCRIBE request is made to a dialplan hint that doesn't exist, + the current NOTICE message informing users of this swaps the context and + extension parameters. This can cause a bit of confusion. + + Thanks to CptBurger in #asterisk for helping to point this out. + + Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43 + +2015-05-17 20:36 +0000 [5ac65ddfb4] Matt Jordan <mjordan@digium.com> + + * res/ari: Register Stasis application on WebSocket attempt + + Prior to this patch, when a WebSocket connection is made, ARI would not + be informed of the connection until after the WebSocket layer had + accepted the connection. This created a brief race condition where the + ARI client would be notified that it was connected, a channel would be + sent into the Stasis dialplan application, but ARI would not yet have + registered the Stasis application presented in the HTTP request that + established the WebSocket. + + This patch resolves this issue by doing the following: + * When a WebSocket attempt is made, a callback is made into the ARI + application layer, which verifies and registers the apps presented in + the HTTP request. Because we do not yet have a WebSocket, we cannot + have an event session for the corresponding applications. Some + defensive checks were thus added to make the application objects + tolerant to a NULL event session. + * When a WebSocket connection is made, the registered application is + updated with the newly created event session that wraps the WebSocket + connection. + + ASTERISK-24988 #close + Reported by: Joshua Colp + + Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636 + +2015-05-20 11:11 +0000 [60e2fbfe62] gtjoseph <george.joseph@fairview5.com> + + * res_pjsip: Refactor endpt_send_transaction (qualify_timeout) + + This patch refactors the transaction timeout processing to eliminate + calling the lower level public pjsip functions and reverts to calling + pjsip_endpt_send_request again. This is the result of me noticing + a possible incompatibility with pjproject-2.4 which was causing + contact status flapping. + + The original version of this feature used the lower level calls to + get access to the tsx structure in order to cancel the transaction + when our own timer expires. Since we no longer have that access, + if our own timer expires before the pjsip timer, we call the callbacks + and just let the pjsip transaction take it's own course. When the + transaction ends, it discovers the callbacks have already been run + and just cleans itself up. + + A few messages in pjsip_configuration were also added/cleaned up. + + ASTERISK-25105 #close + + Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> +2015-05-20 00:45 +0000 [42476e6633] demon-ru <serov.d.p@gmail.com> + + * res_pjsip_outbound_registration: Check request URI for line. + + When an inbound call is received the To header is checked + for the "line" option. Some remote servers will place this + in the request URI instead. This adds an additional check for + the option in the request URI. + + ASTERISK-25072 #close + Reported by: Dmitriy Serov + + Change-Id: Id4e44debbb80baad623b914a88574371575353c8 + +2015-05-21 17:51 +0000 [e7edb59db6] Corey Farrell <git@cfware.com> + + * res_mwi_external_ami: Use module version of AMI registration. + + Use ast_manager_register_xml for res_mwi_external_ami manager + actions. This ensures the module is held open while any of + the actions are being run. + + ASTERISK-25117 #close + Reported by: Corey Farrell + + Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7 +2015-05-21 19:59 +0000 Asterisk Development Team <asteriskteam@digium.com> + + * asterisk 13.4.0-rc1 Released. + +2015-05-21 14:56 +0000 [3fb2b375fe] Matt Jordan <mjordan@digium.com> + + * Release summaries: Remove previous versions + +2015-05-21 14:56 +0000 [9d9ae03842] Matt Jordan <mjordan@digium.com> + + * .version: Update for 13.4.0-rc1 + +2015-05-21 14:56 +0000 [53a39083e5] Matt Jordan <mjordan@digium.com> + + * .lastclean: Update for 13.4.0-rc1 + +2015-05-21 14:56 +0000 [7af8ef9346] Matt Jordan <mjordan@digium.com> + + * realtime: Add database scripts for 13.4.0-rc1 + +2015-05-21 14:52 +0000 [20982c68d4] Matt Jordan <mjordan@digium.com> + + * Release summaries: Correct summaries for 13.4.0-rc1 + +2015-05-21 13:17 +0000 [1bb62b037f] mjordan <mjordan@lunkwill> + + * ChangeLog: Updated for 13.4.0-rc1 + +2015-05-21 13:17 +0000 [1e98a36699] mjordan <mjordan@lunkwill> + + * Release summaries: Add summaries for 13.4.0-rc1 + +2015-05-21 13:15 +0000 [5c12e5ba72] mjordan <mjordan@lunkwill> + + * .version: Update for 13.4.0-rc1 + +2015-05-21 13:15 +0000 [69292a9f11] mjordan <mjordan@lunkwill> + + * .lastclean: Update for 13.4.0-rc1 + +2015-05-21 13:15 +0000 [628680803a] mjordan <mjordan@lunkwill> + + * realtime: Add database scripts for 13.4.0-rc1 + +2015-05-21 13:05 +0000 [9d8a462356] Matt Jordan <mjordan@digium.com> + + * ARI: Update version to 1.7.0 + + This patch updates the version of ARI to 1.7.0 to reflect the backwards + compatible changes that will be introduced in 13.4.0. + + Change-Id: I6c36e6144da426412f25828a868e4df916bff60a + +2015-05-21 07:22 +0000 [620054c527] Matt Jordan <mjordan@digium.com> + + * Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 13 +2015-05-21 07:21 +0000 [f5e195b44e] Matt Jordan <mjordan@digium.com> + + * Merge "Logger: Reset defaults before processing config." into 13 +2015-05-21 07:20 +0000 [e8a4e01c32] Matt Jordan <mjordan@digium.com> + + * Merge "res/res_http_websocket: Add a pre-session established callback" into 13 +2015-05-21 05:15 +0000 [3c98544543] Joshua Colp <jcolp@digium.com> + + * Merge "main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits" into 13 +2015-05-20 20:53 +0000 [9b6e228419] Corey Farrell <git@cfware.com> + + * Logger: Reset defaults before processing config. + + Reset options to default values before reloading config. This ensures + that if a setting is removed or commented out of the configuration file + it is unset on reload. + + ASTERISK-25112 #close + Reported by: Corey Farrell + + Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd + +2015-05-20 19:05 +0000 [7fcf0a97b8] George Joseph <george.joseph@fairview5.com> + + * app_playback: Suppress warnings on playback if channel hung up + + If a channel hangs up while an audio file is playing, there's + no need to clutter up the logs with a warning so suppress it + if ast_check_hangup returns true. + + Also, change warning to debug/2 in file.c if writing a frame + fails. Same reasoning. + + Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> + +2015-05-14 15:21 +0000 [b1e8c0b9eb] Kevin Harwell <kharwell@digium.com> + + * audiohook.c: Difference in read/write rates caused continuous buffer resets + + Currently, everytime a sample rate change occurs (on read or write) the + associated factory buffers are reset. If the requested sample rate on a + read differed from that of a write then the buffers are continually reset + on every read and write. This has the side effect of emptying the buffer, + thus there being no data to read and then write to a file in the case of + call recording. + + This patch fixes it so that an audiohook_list's rate always maintains the + maximum sample rate among hooks and formats. Audiohook sample rates are + only overwritten by this value when slin native compatibility is turned on. + Also, the audiohook sample rate can only overwrite the list's sample rate + when its rate is greater than that of the list or if compatibility is + turned off. This keeps the rate from constantly switching/resetting. + + ASTERISK-24944 #close + Reported by: Ronald Raikes + + Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f + +2015-05-20 15:22 +0000 [4a450f863b] Matt Jordan <mjordan@digium.com> + + * Merge "Fix potential crash after unload of func_periodic_hook or test_message." into 13 +2015-05-19 13:01 +0000 [17d6ede337] Corey Edwards <tensai@zmonkey.org> + + * main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits + + ASTERISK-24887 #close + Reported by: Makoto Dei + Tested by: tensai + + Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf + +2015-05-13 09:55 +0000 [31cc24aad6] Matt Jordan <mjordan@digium.com> + + * res/res_http_websocket: Add a pre-session established callback + + This patch updates http_websocket and its corresponding implementation + with a pre-session established callback. This callback allows for + WebSocket server consumers to be notified when a WebSocket connection is + attempted, but before we accept it. Consumers can choose to reject the + connection, if their application specific logic allows for it. + + As a result, this patch pulls out the previously private + websocket_protocol struct and makes it public, as + ast_websocket_protocol. In order to preserve backwards compatibility + with existing modules, the existing APIs were left as-is, and new APIs + were added for the creation of the ast_websocket_protocol as well as for + adding a sub-protocol to a WebSocket server. + + In particular, the following new API calls were added: + * ast_websocket_add_protocol2 - add a protocol to the core WebSocket + server + * ast_websocket_server_add_protocol2 - add a protocol to a specific + WebSocket server + * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. + Consumers can populate this with whatever callbacks they wish to + support, then add it to the core server or a specified server. + + ASTERISK-24988 + Reported by: Joshua Colp + + Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2 + +2015-05-14 22:05 +0000 [f9114179e6] snuffy <snuffy22@gmail.com> + + * chan_pjsip: Fix crash during off-nominal when no endpoint specified. + + Add missing return -1 when no endpoint name is specified. + + ASTERISK-25086 #close + Reported by: snuffy + + Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e +2015-05-14 18:01 +0000 [dd78ab42e4] George Joseph <george.joseph@fairview5.com> + + * res_pjsip_config_wizard/config: Fix template processing + + The config wizard was always pulling the first occurrence of + a variable from an ast_variable list but this gets the template + value from the list instead of any overridden value. This patch + creates ast_variable_find_last_in_list() in config.c and updates + res_pjsip_config_wizard to use it instead of + ast_variable_find_in_list. Now the overridden values, where they + exist, are used instead of template variables. + + Updated test_config to test the new API. + + ASTERISK-25089 #close + + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> + Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4 + +2015-05-15 01:54 +0000 [091b436007] snuffy <snuffy22@gmail.com> + + * cdr: Fix 'core show channel' CDR variable truncation. + + When the new Bridging API was implemented, the workspace variable + changed to a malloc'd string, causing sizeof() to always be 8 (char). + + Revert back to stored on stack string for workspace. + + ASTERISK-25090 #close + + Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7 +2015-05-14 15:20 +0000 [8697a49ef9] Joshua Colp <jcolp@digium.com> + + * Merge "sorcery: Add API to insert/remove a wizard to/from an object type's list" into 13 +2015-05-14 15:19 +0000 [aea349a87e] Joshua Colp <jcolp@digium.com> + + * Merge "Message.c: Clear message channel frames on cleanup" into 13 +2015-05-14 00:06 +0000 [6b7282ca40] Corey Farrell <git@cfware.com> + + * Fix potential crash after unload of func_periodic_hook or test_message. + + These modules save a pointer to the context they create on load, and + use that pointer to destroy the context at unload. It is not safe + to save this pointer, it is replaced during load of pbx_config, + pbx_lua or pbx_ael. + + This change causes the modules to pass NULL to ast_context_destroy, + a safer way to perform the unregistration since it does not use + a pointer that could become invalid. + + ASTERISK-25085 #close + Reported by: Corey Farrell + + Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835 +2015-05-14 05:02 +0000 [8f8d54a18e] Joshua Colp <jcolp@digium.com> + + * Merge "main/manager.c: Bugfix sort action_manager by alphabetically" into 13 +2015-05-13 15:41 +0000 [02c5130589] Jonathan Rose <jrose@digium.com> + + * Message.c: Clear message channel frames on cleanup + + The message channel is a special channel that doesn't actually process frames. + However, certain actions can cause frames to be placed in the channel's read + queue including the Hangup application which is called on the channel after + each message is processed. Since the channel will continually be reused for + many messages, it's necessary to flush these frames at some point. + + ASTERISK-25083 #close + Reported by: Jonathan Rose + + Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f + +2015-05-13 15:44 +0000 [586da882bc] Joshua Colp <jcolp@digium.com> + + * Merge "app_voicemail: fix moving when old messages full" into 13 +2015-05-12 17:45 +0000 [d49d64b79c] Jonathan Rose <jrose@digium.com> + + * app_voicemail: fix moving when old messages full + + When completing voicemail playback of a message in the 'INBOX', the + message gets moved to the 'Old' messages folder. Without this patch, if + the 'Old' folder is already at its set limit, then the 'INBOX' message will + simply be deleted. With this patch, the flag to delete the message will be + removed if the save_to_folder function indicates that the message could + not be moved due to a full folder. + + ASTERISK-25082 #close + Reported by: Jonathan Rose + Review: https://gerrit.asterisk.org/#/c/448/ + + Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f +2015-05-13 14:20 +0000 [51478575e4] Joshua Colp <jcolp@digium.com> + + * Merge "General: Fix recent menuselect-related cross compile regression" into 13 +2015-05-13 12:26 +0000 [5fcaf727cc] Joshua Colp <jcolp@digium.com> + + * Merge "res_config_mysql: Fix broken column type checking" into 13 +2015-05-13 12:24 +0000 [6a12b0634b] Joshua Colp <jcolp@digium.com> + + * Merge "chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision." into 13 +2015-05-04 20:11 +0000 [9b13536fed] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * main/manager.c: Bugfix sort action_manager by alphabetically + + Fix the alphabetic order added on ast_manager_register_struct. The order + for struct manager_action added is not working, this change fixes the + problem. + + Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b + +2015-05-08 18:01 +0000 [e67e8d5c7f] Alexandre Fournier <alexandre.fournier@kiplink.fr> + + * res_config_mysql: Fix broken column type checking + + MySQL configuration engine contains a bug in require_mysql(). This + function is used for column type checking in tables. This bug only + affects DATETIME, DATE and FLOAT types. + + It came from mixing the first condition (switch-case-like + if/then/else), to check the expected column type, with the second + condition, to check the actual column type against the expected column + type. Both conditions must be checked separately in order to avoid the + execution of the wrong block. + + ASTERISK-18252 #comment This patch might fix the issue + Reported by: Gareth Blades + + ASTERISK-25041 #close + Reported by: Alexandre Fournier + Tested by: Alexandre Fournier + + Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa + +2015-05-10 07:37 +0000 [16f602f5c2] Yousf Ateya <y.ateya@starkbits.com> + + * res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS. + + First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC + https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values. + + Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31 + +2015-05-13 04:35 +0000 [62422712f7] Joshua Colp <jcolp@digium.com> + + * Merge "cdr_pgsql: Use PQescapeStringConn for escaping names." into 13 +2015-05-12 17:34 +0000 [c780b6e431] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision. + + If an ISDN call is hungup by both sides at the same time a crash could + happen. + + * Added missing NULL checks for the owner channel after calling + pri_queue_pvt_cause_data() in two places. Code after those calls need to + check the owner channel pointer for NULL before use because + pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the + owner and the owner may get hung up. + + ASTERISK-21893 #close + Reported by: Alexandr Gordeev + + Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a + +2015-05-10 02:26 +0000 [6627de830b] Sebastian Kemper <sebastian_ml@gmx.net> + + * General: Fix recent menuselect-related cross compile regression + + MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the + target platform. But menuselect is to be run on the build system, so + BUILD_CC needs to be used instead - like it was in the past, before the + recent changes (https://reviewboard.asterisk.org/r/4370/). This is the + patch for ASTERISK-25074. + + ASTERISK-25074 #close + Reported by: Sebastian Kemper + Tested by: Sebastian Kemper + + Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8 + +2015-05-05 15:32 +0000 [637c8f065e] George Joseph <george.joseph@fairview5.com> + + * sorcery: Add API to insert/remove a wizard to/from an object type's list + + Currently you can 'apply' a wizard to an object type but the wizard + always goes at the end of the object type's wizard list. This patch + adds a new ast_sorcery_insert_wizard_mapping function that allows + you to insert a wizard anyplace in the list. I.E. You could + add a caching wizard to an object type and place it before all + wizards. + + ast_sorcery_get_wizard_mapping_count and + ast_sorcery_get_wizard_mapping were added to allow examination + of the mapping list. + + ast_sorcery_remove_mapping was added to remove a mapping by name. + + As part of this patch, the object type's wizard list was converted + from an ao2_container to an AST_VECTOR_RW. + + A new test was added to test_sorcery for this capability. + + ASTERISK-25044 #close + + Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57 + +2015-05-12 01:31 +0000 [3cdb7950f0] Corey Farrell <git@cfware.com> + + * Fix processing of asterisk.conf debug=yes. + + The code which reads asterisk.conf supports processing the debug + option with ast_true, but ast_true returns -1. This causes debug + to still be off, convert to 1 so debug will be on as requested. + + ASTERISK-25042 + Reported by: Corey Farrell + + Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6 + +2015-05-01 23:43 +0000 [6553a00770] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * cdr_pgsql: Use PQescapeStringConn for escaping names. + + Use function PQescapeStringConn for escaping the name + of the table and schema instead of doing it manually. + + Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599 + +2015-05-12 05:38 +0000 [8523a5ed09] Joshua Colp <jcolp@digium.com> + + * Merge "vector: Add REMOVE, ADD_SORTED and RESET macros" into 13 +2015-05-09 16:58 +0000 [ea917fefaf] George Joseph <george.joseph@fairview5.com> + + * vector: Add REMOVE, ADD_SORTED and RESET macros + + Based on feedback from Corey Farrell and Y Ateya, a few new + macros have been added... + + AST_VECTOR_REMOVE which takes a parameter to indicate if + order should be preserved. + + AST_VECTOR_ADD_SORTED which adds an element to + a sorted vector. + + AST_VECTOR_RESET which cleans all elements from the vector + leaving the storage intact. + + Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14 + +2015-05-11 07:07 +0000 [d5864a358c] Ivan Poddubny <ivan.poddubny@gmail.com> + + * pbx/pbx_spool: Fix issue when call files were executed too early + + pbx_spool used to delete/move the call file upon successful outgoing + call completion, but did not delete it from in-memory list of files + (dirlist, used only when compiled with inotify/kqueue support). + That resulted in an extra attempt to process that filename after + retrytime seconds. + Then, if a new file with the same name appears that is scheduled + in future further than the completed one plus its retrytime, + then it gets executed earlier than expected. + + This patch fixes remove_from_queue function to also remove the entry + from the dirlist. + + ASTERISK-17069 #close + Reported by: Jeremy Kister + + ASTERISK-24442 #close + Reported by: tootai + + Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b + +2015-05-08 14:47 +0000 [4dbd4021c9] Rusty Newton <rnewton@digium.com> + + * configs/basic-pbx: Modified main IVR to play new Allison prompt. + + The main IVR was playing demo-congrats. I've switched it over to the + basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt + has Allison prompting the user with the actual IVR menu. + + ASTERISK-24892 #close + + Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d + +2015-05-08 15:55 +0000 [7111ba6df4] Matt Jordan <mjordan@digium.com> + + * Merge "tcptls: Avoiding ERR_remove_state in OpenSSL." into 13 +2015-05-08 10:39 +0000 [613a461c3d] Sean Bright <sean@malleable.com> + + * res_rtp_asterisk: Issue ERROR if res_srtp is not found. + + While trying to get WebRTC working with chan_pjsip, I was running + into the following error: + + Attempted to set an invalid DTLS-SRTP configuration on RTP + instance... + + Josh helpfully pointed out that res_srtp.so might not be loaded, and + sure enough, it wasn't. This patch adds a ERROR indiciating as much + to hopefully help others having a similar problem. + + Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f + +2015-05-07 17:49 +0000 [394fcb5eab] Rusty Newton <rnewton@digium.com> + + * sounds: Add Swedish sounds to Makefile and XML + + Added the necessary lines to the Makefile and sounds.xml so we'll have the + Swedish sounds in all available formats in menuselect. + + See also: Swedish sounds were added into the core sounds release 1.4.27. + + ASTERISK-24744 #close + + Reported by: Tove Hjelm + Tested by: Rusty Newton + + Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908 + +2015-05-08 09:54 +0000 [30c3b254c5] Joshua Colp <jcolp@digium.com> + + * Merge "doc: Make progdocs play nice with git" into 13 +2015-05-05 11:35 +0000 [2115f11b54] Alexander Traud (License 6520) + + * tcptls: Avoiding ERR_remove_state in OpenSSL. + + ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by + ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were + called by SSL_load_error_strings already and got removed. These changes allow + OpenSSL forks like BoringSSL to be used with Asterisk. + + ASTERISK-25043 #close + Reported by: Alexander Traud + patches: + asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520) + + Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629 + (cherry picked from commit 247fef66537b59649e7571d64e2c574a106dbd65) + +2015-05-07 14:54 +0000 [5392e970d0] George Joseph <george.joseph@fairview5.com> + + * doc: Make progdocs play nice with git + + Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in + + Changed /Makefile to copy asterisk-ng-doxygen.in to + asterisk-ng-doxygen then modify it with version instead of + modifying asterisk-ng-doxygen directly. Updated clean + targets as well. + + Updated /.gitignore and doc/.gitignore. + + Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622 + +2015-05-07 15:10 +0000 [1e44d1bef9] Joshua Colp <jcolp@digium.com> + + * Merge "res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination" into 13 +2015-05-04 14:43 +0000 [608f0a94ee] Ivan Poddubny <ivan.poddubny@gmail.com> + + * contrib/editors: Fix vim syntax highlighting of comments in config files + + * Added a lookbehind to one-line comment matcher to skip escaped + semicolons. + * Added support for block comments. + + Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7 + +2015-05-07 13:30 +0000 [22c6c12af2] Matt Jordan <mjordan@digium.com> + + * Merge "vector: Additional enhancements and fixes" into 13 +2015-05-06 13:24 +0000 [d649d682c4] Joshua Colp <jcolp@digium.com> + + * res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination + + The res_pjsip_exten_state module currently has a race condition between + processing the extension state callback from the PBX core and processing + the subscription shutdown callback from res_pjsip_pubsub. There is currently + no synchronization between the two. This can present a problem as while + the SIP subscription will remain valid the tree it points to may not. + This is in particular a problem as a task to send a NOTIFY may get queued + which will try to use the tree that may no longer be valid. + + This change does the following to fix this problem: + + 1. All access to the subscription tree is done within the task that + sends the NOTIFY to ensure that no other thread is modifying or + destroying the tree. This task executes on the serializer for the + subscriptions. + + 2. A reference to the subscription serializer is kept to ensure it + remains valid for the lifetime of the extension state subscription. + + 3. The NOTIFY task has been changed so it will no longer attempt + to send a NOTIFY if the subscription has already been terminated. + + ASTERISK-25057 #close + Reported by: Matt Jordan + + Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643 + +2015-05-07 07:02 +0000 [9322bc6ff6] Matt Jordan <mjordan@digium.com> + + * Merge "chan_dahdi: Improve force_restart_unavailable_chans option description." into 13 +2015-05-07 06:39 +0000 [b1514362ef] Matt Jordan <mjordan@digium.com> + + * Merge "res_stasis_snoop: Spying on a single direction continually increases CPU" into 13 +2015-05-07 06:28 +0000 [652ee2ff83] Joshua Colp <jcolp@digium.com> + + * Merge "features: Fix crash when transferee hangs up during DTMF attended transfer." into 13 +2015-05-05 20:22 +0000 [5f9aea8e3c] George Joseph <george.joseph@fairview5.com> + + * vector: Additional enhancements and fixes + + After using the new vector stuff for real I found... + + A bug in AST_VECTOR_INSERT_AT that could cause a seg fault. + + The callbacks needed to be closer to ao2_callback in behavior + WRT to CMP_MATCH and CMP_STOP behavior and the ability to return + a vector of matched entries. + + A pre-existing issue with APPEND and REPLACE was also fixed. + + I also added a new macro to test.h that acts like ast_test_validate + but also accepts a return code variable and a cleanup label. As well + as printing the error, it sets the rc variable to AST_TEST_FAIL and + does a goto to the specified label on error. I had a local version + of this in test_vector so I just moved it. + + ASTERISK-25045 + + Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc + +2015-05-04 17:28 +0000 [68513e00f7] Kevin Harwell <kharwell@digium.com> + + * res_stasis_snoop: Spying on a single direction continually increases CPU + + Creating a snoop channel in ARI and spying only on a single direction (in or + out) results in CPU utilization continually increasing until the CPU is fully + consumed. This occurs because frames are being put in the opposing direction's + slin factory queue, but not being removed. + + Fixed the problem by always reading and disposing of frames from the opposite + queue of the direction selected. + + ASTERISK-24938 #closes + + Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60 +2015-05-06 16:00 +0000 [904f5d98f6] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi: Improve force_restart_unavailable_chans option description. + + ASTERISK-25034 + Reported by: Richard Mudgett + + Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30 + +2015-05-06 07:42 +0000 [d6ffbe39b0] Joshua Colp <jcolp@digium.com> + + * Merge "app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter" into 13 +2015-05-06 06:13 +0000 [dfb292ce3e] Matt Jordan <mjordan@digium.com> + + * Merge "res_ari_bridges: Add missing dependencies." into 13 +2015-05-05 21:05 +0000 [50e90f9121] Matt Jordan <mjordan@digium.com> + + * Merge "pbx_config: Register manager actions with module version of macro." into 13 +2015-05-05 18:17 +0000 [be1260a35f] Richard Mudgett <rmudgett@digium.com> + + * features: Fix crash when transferee hangs up during DTMF attended transfer. + + A crash happens with this sequence of steps: + 1) Party A is connected to party B. + 2) Party B starts a DTMF attended transfer. + 3) Party A hangs up while party B is dialing party C. + + When party A hangs up the bridge that party A and party B are in is + dissolved and party B is kicked out of the bridge. When party B finishes + dialing party C he attempts to move to the new bridge with party C. Since + party B is no longer in a bridge the attempted move dereferences a NULL + bridge_channel pointer and crashes. + + * Made the hold(), unhold(), ringing(), and the bridge_move() functions + tolerant of the channel not being in a bridge. The assertion that party B + is always in a bridge is not true if the bridged peer of party B hangs up + and dissolves the bridge. Being tolerant of not being in a bridge allows + the peer hangup stimulus to be processed by the FSM. + + * Made the bridge_move() function return void since where the return value + for a failed move was checked generated a FSM coding ERROR message for a + normal off-nominal condition. + + * Eliminated most uses of RAII_VAR in bridge_basic.c. + + ASTERISK-25003 #close + Reported by: Artem Volodin + + Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada + +2015-05-05 15:40 +0000 [8b0f85ac06] George Joseph <george.joseph@fairview5.com> + + * test_vector: Fix build breakage caused by ASTERISK_REGISTER_FILE + + My 13 version of test_vector had an ASTERISK_REGISTER_FILE() macro + call at the top which is only supported in master. Once removed + builds are successful. + + Change-Id: I7cac8b669bed6de543bbf4e2eec3cffc9741acdd + +2015-05-05 14:48 +0000 [87263b47b5] Ivan Poddubny <ivan.poddubny@gmail.com> + + * app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter + + This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3 + parameters: position, original position and waiting time. + + ASTERISK-25038 #close + Reported by: Etienne Lessard + + Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618 + +2015-05-05 13:13 +0000 [2d9081b5ec] Matt Jordan <mjordan@digium.com> + + * Merge "stasis: Fix dial masquerade datastore lifetime" into 13 +2015-05-05 12:45 +0000 [8ca25dfd7e] Matt Jordan <mjordan@digium.com> + + * Merge "vector: Traversal, retrieval, insert and locking enhancements" into 13 +2015-05-05 09:47 +0000 [366ea63438] Corey Farrell <git@cfware.com> + + * res_ari_bridges: Add missing dependencies. + + Missed this module in the previous commit. res_ari_bridges uses symbols + from res_stasis_playback and res_stasis_recording. + + ASTERISK-25027 #close + Reported by: Corey Farrell + + Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f + +2015-05-05 09:27 +0000 [69ae8cf0a4] Corey Farrell <git@cfware.com> + + * pbx_config: Register manager actions with module version of macro. + + Switch manager actions in pbx_config to use the registration macro that + passes the module pointer, allowing pbx_config reference to be bumped + while the manager actions run. + + ASTERISK-25061 #close + Reported by: Corey Farrell + + Change-Id: I422c50dd74814616ac10c5e9c6598a0b1bc2c44e + +2015-05-04 12:16 +0000 [181ae3b8d9] Joshua Colp <jcolp@digium.com> + + * stasis: Fix dial masquerade datastore lifetime + + A recent change went into Asterisk which added reference counts to the + channels stored in a dial masquerade datastore. Unfortunately this + included a reference to the caller in a dialing operation. While all + of the dialed targets have the datastore removed from them upon dialing + completion this did not occur for the caller, causing it to have a + reference to itself that could go never go away (as it depended on + the destruction of the datastore which only happened when the channel + was destroyed). This resulted in the caller channel remaining on the + system despite it having hung up. + + This change does the following to fix this issue: + + 1. The dial masquerade datastore is now removed from the caller upon + dialing completion, just like the dialed targets. + 2. Upon destruction of the caller all the dialed targets are also + removed from the dial masquerade datastore (just in case). + 3. The reference to the caller has been removed as it should not be + possible for the datastore to now be valid/useful after the lifetime + of the caller has ended. + + ASTERISK-25025 #close + + Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f + +2015-05-01 19:25 +0000 [7a7e9733c2] George Joseph <george.joseph@fairview5.com> + + * vector: Traversal, retrieval, insert and locking enhancements + + Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really + does replace not insert. The few users of AST_VECTOR_INSERT were + refactored. Because these are macros, there should be no ABI + compatibility issues. + + Added AST_VECTOR_INSERT_AT that actually inserts an element into the + vector at a specific index pushing existing elements to the right. + + Added AST_VECTOR_GET_CMP that can retrieve from the vector based + on a user-provided compare function. + + Added AST_VECTOR_CALLBACK function that will execute a function + for each element in the vector. Similar to ao2_callback and + ao2_callback_data functions although the vector callback can take + a variable number of arguments. This should allow easy migration + to a vector where a container might be too heavy. + + Added read/write locked vector and lock manipulation macros. + + Added unit tests. + + ASTERISK-25045 #close + + Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0 + +2015-05-03 13:55 +0000 [040d2f8558] Corey Farrell <git@cfware.com> + + * main/test.c: Add test to verify there were no registration errors. + + This adds a test that will fail if any test failed to register. Also fail + if any test registration produced a warning about missing a leading or + trailing slash. + + ASTERISK-25053 #close + Reported by: Corey Farrell + + Change-Id: I93e50b8fcbcfa7f1f5b41b2c44a51685c09529c3 + +2015-05-04 09:26 +0000 [626bffc4c2] Matt Jordan <mjordan@digium.com> + + * Merge "contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update" into 13 +2015-05-04 09:26 +0000 [87fb7fc165] Matt Jordan <mjordan@digium.com> + + * Merge "cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8" into 13 +2015-05-04 09:25 +0000 [81c27127aa] Matt Jordan <mjordan@digium.com> + + * Merge "Format Interfaces: Prevent unload except by shutdown." into 13 +2015-05-04 07:46 +0000 [743fed71fc] Matt Jordan <mjordan@digium.com> + + * Merge "res_odbc: Use negative connection cache for all connections" into 13 +2015-04-21 11:52 +0000 [3dcec04ab5] Martin Tomec <tomec.martin@gmail.com> + + * res_odbc: Use negative connection cache for all connections + + Apply the negative connection cache setting to all connections, + even those that are not pooled. This ensures that the connection + will not be re-established before the negative connection cache + time is met. + + ASTERISK-22708 #close + + Change-Id: I431cc2e8584ab0b6908b3523d0a0e18c9a527271 +2015-05-04 04:03 +0000 [74799b3fe2] Matt Jordan <mjordan@digium.com> + + * Merge "Remove unneeded uses of optional_api providers." into 13 +2015-05-04 04:03 +0000 [78c02f8e88] Matt Jordan <mjordan@digium.com> + + * Merge "Update configure.ac/Makefile for clang" into 13 +2015-05-03 21:03 +0000 [f38066fcad] Corey Farrell <git@cfware.com> + + * Format Interfaces: Prevent unload except by shutdown. + + Format interfaces cannot be unregistered, so the modules that provide them + need to be held open except by shutdown. + + ASTERISK-25054 #close + Reported by: Corey Farrell + + Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3 + +2015-05-03 20:28 +0000 [e76a6a97bf] Matt Jordan <mjordan@digium.com> + + * contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update + + The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755 + failed to add ENUM support for Postgres databases. This requires a + specific import from the sqlalchemy.dialects.postgresql package. This + patch corrects this error, which allows for Postgres update scripts to + be generated. + + ASTERISK-24706 + + Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015 + +2015-05-01 19:50 +0000 [92120247e9] D Tucny <d@tucny.com> + + * term: send proper reset sequence when black background is forced + + When using the force black background command-line option or configuration + option an invalid reset sequence is sent following a coloured output item + in the CLI, the result is that the colour is not 'turned off' and continues + until the next non-default coloured text output. + + A reset sequence is already defined in term.c, but the ast_term_reset + function doesn't use it, instead building it's own invalid sequence and + returning that. + + This patch changes that behaviour, removing the building of a reset sequence + and instead using the pre-built constant 'enddata' which is a suitable reset + sequence for this purpose. + + ASTERISK-24896 #close + Reported by: Dan Tucny + + Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43 + +2015-05-03 09:20 +0000 [13819a34c4] Matt Jordan <mjordan@digium.com> + + * Merge "Build System: Prevent unneeded changes to asterisk/buildopts.h." into 13 +2015-05-03 09:19 +0000 [b518ba1c6c] Matt Jordan <mjordan@digium.com> + + * Merge "res_pjsip_dlg_options: Fix MODULEINFO section." into 13 +2015-05-02 18:58 +0000 [ad6ea29697] Corey Farrell <git@cfware.com> + + * Remove unneeded uses of optional_api providers. + + A few cases exist where headers of optional_api provders are included but + not needed. This causes unneeded calls to ast_optional_api_use. + + * Don't include optional_api.h from sip_api.h. + * Move 'struct ast_channel_monitor' to channel.h. + * Don't include monitor.h from chan_sip.c, channel.c or features.c. + + The move of struct ast_channel_monitor is needed since channel.c depends on + it. This has no effect on users of monitor.h since channel.h is included + from monitor.h. + + ASTERISK-25051 #close + Reported by: Corey Farrell + + Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478 + +2015-05-02 10:19 +0000 [9888562c8c] Matt Jordan <mjordan@digium.com> + + * Merge "include/asterisk/channel.h: Fix typo" into 13 +2015-05-02 10:17 +0000 [b4000f2d44] Matt Jordan <mjordan@digium.com> + + * Merge "Astobj2: Fix initialization order of refdebug and AO2_DEBUG." into 13 +2015-04-30 02:07 +0000 [525c8c8689] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * include/asterisk/channel.h: Fix typo + + Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3 + +2015-05-02 02:15 +0000 [63196a8256] Corey Farrell <git@cfware.com> + + * res_pjsip_dlg_options: Fix MODULEINFO section. + + Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options. + This extra space prevented any of the dependencies from being seen by + menuselect, so building with default options would fail if PJSIP was + not installed. + + This also makes the tool that extracts information for menuselect + tolerant of multiple spaces in the future. + + ASTERISK-25033 #close + Reported by: Peter Whisker + + Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698 + +2015-04-29 03:03 +0000 [ac1f0090eb] Corey Farrell <git@cfware.com> + + * Build System: Prevent unneeded changes to asterisk/buildopts.h. + + * Add AST_DEVMODE to BUILDOPTS + * Remove CFLAGS that do not effect ABI from BUILDOPTS. + * Use BUILDOPTS to generate AST_BUILDOPT_SUM. + * Remove loop that defined AST_MODULE_* + + These changes ensure that only ABI effecting options are considered for + AST_BUILDOPT_SUM. This also reduces unneeded full system rebuilds caused + by enabling or disabling one module that another is dependent on. + + ASTERISK-25028 + Reported by: Corey Farrell + + Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2 + +2015-05-01 13:22 +0000 [5875bf183c] Corey Farrell <git@cfware.com> + + * Astobj2: Fix initialization order of refdebug and AO2_DEBUG. + + This ensures that refdebug is initialized before AO2_DEBUG if + both are enabled, since AO2_DEBUG allocates a container. + + This change also makes AO2_DEBUG initialization critical, a + failure will abort Asterisk startup. This is needed since + the failure would be caused by reg_containers allocation + failure, and that would result in a segmentation fault by + ao2_container_register later in startup. + + ASTERISK-25048 #close + Reported by: Corey Farrell + + Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244 + +2015-04-29 14:49 +0000 [1b19c15f17] Matt Jordan <mjordan@digium.com> + + * main/pbx: Improve performance of dialplan reloads with a large number of hints + + The PBX core maintains two hash tables for hints: a container of the + actual hints (hints), along with a container of devices that are watching that + hint (hintdevices). When a dialplan reload occurs, each hint in the hints + container is destroyed; this requires a lookup in the container of devices to + find the device => hint mapping object. In the current code, this performs an + ao2_callback, iterating over each of the device to hint objects in the + hintdevices container. For a large number of hints, this is extremely + expensive: dialplan reloads with 20000 hints could take several minutes + in just this phase. + + This patch improves the performance of this step in the dialplan reloads + by caching which devices are watching a hint on the hint object itself. + Since we don't want to create a circular reference, we just cache the + name of the device. This allows us to perform a smarter ao2_callback on + the hintdevices container during hint removal, hashing on the name of the + device and returning an iterator to the matching names. The overall + performance improvement is rather large, taking this step down to a number of + seconds as opposed to minutes. + + In addition, this patch also registers the hint containers in the PBX + core with the astobj2 library. This allows for reasonable debugging to + hash collisions in those containers. + + ASTERISK-25040 #close + Reported by: Matt Jordan + + Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360 + +2015-05-01 06:55 +0000 [ec0f80b6e8] Matt Jordan <mjordan@digium.com> + + * Merge "res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback." into 13 +2015-05-01 06:55 +0000 [ed51fbbe9c] Matt Jordan <mjordan@digium.com> + + * Merge "Prevent potential crash on blond transfer." into 13 +2015-04-30 15:54 +0000 [3efe0df044] Corey Farrell <git@cfware.com> + + * Sample Configs: Fix syntax error in pjsip.conf + + The sample pjsip.conf has a few comment lines that are missing the + semicolons at the start of the comment, causing the config to fail + load. + + Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352 + +2015-04-30 15:20 +0000 [077979618b] Mark Michelson <mmichelson@digium.com> + + * Prevent potential crash on blond transfer. + + Scenario: + Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects + the incoming call (or some other immediate circumstance causes Carol not + to answer the call) + + What occurs in this case is that when the bridge between Alice and Bob + breaks, Alice is told to masquerade into Bob's channel that had placed + the call to Carol. The actual masquerade goes down without a hitch. + However, a channel fixup callback that attempts to publish dial events + over Stasis has a crash. The reason for this crash is that the datastore + on Bob's channel that placed the outbound call to Carol only had a bare + pointer to Carol's channel. Since Carol rejected the incoming call, + Carol's channel has been hung up and freed, meaning accessing her + channel results in a crash. + + The fix here is simple. The dial fixup code has been altered to hold + references to the involved channels and to drop those references when + freeing data. + + ASTERISK-25025 #close + Reported by Chet Stevens + + Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197 + +2015-04-30 14:09 +0000 [4b8cddfb36] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback. + + The Asterisk 13 version of the fix for outbound registration was missing + a key component that set the outbound authenticator's callback that + creates an authenticated request based on an old request. This was + picked up by some outbound registration tests failing in the testsuite. + + Change-Id: I5ca9379698c606da36bc38eaffccedaf64211ce3 +2015-04-30 13:42 +0000 [415a0d0745] Joshua Colp <jcolp@digium.com> + + * res_ari_device_states: Fix dependency on res_stasis_device_state. + + The res_ari_device_states module depends on res_stasis_device_state, + not res_stasis_device_states. + + Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de + +2015-04-30 11:11 +0000 [e0c6f88010] Mark Michelson <mmichelson@digium.com> + + * Merge "chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option." into 13 +2015-04-30 10:53 +0000 [d1bc86fc99] Matt Jordan <mjordan@digium.com> + + * Merge "res_pjsip_outbound_registration: Add virtual line support." into 13 +2015-04-29 14:29 +0000 [d3c310a28c] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. + + Some telco switches occasionally ignore ISDN RESTART requests. The fix + for ASTERISK-19608 added an escape clause for B channels in the restarting + state if the telco ignores a RESTART request. If the telco fails to + acknowledge the RESTART then Asterisk will assume the telco acknowledged + the RESTART on the second call attempt requesting the B channel by the + telco. The escape clause is good for dealing with RESTART requests in + general but it does cause the next call for the restarting B channel to be + rejected if the telco insists the call must go on that B channel. + + chan_dahdi doesn't really need to issue a RESTART request in response to + receiving a cause 44 (Requested channel not available) code. Sending the + RESTART in such a situation is not required (nor prohibited) by the + standards. I think chan_dahdi does this for historical reasons to deal + with buggy peers to get channels unstuck in a similar fashion as the + chan_dahdi.conf resetinterval option. + + * Add the chan_dahdi.conf force_restart_unavailable_chans compatability + option that when disabled will prevent chan_dahdi from trying to RESTART + the channel in response to a cause 44 code. + + ASTERISK-25034 #close + Reported by: Richard Mudgett + + Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65 +2015-04-30 06:38 +0000 [7f611fa0e8] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8 + + This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR + columns added in Asterisk 1.8. The columns are: + * peeraccount + * linkedid + * sequence + When enabled, the columns in the database entry will be populated with the data + from the CDR. + + ASTERISK-24976 #close + + Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b + +2015-04-30 06:04 +0000 [e332c7ed5e] Joshua Colp <jcolp@digium.com> + + * res_pjsip_outbound_registration: Fix double unref on error return. + + When the PJSIP pjsip_regc_send function is invoked and an error + status returned the caller currently decrements the reference count + of the client state that it just incremented, assuming the + registration callback would not have been invoked. In practice + this is not correct. If the failure happens after the transaction + has been set up the callback will still be invoked. This will + cause the reference count to be incorrectly decremented twice, once + by the registration callback and second by the caller of + pjsip_regc_send. + + This change makes it so that whether the callback is invoked or + not is known by the caller of pjsip_regc_send. Depending on + this it can know whether it is responsible for decrementing the + reference count of the client state or not. + + ASTERISK-25037 #close + Reported by: Joshua Colp + + Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de + +2015-04-20 13:03 +0000 [9c3ed42875] Diederik de Groot <ddegroot@talon.nl> + + * Update configure.ac/Makefile for clang + + Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which + checks compiler requirements for RAII: + gcc: -fnested-functions support + clang: -fblocks (and if required -lBlocksRuntime) + The original check was implemented in configure.ac and now has it's + own file. This function also sets C_COMPILER_FAMILY to either gcc or + clang for use by makefile + + Created autoconf/ast_check_strsep_array_bounds.m4 (contains + AST_CHECK_STRSEP_ARRAY_BOUNDS): + which checks if clang is able to handle the optimized strsep & strcmp + functions (linux). If not, the standard libc implementation should be + used instead. Clang + the optimized macro's work with: + strsep(char *, char []), but not with strsepo(char *, char *). + Instead of replacing all the occurences throughout the source code, + not using the optimized macro version seemed easier + + See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h': + llvm-comment: Normally, this array-bounds warning are suppressed for + macros, so that unused paths like the one that accesses __s1[3] are + not warned about. But if you preprocess manually, and feed the + result to another instance of clang, it will warn about all the + possible forks of this particular if statement. Instead of switching + of this optimization, another solution would be to run the preproces- + sing step with -frewrite-includes, which should preserve enough + information so that clang should still be able to suppress the diag- + nostic at the compile step later on. + + See also "https://llvm.org/bugs/show_bug.cgi?id=20144" + See also "https://llvm.org/bugs/show_bug.cgi?id=11536" + + Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning + suppressions: + -Wno-unused-value + -Wno-parentheses-equality + In an earlier review (reviewboard: 4550 and 4554), they were deemed a + nuisace and less than benefitial. + + configure.ac: + Added AST_CHECK_RAII() see earlier + Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier + Removed moved content + + ASTERISK-24917 + Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb + +2015-04-29 16:43 +0000 [37a193da18] Matt Jordan <mjordan@digium.com> + + * Merge "ARI: Fix missing dependencies." into 13 +2015-04-29 16:42 +0000 [6a86b3555b] Matt Jordan <mjordan@digium.com> + + * Merge "res_fax: allow 2400 transmission rate according to v.27ter standard" into 13 +2015-04-29 16:15 +0000 [d4e207e27e] Matt Jordan <mjordan@digium.com> + + * main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8 + + The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS + structures in the RTP engine. However, when a 'core reload' is issued, a + double free of the memory pointed to by the char *'s in the DTLS + configuration struct can occur, as ast_rtp_dtls_cfg_free does not set + the pointers to NULL when they are freed. + + This patch sets those pointers to NULL, preventing a second call to + ast_rtp_dtls_cfg_free from corrupting memory. + + ASTERISK-25022 + + Change-Id: I820471e6070a37e3c26f760118c86770e12f6115 + +2015-04-29 13:05 +0000 [3fb6daeb55] Kevin Harwell <kharwell@digium.com> + + * res_fax: allow 2400 transmission rate according to v.27ter standard + + A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so + a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits + per second. This reverts all or some of those patches since according to the + v.27ter standard a rate of 2400 bits per second is also supported. + + One of the original patches also added 9600 bits per second support for v.27. + This patch also removes that since v.27ter only supports 2400/4800 bits per + second. + + Also, since Asterisk specifically supports v.27ter the enum was renamed to + better reflect this. + + ASTERISK-24955 #close + Reported by: Matt Jordan + + Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733 + +2015-04-29 10:46 +0000 [49ef81c15c] Joshua Colp <jcolp@digium.com> + + * res_sorcery_config: Fix build issue due to syntax error. + + Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac + +2015-04-28 00:29 +0000 [3278fe5327] Ashley Sanders <asanders@digium.com> + + * chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR + Sections Exist in pjsip.conf + + This patch modifies the current loading strategy of the pjsip configuration. If + duplicate sections (e.g. sections containing the same [id/type]) are defined in + [pjsip.conf], the loader will consider the configuration for the given type as + invalid when the duplicate section is encountered. The entire configuration + (including what was previously loaded) for the duplicate [id/type] sections + will be rejected and destroyed, an error message is logged and the load + processing for the given stops. + + ASTERISK-24996 + Reported By: Ashley Sanders + + Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef + +2014-11-04 06:03 +0000 [89f6719f7a] Joshua Colp <jcolp@digium.com> + + * res_pjsip_outbound_registration: Add virtual line support. + + Virtual line support establishes a relationship between messages + related to an outbound registration and a local endpoint. This is + accomplished by attaching a parameter to the Contact of the outbound + registration and looking for it on any received requests. If the + parameter exists and can be matched to an outbound registration + the configured endpoint is associated with the request. + + ASTERISK-24949 #close + Reported by: Joshua Colp + + Change-Id: I7df909d2625479110a83fdd354c21ac539e8615d + +2015-04-29 06:39 +0000 [d61f03c4f9] Corey Farrell <git@cfware.com> + + * ARI: Fix missing dependencies. + + ARI modules that are generated by 'make ari-stubs' are all dependent on + res_ari_model. Additionally some of the same modules depend on one or more + res_stasis_* modules. + + ASTERISK-25027 #close + Reported by: Corey Farrell + + Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153 + +2015-04-29 06:26 +0000 [3e4624ad21] Corey Farrell <git@cfware.com> + + * res_pjsip: Remove incorrect MODULEINFO from presence_xml.c. + + Remove incorrect MODULEINFO block and unneeded header includes + from presence_xml.c. + + ASTERISK-25027 + Reported by: Corey Farrell + + Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb + +2015-04-29 06:17 +0000 [fed9faab8d] Corey Farrell <git@cfware.com> + + * Git Migration: Create doc/rest-api when needed. + + Create the directory './doc/rest-api' at the start of 'make ari-stubs' + to prevent an error when documentation is generated. The directory is + also added to git ignores. + + ASTERISK-25027 + Reported by: Corey Farrell + + Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b + +2015-04-29 05:17 +0000 [df23c8a86b] Joshua Colp <jcolp@digium.com> + + * res_pjsip_outbound_registration: Fix build due to removal of transaction. + + Change-Id: I7a8a7beec3334cec304943f2dd7597eabe2e3150 + +2015-04-28 19:18 +0000 [95ab9fdb1a] Joshua Colp <jcolp@digium.com> + + * Merge "res_pjsip_outbound_registration: Add debugging messages." into 13 +2015-04-28 19:18 +0000 [0e70dc0dc8] Joshua Colp <jcolp@digium.com> + + * Merge "res_pjsip_outbound_registration: Don't fail on delayed processing: 13." into 13 +2015-04-27 16:56 +0000 [e39bd6ba46] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_outbound_registration: Don't fail on delayed processing: 13. + + This is the Asterisk 13 version of a change to master that allows for + registration responses to be processed successfully potentially after + the original transaction has timed out. The main difference between this + and the master change is that the master version has API changes that + are unacceptable for 13. For 13, this is worked around by adding a new + API call that the outbound registration code uses instead. + + The following is the text from the master version of this commit: + + Odd behaviors have been observed during outbound registrations. The most + common problem witnessed has been one where a request with + authentication credentials cannot be created after receiving a 401 + response. Other behaviors include apparently processing an incorrect SIP + response. + + Inspecting the code led to an apparent issue with regards to how we + handle transactions in outbound registration code. When a response to a + REGISTER arrives, we save a pointer to the transaction and then push a + task onto the registration serializer. Between the time that we save the + pointer and push the task, it's possible for the transaction to be + destroyed due to a timeout. It's also possible for the address to be + reused by the transaction layer for a new transaction. + + To allow for authentication of a REGISTER request to be authenticated + after the transaction has timed out, we now also hold a reference to the + original REGISTER request instead of the transaction. The function for + creating a request with authentication has been altered to take the + original request instead of the transaction where the original request + was sent. + + ASTERISK-25020 + Reported by Mark Michelson + + Change-Id: If1ee5f601be839479a219424f0358a229f358f7c +2015-04-27 14:44 +0000 [1bf008fc76] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_outbound_registration: Add debugging messages. + + When problems occur regarding outbound registrations, it currently + is difficult to debug. Most off-nominal paths had warning messages, + but sometimes we want to know what's going on before hitting the + off-nominal path. This patch adds lots of debugging output that + should give a clearer picture of what is happening with regards + to outbound registrations. + + ASTERISK-25020 + Reported by Mark Michelson + + Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45 + +2015-04-28 07:13 +0000 [7ee05892d6] Joshua Colp <jcolp@digium.com> + + * Merge "Example script for scan-build (the llvm static analyzer)" into 13 +2015-04-28 05:38 +0000 [0b6410c4f8] Steve Davies <steve@one47.co.uk> + + * res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS + + ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. + The resources are linked into a table, but the original alloc refs + are never released. ast_strdup leak in rtp_engine.c. If + ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, + a pointer to an alloc'd string is overwritten before the string is free'd. + + ASTERISK-25022 + Reported by: one47 + + Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b + +2015-04-28 06:55 +0000 [427209603d] Joshua Colp <jcolp@digium.com> + + * Merge "cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version" into 13 +2015-04-27 12:11 +0000 [99fb87ae13] George Joseph <george.joseph@fairview5.com> + + * res_pjsip: Fix SEGV on pending-qualify contacts + + Permanent contacts that hadn't been qualified yet were missing + their contact_status entries causing SEGVs when running CLI + commands. + + This patch makes sure that contact_statuses are created for + both dynamic and permanent contacts when they are created. + It also adds checks in the CLI code to make sure there's a + contact_status, just in case. + + ASTERISK-25018 #close + Reported-by: Ivan Poddubny + Tested-by: Ivan Poddubny + Tested-by: George Joseph + + Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029 + +2015-04-15 18:55 +0000 [d5dd43856e] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version + + Add new column to INSERT new columns added in cdr 1.8 version. The columns are: + * peeraccount + * linkedid + * sequence + This feature is configurable in cdr_odbc.conf using a new configuration + option, 'newcdrcolumns'. + + ASTERISK-24976 #close + + Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127 +2015-04-26 17:21 +0000 [e9788056e9] Matt Jordan <mjordan@digium.com> + + * channels/chan_skinny: Fix compilation error introduced in f8e21a1adf + + A typo in commit f8e21a1adf resulted in a compilation error in + chan_skinny. This patch fixes the typo. + + ASTERISK-24917 + + Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c + +2015-04-26 15:53 +0000 [2d277996b7] Matt Jordan <mjordan@digium.com> + + * Merge "Clang: Fix some more tautological-compare warnings." into 13 +2015-04-24 13:07 +0000 [145f65598c] Matt Jordan <mjordan@digium.com> + + * Merge "app_confbridge: Default the template option to a compatible default profile." into 13 +2015-04-23 15:11 +0000 [7e5056b393] Kevin Harwell <kharwell@digium.com> + + * app_confbridge: Default the template option to a compatible default profile. + + Confbridge dynamic profiles did not have a default profile unless you + explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a + template was not set prior to the bridge being created then some + options were left with no default values set. This patch makes it so + the default templates are set to the default bridge and user profiles. + + ASTERISK-24749 #close + Reported by: philippebolduc + + Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a + +2015-04-24 09:17 +0000 [1da9ec969d] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_outbound_authenticator: Increase CSeq on authed requests. + + The way PJSIP generates an authenticated request is to use a previous + request as a template. This means that the authenticated request will + have the same Call-ID, From header (including tag), and CSeq as the + original request. PJSIP generates a new branch on the Via header to + indicate that this is a new transaction, though. + + There are some SIP implementations, though, that do not notice the + change in the branch and therefore will match the authed request to the + original request's transaction. Since the CSeq is the same, the server + will repeat the response it sent to the original request. + + This patch aids interoperability by increasing the CSeq of the authed + request by one. + + ASTERISK-24845 #close + Reported by: Carl Fortin + Tested by: Carl Fortin + + Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01 + +2015-04-24 09:24 +0000 [bf3d9db4a6] Matt Jordan <mjordan@digium.com> + + * Merge "res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX." into 13 +2015-04-20 13:06 +0000 [cb318f3960] Diederik de Groot <ddegroot@talon.nl> + + * Example script for scan-build (the llvm static analyzer) + + - Added Pre-amble (Options / Flags / Usage Example / GNU License) + - Extended Configurability + - Made Executable + + ASTERISK-24917 + Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8 + +2015-04-23 17:23 +0000 [b3cd5bc77f] Mark Michelson <mmichelson@digium.com> + + * Merge "Clang: change previous tautological-compare fixes." into 13 +2015-04-23 12:54 +0000 [eabf3b5a3c] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX. + + When Asterisk originates a channel to an application, the channel is + hung up once the application finishes executing. When the application + in question is SendFax, the Asterisk PJSIP code will attempt to reinvite + the T.38 session to audio after the FAX completes. The hangup of the + channel happens in the midst of this reinvite transaction. In most + circumstances, this works out okay because the BYE is delayed until the + reinvite transaction can complete. + + However, if the reinvite that Asterisk sends receives a 401/407 + response, then Asterisk's attempt to re-send the reinvite with + authentication will fail. This is because the session supplement in + res_pjsip_t38 makes the assumption that the channel on the session will + always be non-NULL. Since the channel has been hung up, though, the + channel is now NULL. Attempting to operate on the channel causes a + crash. + + This patch fixes the issue by ensuring that the channel on the session + is not NULL before attempting to mess with the T.38 framehook. + + This patch also contains some corrections for comments that were + incorrect and really confused me when I first started looking at the + code. + + ASTERISK-25004 #close + Reported by Mark Michelson + + Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0 +2015-04-23 09:16 +0000 [f70d21b2cf] George Joseph <george.joseph@fairview5.com> + + * res_pjsip: Validate that contact uris start with sip: or sips: + + Currently we use pjsip_parse_hdr to validate contact uris but it + appears that it allows uris without a scheme if there's a port + supplied. I.E myexample.com will fail but myexample.com:5060 will + pass even though it has no scheme. This causes SEGVs later on + whenever the uri is used. + + To prevent this, permanent_contact_validate has been updated to check + that the scheme is either 'sip' or 'sips'. + + 2 uses of possibly-null endpoint have also been fixed in + create_out_of_dialog_request. + + ASTERISK-24999 + + Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 + Reported-by: Brad Latus + +2015-04-23 08:00 +0000 [1bb16bedc7] Diederik de Groot <ddegroot@talon.nl> + + * Clang: change previous tautological-compare fixes. + + clang can warn about a so called tautological-compare, when it finds + comparisons which are logically always true, and are therefor deemed + unnecessary. + + Exanple: + unsigned int x = 4; + if (x > 0) // x is always going to be bigger than 0 + + Enum Case: + Each enumeration is its own type. Enums are an integer type but they + do not have to be *signed*. C leaves it up to the compiler as an + implementation option what to consider the integer type of a particu- + lar enumeration is. Gcc treats an enum without negative values as + an int while clang treats this enum as an unsigned int. + + rmudgett & mmichelson: cast the enum to (unsigned int) in assert. + The cast does have an effect. For gcc, which seems to treat all enums + as int, the cast to unsigned int will eliminate the possibility of + negative values being allowed. For clang, which seems to treat enums + without any negative members as unsigned int, the cast will have no + effect. If for some reason in the future a negative value is ever + added to the enum the assert will still catch the negative value. + + ASTERISK-24917 + + Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a + +2015-04-23 06:50 +0000 [a06924e9d9] Matt Jordan <mjordan@digium.com> + + * Merge "Astobj2: Ensure all calls to __adjust_lock pass a valid object." into 13 +2015-04-22 16:22 +0000 [1474bb05f6] George Joseph <george.joseph@fairview5.com> + + * res_corosync: Add check for config file before calling corosync apis + + On some systems, res_corosync isn't compatible with the installed version of + corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE, + and Asterisk terminates. The work around has been to remember to add + res_corosync as a noload in modules.conf. A better solution though is to have + res_corosync check for its config file before attempting to call corosync apis + and return LOAD_DECLINE if there's no config file. This lets Asterisk loading + continue. + + If you have a res_corosync.conf file and res_corosync fails, you get the same + behavior as today and the fatal error tells you something is wrong with the + install. + + ASTERISK-24998 + + Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889 +2015-04-22 15:17 +0000 [73efb093b8] Corey Farrell <git@cfware.com> + + * Astobj2: Ensure all calls to __adjust_lock pass a valid object. + + __adjust_lock doesn't check for invalid objects, and doesn't have an + appropriate return value for invalid objects. Most callers of + __adjust_lock pass objects that have already been confirmed valid, + this change adds checks before the remaining calls. + + ASTERISK-24997 #close + Reported by: Corey Farrell + + Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f + +2015-04-22 16:32 +0000 [b0e929219b] George Joseph <george.joseph@fairview5.com> + + * .gitignore: Add .gcno and .gcda + + Products of --enable-coverage + + Change-Id: Ie20882d64b60692e2c941ea8872ab82a86ce77a3 + +2015-04-22 14:25 +0000 [5a3948a66f] Matt Jordan <mjordan@digium.com> + + * Merge "Fix/Update clang-RAII macro implementation" into 13 +2015-04-22 14:07 +0000 [2ef1e1fc68] Mark Michelson <mmichelson@digium.com> + + * Merge "res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers." into 13 +2015-04-22 04:17 +0000 [d6dfc85666] Diederik de Groot <ddegroot@talon.nl> + + * Clang: Fix some more tautological-compare warnings. + + clang can warn about a so called tautological-compare, when it finds + comparisons which are logically always true, and are therefor deemed + unnecessary. + + Exanple: + unsigned int x = 4; + if (x > 0) // x is always going to be bigger than 0 + + Enum Case: + Each enumeration is its own type. Enums are an integer type but they + do not have to be *signed*. C leaves it up to the compiler as an + implementation option what to consider the integer type of a particu- + lar enumeration is. Gcc treats an enum without negative values as + an int while clang treats this enum as an unsigned int. + + rmudgett & mmichelson: cast the enum to (unsigned int) in assert. + The cast does have an effect. For gcc, which seems to treat all enums + as int, the cast to unsigned int will eliminate the possibility of + negative values being allowed. For clang, which seems to treat enums + without any negative members as unsigned int, the cast will have no + effect. If for some reason in the future a negative value is ever + added to the enum the assert will still catch the negative value. + + ASTERISK-24917 + Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62 + +2015-04-22 05:45 +0000 [edd9e54818] Joshua Colp <jcolp@digium.com> + + * Merge "Check for ao2_alloc failure in __ast_channel_internal_alloc." into 13 +2015-04-14 14:04 +0000 [7b57116833] Joshua Colp <jcolp@digium.com> + + * res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers. + + Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon + a mailbox state change (such as a new message being left, or one being deleted). + In practice this is not sufficient to keep clients aware of the current MWI status. + + This change makes the module send unsolicited MWI NOTIFY on startup so that + clients are guaranteed to have the most up to date MWI information. It also makes + clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware + of the current MWI status they receive it. + + ASTERISK-24982 #close + Reported by: Joshua Colp + + Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58 + +2015-04-22 05:29 +0000 [4423d5f755] Joshua Colp <jcolp@digium.com> + + * Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs." into 13 +2015-04-21 15:17 +0000 [ad1a118632] Corey Farrell <git@cfware.com> + + * Check for ao2_alloc failure in __ast_channel_internal_alloc. + + Fix a crash that could occur in __ast_channel_internal_alloc if + ao2_alloc fails. + + ASTERISK-24991 #close + + Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90 + +2015-04-20 14:30 +0000 [3327560cb2] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. + + When SUBSCRIBE dialogs were established, we never associated + the endpoint that created the subscription with the dialog + we end up creating. In most cases, this ended up not causing + any problems. + + The actual bug that was observed was that when a device that + was behind NAT established a subscription with Asterisk, Asterisk + would end up sending in-dialog NOTIFY requests to the device's + private IP addres instead of the public address of the NAT router. + + When Asterisk receives the initial SUBSCRIBE from the device, + res_pjsip_nat rewrites the contact to the public address on which the + SUBSCRIBE was received. This allows for the dialog to have its target + address set to the proper public address. Asterisk then would send a 200 + OK response to the SUBSCRIBE, then a NOTIFY with the initial + subscription state. The device would then send a 200 OK response to + Asterisk's NOTIFY. + + Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat + did not rewrite the address in the Contact header. Then, when the PJSIP + dialog layer processed the 200 OK, PJSIP would perform a comparison + between the IP address in the Contact header and its saved target + address for the dialog. Since they differed, PJSIP would update the + target dialog address to be the address in the Contact header. From this + point, if Asterisk needed to send a NOTIFY to the device, the result was + that the NOTIFY would be sent to the private address that the device + placed in the Contact header. + + The reason why res_pjsip_nat did not rewrite the address when it + received the 200 OK response was that it could not associate the + incoming response with a configured endpoint. This is because on a + response, the only way to associate the response to an endpoint is by + finding the dialog that the response is associated with and then finding + the endpoint that is associated with that dialog. We do not perform + endpoint lookups on responses. res_pjsip_pubsub skipped the step of + associating the endpoint with the dialog we created, so res_pjsip_nat + could not find the associated endpoint and therefore couldn't rewrite + the contact. + + This commit message is like 50x longer than the actual fix. + + ASTERISK 24981 #close + Reported by Mark Michelson + + Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd +2015-04-20 18:00 +0000 [d08446ec36] Richard Mudgett <rmudgett@digium.com> + + * chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels. + + The chan_dahdi channel driver is a very old driver. The ability for it to + support ISDN was added well after the initial analog support. Setting the + softhangup flags is a carry over from the original analog code. The + driver was not updated to call ast_queue_hangup() which will post the AMI + HangupRequest event. + + * Changed sig_pri.c to call ast_queue_hangup() instead of setting the + softhangup flag when the remote party initiates a hangup. + + ASTERISK-24895 #close + Reported by: Andrew Zherdin + + Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325 + +2015-04-20 17:23 +0000 [96e18453f4] Joshua Colp <jcolp@digium.com> + + * Merge "pjsip_options: Fix non-qualified contacts showing as unavailable" into 13 +2015-04-20 13:01 +0000 [2be9cc2643] Diederik de Groot <ddegroot@talon.nl> + + * Fix/Update clang-RAII macro implementation + + - When you need to refer to 'variable XXX' outside a block, it needs + to be declared as '__block XXX', otherwise it will not be available with- + in the block, making updating that variable hard to do, and ast_free + lead to issues. + + - Removed the #error message + because it creates complications when compiling external projects + against asterisk For example when using a different compiler than the + one used to compile asterisk. The warning/error should be generated + during the configure process not the compilation process + + ASTERISK-24917 + Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2 +2015-04-20 09:53 +0000 [b74b2cdcda] George Joseph <george.joseph@fairview5.com> + + * pjsip_options: Fix format specifier for int64_t rtt. + + Contact status rtt is an int64_t and needs the PRId64 macro to + properly create the format specifier on 32-bit systems. + + Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7 + +2015-04-20 06:29 +0000 [27a122af66] Matt Jordan <mjordan@digium.com> + + * Merge "main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple" into 13 +2015-04-20 05:54 +0000 [9581a0ebf3] Joshua Colp <jcolp@digium.com> + + * Merge "Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled." into 13 +2015-04-18 13:36 +0000 [63169e00ff] George Joseph <george.joseph@fairview5.com> + + * pjsip_options: Fix non-qualified contacts showing as unavailable + + The "Add qualify_timeout processing and eventing" patch introduced + an issue where contacts that had qualify_frequency set to 0 were + showing Unavailable instead Unknown. This patch checks for + qualify_frequency=0 and create an "Unknown" contact_status + with an RTT = 0. + + Previously, the lack of contact_status implied Unknown but since + we're now changing endpoint state based on contact_status, I've + had to add new UNKNOWN status so that changes could trigger the + appropriate contact_status observers. + + ASTERISK-24977: #close + + Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7 + +2015-04-19 15:49 +0000 [f0c82a173a] Matt Jordan <mjordan@digium.com> + + * main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple + + When a PBX registrar is unloaded, it will fail to remove its extension from + the context root_table if a dialplan application used by that extension is + still loaded. This can be the case for AGI, which can be unloaded after several + of the standard PBX providers. Often, this is harmless; however, if the + extension's priorities are removed during the failed unloading *and* the + dialplan application later unregisters, it leaves a ticking timebomb for the + next PBX provider that attempts to iterate over the extensions. When that + occurs, the peer_table pointer on the extension will already be set to NULL. + The current code does not check to see if the pointer is NULL before passing + it to a hashtab function this is not NULL tolerant. + + Since it is possible for the peer_table to be NULL when we normally would not + expect that to be the case, the solution in this patch is to simply skip over + processing an extension's priorities if peer_table is NULL. + + Prior to this patch, the tests/pbx/callerid_match test would crash during + module unload. With this patch, the test no longer crashes after running. + + ASTERISK-24774 #close + Reported by: Corey Farrell + + Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40 + +2015-04-17 18:05 +0000 [82bc0fd3ad] Richard Mudgett <rmudgett@digium.com> + + * res_fax: Fix latent bug exposed by ASTERISK-24841 changes. + + Three fax related tests started failing as a result of changes made for + ASTERISK-24841: + tests/fax/pjsip/gateway_t38_g711 + tests/fax/sip/gateway_mix1 + tests/fax/sip/gateway_mix3 + + Historically, ast_channel_make_compatible() did nothing if the channels + were already "compatible" even if they had a sub-optimal translation path + already setup. With the changes from ASTERISK-24841 this is no longer + true in order to allow the best translation paths to always be picked. In + res_fax.c:fax_gateway_framehook() code manually setup the channels to go + through slin and then called ast_channel_make_compatible(). With the + previous version of ast_channel_make_compatible() this was always a + no-operation. + + * Remove call to ast_channel_make_compatible() in fax_gateway_framehook() + that now undoes what was just setup when the framehook is attached. + + * Fixed locking around saving the channel formats in + fax_gateway_framehook() to ensure that the formats that are saved are + consistent. + + * Fix copy pasta errors in fax_gateway_framehook() that confuses read and + write when dealing with saved channel formats. + + ASTERISK-24841 + Reported by: Matt Jordan + + Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d + +2015-04-17 16:19 +0000 [c59a800707] Corey Farrell <git@cfware.com> + + * Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled. + + When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be + called as a function. This causes a compile error with raw threadstorage as + it uses NULL for cleanup. This fix uses a macro that provides NULL when + DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);" + with "{};" when DEBUG_THREADLOCALS is enabled. + + ASTERISK-24975 #close + Reported by: Ashley Sanders + + Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402 + +2015-04-17 15:57 +0000 [e05b076827] Matt Jordan <mjordan@digium.com> + + * Merge "Detect potential forwarding loops based on count." into 13 +2015-04-15 10:38 +0000 [4f1a8dbe92] Mark Michelson <mmichelson@digium.com> + + * Detect potential forwarding loops based on count. + + A potential problem that can arise is the following: + + * Bob's phone is programmed to automatically forward to Carol. + * Carol's phone is programmed to automatically forward to Bob. + * Alice calls Bob. + + If left unchecked, this results in an endless loops of call forwards + that would eventually result in some sort of fiery crash. + + Asterisk's method of solving this issue was to track which interfaces + had been dialed. If a destination were dialed a second time, then + the attempt to call that destination would fail since a loop was + detected. + + The problem with this method is that call forwarding has evolved. Some + SIP phones allow for a user to manually forward an incoming call to an + ad-hoc destination. This can mean that: + + * There are legitimate use cases where a device may be dialed multiple + times, or + * There can be human error when forwarding calls. + + This change removes the old method of detecting forwarding loops in + favor of keeping a count of the number of destinations a channel has + dialed on a particular branch of a call. If the number exceeds the + set number of max forwards, then the call fails. This approach has + the following advantages over the old: + + * It is much simpler. + * It can detect loops involving local channels. + * It is user configurable. + + The only disadvantage it has is that in the case where there is a + legitimate forwarding loop present, it takes longer to detect it. + However, the forwarding loop is still properly detected and the + call is cleaned up as it should be. + + Address review feedback on gerrit. + + * Correct "mfgium" to "Digium" + * Decrement max forwards by one in the case where allocation of the + max forwards datastore is required. + * Remove irrelevant code change from pjsip_global_headers.c + + ASTERISK-24958 #close + + Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23 +2015-04-11 16:56 +0000 [674b18bdf0] George Joseph <george.joseph@fairview5.com> + + * pjsip_options: Add qualify_timeout processing and eventing + + This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the + discussion at + http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html + + The basic issues are that changes in contact status don't cause events to be + emitted for the associated endpoint. Only dynamic contact add/delete actions + update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds + which is a long time. + + This patch makes use of the new transaction timeout feature in r4585 and + provides the following capabilities... + + 1. A new aor/contact variable 'qualify_timeout' has been added that allows the + user to specify the maximum time in milliseconds to wait for a response to an + OPTIONS message. The default is 3000ms. When the timer expires, the contact is + marked unavailable. + + 2. Contact status changes are now propagated up to the endpoint as follows... + When any contact is 'Available', the endpoint is marked as 'Reachable'. When + all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The + existing endpoint events are generated appropriately. + + ASTERISK-24863 #close + + Change-Id: Id0ce0528e58014da1324856ea537e7765466044a + Tested-by: Dmitriy Serov + Tested-by: George Joseph <george.joseph@fairview5.com> + +2015-04-17 15:29 +0000 [f1abf51b73] Matt Jordan <mjordan@digium.com> + + * Merge "res_pjsip: Refactor endpt_send_request to include transaction timeout" into 13 +2015-04-17 10:30 +0000 [ab5b38e434] Matt Jordan <mjordan@digium.com> + + * Merge "res_pjsip: Add global option to limit the maximum time for initial qualifies" into 13 +2015-04-17 10:25 +0000 [ec77b6148f] Joshua Colp <jcolp@digium.com> + + * Merge "res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced" into 13 +2015-04-16 10:51 +0000 [b56c1914fa] Kevin Harwell <kharwell@digium.com> + + * bridge.c: NULL app causes crash during attended transfer + + Due to a race condition there was a chance that during an attended transfer the + channel's application would return NULL. This, of course, would cause a crash + when attempting to access the memory. This patch retrieves the channel's app + at an earlier time in processing in hopes that the app name is available. + However, if it is not then "unknown" is used instead. Since some string value + is now always present the crash can no longer occur. + + ASTERISK-24869 #close + Reported by: viniciusfontes + Review: + + Change-Id: I5134b84c4524906d8148817719d76ffb306488ac + +2015-04-16 13:20 +0000 [8d4ce7cc2b] Scott Griepentrog <scott@griepentrog.com> + + * res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced + + This change makes the send_notify of the sub_tree + not happen when the sub_tree has been deleted due + to the notify call failing, which avoids a crash. + + ASTERISK-24970 #close + + Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf +2015-04-11 16:39 +0000 [bf46799f0e] George Joseph <george.joseph@fairview5.com> + + * res_pjsip: Refactor endpt_send_request to include transaction timeout + + This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the + discussion at + http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html + + Since we currently have no control over pjproject transaction timeout, this + patch pulls the pjsip_endpt_send_request function out of pjproject and into + res_pjsip/endpt_send_transaction in order to implement that capability. + + Now when the transaction is initiated, we also schedule our own pj_timer with + our own desired timeout. + + If the transaction completes before either timeout, pjproject cancels its timer, + and calls our tsx callback where we cancel our timer and run the app callback. + + If the pjproject timer times out first, pjproject calls our tsx callback where + we cancel our timer and run the app callback. + + If our timer times out first, we terminate the transaction which causes + pjproject to cancel its timer and call our tsx callback where we run the app + callback. + + Regardless of the scenario, pjproject is calling the tsx callback inside the + group_lock and there are checks in the callback to make sure it doesn't run + twice. + + As part of this patch ast_sip_send_out_of_dialog_request was created to replace + its similarly named private function. It takes a new timeout argument in + milliseconds (<= 0 to disable the timeout). + + ASTERISK-24863 #close + Reported-by: George Joseph <george.joseph@fairview5.com> + Tested-by: George Joseph <george.joseph@fairview5.com> + + Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747 +2015-04-11 17:04 +0000 [1b6f6ff841] George Joseph <george.joseph@fairview5.com> + + * res_pjsip: Add global option to limit the maximum time for initial qualifies + + Currently when Asterisk starts initial qualifies of contacts are spread out + randomly between 0 and qualify_timeout to prevent network and system overload. + If a contact's qualify_frequency is 5 minutes however, that contact may be + unavailable to accept calls for the entire 5 minutes after startup. So while + staggering the initial qualifies is a good idea, basing the time on + qualify_timeout could leave contacts unavailable for too long. + + This patch adds a new global parameter "max_initial_qualify_time" that sets the + maximum time for the initial qualifies. This way you could make sure that all + your contacts are initialy, randomly qualified within say 30 seconds but still + have the contact's ongoing qualifies at a 5 minute interval. + + If max_initial_qualify_time is > 0, the formula is initial_interval = + min(max_initial_interval, qualify_timeout * random(). If not set, + qualify_timeout is used. + + The default is "0" (disabled). + + ASTERISK-24863 #close + + Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 + Tested-by: George Joseph <george.joseph@fairview5.com> + +2015-04-15 16:08 +0000 [5d218cde87] George Joseph <george.joseph@fairview5.com> + + * More .gitignore updates + + Added .pyc and .sha1 to the top-level .gitignore. + + Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a + Tested-by: George Joseph <george.joseph@fairview5.com> + +2015-04-15 13:36 +0000 [97f83c4c53] Matt Jordan <mjordan@digium.com> + + * Merge "Build System: Replace comment about setting menuselect defaults." into 13 +2015-04-14 13:16 +0000 [abd56db3e0] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * cel_pgsql: Fix name string for log on unable allocate memory. + + The LOG_ERROR has reference to CDR instead of CEL for LENGTHEN_BUF1 and + LENGTHEN_BUF2. + + ASTERISK-24965 #close + Reported by: Rodrigo Ramirez Norambuena + + Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744 +2015-04-14 13:48 +0000 [222fbe1d9a] Corey Farrell <git@cfware.com> + + * Build System: Replace comment about setting menuselect defaults. + + The Makefile claims that you can set default menuselect options by creating + ~/.asterisk.makeopts or /etc/asterisk.makeopts, but those files have never + been respected in Asterisk 11 or 13. This changes the comment to accurately + reflect that these files are not automatically used by the build system. + + ASTERISK-13721 #close + Reported by: pj + + Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2 + +2015-04-12 09:08 +0000 [07e729cc7b] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> + + * cdr_pgsql: Fix CLI "cdr show pgsql status" command. + + The command always showed the usage information. + + * Fix the error in command validation for CLI_SHOWUSAGE. + + ASTERISK-24959 #close + Reported by: Rodrigo Ramirez Norambuena + + Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5 + (cherry picked from commit 23a180cade51e84b9def65b05759c3cb9feba225) + +2015-04-13 19:06 +0000 [7d43d85bea] George Joseph <george.joseph@fairview5.com> + + * .gitignore updates for master/13 + + Added products of ./bootstrap + + Added nmenuselect and gmenuselect to menuselect/ + + Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35 + +2015-04-13 14:41 +0000 [3d27c223a5] David M. Lee <dlee@respoke.io> + + * Fixing extconf compile + + During the mass code deletion for clang support, a stray backslash was + left behind that was causing utils to fail to compile. + + Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1 + +2015-04-13 12:03 +0000 [30045b4e67] Matt Jordan <mjordan@digium.com> + + * Merge "build_tools/make_version: Update version parsing for Git migration" into 13 +2015-04-13 10:47 +0000 [88dbf6653e] Joshua Colp <jcolp@digium.com> + + * Merge "res_monitor: Add dependency on func_periodic_hook." into 13 +2015-04-13 09:54 +0000 [e996d8f728] Matt Jordan <mjordan@digium.com> + + * build_tools/make_version: Update version parsing for Git migration + + External systems - such as the Asterisk Test Suite - require knowledge of the + upstream branch. Unfortunately, after moving to Git, the Asterisk version + currently consists of only a 'GIT" prefix followed by an object blob, + e.g., GIT-as08d7. This makes it difficult for such systems to know what + features are available in a particular check out of Asterisk. + + This patch fixes this by hardcoding the branch in a variable in the + make_version script. Since the mainline branches are not changed often - + typically only once a year - this is a reasonable approach to solving + the problem, and is more reliable than parsing the output of 'git branch + -vv'. Branches that track off of an upstream primary branch will then get the + benefit of knowing which mainline branch they are currently based off + of. + + ASTERISK-24954 #close + + Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799 + +2015-04-12 12:59 +0000 [d1a6f1a9f9] Matt Jordan <mjordan@digium.com> + + * git migration: Remove support for file versions + + Git does not support the ability to replace a token with a version + string during check-in. While it does have support for replacing a + token on clone, this is somewhat sub-optimal: the token is replaced + with the object hash, which is not particularly easy for human + consumption. What's more, in practice, the source file version was often + not terribly useful. Generally, when triaging bugs, the overall version + of Asterisk is far more useful than an individual SVN version of a file. + As a result, this patch removes Asterisk's support for showing source file + versions. + + Specifically, it does the following: + * main/asterisk: + - Refactor the file_version structure to reflect that it no longer + tracks a version field. + - Alter the "core show file version" CLI command such that it always + reports the version of Asterisk. The file version is no longer + available. + + * main/manager: The Version key now always reports the Asterisk version. + + * UPGRADE: Add notes for: + - Modification to the ModuleCheck AMI Action. + - Modification of the "core show file version" CLI command. + + Change-Id: Ia932d3c64cd18a14a3c894109baa657ec0a85d28 + +2015-04-13 06:19 +0000 [0e4b997cd7] Corey Farrell <git@cfware.com> + + * res_monitor: Add dependency on func_periodic_hook. + + OPTIONAL_API has conditionals to define AST_OPTIONAL_API and + AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined. + Unfortunately this is inside the include protection block, so only the + first status of AST_API_MODULE is respected. For example res_monitor + is an optional API provider, but uses func_periodic_hook. This makes + func_periodic_hook non-optional to res_monitor. + + ASTERISK-17608 #close + Reported by: Warren Selby + + Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679 + +2015-04-12 15:27 +0000 [91c1ed7ef6] Matt Jordan <mjordan@digium.com> + + * Merge "main/editline: Add .gitignore." into 13 +2015-04-12 06:12 +0000 [a77c31b99c] Corey Farrell <git@cfware.com> + + * main/editline: Add .gitignore. + + This patch adds a .gitignore for main/editline to ignore all build results. + + Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d + +2015-04-11 23:22 +0000 [d918c3b78e] Matt Jordan <mjordan@digium.com> + + * .gitignore: Ignore tarballs (*.gz) + + This patch updates the root .gitignore file to ignore files with a .gz + extension. This will cause git to ignore downloaded sound tarballs in + the the sounds/ directory. + + Change-Id: I1e42fbfa02a8884231507b683e8e49ac3e278aaa + +2015-04-11 13:20 +0000 [555b5f5d30] George Joseph <george.joseph@fairview5.com> + + * Add .gitignore and .gitreview files + + Add the .gitignore and .gitreview files to the asterisk repo. + + NB: You can add local ignores to the .git/info/exclude file + without having to do a commit. + + Common ignore patterns are in the top-level .gitignore file. + Subdirectory-specific ignore patterns are in their own .gitignore + files. + + Change-Id: I4c8af3b8e3739957db545f7368ac53f38e99f696 + Tested-by: George Joseph + +2015-04-11 10:35 +0000 [5807ca519c] Matthew Jordan <mjordan@digium.com> + + * Blocked revisions 434708 + + ........ + main/event: Remove unnecessary assignment of negative value to enum + + When cleaning up some clang compiler warnings, the comparison of a negative + value to an unsigned enum was removed. However, the initial assignment of a + negative value to said enum remained in the variable declaration. This patch + removes that assignment. + + Thanks to ibercom in #asterisk-bugs for pointing it out. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-11 10:26 +0000 [d0d78d5732] dkdegroot (License 6600) + + * clang compiler warnings: Fix various warnings for tests + + This patch fixes a variety of clang compiler warnings for unit tests. This + includes autological comparison issues, ignored return values, and + interestingly enough, one embedded function. Fun! + + Review: https://reviewboard.asterisk.org/r/4555 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4555.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434705 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-11 10:10 +0000 [4cf7d0bf01] Juergen Spies (License 6698) + + * res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram + + Prior to this patch, the far_max_datagram value on the UDPTL structure would + remain -1 if the remote endpoint fails to provide the SDP media attribute + T38FaxMaxDatagram. This can result in the INVITE request being rejected. With + this patch, we will now properly initialize the value with either the default + value or with the value provided by pjsip.conf's t38_udptl_maxdatagram + parameter. + + Review: https://reviewboard.asterisk.org/r/4589 + + ASTERISK-24928 #close + Reported by: Juergen Spies + Tested by: Juergen Spies + patches: + pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 18:29 +0000 [13cd99682d] Richard Mudgett <rmudgett@digium.com> + + * chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. + + With this patch, chan_pjsip/res_pjsip now sets the native formats to the + codecs negotiated by a call. + + * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native + formats to include all the negotiated audio codecs instead of only the + initial preferred audio codec and later the currently received audio + codec. + + * The audio frame handling in channel.c:ast_read() is more streamlined and + will automatically adjust to changes in received frame formats. The new + policy is to remove translation and pass the new frame format to the + receiver except if the translation was to a signed linear format. A more + long winded version is commented in ast_read() along with some caveats. + + * The audio frame handling in channel.c:ast_write() is more streamlined + and will automatically adjust any needed translation to changes in the + frame formats sent. Frame formats sent can change for many reasons such + as a recording is being played back or the bridged peer changed the format + it sends. Since it is a normal expectation that sent formats can change, + the codec mismatch warning message is demoted to a debug message. + + * Removed the short circuit check in + channel.c:ast_channel_make_compatible_helper(). Two party bridges need to + make channels compatible with each other. However, transfers and moving + channels among bridges can result in otherwise compatible channels having + sub-optimal translation paths if the make compatible check is short + circuited. A result of forcing the reevaluation of channel compatibility + is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc + options take effect consistently now. It is unfortunate that these two + options are enabled by default and negate some of the benefits to the + changes in channel.c:ast_read() by forcing translation through signed + linear on a two party bridge. + + * Improved the softmix bridge technology to better control the translation + of frames to the bridge. All of the incoming translation is now normally + handled by ast_read() instead of splitting any translation steps between + ast_read() and the slin factory. If any frame comes in with an unexpected + format then the translation path in ast_read() is updated for the next + frame and the slin factory handles the current frame translation. + + This is the final patch in a series of patches aimed at improving + translation path choices. The other patches are on the following reviews: + https://reviewboard.asterisk.org/r/4600/ + https://reviewboard.asterisk.org/r/4605/ + + ASTERISK-24841 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4609/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 16:03 +0000 [af458e2e60] Kevin Harwell <kharwell@digium.com> + + * chan_sip: make progressinband default to no + + After the "progressinband" value setting of "never" was updated to never send a + 183 this separated its use from the "no" value. Since "never" was the default, + but most users probably expect "no" this patch updates the default for the + "progressinband" setting to "no." + + ASTERISK-24835 #close + Reported by: Andrew Nagy + Review: https://reviewboard.asterisk.org/r/4606/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434654 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 12:53 +0000 [88b0fa7755] yaron nahum (License 6676) + + * res_pjsip: Add an 'auto' option for DTMF Mode + + This patch adds support for automatically detecting the type of DTMF that a + PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', + the channel created for an endpoint will attempt to determine if RFC 4733 + DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type + for the channel will be set to inband. + + Review: https://reviewboard.asterisk.org/r/4438 + + ASTERISK-24706 #close + Reported by: yaron nahum + patches: + yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 11:59 +0000 [16afee4651] George Joseph <george.joseph@fairview5.com> + + * res_pjsip_config_wizard: Cleanup load unload + + While investigating other unload issues I realized that the load/unload process + for the config wizard was pretty ugly so I've refactored it as follows... + + When the res_pjsip sorcery instance is created the config_wizard bumps it's own + module reference to prevent it from unloading while the sorcery instance is + still active. When res_pjsip unloads and it's sorcery instance is destroyed, + the config wizard unrefs itself which then allows itself to unload cleanly. + Since the config wizard now can't load after res_pjsip or unload before it + (which should have been the correct behavior all along), I was able to remove + the chunks of code in both load_module and unload_module that handled that case. + + Ran the testsuite tests to insure there were no functional changes and REF_DEBUG + to insure that Asterisk was shutting down cleanly with no FRACKs or leaks. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4610/ + + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 11:37 +0000 [125acc52fe] Richard Mudgett <rmudgett@digium.com> + + * bridge_softmix.c,channel.c: Minor code simplification and cleanup. + + * Made code easier to follow in bridge_softmix.c:analyse_softmix_stats() + and made some debug messages more helpful. + + * Made some debug and warning messages more helpful in + channel.c:set_format(). + + Review: https://reviewboard.asterisk.org/r/4607/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 11:28 +0000 [a63f7ad04a] Richard Mudgett <rmudgett@digium.com> + + * translate.c: Only select audio codecs to determine the best translation choice. + + Given a source capability of h264 and ulaw, a destination capability of + h264 and g722 then ast_translator_best_choice() would pick h264 as the + best choice even though h264 is a video codec and Asterisk only supports + translation of audio codecs. When the audio starts flowing, there are + warnings about a codec mismatch when the channel tries to write a frame to + the peer. + + * Made ast_translator_best_choice() only select audio codecs. + + * Restore a check in channel.c:set_format() lost after v1.8 to prevent + trying to set a non-audio codec. + + This is an intermediate patch for a series of patches aimed at improving + translation path choices for ASTERISK-24841. + + This patch is a complete enough fix for ASTERISK-21777 as the v11 version + of ast_translator_best_choice() does the same thing. However, chan_sip.c + still somehow tries to call ast_codec_choose() which then calls + ast_best_codec() with a capability set that doesn't contain any audio + formats for the incoming call. The remaining warning message seems to be + a benign transient. + + ASTERISK-21777 #close + Reported by: Nick Ruggles + + ASTERISK-24380 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4605/ + ........ + + Merged revisions 434614 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 09:55 +0000 [c9791dba1f] Matthew Jordan <mjordan@digium.com> + + * res/ari: Fix model validation for ChannelHold event + + When the ChannelHold event was added, the 'musicclass' parameter was + erroneously removed. This caused the ChannelHold events to be rejected as + they failed model validation. This patch updates the Swagger schema such that + it now properly reflects the event that is being created. + + Hooray for tests that catch things like this. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 07:39 +0000 [c39faa4729] Y Ateya (License 6693) + + * channels/chan_iax2: Improve POKE expiration time calculation for lossy networks + + POKE is used to check for peer availability; however, in networks with packet + loss, the current calculations may result in POKE expiration times that are too + short. This patch alters the expiration/retry time logic to take into account + the last known qualify round trip time, as opposed to always using a static + value for each peer. + + Review: https://reviewboard.asterisk.org/r/4536 + + ASTERISK-22352 #close + Reported by: Frederic Van Espen + + ASTERISK-24894 #close + Reported by: Y Ateya + patches: + poke_noanswer_duration.diff submitted by Y Ateya (License 6693) + ........ + + Merged revisions 434564 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 17:35 +0000 [75c2c85962] George Joseph <george.joseph@fairview5.com> + + * res_pjsip_phoneprov_provider: Fix reference leak on unload + + res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to + a missing OBJ_NODATA in an ao2_callback in load_users(). Rather than adding the + OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator. + This plugged the leak but exposed an unload order issue between + res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip. + + res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip. + Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it + unloads, it's objects are still in the sorcery instance. When res_pjsip + unloads, it destroys all its objects including res_pjsip_phoneprov_provider's. + The phoneprov destructor then attempts to unregister the extension from + res_phoneprov but because res_phoneprov is already cleaned up, its users + container is gone and we get a FRACK. + + Simple solution, check for the NULL users container before attempting to remove + the entry. Duh. + + Ran tests/res_phoneprov/res_phoneprov_provider. No leaks in + res_pjsip_phoneprov_provider and no FRACKs. + + Reported-by: Corey Farrell + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4608/ + ASTERISK-24935 #close + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 17:31 +0000 [73c286a393] George Joseph <george.joseph@fairview5.com> + + * loader/main: Don't set ast_fully_booted until deferred reloads are processed + + Until we have a true module management facility it's sometimes necessary for one + module to force a reload on another before its own load is complete. If + Asterisk isn't fully booted yet, these reloads are deferred. The problem is + that asterisk reports fully booted before processing the deferred reloads which + means Asterisk really isn't quite ready when it says it is. + + This patch moves the report of fully booted after the processing of the deferred + reloads is complete. + + Since the pjsip stack has the most number of related modules, I ran the + channels/pjsip testsuite to make sure there aren't any issues. All tests + passed. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4604/ + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 17:03 +0000 [5737650a67] Kevin Harwell <kharwell@digium.com> + + * res_pjsip: add CLI command to show global and system configuration + + Added a new CLI command for res_pjsip that shows both global and system + configuration settings: pjsip show settings + + ASTERISK-24918 #close + Reported by: Scott Griepentrog + Review: https://reviewboard.asterisk.org/r/4597/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 11:07 +0000 [1695a5b85f] Richard Mudgett <rmudgett@digium.com> + + * chan_iax2.c: Fix ref leak in iax2_request(). + + * Increased warning message format capability string buffer size in + iax2_request(). + + Review: https://reviewboard.asterisk.org/r/4601/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 10:54 +0000 [92c1688edb] Richard Mudgett <rmudgett@digium.com> + + * bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible(). + + Review: https://reviewboard.asterisk.org/r/4601/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 10:42 +0000 [2679d0100a] yaron nahum (License 6676) + + * res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests + + This patch adds a new session supplement that handles in-dialog OPTIONS + requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup + for the OPTIONS request would already have been done by the time the + session supplement receives the inbound request. + + ASTERISK-24862 #close + Reported by: yaron nahum + patches: + res_pjsip_dlg_options.c submitted by yaron nahum (License 6676) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 07:56 +0000 [6ba6e3dffd] dkdegroot (License 6600) + + * clang compiler warnings: Fix autological comparisons + + This fixes autological comparison warnings in the following: + * chan_skinny: letohl may return a signed or unsigned value, depending on the + macro chosen + * func_curl: Provide a specific cast to CURLoption to prevent mismatch + * cel: Fix enum comparisons where the enum can never be negative + * enum: Fix comparison of return result of dn_expand, which returns a signed + int value + * event: Fix enum comparisons where the enum can never be negative + * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be + negative + * presencestate: Use the actual enum value for INVALID state + * security_events: Fix enum comparisons where the enum can never be negative + * udptl: Don't bother to check if the return value from encode_length is less + than 0, as it returns an unsigned int + * translate: Since the parameters are unsigned int, don't bother checking + to see if they are negative. The cast to unsigned int would already blow + past the matrix bounds. + * res_pjsip_exten_state: Use a temporary value to cache the return of + ast_hint_presence_state + * res_stasis_playback: Fix enum comparisons where the enum can never be + negative + * res_stasis_recording: Add an enum value for the case where the recording + operation is in error; fix enum comparisons + * resource_bridges: Use enum value as opposed to -1 + * resource_channels: Use enum value as opposed to -1 + + Review: https://reviewboard.asterisk.org/r/4533 + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4533.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 21:05 +0000 [e05c8ae68e] Stefan Engström (License 6691) + + * apps/app_queue: Prevent possible crash when evaluating queue penalty rules + + Although it only occurred once, a crash occurred when a queue attempted to + evaluate a queue penalty rule that appeared to have already been destroyed. + In many locations in app_queue, a test is done to see if qe->pr is NULL; + however, when we dispose of a queue's penalty rules, we don't set the pointer + to NULL after free'ing it. This patch does that to prevent any dangling + pointers from lingering on the queue object. + + Review: https://reviewboard.asterisk.org/r/4522 + + ASTERISK-23319 #close + Reported by: Vadim + patches: + rb4552.patch submitted by Stefan Engström (License 6691) + ........ + + Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 13:15 +0000 [f21b45db49] Jonathan Rose <jrose@digium.com> + + * res_pjsip_t38: Fix FAX failures when using PJSIP with authentication + + Without this patch, if a PJSIP endpoint with udptl enabled and authentication + set attempted to use sendFax, the FAX session would fail during setup. This + was because the invite issued in response to being auth challenged would cause + the PJSIP channel performing the FAX to receive a second T38 framehook and + this would cause frames to be consumed in an inappropriate manner. + + ASTERISK-24933 #close + Reported by: Jonathan Rose + Review: https://reviewboard.asterisk.org/r/4577/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 13:14 +0000 [4441bb6a25] Richard Mudgett <rmudgett@digium.com> + + * Bridging: Eliminate the unnecessary make channel compatible with bridge operation. + + When a channel enters the bridging system it is first made compatible with + the bridge and then the bridge technology makes the channel compatible + with the technology. For all but the DAHDI native and softmix bridge + technologies the make channel compatible with the bridge step is an + effective noop because the other technologies allow all audio formats. + For the DAHDI native bridge technology it doesn't matter because it is not + an initial bridge technology and chan_dahdi allows only one native format + per channel. For the softmix bridge technology, it is a noop at best and + harmful at worst because the wrong translation path could be setup if the + channel's native formats allow more than one audio format. + + This is an intermediate patch for a series of patches aimed at improving + translation path choices. + + * Removed code dealing with the unnecessary step of making the channel + compatible with the bridge. + + ASTERISK-24841 + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4600/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 11:40 +0000 [f767440906] mhej (license 6085) + + * Security/tcptls: MitM Attack potential from certificate with NULL byte in CN. + + When registering to a SIP server with TLS, Asterisk will accept CA signed + certificates with a common name that was signed for a domain other than the + one requested if it contains a null character in the common name portion of + the cert. This patch fixes that by checking that the common name length + matches the the length of the content we actually read from the common name + segment. Some certificate authorities automatically sign CA requests when + the requesting CN isn't already taken, so an attacker could potentially + register a CN with something like www.google.com\x00www.secretlyevil.net + and have their certificate signed and Asterisk would accept that certificate + as though it had been for www.google.com - this is a security fix and is + noted in AST-2015-003. + + ASTERISK-24847 #close + Reported by: Maciej Szmigiero + Patches: + asterisk-null-in-cn.patch submitted by mhej (license 6085) + ........ + + Merged revisions 434337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 434338 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 11:23 +0000 [1712d16825] Richard Mudgett <rmudgett@digium.com> + + * format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear(). + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 07:33 +0000 [ae39dd1f46] Matthew Jordan <mjordan@digium.com> + + * chan_iax2: Fix compilation issue due to funky merge + + Don't mix declarations and code + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 07:00 +0000 [05397ad01e] Jaco Kroon (License 5671) + + * chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno] + + This patch fixes an access to the peer callnumber that is unprotected by a + corresponding mutex. The peer->callno value can be changed by multiple threads, + and all data inside the iaxs array must be procted by a corresponding lock + of iaxsl. + + The patch moves the unprotected access to a location where the mutex is + safely obtained. + + Review: https://reviewboard.asterisk.org/r/4599/ + + ASTERISK-21211 #close + Reported by: Jaco Kroon + patches: + asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671) + ........ + + Merged revisions 434291 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 06:53 +0000 [be13c72142] Valentin Vidić (License 6697) + + * chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled + + When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will + attempt to handle both IPv4 and IPv6 addresses, although the information will + be stored in a struct with an AF_INET6 address type. However, the current + NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly. + This patch adds an additional check for the mapped address case, allowing + the NAT code to handle clients even when the address is IPv6. + + Review: https://reviewboard.asterisk.org/r/4563/ + + ASTERISK-18032 #close + Reported by: Christoph Timm + patches: + nat_with_ipv6.diff submitted by Valentin Vidić (License 6697) + ........ + + Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 06:44 +0000 [f324870dab] dkdegroot (License 6600) + + * clang compiler warnings: Fix pointer-bool-converesion warnings + + This patch fixes several warnings pointed out by the clang compiler. + * chan_pjsip: Removed check for data->text, as it will always be non-NULL. + * app_minivm: Fixed evaluation of etemplate->locale, which will always + evaluate to 'true'. This patch changes the evaluation to use + ast_strlen_zero. + * app_queue: + - Fixed evaluation of qe->parent->monfmt, which always evaluates to + true. Instead, we just check to see if the dereferenced pointer + evaluates to true. + - Fixed evaluation of mem->state_interface, wrapping it with a call to + ast_strlen_zero. + * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero. + + Review: https://reviewboard.asterisk.org/r/4541 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4541.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 14:38 +0000 [a6aed7f6f6] Scott Griepentrog <sgriepentrog@digium.com> + + * Revert accidental change in r434261 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 14:35 +0000 [0584e29300] Scott Griepentrog <sgriepentrog@digium.com> + + * pjsip: resolve compatibility problem with ast_sip_session + + A change in r430179 inserted a variable near the top of a + structure caused a problem when running DPMA in a version + of Asterisk compiled across the change. This patch moves + the new variable to the end of the structure, eliminating + the problem. + + Review: https://reviewboard.asterisk.org/r/4574/ + ........ + + Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 11:40 +0000 [d754f70239] Kevin Harwell <kharwell@digium.com> + + * bridge.c: Hangup attended transfer target after it has been swapped out + + After completing an attended transfer the transfer target channel (the one that + gets swapped out) was not being hung up after leaving the bridge. This resulted + in a channel possibly being left around. Added an explicit softhangup for the + channel in question after the transfer is successfully completed in order to + make sure the channel is hung up. + + ASTERISK-24782 #close + Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/4575/ + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 10:33 +0000 [c516981dc7] Mark Michelson <mmichelson@digium.com> + + * Do not queue message requests that we do not respond to. + + If we receive a MESSAGE request that we cannot send a response + to, we should not send the incoming MESSAGE to the dialplan. + + This commit should help the bouncing message_retrans test to + pass consistently. + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 10:21 +0000 [ab803ec342] Matthew Jordan <mjordan@digium.com> + + * ARI: Add the ability to intercept hold and raise an event + + For some applications - such as SLA - a phone pressing hold should not behave + in the fashion that the Asterisk core would like it to. Instead, the hold + action has some application specific behaviour associated with it - such as + disconnecting the channel that initiated the hold; only playing MoH to channels + in the bridge if the channels are of a particular type, etc. + + One way of accomplishing this is to use a framehook to intercept the + hold/unhold frames, raise an event, and eat the frame. Tasty. This patch + accomplishes that using a new dialplan function, HOLD_INTERCEPT. + + In addition, some general cleanup of raising hold/unhold Stasis messages was + done, including removing some RAII_VAR usage. + + Review: https://reviewboard.asterisk.org/r/4549/ + + ASTERISK-24922 #close + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 21:09 +0000 [488f093e97] dkdegroot (License 6600) + + * clang compiler warnings: Fix sometimes-initialized warning in func_math + + This patch fixes a bug in a unit test in func_math where a variable could be + passed to ast_free that wasn't allocated. This patch corrects the issue and + ensures that we only attempt to free a variable if we previously allocated + it. + + Review: https://reviewboard.asterisk.org/r/4552 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4552.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434190 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434191 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 21:03 +0000 [c027133f6d] dkdegroot (License 6600) + + * clang compiler warnings: Fix non-literal-null-conversion warnings + + Clang will flag errors when a char pointer is set to '\0', as opposed to a + value that the char pointer points to. This patch fixes this warning + in a variety of locations. + + Review: https://reviewboard.asterisk.org/r/4551 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4551.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434187 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 14:23 +0000 [2270c40d33] Kevin Harwell <kharwell@digium.com> + + * res_pjsip: config option 'timers' can't be set to 'no' + + When setting the configuration option 'timers' equal to 'no' the bit flag was + not properly negated. This patch clears all associated flags and only sets the + specified one. pjsip will handle any necessary flag combinations. Also went + ahead and did similar for the '100rel' option. + + ASTERISK-24910 #close + Reported by: Ray Crumrine + Review: https://reviewboard.asterisk.org/r/4582/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 14:02 +0000 [95de71f247] George Joseph <george.joseph@fairview5.com> + + * build: Fixes for gcc 5 compilation + + These are fixes for compilation under gcc 5.0... + + chan_sip.c: In parse_request needed to make 'lim' unsigned. + inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 + inline semantics (same as clang). + ccss.c: In ast_cc_set_parm, needed to fix weird comparison. + dsp.c: Needed to work around a possible compiler bug. It was throwing + an array-bounds error but neither + sgriepentrog, rmudgett nor I could figure out why. + manager.c: In action_atxfer, needed to correct an array allocation. + + This patch will go to 11, 13, trunk. + + Review: https://reviewboard.asterisk.org/r/4581/ + Reported-by: Jeffrey Ollie + Tested-by: George Joseph + ASTERISK-24932 #close + ........ + + Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 13:18 +0000 [d54ccda3b1] dkdegroot (License 6600) + + * clang compiler warnings: Remove large chunks of unused code from extconf + + This patch fixes a warning caught by clang, in which it detected that large + chunks of extconf were unused. Frankly, I wish we could pretend that all of + extconf was unused, but alas, that is not yet the case. + + A few extraneous functions in the parking tests were removed as well, for + the same reason. + + Review: https://reviewboard.asterisk.org/r/4553 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4553.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434093 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 13:03 +0000 [0ecd472e4f] dkdegroot (License 6600) + + * clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config + + This patch fixes a warning caught by clang, in which a char pointer could be + assigned to before it was initialized. The patch re-organizes the code to + ensure that the pointer is always initialized, even on off nominal paths. + + Review: https://reviewboard.asterisk.org/r/4529 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4529.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434090 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 12:52 +0000 [4e7be5b2dc] dkdegroot (License 6600) + + * clang compiler warnings: Fix format specified in framehook + + This patch fixes an invalid format specifier used in the formatting of an + ERROR message in the framehook code. The format specifier specifies a + type of 'unsigned short', but the argument passed to it is of type 'int'. + The patch changes the format specifier to 'i'. + + Review: https://reviewboard.asterisk.org/r/4540 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4535.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434087 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 11:02 +0000 [2443b40341] Mark Michelson <mmichelson@digium.com> + + * Ensure that a non-zero sample rate is returned for all formats. + + Versions of Asterisk prior to 12 defaulted to 8000 as a sample rate + if one was not provided by a format. In Asterisk 13, this was removed. + The result was that some calculations which involve dividing by the + sample rate resulted in dividing by 0. The fix being put in place + here is to have the same default fallback that was present in previous + versions of Asterisk. + + Asterisk-24914 #close + Reported by Marcello Ceschia + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 10:16 +0000 [b1102cd642] Corey Farrell <git@cfware.com> + + * res_pjsip_phoneprov_provider: Revert 433996 / 433997. + + res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then + ignoring the return. OBJ_NODATA flag was to prevent a reference leak, but + this caused the module to FRACK on unload. Revert change until this can + be investigated further. + + ASTERISK-24935 + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4578/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434025 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 09:50 +0000 [0f25076f67] Mark Michelson (license #5049) + + * ParkedCall: Don't allow dialplan fallthrough after retrieving parked call. + + This is a change to align behavior with that of Asterisk 11 and previous versions. + In those versions, if a parked call were retrieved, and the call ended, the parked + call retriever would be hung up after the ParkedCall application ran. Prior to this + patch, in Asterisk 13, the same situation would result in the parked call retriever + falling through to additional priorities in the extension where the ParkedCall + application was called. With this patch, the behavior between Asterisk 11 and 13 + aligns. + + ASTERISK-24899 #close + Reported by Malcolm Davenport + Patches: + ASTERISK-24899.patch uploaded by Mark Michelson(license #5049) + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-05 07:53 +0000 [709fa14b44] Corey Farrell <git@cfware.com> + + * res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator. + + res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then + ignoring the return. Added OBJ_NODATA flag to prevent a reference leak. + + ASTERISK-24935 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4578/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-03 16:53 +0000 [1ee8424f27] Mark Michelson <mmichelson@digium.com> + + * res_pjsip_messaging: Serialize outbound SIP MESSAGEs + + Outbound SIP MESSAGEs had the potential to be sent out + of order from how they were specified in a set of + dialplan steps. + + This change creates a serializer for sending outbound + MESSAGE requests on. This ensures that the MESSAGEs are + sent by Asterisk in the same order that they were sent + from the dialplan. + + ASTERISK-24937 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4579 + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-02 09:56 +0000 [169e57d2e0] Scott Griepentrog <sgriepentrog@digium.com> + + * pjsip: resolve compatibility problem with ast_sip_session + + A change in r430179 inserted a variable near the top of a + structure caused a problem when running DPMA in a version + of Asterisk compiled across the change. This patch moves + the new variable to the end of the structure, eliminating + the problem. + + Review: https://reviewboard.asterisk.org/r/4574/ + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-02 05:31 +0000 [1eb0c5f4e8] Corey Farrell <git@cfware.com> + + * Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS. + + DEBUG_CHAOS was marked as conflicting with MALLOC_DEBUG, but + for this to work correctly MALLOC_DEBUG must also be marked + as conflicting with DEBUG_CHAOS. + + Review: https://reviewboard.asterisk.org/r/4557/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-01 11:25 +0000 [e301185983] Ashley Sanders <asanders@digium.com> + + * stasis: set a channel variable on websocket disconnect error + + Resolve compile errors caused by r433863 by fixing the + documentation xml to comply with the schema. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 22:26 +0000 [a1f12d9231] Ashley Sanders <asanders@digium.com> + + * stasis: set a channel variable on websocket disconnect error + + Resolve compile errors caused by r433839 by included the missing + header file, pbx.h. + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 17:00 +0000 [7293ecd90b] Ashley Sanders <asanders@digium.com> + + * stasis: set a channel variable on websocket disconnect error + + When an error occurs while writing to a web socket, the web socket is + disconnected and the event is logged. A side-effect of this, however, is that + any application on the other side waiting for a response from Stasis is left + hanging indefinitely (as there is no mechanism presently available for + notifying interested parties about web socket error states in Stasis). + + To remedy this scenario, this patch introduces a new channel variable: + STASISSTATUS. + + The possible values for STASISSTATUS are: + SUCCESS - The channel has exited Stasis without any failures + FAILED - Something caused Stasis to croak. Some (not all) possible + reasons for this: + - The app registry is not instantiated; + - The app requested is not registered; + - The app requested is not active; + - Stasis couldn't send a start message + + ASTERISK-24802 + Reported By: Kevin Harwell + Review: https://reviewboard.asterisk.org/r/4519/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 11:55 +0000 [94949e7f2f] Richard Mudgett <rmudgett@digium.com> + + * chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos. + + Fix misplaced parentheses in original fabs() expression. + ........ + + Merged revisions 433816 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 06:47 +0000 [9967739669] Corey Farrell <git@cfware.com> + + * Re-add _ast_mem_backtrace_buffer variable for ABI compatibility. + + Modules built prior to commit of r4502 expect to link at runtime + to the variable _ast_mem_backtrace_buffer. This change re-adds + the variable to the C file only. + + Review: https://reviewboard.asterisk.org/r/4558/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-30 06:42 +0000 [2d39bc5528] Corey Farrell <git@cfware.com> + + * Fix an ABI compatibility issue with ast_log_safe for modules. + + Binary modules are sometimes built against the latest release of + Asterisk in each branch, and need to be compatible with all + releases of that branch. This change ensures that utils.h only + uses ast_log_safe from the core. For modules and utilities ast_log + is used instead. + + Review: https://reviewboard.asterisk.org/r/4548/ + ........ + + Merged revisions 433772 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 21:44 +0000 [5f8faf16af] dkdegroot (License 6600) + + * clang compiler warnings: Fix -Wabsolute-value warnings + + This patch fixes several warnings caught by clang - in this case, usage of the + abs function on non-integer values. This patch uses labs and fabs, as + appropriate, in the various affected files. + + Review: https://reviewboard.asterisk.org/r/4525 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4525.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 21:39 +0000 [09b681e344] dkdegroot (License 6600) + + * clang compiler warnings: Fix invalid enum conversion + + This patch fixes some invalid enum conversion warnings caught by clang. In + particular: + * chan_sip: Several functions mixed usage of the st_refresher_param + enum and st_refresher enum. This patch corrects the functions to use the + right enum. + * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state. + * strings: Fixed incorrect usage of AO2 flags with strings container. + * res_stasis: Change a return enumeration to stasis_app_user_event_res. + + Review: https://reviewboard.asterisk.org/r/4535 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4535.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 21:29 +0000 [7f33abb827] Matthew Jordan <mjordan@digium.com> + + * main/stdtime/localtime: Fix warning introduced in r433720 + + The patch in r433720 caused a warning to be kicked back by gcc. It occurred + due to this check in unistd.h: + + if (__nbytes > __bos0 (__buf)) + return __read_chk_warn (__fd, __buf, __nbytes, __bos0 (__buf)); + + That is, if __nbytes is greater than the result of GCC's built-in object size + for the struct, we'll kick back a warning. + + As it turns out, this is because there is an error in the code in the patch. + We are passing the address of the pointer to the struct, not iev, which is a + pointer to the struct. Hence, the number of bytes is probably going to be lot + larger than the number of bytes that make up a pointer! This patch changes + the code just read from the pointer to the struct - which fixes the warning. + + ASTERISK-24917 + ........ + + Merged revisions 433743 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 20:56 +0000 [47eeb67e14] dkdegroot (License 6600) + + * clang compiler warnings: Ignore -Wunused-command-line-argument + + Asterisk's build system has a tendency to pass include directives for libraries + to everything compiled within a particular group of source files. This means + we pass the header for libxml2 to things that don't necessarily need it. As a + result, we ignore this particular warning. + + Review: https://reviewboard.asterisk.org/r/4545/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4545.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433720 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 20:52 +0000 [dbb4d6f9e7] dkdegroot (License 6600) + + * clang compiler warnings: Fix warning for -Wgnu-variable-sized-type-not-at-end + + This patch fixes a warning caught by clang, wherein a variable sized struct is + not located at the end of a struct. While the code in question actually + expected this, this is a good warning to watch for. Hence, this patch refactors + the code in question to not have two variable length elements in the same + struct. + + Review: https://reviewboard.asterisk.org/r/4530/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4530.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433717 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:56 +0000 [e126ab9eeb] dkdegroot (License 6600) + + * clang compiler warnings: Fix a variety of "unused" warnings + + This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable + errors caught by clang. Specifically: + + * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], + qsmp_cmd_usage[] + * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" + * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" + * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" + * funcs/func_env.c:729: Fixed ast_str_append_substr. + * main/editline/np/strlcat.c: removed unused rcsid variable + * main/editline/np/strlcpy.c: removed unused rcsid variable + * main/security_events.c: removed unused TIMESTAMP_STR_LEN + * utils/conf2ael.c: removed unused cfextension_states + * utils/extconf.c: removed unused cfextension_states + + Review: https://reviewboard.asterisk.org/r/4526 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4526.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:48 +0000 [2f6534527d] dkdegroot (License 6600) + + * clang compiler warnings: Fix -Wself-assign + + Assigning a variable to itself isn't super useful. However, the WAV format + modules make use of this in order to perform byte endian checks. This patch + works around the warning by only performing the self assignment if we are + going to do more than just assign it to ourselves. Which is odd, but true. + + Review: https://reviewboard.asterisk.org/r/4544/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4544.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433690 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:40 +0000 [eb70993a50] dkdegroot (License 6600) + + * clang compiler warnings: Fix -Wparantheses-equality warnings + + Clang will treat ((a == b)) as a warning, as it reasonably expects that the + developer may have intended to write (a == b) or ((a = b)). This patch cleans + up all instances where equality, not assignment, was intended between two + parantheses. + + Review: https://reviewboard.asterisk.org/r/4531/ + + ASTERISK-24917 + Repoted by: dkdegroot + patches: + rb4531.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:31 +0000 [c0ff16036a] dkdegroot (License 6600) + + * clang compiler warnings: Fix -Wbitfield-constant-conversion warning + + In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by + clang, as it will truncate the -1 to a 1 implicitly. + + Instead, we just assign the value a '1'. + + Review: https://reviewboard.asterisk.org/r/4537/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4537.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433683 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:27 +0000 [844bc76bef] dkdegroot (License 6600) + + * clang compiler warnings: Fix -Winitializer-overrides + + This patch fixes clange compiler warnings for initializer overrides. + Specifically: + + res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration + value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing + those enum values, we therefore initialize the value twice to two different + values, "tlsv1" and "default". This patch changes it to just initialize + the index in the array to "tlsv1". + + Review: https://reviewboard.asterisk.org/r/4539/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4539.patch submitted by dkdegroot (License 6600) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:19 +0000 [5e204042d9] dkdegroot (License 6600) + + * clang compiler warnings: Fix -Wunused-function; make inline function static + + This patch fixes clang compilers warnings for unused functions. Specifically: + * channels/chan_iax2: removed user_ref function + * main/dsp.c: removed goertzel_update function + * main/config.c: made variable_list_switch static + + Review: https://reviewboard.asterisk.org/r/4527 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4527.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433678 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 17:34 +0000 [cfbf5fbe91] Jonathan Rose <jrose@digium.com> + + * SAC: Add a few basic queues + + Review: https://reviewboard.asterisk.org/r/4503/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 17:25 +0000 [1a50d8d4c2] Jonathan Rose <jrose@digium.com> + + * SAC: Add conferencing extensions and configuration + + Review: https://reviewboard.asterisk.org/r/4504/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 16:15 +0000 [c6c08d755d] Rusty Newton <rnewton@digium.com> + + * configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2 + + Example configuration files for a "basic PBX" deployment for the fictitious + Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/ + and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company + + Patch 4488 includes all functionality needed for SAC's outside connectivity + and some externally accessed features, as well as outbound dialing. + + Reported by: Malcolm Davenport + Tested by: Rusty Newton + + Review: https://reviewboard.asterisk.org/r/4488/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 16:04 +0000 [13557675d4] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage. + + * Converted the contact_autoexpire container to use the ao2 template hash + and cmp functions. Also made use the OBJ_SEARCH_xxx names instead of the + deprecated names. + + * Eliminates several unnecessary uses of RAII_VAR(). + + Review: https://reviewboard.asterisk.org/r/4524/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 15:30 +0000 [85feac857c] Mark Michelson <mmichelson@digium.com> + + * Add stateful PJSIP response API call, and use it for out-of-dialog responses. + + Asterisk had an issue where retransmissions of MESSAGE requests resulted in + Asterisk processing the retransmission as if it were a new MESSAGE request. + + This patch fixes the issue by creating a transaction in PJSIP on the incoming + request. This way, if a retransmission arrives, the PJSIP transaction layer + will resend the response and Asterisk will not ever see the retransmission. + + ASTERISK-24920 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4532/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 12:50 +0000 [dc2cf21144] Richard Mudgett <rmudgett@digium.com> + + * res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown. + + Contact expiration object refs were leaked when the module was unloaded. + + * Made empty the scheduler of entries before destroying it to release the + object ref held by the scheduler entry. + + Review: https://reviewboard.asterisk.org/r/4523/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 09:41 +0000 [6e6f5b3a1f] scsiguy (License 6692) + + * res/res_timing_kqueue: Update the module to conform to current timer API + + This patch updates the kqueue timing module to conform to current timer API. + + This fixes issues with using the kqueue timing source on Asterisk 13 on + FreeBSD 10. These issues include: + + - Remove support for kevent64(). The values used to support Asterisk timers + fit within 32bits and so can be handled on all platforms via kevent(). + + - Provide debug logging for, but do not track, unacked events. This matches + the behavior of all other timer implementations. + + - Implement continuous mode by triggering and leaving active, a user event. + This ensures that the file descriptor for the timer returns immediately from + poll(), without placing the load of a high speed timer on the kernel. + + - In kqueue_timer_get_max_rate(), don't overstate the capability of the timer. + On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer + type kqueue supports for timers. + + - In kqueue_timer_get_event(), assume the caller woke up from poll() and just + return the mode the timer is currently in. This matches all other timer + implementations. + + - Adjust the test code now that unacked events are not tracked. + + Review: https://reviewboard.asterisk.org/r/4465/ + + ASTERISK-24857 #close + Reported by: scsiguy + Tested by: Ed Hynan + patches: + rb4465.patch submitted by scsiguy (License 6692) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 07:26 +0000 [b0df413fb2] Corey Farrell <git@cfware.com> + + * Fix link error for utils/aelparse. + + Use the standard ast_log instead of ast_log_safe for STANDALONE programs. + + Review: https://reviewboard.asterisk.org/r/4538/ + ........ + + Merged revisions 433549 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 02:09 +0000 [d01706ce1e] Corey Farrell <git@cfware.com> + + * Improved and portable ast_log recursion avoidance + + This introduces a new logger routine ast_log_safe. This routine should be + used for all error messages in code that can be run as a result of ast_log. + ast_log_safe does nothing if run recursively. All error logging in + astobj2.c, strings.c and utils.h have been switched to ast_log_safe. + + This required adding support for raw threadstorage. This provides direct + access to the void* pointer in threadstorage. In ast_log_safe, NULL is used + to signify that this thread is not already running ast_log_safe, (void*)1 when + it is already running. This was done since it's critical that ast_log_safe + do nothing that could log during recursion checking. + + ASTERISK-24155 #close + Reported by: Timo Teräs + Review: https://reviewboard.asterisk.org/r/4502/ + ........ + + Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 18:07 +0000 [4b225e2104] Corey Farrell <git@cfware.com> + + * Fix compile errors caused by r4500 / r4501. + + * Add ast_register_cleanup to utils/clicompat.c to deal with + any utils that copy sources from main. + * Asterisk 13+: remove unused variables from core_local.c. + + Review: https://reviewboard.asterisk.org/r/4534/ + ........ + + Merged revisions 433499 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 17:19 +0000 [6adf26f14d] Corey Farrell <git@cfware.com> + + * Replace most uses of ast_register_atexit with ast_register_cleanup. + + Since 'core stop now' and 'core restart now' do not stop modules, + it is unsafe for most of the core to run cleanups. Originally all + cleanups used ast_register_atexit, and were only changed when it + was shown to be unsafe. ast_register_atexit is now used only when + absolutely required to prevent corruption and close child processes. + + Exceptions that need to use ast_register_atexit: + * CDR: Flush records. + * res_musiconhold: Kill external applications. + * AstDB: Close the DB. + * canary_exit: Kill canary process. + + ASTERISK-24142 #close + Reported by: David Brillert + + ASTERISK-24683 #close + Reported by: Peter Katzmann + + ASTERISK-24805 #close + Reported by: Badalian Vyacheslav + + ASTERISK-24881 #close + Reported by: Corey Farrell + + Review: https://reviewboard.asterisk.org/r/4500/ + Review: https://reviewboard.asterisk.org/r/4501/ + ........ + + Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 12:46 +0000 [d0df545a44] Corey Farrell <git@cfware.com> + + * res_pjsip: Enable unload of all modules at shutdown. + + * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes + caused by running PJSIP functions from non-PJSIP threads. + * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing + crashes in some cases. In theory pj_shutdown() should take care of this. + * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at + shutdown. + * Resolve leaked config global in res_pjsip_notify. + * Unregister pubsub pjsip service module. + * Implement cleanup for res_pjsip_session. + + ASTERISK-24731 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4498/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 12:04 +0000 [fd434a210f] Kevin Harwell <kharwell@digium.com> + + * app_confbridge: file playback blocks dtmf + + Attempting to execute DTMF in a confbridge while file playback (prompt, + announcement, etc) is occurring is not allowed. You have to wait until + the sound file has completed before entering DTMF. This patch fixes it + so that app_confbridge now monitors for dtmf key presses during menu + driven file playback. If a key is pressed playback stops and it executes + the matched menu option. + + ASTERISK-24864 #close + Reported by: Steve Pitts + Review: https://reviewboard.asterisk.org/r/4510/ + ........ + + Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-25 13:37 +0000 [dea885a607] Richard Mudgett <rmudgett@digium.com> + + * A couple minor cleanup tweaks. + + * In res/res_sorcery_realtime.c: Broke long line. + + * In main/bucket.c: Eliminated unnecessary NULL check as + ast_sorcery_unref() is NULL tolerant and set the global object to NULL + after unref in the system shutdown bucket_cleanup(). + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433420 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-25 10:30 +0000 [05de9082a5] Simon Arlott (License 5756) + + * res_xmpp: Buddies are always auto-registered when processing the roster + + Due to a quirk in the configuration handling of res_xmpp, the 'autoregister' + setting was never actually processed. This was due to not properly copying + over the global settings to the client settings when applying the + configuration to the run-time object. + + Review: https://reviewboard.asterisk.org/r/4496/ + + ASTERISK-14233 + ASTERISK-24780 #close + Reported by: Simon Arlott + patches: + asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756) + ........ + + Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433396 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-24 14:26 +0000 [b1e9552b08] Richard Mudgett <rmudgett@digium.com> + + * chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. + + Incoming PJSIP call legs that have not been answered yet send unnecessary + "180 Ringing" or "183 Progress" messages every time a connected line + update happens. If the outgoing channel is also PJSIP then the incoming + channel will always send a "180 Ringing" or "183 Progress" message when + the outgoing channel sends the INVITE. + + Consequences of these unnecessary messages: + + * The caller can start hearing ringback before the far end even gets the + call. + + * Many phones tend to grab the first connected line information and refuse + to update the display if it changes. The first information is not likely + to be correct if the call goes to an endpoint not under the control of the + first Asterisk box. + + When connected line first went into Asterisk in v1.8, chan_sip received an + undocumented option "rpid_immediate" that defaults to disabled. When + enabled, the option immediately passes connected line update information + to the caller in "180 Ringing" or "183 Progress" messages as described + above. + + * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or + "183 Progress" messages. The default is "no" to disable sending the + unnecessary messages. + + ASTERISK-24781 #close + Reported by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4473/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-23 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.3.0-rc1 Released. + +2015-03-22 23:58 +0000 [r433247-433269] Matthew Jordan <mjordan@digium.com> + + * apps/app_queue.c, main/cli.c, main/cdr.c, main/manager.c, + main/rtp_engine.c, /, funcs/func_cdr.c: Fix compilations errors + on 64-bit OpenBSD systems In versiong 5.5, OpenBSD went to 64-bit + time values. This requires a cast to (long) when printing members + of certain time structs. Review: + https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close + Reported by: snuffy Tested by: snuffy patches: + openbsd-time64.diff uploaded by snuffy (License 5024) ........ + Merged revisions 433268 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * main/asterisk.c, main/loader.c, main/xmldoc.c, /: Fix compilation + issues for OpenBSD This patch addresses compilation issues for + OpenBSD. Specifically, it addresses: * It allows including + <sys/vmmeter.h> in asterisk.c * Provides a needed (size_t) cast + in xmldoc.c In 13+, it also addresses a conditional inclusion in + loader.c. Review: https://reviewboard.asterisk.org/r/4506 + ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy + patches: misc-openbsd.diff uploaded by snuffy (License 5024) + ........ Merged revisions 433245 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-20 19:52 +0000 [r433199-433222] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_messaging.c, res/res_pjsip/pjsip_options.c, + res/res_pjsip.c, res/res_pjsip_nat.c: Audit + ast_pjsip_rdata_get_endpoint() usage for ref leaks. Valgrind + found some memory leaks associated with + ast_pjsip_rdata_get_endpoint(). The leaks would manifest when + sending responses to OPTIONS requests, processing MESSAGE + requests, and res_pjsip supplements implementing the + incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint() + endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(), + res/res_pjsip/pjsip_options.c:send_options_response(), + res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and + res/res_pjsip_messaging.c:send_response(). * Eliminated + RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in + res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but + benign return value in + res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review: + https://reviewboard.asterisk.org/r/4511/ + + * res/res_pjsip_sdp_rtp.c, main/sorcery.c, main/xmldoc.c: + res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak + respectively. Valgrind found a memory leak and invalid access. * + Fix invalid access by sscanf() being fed a non-nul terminated + string of digits in res/res_pjsip_sdp_rtp.c:get_codecs(). * Fix + memory leak in main/sorcery.c:sorcery_object_field_destructor(). + * Fix potential NULL pointer dereference in + main/xmldoc.c:xmldoc_get_syntax_config_option(). Review: + https://reviewboard.asterisk.org/r/4513/ + +2015-03-19 19:19 +0000 [r433174] Matthew Jordan <mjordan@digium.com> + + * funcs/func_env.c, tests/test_func_file.c, /: funcs/func_env: Fix + regression caused in FILE read operation When r432935 was merged, + it did correctly fix a situation where a FILE read operation on + the middle of a file buffer would not read the requested length + in the parameters passed to the FILE function. Unfortunately, it + would also allow the FILE function to append more bytes than what + was available in the buffer if the length exceeded the end of the + buffer length. This patch takes the minimum of the remaining + bytes in the buffer along with the calculated length to append + provided by the original patch, and uses that as the length to + append in the return result. This patch also updates the unit + tests with the scenarios that were originally pointed out in + ASTERISK-21765 that the original implementation treated + incorrectly. ASTERISK-21765 ........ Merged revisions 433173 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-19 10:20 +0000 [r433113-433126] Corey Farrell <git@cfware.com> + + * main/logger.c, /: logger: Apply default console logging when + configuration cannot be loaded. When logger.conf is missing or + invalid enable console logging and display an error message. + ASTERISK-24817 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4497/ ........ Merged + revisions 433122 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * channels/sip/include/dialog.h, channels/chan_sip.c, + channels/sip/include/sip.h: chan_sip: Simplify dialog/peer + references, improve REF_DEBUG output. * Replace functions for + ref/undef of dialogs and peers with macro's to call + ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller + information to sip_alloc and find_call. ASTERISK-24882 #close + Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4189/ + + * /, channels/chan_sip.c: chan_sip: Fix dialog reference leaked to + scheduler for reinvite_timeout. Release the scheduler reference + to the dialog for reinvite timeout during dialog_unlink_all. + ASTERISK-24876 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4491/ ........ Merged + revisions 433112 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-18 02:34 +0000 [r433088] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_session.c: res_pjsip_session: Fix off-nominal extra + unref of session. + +2015-03-17 22:15 +0000 [r433060-433064] Scott Griepentrog <sgriepentrog@digium.com> + + * main/asterisk.c, main/config.c, main/xmldoc.c, main/manager.c, + include/asterisk/config.h, main/utils.c, main/codec_builtin.c, + main/endpoints.c: Various: bugfixes found via chaos Using + DEBUG_CHAOS several instances of a null pointer crash, and one + uninitialized variable were uncovered and fixed. Also added + details on why Asterisk failed to initialize. Review: + https://reviewboard.asterisk.org/r/4468/ + + * build_tools/cflags.xml, include/asterisk/utils.h: core: Introduce + chaos into memory allocations Locate potential crashes by + exercising seldom used code paths. This patch introduces a new + define DEBUG_CHAOS, and mechanism to randomly return an error + condition from functions that will seldom do so. Functions that + handle the allocation of memory get the first treatment. Review: + https://reviewboard.asterisk.org/r/4463/ + +2015-03-17 21:49 +0000 [r433057] Richard Mudgett <rmudgett@digium.com> + + * main/netsock2.c, /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c, + apps/app_externalivr.c, res/res_pjsip_acl.c: Audit + ast_sockaddr_resolve() usage for memory leaks. Valgrind found + some memory leaks associated with ast_sockaddr_resolve(). Most of + the leaks had already been fixed by earlier memory leak hunt + patches. This patch performs an audit of ast_sockaddr_resolve() + and found one more. * Fix ast_sockaddr_resolve() memory leak in + apps/app_externalivr.c:app_exec(). * Made + main/netsock2.c:ast_sockaddr_resolve() always set the addrs + parameter for safety so the pointer will never be uninitialized + on return. The same goes for + res/res_pjsip_acl.c:extract_contact_addr(). * Made functions that + call ast_sockaddr_resolve() with RAII_VAR() controlling the addrs + variable use ast_free instead of ast_free_ptr to provide better + MALLOC_DEBUG information. Review: + https://reviewboard.asterisk.org/r/4509/ ........ Merged + revisions 433056 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-17 18:34 +0000 [r433028-433031] Kevin Harwell <kharwell@digium.com> + + * include/asterisk/res_pjsip.h, + res/res_pjsip_endpoint_identifier_anonymous.c, + res/res_pjsip_endpoint_identifier_ip.c, res/res_pjsip.c, + res/res_pjsip_endpoint_identifier_user.c: res_pjsip: Allow + configuration of endpoint identifier query order Updated some + documentation stating that endpoint identifiers registered + without a name are place at the front of the lookup list. Also + renamed register method + 'ast_sip_register_endpoint_identifier_by_name' to + 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840 + Reported by: Mark Michelson + + * configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c, + res/res_pjsip_endpoint_identifier_user.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + res/res_pjsip_endpoint_identifier_anonymous.c, + res/res_pjsip_endpoint_identifier_ip.c, + contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py: + res_pjsip: Allow configuration of endpoint identifier query order + This patch fixes previously reverted code that caused binary + incompatibility problems with some modules. And like the original + patch it makes sure that no matter what order the endpoint + identifier modules were loaded, priority is given based on the + ones specified in the new global 'endpoint_identifier_order' + option. ASTERISK-24840 Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/4489/ + +2015-03-17 16:10 +0000 [r433005] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip.c: res_pjsip: Add reason comment. + +2015-03-14 02:28 +0000 [r432971] Matthew Jordan <mjordan@digium.com> + + * /, main/format_cap.c: main/frame: Don't report empty disallow + values as an error In realtime, it is normal to have a database + with both 'allow' and 'disallow' columns in the schema. It is + perfectly valid to have an 'allow' value of '!all,g722,ulaw,alaw' + and no 'disallow' value. Unlike in static conf files, you can't + *not* provide the disallow value. Thus, the empty disallow value + causes a spurious WARNING message, which is kind of annoying. + This patch makes it so that a 'disallow' value with no ... value + ... is ignored. Granted, you can still screw this up as well, as + technically specifying 'disallow=all,!ulaw' allows only ulaw, and + then you would have no 'allow' value in your database. But + really, why would you do that? WHY? ASTERISK-16779 #close + Reported by: Atis Lezdins ........ Merged revisions 432970 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-14 02:00 +0000 [r432945-432949] Joshua Colp <jcolp@digium.com> + + * funcs/func_curl.c, /: func_curl: Don't hold exclusive lock when + performing HTTP request. This code originally kept a lock held + when performing the HTTP request to ensure that the options + provided to curl remain valid. This doesn't seem to be necessary + these days and holding the lock caused requests to happen + sequentially instead of in parallel. ASTERISK-18708 #close + Reported by: Dave Cabot ........ Merged revisions 432948 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * main/cli.c, /: core: Fix tab completion of "core set debug + channel" CLI command. The "core set debug channel" CLI command + mistakenly had source filenames added to its tab completion. This + occurred because the CLI generator fell back to the "core set + debug" command which permits setting debug at a source filename + level. ASTERISK-21038 #close Reported by: Richard Kenner ........ + Merged revisions 432944 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-14 01:21 +0000 [r432920-432938] Matthew Jordan <mjordan@digium.com> + + * /, funcs/func_env.c: FILE: fix retrieval of file contents when + offset is specified The loop that reads in a file was not + correctly using the offset when determining what bytes to append + to the output. This patch corrects the logic such that the + correct portion of the file is extracted when an offset is + specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt + Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by + Di-Shi Sun (License 5076) ........ Merged revisions 432935 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * apps/app_amd.c, /, configs/samples/amd.conf.sample: apps/app_amd: + Document maximum_word_length option; fix AMDCAUSE documentation + This patch corrects the documentation for the AMD application. + Specifically: * It documents the maximum_word_length option, + which limits the maximum allowed length of a single utterance. * + It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. + MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was + undocumented. Thanks to the issue reporter, Frank DiGennaro, for + pointing out the issues. ASTERISK-19470 #close Reported by: Frank + DiGennaro ........ Merged revisions 432918 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-13 17:04 +0000 [r432892-432894] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip/pjsip_configuration.c: chan_pjsip: AMI action + PJSIPShowEndpoint closes AMI connection on error. Also fixed + similar problem with AMI action PJSIPShowEndpoints. + ASTERISK-24872 #close Reported by: Dmitriy Serov Review: + https://reviewboard.asterisk.org/r/4487/ + + * channels/chan_pjsip.c, res/res_pjsip_caller_id.c: + chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and + consistent. The res_pjsip modules were manually checking both + name and number presentation values when there is a function that + determines the combined presentation for a party ID struct. The + function takes into account if the name or number components are + valid while the manual code rarely checked if the data was even + valid. * Made use ast_party_id_presentation() rather than + manually checking party ID presentation values. * Ensure that + set_id_from_pai() and set_id_from_rpid() will not return + presentation values other than what is pulled out of the SIP + headers. It is best if the code doesn't assume that + AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * + Fixed copy paste error in add_privacy_params() dealing with RPID + privacy. * Pulled the id->number.valid test from + add_privacy_header() and add_privacy_params() up into the parent + function add_id_headers() to skip adding PAI/RPID headers + earlier. * Made update_connected_line_information() not send out + connected line updates if the connected line number is invalid. + Lower level code would not add the party ID information and thus + the sent message would be unnecessary. * Eliminated RAII_VAR + usage in send_direct_media_request(). Review: + https://reviewboard.asterisk.org/r/4472/ + +2015-03-13 14:48 +0000 [r432868] Kevin Harwell <kharwell@digium.com> + + * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + res/res_pjsip_endpoint_identifier_anonymous.c, + res/res_pjsip_endpoint_identifier_ip.c, + contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py, + configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c, + res/res_pjsip_endpoint_identifier_user.c: Revert - res_pjsip: + Allow configuration of endpoint identifier query order Due to a + break in binary compatibility with some other modules these + changes are being reverted until the issue can be resolved. + ASTERISK-24840 Reported by: Mark Michelson + +2015-03-12 12:58 +0000 [r432808-432811] Matthew Jordan <mjordan@digium.com> + + * /, main/audiohook.c: main/audiohook: Update internal sample rate + on reads When an audiohook is created (which is used by the + various Spy applications and Snoop channel in Asterisk 13+), it + initially is given a sample rate of 8kHz. It is expected, + however, that this rate may change based on the media that passes + through the audiohook. However, the read/write operations on the + audiohook behave very differently. When a frame is written to the + audiohook, the format of the frame is checked against the + internal sample rate. If the rate of the format does not match + the internal sample rate, the internal sample rate is updated and + a new SLIN format is chosen based on that sample rate. This works + just fine. When a frame is read, however, we do something quite + different. If the format rate matches the internal sample rate, + all is fine. However, if the rates don't match, the audiohook + attempts to "fix up" the number of samples that were requested. + This can result in some seriously large number of samples being + requested from the read/write factories. Consider the worst case + - 192kHz SLIN. If we attempt to read 20ms worth of audio produced + at that rate, we'd request 3840 samples (192000 / (1000 / 20)). + However, if the audiohook is still expecting an internal sample + rate of 8000, we'll attempt to "fix up" the requested samples to: + samples_converted = samples * (ast_format_get_sample_rate(format) + / (float) audiohook->hook_internal_samp_rate); which is: 92160 = + 3840 * (192000 / 8000) This results in us attempting to read + 92160 samples from our factories, as opposed to the 3840 that we + actually wanted. On a 64-bit machine, this miraculously survives + - despite allocating up to two buffers of length 92160 on the + stack. The 32-bit machines aren't quite so lucky. Even in the + case where this works, we will either (a) get way more samples + than we wanted; or (b) get about 3840 samples, assuming the + timing is pretty good on the machine. Either way, the calculation + being performed is wrong, based on the API users expectations. My + first inclination was to allocate the buffers on the heap. As it + is, however, there's at least two drawbacks with doing this: (1) + It's a bit complicated, as the size of the buffers may change + during the lifetime of the audiohook (ew). (2) The stack is + faster (yay); the heap is slower (boo). Since our calculation is + flat out wrong in the first place, this patch fixes this issue by + instead updating the internal sample rate based on the format + passed into the read operation. This causes us to read the + correct number of samples, and has the added benefit of setting + the audihook with the right SLIN format. Note that this issue was + caught by the Asterisk Test Suite as a result of r432195 in the + 13 branch. Because this issue is also theoretically possible in + Asterisk 11, the change is being made here as well. Review: + https://reviewboard.asterisk.org/r/4475/ ........ Merged + revisions 432810 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * Makefile, include/asterisk/utils.h, /, configure, main/Makefile, + configure.ac, include/asterisk/inline_api.h, makeopts.in: Add + support for the clang compiler; update RAII_VAR to use + BlocksRuntime RAII_VAR, which is used extensively in Asterisk to + manage reference counted resources, uses a GCC extension to + automatically invoke a cleanup function when a variable loses + scope. While this functionality is incredibly useful and has + prevented a large number of memory leaks, it also prevents + Asterisk from being compiled with clang. This patch updates the + RAII_VAR macro such that it can be compiled with clang. It makes + use of the BlocksRuntime, which allows for a closure to be + created that performs the actual cleanup. Note that this does not + attempt to address the numerous warnings that the clang compiler + catches in Asterisk. Much thanks for this patch goes to: * The + folks on StackOverflow who asked this question and Leushenko for + providing the answer that formed the basis of this code: + http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang + * Diederik de Groot, who has been extremely patient in working on + getting this patch into Asterisk. Review: + https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133 + ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by: + Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik + de Groot (License 6600) ........ Merged revisions 432807 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-11 16:38 +0000 [r432764-432787] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip/config_domain_aliases.c, + res/res_pjsip/include/res_pjsip_private.h, + include/asterisk/res_pjsip.h: res_pjsip: Move internal + init/destroy prototypes to private header file. Done as a + separate commit from a finding in + https://reviewboard.asterisk.org/r/4467/ + + * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix pjsip.conf + type=global object default value handling. When a type=global + section is not defined in pjsip.conf the global defaults are not + applied. As a result the mandatory Max-Forwards header is not + added to SIP messages for res_pjsip/chan_pjsip. The handling of + pjsip.conf type=global objects has several problems: 1) If the + global object is missing the defaults are not applied. 2) If the + global object is missing the default_outbound_endpoint's default + value is not returned by + ast_sip_global_default_outbound_endpoint(). 3) Defines are needed + so default values only need to be changed in one place. * Added a + sorcery instance observer callback to check if there were any + type=global sections loaded. If there were more than one then + issue an error message. If there were none then apply the global + defaults. * Fixed ast_sip_global_default_outbound_endpoint() to + return the documented default when no type=global object is + defined. * Made defines for the global default values. * + Increased the default_useragent[] size because SVN version + strings can get lengthy and 128 characters may not be enough. * + Fixed an off-nominal code path ref leak in global_alloc() if the + string fields fail to initialize. * Eliminated RAII_VAR in + get_global_cfg() and ast_sip_global_default_outbound_endpoint(). + ASTERISK-24807 #close Reported by: Anatoli Review: + https://reviewboard.asterisk.org/r/4467/ + + * res/res_pjsip/pjsip_global_headers.c: res_pjsip: Fixed invalid + empty Server and User-Agent SIP headers. Setting pjsip.conf + useragent to an empty string results in an empty SIP header being + sent. * Made not add an empty SIP header item to the global SIP + headers list. Review: https://reviewboard.asterisk.org/r/4467/ + +2015-03-10 23:09 +0000 [r432742] Joshua Colp <jcolp@digium.com> + + * main/stasis_channels.c, main/endpoints.c, main/stasis_bridges.c: + core: Don't create snapshots with locks. Snapshots are immutable + and are never changed. Allocating them with a lock is wasteful. + Review: https://reviewboard.asterisk.org/r/4469/ + +2015-03-10 21:33 +0000 [r432693-432721] Matthew Jordan <mjordan@digium.com> + + * res/res_config_odbc.c, /: res/res_config_odbc: Fix improper + escaping of backslashes with MySQL When escaping backslashes with + MySQL, the proper way to escape the characters in a LIKE clause + is to escape the '\' four times, i.e., '\\\\'. To quote the MySQL + manual: "Because MySQL uses C escape syntax in strings (for + example, “\n” to represent a newline character), you must double + any “\” that you use in LIKE strings. For example, to search for + “\n”, specify it as “\\n”. To search for “\”, specify it as + “\\\\”; this is because the backslashes are stripped once by the + parser and again when the pattern match is made, leaving a single + backslash to be matched against." ASTERISK-24808 #close Reported + by: Javier Acosta patches: res_config_odbc.diff uploaded by + Javier Acosta (License 6690) ........ Merged revisions 432720 + from http://svn.asterisk.org/svn/asterisk/branches/11 + + * apps/app_voicemail.c, /: app_voicemail: Fix crash with IMAP + backends when greetings aren't present When an IMAP backend is in + use and greetings are set to be used, but aren't present for a + user in their IMAP folder, Asterisk will crash. This occurs due + to the mailstream being set to the 'greetings' folder and being + left in that particular state, regardless of the success/failure + of the attempt to access the folder the mailstream points to. + Later access of the mailstream assumes that it points to the + 'INBOX' (or some other folder), resulting in either a crash (if + the greetings folder didn't exist and the mailstream is invalid) + or an inability to read messages from the 'INBOX' folder. This + patch restores the mailstream to its correct state after + accessing the greetings. This fixes the crash, and sets the + mailstream to the state that VoiceMailMain expects. Note that + while ASTERISK-23390 also contained a patch for this issue, the + patch on ASTERISK-24786 is the one being merged here. Review: + https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close + Reported by: Ben Smithurst ASTERISK-24786 #close Reported by: + Graham Barnett Tested by: Graham Barnett patches: + app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett + (License 6685) ........ Merged revisions 432695 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, main/stdtime/localtime.c: localtime: Fix file descriptor leak + on kqueue(2) systems The localtime management in the Asterisk + core contains a thread that watches for changes in the local + timezone. On systems where the directory containing + /etc/localtime is modified frequently, the thread monitoring the + changes will be woken up to determine if any changes in timezone + have occurred. When using kqueue(2), this can cause a leak of + file descriptors due to some improper management of resources. + This patch updates the kqueue(2) handling in localtime, such that + is no longer leaks resources. Review: + https://reviewboard.asterisk.org/r/4450/ ASTERISK-24739 #close + Reported by: Ed Hynan patches: 11.15.0-u.diff uploaded by Ed + Hynan (Licnese 6680) 11.7.0-u.diff uploaded by Ed Hynan (License + 6680) svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License + 6680) ........ Merged revisions 432691 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-10 16:04 +0000 [r432668] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_refer.c, res/res_pjsip_session.c, + include/asterisk/res_pjsip_session.h, + res/res_pjsip_session.exports.in: res_pjsip_refer: Fix occasional + unexpected BYE sent after receiving a REFER. A race condition + happened between initiating a transfer and requesting that a + dialog termination be delayed. Occasionally, the transferrer + channels would exit the bridge and hangup before the dialog + termination delay was requested. * Made request dialog + termination delay before initiating the transfer action. If the + transfer fails then cancel the delayed dialog termination + request. ASTERISK-24755 #close Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/4460/ + +2015-03-09 16:12 +0000 [r432638] Kevin Harwell <kharwell@digium.com> + + * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + res/res_pjsip_endpoint_identifier_anonymous.c, + res/res_pjsip_endpoint_identifier_ip.c, + contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py + (added), configs/samples/pjsip.conf.sample, CHANGES, + res/res_pjsip.c, res/res_pjsip_endpoint_identifier_user.c: + res_pjsip: Allow configuration of endpoint identifier query order + It's possible to have a scenario that will create a conflict + between endpoint identifiers. For instance an incoming call could + be identified by two different endpoint identifiers and the one + chosen depended upon which identifier module loaded first. This + of course causes problems when, for example, the incoming call is + expected to be identified by username, but instead is identified + by ip. This patch adds a new 'global' option to res_pjsip called + 'endpoint_identifier_order'. It is a comma separated list of + endpoint identifier names that specifies the order by which + identifiers are processed and checked. ASTERISK-24840 #close + Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/4455/ + +2015-03-08 01:46 +0000 [r432614] Joshua Colp <jcolp@digium.com> + + * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix wrongful use of + USE_PJPROJECT define. As pjproject is now used as a shared + library a different define, HAVE_PJPROJECT, is used to specify if + pjproject is present. ASTERISK-24830 #close Reported by: Stefan + Engström + +2015-03-06 22:50 +0000 [r432574-432594] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_refer.c: res_pjsip_refer: Make safely get the + context for a blind transfer. Made safely get the + TRANSFER_CONTEXT channel value while the channel is locked in + refer_incoming_attended_request() and + refer_incoming_blind_request(). The pointer returned by + pbx_builtin_getvar_helper() is only valid while the channel is + locked. + + * res/res_pjsip_refer.c: res_pjsip_refer: Made + refer_attended_alloc() not create the ao2 object with a lock. The + lock is unused. + +2015-03-06 21:11 +0000 [r432556] Jonathan Rose <jrose@digium.com> + + * include/asterisk/app.h, main/app.c: app: Add functions to swap + voicemail function table for testing purposes + +2015-03-06 20:18 +0000 [r432528-432534] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_dahdi.h, channels/sig_analog.h, UPGRADE.txt: + chan_dahdi/sig_analog: Fix distinctive ring detection to suck + less. The distinctive ring feature interferes with detecting + Caller ID and appears to have been broken for years. What happens + is if you have a ring-ring cadence as used in the UK you get too + many DAHDI events for the distinctive ring pattern array and + Caller ID detection is aborted. I think when Zapata/DAHDI added + the ring begin event it broke distinctive ring. More events + happen than before and the code does no filtering of which event + times are recorded in the pattern array. * Made distinctive ring + only record the ringt count when the ring ends instead of on just + any DAHDI event. Distinctive ring can be ring, ring-ring, + ring-ring-ring, or different ring durations for the up to three + rings. * Fixed the distinctive ring detection enable + (chan_dahdi.conf option usedistinctiveringdetection) to be per + port instead of somewhat per port and somewhat global. This has + been broken since v1.8. * Fixed using the default distinctive + ring context when the detected pattern does not match any + configured dringX patterns. The default context did not get set + when the previous call was a matched distinctive ring pattern and + the current call is not matched. This has been broken since v1.8. + * Made distinctive ring have no effect on Caller ID detection + when it is disabled. Caller ID detection just monitors for 10 + seconds before giving up. * Fixed leak of struct callerid_state + memory when a polarity reversal during Caller ID detection causes + the incoming call to be aborted. DAHDI-1143 AST-1545 + ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588 + Reported by: Daniel Flounders Review: + https://reviewboard.asterisk.org/r/4444/ ........ Merged + revisions 432530 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, channels/chan_sip.c: chan_sip: Fix realtime locking inversion + when poking a just built peer. When a realtime peer is built it + can cause a locking inversion when the just built peer is poked. + If the CLI command "sip show channels" is periodically executed + then a deadlock can happen because of the locking inversion. * + Push the peer poke off onto the scheduler thread to avoid the + locking inversion of the just built realtime peer. AST-1540 + ASTERISK-24838 #close Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/4454/ ........ Merged + revisions 432526 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-05 16:38 +0000 [r432485] George Joseph <george.joseph@fairview5.com> + + * apps/app_voicemail.c, /: app_voicemail: Fix compile breaking in + app_voicemail with IMAP_STORAGE. There is a leftover "assert" in + app_voicemail/__messagecount that references variables that don't + exist. This causes the compile to fail when --enable-dev-mode and + IMAP_STORAGE are selected. This patch removes the assert. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4461/ ........ Merged + revisions 432484 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-03-04 18:52 +0000 [r432453] Matthew Jordan <mjordan@digium.com> + + * main/translate.c: translate: Prevent invalid memory accesses on + fast shutdown When a 'core restart now' or 'core stop now' is + executed and a channel is currently in a media operation, the + translator matrix can be destroyed while a channel is currently + blocked on getting the best translation choice (see + ast_translator_best_choice). When the channel gets the mutex, the + translation matrix now has invalid memory, and Asterisk crashes. + This patch does two things: (1) We now only clean up the + translation matrix on a graceful shutdown. In that case, there + are no channels, and so there is no risk of this occurring. (2) + We also now set the __matrix and __indextable to NULL. In some + initial backtraces when this occurred, it looked as if there was + a memory corruption occurring, and it wasn't until we determined + that something had restarted Asterisk that the issue became + clear. By setting these to NULL on shutdown, it becomes a bit + easier to determine why a crash is occurring. Note that we could + litter the code with NULL checks on the __matrix, but the act of + making the translation matrix cleaned up on shutdown should + preclude this issue from occurring in the first place, and this + part of the code needs to be as fast as possible. Review: + https://reviewboard.asterisk.org/r/4457/ + +2015-03-02 19:14 +0000 [r432423] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert portion of + r432195 Unfortunately, while initial testing with ConfBridge did + not reproduce the audio problem alluded to in the comment in + res_pjsip_sdp_rtp, further testing did show that bridge_softmix + and/or ConfBridge has a severe problem bridging two or more + participants at different sampling rates. Sometimes, it even + picks odd sampling rates that cause hideous audio problems. This + patch backs out the offending portion of the code until the + issues in the affected bridging modules can be more properly + analyzed. ASTERISK-24841 + +2015-02-27 18:23 +0000 [r432404] Richard Mudgett <rmudgett@digium.com> + + * main/json.c, rest-api/api-docs/endpoints.json, + res/ari/resource_endpoints.c, res/res_ari_endpoints.c, + include/asterisk/json.h, res/ari/resource_channels.c: ARI: Fix + crash if integer values used in JSON payload 'variables' object. + Sending the following ARI commands caused Asterisk to crash if + the JSON body 'variables' object passes values of types other + than strings. POST /ari/channels POST /ari/channels/{channelid} + PUT /ari/endpoints/sendMessage PUT + /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated + RAII_VAR usage in ast_ari_channels_originate_with_id(), + ast_ari_channels_originate(), ast_ari_endpoints_send_message(), + and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751 + #close Reported by: jeffrey putnam Review: + https://reviewboard.asterisk.org/r/4447/ + +2015-02-26 18:52 +0000 [r432385] Scott Griepentrog <sgriepentrog@digium.com> + + * include/asterisk/dial.h, main/dial.c: Dial API: add self destruct + option when complete This patch adds a self-destruction option to + the dial api. The usefulness of this is mostly when using async + mode to spawn a separate thread used to handle the new call, + while the calling thread is allowed to go on about other + business. The only alternative to this option would be the + calling thread spawning a new thread, or hanging around itself + waiting to destroy the dial struct after completion. Example of + use (minus error checking): struct ast_dial *dial = + ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL); + ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, + "Echo"); ast_dial_option_global_enable(dial, + AST_DIAL_OPTION_SELF_DESTROY, NULL); ast_dial_run(dial, NULL, 1); + The dial_run call will return almost immediately after spawning + the new thread to run and monitor the dial. If the call is + answered, it is placed into the echo app. When completed, it will + call ast_dial_destroy() on the dial structure. Note that any + allocations made to pass values to ast_dial_set_user_data() or + dial options must be free'd in a state callback function on any + of: AST_DIAL_RESULT_UNASWERED, AST_DIAL_RESULT_ANSWERED, + AST_DIAL_RESULT_HANGUP, or AST_DIAL_RESULT_TIMEOUT. Review: + https://reviewboard.asterisk.org/r/4443/ + +2015-02-26 17:07 +0000 [r432363] Kevin Harwell <kharwell@digium.com> + + * /, apps/app_chanspy.c, main/channel.c: app_chanspy, channel: fix + frame leaks Fixed a couple of frame leaks that were found during + testing. ASTERISK-24828 #close Reported by: John Hardin Review: + https://reviewboard.asterisk.org/r/4445/ ........ Merged + revisions 432362 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-26 04:58 +0000 [r432321-432342] Matthew Jordan <mjordan@digium.com> + + * /, apps/Makefile, channels/Makefile: make: Remove 'res_features' + from libraries to link against with cygwin/mingw32 Both the apps + and channels Makefiles still listed 'res_features' as modules to + link against when compiling for cygwin or mingw32. This module + hasn't existed for quite some time. ASTERISK-18105 #close + Reported by: feyfre ........ Merged revisions 432341 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, channels/chan_sip.c: channels/chan_sip: Don't send a BYE after + final response when PBX thread fails When Asterisk fails to start + a PBX thread for a new channel - for example, when the maxcalls + setting in asterisk.conf is exceeded - we currently send a final + response, and then attempt to send a BYE request to the UA. Since + that's all sorts of wrong, this patch fixes that by setting + sipalreadygone on the sip_pvt such that we don't get stuck + sending BYE requests to something that does not want it. Note + that this patch is a slight modification of the one on + ASTERISK-15434. For clarity, it explicitly calls sipalreadygone + with the calls to transmit a final response. ASTERISK-21845 + ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt + Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei + (License 5027) ........ Merged revisions 432320 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-25 23:48 +0000 [r432301] Rusty Newton <rnewton@digium.com> + + * configs/basic-pbx/README (added), configs/basic-pbx (added), + configs/basic-pbx/extensions.conf (added), + configs/basic-pbx/logger.conf (added), + configs/basic-pbx/indications.conf (added), + configs/basic-pbx/musiconhold.conf (added), + configs/basic-pbx/asterisk.conf (added), + configs/basic-pbx/pjsip.conf (added), + configs/basic-pbx/modules.conf (added), + configs/basic-pbx/voicemail.conf (added): configs/basic-pbx - + Super Awesome Company example configs Phase 1, Patch 1 Example + configuration files for a "basic PBX" deployment for the + fictitious Super Awesome Company. Details at + https://reviewboard.asterisk.org/r/4379/ and + https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company + Reported by: Malcolm Davenport Tested by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/4379/ + +2015-02-25 23:09 +0000 [r432258-432281] Matthew Jordan <mjordan@digium.com> + + * /, configure, configure.ac: configure: Promote SQLite3 "not + installed" warning to error Since Asterisk won't build without + the library, not having it is definitely an error. Thanks to Kyle + Kurz for pointing this out. ........ Merged revisions 432280 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, channels/chan_sip.c: channels/chan_sip: Clarify WARNING + message in mismatched SRTP scenario When we receive an SDP as + part of an offer/answer for a peer/friend has been configured to + require encryption, and that SDP offer/answer failed to provide + acceptable crypto attributes, we currently issue a WARNING that + uses the phrase "we" and "requested". In this case, both of those + terms are ambiguous - the user will probably think "we" is + Asterisk (it most likely isn't) and it may not be a "request", so + much as an SDP that was received in some fashion. This patch + makes the WARNING messages slightly less bad and a bit more + accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton + ........ Merged revisions 432277 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * main/sdp_srtp.c, /: channels/sip/sdp_crypto: Handle SRTP keys + negotiated with key lifetime/MKI Prior to this patch, SDP offers + negotiating SDES-SRTP crypto attributes would be rejected if + those crypto attributes contained either a key lifetime or a MKI + parameter. While from a theoretical point of view this was + defensible - Asterisk does not support key lifetimes or multiple + crypto keys - from a practical point of view, this is quite a + problem. A large number of endpoints offer lifetimes/MKI, which + Asterisk can tolerate so long as it doesn't actually have to + support anything more than a single key or refresh the key. In + reality, this is (so far as we've seen) always the case. This + patch is a forward port of Olle's work in the + lingon-srtp-key-lifetime-1.8 branch. To quote Olle from + ASTERISK-17721, it handles lifetime/MKI parameters in the + following fashion: > The Lingon branch now handle lifetime and + MKI parameters. > > We only accept lifetimes up to max for the + crypto and higher than 10 hours > for packetization of 20 ms (50 + pps). > > We only handle MKI with index 1. > > We do not really + bother with counting packets and reinviting at end of > lifetime, + so the min of 10 hours kind of takes care of most calls. If there + > are longer ones, we rely on the other side for re-invites. > > + It's still not perfect, but I personally think this is an + improvement. A > configuration option for minimum lifetime + accepted could be added. When the patch was ported forward, I + decided against adding a configuration option as Olle's handling + was more than sufficient for every case I've seen come through + the issue tracker or through interoperability testing. We can + revisit that decision if it proves to be false. A few small other + tweaks were made to the surrounding code to reduce indentation + and provide better type safety for the 'tag' parameter. Review: + https://reviewboard.asterisk.org/r/4419/ Review: + https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close + Reported by: Terry Wilson ASTERISK-17899 #close Reported by: + Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff + uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai + ASTERISK-22748 Reported by: Alejandro Mejia ........ Merged + revisions 432239 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-25 20:44 +0000 [r432237] David M. Lee <dlee@digium.com> + + * res/res_http_websocket.c, /: Increase WebSocket frame size and + improve large read handling Some WebSocket applications, like + [chan_respoke][], require a larger frame size than the default + 8k; this patch bumps the default to 16k. This patch also fixes + some problems exacerbated by large frames. The sanity counter was + decremented on every fread attempt in ws_safe_read(), regardless + of whether data was read from the socket or not. For large + frames, this could result in loss of sanity prior to reading the + entire frame. (16k frame / 1448 bytes per segment = 12 segments). + This patch changes the sanity counter so that it only decrements + when fread() doesn't read any bytes. This more closely matches + the original intention of ws_safe_read(), given that the error + message is "Websocket seems unresponsive". This patch also + properly logs EOF conditions, so disconnects are no longer + confused with unresponsive connections. [chan_respoke]: + https://github.com/respoke/chan_respoke Review: + https://reviewboard.asterisk.org/r/4431/ ........ Merged + revisions 432236 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-24 22:14 +0000 [r432195-432199] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: channels/chan_sip: Fix crash when + transmitting packet after thread shutdown When the monitor thread + is stopped, its pthread ID is set to a specific value + (AST_PTHREADT_STOP) so that later portions of the code can + determine whether or not it is safe to manipulate the thread. + Unfortunately, __sip_reliable_xmit failed to check for that + value, checking instead only for AST_PTHREAD_STOP. Passing the + invalid yet very specific value to pthread_kill causes a crash. + This patch adds a check for AST_PTHREADT_STOP in + __sip_reliable_xmit such that it doesn't attempt to poke the + thread if the thread has already been stopped. ASTERISK-24800 + #close Reported by: JoshE ........ Merged revisions 432198 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * channels/chan_pjsip.c, main/channel.c, res/res_pjsip_sdp_rtp.c, + res/ari/resource_channels.c: ARI/PJSIP: Apply requesting + channel's format cap to created channels This patch addresses the + following problems: * ari/resource_channels: In ARI, we currently + create a format capability structure of SLIN and apply it to the + new channel being created. This was originally done when the PBX + core was used to create the channel, as there was a condition + where a newly created channel could be created without any + formats. Unfortunately, now that the Dial API is being used, this + has two drawbacks: (a) SLIN, while it will ensure audio will + flows, can cause a lot of needless transcodings to occur, + particularly when a Local channel is created to the dialplan. + When no format capabilities are available, the Dial API handles + this better by handing all audio formats to the requsted + channels. As such, we defer to that API to provide the format + capabilities. (b) If a channel (requester) is causing this + channel to be created, we currently don't use its format + capabilities as we are passing in our own. However, the Dial API + will use the requester channel's formats if none are passed into + it, and the requester channel exists and has format capabilities. + This is the "best" scenario, as it is the most likely to create a + media path that minimizes transcoding. Fixing this simply entails + removing the providing of the format capabilities structure to + the Dial API. * chan_pjsip: Rather than blindly picking the first + format in the format capability structure - which actually *can* + be a video or text format - we select an audio format, and only + pick the first format if that fails. That minimizes the weird + scenario where we attempt to transcode between video/audio. * + res_pjsip_sdp_rtp: Applied the joint capapbilites to the format + structure. Since ast_request already limits us down to one format + capability once the format capabilities are passed along, there's + no reason to squelch it here. * channel: Fixed a comment. The + reason we have to minimize our requested format capabilities down + to a single format is due to Asterisk's inability to convey the + format to be used back "up" a channel chain. Consider the + following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u + That is, we have PJSIP/A dialing a Local channel, where the + Local;2 dials PJSIP/B. PJSIP/A has native format capabilities + g722,ulaw,alaw; the Local channel has inherited those format + capabilities down the line; PJSIP/B supports only ulaw. According + to these format capabilities, ulaw is acceptable and should be + selected across all the channels, and no transcoding should + occur. However, there is no way to convey this: when L;2 and + PJSIP/B are put into a bridge, we will select ulaw, but that is + not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A + <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be + written to PJSIP/B. Even if we can convey the 'ulaw' choice back + up the chain (which through some severe hacking in Local channels + was accomplished), such that the chain looks like: PJSIP/A <=> + L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's + *channel driver* to Answer in the SDP back with only 'ulaw'. This + results in all the channel structures being set up correctly, but + PJSIP/A *still* sending g722 and causing the chain to fall apart. + There's a lot of difficulty just in setting this up, as there are + numerous race conditions in the act of bridging, and no clean + mechanism to pass the selected format backwards down an + established channel chain. As such, the best that can be done at + this point in time is clarifying the comment. Review: + https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close + Reported by: Matt Jordan + +2015-02-24 18:32 +0000 [r432175] Kevin Harwell <kharwell@digium.com> + + * /, bridges/bridge_softmix.c: bridge_softmix: G.729 codec license + held When more than one call using the same codec type enters + into a softmix bridge and no audio is present for a channel the + bridge optimizes the out frame by using the same one for all + channels with the same codec type. Unfortunately, when that + number (channels with same codec type) dropped to <= 1 the codec + was not dereferenced. At least not until all parties left the + bridge. Thus in the case of G.729 the license was not released. + This patch ensures that the codec is dereferenced immediately + when the optimization no longer applies. ASTERISK-24797 #close + Reported by: Luke Hulsey Review: + https://reviewboard.asterisk.org/r/4429/ ........ Merged + revisions 432174 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-21 20:47 +0000 [r432118-432154] Joshua Colp <jcolp@digium.com> + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c: res_ari_channels: Return a 404 response + when a requested channel variable does not exist. This change + makes it so that if a channel variable is requested and it does + not exist a 404 response will be returned instead of an + allocation failed response. This makes it easier to debug and + figure out what is going on for a user. ASTERISK-24677 #close + Reported by: Joshua Colp + + * res/res_pjsip_registrar.c: res_pjsip_registrar: Add Expires + header to 200 OK if present in REGISTER. Some implementations + don't pay attention to the expires for individual contacts. In + this case they may consider the lack of an Expires header in the + 200 OK as unregistered. This change makes it so if an Expires + header is present in the REGISTER we will add one in the 200 OK. + ASTERISK-24785 #close Reported by: Ross Beer + + * res/res_pjsip.c: res_pjsip: Add a log message when creating a UAC + dialog to a target URI that is invalid. ASTERISK-24499 #close + Reported by: Rusty Newton + +2015-02-21 17:35 +0000 [r432099] Matthew Jordan <mjordan@digium.com> + + * apps/app_voicemail.c, /: apps/app_voicemail: Demote an ERROR + message to a WARNING message When using IMAP voicemail with + FreePBX, you will often get ERROR messages complaining about not + being able to find a mailbox. This is due to how FreePBX handles + voicemail mailboxes. Unfortunately, app_voicemail has to consider + this a configuration error, as in any other system it would be + indicative of someone misconfiguring their system. Regardless, a + misconfiguration is a WARNING, and not an ERROR. This patch + demotes the message so that system administrators can hopefully + reduce some of the noise in their log files. Note that in the + original patch this was made into a NOTICE, but that's a too + forgiving. ASTERISK-24790 #close Reported by: Graham Barnett + patches: app_voicemail.c.patch_noise uploaded by Graham Barnett + (License 6685) ........ Merged revisions 432098 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-21 14:05 +0000 [r432079] Joshua Colp <jcolp@digium.com> + + * main/http.c, /: http: Add missing html tag to 'httpstatus' + functionality. ASTERISK-24724 #close Reported by: Ashley Sanders + ........ Merged revisions 432078 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-21 02:56 +0000 [r432055-432059] Corey Farrell <git@cfware.com> + + * /, main/bucket.c, main/codec.c, main/loader.c: Allow shutdown to + unload modules that register bucket scheme's or codec's. * Change + __ast_module_shutdown_ref to be NULL safe (11+). * Allow modules + that call ast_bucket_scheme_register or ast_codec_register to be + unloaded during graceful shutdown only (13+ only). ASTERISK-24796 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4428/ ........ Merged + revisions 432058 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, include/asterisk/lock.h: asterisk/lock.h: Fix syntax errors + for non-gcc OSX with 64-bit integers. Add a couple of missing + closing brackets / parenthesis. ASTERISK-24814 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4436/ ........ Merged + revisions 432054 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-20 17:51 +0000 [r432034] Richard Mudgett <rmudgett@digium.com> + + * /, channels/sig_analog.c: chan_dahdi/sig_analog: Put log message + strings on one line. With the log messages on one line, you can + search for the log message seen in the log and expect to find it. + ........ Merged revisions 432032 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-20 17:46 +0000 [r432033] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_publish_asterisk.c, res/res_pjsip_acl.c: + ASTERISK-24811: Add ast_sorcery_apply_config() to + res_pjsip_publish_asterisk. Matt Hoskins reported that + res_pjsip_publish_asterisk wouldn't pull config from realtime. + Turns out it was just missing a call ast_sorcery_apply_config(). + res_pjsip_acl was missing it as well, so I added it. The other + pjsip modules looked OK. ASTERISK-24811 #close Reported-by: Matt + Hoskins Tested-by: George Joseph Tested-by: Matt Hoskins patches: + res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins + (license 6688) Review: https://reviewboard.asterisk.org/r/4433/ + +2015-02-20 15:47 +0000 [r432013] Matthew Jordan <mjordan@digium.com> + + * apps/app_voicemail.c, /: apps/app_voicemail: Fix IMAP header + compatibility issue with Microsoft Exchange When interfacing with + Microsoft Exchange, custom headers will be returned as all lower + case. Currently, the IMAP header code will fail to parse the + returned custom headers, as it will be performing a case + sensitive comparison. This can cause playback of messages to + fail, as needed information - such as origtime - will not be + present. This patch updates app_voicemail's header parsing code + to perform a case insensitive lookup for the requested custom + headers. Since the headers are specific to Asterisk, e.g., + 'x-asterisk-vm-orig-time', and headers should be unique in an + IMAP message, this should cause no issues with other systems. + ASTERISK-24787 #close Reported by: Graham Barnett patches: + app_voicemail.c.patch_MSExchange uploaded by Graham Barnett + (License 6685) ........ Merged revisions 432012 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-19 21:25 +0000 [r431956-431993] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, channels/sig_analog.c, /: chan_dahdi: + Remove some dead code. ........ Merged revisions 431992 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * main/aoc.c: ISDN AOC: Fix crash from an AOC-E message that + doesn't have a channel association. Processing an AOC-E event + that does not or no longer has a channel association causes a + crash. The problem with posting AOC events to the channel topic + is that AOC-E events don't always have a channel association and + posting the event to the all channels topic is just wrong. AOC-E + events do however have their own charging association method to + refer to the agreement with the charging entity. * Changed the + AOC events to post to the AMI manager topic instead of the + channel topics. If a channel is associated with the event then + channel snapshot information is supplied with the AMI event. * + Eliminated RAII_VAR() usage in aoc_to_ami() and + ast_aoc_manager_event(). This patch supercedes the patch on + Review: https://reviewboard.asterisk.org/r/4427/ ASTERISK-22670 + #close Reported by: klaus3000 ASTERISK-24689 #close Reported by: + Marcel Manz ASTERISK-24740 #close Reported by: Panos Gkikakis + Review: https://reviewboard.asterisk.org/r/4430/ + + * res/res_pjsip_refer.c: res_pjsip_refer: Handle INVITE with + Replaces failure after answer. * Fixed hangup handling of the + session->channel after answer if the ast_channel_move() or + ast_bridge_impart() fails. We are still the thread controlling + the session->channel so we need to call ast_hangup() to kill the + channel. * Fixed debug messages in + refer_incoming_invite_request() referencing incorrect channnels + on success. Code comments now say why the session->channel cannot + be used. Review: https://reviewboard.asterisk.org/r/4422/ + +2015-02-19 15:28 +0000 [r431937] Matthew Jordan <mjordan@digium.com> + + * main/tcptls.c, /: tcptls: Handle new OpenSSL compile time option + to disable SSLv3 Some distributions are going to disable SSLv3 at + compile time. This option can be checked using the directive + OPENSSL_NO_SSL3_METHOD. This patch updates the TCP/TLS handling + in Asterisk to look for that directive before attempting to use + the SSLv3 specific methods. ASTERISK-24799 #close Reported by: + Alexander Traud patches: no-ssl3-method.patch uploaded by + Alexander Traud (License 6520) ........ Merged revisions 431936 + from http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-19 02:01 +0000 [r431917] Corey Farrell <git@cfware.com> + + * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c: + Create work around for scheduler leaks during shutdown. * Added + ast_sched_clean_by_callback for cleanup of scheduled events that + have not yet fired. * Run all pending peercnt_remove_cb and + replace_callno events in chan_iax2. Cleanup of replace_callno + events is only run 11, since it no longer releases any references + or allocations in 13+. ASTERISK-24451 #close Reported by: Corey + Farrell Review: https://reviewboard.asterisk.org/r/4425/ ........ + Merged revisions 431916 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-17 15:31 +0000 [r431898] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip_messaging.c, + res/res_pjsip_caller_id.c, res/res_pjsip_refer.c, + res/res_pjsip_send_to_voicemail.c: res_pjsip_refer: Fix crash + from a REFER and BYE collision. Analyzing a one-off crash on a + busy system showed that processing a REFER request had a NULL + session channel pointer. The only way I can think of that could + cause this is if an outgoing BYE transaction overlapped the + incoming REFER transaction in a collision. Asterisk sends a BYE + while the phone sends a REFER to complete an attended transfer. * + Made check the session channel pointer before processing an + incoming REFER request in res_pjsip_refer. * Fixed similar crash + potential for res_pjsip supplement incoming request processing + for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, + res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail + REFER messages. * Made res_pjsip_messaging respond to a message + body too large with a 413 instead of ignoring it. ASTERISK-24700 + #close Reported by: Zane Conkle Review: + https://reviewboard.asterisk.org/r/4417/ + +2015-02-16 21:29 +0000 [r431879] Matthew Jordan <mjordan@digium.com> + + * res/res_rtp_asterisk.c: res/res_rtp_asterisk: Fix crash in debug + from RTCP reports without report block When RTCP debugging was + enabled, an RTCP report without a report block would cause a + crash. This was due to the verbose output not checking to see if + the report_block pointer was NULl before dereferencing it. This + patch adds the necessary check to prevent printing any verbose + output if the far side hasn't provided us the information they + should have. ASTERISK-24791 #close Reported by: JoshE Tested by: + JoshE + +2015-02-15 19:00 +0000 [r431807-431860] Joshua Colp <jcolp@digium.com> + + * configs/samples/pjsip.conf.sample: pjsip: Remove "contact" type + from pjsip.conf.sample The "contact" object is not meant to be + configured from the pjsip.conf configuration file. It is meant to + be created as a result of a registration and stored elsewhere. + ASTERISK-24085 #close Reported by: Rusty Newton + + * contrib/scripts/install_prereq: install_prereq: Tweak flags when + configuring pjproject. This change does two things: 1. Disables + debugging so assertions which can return an error do, instead of + asserting. 2. Enables IPv6 support. ASTERISK-24632 #close + Reported by: Rusty Newton + + * res/res_sorcery_config.c: res_sorcery_config: Improve object + lookup times. The res_sorcery_config module currently uses a + fixed bucket size of 53. This means that depending on the number + of objects you either end up with excess buckets or a lot of + collisions. Due to the way that res_sorcery_config is implemented + it's actually possible to make the bucket size dynamic based on + the number of objects. This is due to the fact that each loading + of the config file produces a new container and does not modify + the existing one. This change uses the number of expected objects + and finds a prime number near it. In practice depending on the + number of objects this can speed up lookups anywhere from 2X to + 15X. This change also removes the lock from the container as it + is not needed. Review: https://reviewboard.asterisk.org/r/4423/ + + * res/res_pjsip/pjsip_cli.c: res_pjsip: Add "pjsip show version" + CLI command. When debugging things it can be useful to know + absolutely what version of pjproject res_pjsip is running + against. This change adds a "pjsip show version" CLI command + which can be used to query for this. ASTERISK-24685 #close + Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/4424/ + + * res/res_timing_pthread.c: res_timing_pthread: Fix leaky pipes. + During some refactoring the way private information for timers + was stored was changed. As a result of this the action which + normally removed the timer upon closure in res_timing_pthread was + also removed causing the timer to remain after it should using up + resources. This change ensures that the timer is removed upon + closure. ASTERISK-24768 #close Reported by: Matthias Urlichs + patches: timer.patch submitted by Matthias Urlichs (license 5508) + +2015-02-15 00:32 +0000 [r431789] Matthew Jordan <mjordan@digium.com> + + * /, apps/app_mixmonitor.c: apps/app_mixmonitor: Move Test Event + for MIXMONITOR_END to after it finishes The Test Event for + MIXMONITOR_END - which signals that a MixMonitor has completed - + technically fired before the filestream was closed. If a test + used this to trigger a condition to verify that the file was + written, it could result in a race condition where the file size + would not be what the test expected. Luckily, no tests were using + this (although they should have been). Since the test event + needed to be moved after the point where the MixMonitor autochan + has been destroyed, the test event no longer emits the channel + name. Luckily, nothing needs it. ........ Merged revisions 431788 + from http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-14 19:45 +0000 [r431751-431771] Joshua Colp <jcolp@digium.com> + + * main/sorcery.c: sorcery: Output an error message if a wizard is + specified for an object type and it isn't found. ASTERISK-24612 + #close Reported by: Joshua Colp + + * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Improve log + message when a subscription is attempted to a non-existent + extension. ASTERISK-24716 #close Reported by: Rusty Newton + + * channels/pjsip/dialplan_functions.c: 'information' ends with an + 'n'. + + * channels/pjsip/dialplan_functions.c: chan_pjsip: Fix crash when + CHANNEL dialplan function is invoked with pjsip argument and no + type. ASTERISK-24771 #close Reported by: Niklas Larsson + +2015-02-13 17:21 +0000 [r431734] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_session.c: res_pjsip_session: Fix double re-INVITE + collision crash. A multi-asterisk box setup with direct media + enabled would occasionally crash when two re-INVITE collisions on + a call leg happen in a row. The re-INVITE logic only had one + timer struct to defer the re-INVITE. When the second collision + happens the timer struct is overwritten and put into the timer + heap again. Resources for the first timer are leaked and the heap + has two positions occupied by the same timer struct. Now the heap + ordering is potentially corrupted, the timer will fire twice, and + any resources allocated for the second timer will be released + twice. * The solution is to put the collided re-INVITE into the + delayed requests queue with all the other delayed requests and + cherry pick the next request that can come off the queue when an + event happens. * Changed to put delayed BYE requests at the head + of the delayed queue. There is no sense in processing delayed + UPDATEs and re-INVITEs when a BYE has been requested. * Made the + start of a BYE request flush the delayed requests queue to + prevent a delayed request from overlapping the BYE transaction. I + saw a few cases where a delayed re-INVITE got started after the + BYE transaction started. * Changed the delayed_request struct to + use an enum instead of a string for the request method. Cherry + picking the queue is easier with an enum than string comparisons + and the compiler can warn if a switch statement does not cover + all defined enum values. * Improved the debug output to give more + information. It helps to know which channel is involved with an + endpoint. Trunks can have many channels associated with the + endpoint at the same time. ASTERISK-24727 #close Reported by: + Mark Michelson Review: https://reviewboard.asterisk.org/r/4414/ + +2015-02-12 20:32 +0000 [r431717] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_multihomed.c, res/stasis/control.c, + include/asterisk/stasis_app.h, rest-api/api-docs/channels.json, + res/ari/resource_channels.c, CHANGES, res/res_ari_channels.c, + channels/chan_pjsip.c, res/res_pjsip_nat.c, + res/res_pjsip_transport_websocket.c, res/ari/resource_channels.h: + ARI/PJSIP: Add the ability to redirect (transfer) a channel in a + Stasis app This patch adds a new feature to ARI to redirect a + channel to another server, and fixes a few bugs in PJSIP's + handling of the Transfer dialplan application/ARI redirect + capability. *New Feature* A new operation has been added to the + ARI channels resource, redirect. With this, a channel in a Stasis + application can be redirected to another endpoint of the same + underlying channel technology. *Bug fixes* In the process of + writing this new feature, two bugs were fixed in the PJSIP stack: + (1) The existing .transfer channel callback had the limitation + that it could only transfer channels to a SIP URI, i.e., you had + to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan + application. While this is still supported, it is somewhat + unintuitive - particularly in a world full of endpoints. As such, + we now also support specifying the PJSIP endpoint to transfer to. + (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a + 302 redirect by updating its Contact header. Alas, that resulted + in the forwarding destination set by the dialplan application/ARI + resource/whatever being rewritten with very incorrect + information. Hence, we now don't bother updating an outgoing + response if it is a 302. Since this took a looong time to find, + some additional debug statements have been added to those modules + that update the Contact headers. Review: + https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close + Reported by: Private Name ASTERISK-24703 #close Reported by: Matt + Jordan + +2015-02-11 18:02 +0000 [r431693-431698] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip/pjsip_configuration.c: res_pjsip: dtls_handler + causes Asterisk to crash There have been a couple of times where + a crash occurred in the dtls_handler section of the code for + res_pjsip. Unfortunately, in working this issue the problem was + unable to be reproduced. After looking at the backtraces and + through the code the current best guess as to why this happened + might be due to a reentrance problem and the strtok function. So, + the current fix is to convert the strtok function into the + reentrant version of the function, strtok_r. ASTERISK-24741 + #close Reported by: Zane Conkle Review: + https://reviewboard.asterisk.org/r/4409/ + + * res/ari/ari_websockets.c: ari_websockets: removed extra check on + websocket session read When merging the websocket timeout issue + (ASTERISK-24701) an extra, almost duplicate, check was left in + the code that should not have been. This removes it. + ASTERISK-24701 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4412/ + +2015-02-11 17:28 +0000 [r431692] Richard Mudgett <rmudgett@digium.com> + + * main/bridge.c, main/http.c, apps/app_confbridge.c, + include/asterisk/channel.h, res/res_pjsip/pjsip_options.c, + res/res_pjsip_pubsub.c, main/asterisk.c, main/channel.c, + include/asterisk.h, channels/chan_sip.c: HTTP: Stop accepting + requests on final system shutdown. There are three CLI commands + to stop and restart Asterisk each. 1) core stop/restart now - + Hangup all calls and stop or restart Asterisk. New channels are + prevented while the shutdown request is pending. 2) core + stop/restart gracefully - Stop or restart Asterisk when there are + no calls remaining in the system. New channels are prevented + while the shutdown request is pending. 3) core stop/restart when + convenient - Stop or restart Asterisk when there are no calls in + the system. New calls are not prevented while the shutdown + request is pending. ARI has made stopping/restarting Asterisk + more problematic. While a shutdown request is pending it is + desirable to continue to process ARI HTTP requests for current + calls. To handle the current calls while a shutdown request is + pending, a new committed to shutdown phase is needed so ARI + applications can deal with the calls until the system is fully + committed to shutdown. * Added a new shutdown committed phase so + ARI applications can deal with calls until the final committed to + shutdown phase is reached. * Made refuse new HTTP requests when + the system has reached the final system shutdown phase. Starting + anything while the system is actively releasing resources and + unloading modules is not a good thing. * Split the bridging + framework shutdown to not cleanup the global bridging containers + when shutting down in a hurry. This is similar to how other + modules prevent crashes on rapid system shutdown. * Moved + ast_begin_shutdown(), ast_cancel_shutdown(), and + ast_shutting_down(). You should not have to include channel.h + just to access these system functions. ASTERISK-24752 #close + Reported by: Matthew Jordan Review: + https://reviewboard.asterisk.org/r/4399/ + +2015-02-11 17:12 +0000 [r431674] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: channels/chan_sip: Fix RealTime error + during SIP unregistration with MariaDB When a SIP device that has + its registration stored in RealTime unregisters, the entry for + that device is updated with blank values, i.e., "", indicating + that it is no longer registered. Unfortunately, one of those + values that is 'blanked' is the device's port. If the column type + for the port is not a string datatype (the recommended type is + integer), an ODBC or database error will be thrown. MariaDB does + not coerce empty strings to a valid integer value. This patch + updates the query run from chan_sip such that it replaces the + port value with a value of '0', as opposed to a blank value. This + is the value that other database backends coerce the empty string + ("") to already, and the handling of reading a RealTime + registration value from a backend already anticipates receiving a + port of '0' from the backends. ASTERISK-24772 #close Reported by: + Richard Miller patches: chan_sip.diff uploaded by Richard Miller + (License 5685) ........ Merged revisions 431673 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-11 16:51 +0000 [r431670] Kevin Harwell <kharwell@digium.com> + + * res/res_http_websocket.c, res/ari/ari_websockets.c, /: + res_http_websocket: websocket write timeout fails to fully + disconnect When writing to a websocket if a timeout occurred the + underlying socket did not get closed/disconnected. This patch + makes sure the websocket gets disconnected on a write timeout. + Also a notice is logged stating that the websocket was + disconnected. ASTERISK-24701 #close Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged + revisions 431669 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-11 15:51 +0000 [r431663] Corey Farrell <git@cfware.com> + + * include/asterisk/module.h, main/loader.c, /, + bridges/bridge_builtin_features.c: Enable REF_DEBUG for + ast_module_ref / ast_module_unref. Add ast_module_shutdown_ref + for use by modules that can only be unloaded during graceful + shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to + struct ast_module. * Allocate ao2 object when the module is + loaded. * Perform an ao2_ref in each place where mod->usecount is + manipulated. * ao2_cleanup on module unload. ASTERISK-24479 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4141/ ........ Merged + revisions 431662 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-02-10 23:16 +0000 [r431643] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_config_wizard.c, + configs/samples/pjsip_wizard.conf.sample: + res_pjsip_config_wizard: Add ability to auto-create hints. + Looking at the Super Awesome Company sample reminded me that + creating hints is just plain gruntwork. So you can now have the + pjsip conifg wizard auto-create them for you. Specifying + 'hint_exten' in the wizard will create 'exten => + <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for + 'hint_context'. Specifying 'hint_application' in the wizard will + create 'exten => <hint_exten>,1,<hint_application>' in whatever + is specified for 'hint_context'. The default for 'hint_context' + is the endpoint's context. There's no default for + 'hint_application'. If not specified, no app is added. There's no + default for 'hint_exten'. If not specified, neither the hint + itself nor the application will be created. Some may think this + is the slippery slope to users.conf but hints are a basic + necessity for phones unlike voicemail, manager, etc that + users.conf creates. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4383/ + +2015-02-09 03:10 +0000 [r431600-431622] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c: + res/ari/resource_channels: Add missing 'no_answer' reason to + DELETE /channels One of the canonical reasons for hanging up a + channel is because the far end failed to answer - or because + someone else answered, and we want to get rid of this channel. + This patch adds the missing value to the 'reason' query parameter + for the DELETE /channels operation. Review: + https://reviewboard.asterisk.org/r/4400 ASTERISK-24745 #close + Reported by: Ben Merrills patches: + add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills + (License 6678) + + * /, res/res_odbc.c: res/res_odbc: Remove unneeded queries when + determining if a table exists This patch modifies the + ast_odbc_find_table function such that it only performs a lookup + of the requested table if the table is not already known. Prior + to this patch, a queries would be executed against the database + even if the table was already known and cached. Review: + https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close + Reported by: ibercom patches: patch.diff uploaded by ibercom + (License 6599) ........ Merged revisions 431617 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Fix leak of local + ICE candidates when applying to SDP When an SDP is created for an + outgoing request/response, the ICE candidates obtained from the + RTP instance are currently leaked. This causes the ao2 container + that holds the candidates to never properly be reclaimed when the + RTP instance is destroyed. This patch properly decrements the ICE + candidates' container if it is successfully obtained. + ASTERISK-24769 #close Reported by: Matt Jordan + +2015-02-06 21:26 +0000 [r431583] Scott Griepentrog <sgriepentrog@digium.com> + + * main/utils.c, res/res_pjsip.c, main/config.c: various: cleanup + issues found during leak hunt In this collection of small patches + to prevent Valgrind errors are: fixes for reference leaks in + config hooks, evaluating a parameter beyond bounds, and accessing + a structure after a lock where it could have been already free'd. + Review: https://reviewboard.asterisk.org/r/4407/ + +2015-02-04 01:27 +0000 [r431538-431555] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_keepalive.c: res_pjsip_keepalive: Don't crash if + PJSIP module is not loaded. + + * main/sorcery.c: sorcery: Don't try to load object types which + haven't been defined. The act of defining wizards for an object + type in sorcery.conf will create a minimal object type. This can + cause a problem when a module has multiple sorcery instances + (which all get the wizards from sorcery.conf applied) but the + sorcery instances do not all contain full information about the + object types. Upon loading errors will occur stating that the + objects can not be created. This is confusing and is actually + perfectly fine. This change makes it so that only object types + which have been fully defined will be loaded. ASTERISK-24748 + #close Reported by: Joshua Colp + +2015-01-31 16:27 +0000 [r431521] Joshua Colp <jcolp@digium.com> + + * res/res_format_attr_h264.c: res_format_attr_h264: Fix crash when + determining joint capability. The res_format_attr_h264 module + currently incorrectly attempts to copy SPS and PPS information + from the wrong attribute. This change fixes that. ASTERISK-24616 + #close Reported by: Yura Kocyuba Review: + https://reviewboard.asterisk.org/r/4392/ + +2015-02-06 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.2.0 Released. + +2015-01-30 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.2.0-rc1 Released. + +2015-01-30 17:44 +0000 [r431492] Richard Mudgett <rmudgett@digium.com> + + * apps/app_agent_pool.c: app_agent_pool: Fix initial module load + agent device state reporting. When the app_agent_pool module + initially loads there is a race condition between the thread + loading agents.conf and the device state internal processing + thread. If the device state internal processing thread handles + the agent creation state updates before the thread that loaded + agents.conf registers the device state provider callback then the + cached agent state is "Invalid". When a consumer module like + app_queue asks for the agent state it gets the cached "Invalid" + state instead of the real state from the provider. * Moved + loading the agents.conf configuration to the last thing setup by + app_agent_pool in load_module(). Now the device state provider + callback is registered before the config is loaded so the agent + creation state updates are guaranteed to get the initial device + state. * Removed some now redundant config cleanup on error in + load_config(). * Added lock protection when accessing the device + state in agent_pvt_devstate_get() and eliminated the RAII_VAR() + usage. ASTERISK-24737 #close Reported by: Steve Pitts Review: + https://reviewboard.asterisk.org/r/4390/ + +2015-01-30 17:38 +0000 [r431490] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_outbound_publish.c: res_pjsip_outbound_publish: + eventually crashes when no response is ever received When + Asterisk attempts to send SIP outbound publish information and no + response is ever received (no 200 okay, 412, 423) the system + eventually crashes. A response is never received because the + system Asterisk is attempting to send publish information to is + not available. The underlying pjsip framework attempts to send + publish information. After several attempts it calls back into + the Asterisk outbound publish code. At this point if the + "client->queue" is empty Asterisk attempts to schedule a refresh + which utilizes "rdata" and since no response was received the + given "rdata" struture is NULL. Attempting to dereference a NULL + object of course results in a crash. The fix here removes the + dependency on rdata for schedule_publish_refresh. Instead + param->expiration is now passed to it as this is set to -1 if no + response is received. Also added a notification when no response + is received. ASTERISK-24635 #close Reported by: Marco Paland + Review: https://reviewboard.asterisk.org/r/4384/ + +2015-01-30 16:52 +0000 [r431471] asanders <asanders@localhost>: + + * include/asterisk/http.h, configs/samples/http.conf.sample, + main/http.c: HTTP: For httpd server, need option to define server + name for security purposes Added a new config property + [servername] to the http.conf file; updated the http server to + use the new property when sending responses, for showing http + status through the CLI and when reporting status through the + 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew + Nagy Review: https://reviewboard.asterisk.org/r/4374/ + +2015-01-30 16:47 +0000 [r431468] Mark Michelson <mmichelson@digium.com> + + * main/stasis_channels.c, channels/chan_pjsip.c, main/xmldoc.c, + res/res_pjsip_refer.c, main/pbx.c, main/manager.c, + pbx/pbx_spool.c, main/bridge_after.c: Fix some memory leaks. + These memory leaks were found and fixed by John Hardin. I'm just + committing them for him. ASTERISK-24736 #close Reported by Mark + Michelson Review: https://reviewboard.asterisk.org/r/4389 + +2015-01-29 23:02 +0000 [r431450] Scott Griepentrog <sgriepentrog@digium.com> + + * include/asterisk/bridge.h, main/bridge.c, + res/stasis/stasis_bridge.c: stasis transfer: fix stasis bridge + push race part two When swapping a Local channel in place of one + already in a bridge (to complete a bridge attended transfer), the + channel that was swapped out can actually be hung up before the + stasis bridge push callback executes on the independant transfer + thread. This results in the stasis app loop dropping out and + removing the control that has the the app name which the local + replacement channel needs so it can re-enter stasis. To avoid + this race condition a new push_peek callback has been added, and + called from the ast_bridge_impart thread before it launches the + independant thread that will complete the transfer. Now the + stasis push_peek callback can copy the stasis app name before the + swap channel can hang up. ASTERISK-24649 Review: + https://reviewboard.asterisk.org/r/4382/ + +2015-01-29 20:58 +0000 [r431420-431426] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip.c, res/res_pjsip_sips_contact.c (added): Use SIPS + URIs in Contact headers when appropriate. RFC 3261 sections + 8.1.1.8 and 12.1.1 dictate specific scenarios when we are + required to use SIPS URIs in Contact headers. Asterisk's + non-compliance with this could actually cause calls to get + dropped when communicating with clients that are strict about + checking the Contact header. Both of the SIP stacks in Asterisk + suffered from this issue. This changeset corrects the behavior in + res_pjsip/chan_pjsip.c Review: + https://reviewboard.asterisk.org/r/4345 + + * /, channels/chan_sip.c: Use SIPS URIs in Contact headers when + appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate + specific scenarios when we are required to use SIPS URIs in + Contact headers. Asterisk's non-compliance with this could + actually cause calls to get dropped when communicating with + clients that are strict about checking the Contact header. Both + of the SIP stacks in Asterisk suffered from this issue. This + changeset corrects the behavior in chan_sip. ASTERISK-24646 + #close Reported by Stephan Eisvogel Review: + https://reviewboard.asterisk.org/r/4346 ........ Merged revisions + 431423 from http://svn.asterisk.org/svn/asterisk/branches/11 + + * res/res_pjsip/pjsip_configuration.c: Allow disabling of 100rel + support on PJSIP endpoints. Due to an inversion error, setting + 100rel=no would not actually change the current value of the + setting (which defaulted to "yes"). With this fix, the inversion + is corrected. + +2015-01-29 16:46 +0000 [r431403] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Reduce log + clutter... change a WARNING to a VERBOSE/2 Reduce log clutter by + changing the "Watcher for hint %s (removed|deactivated)" message + from WARNING to VERBOSE/2. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4387/ + +2015-01-29 12:09 +0000 [r431385] Joshua Colp <jcolp@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix DTLS when used + with OpenSSL 1.0.1k A recent security fix for OpenSSL broke DTLS + negotiation for many applications. This was caused by read ahead + not being enabled when it should be. While a commit has gone into + OpenSSL to force read ahead on for DTLS it may take some time for + a release to be made and the change to be present in + distributions (if at all). As enabling read ahead is a simple one + line change this commit does that and fixes the issue. + ASTERISK-24711 #close Reported by: Jared Biel ........ Merged + revisions 431384 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-28 17:37 +0000 [r431301-431303] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c, + res/res_pjsip_session.c: Fix file descriptor leak in RTP code. + SIP requests that offered codecs incompatible with configured + values could result in the allocation of RTP and RTCP ports that + would not get reclaimed later. ASTERISK-24666 #close Reported by + Y Ateya Review: https://reviewboard.asterisk.org/r/4323 + AST-2015-001 ........ Merged revisions 431300 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * funcs/func_curl.c, /: Multiple revisions 431297-431298 ........ + r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan + 2015) | 17 lines Mitigate possible HTTP injection attacks using + CURL() function in Asterisk. CVE-2014-8150 disclosed a + vulnerability in libcURL where HTTP request injection can be + performed given properly-crafted URLs. Since Asterisk makes use + of libcURL, and it is possible that users of Asterisk may get + cURL URLs from user input or remote sources, we have made a patch + to Asterisk to prevent such HTTP injection attacks from + originating from Asterisk. ASTERISK-24676 #close Reported by Matt + Jordan Review: https://reviewboard.asterisk.org/r/4364 + AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49 + -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from + previous patch. ........ Merged revisions 431297-431298 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 431299 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2015-01-28 12:18 +0000 [r431267] Sean Bright <sean@malleable.com> + + * res/res_format_attr_silk.c, res/res_format_attr_opus.c: media + formats: update res_format_attr_opus & silk In r419044, we + changed how formats were handled, but the return value of the + format_parse_sdp_fmtp functions in res_format_attr_opus and + res_format_attr_silk were not updated, causing calls to fail. Ran + into this when getting codec_opus working with Asterisk 13. Once + the return value was corrected, we were crashing in opus_getjoint + because of NULL format attributes. I've fixed this as well in + this patch. Review: https://reviewboard.asterisk.org/r/4371/ + +2015-01-28 04:09 +0000 [r431243] Richard Mudgett <rmudgett@digium.com> + + * main/sorcery.c, res/res_pjsip_outbound_registration.c, + res/res_pjsip.c: res_pjsip_outbound_registration: Fix reload race + condition. Performing a CLI "module reload" command when there + are new pjsip.conf registration objects defined frequently failed + to load them correctly. What happens is a race condition between + res_pjsip pushing its reload into an asynchronous task processor + task and the thread that does the rest of the reloads when it + gets to reloading the res_pjsip_outbound_registration module. A + similar race condition happens between a reload and the CLI/AMI + show registrations commands. The reload updates the + current_states container and the CLI/AMI commands call + get_registrations() which builds a new current_states container. + * Made res_pjsip.c reload_module() use + ast_sip_push_task_synchronous() instead of ast_sip_push_task() to + eliminate two threads processing config reloads at the same time. + * Made get_registrations() not replace the global current_states + container so the CLI/AMI show registrations command cannot + interfere with reloading. You could never add/remove objects in + the container without the possibility of the container being + replaced out from under you by get_registrations(). * Added a + registration loaded sorcery instance observer to purge any dead + registration objects since get_registrations() cannot do this job + anymore. The struct ast_sorcery_instance_observer callbacks must + be used because the callback happens inline with the load + process. The struct ast_sorcery_observer callbacks are pushed to + a different thread. * Added some global current_states NULL + pointer checks in case the container disappears because of + unload_module(). * Made sorcery's struct + ast_sorcery_instance_observer.object_type_loaded callbacks + guaranteed to be called before any struct + ast_sorcery_observer.loaded callbacks will be called. * Moved the + check for non-reloadable objects to before the sorcery instance + loading callbacks happen to short circuit unnecessary work. + Previously with non-reloadable objects, the sorcery instance + loading/loaded callbacks would always happen, the individual + wizard loading/loaded would be prevented, and the non-reloadable + type logging message would be logged for each associated wizard. + ASTERISK-24729 #close Review: + https://reviewboard.asterisk.org/r/4381/ + +2015-01-27 22:56 +0000 [r431179-431219] Kevin Harwell <kharwell@digium.com> + + * /, main/tcptls.c: tcptls: Bad file descriptor error when + reloading chan_sip While running through some scenarios using + chan_sip and tcp a problem would occur that resulted in a flood + of bad file descriptor messages on the cli: tcptls.c:712 + ast_tcptls_server_root: Accept failed: Bad file descriptor The + message is received because the underlying socket has been + closed, so is valid. This is probably happening because unloading + of chan_sip is not atomic. That however is outside the scope of + this patch. This patch simply stops the logging of multiple + occurrences of that message. ASTERISK-24728 #close Reported by: + Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/ + ........ Merged revisions 431218 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, channels/chan_sip.c: chan_sip: stale nonce causes failure When + refreshing (with a small expiration) a registration that was sent + to chan_sip the nonce would be considered stale and reject the + registration. What was happening was that the initial + registration's "dialog" still existed in the dialogs container + and upon refresh the dialog match algorithm would choose that as + the "dialog" instead of the newly created one. This occurred + because the algorithm did not check to see if the from tag + matched if authentication info was available after the 401. So, + it ended up assuming the original "dialog" was a match and + stopped the search. The old "dialog" of course had an old nonce, + thus the stale nonce message. This fix attempts to leave the + original functionality alone except in the case of a REGISTER. If + a REGISTER is received if searches for an existing "dialog" + matching only on the callid. If the expires value is low enough + it will reuse dialog that is there, otherwise it will create a + new one. ASTERISK-24715 #close Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/4367/ ........ Merged + revisions 431187 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c, + main/stasis_message_router.c, res/res_pjsip/location.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_distributor.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip/pjsip_global_headers.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + res/res_pjsip/config_transport.c: res_pjsip: make it unloadable + (take 2) Due to the original patch causing memory corruptions it + was removed until the problem could be resolved. This patch is + the original patch plus some added locking around stasis router + subcription that was needed to avoid the memory corruption. + Description of the original problem and patch (still applicable): + The res_pjsip module was previously unloadable. With this patch + it can now be unloaded. This patch is based off the original + patch on the issue (listed below) by Corey Farrell with a few + modifications. Namely, removed a few changes not required to make + the module unloadable and also fixed a bug that would cause + asterisk to crash on unloading. This patch is the first step + (should hopefully be followed by another/others at some point) in + allowing res_pjsip and the modules that depend on it to be + unloadable. At this time, res_pjsip and some of the modules that + depend on res_pjsip cannot be unloaded without causing problems + of some sort. The goal of this patch is to get res_pjsip and only + res_pjsip to be able to unload successfully and/or shutdown + without incident (crashes, leaks, etc...). Other dependent + modules may still cause problems on unload. Basically made sure, + with the patch applied, that res_pjsip (with no other dependent + modules loaded) could be succesfully unloaded and Asterisk could + shutdown without any leaks or crashes that pertained directly to + res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4363/ patches: + pjsip_unload-broken-r1.patch submitted by Corey Farrell (license + 5909) + +2015-01-27 17:36 +0000 [r431160] Richard Mudgett <rmudgett@digium.com> + + * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c: + app_confbridge: Repeatedly starting and stopping recording ref + leaks the recording channel. Starting and stopping conference + recording more than once causes the recording channels to be + leaked. For v13 the channels also show up in the CLI "core show + channels" output. * Reworked and simplified the recording channel + code to use ast_bridge_impart() instead of managing the recording + thread in the ConfBridge code. The recording channel's ref + handling easily falls into place and other off nominal code paths + get handled better as a result. ASTERISK-24719 #close Reported + by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/ + Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged + revisions 431135 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-27 17:32 +0000 [r431157] Joshua Colp <jcolp@digium.com> + + * main/bridge_channel.c, res/res_pjsip_sdp_rtp.c: bridge / + res_pjsip_sdp_rtp: Fix issues with media not being reinvited + during direct media. This change fixes two issues: 1. During a + swap operation bridging added the new channel before having the + swap channel leave. This was not handled in bridge_native_rtp and + could result in a channel not getting reinvited back to Asterisk. + After this change the swap channel will leave first and the new + channel will then join. 2. If a re-invite was received after a + session had been established any upstream elements (such as + bridge_native_rtp) were not notified that they may want to + re-evaluate things. After this change an UPDATE_RTP_PEER control + frame is queued when this situation occurs and upstream can + react. AST-1524 #close Review: + https://reviewboard.asterisk.org/r/4378/ + +2015-01-27 17:22 +0000 [r431153] Jonathan Rose <jrose@digium.com> + + * main/manager.c: Manager: Fix Manager Action ModuleLoad to give + correct response when reloading Prior to this patch, ModuleLoad + would respond with an error indicating that the requested module + wasn't found in spite of finding and reloading the module. + Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721 + #close + +2015-01-27 17:20 +0000 [r431134-431145] Matthew Jordan <mjordan@digium.com> + + * res/ari/resource_bridges.c, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, res/res_ari_bridges.c, + res/ari/resource_bridges.h, rest-api-templates/api.wiki.mustache, + rest-api/api-docs/channels.json, + rest-api-templates/swagger_model.py, + rest-api/api-docs/bridges.json: ARI: Improve wiki documentation + This patch improves the documentation of ARI on the wiki. + Specifically, it addresses the following: * Allowed values and + allowed ranges weren't documented. This was particularly + frustrating, as Asterisk would reject query parameters with + disallowed values - but we didn't tell anyone what the allowed + values were. * The /play/id operation on /channels and /bridges + failed to document all of the added media resource types. * + Documentation for creating a channel into a Stasis application + failed to note when it occurred, and that creating a channel into + Stasis conflicts with creating a channel into the dialplan. * + Some other minor tweaks in the mustache templates, including + italicizing the parameter type, putting the default value on its + own sub-bullet, and some other nicities. Review: + https://reviewboard.asterisk.org/r/4351 + + * apps/confbridge/conf_config_parser.c, + apps/confbridge/include/confbridge.h: app_confbridge: Restore + user's menu name to CLI output of 'confbridge list' When issuing + a 'confbridge list XXXX' CLI command, the resulting output no + longer displays the menu associated with a ConfBridge + participant. The issue was caused by ASTERISK-22760. When that + patch was done, it removed the copying of the menu name + associated with the user from the actual user profile. This patch + fixes the issue by copying the menu name over to the user profile + when the menu hooks are applied to the user. Since that function + now does a little bit more than just apply the hooks, the name of + the function has been changed to cover the copying of the menu + name over as well. In addition, there is a disparity between the + menu name length as it is stored on the conf_menu structure and + the confbridge_user structure; this patch makes the lengths match + so that a strcpy can be used. Review: + https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close + Reported by: Steve Pitts + +2015-01-27 11:47 +0000 [r431114] Joshua Colp <jcolp@digium.com> + + * res/parking/parking_manager.c: res_parking: Fix crash due to race + condition when unloading. There is currently a race condition + when unloading the res_parking module. Depending on the will of + the universe the subscription invocation may occur AFTER the + module is unloaded. This is because the module does NOT use + stasis_unsubscribe_and_join when terminating the subscription. It + merely uses stasis_unsubscribe. This change makes it use + stasis_unsubscribe_and_join which is documented for usage in this + exact scenario. AST-1520 #close Review: + https://reviewboard.asterisk.org/r/4375/ + +2015-01-26 14:49 +0000 [r431092] David M. Lee <dlee@digium.com> + + * channels/sip/include/route.h, funcs/func_presencestate.c, + main/rtp_engine.c, configure, include/asterisk/autoconfig.h.in, + include/asterisk/sem.h, configure.ac, main/app.c, + main/bridge_channel.c, main/sem.c, res/res_timing_kqueue.c, + main/asterisk.c: Various fixes for OS X This patch addresses + compilation errors on OS X. It's been a while, so there's quite a + few things. * Fixed __attribute__ decls in route.h to be + portable. * Fixed htonll and ntohll to work when they are defined + as macros. * Replaced sem_t usage with our ast_sem wrapper. * + Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC + 4.9 warnings using sig*set() functions. * Fixed some format + strings for portability. * Fixed compilation issues with + res_timing_kqueue (although tests still fail on OS X). * Fixed + menuconfig /sbin/launchd detection, which disables + res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: + George Joseph ASTERISK-24544 #close Reported by: George Joseph + Review: https://reviewboard.asterisk.org/r/4327/ + +2015-01-25 13:42 +0000 [r431072] Matthew Jordan <mjordan@digium.com> + + * main/config.c: dynamic realtime: Updates fail to work due to + update fields being passed over When a crash was fixed due to + usage of the REALTIME function in r423003, a regression was + introduced into ast_update2_realtime where the update fields + passed to the function would be skipped and the lookup field + processed twice. The use of this function is a bit interesting: A + variable argument list is used with two sentinel values - the + first marks the end of the lookup fields/values; the second marks + the end of the update fields/values. Unfortunately, + ast_update2_realtime parses over the lookup fields twice, as + opposed to parsing over the update fields. This causes the + lookups to succeed, but the updates itself to have no effect. + Note that the most common instance of this problem occurred in + app_voicemail during the updating of a mailbox password. Thanks + to the issue reporter, Paddy Grice, for pointing out the problem. + Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231 + ASTERISK-24626 #close Reported by: Paddy Grice + +2015-01-23 20:13 +0000 [r431050-431052] Richard Mudgett <rmudgett@digium.com> + + * apps/confbridge/conf_chan_record.c: app_confbridge: Make CBRec + channel names more unique. Channel names should be different from + other channels in the system while the channel exists. * Use a + sequence number for CBRec channels instead of a random number + because the same random number could be picked again for the next + CBRec channel. + + * /, apps/app_confbridge.c: app_confbridge: Whitespace Because + there is sometimes no sence to any whitespace. ........ Merged + revisions 431049 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-23 17:08 +0000 [r431030] David M. Lee <dlee@digium.com> + + * res/res_pjsip_config_wizard.c: Add depend on pjproject to + res_pjsip_config_wizard.c + +2015-01-23 15:12 +0000 [r430999] Kevin Harwell <kharwell@digium.com> + + * res/parking/parking_applications.c, channels/chan_iax2.c, + res/res_pjsip/pjsip_global_headers.c, res/res_pjsip_pubsub.c, + res/res_ari_channels.c, res/res_stasis.c, + rest-api-templates/param_parsing.mustache, + res/res_ari_endpoints.c, res/res_ari_events.c, + include/asterisk/stasis_app.h, res/res_pjsip_mwi.c: Investigate + and fix memory leaks in Asterisk Fixed memory leaks that were + found in Asterisk. ASTERISK-24693 #close Reported by: Kevin + Harwell Review: https://reviewboard.asterisk.org/r/4347/ + +2015-01-23 15:03 +0000 [r430994-430998] Walter Doekes <walter+asterisk@wjd.nu> + + * apps/app_voicemail.c, channels/chan_unistim.c, + funcs/func_hangupcause.c, main/manager_bridges.c, + channels/chan_misdn.c, funcs/func_groupcount.c, /, + addons/ooh323c/src/ooh245.c, channels/chan_sip.c, res/res_fax.c, + res/res_pjsip_outbound_registration.c, apps/app_minivm.c, + apps/app_alarmreceiver.c, include/asterisk/channel.h, + contrib/utils/eagi_proxy.c: Fix typo's (retrieve, specified, + address). ........ Merged revisions 430996 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * /, channels/chan_sip.c: chan_sip: Case insensitive comparison of + "defaultuser" parameter. All the other configuration options are + case insensitive, so this one should be too. ASTERISK-24355 + #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded + by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions + 430993 from http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-22 19:24 +0000 [r430957-430975] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/bridge.h, + include/asterisk/bridge_channel_internal.h, main/bridge.c, + include/asterisk/bridge_internal.h, main/bridge_channel.c: Bridge + core: Pass a ref with the swap channel when joining a bridge. + When code imparts a channel into a bridge to swap with another + channel, a ref needs to be held on the swap channel to ensure + that it cannot dissapear before finding it in the bridge. * The + ast_bridge_join() swap channel parameter now always steals a ref + for the swap channel. This is the only change to the bridge + framework's public API semantics. * + bridge_channel_internal_join() now requires the + bridge_channel->swap channel to pass in a ref. ASTERISK-24649 + Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/4354/ + + * res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration.c: Minor code cleanup. * Add an + allocation failure check and assert in + sip_outbound_registration_response_cb(). * Made + sip_outbound_registration_state_destroy() handle partially + created state objects from + sip_outbound_registration_state_alloc(). Review: + https://reviewboard.asterisk.org/r/4366/ + +2015-01-22 18:09 +0000 [r430939] Scott Griepentrog <sgriepentrog@digium.com> + + * res/stasis/app.c, res/stasis/stasis_bridge.c: stasis transfer: + fix a race condition on stasis bridge push After a bridge + transfer completes where a local replacement channel is used, a + stasis transfer message with the details of the transfer is sent. + This is processed by stasis which then sets the stasis app name + and replaced channel snapshot on the replacement channel. + However, since a separate thread was already started to run + stasis on the new replacement channel, a race was on to see if + the message processing would be completed before the app name was + needed, otherwise the channel would be hung up. This change moves + the calls used to set the stasis app name and the replace + snapshot to the bridge_stasis_push function callback from the + bridge transfer logic, allowing the steps to be completed earlier + and more deterministically, and the race elimianted. NOTE: the + swap channel parameter to bridge_stasis_push (and thus all bridge + push callbacks) must always be present when performing a swap + with another channel. ASTERISK-24649 #close Reported by: John + Bigelow Review: https://reviewboard.asterisk.org/r/4341/ + +2015-01-22 14:23 +0000 [r430921] Matthew Jordan <mjordan@digium.com> + + * /, apps/app_voicemail.c: apps/app_voicemail: Trigger MWI + notification with MixMonitor m() option The MixMonitor m() option + allows a recording to be pushed to a specific voicemail mailbox. + If the message is delivered to the mailbox's INBOX, however, no + MWI notification is currently raised. This patch corrects the + issue by properly calling notify_new_state from the + msg_create_from_file function. This will cause MWI to be + triggered if the message was placed in the mailbox's INBOX. + ASTERISK-24709 #close Reported by: Gareth Palmer patches: + app_voicemail-430919.patch uploaded by Gareth Palmer (License + 5169) ........ Merged revisions 430920 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-21 21:53 +0000 [r430902] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration.c: Move unref to a better place. + Move an unconditional unref of client_state so it doesn't look + like it could be used after the last ref has destroyed it. + +2015-01-21 13:33 +0000 [r430840-430864] Matthew Jordan <mjordan@digium.com> + + * channels/chan_sip.c: channels/chan_sip: Fix registration leak + during reload When the SIP registrations were migrated to using + ao2 in what was then trunk, the explicit destruction of the + registrations on module reload was removed and not replaced with + an ao2 equivalent. Debugging done by Stefan Engström, the issue + reporter, on ASTERISK-24673 confirmed that the reference in the + registry_list container was being leaked. Since the purpose of + cleanup_all_regs is to prep a registration for destruction, this + function now calls an ao2_callback function callback with the + OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the + registrations. This cleans up each registration, and also removes + it from the registration container registry_list. Review: + https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close + Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan + Engström Tested by: Stefan Engström + + * cdr/cdr_manager.c, cel/cel_manager.c: AMI: Add documentation for + the missing Cdr/CEL events. This patch adds AMI event + documentation for the Cdr and CEL AMI events. Note that while + these events do share fields with each other and with other + channel related events, they do not contain all of the fields in + a standard channel snapshot, nor is the description of the fields + identical. As such, the patch opts for documentation for each + field, for each event. Review: + https://reviewboard.asterisk.org/r/4350/ ASTERISK-24671 #close + Reported by: Dan Jenkins + + * apps/app_dial.c: apps/app_dial: Don't publish DialEnd twice on + unexpected GoSub/Macro values The Dial application has some + interesting options with the mid-call Macro (M) and GoSub (U) + options. If the MACRO_RESULT/GOSUB_RESULT returns specific + values, the Dial application will take some action upon the + channels involved in the dial operation (such as hanging up a + particular party, etc.) The Dial application ensures that a + Stasis message is published in the event that + MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial + operation, so that there is a corresponding DialEnd event + published in AMI/ARI for the DialBegin event that preceeded it. A + bug exists where that same DialEnd event will be published on + Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is + not one that the Dial application cares about. This causes two + DialEnd events to be published - one with the + MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is + all sorts of wrong. This patch fixes the bug by ensuring that we + only publish a DialEnd message to Stasis if the Dial + application's mid-call Macro/GoSub returns something that Dial + cares about. Review: https://reviewboard.asterisk.org/r/4336 + ASTERISK-24682 #close Reported by: Matt Jordan + + * main/rtp_engine.c: main/rtp_engine: Format NTP timestamps as + unsigned longs When the RTCP reports are created, the NTP + timestamps are stored as strings, as JSON does not have an + integer type long enough to store the value. However, on 32-bit + systems, a signed long may overflow for some portion of the + timestamp. This patch corrects the overflow by formatting the + timestamps as unsigned longs. + +2015-01-20 16:51 +0000 [r430818] asanders <asanders@localhost>: + + * res/ari/resource_bridges.c: ARI: Fixed crash that occurred when + updating a bridge when the optional query parameter 'name' was + not supplied. Prior to this changeset, posting to the: + /ari/bridges/{bridgeId} endpoint without specifying a value for + the [name] query parameter, would crash Asterisk if the bridge + you are attempting to create (or update) had the same ID as an + existing bridge. The internal mechanism of the POST operation + interpreted a null value for name, thus resulting in an error + condition that crashed Asterisk. ASTERISK-24560 #close Reported + By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/ + +2015-01-20 16:46 +0000 [r430817] Richard Mudgett <rmudgett@digium.com> + + * configs/samples/iax.conf.sample, res/res_fax.c, + funcs/func_channel.c, UPGRADE.txt, res/snmp/agent.c, + channels/chan_iax2.c: CHANNEL(peer), chan_iax2, res_fax, SNMP + agent: Fix deadlock from reaching across a bridge. Calling + ast_channel_bridge_peer() cannot be done while holding any + channel locks. The reported issue hit the deadlock in chan_iax2, + but an audit of the ast_channel_bridge_peer() calls found three + more locations where the same deadlock can occur. * Made + CHANNEL(peer), res_fax, and the SNMP agent not call + ast_channel_bridge_peer() with any channel locked. For + CHANNEL(peer) I had to rework the logic to not hold the channel + lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). + It was done for legacy reasons that no longer apply. * Removed + the iax.conf forcejitterbuffer option. It is now always enabled + when the jitterbuffer option is enabled. If you put a jitter + buffer on a channel it will be on the channel. ASTERISK-24600 + #close Reported by: Jeff Collell Review: + https://reviewboard.asterisk.org/r/4342/ + +2015-01-20 02:39 +0000 [r430796-430799] Matthew Jordan <mjordan@digium.com> + + * contrib/scripts/install_prereq, /: + contrib/scripts/install_prereq: Don't install 32-bit packages on + 64-bit hosts On Debian based systems, the install_prereq tool + uses a search command on Debian that results in selecting both + 64-bit and 32-bit packages. Besides the waste of disk space, this + can actually cause aptitude use 100% of memory on a VM with 1GB + of RAM as it tried to work out all of the 32-bit package + dependencies. This patch filters out the 32-bit packages on a + 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048 + #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan + patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang + (License 5876) ........ Merged revisions 430798 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * apps/app_voicemail.c, /: app_voicemail: Temp message left after + review/hangup with ODBC/IMAP backend When using ODBC or IMAP + storage, temporary files created on the file system must be + disposed of using the DISPOSE macro. The DELETE macro will map to + a deletion function for the backend storage, but does not clean + up any local files created as a result of the operation. When + using voicemail with the operator and review options enabled, + pressing 0 to enter the menu, followed by 1 to save the message, + followed by any other DTMF press to delete the message, will + result in the temporary file lingering on the file system. This + patch properly calls DISPOSE after the DELETE. This causes the + local file to be disposed of. ASTERISK-24288 #close Reported by: + LEI FU patches: voicemail_odbc_review_fix.diff uploaded by LEI FU + (License 6640) ........ Merged revisions 430795 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-19 18:05 +0000 [r430776] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c: Call extension state callbacks at hint creation. When + a hint gets created, any subsequent device or presence state + changes result in extension status events getting sent out to + interested parties. However, at the time of hint creation, no + such event gets sent out, so watchers of extension state are + potentially left in the dark until the first state change after + hint creation. Patch contributed by John Hardin (License #6512) + +2015-01-19 13:18 +0000 [r430755] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_multihomed.c, res/res_pjsip.c: res_pjsip / + res_pjsip_multihomed: Use the correct transport and addressing + information on UAS sessions. The first thing this patch fixes is + UAS dialogs. Previously if a transport was configured on an + endpoint and an inbound session was created there was no + guarantee that requests sent on the dialog would use the correct + transport and address information. This has now been fixed so an + explicitly configured transport is taken into account. The second + thing this patch fixes is res_pjsip_multihomed. The + res_pjsip_multihomed module attempts to determine what transport + a message should go out on and what addressing information should + go into the message itself. In a scenario where multiple + transports exist bound to the same IP address but a different + port the code would incorrectly alter the transport and change + the message to the wrong transport. This change makes the + res_pjsip_multihomed module smarter so it will only change the + transport and address information in the message when it is + possible and makes sense. ASTERISK-24615 #close Reported by: + David Justl Review: https://reviewboard.asterisk.org/r/4331/ + +2015-01-17 00:31 +0000 [r430734] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip/config_transport.c, + res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c, + main/stasis_message_router.c, res/res_pjsip/location.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_distributor.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip/pjsip_global_headers.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c: REVERTING + res_pjsip: make it unloadable Due to the original patch causing + memory corruptions the patch is being removed until the problem + can be resolved. + +2015-01-16 22:13 +0000 [r430709-430716] Mark Michelson <mmichelson@digium.com> + + * CHANGES: Change PJProject version requirement for ca_list_path + transport option in CHANGES file. + + * channels/chan_pjsip.c, res/res_pjsip_session.c: Fix problem where + a hung channel could occur on a failed blind transfer. Different + clients react differently to being told that a blind transfer has + failed. Some will simply send a BYE and be done with it. Others + will attempt to reinvite themselves back onto the call. In the + latter case, we were creating a new channel and then leaving it + to sit forever doing nothing. With this code change, that new + channel will not be created and the dialog with the transferring + channel will be cleaned up properly. ASTERISK-24624 #close + Reported by Zane Conkle Review: + https://reviewboard.asterisk.org/r/4339 + + * include/asterisk/res_pjsip.h, configure, + include/asterisk/autoconfig.h.in, configure.ac, + configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c, + res/res_pjsip/config_transport.c: Add support for the + ca_list_path option for PJSIP transports. This allows for a path + to be specified that has a collection of CA certificates in it. + ASTERISK-24575 #close Reported by cloos Patches: + pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: + https://reviewboard.asterisk.org/r/4344 + +2015-01-15 17:35 +0000 [r430685-430687] Richard Mudgett <rmudgett@digium.com> + + * res/res_fax_spandsp.c, res/res_fax.c: res_fax.c, + res_fax_spandsp.c: Remove redundant locking. When FAX was + developed, apparently the faxregistry.container used to be a + linked list that was converted to an ao2 container. Some of the + replacement ao2 container operations still had explicit + lock/unlocks around them. Three off nominal code paths in + res_fax.c and res_fax_spandsp.c unlock the channel even though + the routine did not lock the channel and other code paths in the + routine do not unlock the channel. Review: + https://reviewboard.asterisk.org/r/4340/ + + * res/res_fax_spandsp.c, res/res_fax.c: res_fax.c, + res_fax_spandsp.c: Fix some curlies on the end of function + definitions. + +2015-01-15 12:09 +0000 [r430664] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Fix race condition when + reloading and listing registrations. Due to the split of outbound + registration state from configuration it is possible during a + reload for a "pjsip show registrations" CLI command to be + executed which gets an older snapshot of the configuration. This + configuration may include outbound registrations which have been + removed due to a reload operation occurring at the same time. The + code for printing the outbound registration did not take this + into account but now it does. AST-1506 #close Review: + https://reviewboard.asterisk.org/r/4338/ + +2015-01-15 02:18 +0000 [r430646] Matthew Jordan <mjordan@digium.com> + + * configure, configure.ac: configure: If cross-compiling, assume we + have working semaphores The Asterisk 13 configure.ac checks for + HAS_WORKING_SEMAPHORE but does not have an option for + cross-compiling so it fails with an exit. Since we're cross- + compiling, we can't exactly go looking for the header. The + semaphore.h header is relatively common: * It's part of the POSIX + standard * It's part of GNU C Library As such, we assume that it + will be present when cross-compiling. As such, this patch + defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling is + detected. If you're cross-compiling to a platform that doesn't + support this, then make sure you re-define this to 0. + ASTERISK-24663 #close Reported by: abelbeck patches: + asterisk-13-anonymous-semaphores.patch uploaded by abelbeck + (License 5903) + +2015-01-14 23:14 +0000 [r430628] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_distributor.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip/pjsip_global_headers.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + res/res_pjsip/config_transport.c, + res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c, + main/stasis_message_router.c, res/res_pjsip/location.c: + res_pjsip: make it unloadable The res_pjsip module was previously + unloadable. With this patch it can now be unloaded. This patch is + based off the original patch on the issue (listed below) by Corey + Farrell with a few modifications. Namely, removed a few changes + not required to make the module unloadable and also fixed a bug + that would cause asterisk to crash on unloading. This patch is + the first step (should hopefully be followed by another/others at + some point) in allowing res_pjsip and the modules that depend on + it to be unloadable. At this time, res_pjsip and some of the + modules that depend on res_pjsip cannot be unloaded without + causing problems of some sort. The goal of this patch is to get + res_pjsip and only res_pjsip to be able to unload successfully + and/or shutdown without incident (crashes, leaks, etc...). Other + dependent modules may still cause problems on unload. Basically + made sure, with the patch applied, that res_pjsip (with no other + dependent modules loaded) could be succesfully unloaded and + Asterisk could shutdown without any leaks or crashes that + pertained directly to res_pjsip. ASTERISK-24485 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4311/ patches: + pjsip_unload-broken-r1.patch submitted by Corey Farrell (license + 5909) + +2015-01-14 20:27 +0000 [r430608] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_outbound_publish.c: Prevent slow graceful shutdown + when outbound publications never started. The code was missing + the case for explicitly destroying an outbound publication when + Asterisk had never actually published anything. The result was + that Asterisk would hang for a while on a graceful shutdown. With + this change, the case is taken into account, and on a graceful + shutdown, these publications are destroyed without the need to + actually send a PUBLISH request. ASTERISK-24655 #close Reported + by Kevin Harwell Review: https://reviewboard.asterisk.org/r/4325 + +2015-01-14 15:39 +0000 [r430590] Matthew Jordan <mjordan@digium.com> + + * build_tools/mkpkgconfig, /: build_tools/mkpkgconfig: Fix Cflags + concatenation error in asterisk.pc The mkpkgconfig script + incorrectly concatenates Cflags options together. As an example, + the following: Cflags: -I/usr/include/libxml2 -g3 Is instead + generated as: Cflags: -I/usr/include/libxml2-g3 This patch + corrects the generation of Cflags in mkpkgconfig such that the + Cflags options are output correctly. Review: + https://reviewboard.asterisk.org/r/3707/ ASTERISK-23991 #close + Reported by: Diederik de Groot patches: fix_mkpkgconfig.diff + uploaded by Diederik de Groot (License 6600) ........ Merged + revisions 430589 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-13 18:16 +0000 [r430565] Richard Mudgett <rmudgett@digium.com> + + * apps/app_macro.c, /: app_macro: Don't restore the calling + location on a channel redirect. v11: If a channel redirect to a + macro exten of a macro that is active happens, the redirect + location doesn't get executed. Instead the original macro + location is restored and gets reexecuted. v13: An additional + effect happens if a parked call times out to an extension in the + macro that parked the call then the macro is reexecuted instead + of the expected park return location. * Made not restore the + macro calling location on an AST_SOFTHANGUP_ASYNCGOTO. * + Increased the locked channel range when setting up the macro + execution environment to cover things that should be done while + the channel is locked. * Removed unnecessary NULL tests before + calling ast_free() in _macro_exec(). ASTERISK-23850 #close + Reported by: Andrew Nagy Review: + https://reviewboard.asterisk.org/r/4292/ ........ Merged + revisions 430564 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-13 12:06 +0000 [r430546] Joshua Colp <jcolp@digium.com> + + * channels/pjsip/dialplan_functions.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: chan_pjsip: Add + configure check for 'pjsip_get_dest_info' function. The + 'pjsip_get_dest_info' function is used to determine if the + signaling transport of the dialog is secure or not. This function + was added in PJSIP 2.3 and does not exist in earlier versions. + This configure check allows Asterisk to build and run with older + versions at the loss of the 'secure' argument for the PJSIP + CHANNEL dialplan function. Usage of this argument will require + upgrading to PJSIP 2.3. ASTERISK-24665 #close Reported by: Mark + Michelson Review: https://reviewboard.asterisk.org/r/4329/ + +2015-01-12 18:34 +0000 [r430528] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/manager.h, main/manager.c: AMI: Revert + non-backwards compatible changes from earlier commit. * Reverted + the change to astman_send_listack() to not use the listflag + parameter and always set the value to "Start" so the start + capitalization is consistent. Unfortunately changing the case of + a returned value is not a backward compatible change so for now + FAXSessions is going to have to remain inconsistent with all of + the other AMI list actions. * Reverted the minor protocol error + fix in action_getconfig() when no requested categories are found. + Each line needs to be formatted as "Header: text". Caught by the + testsuite. ASTERISK-24049 + +2015-01-12 18:28 +0000 [r430488-430526] Matthew Jordan <mjordan@digium.com> + + * configs/samples/features.conf.sample: + configs/samples/features.conf.sample: Document attended transfer + DTMF options The sample config was missing the configuration + options for DTMF attended transfer completion scenarios. The + configuration options 'atxferabort', 'atxfercomplete', + 'atxferthreeway', and 'atxferswap' are now documented in the + appropriate configuration file. ASTERISK-24678 #close Reported + by: Niklas Larsson patches: features.conf.sample.diff uploaded by + Niklas Larsson (License 5068) + + * main/syslog.c, include/asterisk/syslog.h, /: main/syslog: Allow + dynamic logs, such as security events, to log to the syslog The + security event log uses a dynamic log level (SECURITY) that is + registered with the Asterisk logging core. Unfortunately, the + syslog would ignore log statements that had a dynamic log level + associated with them. Because the syslog cannot handle ad hoc + dynamic log levels, this patch treats any dynamic log entries + sent to the syslog as logs with a level of NOTICE. ASTERISK-20744 + #close Reported by: Michael Keuter Tested by: Michael L. Young, + Jacek Konieczny patches: + asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by + Michael L. Young (license 5026) ........ Merged revisions 430506 + from http://svn.asterisk.org/svn/asterisk/branches/11 + + * funcs/func_curl.c, /: funcs/func_curl: Fix memory leak when + CURLOPT channel datastore is destroyed When the channel datastore + associated with the usage of CURLOPT on a specific channel is + freed, the underlying structure holding the list of options is + not disposed of. This patch properly frees the structure in the + datastore .destroy callback. ASTERISK-24672 #close Reported by: + Kristian Hogh patches: func_curl-memory-leak.diff uploaded by + Kristian Hogh (License 6639) ........ Merged revisions 430487 + from http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-09 22:08 +0000 [r430467-430469] Scott Griepentrog <sgriepentrog@digium.com> + + * contrib/scripts/sip_to_pjsip/sip_to_pjsip.py, + contrib/scripts/sip_to_pjsip/astconfigparser.py: sip_to_pjsip: + improve ability to parse input files General improvements to SIP + to PJSIP conversion utility: 1) track default section of input + file to allow parsing an include file that doesn't specify a + [section] 2) informatively handle case of assignment without + [section] 3) correctly handle getting sections from included + files - [section]'s are inherited by included file 4) provide + null string as default transport bind ip 5) gracefully handle + missing portions of registration string 6) denote steps of + operation during conversion and confirm top level files as a + convenience ASTERISK-24474 #close Review: + https://reviewboard.asterisk.org/r/4280/ Reported by: John + Kiniston + + * main/features.c: app_bridge: return to the next dialplan priority + When app_bridge grabs a channel and puts it into a bridge, the + channel should then continue where it left off in the dialplan + after the bridge has ended. Although it stores the current + dialplan location as an after bridge goto on the channel, it was + executing the same priority again instead of going to the next + priority. By swapping the "specific" version of + bridge_set_after_goto with bridge_set_after_go_on, the next + priority in the dialplan is executed instead. ASTERISK-24637 + #close Review: https://reviewboard.asterisk.org/r/4322/ Reported + by: John Bigelow + +2015-01-09 17:54 +0000 [r430434] Richard Mudgett <rmudgett@digium.com> + + * UPGRADE.txt, res/res_mwi_external_ami.c, CHANGES, + include/asterisk/manager.h, channels/chan_iax2.c, + apps/app_queue.c, apps/app_agent_pool.c, + res/res_manager_devicestate.c, main/manager_bridges.c, + channels/chan_dahdi.c, main/manager.c, channels/chan_skinny.c, + res/res_pjsip_outbound_registration.c, + res/res_manager_presencestate.c, + res/res_pjsip/pjsip_configuration.c, apps/app_confbridge.c, + res/res_pjsip_pubsub.c, main/db.c, res/parking/parking_manager.c, + res/res_pjsip_registrar.c, apps/app_voicemail.c, main/pbx.c, + channels/chan_sip.c, apps/app_meetme.c, main/bridge.c, + res/res_fax.c: AMI: Make AMI actions that generate event lists + consistent. * Made the following AMI actions use list API calls + for consistency: Agents BridgeInfo BridgeList + BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms + CoreShowChannels DAHDIShowChannels DBGet DeviceStateList + ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers + IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls + Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints + PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound + PJSIPShowResourceLists PJSIPShowSubscriptionsInbound + PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans + QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus + SIPshowregistry SKINNYdevices SKINNYlines Status + VoicemailUsersList * Incremented the AMI version to 2.7.0. * + Changed astman_send_listack() to not use the listflag parameter + and always set the value to "Start" so the start capitalization + is consistent. i.e., The FAXSessions used "Start" while the rest + of the system used "start". The corresponding complete event + always used "Complete". * Fixed ami_show_resource_lists() + "PJSIPShowResourceLists" to output the AMI ActionID for all of + its list events. * Fixed off-nominal AMI protocol error in + manager_bridge_info(), manager_parking_status_single_lot(), and + manager_parking_status_all_lots(). Use of astman_send_error() + after responding to the original AMI action request violates the + action response pattern by sending two responses. * Fixed minor + protocol error in action_getconfig() when no requested categories + are found. Each line needs to be formatted as "Header: text". * + Fixed off-nominal memory leak in + manager_build_parked_call_string(). * Eliminated unnecessary use + of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close + Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/4315/ + +2015-01-09 14:51 +0000 [r430416] Kinsey Moore <kmoore@digium.com> + + * /, res/res_fax.c, include/asterisk/res_fax.h, + configs/samples/res_fax.conf.sample, CHANGES: res_fax: Add T.38 + negotiation timeout option This change makes the T.38 negotiation + timeout configurable via 't38timeout' in res_fax.conf or + FAXOPT(t38timeout). It was previously hard coded to be 5000 + milliseconds. This change also handles T.38 switch failures by + aborting the fax since in the case where this can happen, both + sides have agreed to switch to T.38 and Asterisk is unable to do + so. Review: https://reviewboard.asterisk.org/r/4320/ ........ + Merged revisions 430415 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2015-01-08 21:40 +0000 [r430373-430397] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_pubsub.c: res_pjsip_pubsub: Fix persistent + subscriptions not surviving graceful shutdown If you do a 'core + (shutdown|restart) graceful' persistent subscriptions won't + survive. If you do a 'core (shutdown|restart) now' or asterisk + terminates for some reason, they do. Here's why... When asterisk + shuts down gracefully, it sends a 'NOTIFY/terminated' to + subscribers for each subscription. This not only tells the + subscribers that the dialog/state machine is done, it also frees + the last reference to the subscription tree which causes the + persistent subscription to get deleted from astdb. When asterisk + restarts, nothing's left. Just preventing the delete from astdb + doesn't work because we already told the subscriber to terminate + the dialog so we can't restart it even if it was still in astdb. + Everything works OK if asterisk terminates unexpectedly because + we never send the 'terminated' message so on restart, the + subscription is still in astdb and the subscriber is none the + wiser. This patch suppresses the sending of 'NOTIFY/terminated' + on shutdown for persistent connections. Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4318/ + + * res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Fix reference leak. Every time a + registration started, sip_outbound_registration_response_cb bumps + the ref count on client_state then pushes a + handle_registration_response task. handle_registration_response + never unreffed it though. So every time a registration goes out, + the ref count goes up by one. This patch adds the unreffs to + handle_registration_response. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4303/ + + * res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Fix several reload issues There + are 2 issues with reloading registrations... 1. The + 'can_reuse_registration' test wasn't considering the intervals or + expiration in its determination of whether a registration changed + or not so if you changed any of the intervals or the expiration + and reloaded, the object would get reloaded but the actual timers + wouldn't change. can_reuse_registration now does a sorcery diff + on the old and new objects instead of discretely testing certain + fields. Now if you change expiration for instance, and reload, + the timer is updated and re-registration will occur on the new + value. 2. If you mung up your password on an outbound + registration you get a permanent failure. If you fix the password + (on the outbound_auth object) and reload, nothing tells + outbound_registration to try again because the registration + itself didn't change. This patch adds an observer on the "auth" + object type and if any auth changes, existing registration states + are searched and those in a REJECTED_PERMANENT state are retried. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4304/ + +2015-01-07 21:25 +0000 [r430355] Kinsey Moore <kmoore@digium.com> + + * res/res_stasis.c: ARI: Allow usage of ASYNCGOTO with Stasis() + When the AMI Redirect action is used with a channel bridged + inside Stasis() and not running a pbx, the channel is hung up + instead of proceeding to the desired location in dialplan. This + change allows such channels to be Redirected properly by + detecting the operation used by Redirect (ASYNCGOTO) and using + the code already established for functionality of the ARI channel + continue operation. ASTERISK-24591 #close Review: + https://reviewboard.asterisk.org/r/4271/ + +2015-01-07 18:53 +0000 [r430337] Mark Michelson <mmichelson@digium.com> + + * rest-api/api-docs/channels.json, rest-api/resources.json, + res/ari/resource_channels.c, CHANGES, res/res_ari_channels.c, + res/ari/resource_channels.h: Add the ability to continue and + originate using priority labels. With this patch, the following + two ARI commands POST /channels POST /channels/{id}/continue + Accept a new parameter, label, that can be used to continue to or + originate to a priority label in the dialplan. Because this is + adding a new parameter to ARI commands, the API version of ARI + has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of + Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412 + #close Reported by Nir Simionovich Review: + https://reviewboard.asterisk.org/r/4285 + +2015-01-07 18:17 +0000 [r430315-430319] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Change 'does + not exist' warning to notice The 'new_subscribe: Extension <> + does not exist or has no associated hint' is a config issue and + doesn't need to clutter up logs with warnings. Changed to notice. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4307/ + + * res/res_pjsip_mwi.c: res_pjsip_mwi: Change "MWI Subscription + failed" message from warning to notice The "MWI Subscription + failed" message means the client is trying to subscribe to a + mailbox that doesn't exist. There's no need to clutter up logs + with warnings for a client misconfiguration so I changed it to a + notice. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4306/ + + * funcs/func_config.c, tests/test_config.c: func_config: Add + ability to retrieve specific occurrence of a variable I guess + nobody uses templates with AST_CONFIG because today if you have a + context that inherits from a template and you call AST_CONFIG on + the context, you'll get the value from the template even if + you've overridden it in the context. This is because AST_CONFIG + only gets the first occurrence which is always from the template. + This patch adds an optional 'index' parameter to AST_CONFIG which + lets you specify the exact occurrence to retrieve, or '-1' to + retrieve the last. The default behavior is the current behavior. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4313/ + +2015-01-07 17:35 +0000 [r430313] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_refer.c: Fix ability to perform a remote attended + transfer with PJSIP. This fix has two parts: * Corrected an error + message to properly state that external_replaces is an extension. + The error message also prints what dialplan context the + external_replaces extension was being looked for in. * Corrected + the printing of the Replaces: header in an INVITE request. We + were duplicating "Replaces: " in the header. ASTERISK-24376 + #close Reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/4296 + +2015-01-07 16:55 +0000 [r430295] George Joseph <george.joseph@fairview5.com> + + * include/asterisk/config.h, main/config.c, main/manager.c: config: + Add option to NOT preserve effective context when changing a + template Let's say you have a template T with variable VAR1 = ON + and you have a context C(T) that doesn't specify VAR1. If you + read C, the effective value of VAR1 is ON. Now you change T VAR1 + to OFF and call ast_config_text_file_save. The current behavior + is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is + added. Personally, I think this is a bug. It's preserving the + effective state of C even though I didn't specify C/VAR1 in th + first place. I believe the behavior should be that if I didn't + specify C/VAR1 originally, then the effective value of C/VAR1 + should continue to follow the inherited state. Now, if I DID + explicitly specify C/VAR1, the it should be preserved even if the + template changes. Even though I think the existing behavior is a + bug, it's been that way forever so I'm not changing it. Instead, + I've created ast_config_text_file_save2() that takes a bitmask of + flags, one of which is to preserve the effective context (the + current behavior). The original ast_config_text_file_save calls + *2 with the preserve flag. If you want the new behavior, call *2 + directly without a flag. I've also updated Manager UpdateConfig + with a new parameter 'PreserveEffectiveContext' whose default is + 'yes'. If you want the new behavior with UpdateConfig, set + 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4297/ + +2015-01-07 02:52 +0000 [r430274] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + main/rtp_engine.c: Fix dev-mode build on recent gcc + +2015-01-06 22:46 +0000 [r430252] Matthew Jordan <mjordan@digium.com> + + * contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py: + contrib/ast-db-manage: Correct down_revision path for + user_eq_phone When the user_eq_phone patch was backported to 13, + it referenced the downward revision that the PJSIP optimistic + encryption option also references. This creates a multi-path + upgrade Exception when generating the SQL files. This patch + corrects this in the 13 branch. Note that trunk, which already + contained both of these features, is unaffected by this problem. + +2015-01-06 17:52 +0000 [r430221-430227] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_mwi.c: res_pjsip_mwi: Change warning to notice When + res_pjsip loads and an endpoint auto-subscribes a mailbox for + mwi, if a contact hasn't registered yet, res_pjsip_mwi spits out + a warning. This is a perfectly normal situation though and + doesn't require something as serious as a warning. It's also self + correcting. The device will start getting mwi as soon as it + registers. This patch changes the warning to a notice. Tested-by: + George Joseph Review: https://reviewboard.asterisk.org/r/4314/ + + * bridges/bridge_native_rtp.c: bridge_native_rtp: Change + local/remote message from debug/2 to verb/4 Change the "Locally + bridged"/"Remotely bridged" messages from dbg/2 to verb/4. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4300/ + + * res/res_pjsip_outbound_registration.c, CHANGES: + outbound_registration: Add 'pjsip send register' and update 'send + unregister' The current behavior of 'pjsip send unregister' is to + send the unregister (REGISTER with 0 exp) but let the next + scheduled register proceed normally. I don't think that's a good + idea. If you unregister, it should stay unregistered until you + decide to start registrations again. So this patch just adds a + cancel_registration call to the current unregister_task to cancel + the timer. Of course, now you need a way to start registration + again so I've added a 'pjsip send register' command that + unregisters and cancels any existing registration (the same as + send unregister), then sends an immediate registration and starts + the timer back up again. Both changes also ripple to AMI. There's + a new PJSIPRegister command. There's no harm in calling either + command repeatedly. They don't care about the actual state. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4301/ + + * res/res_pjsip/location.c: pjsip cli: Fix sorting of contacts for + 'pjsip list contacts' For some reason I was using a hash + container instead of a list to gather the contacts for 'pjsip + list/show contacts' so even though I had a sort function, the + output wasn't sorted. This patch just changes the hash container + to a list container and the contacts now appear sorted in the + CLI. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4305/ + +2015-01-05 22:49 +0000 [r430200] Scott Griepentrog <sgriepentrog@digium.com> + + * /, main/bridge_basic.c: bridge: avoid leaking channel during + blond transfer pt2 A blond transfer to a failed destination, when + followed by a recall attempt, lead to a leak of the reference to + the destination channel. In addition to correcting the regression + on the previous attempt (r429826) this fixes the leak and two + additional reference leaks on failures of bridge_import. + ASTERISK-24513 #close Review: + https://reviewboard.asterisk.org/r/4302/ ........ Merged + revisions 430199 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2015-01-05 17:56 +0000 [r430179-430181] Joshua Colp <jcolp@digium.com> + + * CHANGES: pjsip: Document addition of 'PJSIP_AOR' and + 'PJSIP_CONTACT' in CHANGES file. + + * funcs/func_pjsip_contact.c (added), res/res_pjsip_session.c, + include/asterisk/res_pjsip.h, + channels/pjsip/dialplan_functions.c, + include/asterisk/res_pjsip_session.h, funcs/func_pjsip_aor.c + (added), res/res_pjsip/location.c: pjsip: Add 'PJSIP_AOR' and + 'PJSIP_CONTACT' dialplan functions. The PJSIP_AOR dialplan + function allows inspection of configured AORs including what + contacts are currently bound to them. The PJSIP_CONTACT dialplan + function allows inspection of contacts in existence. These can + include both externally added (by way of registration) or + permanent ones. ASTERISK-24341 Reported by: xrobau Review: + https://reviewboard.asterisk.org/r/4308/ + +2014-12-29 13:10 +0000 [r430145] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip.c: PJSIP: Update transport method documentation + This updates the documentation for the 'method' configuration + option to be more verbose about the behaviors of values + 'unspecified' and 'default'. They do exactly the same thing which + is to select the default as defined by PJSIP which is currently + TLSv1. Review: https://reviewboard.asterisk.org/r/4264/ + +2014-12-24 21:27 +0000 [r430127] Kevin Harwell <kharwell@digium.com> + + * /, configs/samples/queues.conf.sample: app_queue: Update sample + conf documenation Updated the queues.conf.sample file to + explicitly state which channel queue variables are propagated to. + ASTERISK-24267 Reported by: Mitch Claborn ........ Merged + revisions 430126 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-24 15:26 +0000 [r430083-430092] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip.c, + contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py + (added): res_pjsip: Backport missing commits for user_eq_phone + This backports the following from trunk, which were missed: + r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2 + lines res_pjsip: Allow + at the beginning of a phone number when + user_eq_phone is enabled. r427259 | file | 2014-11-04 16:51:32 + -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip: Apply the + 'user_eq_phone' setting to the To header as well. It also adds + the Alembic script for the option. ASTERISK-24643 + + * CHANGES, res/res_pjsip.c, include/asterisk/res_pjsip.h, + res/res_pjsip/config_global.c, res/res_pjsip_keepalive.c (added), + configs/samples/pjsip.conf.sample: res_pjsip_keepalive: Add + runtime configurable keepalive module for connection-oriented + transports. Note that this is backport from trunk of r425825. + This change adds a module which is configurable using the + keep_alive_interval setting in the global section that will send + a CRLF keep alive to all active connection-oriented transports at + the provided interval. This is useful because it can help keep + connections open through NATs. This functionality also exists + within PJSIP but can not be controlled at runtime and requires + recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ + ASTERISK-24644 #close + + * res/res_pjsip/pjsip_configuration.c, res/res_pjsip_caller_id.c, + CHANGES, res/res_pjsip.c, include/asterisk/res_pjsip.h: + res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' + parameter when applicable. Note that this is a backport of + r425804 from trunk. This change adds a configuration option which + adds a 'user=phone' parameter if the user portion of the request + URI or the From URI is determined to be a number. Review: + https://reviewboard.asterisk.org/r/4073/ ASTERISK-24643 #close + +2014-12-23 23:18 +0000 [r430059-430064] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip/pjsip_options.c: pjsip_options: Fix continued + qualifies after endpoint/aor deletion If you remove an + endpoint/aor from pjsip.conf then do a core reload, qualifies + will continue even though the object are gone. This happens + because nothing clears out the qualify tasks. This patch + unschedules all existing qualify tasks before scheduling new ones + on reload. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4290/ + + * tests/test_astobj2.c: test_astobj2: Fix warning for missing + trailing slash in category This patch adds a trailing slash to + the category for this test. No more warning. Tested-by: George + Joseph Review: https://reviewboard.asterisk.org/r/4295/ + +2014-12-22 21:18 +0000 [r430010-430034] Richard Mudgett <rmudgett@digium.com> + + * main/bridge_basic.c: DTMF atxfer: Setup recall channels as if the + transferee initiated the call. After the initial DTMF atxfer call + attempt to the transfer target fails to answer during a blonde + transfer, the recall callback channels do not get setup with + information from the initial transferrer channel. As a result, + the recall callback to the transferrer does not have callid, + channel variables, datastores, accountcode, peeraccount, COLP, + and CLID setup. A similar situation happens with the recall + callback to the transfer target but it is less visible. The + recall callback to the transfer target does not have callid, + channel variables, datastores, accountcode, peeraccount, and COLP + setup. * Added missing information to the recall callback + channels before initiating the call. callid, channel variables, + datastores, accountcode, peeraccount, COLP, and CLID * Set callid + of the transferrer channel on the DTMF atxfer controller thread + attended_transfer_monitor_thread(). * Added missing channel + unlocks and props unref to off nominal paths in + attended_transfer_properties_alloc(). ASTERISK-23841 #close + Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/4259/ + + * /, main/logger.c, include/asterisk/_private.h, main/asterisk.c: + queue_log: Post QUEUESTART entry when Asterisk fully boots. The + QUEUESTART log entry has historically acted like a fully booted + event for the queue_log file. When the QUEUESTART entry was + posted to the log was broken by the change made by + ASTERISK-15863. * Made post the QUEUESTART queue_log entry when + Asterisk fully boots. This restores the intent of that log entry + and happens after realtime has had a chance to load. AST-1444 + #close Reported by: Denis Martinez Review: + https://reviewboard.asterisk.org/r/4282/ ........ Merged + revisions 430009 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-22 15:40 +0000 [r429983] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: chan_sip: Send CANCEL via original INVITE + destination even after UPDATE request Given the following + scenario: * Three SIP phones (A, B, C), all communicating via a + proxy with Asterisk * A call is established between A and B. B + performs a SIP attended transfer of A to C. B sets the call on + hold (A is hearing MOH) and dials the extension of C. While phone + C is ringing, B transfers the call (that is, what we typically + call a 'blond transfer'). * When the transfer completes, A hears + the ringing of phone C, while B is idle. In the SIP messaging for + the above scenario, a REFER request is sent to transfer the call. + When "sendrpid=yes" is set in sip.conf, Asterisk may send an + UPDATE request to phone C to update party information. This + update is sent directly to phone C, not through the intervening + proxy. This has the unfortunate side effect of providing route + information, which is then set on the sip_pvt structure for C. If + someone (e.g. B) is trying to get the call back (through a + directed pickup), Asterisk will send a CANCEL request to C. + However, since we have now updated the route set, the CANCEL + request will be sent directly to C and not through the proxy. The + phone ignores this CANCEL according to RFC3261 (Section 9.1). + This patch updates reqprep such that the route is not updated if + an UPDATE request is being sent while the INVITE state is + INV_PROCEEDING or INV_EARLY_MEDIA. This ensures that a subsequent + CANCEL request is still sent to the correct location. Review: + https://reviewboard.asterisk.org/r/4279 ASTERISK-24628 #close + Reported by: Karsten Wemheuer patches: issue.patch uploaded by + Karsten Wemheuer (License 5930) ........ Merged revisions 429982 + from http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-22 00:17 +0000 [r429914] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_phoneprov_provider.c: + res_pjsip_phoneprovi_provider: Fix reload Reloading wasn't + working correctly because on a reload, the sorcery apply handler + was never being called for unchanged users. So, instead of using + an apply handler, I'm now iterating over all users. Works much + more reliably. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4288/ + +2014-12-20 20:57 +0000 [r429894] Joshua Colp <jcolp@digium.com> + + * main/named_acl.c, /: acl: Fix reloading of configuration if + configuration file does not exist at startup. The named ACL code + incorrectly destroyed the config options information if loading + of the configuration file failed at startup. This would result in + reloading also failing even if a valid configuration file was put + in place. ASTERISK-23733 #close Reported by: Richard Kenner + ........ Merged revisions 429893 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-19 20:54 +0000 [r429829-429868] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_http_websocket.c: res_http_websocket.c: Fix incorrect + use of sizeof in ast_websocket_write(). This won't fix the + reported issue but it is an incorrect use of sizeof. + ASTERISK-24566 Reported by: Badalian Vyacheslav ........ Merged + revisions 429867 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * channels/chan_dahdi.c, /: chan_dahdi: Don't ignore setvar when + using configuration section scheme. When the configuration + section scheme of chan_dahdi.conf is used (keyword dahdichan + instead of channel) all setvar= options are completely ignored. + No variable defined this way appears in the created DAHDI + channels. * Move the clearing of setvar values to after the + deferred processing of dahdichan. AST-1378 #close Reported by: + Guenther Kelleter Patch by: Guenther Kelleter ........ Merged + revisions 429825 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-19 17:26 +0000 [r429827] Scott Griepentrog <sgriepentrog@digium.com> + + * /, main/bridge_basic.c: bridge: avoid leaking channel during + blond transfer After a blond transfer (start attended and hang + up) to a destination that also hangs up without answer, the + Local;1 channel was leaked and would show up on core show + channels. This was happening because the attended state + blond_nonfinal_enter() resetting the props->transfer_target to + null while releasing it's own reference, which would later + prevent props from releasing another reference during + destruction. The change made here is simply to not assign the + target to NULL. ASTERISK-24513 #close Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4262/ ........ Merged + revisions 429826 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-18 22:38 +0000 [r429784-429805] Richard Mudgett <rmudgett@digium.com> + + * res/res_rtp_asterisk.c, channels/chan_dahdi.c, /: chan_dahdi.c, + res_rtp_asterisk.c: Change some spammy debug messages to level 5. + ASTERISK-24337 #close Reported by: Rusty Newton ........ Merged + revisions 429804 from + http://svn.asterisk.org/svn/asterisk/branches/11 + + * UPGRADE.txt, channels/sig_analog.c, /: chan_dahdi: Populate + CALLERID(ani2) for incoming calls in featdmf signaling mode. For + the featdmf signaling mode the incoming MF Caller-ID information + is formatted as follows: + *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}# Rather than + discarding the ani2 digits, populate the CALLERID(ani2) value + with what is received instead. AST-1368 #close Reported by: Denis + Martinez Patches: extract_ani2_for_featdmf_v11.patch (license + #5621) patch uploaded by Richard Mudgett ........ Merged + revisions 429783 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-18 15:50 +0000 [r429763] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: wrong bridge chosen + when the DTMF mode is not compatible A native rtp bridge was + being chosen (it shouldn't have been) when using two pjsip + channels with incompatible DTMF modes. This patch sets the rtp + instance property, AST_RTP_PROPERTY_DTMF, for the appropriate + DTMF mode(s) for pjsip. It was not being set before, meaning all + DTMF modes for pjsip were being treated as compatible, thus + native bridging would be chosen as the bridge type when it + shouldn't have been. ASTERISK-24459 #close Reported by: Yaniv + Simhi Review: https://reviewboard.asterisk.org/r/4265/ + +2014-12-18 15:34 +0000 [r429739-429761] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_outbound_registration.c: Prevent potential infinite + outbound authentication loops in registration. Prior to this + patch, Asterisk would always respond to 401 responses to + registration attempts by trying to provide a registration with + authentication credentials. Even if subsequent attempts were + rejected with 401 responses, Asterisk would continue this + behavior. If authentication credentials were incorrect, this + could continue forever. With this patch, we keep track of whether + we have attempted authentication on an outbound registration + attempt. If we already have, we don not try again until the next + attempt. This prevents the infinite loop scenario. Review: + https://reviewboard.asterisk.org/r/4273 + + * main/manager.c: Prevent possible race condition on dual redirect + of channels in the same bridge. The + AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent + bridges from prematurely acting on orphaned channels in bridges. + The problem with the AMI redirect action was that it was setting + this flag on channels based on the presence of a PBX, not whether + the channel was in a bridge. Whether a channel has a PBX is + irrelevant, so the condition has been altered to check if the + channel is in a bridge. ASTERISK-24536 #close Reported by Niklas + Larsson Review: https://reviewboard.asterisk.org/r/4268 + + * channels/pjsip/dialplan_functions.c: Ensure the correct value is + returned for CHANNEL(pjsip, secure) Prior to this patch, we were + using the PJSIP dialog's secure flag to determine if a secure + transport was being used. Unfortunately, the dialog's secure flag + was only set if a SIPS URI were in use, as required by RFC 3261 + sections 12.1.1 and 12.1.2. What we're interested in is not + dialog security, but transport security. This code change + switches to a model where we use the dialog's target URI to + determine what transport would be used to communicate, and then + check if that transport is secure. AST-1450 #close Reported by + John Bigelow Review: https://reviewboard.asterisk.org/r/4277 + +2014-12-18 00:10 +0000 [r429699-429719] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_config_wizard.c: res_pjsip_config_wizard: fix + unload SEGV If certain pjsip modules aren't loaded, the wizard + causes a SEGV when it unloads. Added a check for the presense of + the object type wizard before trying to clean it up. Tested-by: + George Joseph + + * res/res_pjsip_config_wizard.c: res_pjsip_config_wizard: Change + FILEUNCHANGED config_load2 flag determination The module now + applies the FILEUNCHANGED flag when both reloaded is specified + AND there's no last_config for the object type. Tested-by: George + Joseph Review: https://reviewboard.asterisk.org/r/4276/ + +2014-12-17 09:54 +0000 [r429675] Walter Doekes <walter+asterisk@wjd.nu> + + * addons/ooh323c/src/printHandler.c, apps/app_adsiprog.c, + channels/chan_unistim.c, main/udptl.c, res/res_rtp_asterisk.c, /, + channels/chan_sip.c, channels/vcodecs.c, res/res_crypto.c, + utils/astman.c, utils/smsq.c, main/utils.c, pbx/dundi-parser.c, + apps/app_getcpeid.c, channels/chan_iax2.c, channels/sig_pri.c, + res/res_pktccops.c, main/loader.c, channels/iax2/parser.c, + main/uuid.c, main/manager.c, channels/chan_misdn.c, + apps/app_osplookup.c, channels/misdn/ie.c, main/http.c, + apps/app_sms.c: Fix printf problems with high ascii characters + after r413586 (1.8). In r413586 (1.8) various casts were added to + silence gcc 4.10 warnings. Those fixes included things like: -out + += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += + sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii + characters, but for the high range that yields e.g. FFFFFFC3 when + C3 is expected. This changeset: - fixes those casts to use the + 'hh' unsigned char modifier instead - consistently uses %02x + instead of %2.2x (or other non-standard usage) - adds a few 'h' + modifiers in various places - fixes a 'replcaes' typo - + dev/urandon typo (in 13+ patch) Review: + https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close + Reported by: Stefan27 (on IRC) ........ Merged revisions 429673 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + Merged revisions 429674 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-16 17:53 +0000 [r429653] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_config_wizard.c: res_pjsip_config_wizard: fix test + breakage Fix test breakage caused by not checking for res_pjsip + before calling ast_sip_get_sorcery. Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4269/ + +2014-12-16 16:38 +0000 [r429612-429633] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: chan_sip: Allow T.38 switch-over when + SRTP is in use. Previously when SRTP was enabled on a channel it + was not possible to switch to T.38 as no crypto attributes would + be present. This change makes it so it is now possible. If a T.38 + re-invite comes in SRTP is terminated since in practice you can't + encrypt a UDPTL stream. Now... if we were doing T.38 over RTP + (which does exist) then we'd have a chance but almost nobody does + that so here we are. ASTERISK-24449 #close Reported by: Andreas + Steinmetz patches: udptl-ignore-srtp-v2.patch submitted by + Andreas Steinmetz (license 6523) ........ Merged revisions 429632 + from http://svn.asterisk.org/svn/asterisk/branches/11 + + * res/res_pjsip_t38.c: res_pjsip_t38: Fix T.38 failure when peer + reinvites immediately. If a remote endpoint reinvites to T.38 + immediately the state machine will go into a peer reinvite state. + If a T.38 capable application (such as ReceiveFax) queries it + will receive this state. Normally the application will then + indicate so that the channel driver will queue up the T.38 offer + previously received. Once it receives this offer the application + will act normally and negotiate. The res_pjsip_t38 module + incorrectly partially squashed this indication. This would cause + the application to think the request had failed when in reality + it had actually worked. This change makes it so that no T.38 + control frames (or indications) are squashed. + +2014-12-15 17:07 +0000 [r429592] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_phoneprov_provider.c, + configs/samples/pjsip_wizard.conf.sample (added), CHANGES, + res/res_pjsip_config_wizard.c (added): res_pjsip_config_wizard: + Allow streamlined config of common pjsip scenarios + res_pjsip_config_wizard ------------------ * This is a new module + that adds streamlined configuration capability for chan_pjsip. + It's targetted at users who have lots of basic configuration + scenarios like 'phone' or 'agent' or 'trunk'. Additional + information can be found in the sample configuration file at + config/samples/pjsip_wizard.conf.sample. Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4190/ + +2014-12-15 15:36 +0000 [r429571] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c: Activate persistent subscriptions when + they are recreated. Prior to this change, recreating persistent + subscriptions would create the subscription but would not + activate it. This led to subscriptions being listed in the "NULL" + state by diagnostics and not sending NOTIFYs when expected. + Review: https://reviewboard.asterisk.org/r/4261 + +2014-12-12 23:54 +0000 [r429542] George Joseph <george.joseph@fairview5.com> + + * main/manager.c, include/asterisk/module.h, + include/asterisk/_private.h: loader: Move definition of + ast_module_reload from _private.h to module.h No functionality + change. Just move the definition of ast_module_reload from + _private.h to module.h so it can be public. Also removed the + include of _private.h from manager.c since ast_module_load was + the only reason for including it. Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4251/ + +2014-12-12 23:40 +0000 [r429540] Richard Mudgett <rmudgett@digium.com> + + * main/lock.c, /, include/asterisk/lock.h: DEBUG_THREADS: Fix + regression and lock tracking initialization problems. This patch + started with David Lee's patch at + https://reviewboard.asterisk.org/r/2826/ and includes a + regression fix introduced by the ASTERISK-22455 patch. The + initialization of a mutex's lock tracking structure was not + protected in a critical section. This is fine for any mutex that + is explicitly initialized, but a static mutex may have its lock + tracking double initialized if multiple threads attempt the first + lock simultaneously. * Added a global mutex to properly serialize + initialization of the lock tracking structure. The painful global + lock can be mitigated by adding a double checked lock flag as + discussed on the original review request. * Defer lock tracking + initialization until first use. * Don't be "helpful" and + initialize an uninitialized lock when DEBUG_THREADS is enabled. + Debug code is not supposed to fix or change normal code behavior. + We don't need a lock initialization race that would force a + re-setup of lock tracking. Lock tracking already handles + initialization on first use. * Properly handle allocation + failures of the lock tracking structure. * No need to initialize + tracking data in __ast_pthread_mutex_destroy() just to turn + around and destroy it. The regression introduced by + ASTERISK-22455 is the result of manipulating a pthread_mutex_t + struct outside of the pthread library code. The pthread_mutex_t + struct seems to have a global linked list pointer member that can + get changed by other threads. Therefore, saving and restoring the + contents of a pthread_mutex_t struct is a bad thing. Thanks to + Thomas Airmont for finding this obscure regression. * Don't + overwrite the struct ast_lock_track.reentr_mutex member to + restore tracking data in __ast_cond_wait() and + __ast_cond_timedwait(). The pthread_mutex_t struct must be + treated as a read-only opaque variable. Miscellaneous other items + fixed by this patch: * Match ast_suspend_lock_info() with + ast_restore_lock_info() in __ast_cond_timedwait(). * Made some + uninitialized lock sanity checks return EINVAL and try a + DO_THREAD_CRASH. * Fix bad canlog initialization expressions. + ASTERISK-24614 #close Reported by: Thomas Airmont Review: + https://reviewboard.asterisk.org/r/4247/ Review: + https://reviewboard.asterisk.org/r/2826/ ........ Merged + revisions 429539 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-12-12 22:53 +0000 [r429518-429519] Matthew Jordan <mjordan@digium.com> + + * res/res_agi.c: res/res_agi: Make Verbose message for 'stream + file' match other playbacks The Verbose message displayed when a + file is played back via 'stream file' was formatted differently + than other playbacks: * It didn't include the channel name * It + didn't include the channel language It does, however, include the + playback offset as well as any escape digits. That information + was kept; however, this patch updates the formatting to more + closely match the Verbose messages displayed when a file is + played back by 'control stream file', Playback, ControlPlayback, + or any other file playback operation. + + * /: Add 11 merge properties + +2014-12-12 16:57 +0000 [r429497] Joshua Colp <jcolp@digium.com> + + * main/format.c, main/codec.c, include/asterisk/format.h: media: + Fix crash when determining sample count of a frame during + shutdown. When shutting down Asterisk the codecs are cleaned up. + As a result anything attempting to get a codec based on ID or + details will find that no codec exists. This currently occurs + when determining the sample count of a frame. This code did not + take this situation into account. This change fixes this by + getting the codec directly from the format and eliminates the + lookup. This is both faster and also provides a guarantee that + the codec will exist and will be valid. ASTERISK-24604 #close + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4260/ + +2014-12-12 15:30 +0000 [r429477] Kevin Harwell <kharwell@digium.com> + + * channels/chan_pjsip.c: chan_pjsip: Race between channel answer + and bridge setup when using direct media When direct media is + enabled and a pjsip channel is answered a race would occur + between the handling of the answer and bridge setup. Sometimes + the media negotiation would take place after the native bridge + was setup. This resulted in a NULL media address, which in turn + resulted in Asterisk using its address as the remote media + address when sending a reinvite. This patch makes the chan_pjsip + answer handler synchronous thus alleviating the race condition + (the bridge won't start setting things up until after it + returns). ASTERISK-24563 #close Reported by: Steve Pitts Review: + https://reviewboard.asterisk.org/r/4257/ + +2014-12-12 15:00 +0000 [r429457] David M. Lee <dlee@digium.com> + + * res/res_pjsip_outbound_publish.c: Fix crash for sorcery + misconfigs res_pjsip_outbound_publish was missing the + CHECK_PJSIP_MODULE_LOADED() call in load_module, and would crash + with a segfault if res_pjsip declined to load. Review: + https://reviewboard.asterisk.org/r/4258/ + +2014-12-12 14:12 +0000 [r429430-429433] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip_sdp_rtp.c: PJSIP: Allow use of 'inactive' + streams for hold This allows use of the 'inactive' stream + direction identifier to be used for hold where 'sendonly' is + normally used. Some Seimens phones use 'inactive' and this change + allows music on hold to operate properly. Review: + https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts + ........ Merged revisions 429432 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_sorcery_config.c, /: Sorcery: Log when old config remains + in use This adds a log message notifying the user that a stale + configuration is in place upon reload when a config object fails + to load. This situation can end up causing confusion when the + object failed to load but exists from a previous config load + especially when the old config is significantly different from + the new config. Review: https://reviewboard.asterisk.org/r/4250/ + Reported by: Thomas Thompson ........ Merged revisions 429429 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-12 13:05 +0000 [r429407-429409] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.exports.in, channels/chan_pjsip.c, + res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h: + res_pjsip_session: Delay sending BYE if a re-INVITE transaction + is in progress. Given the scenario where a PJSIP channel is in a + native RTP bridge with direct media and the channel is then hung + up the code will currently re-INVITE the channel back to Asterisk + and send a BYE at the same time. Many SIP implementations dislike + this greatly. This change makes it so that if a re-INVITE + transaction is in progress the BYE is queued to occur after the + completion of the transaction (be it through normal means or a + timeout). Review: https://reviewboard.asterisk.org/r/4248/ + + * res/res_pjsip_session.c: res_pjsip_session: Fix issue where a + declined media stream in a re-INVITE would fail SDP negotiation. + In the past the SDP negotiation within res_pjsip_session was made + more tolerant of certain situations. The only case where SDP + negotiation will fail is when a major error occurs during + negotiation. Receiving an already declined media stream is not + considered a major error. When producing the local SDP the logic + took this into account so on the initial INVITE the declined + media stream did not cause an SDP negotiation failure. + Unfortunately the logic for handling media streams with a handler + did not mirror this logic and considered an already declined + media stream an error and thus failed the SDP negotiation. This + change makes the logic between both situations match so only + under major errors will the SDP negotiation fail. ASTERISK-24607 + #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4254/ + +2014-12-11 20:31 +0000 [r429387] Kevin Harwell <kharwell@digium.com> + + * CHANGES: ARI/AMI: Include language in standard channel snapshot + output The CHANGES verbiage for the "language" addition had been + put under the wrong release. This moves it to be under 13.1 to + 13.2 changes. ASTERISK-24553 Reported by: Matt Jordan + +2014-12-11 17:21 +0000 [r429352-429379] Kinsey Moore <kmoore@digium.com> + + * /: Recorded merge of revisions 429378 from + http://svn.asterisk.org/svn/asterisk/branches/12 ........ Fix + incorrect patch applied in r429354 The patch that was applied was + another pending patch. This swaps them out. + + * /: Recorded merge of revisions 429354 from + http://svn.asterisk.org/svn/asterisk/branches/12 ........ Stasis: + Update unittest for channel snapshots This adjusts the unit test + for channel snapshots to take the new language key into account. + + * tests/test_stasis_channels.c: Stasis: Update unittest for channel + snapshots This adjusts the unit test for channel snapshots to + take the new language key into account. + +2014-12-10 15:42 +0000 [r429326] Kevin Harwell <kharwell@digium.com> + + * /, CHANGES: ARI/AMI: Include language in standard channel + snapshot output Adding information about including "language" in + the standard channel snapshot output to the CHANGES file. Note + the actual source changes have already been previously committed. + ASTERISK-24553 Reported by: Matt Jordan ........ Merged revisions + 429325 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-10 13:34 +0000 [r429273] Joshua Colp <jcolp@digium.com> + + * res/res_http_websocket.c, res/res_pjsip_transport_websocket.c, /, + channels/chan_sip.c: res_http_websocket: Fix crash due to double + freeing memory when receiving a payload length of zero. Frames + with a payload length of 0 were incorrectly handled in + res_http_websocket. Provided a frame with a payload had been + received prior it was possible for a double free to occur. The + realloc operation would succeed (thus freeing the payload) but be + treated as an error. When the session was then torn down the + payload would be freed again causing a crash. The read function + now takes this into account. This change also fixes assumptions + made by users of res_http_websocket. There is no guarantee that a + frame received from it will be NULL terminated. ASTERISK-24472 + #close Reported by: Badalian Vyacheslav Review: + https://reviewboard.asterisk.org/r/4220/ Review: + https://reviewboard.asterisk.org/r/4219/ ........ Merged + revisions 429270 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 429272 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-10 13:14 +0000 [r429246] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip/pjsip_options.c: PJSIP: Fix assert on initial + mass qualify This fixes the MWI test regressions caused by + r429127 and ensures that contacts have non-zero qualify_frequency + before attempting scheduling. ........ Merged revisions 429245 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-09 20:46 +0000 [r429223] Scott Griepentrog <sgriepentrog@digium.com> + + * main/asterisk.c: core: avoid possible asterisk -r crash from long + id When connecting to the remote console, an id string is first + provided that consts of the hostname, pid, and version. This is + parsed by the remote instance using a buffer that may be too + short, and can allow a buffer overrun because it is not + terminated. This patch adds termination and a larger buffer. + Review: https://reviewboard.asterisk.org/r/4182/ + +2014-12-09 20:19 +0000 [r429175-429206] Kevin Harwell <kharwell@digium.com> + + * res/ari/ari_model_validators.h, /, main/stasis_channels.c, + rest-api/api-docs/channels.json, res/ari/ari_model_validators.c, + main/manager_channels.c: ARI/AMI: Include language in standard + channel snapshot output The channel "language" was already part + of a channel snapshot, however is was not sent out over AMI or + ARI. This patch makes it so the channel "language" is included in + the appropriate AMI or ARI events. ASTERISK-24553 #close Reported + by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4245/ + ........ Merged revisions 429204 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, + main/rtp_engine.c, /, channels/chan_sip.c: Direct Media calls + within private network sometimes get one way audio When endpoints + with direct_media enabled, behind a firewall (Asterisk on a + separate network) and were bridged sometimes Asterisk would send + the ip address of the firewall in the sdp to one of the phones in + the reinvite resulting in one way audio. When sending the + reinvite Asterisk will retrieve the media address from the + associated rtp instance, but if frames were being read this can + be overwritten with another address (in this case the + firewall's). This patch ensures that Asterisk uses the original + device address when using direct media. ASTERISK-24563 Reported + by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/ + ........ Merged revisions 429195 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_outbound_publish.c: res_pjsip_outbound_publish: + stack overflow when using non-default sorcery wizard When using a + non-default sorcery wizard (in this instance realtime) for + outbound publishes Asterisk will crash after a stack overflow + occurs due to the code infinitely recursing. The fix entails + removing the outbound publish state dependency from the outbound + publish sorcery object and instead keeping an in memory container + that can be used to lookup the state when needed. ASTERISK-24514 + #close Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/4178/ + +2014-12-09 15:44 +0000 [r429153] Joshua Colp <jcolp@digium.com> + + * res/ari/resource_channels.h, rest-api/api-docs/channels.json, + res/ari/resource_channels.c, CHANGES, res/res_ari_channels.c: + ari: Add support for specifying an originator channel when + originating. If an originator channel is specified when + originating a channel the linked ID of it will be applied to the + newly originated outgoing channel. This allows an association to + be made between the two so it is known that the originator has + dialed the originated channel. ASTERISK-24552 #close Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/ + +2014-12-09 14:00 +0000 [r429128] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip/pjsip_options.c: PJSIP: Stagger outbound + qualifies This change staggers initiation of outbound qualify + (OPTIONS) attempts to reduce instantaneous server load and + prevent network congestion. Review: + https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close + Reported by: Richard Mudgett ........ Merged revisions 429127 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-15 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.1.0 Released. + +2014-12-10 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.1.0-rc2 Released. + + * AST-2014-019: Fix crash when receiving a WebSocket packet with a + payload length of zero. + + Frames with a payload length of 0 were incorrectly handled in + res_http_websocket. Provided a frame with a payload had been + received prior it was possible for a double free to occur. The + realloc operation would succeed (thus freeing the payload) but be + treated as an error. When the session was then torn down the payload + would be freed again causing a crash. The read function now takes + this into account. + + This change also fixes assumptions made by users of + res_http_websocket. There is no guarantee that a frame received from + it will be NULL terminated. + + ASTERISK-24472 #close + Reported by: Badalian Vyacheslav + +2014-12-08 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.1.0-rc1 Released. + +2014-12-08 16:53 +0000 [r429091] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/playbacks.json, UPGRADE.txt, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, CHANGES, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json: AMI/ARI: Update version to + 2.6.0/1.6.0 respectively for new features AMI/ARI are getting a + few enhancements in the next release of Asterisk 13. Per semantic + versioning, that warrants a bump in the minor version number, as + it reflects a backwards compatible change. Hence, this commit. + +2014-12-08 16:41 +0000 [r429064-429089] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_session.c: Fix a crash that would occur when + receiving a 491 response to a reinvite. The reviewboard + description does a fine job of summarizing this, so here it is: A + reporter discovered that Asterisk would crash when attempting to + retransmit a reinvite that had previously received a 491 + response. The crash occurred because a pjsip_tx_data structure + was being saved for reuse, but its reference count was not being + increased. The result was that the pjsip_tx_data was being freed + before we were actually done with it. When we attempted to re-use + the structure when re-sending the reinvite, Asterisk would crash. + The fix implemented here is not to try holding onto the + pjsip_tx_data at all. Instead, when we reschedule sending the + reinvite, we create a brand new pjsip_tx_data and send that + instead. Because of this change, there is no need for an + ast_sip_session_delayed_request structure to have a pjsip_tx_data + on it any more. So any code referencing its use has been removed. + When this initial fix was introduced, I encountered a second + crash when processing a subsequent 200 OK on a rescheduled + reinvite. The reason was that when rescheduling the reinvite, we + gave the wrong location for a response callback. This has been + fixed in this patch as well. ASTERISK-24556 #close Reported by + Abhay Gupta Review: https://reviewboard.asterisk.org/r/4233 + + * main/stasis_channels.c, CHANGES, res/ari/ari_model_validators.c, + main/manager_channels.c, main/channel.c, + res/ari/ari_model_validators.h, + include/asterisk/stasis_channels.h, + rest-api/api-docs/events.json, res/stasis/app.c: Add new AMI and + ARI events for connected line changes on a channel. The AMI event + is called NewConnectedLine and the ARI event is called + ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt + Jordan Review: https://reviewboard.asterisk.org/r/4231 + +2014-12-08 15:43 +0000 [r429062] Kinsey Moore <kmoore@digium.com> + + * /, res/stasis/app.c, main/channel_internal_api.c, + res/stasis/stasis_bridge.c, res/stasis/app.h, + include/asterisk/channel.h, res/res_stasis.c, main/channel.c: + Stasis: Fix StasisStart/End order and missing events This + corrects several bugs that currently exist in the stasis + application code. * After a masquerade, the resulting channels + have channel topics that do not match their uniqueids ** + Masquerades now swap channel topics appropriately * StasisStart + and StasisEnd messages are leaked to observer applications due to + being published on channel topics ** StasisStart and StasisEnd + publishing is now properly restricted to controlling apps via app + topics * Race conditions exist where StasisStart and StasisEnd + messages due to a masquerade may be received out of order due to + being published on different topics ** These messages are now + published directly on the app topic so this is now a non-issue * + StasisEnds are sometimes missing when sent due to masquerades and + bridge swaps into and out of Stasis() ** This was due to + StasisEnd processing adjusting message-sent flags after Stasis() + had already exited and Stasis() had been re-entered ** This was + corrected by adjusting these flags prior to sending the message + while the initial Stasis() application was still shutting down + Review: https://reviewboard.asterisk.org/r/4213/ ASTERISK-24537 + #close Reported by: Matt DiMeo ........ Merged revisions 429061 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-06 18:16 +0000 [r429029-429033] Matthew Jordan <mjordan@digium.com> + + * res/res_monitor.c, /: res/res_monitor: Reset in/out sample counts + on Monitor start When repeatedly starting/stopping a Monitor on a + channel, the accumulated in/out sample counts are never reset to + 0. This can cause inadvertent jumps in the recordings, as the + code in the channel core will determine incorrectly that a jump + in the recorded file position should occur. Setting the sample + counts to 0 simply reflects the initial state a Monitor should be + in when it is started, as this is the initial count that would be + on the channels at that time. ASTERISK-24573 #close Reported by: + Nuno Borges patches: 24573.patch uploaded by Nuno Borges (License + 6116) ........ Merged revisions 429031 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 429032 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_meetme.c: apps/app_meetme: Apply default values on + initial load with no config file When the app_meetme module is + loaded without its configuration file, the module settings aren't + initialized. In particular, this impacts the use of logging + realtime members. This patch guarantees that we always set the + default module settings on initial load. Review: + https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close + Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno + Borges (License 6116) ........ Merged revisions 429027 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 429028 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-05 17:06 +0000 [r429000] George Joseph <george.joseph@fairview5.com> + + * tests/test_sorcery.c, main/sorcery.c, include/asterisk/test.h, /, + include/asterisk/sorcery.h: sorcery: Add additional observer + capabilities. Add new global, instance and wizard observers. + instance_created wizard_registered wizard_unregistered + instance_destroying instance_loading instance_loaded + wizard_mapped object_type_registered object_type_loading + object_type_loaded wizard_loading wizard_loaded Tested-by: George + Joseph Review: https://reviewboard.asterisk.org/r/4215/ ........ + Merged revisions 428999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-04 17:13 +0000 [r428865-428973] Matthew Jordan <mjordan@digium.com> + + * /, main/test.c: main/test: Fix compilation issue on 32-bit + systems On a 32-bit system, a type of intmax_t will result in a + compilation warning when formatted as a 'long int'. Use the + format specifier of %jd (which was what was used originally in + manager.c) to format the JSON extracted integer on both + 32-/64-bit systems. ........ Merged revisions 428972 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /, main/test.c: main/test: Fix race condition + between AMI topic and Test Suite topic This patch fixes a race + condition between the raising of test AMI events (which drive + many tests in the Asterisk Test Suite) and other AMI events. + Prior to this patch, the Stasis messages published to the test + topic were not forwarded to the AMI topic. Instead, the code in + manager had a dedicated handler for test messages that was + independent of the topics forwarded to the AMI topic. This + results in no synchronization between the test messages and the + rest of the Stasis messages published out over AMI. In some test + with very tight timing constraints, this can result in out of + order messages and spurious test failures. Properly forwarding + the Test Suite topic to the AMI topic ensures that the messages + are synchronized properly. This patch does that, and moves the + message handling to the Stasis definition of the Test Suite + message in test.c as well. Review: + https://reviewboard.asterisk.org/r/4221/ ........ Merged + revisions 428945 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c, /: tests/test_cel: Add + test_cel_attended_transfer_bridges_link to racey tests Despite + failing less often, the ordering of the ATTENDEDTRANSFER event + and the BRIDGE_EXIT event for the Alice and David channels is not + defined. This makes the test still fail. ........ Merged + revisions 428918 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c, /: tests/test_cel: Fix CEL unit test failures + caused by attended transfer changes When the publication of + attended transfer messages were pushed to another thread, some + subtle race conditions were introduced with the CEL unit tests. + This patch fixes one of them, and pushes the other to + ASTERISK-22367, which already exists to fix another bouncy CEL + unit test. In particular, this patch fixes the + test_cel_attended_transfer_bridges_link test, and defers the + test_cel_attended_transfer_bridges_swap test to the + aforementioned JIRA issue. ASTERISK-22367 ........ Merged + revisions 428891 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_voicemail.c, /: apps/app_voicemail: Fix crash with IMAP + when streams are opened simultaneously The UW IMAP library is + instrinsically not thread-safe, and relies upon higher level + applications to guarantee thread safety. For the most part, this + is provided by the vms object, which provides locking for + individual streams. Unfortunately, this is not sufficient for + calls to mail_open which create the IMAP stream. mail_open can, + on some systems, call into a UW IMAP specific function for + determining the address of a system based on a hostname, + ip_nametoaddr. In the ip6_unix implementation of this function, + static variables are used to hold parsing buffers. This can cause + a crash if multiple threads attempt to convert a hostname to an + address at the same time. Locking on a single mail stream is not + sufficient to prevent simultaneous access to these static + variables. In the IMAP library, this function can be called from + the mail_open and imap_status functions. As the imap_status + function is not used by app_voicemail, locking on access to + mail_open is sufficient to prevent any mangling of the buffers. + Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516 + #close Reported by: David Duncan Ross Palmer Tested by: David + Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David + Duncan Ross Palmer (License 6660) ........ Merged revisions + 428863 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 428864 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-02 21:53 +0000 [r428837] George Joseph <george.joseph@fairview5.com> + + * CHANGES, /: CHANGES: Add item for new 'pjsip show identif(y|ies) + commands Tested-by: George Joseph ........ Merged revisions + 428836 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-02 19:03 +0000 [r428789-428815] Matthew Jordan <mjordan@digium.com> + + * tests/test_stasis.c: tests/test_stasis: Resolve compilation + issues from Asterisk 12 merge When merging the changes up stream + in r428687, I missed the fact that the signature for + stasis_message_type_create was changed. This patch fixes the + compilation issues introduced by that merge. + + * pbx/pbx_loopback.c, /: pbx/pbx_loopback: Speed up switches by + avoiding unneeded lookups This patch makes a small rearrangement + to only do dialplan lookups during loopback switches if the + pattern matches. Prior to this patch, the dialplan lookups were + always performed, even when the result would be discarded. + Dialplan lookups can be very costly if remote switches - like + DUNDi - are present. In those cases extension matching is sped up + considerably, making the issue of lost digits more manageable. As + collateral damage, 6 trailing spaces were killed. Review: + https://reviewboard.asterisk.org/r/4211 ASTERISK-24577 #close + Reported by: Birger Harzenetter patches: ast-loopback.patch + uploaded by Birger Harzenetter (License 5870) ........ Merged + revisions 428787 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428788 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-02 12:20 +0000 [r428761] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_refer.c, /: res_pjsip_refer: Fix issue where native + bridge may not occur upon completion of a transfer. There are two + methods within res_pjsip_refer for keeping track of the state of + a transfer. The first is a framehook which looks at frames + passing by to determine the state. The second subscribes to know + when the channel joins a bridge. In the case when the channel + joins the bridge the framehook is *NOT* removed and this prevents + the native RTP bridging technology from getting used. This change + gets the channel and if it still exists remove the framehook. + Review: https://reviewboard.asterisk.org/r/4218/ ........ Merged + revisions 428760 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-02 00:38 +0000 [r428731-428734] George Joseph <george.joseph@fairview5.com> + + * /, include/asterisk/config.h, main/config.c: config: Create + ast_variable_find_in_list() Add const char + *ast_variable_find_in_list(const struct ast_variable *list, const + char *variable); ast_variable_find() requires a config category + to search whereas ast_variable_find_in_list() just needs the root + list element which is useful if you don't have a category. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4217/ ........ Merged + revisions 428733 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip/pjsip_cli.c: res_pjsip_endpoint_identifier_ip: Add + 'show identify(ies)' cli commands While troubleshooting other + things I realized there were no pjsip cli commands for identify. + This patch adds them. It also also fixes a reference leak when a + 'show endpoint' displayed identifies and properly sets the return + code if load_module can't allocate a cli formatter structure. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4212/ ........ Merged + revisions 428725 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-01 17:57 +0000 [r428687] Matthew Jordan <mjordan@digium.com> + + * channels/chan_skinny.c, res/res_pjsip_mwi.c, tests/test_stasis.c, + res/res_pjsip_pubsub.c, res/res_pjsip_refer.c, + channels/chan_mgcp.c, main/stasis_cache.c, channels/chan_sip.c, + include/asterisk/stasis_internal.h, /, include/asterisk/stasis.h, + UPGRADE.txt, configs/samples/stasis.conf.sample, + res/parking/parking_applications.c, res/res_xmpp.c, + channels/chan_iax2.c, apps/app_queue.c, + res/res_stasis_device_state.c, channels/sig_pri.c, + include/asterisk/stasis_message_router.h, main/endpoints.c, + res/parking/parking_bridge_features.c, main/stasis.c, + channels/chan_dahdi.c, main/stasis_message_router.c: main/stasis: + Allow subscriptions to use a threadpool for message delivery + Prior to this patch, all Stasis subscriptions would receive a + dedicated thread for servicing published messages. In contrast, + prior to r400178 (see review + https://reviewboard.asterisk.org/r/2881/), the subscriptions + shared a thread pool. It was discovered during some initial work + on Stasis that, for a low subscription count with high message + throughput, the threadpool was not as performant as simply having + a dedicated thread per subscriber. For situations where a + subscriber receives a substantial number of messages and is + always present, the model of having a dedicated thread per + subscriber makes sense. While we still have plenty of + subscriptions that would follow this model, e.g., AMI, CDRs, CEL, + etc., there are plenty that also fall into the following two + categories: * Large number of subscriptions, specifically those + tied to endpoints/peers. * Low number of messages. Some + subscriptions exist specifically to coordinate a single message - + the subscription is created, a message is published, the delivery + is synchronized, and the subscription is destroyed. In both of + the latter two cases, creating a dedicated thread is wasteful + (and in the case of a large number of peers/endpoints, harmful). + In those cases, having shared delivery threads is far more + performant. This patch adds the ability of a subscriber to Stasis + to choose whether or not their messages are dispatched on a + dedicated thread or on a threadpool. The threadpool is + configurable through stasis.conf. Review: + https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close + Reported by: xrobau Tested by: xrobau ........ Merged revisions + 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-12-01 13:41 +0000 [r428632-428655] Joshua Colp <jcolp@digium.com> + + * /, apps/app_record.c: app_record: Fix bug where using the 'k' + option and hanging up would trim 1/4 of a second of the + recording. The Record dialplan function trims 1/4 of a second + from the end of recordings in case they are terminated because of + DTMF. When hanging up, however, you don't want this to happen. + This change makes it so on hangup this does not occur. + ASTERISK-24530 #close Reported by: Ben Smithurst patches: + app_record_v2.diff submitted by Ben Smithurst (license 6529) + Review: https://reviewboard.asterisk.org/r/4201/ ........ Merged + revisions 428653 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428654 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel.c: channel: Extend size of buffer for codecs in + "core show channeltype" CLI command. The static buffer for codecs + when invoking the "core show channeltype" CLI command did not + have enough room for all codecs. This has been extended so it + does. ASTERISK-24542 #close Reported by: snuffy patches: + channeltype-tech.diff submitted by snuffy (license 5024) Review: + https://reviewboard.asterisk.org/r/4204/ + +2014-11-24 20:37 +0000 [r428602-428604] Richard Mudgett <rmudgett@digium.com> + + * tests/test_channel_feature_hooks.c: test_channel_feature_hooks.c: + Fix unit test for DTMF hooks. Fix the failing + /channels/features/test_features_channel_dtmf unit test. DTMF + emulation does not work without a stream of packets to prod the + emulation code. Review: https://reviewboard.asterisk.org/r/4199/ + + * /, main/bridge.c, main/bridge_channel.c: DTMF hooks: Leaving + channels need to push any collected digits into the bridge. Any + partially collected DTMF digits for a DTMF hook need to be pushed + into the bridge when a channel leaves the bridging system as if + there were a timeout. Review: + https://reviewboard.asterisk.org/r/4199/ ........ Merged + revisions 428601 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-21 19:09 +0000 [r428572] Richard Mudgett <rmudgett@digium.com> + + * main/manager.c, /: manager: Fix could not extend string messages. + When shutting down Asterisk that has an active AMI connection, + you get several "failed to extend from %d to %d" messages because + use of the EVENT_FLAG_SHUTDOWN attempts to add all AMI permission + strings to the event. * Created MAX_AUTH_PERM_STRING to use when + creating stack based struct ast_str variables used with the + authority_to_str() and user_authority_to_str() functions instead + of a variety of magic numbers that could be too small. * Added a + special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so it + will not attempt to add all permission level strings. Review: + https://reviewboard.asterisk.org/r/4200/ ........ Merged + revisions 428570 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428571 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-21 17:45 +0000 [r428544] George Joseph <george.joseph@fairview5.com> + + * main/sorcery.c, /, res/res_pjsip_phoneprov_provider.c, + tests/test_sorcery.c: sorcery: Make is_object_field_registered + handle field names that are regexes. As a result of + https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime was + tossing database fields that didn't have an exact match to a + sorcery registered field. This broke the ability to use regexes + as field names which manifested itself as a failure of + res_pjsip_phoneprov_provider which uses this capability. It also + broke handling of fields that start with '@' in realtime but I + don't think anyone noticed. This patch does the following... * + Modifies ast_sorcery_fields_register to pre-compile the name + regex. * Modifies ast_sorcery_is_object_field_registered to test + the regex if it exists instead of doing an exact strcmp. * + Modifies res_pjsip_phoneprov_provider with a few tweaks to get it + to work with realtime. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4185/ ........ Merged + revisions 428543 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-21 02:16 +0000 [r428505] Matthew Jordan <mjordan@digium.com> + + * main/bridge_basic.c: main/bridge_basic: Fix features regressions + introduced by r428165 In r428165, two bugs were introduced: * + Prior to entering the features retry loop, the buffer that holds + the collected digits is wiped. However, this inadvertently wipes + out the first collected digit on the first pass through, which is + obtained in ast_stream_and_wait. This caused all of the features + tests to fail. * If ast_app_dtget returns a hangup (-1), the loop + would retry incorrectly. If we detect a hangup, we have to stop + trying the feature. This patch fixes both issues. Review: + https://reviewboard.asterisk.org/r/4196/ + +2014-11-20 16:36 +0000 [r428425] Mark Michelson <mmichelson@digium.com> + + * main/acl.c, /: Fix error with mixed address family ACLs. Prior to + this commit, the address family of the first item in an ACL was + used to compare all incoming traffic. This could lead to traffic + of other IP address families bypassing ACLs. ASTERISK-24469 + #close Reported by Matt Jordan Patches: ASTERISK-24469-11.diff + uploaded by Matt Jordan (License #6283) AST-2014-012 ........ + Merged revisions 428402 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 428417 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428422 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-20 16:34 +0000 [r428413] Kevin Harwell <kharwell@digium.com> + + * funcs/func_db.c, /: AST-2014-018 - func_db: DB Dialplan function + permission escalation via AMI. The DB dialplan function when + executed from an external protocol (for instance AMI), could + result in a privilege escalation. Asterisk now inhibits the DB + function from being executed from an external interface if the + live_dangerously option is set to no. ASTERISK-24534 Reported by: + Gareth Palmer patches: submitted by Gareth Palmer (license 5169) + ........ Merged revisions 428331 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 428363 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428409 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-20 16:13 +0000 [r428343] Jonathan Rose <jrose@digium.com> + + * res/res_pjsip_acl.c, /: PJSIP ACLs: Fix ACLs not loading on + startup and apply/acl issues on contact The biggest problem this + patch fixes is that ACLs weren't previously being loaded when the + res_pjsip_acl module was loaded. Yikes. In addition, the ACL + options contact_permit and contact_acl were effectively + interpreted as contact_deny and this patch fixes that as well. + AST-1418 #close Reported by: Thomas Thompson Review: + https://reviewboard.asterisk.org/r/4120/ ASTERISK-24531 #close + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4171/ ........ Merged + revisions 428333 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-20 15:50 +0000 [r428339] Kevin Harwell <kharwell@digium.com> + + * apps/app_confbridge.c, /: AST-2014-017 - app_confbridge: + permission escalation/ class authorization. Confbridge dialplan + function permission escalation via AMI and inappropriate class + authorization on the ConfbridgeStartRecord action. The CONFBRIDGE + dialplan function when executed from an external protocol (for + instance AMI), could result in a privilege escalation. Also, the + AMI action “ConfbridgeStartRecord” could also be used to execute + arbitrary system commands without first checking for system + access. Asterisk now inhibits the CONFBRIDGE function from being + executed from an external interface if the live_dangerously + option is set to no. Also, the “ConfbridgeStartRecord” AMI action + is now only allowed to execute under a user with system level + access. ASTERISK-24490 Reported by: Gareth Palmer ........ Merged + revisions 428332 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428334 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-20 14:55 +0000 [r428302-428305] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_refer.c, /: AST-2014-016: Fix crash when receiving + an in-dialog INVITE with Replaces in res_pjsip_refer. The + implementation of INVITE with Replaces in res_pjsip_refer did not + expect them to occur in-dialog. As a result it would incorrectly + attempt to hang up a channel it thought was under its control. In + reality the channel would be under the control of another thread. + When the other thread accessed the channel it would be accessing + freed memory and could crash. This change makes res_pjsip_refer + not act on an in-dialog INVITE with Replaces. ASTERISK-24528 + #close Reported by: Joshua Colp ........ Merged revisions 428304 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_pjsip.c, /: AST-2014-015: Fix race condition in + chan_pjsip when sending responses after a CANCEL has been + received. Due to the serialized architecture of chan_pjsip there + exists a race condition where a CANCEL may be received and + processed before responses (such as 180 Ringing, 183 Session + Progress, and 200 OK) are sent. Since the session is in an + unexpected state PJSIP will assert when this is attempted. This + change makes it so that these responses are not sent on + disconnected sessions. ASTERISK-24471 #close Reported by: yaron + nahum ........ Merged revisions 428301 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-19 19:31 +0000 [r428273] Corey Farrell <git@cfware.com> + + * include/asterisk/stringfields.h, /: stringfields: Fix bug in + ast_string_fields_copy. ast_string_fields_copy relies on the fact + that __ast_string_field_release_active never previously zeroed + pool->used, so keeping the existing pointer was "ok". Now that + existing pools can be reset to 'empty', it is important to set + each field to __ast_string_field_empty after releasing the + memory. ASTERISK-24535 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4186/ ........ Merged + revisions 428272 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-19 17:13 +0000 [r428246] Richard Mudgett <rmudgett@digium.com> + + * res/res_calendar.c, main/manager.c, /, channels/chan_sip.c, + channels/sip/security_events.c: ast_str: Fix improper member + access to struct ast_str members. Accessing members of struct + ast_str outside of the string manipulation API routines is + invalid since struct ast_str is supposed to be treated as opaque. + Review: https://reviewboard.asterisk.org/r/4194/ ........ Merged + revisions 428244 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428245 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-19 12:40 +0000 [r428196-428222] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, include/asterisk/res_pjsip.h, + include/asterisk/res_pjsip_session.h, res/res_pjsip_sdp_rtp.c, + res/res_pjsip/pjsip_configuration.c, + configs/samples/pjsip.conf.sample, + contrib/ast-db-manage/config/versions/eb88a14f2a_add_media_encryption_optimistic_to_pjsip.py + (added), CHANGES, res/res_pjsip.c: res_pjsip_sdp_rtp: Add support + for optimistic SRTP. Optimistic SRTP is the ability to enable + SRTP but not have it be a fatal requirement. If SRTP can be used + it will be, if not it won't be. This gives you a better chance of + using it without having your sessions fail when it can't be. + Encrypt all the things! Review: + https://reviewboard.asterisk.org/r/3992/ + + * res/res_pjsip_refer.c, /: res_pjsip_refer: Ensure Refer-To is + NULL terminated and parse it as a URI. There is no guarantee that + when we get a Refer-To that it will be NULL terminated. As the + URI parsing function requires it to be we now NULL terminate it. + Additionally parsing the Refer-To as a 'To' header is needless + and it can simply be done as a URI. This also fixes a problem + where certain Refer-To headers would not be parsed as a 'To' + header causing the REFER to fail. ASTERISK-24508 #close Reported + by: Beppo Mazzucato Review: + https://reviewboard.asterisk.org/r/4187/ ........ Merged + revisions 428195 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-18 18:54 +0000 [r428169] Richard Mudgett <rmudgett@digium.com> + + * /, res/parking/parking_tests.c: parking_tests.c: Add missing + newline on a unit test message. ........ Merged revisions 428168 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-17 16:51 +0000 [r428145] Mark Michelson <mmichelson@digium.com> + + * CHANGES, main/features_config.c, + configs/samples/features.conf.sample, + include/asterisk/features_config.h, main/bridge_basic.c: Allow + for transferer to retry when dialing an invalid extension. This + allows for a configurable number of attempts for a transferer to + dial an extension to transfer the call to. For Asterisk 13, the + default values are such that upgrading between versions will not + cause a behaivour change. For trunk, though, the defaults will be + changed to be more user-friendly. Review: + https://reviewboard.asterisk.org/r/4167 + +2014-11-17 16:00 +0000 [r428119] Corey Farrell <git@cfware.com> + + * /, channels/chan_sip.c: chan_sip: Fix theoretical leak of + p->refer. If transmit_refer is called when p->refer is already + allocated, it leaks the previous allocation. Updated code to + always free previous allocation during a new allocation. Also + instead of checking if we have a previous allocation, always + create a clean record. ASTERISK-15242 #close Reported by: David + Woolley Review: https://reviewboard.asterisk.org/r/4160/ ........ + Merged revisions 428117 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428118 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-17 15:27 +0000 [r428079-428115] Matthew Jordan <mjordan@digium.com> + + * /, apps/confbridge/conf_state_multi_marked.c: + apps/app_confbridge: Ensure 'normal' users hear message when last + marked leaves When r428077 was made for ASTERISK-24522, it failed + to take into account users who are neither wait_marked nor + end_marked. These users are *also* supposed to hear the 'leader + has left the conference' message. Granted, this behaviour is a + bit odd; however, that is how it used to work... and behaviour + changes are not good. This patch ensures that if there are any + 'normal' users present when the last marked user leaves the + conference, the message will still be played to them. Note that + this regression was caught by the Asterisk Test Suite's + confbridge_nominal test, which has a quirky combination of users. + ........ Merged revisions 428113 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428114 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/confbridge/conf_state_multi_marked.c: app_confbridge: + Don't play leader leaving prompt if no one will hear it Consider + the following: - A marked user in a conference - One or more + end_marked only users in the conference When the marked users + leaves, we will be in the conf_state_multi_marked state. This + currently will traverse the users, kicking out any who have the + end_marked flags. When they are kicked, a full ast_bridge_remove + is immediately called on the channels. At this time, we also + unilaterally set the need_prompt flag. When the need_prompt flag + is set, we then playback a sound to the bridge informing everyone + that the leader has left; however, no one is left in the bridge. + This causes some odd behaviour for the end_marked users - they + are stuck waiting for the bridge to be unlocked. This results in + them waiting for 5 or 6 seconds of dead air before hearing that + they've been kicked. Unfortunately, we do have to keep the bridge + locked while we're playing back the 'leader-has-left' prompt. If + there are any wait_marked users in the conference, this behaviour + can't be easily changed - but we do make the case of the + end_marked users better with this patch. Review: + https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close + Reported by: Matt Jordan ........ Merged revisions 428077 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 428078 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-16 21:12 +0000 [r427979-428052] Joshua Colp <jcolp@digium.com> + + * channels/chan_pjsip.c, /: chan_pjsip: Remove AOR check when + dialing and one is specified. The AOR value may contain the name + of an AOR or a full SIP URI. Checking if the AOR exists can't be + done as a result of this. ........ Merged revisions 428051 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_pjsip.c: chan_pjsip: Add additional log message + when an AOR is specified when dialing and it does not exist. + ASTERISK-24499 #close Reported by: Rusty Newton ........ Merged + revisions 428007 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_motif.c, channels/chan_pjsip.c, /: chan_motif / + chan_pjsip: Fix incorrect "No such module" messages when + reloading. For chan_motif the direct return value of the + underlying config options framework was passed back. This can + relay various states which the module loader would not interpet + as success. It has been changed so only on errors will it report + back an error. For chan_pjsip the code implemented a dummy reload + function which always returned an error. This has been removed as + all configuration is held within res_pjsip instead. + ASTERISK-23651 #close Reported by: Rusty Newton ........ Merged + revisions 427981 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Enforce + requirements for session timer minimum expiration period and + normal expiration period. This change enforces the requirements + in PJSIP for session timer configuration. The minimum expiration + period must be 90 seconds or higher and the normal expiration + period can not be lower than the minimum expiration period. If + either of these were done the code would assert at session setup + time. ASTERISK-24336 #close Reported by: Leon Rowland ........ + Merged revisions 427978 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-15 16:56 +0000 [r427927-427954] Matthew Jordan <mjordan@digium.com> + + * cel/cel_odbc.c, /: cel/cel_odbc: Provide microsecond precision in + 'eventtime' column when possible This patch adds microsecond + precision when inserting a CEL record into a table with an + "eventtime" column of type timestamp, instead of second + precision. The documentation (configs/cel_odbc.conf.sample) was + already saying that the eventtime column included microseconds + precision, but that was not the case. Also, without this patch, + if you had a table with an "eventtime" column of type varchar, + you had millisecond precision. With this patch, you also get + microsecond precision in this case. Review: + https://reviewboard.asterisk.org/r/3980 ASTERISK-24283 #close + Reported by: Etienne Lessard patches: + cel_odbc_time_precision.patch uploaded by Etienne Lessard + (License 6394) ........ Merged revisions 427952 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427953 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c: tests/test_cel: Unlock bridge on off nominal + paths If the test fails due to memory allocation errors, we may + as well attempt to unlock the bridge on the way out. + +2014-11-14 17:45 +0000 [r427902] Jonathan Rose <jrose@digium.com> + + * configs/samples/cdr.conf.sample, main/cdr.c, /: Documentation: + Revise explanation of cdr.conf option 'Unanswered' ASTERISK-24279 + #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4109/ ........ Merged + revisions 427901 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-14 15:51 +0000 [r427876] Scott Griepentrog <sgriepentrog@digium.com> + + * /, main/stun.c: stun: correct attribute string padding to match + rfc When sending the USERNAME attribute in an RTP STUN response, + the implementation in append_attr_string passed the actual + length, instead of padding it up to a multiple of four bytes as + required by the RFC 3489. This change adds separate variables for + the string and padded attributed lengths, and performs padding + correctly. Reported by: Thomas Arimont Review: + https://reviewboard.asterisk.org/r/4139/ ........ Merged + revisions 427874 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427875 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-14 15:24 +0000 [r427870] Mark Michelson <mmichelson@digium.com> + + * main/bridge.c, main/bridge_basic.c, + include/asterisk/stasis_bridges.h, tests/test_cel.c, + apps/app_queue.c, main/cel.c, main/stasis_bridges.c, /, + res/stasis/app.c: Fix race condition that could result in ARI + transfer messages not being sent. From reviewboard: "During blind + transfer testing, it was noticed that tests were failing + occasionally because the ARI blind transfer event was not being + sent. After investigating, I detected a race condition in the + blind transfer code. When blind transferring a single channel, + the actual transfer operation (i.e. removing the transferee from + the bridge and directing them to the proper dialplan location) is + queued onto the transferee bridge channel. After queuing the + transfer operation, the blind transfer Stasis message is + published. At the time of publication, snapshots of the channels + and bridge involved are created. The ARI subscriber to the blind + transfer Stasis message then attempts to determine if the bridge + or any of the involved channels are subscribed to by ARI + applications. If so, then the blind transfer message is sent to + the applications. The way that the ARI blind transfer message + handler works is to first see if the transferer channel is + subscribed to. If not, then iterate over all the channel IDs in + the bridge snapshot and determine if any of those are subscribed + to. In the test we were running, the lone transferee channel was + subscribed to, so an ARI event should have been sent to our + application. Occasionally, though, the bridge snapshot did not + have any channels IDs on it at all. Why? The problem is that + since the blind transfer operation is handled by a separate + thread, it is possible that the transfer will have completed and + the channels removed from the bridge before we publish the blind + transfer Stasis message. Since the blind transfer has completed, + the bridge on which the transfer occurred no longer has any + channels on it, so the resulting bridge snapshot has no channels + on it. Through investigation of the code, I found that attended + transfers can have this issue too for the case where a transferee + is transferred to an application." The fix employed here is to + decouple the creation of snapshots for the transfer messages from + the publication of the transfer messages. This way, snapshots can + be created to reflect what they are at the time of the transfer + operation. Review: https://reviewboard.asterisk.org/r/4135 + ........ Merged revisions 427848 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-14 14:56 +0000 [r427846] Joshua Colp <jcolp@digium.com> + + * /, apps/confbridge/conf_state_multi_marked.c: app_confbridge: + Play "leader has left" sound even when musiconhold is enabled. + Currently if the leader of a conference bridge leaves any + participant that has musiconhold enabled will not hear the + "leader has left" sound. This is because musiconhold is started + and THEN the sound is played. This change makes it so that the + sound is played and THEN musiconhold is started. This provides a + better experience for users as they may not have known previously + why they went back to musiconhold. Review: + https://reviewboard.asterisk.org/r/4177/ ........ Merged + revisions 427844 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427845 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-14 14:24 +0000 [r427841] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c, + include/asterisk/res_pjsip.h: Fix race condition where duplicated + requests may be handled by multiple threads. This is the Asterisk + 13 version of the patch. The main difference is in the pubsub + code since it was completely refactored between Asterisk 12 and + 13. Review: https://reviewboard.asterisk.org/r/4175 + +2014-11-13 22:03 +0000 [r427815] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip_outbound_registration.c: res_pjsip_exten_state: + PJSIPShowSubscriptionsInbound causes crash When using a + non-default sorcery wizard (in this instance realtime) for + outbound registrations and after adding in an appropriate call to + ast_sorcery_apply_config() (since it is missing) Asterisk will + crash after a stack overflow occurs due to the code infinitely + recursing. The fix entails removing the outbound registration + state dependency from the outbound registration sorcery object + and instead keeping an in memory container that can be used to + lookup the state when needed. ASTERISK-24514 Reported by: Mark + Michelson Review: https://reviewboard.asterisk.org/r/4164/ + ........ Merged revisions 427814 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-13 15:44 +0000 [r427789] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/stasis.h, include/asterisk/stasis_app.h, + res/stasis/app.h, res/res_stasis.c, /, res/stasis/app.c, + res/stasis/stasis_bridge.c: Stasis: Fix StasisEnd message + ordering This change corrects message ordering in cases where a + channel-related message can be received after a Stasis/ARI + application has received the StasisEnd message. The StasisEnd + message was being passed to applications directly without waiting + for the channel topic to empty. As a result of this fix, other + bugs were also identified and fixed: * StasisStart messages were + also being sent directly to apps and are now routed through the + stasis message bus properly * Masquerade monitor datastores were + being removed at the incorrect time in some cases and were + causing StasisEnd messages to not be sent * General refactoring + where necessary for the above * Unsubscription on StasisEnd + timing changes to prevent additional messages from following the + StasisEnd when they shouldn't A channel sanitization function + pointer was added to reduce processing and AO2 lookups. Review: + https://reviewboard.asterisk.org/r/4163/ ASTERISK-24501 #close + Reported by: Matt Jordan ........ Merged revisions 427788 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-13 00:00 +0000 [r427763] Matthew Jordan <mjordan@digium.com> + + * main/rtp_engine.c, /: main/rtp_engine: Fix crash when processing + more than one RTCP report info block Asterisk - in + res_rtp_asterisk - only understands a single RTCP report info + block. When the RTCP information was refactored in the RTP Engine + to be pushed over the Stasis message bus, I put in the hooks into + the engine to handle multiple RTCP report info blocks, in the + hope that a future RTP implementation would be able to provide + that data. Unfortunately, res_rtp_asterisk has a tendency to + "lie": (1) It will send RTCP reports with a + reception_report_count greater than 1 (which is pulled directly + from the RTCP packet itself, so that part is correct) (2) It will + only provide a single report block When the rtp_engine goes to + convert this to a JSON blob, hilarity ensues as it looks for a + report block that doesn't exist. This patch updates the + rtp_engine to be a bit more skeptical about what it is presented + with. While this could also be fixed in res_rtp_asterisk, this + patch prefers to fix it in the engine for two reasons: (1) The + engine is designed to work with multiple RTP implementation, and + hence having it be more robust is a good thing (tm) (2) + res_rtp_asterisk's handling of RTCP information is "fun". It + should report the correct reception_report_count; ideally it + should also be giving us all of the blocks - but it is + *definitely* not designed to do that. Going down that road is a + non-trivial effort. Review: + https://reviewboard.asterisk.org/r/4158/ ASTERISK-24489 #close + Reported by: Gregory Malsack Tested by: Gregory Malsack + ASTERISK-24498 #close Reported by: Beppo Mazzucato Tested by: + Beppo Maazucato ........ Merged revisions 427762 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-12 20:39 +0000 [r427737] Corey Farrell <git@cfware.com> + + * /, main/features.c: Fix leak in AMI Action Bridge Add missing + reference cleanup for newly created bridge. ASTERISK-24281 + Reported by: Stefan Engström Review: + https://reviewboard.asterisk.org/r/4154/ ........ Merged + revisions 427736 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-12 16:12 +0000 [r427711] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, /: pbx: Fix off-nominal case where a freed extension + may still be used. If during the operation of adding an extension + a priority is added but fails it is possible for the extension to + be freed but still exist in the PBX core. If this occurs + subsequent lookups may try to access the extension and end up in + freed memory. This change removes the extension from the PBX core + when the priority addition fails and then frees the extension. + ASTERISK-24444 #close Reported by: Leandro Dardini Review: + https://reviewboard.asterisk.org/r/4162/ ........ Merged + revisions 427709 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427710 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-12 13:46 +0000 [r427684] Corey Farrell <git@cfware.com> + + * codecs/ilbc, /, tests, codecs/speex, apps/confbridge, + Makefile.rules: Fix compiler error when using ./configure + --enable-dev-mode --enable-coverage When DONT_OPTIMIZE is enabled + with dev-mode, it causes a shadow compilation to be done with + output to /dev/null. This can cause errors with coverage when GCC + attempts to write to /dev/null.gcno. This change disables + coverage for the shadow compilation. ASTERISK-24502 #close + Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4151/ ........ Merged + revisions 427682 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427683 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-09 08:00 +0000 [r427643] Corey Farrell <git@cfware.com> + + * main/manager.c, /: manager: Fix HTTP connection reference leaks. + Fix reference leak that happens if (session && !blastaway). + ASTERISK-24505 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4153/ ........ Merged + revisions 427641 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427642 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-09 00:38 +0000 [r427583-427615] Matthew Jordan <mjordan@digium.com> + + * channels/chan_mgcp.c, /: channels/chan_mgcp: Fix regression which + causes gateways to be skipped In r227276, a while loop was turned + into a for loop. Unfortunately, a portion of the while loop was + left in the code such that, when a static gateway is encountered + in the list of MGCP gateways, the next gateway would be skipped. + At best, we would simply flip past a gateway; at worst, this + could lead to a crash. ASTERISK-24500 #close Reported by: Xavier + Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne + (License 6657) ........ Merged revisions 427613 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427614 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, addons/chan_mobile.c: addons/chan_mobile: Increase buffer size + of UCS2 encoded SMS messages When UCS2 character encoding is + used, one symbol in national language can be expanded to 4 bytes. + The current buffer used for receiving message in do_monitor_phone + is 256 bytes, which is not large enough for incoming messages. + For example: * AT+CMGR phone response prefix '+CMGR: "REC + UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes * + SMS body with UCS2 encoding (max) - 280 bytes * AT+CMGR phone + response suffix '\r\n\r\nOK\r\n' - 8 bytes * Terminating null + character - 1 byte This results in a needed buffer size of 349 + bytes. Hence, this patch opts for a 350 byte buffer. + ASTERISK-24468 #close Reported by: Dmitriy Bubnov patches: + chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651) + chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651) + ........ Merged revisions 427607 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427610 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_voicemail.c: app_voicemail: Fix enhancement that allowed + multiple recipients in To: header An issue existed in r420577, + which added multiple recipients to voicemail emails. The patch, + when looking at the intended recipients, looked ahead for the '|' + character inside a while loop which already had pulled out the + appropriate field parsing on the '|' character. This would cause + it to skip the recipients. This patch fixes it such that it + relies completely on the while loop to parse through the e-mail + fields. Note that the original author of the patch looked at this + fix and approved it. ASTERISK-24250 #close Reported by: abelbeck + patches: voicemail-420577-to-comma-fix.diff uploaded by abelbeck + (License 5903) + + * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix T.38 + issues with remote bridges After r425242 the + fax/sip/directmedia_reinvite_t38 test started failing due to the + surviving channel not being re-INVITEd back from T.38 to audio. + This patch fixes that bug - a deeper explanation of what happened + follows. When two RTP channels are in a native bridge, the + bridging layer will investigate each via the get_rtp_info glue + callback. This callback returns the native bridge preference of + the channel *at that moment in time* (that part is key). At + different points during the bridging, the native bridging layer + will inform the RTP capable channels of the status of the bridge + via the update_peer glue callback. In a T.38 scenario with audio + direct media, the sequence of events will often look like the + following: * SIP/A and SIP/B both have audio and enter a native + bridge. * Asterisk re-INVITEs audio between SIP/A and SIP/B + directly (via an update_peer callback). * SIP/A sends a re-INVITE + to T.38, which causes Asterisk to send a re-INVITE to T.38 to + SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack + receives UDPTL packets in Asterisk from both endpoints. From the + perspective of the channels, we are now in a local bridge for + T.38, even though we are technically still in a remote bridge in + bridge_native_rtp. (YAY!) * When one side hangs up, + bridge_native_rtp is told to stop bridging. It then re-evaluates + the channels and asks them how they are bridged - and since T.38 + is enabled, they reply with a Local bridge (which is correct), + but is wrong because the audio portion is still technically in a + remote bridge. * Asterisk releases the surviving channel, whose + audio is *not* re-INVITED back to Asterisk as bridge_native_rtp + incorrectly assumes that it was in a local bridge. Ironically, + prior to r425242, this used to work mostly due to a fluke in the + bridging layer. The purpose of the get_rtp_info callback + shouldn't be modified: it should tell the bridging layer what + kind of bridge the channel prefers at that moment in time. If you + have T.38 enabled, that *must* be a local bridge, as the UDPTPL + stack must be in the media path. As such, this patch does not + modify that part of the code. However, we have to tell the + channels to re-evaluate themselves when they come out of a native + bridge, since we can no longer trust the get_rtp_info callbacks + when the native bridge is being stopped. Something else may have + changed in the channels, and they may now be lying to us. As + such, this patch makes it so that we unilaterally tell the + channels that they are no longer bridged via the update_peer + callback. This is actually what the channels expect anyway: code + in both chan_sip and chan_pjsip's callbacks look at the T.38 + state and - if they were in T.38 - send a re-INVITE to get the + audio back to Asterisk. Review: + https://reviewboard.asterisk.org/r/4157/ ........ Merged + revisions 427582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-08 18:17 +0000 [r427557] Corey Farrell <git@cfware.com> + + * /, channels/chan_console.c: chan_console: Fix reference leaks to + pvt. Fix a bunch of calls to get_active_pvt where the reference + is never released. ASTERISK-24504 #close Reported by: Corey + Farrell Review: https://reviewboard.asterisk.org/r/4152/ ........ + Merged revisions 427554 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427555 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-06 19:22 +0000 [r427494-427512] Richard Mudgett <rmudgett@digium.com> + + * apps/app_agent_pool.c, /: app_agent_pool: Made agent alert + interruptable by DTMF. Made agent able to interrupt the alerting + beep playback with DTMF. Any digit can interrupt if the call does + not need to be acknowledged. Only the first digit of the + acknowledgement can interrupt if the call needs to be + acknowledged. The agent interrupting the alerting playback builds + on the ASTERISK-24447 patch because it knows what digit + interrupted the playback and needs to be able to pass that digit + to the DTMF hook digit collection code. ASTERISK-24257 #close + Reported by: Steve Pitts Review: + https://reviewboard.asterisk.org/r/4123/ ........ Merged + revisions 427508 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, include/asterisk/bridge_channel.h, main/bridge_channel.c: + Bridge DTMF hooks: Made audio pass from the bridge while waiting + for more matching digits. * Made collecting DTMF digits for the + DTMF feature hooks pass frames from the bridge. * Made collecting + DTMF digits possible by other bridge hooks if there is a need. + ASTERISK-24447 #close Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/4123/ ........ Merged + revisions 427493 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-06 18:20 +0000 [r427491] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/pjsip_distributor.c: res_pjsip: Ensure in-dialog + responses have an endpoint associated. When handling incoming + messages we determine if it is associated with a dialog. If so we + use that to determine what serializer and endpoint to use for the + message. Previously this would pass the endpoint to the endpoint + lookup module to actually place the endpoint completely on the + message. For in-dialog responses, however, this did not occur as + dialog processing took over and the endpoint lookup did not + occur. This change just places the endpoint in the expected spot + immediately instead of relying on the endpoint lookup module. + In-dialog responses thus have the expected endpoint. AST-1459 + #close Review: https://reviewboard.asterisk.org/r/4146/ ........ + Merged revisions 427490 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-06 12:13 +0000 [r427384-427466] Corey Farrell <git@cfware.com> + + * main/file.c, /: main/file.c: fix possible extra ast_module_unref + to format modules. fn_wrapper only adds a reference to the + format's module if the file was able to be opened. If not this + causes an unmatched ast_module_unref in filestream_destructor. + Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4149/ ........ Merged + revisions 427464 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_hep.c, /: res_hep: fix major leak that occurs when config + is missing or enabled=no. Add missing unreference in + hepv3_send_packet. ASTERISK-24491 #close Reported by: Zane Conkle + Tested by: Zane Conkle Review: + https://reviewboard.asterisk.org/r/4150/ ........ Merged + revisions 427400 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/utils.c, include/asterisk/stringfields.h: Fix unintential + memory retention in stringfields. * Fix missing / unreachable + calls to __ast_string_field_release_active. * Reset pool->used to + zero when the current pool->active reaches zero. ASTERISK-24307 + #close Reported by: Etienne Lessard Tested by: ibercom, Etienne + Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ + Merged revisions 427380 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 427381 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427382 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-06 02:37 +0000 [r427356] George Joseph <george.joseph@fairview5.com> + + * tests/test_strings.c, /: test_strings: Remove string tests that + exercise asserts. Since unit tests are run with DO_CRASH, those + tests were causing the test to fail. Tested-by: George Joseph + ........ Merged revisions 427354 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427355 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-05 19:52 +0000 [r427334] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip/config_system.c, configs/samples/pjsip.conf.sample, + res/res_pjsip.c: Make the disable_tcp_switch PJSIP system object + enabled by default. Testing has shown repeatedly that PJSIP's + default behavior of switching automatically to TCP for large + messages can cause issues. The most common issues are that + devices that we are communicating with do not handle the switch + to TCP gracefully, thus causing situations such as broken calls + or broken subscriptions. Now, in order to have this behavior + happen, you must opt into it. The sample file has been updated to + warn that enabling the TCP switch behavior may cause issues for + you, so use at your own risk. + +2014-11-05 12:18 +0000 [r427303] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Add logging + during startup to aid debugging if local DNS is misbehaving. This + change adds a bit of logging so if the local DNS is misbehaving + it is easier to track down what is going on and where Asterisk + may be hanging. ASTERISK-24438 #close Reported by: Melissa + Shepherd Review: https://reviewboard.asterisk.org/r/4148/ + ........ Merged revisions 427300 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-05 00:15 +0000 [r427228-427276] George Joseph <george.joseph@fairview5.com> + + * pbx/pbx_config.c, main/config.c, tests/test_strings.c, + include/asterisk/utils.h, /, main/utils.c: config: Make + text_file_save and 'dialplan save' escape semicolons in values. + When a config file is read, an unescaped semicolon signals + comments which are stripped from the value before it's stored. + Escaped semicolons are then unescaped and become part of the + value. Both of these behaviors are normal and expected. When the + config is serialized either by 'dialplan save' or + AMI/UpdateConfig however, the now unescaped semicolons are + written as-is. If you actually reload the file just saved, the + unescaped semicolons are now treated as start of comments. Since + true comments are stripped on read, any semicolons in + ast_variable.value must have been escaped originally. This patch + re-escapes semicolons in ast_variable.values before they're + written to file either by 'dialplan save' or + config/ast_config_text_file_save which is called by + AMI/UpdateConfig. I also fixed a few pre-existing formatting + issues nearby in pbx_config.c Tested-by: George Joseph + ASTERISK-20127 #close Review: + https://reviewboard.asterisk.org/r/4132/ ........ Merged + revisions 427275 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, /: config: BUG: Restore ability for non-templ to + be used as base objs in config. My recent refactor of config.c + accidentally removed the capability for an object to inherit from + a non-template object. This patch restores the capability to + inherit from both template and non-template objects. Tested-by: + George Joseph Reported-by: Scott Griepentrog ASTERISK-24487 + #close Review: https://reviewboard.asterisk.org/r/4147/ ........ + Merged revisions 427227 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-04 19:44 +0000 [r427181-427204] Corey Farrell <git@cfware.com> + + * funcs/func_talkdetect.c, /: func_talkdetect: Fix stasis message + leak in audiohook callback. ASTERISK-24482 #close Reported by: + Corey Farrell Review: https://reviewboard.asterisk.org/r/4142/ + ........ Merged revisions 427203 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_http_websocket.c: res_http_websockets: Fix extra unref + of module In websocket_add_protocol_internal is used to add the + "echo" protocol, but ast_websocket_remove_protocol is used to + remove it. This causes an extra call to ast_module_unref. + ASTERISK-24480 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4140/ ........ Merged + revisions 427200 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/app.c: Fix crash caused by merge error on review 4138 When + merging from 12 to 13 there were conflicts, I mistakenly had the + loop run ast_closestream(others[0]) when it should be + ast_closestream(others[x]). + +2014-11-03 18:15 +0000 [r427130] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_pjsip/config_system.c, UPGRADE.txt, + configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip: + Add disable_tcp_switch option. When a packet exceeds the MTU, + pjproject will switch from UDP to TCP. In some circumstances (on + some networks), this can cause some issues with messages not + getting sent to the correct destination - and can also cause + connections to get dropped due to quirks in pjproject deciding to + terminate TCP connections with no messages. While fixing the + routing/messaging issues is important, having a configuration + option in Asterisk that tells pjproject to not switch over to TCP + would be useful. That way, if some glitch is discovered on some + other network/site, we can at least disable the behavior until a + fix is put into place. AFS-197 #close Review: + https://reviewboard.asterisk.org/r/4137/ ........ Merged + revisions 427129 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-03 02:34 +0000 [r427021-427089] Corey Farrell <git@cfware.com> + + * apps/app_voicemail.c, /: Fix compile error caused by review 4138 + There is no procedure called ast_closeframe, fix code to use + ast_closestream. Reported By: Matt Jordan ........ Merged + revisions 427087 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427088 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/app.c, apps/app_voicemail.c, /: Fix ast_writestream leaks + Fix cleanup in __ast_play_and_record where others[x] may be + leaked. This was caught where prepend != NULL && outmsg != NULL, + once realfile[x] == NULL any further others[x] would be leaked. A + cleanup block was also added for prepend != NULL && outmsg == + NULL. 11+: Fix leak of ast_writestream recording_fs in + app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: + Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ + ........ Merged revisions 427023 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 427024 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427025 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/abstract_jb.c: func_jitterbuffer: fix frame leaks. Fix + code paths where it is possible for frames to leak. Fix + uninitialized variable in jb_get_fixed and jb_get_adaptive. + ASTERISK-22409 #related Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4128/ ........ Merged + revisions 427019 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 427020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-11-02 01:01 +0000 [r426996] Matthew Jordan <mjordan@digium.com> + + * /, res/res_stasis.c: res/res_stasis: Fix crash on module unload + while performing operation When the res_stasis module is + unloaded, it will dispose of the apps_registry container. This is + a problem if an ARI operation is in flight that attempts to use + the registry, as the shutdown occurs in a separate thread. This + patch adds some sanity checks to the various routines that access + the registry which cause the operations to fail if the + apps_registry does not exist. Crash caught by the Asterisk Test + Suite. ........ Merged revisions 426995 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-31 16:50 +0000 [r426934] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * Makefile, /: install init.d files on GNU/kFreeBSD Review: + https://reviewboard.asterisk.org/r/4118/ ........ Merged + revisions 426926 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 426927 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426933 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-31 16:40 +0000 [r426924-426930] Scott Griepentrog <sgriepentrog@digium.com> + + * /, configs/samples/pjsip.conf.sample, res/res_pjsip.c: pjsip: + clarify tls cert and key file usage A question arose as to + whether a .pem file could be provided in place of the .crt and + .key files in a PJSIP TLS configuration. I tested this and + discovered that although a cert will be read from the pem file, a + key will not, and thus the priv_key_file entry is still required. + This update to the fine documentation clarifies the option usage. + AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/ + Reported by: John Bigelow ........ Merged revisions 426928 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c: pjsip: Handle outbound + unregister correctly This updates the status of the outbound + registration to reflect when it has been unregistered. Since the + registration is unregistered but is not stopped, the registration + schedule remains active as before. The patch also updates the + documentation of both the AMI and CLI commands. ASTERISK-24411 + #close Review: https://reviewboard.asterisk.org/r/4119/ Reported + by: John Bigelow patches: unregister-patch1.txt uploaded by John + Bigelow (License 5091) ........ Merged revisions 426923 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-31 03:26 +0000 [r426865] Matthew Jordan <mjordan@digium.com> + + * /, channels/sip/reqresp_parser.c, + channels/sip/include/reqresp_parser.h: + channels/sip/reqresp_parser: Fix unit tests for r426594 When + r426594 was made, it did not take into account a unit test that + verified that the function properly populated the unsupported + buffer. The function would previously memset the buffer if it + detected it had any contents; since this function can now be + called iteratively on successive headers, the unit tests would + now fail. This patch updates the unit tests to reset the buffer + themselves between successive calls, and updates the + documentation of the function to note that this is now required. + ........ Merged revisions 426858 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 426860 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426863 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-31 03:08 +0000 [r426803-426833] Corey Farrell <git@cfware.com> + + * contrib/Makefile (added), Makefile, /: REF_DEBUG: Install + refcounter.py to $(ASTDATADIR)/scripts This change ensures + refcounter.py is installed to a place where it can be found by + the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4094/ ........ Merged + revisions 426830 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 426831 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426832 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_queue.c: app_queue: fix a couple leaks to struct + call_queue in set_member_value set_member_value has a couple + leaks to references in the variable q found through testsuite + tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES + compiler declaration, this is no longer possible with the updated + REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged + revisions 426805 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426806 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/audiohook.c: audiohooks: Clean references to formats Cleanup + references to in_translate[x].format and out_translate[x].format + in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: + Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ + +2014-10-30 21:13 +0000 [r426757-426780] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_exten_state.c, /: res_pjsip_exten_state: + PJSIPShowSubscriptionsInbound causes crash Currently, it is + possible for some subscriptions to get into a NULL state. When + this occurs and the PJSIPShowSubscriptionsInbound ami action is + issued and a device is subscribed for extension state then the + associated subscription state object can't be located. The code + then attempts to dereference a NULL object. Added a NULL check to + avoid the problem. Reported by: John Bigelow ........ Merged + revisions 426779 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/pjsip_options.c, /: res_pjsip: incorrect qualify + statistics after disabling for contact When removing the + qualify_frequency from an AoR or a contact the statistics shown + when issuing "pjsip show aors" from the CLI are incorrect. This + patch deletes the contact's status object from sorcery, + disassociating it from the contact, if the qualify_freqency is + removed from configuration. ASTERISK-24462 #close Reported by: + Mark Michelson Review: https://reviewboard.asterisk.org/r/4116/ + ........ Merged revisions 426755 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-30 09:20 +0000 [r426702] Walter Doekes <walter+asterisk@wjd.nu> + + * apps/app_voicemail.c, /: app_voicemail: Fix unchecked bounds of + myArray in IMAP_STORAGE. In update_messages_by_imapuser(), + messages were appended to a finite array which resulted in a + crash when an IMAP mailbox contained more than 256 entries. This + memory is now dynamically increased as needed. Observe that this + patch adds a bunch of XXX's to questionable code. See the review + (url below) for more information. ASTERISK-24190 #close Reported + by: Nick Adams Tested by: Nick Adams Review: + https://reviewboard.asterisk.org/r/4126/ ........ Merged + revisions 426691 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 426692 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426696 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-30 06:09 +0000 [r426668] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * channels/chan_unistim.c, /: Add additional checks for NULL + pointers to fix several crashes reported. ASTERISK-24304 #close + Reported by: dhanapathy sathya ........ Merged revisions 426666 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + Merged revisions 426667 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-30 01:59 +0000 [r426597-426602] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: channels/chan_sip: Add improved support + for 4xx error codes This patch adds support for 414, 493, 479, + and a stray 400 response in REGISTER response handling. This + helps interoperability in a number of scenarios. Review: + https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch + uploaded by oej (License 5267) ........ Merged revisions 426599 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 426600 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426601 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sip/reqresp_parser.c, /, channels/chan_sip.c: + channels/chan_sip: Support mutltiple Supported and Required + headers A SIP request may contain multiple Supported: and + Required: headers. Currently, chan_sip only parses the first + Supported/Required header it finds. This patch adds support for + multiple Supported/Required headers for INVITE requests. Review: + https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close + Reported by: Olle Johansson patches: rb2478.patch uploaded by oej + (License 5267) ........ Merged revisions 426594 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 426595 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426596 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-29 10:33 +0000 [r426570] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_phone.c: Fix building chan_phone on big endian + systems A left over from the formats conversion (Corey Farrell). + ASTERISK-24458 #close Review: + https://reviewboard.asterisk.org/r/4117/ + +2014-10-28 21:26 +0000 [r426552] Richard Mudgett <rmudgett@digium.com> + + * /, bridges/bridge_builtin_features.c: bridge_builtin_features: + Add missing channel locks around + ast_get_chan_features_general_config(). The feature_automonitor() + and feature_automixmonitor() functions were not locking the + channel around ast_get_chan_features_general_config(). Accessing + the channel datastore list without the channel locked is a good + way to corrupt the list or follow the pointer chain into + oblivion. ........ Merged revisions 426531 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-28 21:05 +0000 [r426525-426529] Corey Farrell <git@cfware.com> + + * /, res/res_fax.c: res_fax: Resolve T38 gateway frame leak. When + frames are translated by a fax gateway they need to be freed. The + existing call to ast_frfree was unreachable. This change + reorganizes fax_gateway_framehook to ensure that ast_frfree is + called when needed. ASTERISK-24457 #close Reported by: Corey + Farrell Review: https://reviewboard.asterisk.org/r/4115/ ........ + Merged revisions 426527 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Unsubscribe from acl_change_sub at + shutdown. ASTERISK-24453 #close Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged + revisions 426524 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-28 18:09 +0000 [r426459] mdavenport <mdavenport@localhost>: + + * configs/samples/manager.conf.sample: ASTERISK-23512, correct + inaccurate comment in manager.conf.sample + +2014-10-28 16:40 +0000 [r426368-426432] Matthew Jordan <mjordan@digium.com> + + * /, main/bridge.c: main/bridge: Destroy features struct on off + nominal path during bridge impart When a channel is imparted to a + bridge, the invocation of the function may provide an + ast_bridge_features struct. Upon passing this to + ast_bridge_impart, the caller must assume that ownership has + passed to the function, as in all paths the function destroys the + struct prior to returning (as its purpose is to configure the + behavior of the channel while in the bridge). On one off nominal + path - where the channel already has a PBX thread - the struct + was not being destroyed. This patch fixes that glitch. + ASTERISK-24437 #close Reported by: Scott Griepentrog ........ + Merged revisions 426431 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: main/manager: Fix typo in AMI event + documentation of "OriginateResponse" The parameter name is + "Response", not "Resonse". ASTERISK-24430 #close Reported by: + Dafi Ni ........ Merged revisions 426366 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426367 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-28 14:56 +0000 [r426294-426362] mdavenport <mdavenport@localhost>: + + * res/res_agi.c: ASTERISK-24323, fix bug in documentation of AGI + STREAM FILE CONTROL + + * configs/samples/extensions.conf.sample: ASTERISK-24419, fix + incorrect syntax for setting language in extensions.conf.sample + +2014-10-28 11:20 +0000 [r426252-426266] Corey Farrell <git@cfware.com> + + * apps/app_queue.c, /: app_queue: Cleanup ao2_iterator Clean + ao2_iterator, resolving reference leak to queue members. + ASTERISK-24454 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4111/ ........ Merged + revisions 426255 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426260 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * funcs/func_cdr.c: func_cdr: Fix CDR_PROP payload leak Remove + duplicate allocation of payload, preventing leak. ASTERISK-24455 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4113/ + +2014-10-27 17:54 +0000 [r426234] Sean Bright <sean@malleable.com> + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: + configure: Add autoconf check for libopus. Because opus + transcoding support cannot be included in the standard Asterisk + distribution, a few codec_opus implementations have popped up. To + make it easier for people to drop in opus support in their own + installations, this patch adds configure checks for libopus. + Review: https://reviewboard.asterisk.org/r/4106/ + +2014-10-27 02:46 +0000 [r426143-426211] Matthew Jordan <mjordan@digium.com> + + * res/res_http_websocket.c, /: res/res_http_websocket: Fix minor + nits found by wdoekes on r409681 When Moises committed the fixes + for WSS (which was a great patch), wdoekes had a few style nits + that were on the review that got missed. This patch resolves what + I *think* were all of the ones that were still on the review. + Thanks to both moy for the patch, and wdoekes for the reviews. + Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged + revisions 426209 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426210 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_phoneprov.c: res/res_phoneprov: Fix crash on shutdown + caused by container cleanup In res_phoneprov, unloading the + module first destroys the http_routes container, followed by the + users. However, users may have a route in the http_routes + container; the validity of this container is not checked in the + users destructor. Hence, we hit an assert as the container has + already been set to NULL. This patch does two things: (1) It adds + a sanity check in the user destructor (because why not) (2) It + switches the order of destruction, so that users are disposed of + prior to the HTTP routes they may hold a reference to. Note that + this crash was caught by the Test Suite (go go testing!) ........ + Merged revisions 426174 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_srtp.c, /: res/res_srtp: Fix include issue for libsrtp + 1.5.0 In libsrtp 1.5.0, crypto_get_random is no longer resolved + simply by including srtp.h. Now, one must include crypto_kernel.h + as well. As it turns out, this header file has been provided by + the library since 2006, so this is a relatively benign change. + ASTERISK-24436 #close Reported by: Patrick Laimbock ........ + Merged revisions 426140 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 426141 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 426142 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-24 15:17 +0000 [r426120] Jonathan Rose <jrose@digium.com> + + * main/manager.c: Documentation: Improve documentation for + ExtensionStatus AMI events Review: + https://reviewboard.asterisk.org/r/4085/ + +2014-10-24 Asterisk Development Team <asteriskteam@digium.com> + + * Asterisk 13.0.0 Released. + +2014-10-22 21:27 +0000 [r426097] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: codec_dahdi: Cannot use struct + ast_translator.core_{src,src}_codec. This fixes a Segmentation + fault introduced in r419044 "media formats: re-architect handling + of media for performance improvements". The problem is that + codec_dahdi was using core_src_codec and core_dst_codec in the + ast_translator structure when these fields were never set. Now + instead of trying to map the new core codec descriptions to the + way DAHDI defines different codecs, we will store the DAHDI + specific formats in 'struct translator' directly so we can refer + to them without mapping. This also allows us to remove the + "global_format_map" structure, since we can now query the list of + translators directly to make sure we do not ever register a DAHDI + based translator for a specific path more than once and eliminate + the need to keep the list and the map in sync. ASTERISK-24435 + #close Reported by: Marian Koniuszko Review: + https://reviewboard.asterisk.org/r/4105/ + +2014-10-21 17:47 +0000 [r426079] Richard Mudgett <rmudgett@digium.com> + + * main/translate.c: translage.c: Fix regression when generating + translation path strings. Fix the AMI Status action read and + write translation path strings from growing for each channel in + the status event list by reseting the ast string given to + ast_translate_path_to_str() to fill in the given translation + path. + +2014-10-20 14:15 +0000 [r425991] Matthew Jordan <mjordan@digium.com> + + * res/res_xmpp.c, main/tcptls.c, /: AST-2014-011: Fix POODLE + security issues There are two aspects to the vulnerability: (1) + res_jabber/res_xmpp use SSLv3 only. This patch updates the module + to use TLSv1+. At this time, it does not refactor + res_jabber/res_xmpp to use the TCP/TLS core, which should be done + as an improvement at a latter date. (2) The TCP/TLS core, when + tlsclientmethod/sslclientmethod is left unspecified, will default + to the OpenSSL SSLv23_method. This method allows for all + encryption methods, including SSLv2/SSLv3. A MITM can exploit + this by forcing a fallback to SSLv3, which leaves the server + vulnerable to POODLE. This patch adds WARNINGS if a user uses + SSLv2/SSLv3 in their configuration, and explicitly disables + SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk + will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly + chosen. For TLS servers, Asterisk will no longer support SSLv2 or + SSLv3. Much thanks to abelbeck for reporting the vulnerability + and providing a patch for the res_jabber/res_xmpp modules. + Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 + #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, + gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by + abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch + uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff + uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded + by mjordan (License 6283) ........ Merged revisions 425987 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-19 17:07 +0000 [r425965] George Joseph <george.joseph@fairview5.com> + + * Makefile, /, configure, include/asterisk/autoconfig.h.in, + configure.ac, makeopts.in: build: Force -fsigned-char on + platforms where the default for char is unsigned gcc on the ARM + platform defaults 'char' to 'unsigned char' whereas Intel and + SPARC default to 'signed char'. This is only an issue in the rare + cases where negative values are assigned to a 'char' but this + this patch insures compatibility by detecting platforms that + default to 'unsigned' and adding an '-fsigned-char' flag to + _ASTCFLAGS. If compiling for ARM (native or cross-compile) be + sure to run ./bootstrap.sh and ./configure to regenerate the + build files. You shouldn't have to do this for Intel or SPARC. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4091/ ........ Merged + revisions 425964 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-19 04:01 +0000 [r425923-425944] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert 425922 + This patch for r425922 introduced a bug, wherein sending an + INVITE request with no SDP would cause Asterisk to not send an + SDP Offer in the 200 OK. The current structure of + res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as + create_outgoing_sdp has no knowledge of whether or not it is + creating an SDP as a new Offer or an Answer. This is something of + an oversight in the callback definition, as the caller of it does + have this information. + + * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Remove left over + reference to override_prefs The usage of the local override_prefs + variable in create_outgoing_sdp_stream was previously to track an + override format preference set by PJSIP_MEDIA_OFFER. Now, + however, that function simply sets the joint capabilities + structure, session->req_caps. During the media format rework, the + override_prefs was instead used to check if there were any + formats in session->req_caps. However, this usage isn't useful in + create_outgoing_sdp_stream. session->req_caps contains the + negotiated formats for *all* streams, not just the current one + being created. Thus, so long as any stream of any type has + provided a format, override_prefs will be non-zero. Hence, its + usage in checking whether or not we should look at the formats on + the endpoint or the joint capabilities is generally useless. + There's only two things useful to check: (1) Does the endpoint + have a format for the media type? (2) Did we negotiate a format + for the media type? If either of those is a 'no', then we must + kill the media stream. + +2014-10-17 22:43 +0000 [r425905] Jonathan Rose <jrose@digium.com> + + * configs/samples/cli_aliases.conf.sample: Sample Configurations: + make 'pjsip reload' reload all reloadable pjsip modules AST-1432 + #close Reported by: John Bigelow + +2014-10-17 13:35 +0000 [r425821-425879] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip.c, + res/res_pjsip_session.c, /: res_pjsip_session/res_pjsip_sdp_rtp: + Be more tolerant of offers When an inbound SDP offer is received, + Asterisk currently makes a few incorrection assumptions: (1) If + the offer contains more than a single audio/video stream, + Asterisk will reject the entire stream with a 488. This is an + overly strict response; generally, Asterisk should accept the + media streams that it can accept and decline the others. (2) If + the offer contains a declined media stream, Asterisk will attempt + to process it anyway. This can result in attempting to match + format capabilities on a declined media stream, leading to a 488. + Asterisk should simply ignore declined media streams. (3) + Asterisk will currently attempt to handle offers with AVPF with + use_avpf=No/AVP with use_avpf=Yes. This mismatch results in + invalid SDP answers being sent in response. If there is a + mismatch between the media type being offered and the + configuration, Asterisk must reject the offer with a 488. This + patch does the following: * Asterisk will accept SDP offers with + at least one media stream that it can use. Some WARNING messages + have been dropped to NOTICEs as a result. * Asterisk will not + accept an offer with a media type that doesn't match its + configuration. * Asterisk will ignore declined media streams + properly. #SIPit31 Review: + https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close + Reported by: James Van Vleet ASTERISK-24381 #close Reported by: + Matt Jordan ........ Merged revisions 425868 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy + setting when sending qualify requests The outboundproxy setting + is currently ignored when sending OPTIONS requests as a result of + the qualify setting. This means that if an Asterisk server is + unable to send the packet directly to a peer, it is unable to + qualify any non-inbound registered peer (e.g. a peer SIP Trunk). + This patch grabs the outboundproxy information for a peer when a + qualify attempt is being constructed and, if it finds the + information, uses it when sending the OPTIONS request. Review: + https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close + Reported by: Damian Ivereigh patches: outboundproxy-dai.patch + uploaded by Damian Ivereigh (License 6632) ........ Merged + revisions 425818 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425819 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425820 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-17 02:41 +0000 [r425783] Richard Mudgett <rmudgett@digium.com> + + * main/core_unreal.c, main/channel.c, /: AMI: Add missing VarSet + events when a channel inherits variables. There should be AMI + VarSet events when channel variables are inherited by an outgoing + channel. Also local;2 should generate VarSet events when it gets + all of its channel variables from channel local;1. ASTERISK-24415 + #close Reported by: Richard Mudgett Patches: + jira_asterisk_24415_v12.patch (license #5621) patch uploaded by + Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/ + ........ Merged revisions 425782 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-17 01:57 +0000 [r425736-425761] Matthew Jordan <mjordan@digium.com> + + * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio + issues when moving from remote bridge to softmix When a native + RTP bridge that is remotely bridging its participants switches to + a softmix bridge, it may not properly re-INVITE the media for one + or both participants back to Asterisk. This is due to the current + bridge_native_rtp code only re-INVITEs if it believes the channel + will survive the bridge operation. Currently, that code is + failing, as it expects the channels to have a soft hangup flag + set on it indicating that a redirect has occurred or that the + channel is going to leave the bridge. (The code did not take into + account a smart bridge operation). This patch also renames a few + things to be more reflective of the underlying types. Review: + https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close + ........ Merged revisions 425760 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_cel.c: test_cel: Update pickup test to expect + CANCEL instead of ANSWSER The CEL pickup test previously looked + for a disposition of ANSWER between the original caller/peer when + the call is picked up. This is actually incorrect: the + disposition should, at the very least, not be ANSWER as the call + was never ANSWERed. The disposition is now CANCEL; this patch + updates the test accordingly. ........ Merged revisions 425757 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: main/cdr: Use 'time' when rescheduling batched + CDRs as opposed to 'size' When refactoring CDRs to use the + configuration framework, a 'whoops' was introduced where the CDR + batch size was used when rescheduling a batch, as opposed to the + time duration. This patch corrects that obvious mistake. + ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged + revisions 425735 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 17:30 +0000 [r425714] George Joseph <george.joseph@fairview5.com> + + * include/asterisk/config.h, tests/test_config.c, main/config.c, /: + config: Fix inf loop using ast_category_browse and + ast_variable_retrieve Fix infinite loop when calling + ast_variable_retrieve inside an ast_category_browse loop when + there is more than 1 category with the same name. Tested-by: + George Joseph Review: https://reviewboard.asterisk.org/r/4089/ + ........ Merged revisions 425713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 14:35 +0000 [r425691] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip_t38.c, res/res_pjsip_registrar_expire.c, + res/res_pjsip_mwi_body_generator.c, + res/res_pjsip_endpoint_identifier_user.c, + res/res_pjsip_send_to_voicemail.c, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip_endpoint_identifier_anonymous.c, + res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c, + res/res_pjsip_acl.c, res/res_pjsip_pubsub.c, + res/res_pjsip_diversion.c, res/res_pjsip_refer.c, + include/asterisk/res_pjsip.h, + res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c, + res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c, + res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c, + res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c, + res/res_pjsip_logger.c, res/res_pjsip_nat.c, + res/res_pjsip_session.c, res/res_pjsip_exten_state.c, + res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c, + res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c, + res/res_pjsip_dialog_info_body_generator.c, + res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_registrar.c, + channels/chan_pjsip.c, res/res_pjsip_transport_websocket.c, + res/res_pjsip_pidf_eyebeam_body_supplement.c, + include/asterisk/res_pjsip_session.h, /, res/res_pjsip_notify.c, + res/res_pjsip_pidf_digium_body_supplement.c, + res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip_publish_asterisk.c: PJSIP: Enforce module load + dependencies This enforces that res_pjsip, res_pjsip_session, and + res_pjsip_pubsub have loaded properly before attempting to load + any modules that depend on them since the module loader system is + not currently capable of resolving module dependencies on its + own. ASTERISK-24312 #close Reported by: Dafi Ni Review: + https://reviewboard.asterisk.org/r/4062/ ........ Merged + revisions 425690 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 06:11 +0000 [r425669] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * channels/chan_unistim.c, /: Fix loss of voice after second call + drops (on a second line) in case using multiple lines on unistim + phones. There is regression was introduced in r391379. Reported + by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........ + Merged revisions 425667 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425668 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 01:25 +0000 [r425646] Joshua Colp <jcolp@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix a bug where ICE + state would get reset when it shouldn't. In the case where the + ICE negotiation had not yet started current state would get wiped + when it shouldn't. This also removes channel binding as in + practice this does not work well with other implementations. + ........ Merged revisions 425644 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425645 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-15 19:31 +0000 [r425627] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_motif.c: chan_motif: Cleanup + jingle_tech.capabilities only once. + +2014-10-15 19:05 +0000 [r425611] Jonathan Rose <jrose@digium.com> + + * res/parking/parking_tests.c: parking_tests: Fix assertions and + possibly crashes in res_parking unit tests Assertions were caused + by attempting to play music on hold to a channel with no formats. + Parking unit test channels were given formats and a technology so + that they would be able to pretend to read/write frames. + ASTERISK-24413 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4075/ + +2014-10-15 09:59 +0000 [r425590] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c, /: chan_ooh323: fix rtptimeout general + value checking correct condition to check rtptimeout in [general] + config section ASTERISK-24393 #close Reported by: Dmitry Melekhov + Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........ + Merged revisions 425547 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425548 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425589 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 20:46 +0000 [r425526] George Joseph <george.joseph@fairview5.com> + + * /, include/asterisk/config.h, tests/test_config.c, main/config.c: + config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG + the /main/config config_basic_ops test was causing a SEGV while + doing an ast_category_delete in an ast_category_browse loop. + Apparently this never worked but was also never tested. I removed + the test, added 2 notes to config.h indicating that it's not + supported and added a few lines of code to ast_category_delete to + prevent the SEGV should someone attempt it in the future. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4078/ ........ Merged + revisions 425525 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 19:00 +0000 [r425504] Jonathan Rose <jrose@digium.com> + + * main/sched.c, /: Scheduler: Fix a nasty scheduler caching bug + which makes new tasks not execute Tasks that were marked for + pending deletion in the scheduler would be moved to the cache for + later reuse, but after being recycled the deleted mark wouldn't + be removed resulting in fresh tasks being deleted without + reason... and immediately moved back into the cache where they + could be reused again. This could cause horrendous things to + happen in just about anything that used a scheduler. + ASTERISK-24321 #close Reported by: Steve Pitts Review: + https://reviewboard.asterisk.org/r/4071/ ........ Merged + revisions 425503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 18:12 +0000 [r425481] George Joseph <george.joseph@fairview5.com> + + * res/res_phoneprov.c, include/asterisk/phoneprov.h, /, + res/res_pjsip_phoneprov_provider.c: res_phoneprov: Create + accessor for ast_phoneprov_std_variable_lookup Based on feedback + from Richard, I created an accessor for + res_phoneprov/ast_phoneprov_std_variable_lookup and added load + priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by: + Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/ + ........ Merged revisions 425480 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 16:46 +0000 [r425459] Corey Farrell <git@cfware.com> + + * /, res/res_fax.c: res_fax: Fix reference leak caused by gateway + sessions Fax gateway session objects can be re-used, causing the + same gateway session to be added to faxregistry.container more + than once. This change causes fax_session_new to remove the + reserved session from the container before it's id is changed, + ensuring it's possible for the session to be freed. + ASTERISK-24392 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4049/ ........ Merged + revisions 425457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 16:35 +0000 [r425455] Richard Mudgett <rmudgett@digium.com> + + * /, main/stasis_channels.c: stasis_channels.c: Resolve unfinished + Dials when doing masquerades (Part 2) Masquerades into and out of + channels that are involved in a dial operation don't create the + expected dial end event. The missing dial end event goes against + the model for things like CDRs and generating Dial end manager + actions and such. There are four cases: 1) A channel masquerades + into the caller channel. The case happens when performing a + blonde transfer using the channel driver's protocol. 2) A channel + masquerades into a callee channel. The case happens when + performing a directed call pickup. 3) The caller channel + masquerades out of dial. The case happens when using the Bridge + application on the caller channel. 4) A callee channel + masquerades out of dial. The case happens when using the Bridge + application on a peer channel. As it turned out, all four cases + need to be handled instead of just the first one. ASTERISK-24237 + Reported by: Richard Mudgett ASTERISK-24394 #close Reported by: + Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/ + ........ Merged revisions 425430 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 16:19 +0000 [r425415] Corey Farrell <git@cfware.com> + + * /, res/res_fax.c: res_fax: Resolve module reference leak caused + by reserved sessions Remove reference to module providing + reserved session after adding a reference to the final module. + This re-reference is done to ensure that module references are + correct even if the final session selects a different module than + the reserved session. ASTERISK-18923 #close Reported by: Grigoriy + Puzankin Review: https://reviewboard.asterisk.org/r/4048/ + ........ Merged revisions 425405 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425407 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425411 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-13 16:10 +0000 [r425384] George Joseph <george.joseph@fairview5.com> + + * apps/app_directory.c, tests/test_sorcery.c, main/config.c, + tests/test_sorcery_realtime.c, res/res_sorcery_realtime.c, + apps/app_voicemail.c, res/res_sorcery_config.c, main/manager.c, + /, include/asterisk/config.h, pbx/pbx_realtime.c, + tests/test_config.c: manager/config: Support templates and + non-unique category names via AMI This patch provides the + capability to manipulate templates and categories with non-unique + names via AMI. Summary of changes: GetConfig and GetConfigJSON: + Added "Filter" parameter: A comma separated list of + name_regex=value_regex expressions which will cause only + categories whose variables match all expressions to be + considered. The special variable name TEMPLATES can be used to + control whether templates are included. Passing 'include' as the + value will include templates along with normal categories. + Passing 'restrict' as the value will restrict the operation to + ONLY templates. Not specifying a TEMPLATES expression results in + the current default behavior which is to not include templates. + UpdateConfig: NewCat now includes options for allowing duplicate + category names, indicating if the category should be created as a + template, and specifying templates the category should inherit + from. The rest of the actions now accept a filter string as + defined above. If there are non-unique category names, you can + now update specific ones based on variable values. To facilitate + the new capabilities in manager, corresponding changes had to be + made to config, most notably the addition of filter criteria to + many of the APIs. In some cases it was easy to change the + references to use the new prototype but others would have + required touching too many files for this patch so a wrapper with + the original prototype was created. Macros couldn't be used in + this case because it would break binary compatibility with + modules such as res_digium_phone that are linked to real symbols. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4033/ ........ Merged + revisions 425383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-12 21:09 +0000 [r425362] Joshua Colp <jcolp@digium.com> + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Make the ICE + transport check case insensitive as some implementations use + 'udp'. ........ Merged revisions 425360 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425361 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-12 08:15 +0000 [r425289-425299] Walter Doekes <walter+asterisk@wjd.nu> + + * /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send + reINVITE after a BYE. After a reINVITE glare situation, Asterisk + would re-send the reINVITE even though the call had been hung up + in the mean time. This patch unschedules the reinvite when + handling the BYE. ASTERISK-22791 #close Reported by: Paolo + Compagnini Tested by: Paolo Compagnini Review: + https://reviewboard.asterisk.org/r/4056/ (testcase is in review + r4055) ........ Merged revisions 425296 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425297 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425298 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, Makefile: build: Relax badshell tilde test to allow for ~ in + middle of DESTDIR. The main Makefile has a target test called + 'badshell' that tests if DESTDIR does not happen to have an + an-expanded tilde (~). This might be the case if you run: make + install DESTDIR=~/somewhere/ That test also disallowed valid + tildes in directory names. The test is now changed to only + trigger on a tilde at the start of the path. ASTERISK-13797 + #close Reported by: Tzafrir Cohen Review: + https://reviewboard.asterisk.org/r/4064/ ........ Merged + revisions 425291 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425292 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425293 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_calendar_ews.c: res_calendar_ews: Relax neon version + check to work with 0.30 too. Allow res_calendar_ews to work not + only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close + Reported by: Tzafrir Cohen Review: + https://reviewboard.asterisk.org/r/4068/ ........ Merged + revisions 425286 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425287 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425288 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-11 21:08 +0000 [r425265] George Joseph <george.joseph@fairview5.com> + + * /, res/res_phoneprov.c: res_phoneprov: Cleanup module load error + handling Tested module load/reload interaction between + res_phoneprov and res_pjsip_phoneprov_provider in cases where + res_phoneprov didn't load correctly (usually misconfiguration or + missing phoneprov.conf) Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4069/ ........ Merged + revisions 425264 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 20:48 +0000 [r425243] Joshua Colp <jcolp@digium.com> + + * /, main/bridge.c, bridges/bridge_native_rtp.c: bridge: During a + smart bridge operation provide a more complete bridge to the old + technology. When a smart bridge operation occurs and a bridge + transitions from one technology to another the old technology is + provided the channels formerly in it and told that they are + leaving. Unfortunately the bridge provided along with them is + incomplete. The bridge, despite there being channels in it, + contains none. This forces technology implementations to have + additional logic when channels are leaving or to store their own + duplicated state. This change makes the bridge more complete so + it contains the expected channels. Now that the bridge is + complete special logic within bridge_native_rtp is no longer + needed and has been removed. Review: + https://reviewboard.asterisk.org/r/4057/ ........ Merged + revisions 425242 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 14:31 +0000 [r425221] Matthew Jordan <mjordan@digium.com> + + * /, res/res_phoneprov.c: res/res_phoneprov: Bail on registration + if res_phoneprov didn't load If res_phoneprov failed to fully + load (due to not being configured), the providers container will + be NULL. If a module attempts to register a phone provisioning + provider, it should check for the presence of the container. If + there is no providers container, it should return an error. This + patch makes the ast_phoneprov_provider_register function do + that... otherwise this would be a silly commit message. ........ + Merged revisions 425220 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 14:23 +0000 [r425217] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_phoneprov_provider.c: + res_pjsip_phoneprov_provider: Add missing dependency on + pjproject. ........ Merged revisions 425216 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 13:01 +0000 [r425155] Kinsey Moore <kmoore@digium.com> + + * /, tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing + regression This fixes a regression in callerid parsing introduced + when another bug was fixed. This bug occurred when the name was + composed entirely of DTMF keys and quoted without a number + section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard + Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by + Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ + ........ Merged revisions 425152 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425153 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425154 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 12:10 +0000 [r425132] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_nat.c, /: res_pjsip_nat: Place source port into + rport of responses if 'force_rport' is on. When the 'force_rport' + option is enabled the behavior should be the same as if the + remote side placed rport into the message themselves. Therefore + any responses we send should include the source port of the + request in the rport of the Via header. #SIPit31 ASTERISK-24387 + #close Reported by: Matt Jordan ........ Merged revisions 425131 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 07:32 +0000 [r425071] Walter Doekes <walter+asterisk@wjd.nu> + + * /, channels/chan_sip.c: chan_sip: Fix dialog leak resulting from + missing ACK to re-INVITE. If a device re-INVITEs at the same time + as the dialog is hung up, and if then the ACK to the re-INVITE + never reaches Asterisk, chan_sip would fail to destroy the dialog + after a while. This resulted in (most prominently) file handle + leaks. (Patch reindented by me.) ASTERISK-20784 #close + ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal + Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle + (License #5334) patch_asterisk_20784.txt uploaded by Nitesh + Bansal (License #6418) Reviewboard: + https://reviewboard.asterisk.org/r/4052/ (testcase can be found + at r4051) ........ Merged revisions 425068 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425069 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425070 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 23:35 +0000 [r425052] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip_phoneprov_provider.c: res_pjsip_phoneprov_provider: + fix compile breakage on AST_VECTOR endpoint->inbound_auths was + changed to a vector in 13 and I committed the 12 patch instead of + the 13 patch. Tested-by: George Joseph + +2014-10-09 21:38 +0000 [r425031] Kevin Harwell <kharwell@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Crash if no + candidates received for component When starting ice if there is + not at least one remote ice candidate with an RTP component + asterisk will crash. This is due to an assertion in pjnath as it + expects at least one candidate with an RTP component. Added a + check to make sure at least one candidate contains an RTP + component and at least one candidate has an RTCP component. + ASTERISK-24383 #close Review: + https://reviewboard.asterisk.org/r/4039/ ........ Merged + revisions 425030 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 20:54 +0000 [r425008] George Joseph <george.joseph@fairview5.com> + + * /, res/res_pjsip_phoneprov_provider.c (added), + configs/samples/pjsip.conf.sample: res_pjsip_phoneprov_provider: + Provides pjsip integration with res_phoneprov This module allows + res_pjsip to integrate with res_phoneprov. It handles the pjsip + 'phoneprov' object type. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/3976/ ........ Merged + revisions 425007 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 18:37 +0000 [r424986] Matthew Jordan <mjordan@digium.com> + + * /, res/res_phoneprov.c: res/res_phoneprov: Don't cancel Asterisk + load on module load failure ........ Merged revisions 424985 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 17:45 +0000 [r424964] George Joseph <george.joseph@fairview5.com> + + * include/asterisk/phoneprov.h (added), /, + configs/samples/phoneprov.conf.sample, + include/asterisk/chanvars.h, res/res_phoneprov.c, + res/res_phoneprov.exports.in (added), main/chanvars.c: + res_phoneprov: Refactor phoneprov to allow pluggable config + providers This patch makes res_phoneprov more modular so other + modules (like pjsip) can provide configuration information + instead of res_phoneprov relying solely on users.conf and + sip.conf. To accomplish this a new ast_phoneprov public API is + now exposed which allows config providers to register themselves, + set defaults (server profile, etc) and add user extensions. * + ast_phoneprov_provider_register registers the provider and + provides callbacks for loading default settings and loading + users. * ast_phoneprov_provider_unregister clears the defaults + and users. * ast_phoneprov_add_extension should be called once + for each user/extension by the provider's load_users callback to + add them. * ast_phoneprov_delete_extension deletes one extension. + * ast_phoneprov_delete_extensions deletes all extensions for the + provider. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/3970/ ........ Merged + revisions 424963 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 16:36 +0000 [r424942] Richard Mudgett <rmudgett@digium.com> + + * /, main/cdr.c: cdr.c: Make turning on CDR debug a one step + process instead of two. Now "cdr set debug on" doesn't also + require "core set verbose 1" to see CDR debug output. ........ + Merged revisions 424941 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 08:08 +0000 [r424880] Walter Doekes <walter+asterisk@wjd.nu> + + * /, contrib/scripts/safe_asterisk: safe_asterisk: Don't + automatically exceed MAXFILES value of 2^20. On systems with lots + of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can + exceed the per-process file limit of 2^20. This patch ensures the + value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close + Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff + uploaded by Michael Myles (License #6626) ........ Merged + revisions 424875 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 424878 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424879 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-08 18:46 +0000 [r424854] Joshua Colp <jcolp@digium.com> + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Allow only UDP ICE + candidates. The underlying library, pjnath, that res_rtp_asterisk + uses for ICE support does not have support for ICE-TCP. As + candidates are passed through directly to it this can cause error + messages to occur when it receives something unexpected (such as + a TCP candidate). This change merely ignores all non-UDP + candidates so they never reach pjnath. ASTERISK-24326 #close + Reported by: Joshua Colp ........ Merged revisions 424852 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424853 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-08 18:24 +0000 [r424769-424850] Kinsey Moore <kmoore@digium.com> + + * main/stasis.c: Stasis: Relegate log message to dev-mode This + error message primarily applies to development tasks and will now + only show up when dev-mode is enabled via configure. + + * main/sounds_index.c: Indexer: Format message types may not exist + In Asterisk 13+, any given message type is not guaranteed to + exist even if Asterisk comes up correctly since creation of the + message type could be declined. The indexer should not prevent + Asterisk from starting under these conditions. + + * main/stasis.c: Stasis: Only log errors for non-declined types + When message type creation is declined via stasis.conf, certain + operations log errors assuming that the declined type is being + used before initialization or after destruction. These error + messages get quite spammy for oft used message types and should + not be logged in the first place since the message type is + validly NULL. Reported by: Matt DiMeo + +2014-10-07 18:33 +0000 [r424752] Joshua Colp <jcolp@digium.com> + + * main/data.c: data: Properly access formats in capabilities + structure when adding codecs. Formats within a capabilities + structure are addressed starting at 0, not 1. Assuming 1 causes + it to exceed an array. ASTERISK-24389 #close Reported by: Kevin + Harwell + +2014-10-07 17:41 +0000 [r424692-424731] Matthew Jordan <mjordan@digium.com> + + * /, res/res_pjsip_outbound_registration.c: + res/res_pjsip_outbound_registration: Initialize + auth_reject_permanent parameter Prior to this patch, the + auth_reject_permanent parameter was not initialized on the + registration client state, leading to the parameter being + disabled regardless of the value specified in pjsip.conf. This + patch initialized the setting on the registration client state to + the provided configuration value. ASTERISK-24398 #close ........ + Merged revisions 424730 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Fix typo in WARNING + message + + * main/message.c, /: message: Don't close an AMI connection on + SendMessage action error If SendMessage encounters an error (such + as incorrect input provided to the action), it will currently + return -1. Actions should only return -1 if the connection to the + AMI client should be closed. In this case, SendMessage causing + the client to disconnect is inappropriate. This patch causes the + action to return 0, which simply causes the action to fail. + Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 + #close Reported by: Peter Katzmann patches: sendMessage.patch + uploaded by Peter Katzmann (License 5968) ........ Merged + revisions 424690 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424691 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-06 15:38 +0000 [r424669] Richard Mudgett <rmudgett@digium.com> + + * main/features.c, /: features.c: Fix lingering channel ref while + Bridge() application is active. Using the Bridge application to + bridge a channel that is executing an applicaiton such as Wait + results in a lingering Surrogate channel in the CLI "core show + channels" output even though it has already hungup. * Fix + bridge_exec() to not hold onto the current_dest_chan ref once it + has been put into the bridge. * Eliminated bridge_exec()'s use of + RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged + revisions 424668 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-06 12:38 +0000 [r424601-424647] Matthew Jordan <mjordan@digium.com> + + * /, main/sdp_srtp.c: sdp_srtp: Add new lines to some WARNING + messages ........ Merged revisions 424646 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_options.c: res_pjsip/pjsip_options: Do not + 404 an OPTIONS request not sent to an endpoint An OPTIONS request + that is sent to Asterisk but not to a specific endpoint is + currently sent a 404 in response. This is because, not + surprisingly, an empty extension is never going to be found in + the dialplan. This patch makes it so that we only attempt to look + up the endpoint in the dialplan if it is specified in the OPTIONS + request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt + Jordan ........ Merged revisions 424624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions: + Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels Calling + PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your + health. It will treat the channels as a PJSIP channel, eventually + hitting an ao2 error, FRACKing on assertion error, and quite + likely crashing. This patch adds checks to the read/write + callbacks that ensure that the channel technology is of type + 'PJSIP' before attempting to operate on the channel. #SIPit31 + ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged + revisions 424621 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_hep_pjsip.c, res/res_pjsip/pjsip_distributor.c, + res/res_pjsip_logger.c: res_pjsip: Prevent crashes when PJPROJECT + presents an rdata with no message When a message that exceeds the + PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible + (although it shouldn't occur) for pjproject to pass up an rdata + object with a NULL msg in the msg_info. Needless to say, things + that attempt to dereference this are in for a rough ride. In + particular, this caused crashes in three different locations, all + of which are 'low level' enough to intercept an rdata object + early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) + res_pjsip/distributor Anything that can intercept an rdata object + before res_pjsip/distributor should be defensive when looking at + the received packet. #SIPit31 ASTERISK-24369 #close Reported by: + Matt Jordan ........ Merged revisions 424618 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Gracefully handle + errors when re-creating subscriptions A subscription that has + been persisted can - for various reasons - fail to be re-created + on startup. This patch resolves a number of crashes that occurred + when a subscription cannot be re-created on several off-nominal + paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan + +2014-10-05 00:48 +0000 [r424552-424580] Corey Farrell <git@cfware.com> + + * main/manager.c, /: Release AMI connections on shutdown. + ASTERISK-24378 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4037/ ........ Merged + revisions 424578 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_motif.c: chan_motif: Correct last commit to use + ao2_cleanup to free format cap This fix applies to 13 and trunk. + ASTERISK-24384 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4043/ + + * /, channels/chan_motif.c: chan_motif: Release format capabilities + and config on module load error ASTERISK-24384 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4043/ ........ Merged + revisions 424550 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424551 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 21:56 +0000 [r424472-424529] Richard Mudgett <rmudgett@digium.com> + + * /, CHANGES, res/res_pjsip.c: res_pjsip: Fix XML typo and update + CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, /, + main/framehook.c: audiohooks: Reevaluate the bridge technology + when an audiohook is added or removed. Adding a mixmonitor to a + channel causes the bridge to change technologies from native to + simple_bridge so the call can be recorded. However, when the + mixmonitor is stopped the bridge does not switch back to the + native technology. * Added unbridge requests to reevaluate the + bridge when a channel audiohook is removed. * Moved the unbridge + request into ast_audiohook_attach() ensure that the bridge + reevaluates whenever an audiohook is attached. This simplified + the mixmonitor and chan_spy start code as well. * Added defensive + code to stop_mixmonitor_full() in case additional arguments are + ever added to the StopMixMonitor application. * Made + ast_framehook_detach() not do an unbridge request if the + framehook does not exist. * Made ast_framehook_list_fixup() do an + unbridge request if there are any framehooks. Also simplified the + loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/4046/ ........ Merged + revisions 424506 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/core_unreal.c, main/taskprocessor.c, channels/chan_iax2.c, + res/res_pjsip_session.c, main/channel.c, channels/chan_misdn.c, + channels/chan_skinny.c, funcs/func_frame_trace.c, + channels/chan_motif.c, include/asterisk/frame.h, + main/bridge_channel.c, channels/chan_pjsip.c, + channels/chan_unistim.c, include/asterisk/res_pjsip_session.h, + addons/chan_ooh323.c, /, include/asterisk/taskprocessor.h, + channels/chan_sip.c, res/res_pjsip_session.exports.in: + chan_pjsip: Fix deadlock when masquerading PJSIP channels. + Performing a directed call pickup resulted in a deadlock when + PJSIP channels were involved. A masquerade needs to hold onto the + channel locks while it swaps channel information between the two + channels involved in the masquerade. With PJSIP channels, the + fixup routine needed to push a fixup task onto the PJSIP + channel's serializer. Unfortunately, if the serializer was also + processing a task that needed to lock the channel, you get + deadlock. * Added a new control frame that is used to notify the + channels that a masquerade is about to start and when it has + completed. * Added the ability to query taskprocessors if the + current thread is the taskprocessor thread. * Added the ability + to suspend/unsuspend the PJSIP serializer thread so a masquerade + could fixup the PJSIP channel without using the serializer. + ASTERISK-24356 #close Reported by: rmudgett Review: + https://reviewboard.asterisk.org/r/4034/ ........ Merged + revisions 424471 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 15:54 +0000 [r424448] George Joseph <george.joseph@fairview5.com> + + * /, main/sorcery.c: sorcery: Prevent SEGV in sorcery_wizard_create + when there's no create function When you call + ast_sorcery_create() you don't necessarily know which wizard is + going to be invoked. If it happens to be a wizard like 'config' + that doesn't have a 'create' virtual function you get a segfault + in the sorcery_wizard_create callback. This patch catches the + null function pointer, does an ast_assert, and logs an error. + Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged + revisions 424447 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 13:58 +0000 [r424424-424427] Kinsey Moore <kmoore@digium.com> + + * configs/samples/pjsip.conf.sample, /, + res/res_pjsip/pjsip_configuration.c: PJSIP: Restore functional + default for callerid_privacy The pjsip config option default + fixups from r424263 altered the functional default from + "allowed_not_screened" to "allowed". This change restores the + functional default value when none is provided. ........ Merged + revisions 424426 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: Manager: Add missing fields and documentation + for CoreShowChannels This corrects some issues introduced in the + responses to the CoreShowChannels AMI command as well as adding + documentation for the responses. The command in Asterisk 12 was + missing the following fields: Duration, Application, + ApplicationData, and BridgedChannel and BridgedUniqueID (replaced + with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn + Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged + revisions 424423 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 07:54 +0000 [r424415] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, /: res_pjsip_session: Reduce SDP size by + removing duplicate connection lines. Due to the architecture of + how media streams are handled each individual handler adds + connection details (IP address) for it. The first media stream is + then used as the top level SDP connection line. In practice each + line ends up being the same so to reduce the SDP size + stream-level connection information is also added to the SDP if + it differs from the top level SDP connection line. ........ + Merged revisions 424414 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-02 21:52 +0000 [r424394] Richard Mudgett <rmudgett@digium.com> + + * /, configs/samples/pjsip.conf.sample, res/res_pjsip.c, + res/res_pjsip/config_transport.c: res_pjsip: Make transport + cipher option accept a comma separated list of cipher names. + Improvements to the res_pjsip transport cipher option. * Made the + cipher option accept a comma separated list of OpenSSL cipher + names. Users of realtime will be glad if they have more than one + name to list. * Added the CLI command 'pjsip list ciphers' so a + user can know what OpenSSL names are available for the cipher + option. * Updated the cipher option online XML documentation to + specify what is expected for the value. * Updated + pjsip.conf.sample to not indicate that ALL is acceptable since + ALL does not imply a preference order for the ciphers and PJSIP + does not simply pass the string to OpenSSL for interpretation. + ASTERISK-24199 #close Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/4018/ ........ Merged + revisions 424393 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-02 20:15 +0000 [r424373] Jonathan Rose <jrose@digium.com> + + * /, + contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py + (added): Alembic: Add enumerator value to sippeers -> directmedia + - 'outgoing' The 'outgoing' value was left off of the enumerator + when first creating the column. This patch adds it, and should + gracefully upgrade keeping the existing data in tact. + ASTERISK-23781 #close Reported by: Stephen More Review: + https://reviewboard.asterisk.org/r/4013/ ........ Merged + revisions 424372 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-02 13:35 +0000 [r424338] Scott Griepentrog <sgriepentrog@digium.com> + + * /, configs/samples/pjsip.conf.sample: res_pjsip: document use of + rewrite_contact in sample conf Without setting rewrite_contact, + an invite to an endpoint behind NAT will not reach it - unless + the endpoint itself uses STUN or TURN to discover it's public + URI. Thus, the use of this should be in the sample documentation. + Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged + revisions 424337 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 22:52 +0000 [r424333] Jonathan Rose <jrose@digium.com> + + * channels/chan_pjsip.c: chan_pjsip: Fix an assertion for channels + that lack formats on creation ASTERISK-24222 #close Reported by: + Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/ + +2014-10-01 20:36 +0000 [r424313] Corey Farrell <git@cfware.com> + + * res/res_hep.c, /: res_hep: Release allocation reference to + configuration. ASTERISK-24362 #close Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged + revisions 424312 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 16:37 +0000 [r424288-424291] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/pjsip_configuration.c, + configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip: + Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. + During the latest update to DTLS-SRTP support the ability to + configure the hash used for fingerprints was added. This gave us + two supported ones: SHA-1 and SHA-256. The default was + accordingly updated to SHA-256. Unfortunately this configuration + ability was not exposed within res_pjsip. This change adds a + dtls_fingerprint option that controls it. #SIPit31 ........ + Merged revisions 424290 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Accept DTLS + attributes in top level, not just media session. #SIPit31 + ........ Merged revisions 424287 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 12:27 +0000 [r424245-424266] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip/config_transport.c, /, res/res_pjsip/location.c, + res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip/pjsip_configuration.c, + configs/samples/pjsip.conf.sample: PJSIP: Handle defaults + properly This updates the code behind PJSIP configuration options + with custom handlers to deal with the assigned default values + properly where it makes sense and adjusting the default value + where it doesn't. Before applying this patch, there were several + cases where the default value for an option would prevent that + config section from loading properly. Reported by: Thomas + Thompson Review: https://reviewboard.asterisk.org/r/4019/ + ........ Merged revisions 424263 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_nat.c: PJSIP: Force transport on contact rewrite + If contact rewriting is enabled but the contact differs in + transport from what is actually being used, messages after the + initial INVITE transaction can be sent to an incorrect + transport/port combination. In the case where this bug occurred + the remote party never received a BYE since it was sent to the + remote party's TCP port over UDP. Review: + https://reviewboard.asterisk.org/r/4032/ ........ Merged + revisions 424244 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 10:09 +0000 [r424179-424184] Walter Doekes <walter+asterisk@wjd.nu> + + * /, channels/chan_sip.c: chan_sip: Simplify some unref code by + removing unlink_peer_from_tables. ASTERISK-22945 #related + Reported by: ibercom Patches: + asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License + #6599) ........ Merged revisions 424181 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 424182 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424183 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Remove excess ref of realtime + peer before sip_poke_peer. The peer is referenced at the end of + sip_poke_peer, it should not get an extra ref before the call to + sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close + Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: + asterisk11.patch uploaded by ibercom (License #6599) Review: + https://reviewboard.asterisk.org/r/4031/ ........ Merged + revisions 424176 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 424177 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424178 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-30 11:40 +0000 [r424153-424156] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't place an + extra whitespace before 'rport' and don't put IPv6 addresses in + brackets. #SIPit31 ........ Merged revisions 424155 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the base + and mapped address for candidates is present in SDP. This change + fixes an issue where ICE candidates put into the SDP did not + contain the 'raddr' and 'rport' information for server reflexive + and relay candidates. #SIPit31 ........ Merged revisions 424151 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + Merged revisions 424152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-29 21:59 +0000 [r424129] George Joseph <george.joseph@fairview5.com> + + * /, res/res_pjsip/pjsip_cli.c: pjsip_cli: Suppress header print on + error or no objects If there's an error on the pjsip command line + or there are no objects, don't print the column headers. + ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George + Joseph Tested-by: Brad Latus Review: + https://reviewboard.asterisk.org/r/4025/ ........ Merged + revisions 424128 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-29 21:26 +0000 [r424126] Walter Doekes <walter+asterisk@wjd.nu> + + * /, contrib/scripts/autosupport: autosupport: Fix bashism. '==' is + bashism (bashspecific, fails when dash is /bin/sh). Anyway, a + 'case' works better there. Originally committed in r375059 and + r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported + by: Tzafrir Cohen ........ Merged revisions 424117 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424125 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-29 21:17 +0000 [r424097-424105] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c, + /, res/res_pjsip_authenticator_digest.c: Simplify UUID generation + in several places. Replace code using ast_uuid_generate() with + simpler and faster code using ast_uuid_generate_str(). The new + code avoids a malloc(), free(), and copy. ........ Merged + revisions 424103 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/threadpool.c: threadpool.c: Minor cleanup fixes. * Fix + threadpool_alloc() prototype. * Add missing off-nominal NULL + check of pool in threadpool_alloc(). * searializer_create() does + not need to create the object with a lock as the lock is not + used. ........ Merged revisions 424096 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-27 12:43 +0000 [r424057] Joshua Colp <jcolp@digium.com> + + * channels/chan_pjsip.c, res/res_pjsip_session.c, /: + res_pjsip_session: Add additional checks for delaying session + refreshes. There are certain situations which no checks existed + for which need to prevent session refreshes. This includes + sending a session refresh with SDP before SDP negotiation has + completed and sending a session refresh before the dialog itself + has been established. Checks for these have been added. + Additionally COLP related UPDATEs were including SDP when it is + not needed. Review: https://reviewboard.asterisk.org/r/4008/ + ........ Merged revisions 424056 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-26 15:21 +0000 [r423992] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_fax.c: res_fax: Fix out of bounds error in + update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy + Laine Patches: res_fax_bounds.patch (license #6561) patch + uploaded by Jeremy Laine Modified patch to not use magic numbers. + ........ Merged revisions 423979 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423983 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423987 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-26 08:25 +0000 [r423918] Walter Doekes <walter+asterisk@wjd.nu> + + * /, doc/asterisk.8: docs: Escape unescaped minus sign in + asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy + Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy + Lainé (License #6561) ........ Merged revisions 423915 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423916 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423917 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-25 21:01 +0000 [r423895] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip.c, /: res_pjsip.c: Add missing off nominal cleanup + in ast_sip_push_task_synchronous(). * Made memset the std struct + in ast_sip_push_task_synchronous() because if DEBUG_THREADS is + enabled then uninitialized lock tracking data is used. ........ + Merged revisions 423894 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-24 18:32 +0000 [r423867] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c: + pjsip_options.c: Fix race condition stopping periodic out of + dialog OPTIONS request. The crash on the issues is a result of an + invalid transport configuration change when asterisk is + restarted. The attempt to send the qualify request fails and we + cleaned up. However, the callback is also called which results in + a double unref of the objects involved. * Put a wrapper around + pjsip_endpt_send_request() to detect when the passed in callback + is called because of an error so callers can know to not cleanup. + * Made send_request_cb() able to handle repeated challenges (Up + to 10). * Fix periodic endpoint qualify OPTIONS sched deletion + race by avoiding it. The sched entry will no longer self stop and + must be externally stopped. * Added REF_DEBUG description tags to + struct sched_data in pjsip_options.c. * Fix some off-nominal ref + leaks in schedule_qualify(), qualify_and_schedule(). * Reordered + pjsip_options.c module start/stop code to cleanup better on + error. ASTERISK-24295 #close Reported by: Rogger Padilla Review: + https://reviewboard.asterisk.org/r/3954/ ........ Merged + revisions 423866 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-24 08:53 +0000 [r423803] Walter Doekes <walter+asterisk@wjd.nu> + + * /, channels/chan_sip.c: chan_sip: Unref outbound proxy structure + on dialog/pvt destruction. Make sure outbound proxy refs are + always unreffed on dialog destruction. Review: + https://reviewboard.asterisk.org/r/4016/ ........ Merged + revisions 423800 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423801 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423802 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-23 14:29 +0000 [r423783] Mark Michelson <mmichelson@digium.com> + + * tests/test_cel.c, tests/test_cdr.c: Make CDR and CEL unit tests + less FRACKy. Prior to this commit, CDR and CEL tests were + expected to trigger FRACKs (i.e. assertions) due to the fact that + the channels they create have no formats on them. Some code was + independently added recently that attempts to prevent FRACKs from + occurring by failing early when attempting to set up translation + paths if one or both channels support no formats. Unfortunately, + this attempt to be helpful made the CDR and CEL tests go from + simply FRACKing to outright failing and in some cases, failing so + badly as to crash Asterisk. This commit seeks to correct past + mistakes by adding the ulaw format to channels created by the CDR + and CEL unit tests. This makes setting up translation paths + succeed, eliminates previously-seen FRACKs, and ultimately causes + the unit tests to succeed again. Review: + https://reviewboard.asterisk.org/r/4014 + +2014-09-22 19:48 +0000 [r423660-423723] Walter Doekes <walter+asterisk@wjd.nu> + + * /, channels/chan_sip.c: chan_sip: On INVITE retransmission, don't + add an extra 503 response. INVITE arrives to asterisk, asterisk + responds Busy(). If the INVITE is retransmitted, asterisk would + generate a 503 in addition to the 486. Thanks Torrey Searle for + providing a working regression test. ASTERISK-24335 #close + Review: https://reviewboard.asterisk.org/r/4003/ Patches: + retrans_486_invite.patch uploaded by Torrey Searle (License + #5334) ........ Merged revisions 423720 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423721 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423722 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/editline/readline.c: cli.c: Fix tab completion "module + load" when MALLOC_DEBUG is enabled. r421600 conflicted with + r155763. ASTERISK-24348 #close ........ Merged revisions 423657 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 423658 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423659 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-21 01:15 +0000 [r423618-423641] Matthew Jordan <mjordan@digium.com> + + * main/channel.c: main/channel: Unlock channel in off-nominal path + In r423414 (13) / r423415 (trunk), an API call that determines if + a format capability structure is empty was added. This returns + true if the format capability structure is completely empty or + "none". A check for this was added in channel.c's set_format + call. Unfortunately, when this check was true, it returned from + the function while still holding the channel lock. This caused + the CDR unit tests - which have a tendency to create channels + with no formats - to deadlock. Whoops. This patch unlocks the + channel on the off-nominal path. + + * rest-api/api-docs/events.json, /: rest-api/api-docs/events.json: + Remove non-compliant 'extends' attribute Prior to the release of + Swagger 1.2, the attribute 'extends' was being promoted as a + possible way to show that a particular object extends an existing + object. Instead, the Swagger specification went with the + 'subTypes' attribute in the base object. This patch removes the + unsupported attribute; the object that the offending objects + proposed to extend already lists them in its 'subTypes' + attribute. ASTERISK-24300 #close Reported by: Bradley Watkins + ........ Merged revisions 423620 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json: rest-api/api-docs: Correct + basePath in resources to match top resources file The + resources.json file that defines the resource JSON files used + with ARI references a basePath of 'http://localhost:8088/ari'. + This does not match what is defined in the resource files + themselves, 'http://localhost:8088/stasis'. The correct base path + is the one that includes 'ari' in the URL; this patch updates the + various resource JSON files to have the correct basePath. + ASTERISK-24339 #close Reported by: Bradley Watkins ........ + Merged revisions 423617 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-19 19:51 +0000 [r423580] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on + unload/load and don't say the module doesn't exist on reload. + When unloading the module did not unregister the CLI commands + causing a crash upon load when they were registered again. When + reloading the module the return value from the config options + framework was not checked to determine if an error occurred or + not. This caused a message to be output saying the module did not + exist when reloading if no changes were present. AST-1433 #close + AST-1434 #close ........ Merged revisions 423579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-19 17:08 +0000 [r423561] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c: + res_pjsip_sdp_rtp.c: Fix native formats containing formats that + were not negotiated. Outgoing PJSIP calls can result in + non-negotiated formats listed in the channel's native formats if + video formats are listed in the endpoint's configuration. The + resulting call could then use a non-negotiated format resulting + in one way audio. * Simplified the update of session->req_caps in + set_caps(). Why do something in five steps when only one is + needed? AFS-162 #close Review: + https://reviewboard.asterisk.org/r/4000/ + +2014-09-19 15:18 +0000 [r423524-423530] Jonathan Rose <jrose@digium.com> + + * /, main/stasis_channels.c: Stasis_channels: Resolve unfinished + Dials when doing masquerades Masquerades into channels that are + in the dialing state don't end their dial and this goes against + the model for things like CDRs and generating Dial end manager + actions and such. ASTERISK-24237 #close Reported by: Richard + Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........ + Merged revisions 423525 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2 + jitterbuffer settings Caused by format changes in Asterisk 13 + ASTERISK-24265 #close Reported by: Dafi Ni Review: + https://reviewboard.asterisk.org/r/3999/ + +2014-09-19 12:45 +0000 [r423504] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/framehook.h, /, main/framehook.c, + res/res_pjsip_t38.c: PJSIP: Prevent T38 framehook being put on + wrong channel This change gives framehooks a reverse-direction + masquerade callback in addition to chan_fixup_cb similar to the + callback added to datastores to handle the same situation. The + new callback provides the same parameters as the fixup callback, + but is called on the new channel's framehooks before moving + framehooks from the old channel to the new channel. This gives + the framehooks an oppurtunity to decide whether they should + remain on the new channel or be removed. This new callback is + used to prevent the PJSIP T.38 framehook from remaining on a + masqueraded channel if the new channel is not also a PJSIP + channel. This was causing a crash when a local channel was + masqueraded into a PJSIP channel and the framehook was executed + on the local channel since the channel's tech private data was + not structured as expected. Review: + https://reviewboard.asterisk.org/r/4001/ ........ Merged + revisions 423503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 19:30 +0000 [r423482] Sean Bright <sean@malleable.com> + + * res/res_pjsip/config_auth.c, /: res_pjsip: Don't require a + password when doing userpass authentication. An empty password is + valid for username/password authentication so we should allow + password to be empty/not supplied. Review: + https://reviewboard.asterisk.org/r/3988 ........ Merged revisions + 423481 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 19:22 +0000 [r423478] George Joseph <george.joseph@fairview5.com> + + * tests/test_strings.c, /, main/utils.c, + include/asterisk/strings.h: utils: Create ast_strsep function + that ignores separators inside quotes This function acts like + strsep with three exceptions... * The separator is a single + character instead of a string. * Separators inside quotes are + treated literally instead of like separators. * You can elect to + have leading and trailing whitespace and quotes stripped from the + result and have '\' sequences unescaped. Like strsep, ast_strsep + maintains no internal state and you can call it recursively using + different separators on the same storage. Also like strsep, for + consistent results, consecutive separators are not collapsed so + you may get an empty string as a valid result. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ + Merged revisions 423476 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 18:31 +0000 [r423462] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c: Add subscription state test events. These + are needed for a set of batched notification RLS tests that are + about to be committed to the testsuite. Review: + https://reviewboard.asterisk.org/r/3967 + +2014-09-18 17:11 +0000 [r423425] Jonathan Rose <jrose@digium.com> + + * res/res_pjsip_endpoint_identifier_ip.c, /: + res_pjsip_endpoint_identifier_ip: Fix parsing of match value with + CIDR Also fixes comma separates match lists ASTERISK-24290 #close + Reported by: Ray Crumrine Review: + https://reviewboard.asterisk.org/r/3995/ ........ Merged + revisions 423417 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 17:09 +0000 [r423418-423423] Richard Mudgett <rmudgett@digium.com> + + * bridges/bridge_softmix.c: bridge_softmix.c: Made use + ao2_replace() instead of the inline equivalent. * Clarified some + read/write format comments. * Fixed a doxygen tag typo. + + * main/astobj2.c, contrib/scripts/refcounter.py, /: + astobj2.c/refcounter.py: Fix to deal with invalid object refs. * + Make astob2 REF_DEBUG output an invalid object line when an + invalid ao2 object ref/unref is attempted. This is similar to the + constructor/destructor lines. * Fixed refcounter.py to handle + skewed objects that have constructor/destructor states. * Made + refcounter.py highlight the invalid ao2 object refs by putting + them in their own section of the processed output file. * Made + refcounter.py highlight unreffing an object by more than one that + results in a negative ref count and the object being destroyed. + The abnormally destroyed object is reported in the invalid and + finalized object sections of the output. Review: + https://reviewboard.asterisk.org/r/3971/ ........ Merged + revisions 423349 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423400 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423416 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 16:37 +0000 [r423348-423414] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/format_cap.h, main/channel.c, main/format_cap.c, + main/translate.c: Add API call to determine if format capability + structure is "empty". Empty here means that there are no formats + in the format_cap structure or the only format in it is the + "none" format. I've added calls to check the emptiness of a + format_cap in a few places in order to short-circuit operations + that would otherwise be pointless as well as to prevent some + assertions from being triggered in cases where channels with no + formats are used. + + * /, res/res_fax_spandsp.c: res_fax_spandsp: Properly handle + cleanup before starting FAXes. If faxing fails at a very early + stage, then it is possible for us to pass a NULL t30 state + pointer to spandsp, which spandsp is none too pleased with. This + patch ensures that we pass the correct pointer to spandsp in the + situation where we have not yet set our local t30 state pointer. + ASTERISK-24301 #close Reported by Matt Jordan Patches: + ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License + #5049) ........ Merged revisions 423360 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423365 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c, + res/res_pjsip_dialog_info_body_generator.c, + res/res_pjsip_xpidf_body_generator.c, + res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c, + res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pidf_body_generator.c: res_pjsip_pubsub: Add some + type safety when generating NOTIFY bodies. res_pjsip_pubsub has + two separate checks that it makes when a SUBSCRIBE arrives. * It + checks that there is a subscription handler for the Event * It + checks that there are body generators for the types in the Accept + header The problem is, there's nothing that ensures that these + two things will actually mesh with each other. For instance, + Asterisk will accept a subscription to MWI that accepts pidf+xml + bodies. That doesn't make sense. With this commit, we add some + type information to the mix. Subscription handlers state they + generate data of type X, and body generators state that they + consume data of type X. This way, Asterisk doesn't end up in some + hilariously mismatched situation like the one in the previous + paragraph. ASTERISK-24136 #close Reported by Mark Michelson + Review: https://reviewboard.asterisk.org/r/3877 Review: + https://reviewboard.asterisk.org/r/3878 ........ Merged revisions + 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 15:13 +0000 [r423284] George Joseph <george.joseph@fairview5.com> + + * /, res/res_pjsip/location.c, + res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_auth.c: + res_pjsip: ami: Fix error in AMI output when an endpoint has no + transport When no transport is associated to an endpoint, the AMI + output for PJSIPShowEndpoint indicates an error instead of + silently ignoring the missing transport. This patch causes the + error to appear only if a transport was specified on the endpoint + and the transport doesn't exist. It also fixes an issue with + counting the objects that were actually found. ASTERISK-24161 + #close ASTERISK-24331 #close Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3998/ ........ Merged + revisions 423282 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 15:00 +0000 [r423281] David M. Lee <dlee@digium.com> + + * makeopts.in, Makefile: Only install dahdi_span_config_hook if + DAHDI is enabled This patch changes the install to only install + the hook script if DAHDI is enabled. It also adds the script to + the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so + that it's not between the _MAKEOPTS variables and their comment. + This allows installs which specify a --prefix to work normally, + as long as they don't enable DAHDI. Review: + https://reviewboard.asterisk.org/r/3972/ + +2014-09-18 14:45 +0000 [r423279] George Joseph <george.joseph@fairview5.com> + + * main/manager.c, /, include/asterisk/config.h, main/config.c: + config: bug: Fix SEGV in ast_category_insert when matching + category isn't found If you call ast_category_insert with a match + category that doesn't exist, the list traverse runs out of 'next' + categories and you get a SEGV. This patch adds check for the + end-of-list condition and changes the signature to return an int + for success/failure indication instead of a void. The only + consumer of this function is manager and it was also changed to + use the return value. Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3993/ ........ Merged + revisions 423276 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423277 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423278 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-17 18:05 +0000 [r423209-423255] Joshua Colp <jcolp@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the + thread terminating pj stuff is registered. ........ Merged + revisions 423253 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423254 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix 100% CPU usage + due to timer heap thread spinning. Side note: I need a vacation. + ........ Merged revisions 423210 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423211 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix building when + pjproject is not used. ........ Merged revisions 423207 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423208 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-16 16:32 +0000 [r423192] Scott Griepentrog <sgriepentrog@digium.com> + + * apps/app_voicemail.c, include/asterisk/file.h, main/file.c: + Voicemail: get correct duration when copying file to vm Changes + made during format improvements resulted in the recording to + voicemail option 'm' of the MixMonitor app writing a zero length + duration in the msgXXXX.txt file. This change introduces a new + function ast_ratestream(), which provides the sample rate of the + format associated with the stream, and updates the app_voicemail + function for ast_app_copy_recording_to_vm to calculate the right + duration. Review: https://reviewboard.asterisk.org/r/3996/ + ASTERISK-24328 #close + +2014-09-16 12:12 +0000 [r423152-423173] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, /: res_pjsip_session: Fix usage of wrong + memory pool when creating local SDP. ........ Merged revisions + 423172 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, /: + res_rtp_asterisk: Fix a myriad of TURN client issues. 1. The + number of file descriptors an ioqueue instance can handle is + fixed, so we now spawn the required number to handle the load. 2. + Our transport identifiers were exceeding the range supported by + pjnath. 3. The TURN client did not set up client binding causing + needless bandwidth usage. 4. The code no longer updates address + information on each packet. 5. STUN traffic was getting looped + back to Asterisk instead of going through the TURN server. 6. + Synchronization now ensures things are completely setup or + destroyed. 7. Logging now reflects the target the TURN server is + sending to/receiving from on our behalf. ASTERISK-23577 #close + Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: + Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ + ........ Merged revisions 423150 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423151 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-15 10:49 +0000 [r423069-423129] Walter Doekes <walter+asterisk@wjd.nu> + + * /, + contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py + (added): contrib: Fix verifyi typo in alembic DB script + ps_transport table. Reported by: Zogot (on IRC) Patches: tmp.diff + uploaded by Zogot, cleaned up by me. ........ Merged revisions + 423128 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/samples/sip.conf.sample, /: chan_sip: Clarify that + sipdebug=yes cannot be undone by the CLI. Document it in + sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod + Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged + revisions 423066 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423067 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423068 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-12 16:09 +0000 [r422985] Jonathan Rose <jrose@digium.com> + + * main/config.c, /: Realtime: Fix a bug that caused realtime + destroy command to crash Also has could affect with anything that + goes through ast_destroy_realtime. If a CLI user used the command + 'realtime destroy <family>' with only a single column/value pair, + Asterisk would crash when trying to create a variable list from a + NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson + Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged + revisions 422984 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-11 22:16 +0000 [r422965] Mark Michelson <mmichelson@digium.com> + + * /, main/app.c: Remove undocumented default behavior of + ast_play_and_record_full acceptdtmf. ast_play_and_record_full() + has a parameter called "acceptdtmf" that is a string of + acceptable DTMF digits that may be pressed by a caller to end and + accept the recording. ARI uses this function in order to perform + recording, and it provides options for what is passed as + acceptdtmf to ast_play_and_record_full(). By default, ARI passes + an empty string, with the intention that no DTMF can be used to + end the recording. The problem is that ast_play_and_record_full() + attempts to be "helpful" by setting "#" as the acceptdtmf if an + empty string or NULL pointer has been passed in. With ARI, this + results in unexpected behavior occurring if you have attempted to + intercept "#" yourself in order to perform some other + manipulation of the live recording. This change removes the + "helpful" behavior by no longer accepting "#" as a default + acceptdtmf if none is specified by the caller of + ast_play_and_record_full(). This makes the ARI scenario work as + expected. The other callers of ast_play_and_record_full() are + app_voicemail and app_minivm, and in both cases, they pass an + explicit "#" to ast_play_and_record_full() as acceptdtmf, so they + are unaffected by this change. ........ Merged revisions 422964 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-10 16:04 +0000 [r422905] George Joseph <george.joseph@fairview5.com> + + * /, main/config.c: config: bug: fix truncation of included config + files on permissions error ast_config_text_file_save() currently + truncates include files as they are processed. If a subsequent + include file or the main config file has a permissions error that + prevents writing, earlier include files are left truncated + resulting in a frantic search for backups. This patch causes + ast_config_text_file_save to check for write access on all files + before it truncates any of them. Will be applied 1.8 > trunk. + Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3986/ ........ Merged + revisions 422900 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422903 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422904 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-10 15:59 +0000 [r422901] Sean Bright <sean@malleable.com> + + * res/res_pjsip/config_auth.c, /: pjsip/config_auth.c: Add missing + whitespace to log messages. The errors generated when validating + 'auth' settings are missing a space which makes the messages a + little confusing. ........ Merged revisions 422899 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-09 20:01 +0000 [r422883] Rusty Newton <rnewton@digium.com> + + * /, sounds/sounds.xml, sounds/Makefile: Sounds/BuildSystem: + Modifications to include new releases and Japanese language. + Modifying Makefile and sounds.xml to include new core 1.4.26 and + extra 1.4.15 sound prompt releases, plus the new Japanese core + sound prompts contributed by QLOOG. ASTERISK-23324 Reported by: + Kevin McCoy Tested by: Rusty Newton ........ Merged revisions + 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 422790 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422791 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-08 18:03 +0000 [r422851-422855] Mark Michelson <mmichelson@digium.com> + + * configs/samples/pjsip.conf.sample: Add note about configuring + list_items on a single line. + + * configs/samples/pjsip.conf.sample: Add sample configuration for + resource lists. On review /r/3977, it was recommended to note in + the sample configuration about the size limitation for resource + lists. However, since there was no section in the sample + configuration at all for resource list subscriptions, I decided + to make a separate commit where I have added the necessary sample + configuration as well as the size limitation warning. + + * res/res_pjsip_pubsub.c: Pre-allocate transmission data buffer for + RLS NOTIFY requests. PJSIP, unless a constant is modified at + compilation time, limits SIP requests to 4000 bytes. Full-state + RLS notifications can easily exceed this limit with moderately + small lists. This changeset allows for Asterisk to work around + this size limit by performing its own allocation of the + transmission data buffer. This way, Asterisk can allocate a + buffer that exceeds the built-in maximum. We still impose our own + limit of 64000 bytes, mainly because making allocations larger + than that is a bit absurd. ASTERISK-24181 #close Reported by Mark + Michelson Review: https://reviewboard.asterisk.org/r/3977 + +2014-09-08 15:41 +0000 [r422836] Jonathan Rose <jrose@digium.com> + + * res/res_pjsip_pubsub.c: res_pjsip_pubsub: Check supported headers + for eventlist when subscribing to resource list + https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan + According to the off-nominal plan, if evenlist support is not + specified in a SUBSCRIBE's supported header(s), that subscription + should be rejected with an error. ASTERISK-23871 Reported by: + Mark Michelson Review: + https://reviewboard.asterisk.org/r/3960/diff/#index_header + +2014-09-06 22:49 +0000 [r422767-422770] Matthew Jordan <mjordan@digium.com> + + * /, main/cdr.c: main/cdr: Copy over location information during a + fork When a CDR is forked, a new CDR is created and appended to + the CDR chain for the Party A. The forked CDR starts life off as + a clone of the last non-finalized for the particular Party A. In + the past, merely copying over the snapshots for Party A/Party B + would be sufficient. However, as the CDRs now contain cached + information from Party A - specifically application/data, + context, and extension - we need to copy that over during a fork + as well. Huzzah for unit tests catching this when the + context/extension were derived from a cached value on the CDR + instead of on Party A. ........ Merged revisions 422769 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/rtp_engine.c, /: main/rtp_engine: Format NTP timestamps as + unsigned ints On some systems, a timeval's tv_sec/tv_usec will be + unsigned lont ints, as opposed to long ints. When the RTP engine + formats these as strings, it was previously formatting them as + signed integers, which can result in some odd negative timestamp + values (particularly on 32-bit systems). This patch formats the + values as unsigned long integers. ........ Merged revisions + 422766 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-06 19:12 +0000 [r422747] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix retrieval of + "ice-pwd" attribute if in session and not media stream. ........ + Merged revisions 422746 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-05 22:03 +0000 [r422716-422719] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /, apps/app_macro.c, include/asterisk/channel.h, + apps/app_stack.c: main/cdrs: Preserve context/extension when + executing a Macro or GoSub The context/extension in a CDR is + generally considered the destination of a call. When looking at a + 2-party call CDR, users will typically be presented with the + following: context exten channel dest_channel app data default + 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial + actually takes place in a Macro, the current behaviour in 12 will + result in the following CDR: context exten channel dest_channel + app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The + same is true of a GoSub: context exten channel dest_channel app + data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This + generally makes the context/exten fields less than useful. It + isn't hard to preserve these values in the CDR state machine; + however, we need to have something that informs us when a channel + is executing a subroutine. Prior to this patch, there isn't + anything that does this. This patch solves this problem by adding + a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on + a channel when it executes a Macro or a GoSub. The CDR engine + looks for this value when updating a Party A snapshot; if the + flag is present, we don't override the context/exten on the main + CDR object. In a funny quirk, executing a hangup handler must + *not* abide by this logic, as the endbeforehexten logic assumes + that the user wants to see data that occurs in hangup logic, + which includes those subroutines. Since those execute outside of + a typical Dial operation (and will typically have their own + dedicated CDR anyway), this is unlikely to cause any heartburn. + Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254 + #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis + ........ Merged revisions 422718 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: main/cdr: Fix crash/memory consumption in CDRs in + multi-party bridge scenarios This patch fixes an issue where CDRs + would get stuck generating an infinite number of CDRs, eventually + crashing Asterisk (and consuming a lot of memory along the way). + When a channel enters into a multi-party bridge, the CDR engine + creates mappings of each participant to each other participant, + picking the 'A' party as it goes. So, if we have four channels in + a multi-party bridge (Alice, Bob, Charlie, Denise), we would have + something like: Alice => Bob Alice => Charlie Alice => Denise Bob + => Charlie Bob => Denise Charlie => Denise This works fine when + participants enter the bridge a single time. When a participant + leaves a bridge, the CDRs for that channel are transitioned to a + finalized state. The bug occurs if Bob rejoins. When the CDR + engine creates mappings between the channels, it walks through + all the participants currently in the bridge, and realizes that + no one in the bridge can create a CDR with the channel (Bob). As + such it creates a new CDR for the candidate and appends it to + that candidate's chain. Unfortunately, on this particular code + path, it doesn't stop traversing the candidate's chain. Since we + just added ourselves to the chain, this causes the loop to keep + going, constantly adding new CDRs. This patch makes it so the + engine bails when it creates a CDR match in this case. Review: + https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close + Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat + ASTERISK-24208 Reported by: Frankie Chin ........ Merged + revisions 422715 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-05 20:35 +0000 [r422700] Richard Mudgett <rmudgett@digium.com> + + * funcs/func_channel.c: func_channel.c: Add missing locking to some + CHANNEL() requests. * The CHANNEL() audionativeformat, + videonativeformat, audioreadformat, and audiowriteformat now need + locking since the media format rework when accessing the + channel's format pointers. * Increased the buffer size for + CHANNEL() audionativeformat and videonativeformat output strings + since the allow=all can be a lengthy list. * Tweaked the + CHANNEL() XML documentation for secure_bridge_signaling, + secure_bridge_media, and state. * Ensured the output buffer is + initialized for secure_bridge_signaling and secure_bridge_media. + * Made use the locked_copy_string() macro instead of inlining it + for trace and checkhangup. + +2014-09-05 20:11 +0000 [r422665-422684] Jonathan Rose <jrose@digium.com> + + * main/dial.c, include/asterisk/dial.h: Dial API: Add a dial option + to indicate the dialed channel will replace dialer Adds an option + to the dial API that marks an outgoing dial as replacing the + dialing channel for the purpose of propagating accountcode. When + it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of + AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on + the involved channels with ast_channel_req_accountcodes. Review: + https://reviewboard.asterisk.org/r/3968/ + + * main/cli.c, /: Call IDs: Fix appearance of call ID in core show + channels when NULL NULL call IDs were meant to appear as '(none)' + but instead were showing the contents of an uninitialized + character buffer. ASTERISK-24223 Review: + https://reviewboard.asterisk.org/r/3979/ ........ Merged + revisions 422664 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-05 17:36 +0000 [r422661] Richard Mudgett <rmudgett@digium.com> + + * main/devicestate.c, channels/chan_iax2.c: devicestate.c: Minor + tweaks * In ast_state_chan2dev() use ARRAY_LEN() instead of a + sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c. + +2014-09-05 13:28 +0000 [r422646] Kinsey Moore <kmoore@digium.com> + + * menuselect/menuselect.c: Menuselect: Fix incorrect enabling on + failed deps This corrects a situation where menuselect can + incorrectly enable a module by default that has defaultenabled + set to "no" and has failed/non-selected dependencies. The bug is + due to an inverted test when checking for whether the given + module should be set to enabled by default on load. Review: + https://reviewboard.asterisk.org/r/3975/ Reported by: John + Bigelow + +2014-09-04 21:23 +0000 [r422631] Jonathan Rose <jrose@digium.com> + + * main/manager.c, /: Manager: Require read permission for SYSTEM in + order to send FullyBooted Review: + https://reviewboard.asterisk.org/r/3969/ ........ Merged + revisions 422584 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422625 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422626 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-03 14:05 +0000 [r422558] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_transport_websocket.c, /: + res_pjsip_transport_websocket: Fix crash when the Contact header + is not a URI. The code for changing the Contact header wrongly + assumed that the Contact would always contain a URI. This is + incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged + revisions 422557 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-02 20:29 +0000 [r422542] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_pjsip.c, res/res_pjsip_diversion.c, + res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h: + Resolve race condition where channels enter dialplan application + before media has been negotiated. Testsuite tests will + occasionally fail because on reception of a 200 OK SIP response, + an AST_CONTROL_ANSWER frame is queued prior to when media has + finished being negotiated. This is because session supplements + are called into before PJSIP's inv_session code has told us that + media has been updated. Sometimes the queued answer frame is + handled by the PBX thread before the ensuing media negotiations + occur, causing a test failure. As it turns out, there is another + place that session supplements could be called into, which is + after media has finished getting negotiated. What this commit + introduces is a means for session supplements to indicate when + they wish to be called into when handling an incoming SIP + response. By default, all session supplements will be run at the + same point that they were prior to this commit. However, session + supplements may indicate that they wish to be handled earlier + than normal on redirects, or they may indicate they wish to be + handled after media has been negotiated. In this changeset, two + session supplements have been updated to indicate a preference + for when they should be run: res_pjsip_diversion executes before + handling redirection in order to get information from the + Diversion header, and chan_pjsip now handles responses to INVITEs + after media negotiation to fix the race condition mentioned + previously. ASTERISK-24212 #close Reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/3930 ........ Merged revisions + 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-01 14:16 +0000 [r422504-422507] Matthew Jordan <mjordan@digium.com> + + * main/cli.c, /: main/cli: Do not attempt to show CDR data for + internal channels Internal channels don't have CDRs. Querying the + CDR engine for their variables will make it cranky. ........ + Merged revisions 422506 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis.c, /, res/stasis/stasis_bridge.c: res_stasis: + Don't play MoH to channels by default when added to holding + bridges When ARI manipulates a bridge, it generally doesn't care + what the mixing technology is. Operations on a bridge initiated + through ARI should perform their action in generally the same + way, regardless of the bridge's mixing technology. While the + mixing technology may determine how media flows to channels, the + actual operations on a bridge themselves should be the same. + Currently, this isn't the case with holding bridges. When a + channel joins without a role, MoH is started on that channel + automatically. Subsequent bridge operations that would stop MoH + would fail (as there is no Announcer channel playing MoH to the + bridge). Starting MoH on the bridge will also create two MoH + streams: one from the MoH being played on the participant + channel, and one from the announcer channel. From the perspective + of ARI users, this is counter-intuitive - I would not expect MoH + to be started for me. The mixing technology determines how media + is shared between participants, not the application experience. + This patch does the following: * The Stasis bridge class now + inspects channels as they are going into a bridge. If the bridge + has a holding capability, and the channel has no roles, we give + it a participant role and mark the default behaviour to have no + entertainment. This allows addChannel operations to continue to + set a participant role with an entertainment option if it felt + like it (or could do it). * The music on hold channel is now + Stasis approved (tm) Review: + https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close + Reported by: Samuel Galarneau Tested by: Samuel Galarneau + ........ Merged revisions 422503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-30 17:32 +0000 [r422442-422445] George Joseph <george.joseph@fairview5.com> + + * apps/app_confbridge.c, /: confbridge: Add Duration to + ConfbridgeList event The ConfbridgeList event doesn't include how + long the user has been a member of the conference. This patch + adds Duration (seconds) which is based on user->chan->answertime. + Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3955/ ........ Merged + revisions 422444 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Make WaitEvent action respect + eventfilters A WaitEvent issued via an http session isn't + respecting eventfilters defined for the user. I just added a + match_filter to the predicate that controls astman_append. Tested + by: George Joseph Review: + https://reviewboard.asterisk.org/r/3958/ ........ Merged + revisions 422439 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422440 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422441 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-29 19:40 +0000 [r422374-422379] Matthew Jordan <mjordan@digium.com> + + * doc/smsq.8 (added), /: doc: Add a manpage for the smsq utility + This patch adds a manpage for the smsq utility. Note that this is + one of the patches the Debian distro applies for the Asterisk + project, as per ASTERISK-24191. Review: + https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close + Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy + Laine (License 6561) ........ Merged revisions 422376 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422377 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422378 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * doc/aelparse.8 (added), /: doc: Add a manpage for the aelparse + utility This patch adds a manpage for the aelparse utility. Note + that this is one of the patches the Debian distro applies for the + Asterisk project, as per ASTERISK-24191. Review: + https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close + Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy + Laine (License 6561) ........ Merged revisions 422371 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422372 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422373 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-29 19:05 +0000 [r422359] Scott Griepentrog <sgriepentrog@digium.com> + + * channels/chan_sip.c: The assertion that peer was not found on + final event message was being triggered on configuration reload. + This patch changes that case to just return instead. Review: + https://reviewboard.asterisk.org/r/3953/ Commited in trunk + revision 422358 + +2014-08-28 21:54 +0000 [r422296] Matthew Jordan <mjordan@digium.com> + + * LICENSE, /: LICENSE: Clarify language in Asterisk's LICENSE to + allow for linking to UniMRCP The UniMRCP project distributes + Asterisk modules that integrate Asterisk with UniMRCP, and other + Asterisk users use the UniMRCP library as well. Unfortunately, + the UniMRCP license is Apache 2.0, which per the Free Software + Foundation, is not a compatible license with the GPLv2. "Please + note that this license is not compatible with GPL version 2, + because it has some requirements that are not in that GPL + version. These include certain patent termination and + indemnification provisions. The patent termination provision is a + good thing, which is why we recommend the Apache 2.0 license for + substantial programs over other lax permissive licenses." On the + other hand, UniMRCP is a great project and we'd like to let + people use it with Asterisk. This patch updates the LICENSE text + to allow users to link Asterisk with UniMRCP and distribute the + resulting binaries. ........ Merged revisions 422293 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422294 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422295 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-28 20:30 +0000 [r422276] Michael L. Young <elgueromexicano@gmail.com> + + * /, channels/chan_iax2.c: chan_iax2: Fix Dynamic IAX2 + Registrations After Temporary DNS Failure The reporter on the + issue found some issues when upgrading from version 10 to 11 on + 55 hosts. Two situations that can occur with dynamic + registrations. 1. With dnsmgr disabled, if the host is not + resolvable we are not trying to resolve the host again when it is + time to attempt to register again. This results in never + registering to the host. 2. With dnsmgr enabled, when the host is + temporarily not resolvable the address is set to 0.0.0.0:0 and + then when the host is resolvable the port is not being restored + and stays set to 0. This patch resolves these two issues by: * + Storing the hostname so that it can be used for resolving with + DNS. * Resolve the hostname on the next scheduled attempt to + register. * Storing the port used to reach the host so that when + the hostname is resolvable again, we can set the port again if + the port is still unset after looking up the host. ASTERISK-23767 + #close Reported by: David Herselman Tested by: David Herselman, + Michael L. Young Patches: + asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3856/ ........ Merged + revisions 422274 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422275 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-28 17:25 +0000 [r422256] Richard Mudgett <rmudgett@digium.com> + + * /, UPGRADE.txt: Added ConfBridge AMI event note to UPGRADE.txt. + ........ Merged revisions 422255 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-28 15:49 +0000 [r422239] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c: Fix bug that did not allow for multiple + batched RLS notifications to be sent. A misunderstanding of how + the scheduler worked caused further batched notifications beyond + the first not to get scheduled. Now we reset our scheduler ID to + -1 after the batched notification is sent. This way, further + notifications can be scheduled when they arise. + +2014-08-28 00:36 +0000 [r422200-422215] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip/pjsip_options.c, /: res/res_pjsip/pjsip_options.c: + Eliminate excessive RAII_VAR usage. * Fix off nominal ref leak in + find_or_create_contact_status(). * Add missing NULL check of + status in update_contact_status() and init_start_time(). ........ + Merged revisions 422214 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/sched.c, include/asterisk/sched.h: sched: Fix typo and + whitespace change. + +2014-08-27 17:29 +0000 [r422177] George Joseph <george.joseph@fairview5.com> + + * /, apps/confbridge/confbridge_manager.c, apps/app_confbridge.c: + confbridge: Add 'Admin' param to join, leave, mute, unmute and + talking events Currently there's no way to tell if a user is an + admin or not when receiving the join, leave, mute, unmute and + talking events. This patch adds that capability. Tested by: + George Joseph Review: https://reviewboard.asterisk.org/r/3950/ + ........ Merged revisions 422176 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-27 15:31 +0000 [r422154] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/utils.h, /, channels/chan_sip.c, + tests/test_callerid.c (added), tests/test_utils.c, + main/callerid.c, main/utils.c, res/res_pjsip_caller_id.c: + CallerID: Fix parsing of malformed callerid This allows the + callerid parsing function to handle malformed input strings and + strings containing escaped and unescaped double quotes. This also + adds a unittest to cover many of the cases where the parsing + algorithm previously failed. Review: + https://reviewboard.asterisk.org/r/3923/ Review: + https://reviewboard.asterisk.org/r/3933/ ........ Merged + revisions 422112 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422113 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422114 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-26 23:28 +0000 [r422091] George Joseph <george.joseph@fairview5.com> + + * apps/app_confbridge.c, /: confbridge: Make kick, mute and unmute + handle channel targets consistently. Kick, mute and unmute were a + little inconsistent in their handling of channel targets. This + patch cleans that up by insuring they all handle the 'all' target + consistently and adds the 'participants' target which acts on + non-admins. Documentation for kick was also cleaned up as it + never supported partial channel names. Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged + revisions 422090 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-26 22:13 +0000 [r422071] Mark Michelson <mmichelson@digium.com> + + * main/sched.c, /: Fix race condition in the scheduler when + deleting a running entry. When scheduled tasks run, they are + removed from the heap (or hashtab). When a scheduled task is + deleted, if the task can't be found in the heap (or hashtab), an + assertion is triggered. If DO_CRASH is enabled, this assertion + causes a crash. The problem is, sometimes it just so happens that + someone attempts to delete a scheduled task at the time that it + is running, leading to a crash. This change corrects the issue by + tracking which task is currently running. If that task is + attempted to be deleted, then we mark the task, and then wait for + the task to complete. This way, we can be sure to coordinate task + deletion and memory freeing. ASTERISK-24212 Reported by Matt + Jordan Review: https://reviewboard.asterisk.org/r/3927 ........ + Merged revisions 422070 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-25 16:44 +0000 [r421979-422037] Richard Mudgett <rmudgett@digium.com> + + * res/res_musiconhold.c: res_musiconhold.c: Release any format refs + before memset(). * Clear the channel music_state pointer before + destroying the music_state object for safety. + + * res/res_musiconhold.c, /: res_musiconhold: Fix MOH restarting + where it left off from the last hold. Restore code removed by + https://reviewboard.asterisk.org/r/3536/ that introduced a + regression that prevents MOH from restarting were it left off the + last time. ASTERISK-24019 #close Reported by: Jason Richards + Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch + uploaded by rmudgett Review: + https://reviewboard.asterisk.org/r/3928/ ........ Merged + revisions 421976 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421977 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421978 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-24 19:36 +0000 [r421911-421956] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_transport_websocket.c, /: + res_pjsip_transport_websocket: Attach the Websocket module on + outgoing INVITEs. In order to alter the Contact header on + in-dialog requests and responses the Websocket module must be + attached on outgoing INVITEs. The Contact header is modified so + that the PJSIP transport layer can find and use the existing + Websocket connection based on the source IP address, port, and + transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov + ........ Merged revisions 421955 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_transport_websocket.c: + res_pjsip_transport_websocket: Fix a progressive memory growth. + The packet structure used to receive messages was using the + transport pool. This meant that for each parsing the pool would + grow accordingly. Since memory can not be reclaimed without + resetting it this would cause the memory pool to grow and grow. + This change uses a specific memory pool for the packet structure + and resets it to a fresh state after the message has been + received and handled. ........ Merged revisions 421939 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_transport_websocket.c: + res_pjsip_transport_websocket: Ensure secure Websocket clients + can be called. This change enforces the transport in the Contact + header for Websocket clients. Previously a client may provide a + transport of 'ws' when it is actually using a transport of 'wss'. + This would cause outgoing calls to fail as the existing + connection could not be found. ........ Merged revisions 421931 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Use the server reflexive ICE + candidate RTCP port as provided. This code originally worked + around an issue within res_rtp_asterisk itself. The wrong socket + was being used for the STUN check for RTCP, causing the port to + be the same as RTP. This was subsequently fixed and the RTCP port + provided for the ICE candidate is correct and does not need to be + incremented. ASTERISK-23997 #close Reported by: Badalian + Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav + (license 5249) ........ Merged revisions 421909 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421910 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-22 16:56 +0000 [r421882] Mark Michelson <mmichelson@digium.com> + + * apps/app_mixmonitor.c: Fix a locking inversion in MixMonitor. We + need to unlock the audiohook before trying to lock the channel, + since the correct locking order is channel then audiohook. + +2014-08-22 16:44 +0000 [r421880] Jonathan Rose <jrose@digium.com> + + * res/res_stasis_answer.c, res/res_stasis.c, res/stasis/command.c, + res/res_stasis_playback.c, /, res/stasis/control.c, + res/stasis/stasis_bridge.c, res/stasis/command.h, + include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c: + ARI: Fix a crash caused by hanging during playback to a channel + in a bridge ASTERISK-24147 #close Reported by: Edvin Vidmar + Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged + revisions 421879 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-22 14:08 +0000 [r421860] Matthew Jordan <mjordan@digium.com> + + * main/message.c, /: main/message: Add a new-line to a DEBUG + message ........ Merged revisions 421859 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 22:07 +0000 [r421802] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_musiconhold.c: res_musiconhold.c: Remove obsolete + REF_DEBUG code. Remove unneeded code that writes to the wrong + file location in an obsolete format. ........ Merged revisions + 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 421800 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421801 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 21:42 +0000 [r421790-421797] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_session.c, /: Switch from hostname to an IP address + in the SDP origin line. Using the hostname in the SDP origin line + may not satisfy the requirement of RFC 4566 that we use a FQDN or + IP address. This change has us use the same information from the + SDP connection line if possible. If not possible, we'll use the + configured media address. And if that's not possible, we use the + result of a PJLIB call to get the IP address of ourself. + ASTERISK-23994 #close Reported by Private Name Review: + https://reviewboard.asterisk.org/r/3925 ........ Merged revisions + 421796 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c: Ensure after-bridge behavior is correct + when moving from Stasis to a non-Stasis bridge. Because of the + departable state of channels that enter Stasis bridges, Stasis + has to take responsibility for directing the channel to its + intended after-bridge destination if the channel moves from a + Stasis bridge to a non-Stasis bridge. This change ensures that + when such a move occurs, when the channel leaves the bridging + system, any after bridge gotos are honored. Review: + https://reviewboard.asterisk.org/r/3920 ........ Merged revisions + 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_caller_id.c, /: Let's try checking the name and + number, instead of the name twice. ........ Merged revisions + 421789 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 21:25 +0000 [r421788] Jonathan Rose <jrose@digium.com> + + * /, res/res_musiconhold.c: res_musiconhold: Fix reference leaks + caused when reloading with REF_DEBUG set Due to a faulty function + for debugging reference decrementing, it was possible to reduce + the refcount on the wrong object if two moh classes of the same + name were in the moh class container. (closes issue + ASTERISK-22252) Reported by: Walter Doekes Patches: + 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license + 6182) ........ Merged revisions 398937 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421777 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421779 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 21:18 +0000 [r421783] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_caller_id.c: Improve consistency of party ID + privacy usage. Prior to this change, the Remote-Party-ID header + took the position of "If caller name and number are not + explicitly allowed, then they are private" and + P-Asserted-Identity took the position of "Caller name and number + are only private if marked explicitly so" Now both mechanisms of + conveying party identification use the former approach. ........ + Merged revisions 421778 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 17:34 +0000 [r421675-421720] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: chan_sip: Don't use port derived from + fromdomain if it isn't set If a user does not provide a port in + the fromdomain setting, chan_sip will set the fromdomainport to + STANDARD_SIP_PORT (5060). The fromdomainport value will then get + used unilaterally in certain places. This causes issues with TLS, + where the default port is expected to be 5061. This patch + modifies chan_sip such that fromdomainport is only used if it is + not the standard SIP port; otherwise, the port from the SIP pvt's + recorded self IP address is used. Review: + https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close + Reported by: Elazar Broad patches: fromdomainport_fix.diff + uploaded by Elazar Broad (License 5835) ........ Merged revisions + 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 421718 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421719 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, UPGRADE.txt, main/app.c: ARI: Fix implicit answer when + playback is initiated on unanswered channel When issuing a POST + /channels/{channel_id}/play on a channel that is not yet + answered, ARI is supposed to: * Queue up an AST_CONTROL_PROGRESS + on the channel * Start up the playback of the media Instead, we + sneak an answer on the channel right before starting playing + media. This is due to ARI's usage of control_streamfile. This + function implicitly answers the channel (and doesn't give ARI the + option to stop it). The answering of the channel here is probably + unnecessary: * app_voicemail, by far the biggest consumer of this + function, always answers the channels anyway * control stream + file (in res_agi) and ControlPlayback probably shouldn't be + implicitly answering the channel. Answering should not be tied + directly to playing back media. As it turns out, the answering of + the channel here is pretty old: 356042 twilson if + (ast_channel_state(chan) != AST_STATE_UP) { 3087 anthm res = + ast_answer(chan); 180259 tilghman } (As in, ancient?) Note that + others ran into this problem and commented about it on various + mailing lists. Review: https://reviewboard.asterisk.org/r/3907/ + ASTERISK-24229 #close Reported by: Matt Jordan ........ Merged + revisions 421695 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/stasis/messaging.h, main/dns.c, /, main/format_cache.c: Clean + up files that do not end with newlines Trivial patch to add new + lines to several files missing them. This fixes warnings when + compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close + Reported by: Shaun Ruffell patches: + 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch + uploaded by Shaun Ruffell (License 5417) ........ Merged + revisions 421677 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/uri.h, main/uri.c: uri: Quiet warning about type + qualifiers ignored on function return type This patch fixes gcc + warnings that occur due to the type qualifier 'const' being + ignored on a return type of int. ASTERISK-24246 #close Reported + by: Shaun Ruffell patches: + 0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch + uploaded by Shaun Ruffell (License 5417) + +2014-08-20 22:49 +0000 [r421616-421645] Richard Mudgett <rmudgett@digium.com> + + * main/bridge.c, res/res_pjsip_sdp_rtp.c, main/file.c, + main/bridge_channel.c, channels/chan_pjsip.c, main/channel.c: + chan_pjsip: Update media translation paths when new SDP + negotiated. On a SIP reinvite that changes media strams, the + PJSIP channel driver was flooding the log with "Asked to transmit + frame type %s, while native formats is %s" warnings. * Fixes + PJSIP not setting up translation paths when the formats change on + a reinvite. AFS-63 was effectively reintroduced because of the + media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the + unexpected frame format WARNING message to include more + information. * Added protective locking while altering formats on + a channel. Reworked set_format() to simplify and protect the + formats under manipulation. * Restored some code that got lost in + the media_formats work. (channel.c:set_format() and + res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark + Michelson Review: https://reviewboard.asterisk.org/r/3906/ + + * /, main/cli.c: cli.c: Fix tab completion of "module load" when + MALLOC_DEBUG is enabled. filename_completion_function() returns + memory that was not allocated by the MALLOC_DEBUG allocation + tracker so the memory must be freed by ast_std_free(). ........ + Merged revisions 421600 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421602 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421608 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-20 20:40 +0000 [r421566-421585] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c: Set the role for inbound subscriptions + correctly. This was causing the AMI show_subscriptions test in + the testsuite to fail since all subscriptions were being seen as + subscribers instead of notifiers. + + * /, channels/chan_pjsip.c: Move evaluation of set_var options in + pjsip to the end of channel initialization. This allows for + set_var to override certain defaults such as caller ID and codec + values. This also fixes a test suite regression. The "set_var" + test suite test attempted to use set_var to override caller ID, + but a recent change caused that to no longer work. ........ + Merged revisions 421565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-20 13:04 +0000 [r421538] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/stasis_bridges.h, tests/test_cel.c, + res/ari/ari_model_validators.c, main/stasis_bridges.c, + res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /, + res/stasis/app.c, main/bridge.c: Stasis: Add information to blind + transfer event When a blind transfer occurs that is forced to + create a local channel pair to satisfy the transfer request, + information about the local channel pair is not published. This + adds a field to describe that channel to the blind transfer + message struct so that this information is conveyed properly to + consumers of the blind transfer message. This also fixes a bug in + which Stasis() was unable to properly identify the channel that + was replacing an existing Stasis-controlled channel due to a + blind transfer. Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3921/ ........ Merged + revisions 421537 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 20:28 +0000 [r421448-421488] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip.c: Alter documentation for callerid_privacy to + use correct values. ........ Merged revisions 421485 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis.c, /: Fix compilation error on certain versions of + GCC. ........ Merged revisions 421447 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 19:42 +0000 [r421445] Kinsey Moore <kmoore@digium.com> + + * main/manager.c, /: AMI Docs: Fix Status channel parameter + optionality ........ Merged revisions 421442 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421443 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421444 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 16:28 +0000 [r421423] Jonathan Rose <jrose@digium.com> + + * res/res_stasis.c, /: ARI: Fix a bug where + /channels/{channelID}/continue doesn't execute PBX If + /channels/{channelID}/continue is called on a channel that was + originated without a PBX (such as the ARI command POST channel + with a stasis application argument), the channel will not start + dialplan execution. This patch will now run the PBX out of the + stasis execution if the channel doesn't currently have an active + PBX upon continuing. ASTERISK-24043 #close Reported by: Krandon + Bruse Review: https://reviewboard.asterisk.org/r/3917/ Patches: + stasis-continue.diff submitted by Krandon Bruse (license 6631) + ........ Merged revisions 421416 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 16:11 +0000 [r421403] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_pjsip_caller_id.c, channels/chan_pjsip.c, + res/res_pjsip_session.c: chan_pjsip: Fix attended transfer + connected line name update. A calls B B answers B SIP attended + transfers to C C answers, B and C can see each other's connected + line information B completes the transfer A has number but no + name connected line information about C while C has the full + information about A I examined the incoming and outgoing party id + information handling of chan_pjsip and found several issues: * + Fixed ast_sip_session_create_outgoing() not setting up the + configured endpoint id as the new channel's caller id. This is + why party A got default connected line information. * Made + update_initial_connected_line() use the channel's CALLERID(id) + information. The core, app_dial, or predial routine may have + filled in or changed the endpoint caller id information. * Fixed + chan_pjsip_new() not setting the full party id information + available on the caller id and ANI party id. This includes the + configured callerid_tag string and other party id fields. * Fixed + accessing channel party id information without the channel lock + held. * Fixed using the effective connected line id without doing + a deep copy outside of holding the channel lock. Shallow copy + string pointers can become stale if the channel lock is not held. + * Made queue_connected_line_update() also update the channel's + CALLERID(id) information. Moving the channel to another bridge + would need the information there for the new bridge peer. * Fixed + off nominal memory leak in update_incoming_connected_line(). * + Added pjsip.conf callerid_tag string to party id information from + enabled trust_inbound endpoint in caller_id_incoming_request(). + AFS-98 #close Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3913/ ........ Merged + revisions 421400 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-18 21:10 +0000 [r421376] Damien Wedhorn <voip@facts.com.au> + + * channels/chan_skinny.c: Skinny: Fixup compile warning for non + dev-mode. + +2014-08-18 20:19 +0000 [r421337] George Joseph <george.joseph@fairview5.com> + + * funcs/func_config.c, /: func_config: Change 'Not Found' message + from ERROR to DEBUG When you call the CONFIG dialplan function + with the name of a variable that doesn't exist in the target + context you get an ERROR. This does nothing but clutter up the + logs with messages that may be perfectly acceptable. Just because + a variable wasn't in the context doesn't mean it's an error. + Maybei t's optional or just needs to be defaulted or ignored. + This patch changes the log level from ERROR to DEBUG. If a + dialplan developer wants to debug their dialplan they still canby + setting the console debug level as needed. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........ + Merged revisions 421327 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421328 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421329 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-18 01:13 +0000 [r421230-421312] Matthew Jordan <mjordan@digium.com> + + * res/ari/resource_channels.c: res/ari/resource_channels: Fix + compilation issue Forgot a parameter. Whoops. + + * res/ari/resource_channels.c: res/ari/resource_channels: Don't + return allocation failure on failed function If a function fails + to execute, it is most likely due to one of two reasons: (1) The + function doesn't exist or can't be read from (2) The function is + dangerous and is restricted based on the user's permissions + Currently we return allocation failure, which is incorrect. This + updates the reason code to more accurately reflect why the + request failed. ASTERISK-24215 + + * /, apps/app_meetme.c: apps/app_meetme: Fix crash when publishing + MeetMe messages with no channel The same function, + meetme_stasis_generate_msg, handles creating and publishing + Stasis message both when there are channels in the MeetMe + conference and when there are no channels in the conference. When + the performance improvement was made to use cached snapshots, + this created a situation where Asterisk would crash: obtaining a + cached snapshot is not NULL tolerant. This patch restores the + previous implementation, which used a NULL safe set of routines + to produce a blob containing the channel snapshot (if available) + and information about the MeetMe conference. ASTERISK-24234 + #close Reported by: Shaun Ruffell Tested by: Shaun Ruffell + ........ Merged revisions 421270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_dial.c, /: apps/app_dial: Fix Dial 'z' option The 'z' + option is supposed to disable the dial timeout in the case of a + call forward. Unfortunately, the wrong timeout timer was passed + to the do_forward function, resulting in the option not working. + ASTERISK-24225 #close Reported by: dimitripietro Tested by: + dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by + rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by + rmudgett (License 5621) ........ Merged revisions 421232 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421233 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421234 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, configure.ac: configure: Undefine FORTIFY_SOURCE + prior to defining it for patched gcc Some distributions of Linux + patch gcc to define FORTIFY_SOURCE when gcc is executed with + optimization. This "help" unfortunately results in re-definition + warnings when FORTIFY_SOURCE is later defined in Asterisk's build + system. This patch undefines FORTIFY_SOURCE prior to defining it + to prevent this warning. Review: + https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close + Reported by: Kilburn Tested by: Kilburn, wdoekes patches: + 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by + cloos (License 5956) 11.diff uploaded by cloos (License 5956) + 12.diff uploaded by cloos (License 5956) 13.diff uploaded by + cloos (License 5956) ........ Merged revisions 421227 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421228 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421229 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-17 16:10 +0000 [r421210] Joshua Colp <jcolp@digium.com> + + * res/res_http_websocket.c: res_http_websocket: Include query + parameters in client connection requests. Review: + https://reviewboard.asterisk.org/r/3914/ + +2014-08-15 17:08 +0000 [r421187] Jonathan Rose <jrose@digium.com> + + * main/channel.c, /: Bridging: Fix a behavioral change when + checking if a channel is leaving a bridge r420934 introduced some + failures in the test suite. Upon investigating, it was discovered + that differences in the way we were evaluating whether a channel + was in the process of leaving a bridge were causing some + reinvites not to occur (mostly reinvites back to Asterisk when + ending a call). This patch fixes that behavioral change. + ASTERISK-24027 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3910/ ........ Merged + revisions 421186 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-15 15:45 +0000 [r421042-421166] Matthew Jordan <mjordan@digium.com> + + * apps/app_voicemail.c, /, main/app.c: app_voicemail/app: Remove + test events that were duplicated by r421059 Moving the test event + raised when a file is played back (which occurred in r421059) + broke the ever loving snot out of the voicemail tests. This + caused duplicate test events to get raised, as app_voicemail and + main/app were raising events prior to call ast_streamfile. The + voicemail tests did not enjoy getting multiple events. Since + raising the playback event in ast_streamfile is far more useful + to the vast majority of tests, this patch keeps the call there + and simply removes the extraneous calls that duplicated the + event. ........ Merged revisions 421125 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421164 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421165 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_hep_rtcp.c, /: res/res_hep_rtcp: Remove dependency on + PJSIP The res_hep_rtcp module was incorrectly including + <pjsip.h>. This didn't need to be included, as the module does + not using PJPROJECT any fashion. Unfortunately, because + res_hep_rtcp did not include pjsip in its MODULEINFO as a + dependency, this also meant that res_hep_rtcp will fail to + compile on a system without PJPROJECT. This patch removes the + include. Thanks to Damien Wedhorn for pointing this out in + #asterisk-dev. ASTERISK-24236 #close Reported by: Damien Wedhorn, + Matt Jordan Tested by: Damien Wedhorn ........ Merged revisions + 421064 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/file.c, main/app.c: main/file: Move test event to emit + PLAYBACK event more consistently This is being done in advance of + the test for ASTERISK-23953 ........ Merged revisions 421059 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421060 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421061 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c, main/cel.c, /: cel: Make sure channels in extra + fields include their unique IDs as well CEL typically tracks a + lot of information using the unique ID of the channel. This is + typically needed due to tying events together using the linked ID + of the various channels involved in a "call", which is derived + from the channel ID of the oldest channel involved in a bridge + (or in the case of a Dial, the parent channel). Previously, we + had updated the extra fields to include the involved channel + names, but forgot to put in the unique ID. This patch corrects + that error. ........ Merged revisions 421037 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-14 16:32 +0000 [r420957-421010] Richard Mudgett <rmudgett@digium.com> + + * /, res/ari/resource_channels.c: ARI: Originate to app local + channel subscription code optimization. Reduce the scope of + local_peer and only get it if the ARI originate is subscribing to + the channels. Review: https://reviewboard.asterisk.org/r/3905/ + ........ Merged revisions 421009 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel_internal_api.c, main/channel.c: + channel_internal_api.c: Replace some code with ao2_replace(). Use + ao2_replace() instead of ao2_cleanup(); ao2_bump(). ao2_replace() + has the advantange of not altering the ref count if the replaced + pointer is the same. Review: + https://reviewboard.asterisk.org/r/3904/ + + * /, res/res_pjsip_send_to_voicemail.c: + res_pjsip_send_to_voicemail.c: Fix svn file properties. ........ + Merged revisions 420956 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-13 16:53 +0000 [r420950] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip.c, /: PJSIP: Prevent crash no-URI contacts This + prevents a crash from occurring when a contact with no URI is + used for the creation of an outbound out-of-dialog request with + no associated endpoint. ........ Merged revisions 420949 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-13 16:07 +0000 [r420940] Jonathan Rose <jrose@digium.com> + + * main/bridge_after.c, main/channel_internal_api.c, + include/asterisk/channel.h, apps/app_chanspy.c, + apps/app_mixmonitor.c, apps/app_stack.c, main/bridge_channel.c, + main/channel.c, main/pbx.c, /, main/framehook.c: Bridges: Fix + feature interruption/unintended kick caused by external actions + If a manager or CLI user attached a mixmonitor to a call running + a dynamic bridge feature while in a bridge, the feature would be + interrupted and the channel would be forcibly kicked out of the + bridge (usually ending the call during a simple 1 to 1 call). + This would also occur during any similar action that could set + the unbridge soft hangup flag, so the fix for this was to remove + unbridge from the soft hangup flags and make it a separate thing + all together. ASTERISK-24027 #close Reported by: mjordan Review: + https://reviewboard.asterisk.org/r/3900/ ........ Merged + revisions 420934 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-13 14:24 +0000 [r420919] Kinsey Moore <kmoore@digium.com> + + * main/manager.c: AMI: Improve documentation for Status action + +2014-08-13 07:52 +0000 [r420899] Walter Doekes <walter+asterisk@wjd.nu> + + * /, main/logger.c: logger: Don't store verbose-magic in the log + files. In r399267, the verbose2magic stuff was edited. This time + it results in magic characters in the log files for multiline + messages. In trunk (and 13) this was fixed by the "stripping" of + those characters from multiline messages (in r414798). This fix + is altered to actually strip the characters and not replace them + with blanks. Review: https://reviewboard.asterisk.org/r/3901/ + Review: https://reviewboard.asterisk.org/r/3902/ ........ Merged + revisions 420897 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420898 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-12 23:43 +0000 [r420879-420881] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_sip.c: chan_sip: Fix type mismatch when the format + is changed. Symptom is most likely an invalid ao2 object bad + magic number message or a less likely crash. + + * res/res_stasis_snoop.c: res_stasis_snoop.c: Fix off nominial exit + path leaving Snoop channel locked and not hungup. * Made use + ast_copy_string() instead of strcpy() for snoop uniqueid for + safety. There is no guarantee that the max channel uniqueid + length will remain the same as the snoop uniqueid space. + +2014-08-12 11:17 +0000 [r420856] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: app_voicemail: Fix the + "test_voicemail_vm_info" unit test. + +2014-08-11 20:53 +0000 [r420837] Richard Mudgett <rmudgett@digium.com> + + * res/stasis/command.c, /: res/stasis/command.c: Fix recent commit + using spaces instead of tabs. ........ Merged revisions 420836 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 18:50 +0000 [r420808] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json: AMI/ARI: Update version to + 2.5.0/1.5.0 respectively This is to support the backwards + compatible changes made in the next version of Asterisk. ........ + Merged revisions 420805 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 18:46 +0000 [r420796-420803] Kinsey Moore <kmoore@digium.com> + + * /, res/res_stasis.c: Stasis: Use the correct return value Return + the correct value instead of always returning 0 when setting + internal status on unreal channels. Reported by: Richard Mudgett + ........ Merged revisions 420802 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis.c, res/ari/resource_bridges.c, /, + res/stasis/stasis_bridge.c, include/asterisk/stasis_app.h: + Stasis: Allow internal channels directly into bridges The patch + to catch channels being shoehorned into Stasis() via external + mechanisms also happens to catch Announcer and Recorder channels + because they aren't known to be stasis-controlled channels in the + usual sense. This marks those channels as Stasis()-internal + channels and allows them directly into bridges. Review: + https://reviewboard.asterisk.org/r/3903/ ........ Merged + revisions 420795 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 18:32 +0000 [r420758-420794] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/stasis_app.h, main/stasis_channels.c, + res/ari/resource_channels.c, CHANGES, res/res_pjsip_pubsub.c, + main/manager_channels.c, apps/app_dial.c, res/stasis/app.c, + res/stasis/control.c: Improve call forwarding reporting, + especially with regards to ARI. This patch addresses a few + issues: 1) The order of Dial events have been changed when + performing a call forward. The order has now been altered to 1) + Dial begins dialing channel A. 2) When A forwards the call to B, + we issue the dial end event to channel A, indicating the dial is + being canceled due to a forward to B. 3) When the call to channel + B occurs, we then issue a new dial begin to channel B. 2) Call + forwards are now reported on the calling channel, not the peer + channel. 3) AMI DialEnd events have been altered to display the + extension the call is being forwarded to when relevant. 4) You + can now get the values of channel variables for channels that are + not currently in the Stasis application. This brings the + retrieval of channel variables more in line with the rest of + channel read operations since they may be performed on channels + not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan + ASTERISK-24138 #close Reported by Matt Jordan Patches: + forward-shenanigans.diff uploaded by Matt Jordan (License #6283) + Review: https://reviewboard.asterisk.org/r/3899 + + * res/res_pjsip_pubsub.c: Fix crashing unit tests with regards to + RLS. The unit tests require a sorcery.conf file that has been set + up to store resource lists in memory rather than retrieving from + configuration. With a setup that is not conducive to running the + tests, a fault in sorcery currently causes Asterisk to crash when + attempting to run any of the tests. To get around the crash, this + adds a function that verifies the current environment and marks + the tests as "not run" if the setup is not correct. + + * res/res_pjsip_pubsub.c: Fix crash encountered by the testsuite. + Running testsuite tests locally produced no errors, but when run + using the continuous integration framework, crashes occurred. The + crashes occurred due to a refcounting error that had been fixed + for a similar situation. + +2014-08-11 13:57 +0000 [r420742] Matthew Jordan <mjordan@digium.com> + + * res/res_hep.c, res/res_hep_pjsip.c, res/res_hep_rtcp.c: res_hep: + Remove disabling of modules These modules were originally + specified as being disabled, as they were introduced midstream in + Asterisk 12. That makes it nicer for folks who are upgrading to a + new release in the middle of Asterisk 12. That's not the case for + Asterisk 13: it's a brand new release. There's no reason to have + the modules disabled by default in that case. + +2014-08-11 10:40 +0000 [r420657-420717] Walter Doekes <walter+asterisk@wjd.nu> + + * /, main/utils.c: general: Fix memory Corruption in + __ast_string_field_ptr_build_va. If the space left in a + stringfield is between 0 and + (alignof(ast_string_field_allocation)-1) adding new data would + cause memory corruption, because we would assume enough space + (unsigned underrun). Thanks Arnd Schmitter for reporting and + finding out the cause! ASTERISK-23508 #close Reported by: Arnd + Schmitter Tested by: Arnd Schmitter, JoshE Review: + https://reviewboard.asterisk.org/r/3898/ ........ Merged + revisions 420680 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420715 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420716 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode. + ........ Merged revisions 420654 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420655 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420656 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 01:31 +0000 [r420607-420639] Matthew Jordan <mjordan@digium.com> + + * funcs/func_jitterbuffer.c: funcs/func_jitterbuffer: Tweak + documentation This patch merely reformats and cleans up a bit of + the jitterbuffer documentation for the wiki. + + * UPGRADE.txt, configs/samples/extconfig.conf.sample, CHANGES, + apps/app_queue.c, + contrib/ast-db-manage/config/versions/d39508cb8d8_create_queue_rules.py + (added), configs/samples/queuerules.conf.sample: app_queue: Add + RealTime support for queue rules This patch gives the optional + ability to keep queue rules in RealTime. It is important to note + that with this patch: (a) Queue rules in RealTime are only + examined on module load/reload (b) Queue rules are loaded both + from the queuerules.conf file as well as the RealTime backend To + inform app_queue to examine RealTime for queue rules, a new + setting has been added to queuerules.conf's general section + "realtime_rules". RealTime queue rules will only be used when + this setting is set to "yes". The schema for the database table + supports a rule_name, time, min_penalty, and max_penalty columns. + min_penalty and max_penalty can be relative, if a '-' or '+' + literal is provided. Otherwise, the penalties are treated as + constants. For example: rule_name, time, min_penalty, max_penalty + 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', + '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', + '4564', '46546' 'test_rule', '40', '15', '50' which would result + in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY + to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 + seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust + QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust + QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - + After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust + QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust + QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 + Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to + 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the + queue rules will be always reloaded on a module reload, even if + the underlying file did not change. With the option disabled, the + rules will only be reloaded if the file was modified. Review: + https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close + Reported by: Michael K patches: app_queue.c_realtime_trunk.patch + uploaded by Michael K (License 6621) + + * CHANGES: Update CHANGES file + + * UPGRADE.txt: Update UPGRADE.txt file + +2014-08-08 20:08 +0000 [r420577-420592] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: Fix build in devmode. + + * CHANGES, configs/samples/voicemail.conf.sample, + apps/app_voicemail.c: app_voicemail: Add the ability to specify + multiple email addresses. ASTERISK-24045 Reported by: Jacob + Barber Review: https://reviewboard.asterisk.org/r/3833/ + +2014-08-08 17:53 +0000 [r420534-420562] Matthew Jordan <mjordan@digium.com> + + * channels/chan_sip.c, channels/sip/security_events.c, + channels/sip/dialplan_functions.c, channels/sip/reqresp_parser.c, + channels/sip/route.c, channels/sip/utils.c, + channels/sip/config_parser.c: chan_sip: Mark chan_sip and its + files as extended support + + * rest-api-templates/make_ari_stubs.py: make_ari_stubs: Update wiki + prefix to '13' + + * rest-api-templates/res_ari_resource.c.mustache: + res_ari_resource.c.mustache: Update template to emit module + support level + + * main/message.c, /: main/message: remove debug message ........ + Merged revisions 420533 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-08 03:03 +0000 [r420514] Kinsey Moore <kmoore@digium.com> + + * tests/test_cel.c, /: CEL: Update unit tests for additional + information This updates the CEL unit tests for the new + information contained in the attended transfer CEL extra field. + ........ Merged revisions 420513 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-08 01:31 +0000 [r420494-420496] Matthew Jordan <mjordan@digium.com> + + * UPGRADE.txt: Update UPGRADE file for 13 branch + + * /: Remove old properties + + * / (added): ___ _ _ _ __ _____ / _ \ | | (_) | | / ||____ | / /_\ + \___| |_ ___ _ __ _ ___| | __ `| | / / | _ / __| __/ _ | '__| / + __| |/ / | | \ \ | | | \__ | || __| | | \__ | < _| |.___/ / \_| + |_|___/\__\___|_| |_|___|_|\_\ \___\____/ + +2014-08-07 21:58 +0000 [r420437] Richard Mudgett <rmudgett@digium.com> + + * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and + resolve the large SDP poll issue. Replace sip_tls_read() and + sip_tcp_read() with a single function and resolve the poll/wait + issue with large SDP payloads. ASTERISK-18345 #close Reported by: + Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) + patch uploaded by Elazar Broad Review: + https://reviewboard.asterisk.org/r/3882/ ........ Merged + revisions 420434 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420435 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420436 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-07 21:17 +0000 [r420389-420415] Kinsey Moore <kmoore@digium.com> + + * main/stasis_bridges.c, /: Stasis: Correct blind transfer message + generation This fixes the json object creation format string and + key name for the BridgeBlindTransfer Stasis event allowing it to + be published properly. ........ Merged revisions 420414 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_bridges.c, /: Stasis: Ensure transfer messages follow + validation rules This makes Stasis() event generation for + transfer messages follow validation rules. Currently, + ast_json_null() is being used in place of omitting a key entirely + which falls afoul of these validation rules. + https://reviewboard.asterisk.org/r/3892/ ........ Merged + revisions 420408 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c: Fix build in dev mode + +2014-08-07 19:44 +0000 [r420384-420388] Mark Michelson <mmichelson@digium.com> + + * /, main/bridge.c: Ensure bridges exist when trying to determine + bridged parties when publishing transfer information. ........ + Merged revisions 420387 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/strings.c, include/asterisk/res_pjsip_presence_xml.h, + res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c, + res/res_pjsip_xpidf_body_generator.c, include/asterisk/strings.h, + res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pidf_body_generator.c: Add support for RFC 4662 + resource list subscriptions. This commit adds the ability for a + user to configure a resource list in pjsip.conf. Subscribing to + this list simultaneously subscribes the subscriber to all + resources listed. This has the potential to reduce the amount of + SIP traffic when loads of subscribers on a system attempt to + subscribe to each others' states. + +2014-08-07 18:51 +0000 [r420364] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/format_compatibility.h, + channels/iax2/format_compatibility.c, + channels/iax2/include/codec_pref.h, main/format_compatibility.c, + channels/chan_iax2.c, channels/iax2/codec_pref.c, + channels/iax2/include/format_compatibility.h: chan_iax2: Several + media format fixes. * Fixed the iax.conf bandwidth option. This + is the root cause of ASTERISK-24150. * Added checks in + iax2_request() to ensure that there are actual formats requested + for the new channel to prevent any more fracks from issues like + ASTERISK-24150. This is a consequence of the iax.conf bandwidth + option not working. * Fixed struct iax2_codec_pref.order member + size mismatch issue when converting to and from the codec + preference order list passed over the wire. In addition the + values sent over the wire are now compatible with previous + Asterisk versions. * Fixed several issues dealing with the struct + iax2_codec_pref members. Off-by-one, array limit errors, and the + order/framing members always need to be updated together. * Made + iax2_request() setup the channel's native format preference order + according to the user's wishes. The new media format strategy + needs the order specified earler. * Fixed usage of + ast_format_compatibility_bitfield2format(). The function can + return NULL if the bitfield was not associated with a function. * + Deleted dead code iax2_codec_pref_getsize() and + iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and + iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of + inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH, + IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants + again as they were in Asterisk v1.8. * Renamed prefs to + prefs_global so it won't get confused with the local pref + versions. * Fixed too small buffer in + handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in + handle_cli_iax2_show_peer() to output complete lines. * Changed + struct create_addr_info.prefs to be struct iax2_codec_pref as an + optimization so iax2_request() and iax2_call() do less work. * + Fixed a potential deadlock in ast_iax2_new() on an off-nominal + path when the pbx could not get started. * Made set_config() + setup a local prefs list along side the local capability format + bitfield. Once the config is loaded, then the local copies are + put into the global versions. * Fix unininialized codec_buf in + function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott + Griepentrog Review: https://reviewboard.asterisk.org/r/3890/ + +2014-08-07 15:30 +0000 [r420338] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/bridge_features.h, res/res_stasis.c, + res/stasis/command.c, rest-api/api-docs/events.json, /, + res/stasis/app.c, res/stasis/control.c, main/bridge.c, + main/bridge_basic.c, res/stasis/stasis_bridge.c, + include/asterisk/stasis_bridges.h, res/stasis/command.h, + include/asterisk/stasis_app.h, res/stasis/app.h, + res/stasis/control.h, apps/app_queue.c, + res/ari/ari_model_validators.c, main/cel.c, + main/stasis_bridges.c, res/ari/ari_model_validators.h, + main/channel.c, include/asterisk/datastore.h, tests/test_cel.c: + Stasis: Convey transfer information to applications This fixes a + class of issues where Stasis applications were not made aware + that their channels were being manipulated or replaced by + external entitiessuch as transfers, AMI commands, or dialplan + applications such as Bridge(). Inconsistent information such as + StasisEnd events with unknown channels as a result of masquerades + has also been corrected. To accomplish these fixes, several new + fields were added to blind and attended transfer messages as well + as StasisStart and BridgeAttendedTransfer Stasis events. + ASTERISK-23941 #close Review: + https://reviewboard.asterisk.org/r/3865/ Review: + https://reviewboard.asterisk.org/r/3857/ Review: + https://reviewboard.asterisk.org/r/3852/ Review: + https://reviewboard.asterisk.org/r/3816/ Review: + https://reviewboard.asterisk.org/r/3731/ Review: + https://reviewboard.asterisk.org/r/3729/ Review: + https://reviewboard.asterisk.org/r/3728/ ........ Merged + revisions 420325 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-07 14:37 +0000 [r420314-420315] Joshua Colp <jcolp@digium.com> + + * include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pubsub.exports.in, res/res_pjsip_publish_asterisk.c + (added), res/res_pjsip_pubsub.c: res_pjsip_publish_asterisk: Add + support for exchanging device and mailbox state using SIP. This + module uses the inbound and outbound PUBLISH support to exchange + device and mailbox state between Asterisk instances. Each + instance is configured to publish to the other and requires no + intermediary server. The functionality provided is similar to the + XMPP and Corosync support. Review: + https://reviewboard.asterisk.org/r/3780/ + + * include/asterisk/res_pjsip_outbound_publish.h (added), + res/res_pjsip_outbound_publish.exports.in (added), + res/res_pjsip_outbound_publish.c (added): + res_pjsip_outbound_publish: Add module which provides outbound + PUBLISH support. This module implements the core parts required + for doing outbound PUBLISH. It takes care of configuration, + lifetime management, and authentication. Additional modules + implement the specific events that are published. Review: + https://reviewboard.asterisk.org/r/3780/ + +2014-08-07 14:17 +0000 [r420289-420309] Matthew Jordan <mjordan@digium.com> + + * main/pbx.c: pbx: Filter out pattern matching hints in responses + sent to ExtensionStateList Hints that are a pattern match are + technically stored in the hint container in the same fashion as + concrete implementations of hints. The pattern matching hints, + however, are not "real" in the sense that things can subscribe to + them: rather, they are stored in the hints container so that when + a subscription is made a "real" hint can be generated for the + subscription if one does not yet exist. The extension state core + takes care of this correctly by matching against non-pattern + matching extensions prior to pattern matching extensions. Because + of this, however, the ExtensionStateList AMI action was returning + pattern matching hints when executed. These hints are meaningless + from the perspective of AMI clients: their state will never + change, they cannot be subscribed to, and events would never + normally be generated from them. As such, we now filter these out + of the response. + + * build_tools/post_process_documentation.py: build_tools: Skip + managerEvent combining for AMI action responses AMI action + responses can (and will) reference AMI events that they return. + These event references and definitions should not be combined + with AMI events raised elsewhere in the code, as they are + specifically tied to the AMI action that raised them. + ASTERISK-24156 #close Reported by: Rusty Newton + +2014-08-06 18:12 +0000 [r420212-420237] Richard Mudgett <rmudgett@digium.com> + + * contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py, + /: Fix alembic script to work properly in offline mode. When run + in offline mode, this would attempt to check the database for the + presence of a type it was going to try to create. I now check the + context to see if we're running in offline mode and change a + parameter accordingly. ........ Merged revisions 407567 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py + (added), /: Add alembic script that adds contact user_agent and + endpoint message_context. ........ Merged revisions 411514 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py + (added), /, + contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py, + contrib/ast-db-manage/config.ini.sample, + contrib/ast-db-manage/config/versions/1758e8bbf6b_increase_useragent_column_size.py + (added), + contrib/ast-db-manage/config/versions/5139253c0423_make_q_member_uniqueid_autoinc.py + (added), contrib/ast-db-manage/cdr.ini.sample, + contrib/ast-db-manage/voicemail.ini.sample: alembic: Adjust + sippeers, queue_members, and voicemail_messages tables. * + Increased the sippeers useragent max string size to 255. * + Changed the queue_members uniqueid to an auto incremented integer + instead of a string. * Increased the voicemail_messages BLOB size + to LONGBLOB on mysql. * Fixed the add_tables_for_pjsip config + change version downgrade actions to drop a table it created. * + Adjusted the sample alembic.ini files cdr.ini.sample, + config.ini.sample, and voicemail.ini.sample to give a mysql and + postgres sqlalchemy.url lines. ASTERISK-23847 #close Reported by: + Stephen More ASTERISK-23825 #close Reported by: Stephen More + ASTERISK-23909 #close Reported by: Stephen More Review: + https://reviewboard.asterisk.org/r/3870/ ........ Merged + revisions 420211 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-06 16:12 +0000 [r420149] George Joseph <george.joseph@fairview5.com> + + * /, pbx/pbx_lua.c, main/pbx.c: pbx_lua: fix regression with global + sym export and context clash by pbx_config. ASTERISK-23818 (lua + contexts being overwritten by contexts of the same name in + pbx_config) surfaced because pbx_lua, having the + AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before + pbx_config. Since I couldn't find any reason for pbx_lua to + export it's symbols to the rest of Asterisk, I simply changed the + flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't + realize was that the symbols need to be exported not because + Asterisk needs them but because any external Lua modules like + luasql.mysql need the base Lua language APIs exported + (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's + an issue in pbx.c where context_merge was only merging includes, + switches and ignore patterns if the context was already existing + AND has extensions, or if the context was brand new. If pbx_lua + is loaded before pbx_config, the context will exist BUT pbx_lua, + being implemented as a switch, will never place extensions in it, + just the switch statement. The result is that when pbx_config + loads, it never merges the switch statement created by pbx_lua + into the final context. This patch sets pbx_lua's modflag back to + AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge + that catches the case where an existing context has includes, + switchs or ingore patterns but no actual extensions. + ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo + Teräs Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3891/ ........ Merged + revisions 420146 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420147 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420148 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-06 15:32 +0000 [r420144] Walter Doekes <walter+asterisk@wjd.nu> + + * funcs/func_channel.c: Add documentation to the ability to + retrieve the source port of a SIP call. (belongs with r419970) + ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by + dtryba Review: https://reviewboard.asterisk.org/r/3781/ + +2014-08-06 12:55 +0000 [r420124] Kinsey Moore <kmoore@digium.com> + + * configs/samples/stasis.conf.sample (added), main/named_acl.c, + apps/app_queue.c, main/stasis_bridges.c, main/loader.c, + main/stasis.c, apps/app_forkcdr.c, main/stasis_message.c, + funcs/func_cdr.c, res/res_corosync.c, res/res_stun_monitor.c, + res/res_stasis_test.c, res/res_stasis.c, apps/app_chanspy.c, + main/stasis_cache.c, main/pickup.c, main/security_events.c, + include/asterisk/stasis.h, main/devicestate.c, main/core_local.c, + res/res_stasis_snoop.c, main/endpoints.c, main/presencestate.c, + main/cdr.c, main/channel.c, main/stasis_system.c, main/manager.c, + main/test.c, main/file.c, main/app.c, pbx/pbx_realtime.c, + main/stasis_channels.c, tests/test_stasis.c, + res/parking/parking_manager.c, main/stasis_endpoints.c, + main/rtp_engine.c, main/ccss.c, main/bridge.c, + tests/test_stasis_channels.c: Stasis: Allow message types to be + blocked This introduces stasis.conf and a mechanism to prevent + certain message types from being published. Internally, this + works by preventing the chosen message types from being created + which ensures that those message types can never be published. + This patch also adjusts message publishers such that message + payloads are not created if the related message type is not + available. ASTERISK-23943 #close Review: + https://reviewboard.asterisk.org/r/3823/ + +2014-08-05 21:48 +0000 [r420098-420100] Matthew Jordan <mjordan@digium.com> + + * res/stasis/messaging.c, /: stasis: Fix compilation issue with ao2 + tagged objects ........ Merged revisions 420099 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /, + channels/chan_sip.c, res/stasis/app.c, res/stasis/messaging.h + (added), res/ari/resource_endpoints.h, res/res_pjsip_messaging.c, + tests/test_message.c (added), res/res_xmpp.c, + include/asterisk/json.h, CHANGES, include/asterisk/manager.h, + res/ari/ari_model_validators.c, res/ari/ari_model_validators.h, + main/json.c, res/res_ari_endpoints.c, include/asterisk/message.h, + res/ari/resource_channels.c, main/message.c, res/res_stasis.c, + res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json: + Multiple revisions 420089-420090,420097 ........ r420089 | + mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines + ARI: Add channel technology agnostic out of call text messaging + This patch adds the ability to send and receive text messages + from various technology stacks in Asterisk through ARI. This + includes chan_sip (sip), res_pjsip_messaging (pjsip), and + res_xmpp (xmpp). Messages are sent using the endpoints resource, + and can be sent directly through that resource, or to a + particular endpoint. For example, the following would send the + message "Hello there" to PJSIP endpoint alice with a display URI + of sip:asterisk@mycooldomain.org: + ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There + This is equivalent to the following as well: + ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There + Both forms are available for message technologies that allow for + arbitrary destinations, such as chan_sip. Inbound messages can + now be received over ARI as well. An ARI application that + subscribes to endpoints will receive messages from those + endpoints: { "type": "TextMessageReceived", "timestamp": + "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": + "PJSIP", "resource": "alice", "state": "online", "channel_ids": + [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", + "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", + "variables": [] }, "application": "testsuite" } The above was + made possible due to some rather major changes in the message + core. This includes (but is not limited to): - Users of the + message API can now register message handlers. A handler has two + callbacks: one to determine if the handler has a destination for + the message, and another to handle it. - All dialplan + functionality of handling a message was moved into a message + handler provided by the message API. - Messages can now have the + technology/endpoint associated with them. Various other + properties are also now more easily accessible. - A number of ao2 + containers that weren't really needed were replaced with vectors. + Iteration over ao2_containers is expensive and pointless when the + lifetime of things is well defined and the number of things is + very small. res_stasis now has a new file that makes up its + structure, messaging. The messaging functionality implements a + message handler, and passes received messages that match an + interested endpoint over to the app for processing. Note that + inadvertently while testing this, I reproduced ASTERISK-23969. + res_pjsip_messaging was incorrectly parsing out the 'to' field, + such that arbitrary SIP URIs mangled the endpoint lookup. This + patch includes the fix for that as well. Review: + https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close + Reported by: Matt Jordan ASTERISK-23969 #close Reported by: + Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 + -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties + :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, + 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing + compilation issue ........ Merged revisions 420089-420090,420097 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-05 13:59 +0000 [r420028] Jonathan Rose <jrose@digium.com> + + * main/format.c: chan_iax2: Fix a crash that occurs when using + allow=all for an IAX2 peer Or any combination of codecs that + includes Opus. ASTERISK-24107 #close Review: + https://reviewboard.asterisk.org/r/3885/ + +2014-08-04 21:00 +0000 [r420007] Richard Mudgett <rmudgett@digium.com> + + * main/format_cache.c, include/asterisk/format_cache.h: Remove + duplicate definitions of ast_format_vp8. + +2014-08-04 20:25 +0000 [r419970] Mark Michelson <mmichelson@digium.com> + + * channels/sip/dialplan_functions.c: Add the ability to retrieve + the source port of a SIP call. This adds the ability to call + CHANNEL(recvport) on chan_sip channels to see the port on which + an INVITE was received. ASTERISK-24040 #close Reported by dtryba + Patches: dialplan_functions.patch uploaded by dtryba (License + #6628) Review: https://reviewboard.asterisk.org/r/3781 + +2014-08-04 19:47 +0000 [r419945] Rusty Newton <rnewton@digium.com> + + * main/manager.c, /: Manager - Improve documentation for manager + commands Getvar and Setvar. The documentation for these commands + did not make it clear that they could accept expressions and + functions. Modified to make this clear, but tried not to be + overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton + Tested by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3854 ........ Merged revisions + 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 419943 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419944 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-02 03:37 +0000 [r419914] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip.c: Manager: Add PJSIPShowEndpoint[s] documentation + This adds a large swath of response documentation for + PJSIPShowEndpoint and PJSIPShowEndpoints AMI commands. It relies + heavily on the existing text in the configInfo documentation via + xi:include tags to avoid documentation duplication. Review: + https://reviewboard.asterisk.org/r/3888/ + +2014-08-01 14:48 +0000 [r419888] Mark Michelson <mmichelson@digium.com> + + * CHANGES, res/res_pjsip/pjsip_options.c: Add ContactStatusDetail + to PJSIPShowEndpoint AMI output. Now when running + PJSIPShowEndpoint, you will receive a ContactStatusDetail for + each bound contact that Asterisk is qualifying. This information + includes the URI of the contact, current reachability, and + roundtrip time. AFS-91 #close Reported by Mark Michelson Review: + https://reviewboard.asterisk.org/r/3797 + +2014-07-31 16:19 +0000 [r419851] Jonathan Rose <jrose@digium.com> + + * CHANGES, res/res_pjsip_notify.c: PJSIP: Send Notify AMI and CLI + commands can now send to URI instead of endpoint Review: + https://reviewboard.asterisk.org/r/3817/ + +2014-07-31 11:57 +0000 [r419822-419825] Matthew Jordan <mjordan@digium.com> + + * main/rtp_engine.c, /, res/res_hep_rtcp.c (added), CHANGES, + channels/chan_pjsip.c, res/res_rtp_asterisk.c: res_hep_rtcp: Add + module that sends RTCP information to a Homer Server This patch + adds a new module to Asterisk, res_hep_rtcp. The module + subscribes to the RTCP topics in Stasis and receives RTCP + information back from the message bus. It encodes into HEPv3 + packets and sends the information to the res_hep module for + transmission. Using this, someone with a Homer server can get + live call quality monitoring for all RTP-based channels in their + Asterisk 12+ systems. In addition, there were a few bugs in the + RTP engine, res_rtp_asterisk, and chan_pjsip that were uncovered + by the tests written for the Asterisk Test Suite. This patch + fixes the following: 1) chan_pjsip failed to set its channel + unique ids on its RTP instance on outbound calls. It now does + this in the appropriate location, in the serialized call + callback. 2) The rtp_engine was overflowing some values when + packed into JSON. Specifically, some longs and unsigned ints + can't be be packed into integer values, for obvious reasons. + Since libjansson only supports integers, floats, strings, + booleans, and objects, we print these values into strings. 3) + res_rtp_asterisk had a few problems: (a) it would emit a source + IP address of 0.0.0.0 if bound to that IP address. We now use + ast_find_ourip to get a better IP address, and properly marshal + the result into an ast_strdupa'd string. (b) Reports can be + generated with no report bodies. In particular, this occurs when + a sender is transmitting information to a receiver (who will send + no RTP back to the sender). As such, the sender has no report + body for what it received. We now properly handle this case, and + the sender will emit SR reports with no body. Likewise, if we + receive an RTCP packet with no report body, we will still + generate the appropriate events. ASTERISK-24119 #close ........ + Merged revisions 419823 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * funcs/func_jitterbuffer.c, doc/appdocsxml.dtd, main/xmldoc.c: + xmldocs: Add support for an <example> tag in the Asterisk XML + Documentation This patch adds support for an <example /> tag in + the XML documentation schema. For CLI help, this doesn't change + the formatting too much: - Preceeding white space is removed - + Unlike with para elements, new lines are preserved However, + having an <example /> tag in the XML schema allows for the wiki + documentation generation script to surround the documentation + with {code} or {noformat} tags, generating much better content + for the wiki - and allowing us to put dialplan examples (and + other code snippets, if desired) into the documentation for an + application/function/AMI command/etc. Review: + https://reviewboard.asterisk.org/r/3807/ + +2014-07-30 18:32 +0000 [r419806] Kinsey Moore <kmoore@digium.com> + + * main/manager.c, res/res_manager_presencestate.c, + res/res_manager_devicestate.c, main/pbx.c: manager: Add state + list commands This patch adds three new AMI commands: * + ExtensionStateList (pbx.c) - list all known extension state hints + and their current statuses. Events emitted by the list action are + equivalent to the ExtensionStatus events. * PresenceStateList + (res_manager_presencestate) - list all known presence state + values. Events emitted are generated by the stasis message type, + and hence are PresenceStateChange events. * DeviceStateList + (res_manager_devicestate) - list all known device state values. + Events emitted are generated by the stasis message type, and + hence are DeviceStateChange events. Patch-by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3799/ + +2014-07-29 19:41 +0000 [r419789] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: Do not omit the first header of a UserEvent AMI + action from the corresponding emitted UserEvent. ASTERISK-24124 + #close Reported by Matt Jordan AFS-131 #close Reported by Matt + Jordan Patches: userevent.patch uploaded by Matt Jordan (License + #6283) + +2014-07-29 10:56 +0000 [r419751-419766] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, /: res_pjsip_session: Fix race condition + where redirecting information may not be set. Since the PJSIP + INVITE session module is invoked before any session supplements + it was possible for it to handle a redirect before the + res_pjsip_diversion module interpreted and set redirecting + information on the channel. This would cause the redirecting + information to get lost. This patch ensures that session + supplements are *always* invoked before a redirect occurs by + explicitly calling them in the redirect handler. Review: + https://reviewboard.asterisk.org/r/3850/ ........ Merged + revisions 419764 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_xpidf_body_generator.c, + res/res_pjsip_pidf_body_generator.c: + res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator: + Ensure local entity is unquoted. The local entity as provided by + PJSIP is quoted within '<' and '>'. As a result placing this + value into XML will result in malformed XML being produced. This + patch now unquotes the local entity so it can go safely into the + XML. Review: https://reviewboard.asterisk.org/r/3851/ ........ + Merged revisions 419750 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-28 18:58 +0000 [r419688] Richard Mudgett <rmudgett@digium.com> + + * apps/app_speech_utils.c, main/channel.c, /, + funcs/func_frame_trace.c, main/abstract_jb.c: datastores: Audit + ast_channel_datastore_remove usage. Audit of v1.8 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leaks in app_speech_utils and func_frame_trace. * Fixed + app_speech_utils not locking the channel when accessing the + channel datastore list. Review: + https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leak in func_jitterbuffer. (Was not in v12) Review: + https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leaks in abstract_jb. * Fixed leak in + ast_channel_unsuppress(). Used by ARI mute control and + res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used + by ARI mute control and res_mutestream. Review: + https://reviewboard.asterisk.org/r/3861/ ........ Merged + revisions 419684 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 419685 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419686 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-25 18:09 +0000 [r419612] Joshua Colp <jcolp@digium.com> + + * main/loader.c: loader: Fix an infinite loop when printing modules + using "module show". When creating the alphabetical sorted list + each module is added to a list temporarily. On the second + iteration each module already has a pointer to another module, + causing stuff to go into a loop. ASTERISK-24123 #close Reported + by: Malcolm Davenport + +2014-07-25 16:47 +0000 [r419592] Mark Michelson <mmichelson@digium.com> + + * res/res_ari_sounds.c, res/res_stasis.c, res/res_fax_spandsp.c, + res/res_timing_kqueue.c, res/res_odbc.c, + res/res_pjsip_transport_websocket.c, apps/app_voicemail.c, + res/res_calendar.c, channels/chan_unistim.c, cel/cel_radius.c, + channels/chan_multicast_rtp.c, res/res_pjsip_notify.c, + res/res_snmp.c, formats/format_sln.c, apps/app_meetme.c, + apps/app_dictate.c, codecs/codec_gsm.c, res/res_stasis_snoop.c, + res/res_musiconhold.c, res/res_format_attr_h264.c, + res/res_http_websocket.c, apps/app_followme.c, + res/res_config_sqlite.c, formats/format_siren7.c, cdr/cdr_csv.c, + formats/format_ilbc.c, channels/chan_phone.c, + apps/app_setcallerid.c, apps/app_osplookup.c, cel/cel_custom.c, + apps/app_mp3.c, res/res_agi.c, channels/chan_motif.c, + res/res_timing_timerfd.c, apps/app_confbridge.c, + res/res_format_attr_silk.c, formats/format_siren14.c, + res/res_sorcery_realtime.c, channels/chan_mgcp.c, + apps/app_jack.c, codecs/codec_lpc10.c, + res/res_pjsip_pidf_body_generator.c, res/res_config_pgsql.c, + funcs/func_dialplan.c, apps/app_nbscat.c, cdr/cdr_syslog.c, + res/res_pjsip_authenticator_digest.c, apps/app_festival.c, + res/res_fax.c, apps/app_waitforsilence.c, res/res_adsi.c, + res/res_crypto.c, res/res_ari_applications.c, + res/res_hep_pjsip.c, pbx/pbx_lua.c, res/res_pjsip_messaging.c, + res/res_pjsip_caller_id.c, channels/chan_console.c, + apps/app_getcpeid.c, res/res_stasis_answer.c, + channels/chan_oss.c, res/res_pjsip_nat.c, + res/res_pjsip_session.c, cdr/cdr_tds.c, + res/res_pjsip_header_funcs.c, res/res_parking.c, + formats/format_vox.c, res/res_pjsip_rfc3326.c, + res/res_ari_endpoints.c, res/res_stun_monitor.c, + res/res_pjsip_mwi.c, res/res_stasis_recording.c, + res/res_pjsip_xpidf_body_generator.c, apps/app_sms.c, + codecs/codec_ulaw.c, channels/chan_nbs.c, apps/app_stack.c, + channels/chan_pjsip.c, formats/format_g729.c, cel/cel_pgsql.c, + res/res_sorcery_config.c, res/res_ari.c, addons/chan_ooh323.c, + cdr/cdr_sqlite3_custom.c, codecs/codec_adpcm.c, + res/res_ari_asterisk.c, res/res_calendar_caldav.c, + apps/app_image.c, apps/app_ices.c, formats/format_wav_gsm.c, + main/cli.c, res/res_format_attr_celt.c, res/res_rtp_multicast.c, + channels/chan_dahdi.c, funcs/func_pitchshift.c, res/res_smdi.c, + res/res_pjsip_one_touch_record_info.c, pbx/pbx_ael.c, + pbx/pbx_realtime.c, apps/app_amd.c, channels/chan_alsa.c, + formats/format_h263.c, apps/app_url.c, res/res_pjsip_acl.c, + apps/app_externalivr.c, res/res_curl.c, formats/format_gsm.c, + res/res_speech.c, cdr/cdr_manager.c, res/res_calendar_exchange.c, + codecs/codec_g722.c, res/res_pjsip_multihomed.c, + res/res_ari_mailboxes.c, cel/cel_tds.c, res/res_sorcery_memory.c, + apps/app_fax.c, codecs/codec_speex.c, res/res_pjsip_sdp_rtp.c, + codecs/codec_g726.c, formats/format_ogg_vorbis.c, + apps/app_talkdetect.c, res/res_ari_channels.c, + res/res_pjsip_exten_state.c, apps/app_speech_utils.c, + apps/app_agent_pool.c, apps/app_waitforring.c, res/res_statsd.c, + addons/cdr_mysql.c, formats/format_g726.c, res/res_ari_bridges.c, + addons/app_mysql.c, res/res_stasis_playback.c, + addons/format_mp3.c, res/res_pjsip_endpoint_identifier_ip.c, + res/res_phoneprov.c, res/res_pjsip_t38.c, + res/res_pjsip_registrar_expire.c, cdr/cdr_pgsql.c, + cdr/cdr_radius.c, res/res_chan_stats.c, + res/res_format_attr_opus.c, res/res_config_odbc.c, + funcs/func_audiohookinherit.c, + res/res_pjsip_outbound_registration.c, cel/cel_manager.c, + funcs/func_odbc.c, res/res_pjsip_endpoint_identifier_anonymous.c, + funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c, + apps/app_minivm.c, res/res_pjsip_log_forwarder.c, + formats/format_h264.c, res/res_config_ldap.c, apps/app_ivrdemo.c, + addons/chan_mobile.c, apps/app_stasis.c, + res/res_pjsip_diversion.c, cdr/cdr_custom.c, apps/app_adsiprog.c, + res/res_pjsip_dtmf_info.c, res/res_rtp_asterisk.c, + res/res_calendar_icalendar.c, res/res_hep.c, channels/chan_sip.c, + channels/chan_bridge_media.c, codecs/codec_alaw.c, + apps/app_queue.c, res/res_srtp.c, funcs/func_presencestate.c, + res/res_timing_pthread.c, res/res_manager_presencestate.c, + res/res_corosync.c, apps/app_celgenuserevent.c, + cel/cel_sqlite3_custom.c, res/snmp/agent.c, pbx/pbx_dundi.c, + formats/format_g723.c, funcs/func_devstate.c, + res/res_pjsip_registrar.c, + res/res_pjsip_pidf_eyebeam_body_supplement.c, + addons/res_config_mysql.c, + res/res_pjsip_pidf_digium_body_supplement.c, apps/app_test.c, + res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c, + apps/app_alarmreceiver.c, apps/app_chanisavail.c, + res/res_format_attr_h263.c, res/res_pjsip_mwi_body_generator.c, + res/res_xmpp.c, res/res_http_post.c, channels/chan_iax2.c, + res/res_pjsip_endpoint_identifier_user.c, res/res_pjsip.c, + res/res_pktccops.c, res/res_pjsip_send_to_voicemail.c, + main/loader.c, cel/cel_odbc.c, res/res_ari_model.c, + channels/chan_skinny.c, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_mwi_external.c, apps/app_skel.c, formats/format_pcm.c, + include/asterisk/module.h, res/res_pjsip_path.c, + res/res_ari_playbacks.c, res/res_pjsip_pubsub.c, cdr/cdr_odbc.c, + funcs/func_periodic_hook.c, res/res_stasis_test.c, + formats/format_jpeg.c, res/res_pjsip_refer.c, + formats/format_g719.c, res/res_clialiases.c, + res/res_config_sqlite3.c, res/res_ari_device_states.c, + formats/format_wav.c, apps/app_saycounted.c, apps/app_dahdiras.c, + apps/app_morsecode.c, res/res_stasis_mailbox.c, + res/res_ael_share.c, res/res_mwi_external_ami.c, + res/res_pjsip_logger.c, res/res_stasis_device_state.c, + res/res_calendar_ews.c, res/res_monitor.c, apps/app_playback.c, + res/res_ari_recordings.c, res/res_manager_devicestate.c, + res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c, + res/res_ari_events.c, res/res_pjsip_dialog_info_body_generator.c, + res/res_sorcery_astdb.c, codecs/codec_dahdi.c, + apps/app_zapateller.c, pbx/pbx_config.c: Add module support level + to ast_module_info structure. Print it in CLI "module show" . + ASTERISK-23919 #close Reported by Malcolm Davenport Review: + https://reviewboard.asterisk.org/r/3802 + +2014-07-25 14:47 +0000 [r419563-419567] Matthew Jordan <mjordan@digium.com> + + * CHANGES, res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, + res/ari/ari_model_validators.h, /, res/res_stasis_recording.c: + Multiple revisions 419565-419566 ........ r419565 | mjordan | + 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines ARI: + report duration values in LiveRecording objects This patch adds + three new fields to the LiveRecording model: - total_duration: + the total length of the live recording - talking_duration: + optional. The duration of talking energy that was detected while + the recording was made. - silence_duration: optional. The + duration of silence that was detected while the recording was + made. These values are reported in the RecordingFinished ARI + event. When a DSP is enabled on the channel during the recording + - which occurs when the recording is created with + max_silence_seconds (indicating that the user actually cares + about how much silence is in the file), we will report the + talking_duration and silence_duration in addition to the + total_duration. Review: https://reviewboard.asterisk.org/r/3770/ + ASTERISK-24037 #close Reported by: Samuel Galarneau ........ + r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) + | 1 line Update CHANGES for r419565 ........ Merged revisions + 419565-419566 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/loader.c, res/res_calendar.c: module loader: Unload modules + in reverse order of their start order When Asterisk starts a + module (calling its load_module function), it re-orders the + module list, sorting it alphabetically. Ostensibly, this was done + so that the output of 'module show' listed modules in alphabetic + order. This had the unfortunate side effect of making modules + with complex usage patterns unloadable. A module that has a large + number of modules that depend on it is typically abandoned during + the unloading process. This results in its memory not being + reclaimed during exit. Generally, this isn't harmful - when the + process is destroyed, the operating system will reclaim all + memory allocated by the process. Prior to Asterisk 12, we also + didn't have many modules with complex dependencies. However, with + the advent of ARI and PJSIP, this can make make unloading those + modules successfully nearly impossible, and thus tracking memory + leaks or ref debug leaks a real pain. While this patch is not a + complete overhaul of the module loader - such an effort would be + beyond the scope of what could be done for Asterisk 13 - this + does make some marginal improvements to the loader such that + modules like res_pjsip or res_stasis *may* be made properly + un-loadable in the future. 1. The linked list of modules has been + replaced with a doubly linked list. This allows traversal of the + module list to occur backwards. The module shutdown routine now + walks the global list backwards when it attempts to unload + modules. 2. The alphabetic reorganization of the module list on + startup has been removed. Instead, a started module is placed at + the end of the module list. 3. The ast_update_module_list + function - which is used by the CLI to display the modules - now + does the sorting alphabetically itself. It creates its own linked + list and inserts the modules into it in alphabetic order. This + allows for the intent of the previous code to be maintained. This + patch also contains a fix for res_calendar. Without + calendar.conf, the calendar modules were improperly bumping the + use count of res_calendar, then failing to load themselves. This + patch makes it so that we detect whether or not calendaring is + enabled before altering the use count. Review: + https://reviewboard.asterisk.org/r/3777/ + +2014-07-25 10:54 +0000 [r419537-419539] Joshua Colp <jcolp@digium.com> + + * apps/app_bridgewait.c, /: app_bridgewait: Remove possibility of + race condition between channels leaving/joining. Bridges created + by app_bridgewait previously had the "dissolve when empty" flag + set. This caused the bridge core to destroy them when the last + channel had left. This introduced a race condition where we may + have a reference to the bridge but it is not actually joinable + when we try to join it. This flag has now been removed and the + bridge is guaranteed to be joinable at all times. ASTERISK-23987 + #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3836/ ........ Merged + revisions 419538 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/bridge.c: bridge: Make "bridge destroy" only available in + developer mode and add "all" to "bridge kick". The "bridge + destroy" CLI command is invasive to bridges and can leave them in + an unexpected state for the users of them. Since this command may + be useful for developers it is now only available when developer + mode is available. To take its place "all" has been added as a + valid option to the "bridge kick" CLI command. It will kick all + of the channels in the bridge out. ASTERISK-23987 Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/ + ........ Merged revisions 419536 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-24 22:48 +0000 [r419520] Richard Mudgett <rmudgett@digium.com> + + * main/bridge.c, main/bridge_basic.c, main/core_unreal.c, + UPGRADE.txt, include/asterisk/channel.h, CHANGES, + apps/app_followme.c, apps/app_queue.c, main/cel.c, + res/parking/parking_bridge_features.c, apps/app_dial.c, + main/channel.c, main/dial.c, main/pbx.c: accountcode: Slightly + change accountcode propagation. The previous behavior was to + simply set the accountcode of an outgoing channel to the + accountcode of the channel initiating the call. It was done this + way a long time ago to allow the accountcode set on the SIP/100 + channel to be propagated to a local channel so the dialplan + execution on the Local;2 channel would have the SIP/100 + accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 + Propagating the SIP/100 accountcode to the local channels is very + useful. Without any dialplan manipulation, all channels in this + call would have the same accountcode. Using dialplan, you can set + a different accountcode on the SIP/200 channel either by setting + the accountcode on the Local;2 channel or by the Dial + application's b(pre-dial), M(macro) or U(gosub) options, or by + the FollowMe application's b(pre-dial) option, or by the Queue + application's macro or gosub options. Before Asterisk v12, the + altered accountcode on SIP/200 will remain until the local + channels optimize out and the accountcode would change to the + SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount + support but ultimately had to punt on the support. The + peeraccount support was rendered useless because of how the CDR + code needed to unconditionally force the caller's accountcode + onto the peer channel's accountcode. The CEL events were thus + intentionally made to always use the channel's accountcode as the + peeraccount value. With the arrival of Asterisk v12, the + situation has improved somewhat so peeraccount support can be + made to work. Using the indicated example, the the accountcode + values become as follows when the peeraccount is set on SIP/100 + before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> + SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: + 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already + has an accountcode it can only change by the following explicit + user actions: 1) A channel originate method that can specify an + accountcode to use. 2) The calling channel propagating its + non-empty peeraccount or its non-empty accountcode if the + peeraccount was empty to the outgoing channel's accountcode + before initiating the dial. e.g., Dial and FollowMe. The + exception to this propagation method is Queue. Queue will only + propagate peeraccounts this way only if the outgoing channel does + not have an accountcode. 3) Dialplan using CHANNEL(accountcode). + 4) Dialplan using CHANNEL(peeraccount) on the other end of a + local channel pair. If a channel does not have an accountcode it + can get one from the following places: 1) The channel driver's + configuration at channel creation. 2) Explicit user action as + already indicated. 3) Entering a basic or stasis-mixing bridge + from a peer channel's peeraccount value. You can specify the + accountcode for an outgoing channel by setting the + CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue + applications. Queue adds the wrinkle that it will not overwrite + an existing accountcode on the outgoing channel with the calling + channels values. Accountcode and peeraccount values propagate to + an outgoing channel before dialing. Accountcodes also propagate + when channels enter or leave a basic or stasis-mixing bridge. The + peeraccount value only makes sense for mixing bridges with two + channels; it is meaningless otherwise. * Made peeraccount + functional by changing accountcode propagation as described + above. * Fixed CEL extracting the wrong ie value for the + peeraccount. This was done intentionally in Asterisk v1.8 when + that version had to punt on peeraccount. * Fixed a few places + dealing with accountcodes that were reading from channels without + the lock held. AFS-65 #close Review: + https://reviewboard.asterisk.org/r/3601/ + +2014-07-24 21:01 +0000 [r419504] Michael L. Young <elgueromexicano@gmail.com> + + * main/db.c, include/asterisk/astdb.h: core/db: Revert Patch Added + In Attempt To Improve I/O Performance Reverting the patch since + it was causing a regression and after fixing the regression, + there were no performance gains. At least based on my method for + measurement. ASTERISK-24050 Review: + https://reviewboard.asterisk.org/r/3841/ + +2014-07-24 17:50 +0000 [r419438-419439] Corey Farrell <git@cfware.com> + + * include/asterisk/astobj.h: Deprecate astobj.h This flags astobj.h + as deprecated, warns people to use astobj2.h instead. Only + netsock.c (also deprecated) still uses astobj.h. ASTERISK-24069 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3818/ + + * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: + complete upgrade to ao2 This change upgrades sip_registry and + sip_subscription_mwi to astobj2. ASTERISK-24067 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3759/ + +2014-07-24 16:52 +0000 [r419377] Jason Parker <jparker@digium.com> + + * addons/chan_ooh323.c, /: Don't cause Asterisk to exit if + ooh323.conf not found. (closes issue ASTERISK-23814) ........ + Merged revisions 419374 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 419375 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419376 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-24 15:20 +0000 [r419358] Matthew Jordan <mjordan@digium.com> + + * main/devicestate.c, channels/chan_pjsip.c: device state: Update + the core to report ONHOLD if a channel is on hold In Asterisk, it + is possible for a device to have a status of ONHOLD. This is not + typically an easy thing to determine, as a channel being on hold + is not a direct channel state. Typically, this has to be + calculated outside of the core independently in channel drivers, + notably, chan_sip and chan_pjsip. Both of these channel drivers + already have to calculate device state in a fashion more complex + than the core can handle, as they aggregate all state of all + channels associated with a peer/endpoint; they also independently + track whether or not one of those channels is currently on hold + and mark the device state appropriately. In 12+, we now have the + ability to report an AST_DEVICE_ONHOLD state for all channels + that defer their device state to the core. This is due to channel + hold state actually now being tracked on the channel itself. If a + channel driver defers its device state to the core (which many, + such as DAHDI, IAX2, and others do in most situations), the + device state core already goes out to get a channel associated + with the device. As such, it can now also factor the channel hold + state in its calculation. This patch adds this logic to the + device state core. It also uses an existing mapping between + device state and channel state to handle more channel states. + chan_pjsip has been updated slightly as well to make use of this + (as it was, for some reason, reporting a channel state of BUSY as + a device state of INUSE, which feels slightly wrong). Review: + https://reviewboard.asterisk.org/r/3771/ ASTERISK-24038 #close + +2014-07-24 13:00 +0000 [r419342] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/manager.h, doc/appdocsxml.dtd, main/xmldoc.c, + main/manager_bridges.c, main/manager.c, + include/asterisk/xmldoc.h, main/config_options.c: AMI: Allow for + command response documentation Allow for responses to AMI + actions/commands to be documented properly in XML and displayed + via the CLI. Response events are documented exactly as standard + AMI events are documented. Review: + https://reviewboard.asterisk.org/r/3812/ + +2014-07-23 16:46 +0000 [r419319] Matthew Jordan <mjordan@digium.com> + + * main/endpoints.c, tests/test_stasis_endpoints.c, /: endpoints: + Fix failing unit tests from r419196 This patch does two things: + (1) It updates the unit tests to expect additional stasis + messages. More messages are now sent to the endpoint topic, due + to forwarding all channel messages and the forwarding + relationship set up between endpoints themselves. (2) Remove the + technology forwarding subscription during ast_endpoint_shutdown. + This prevents an improper double shutdown of an endpoint from + occurring. ........ Merged revisions 419318 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-23 14:00 +0000 [r419286] Scott Griepentrog <sgriepentrog@digium.com> + + * apps/app_voicemail.c, /: app_voicemail: use a consistent + generator string When updating voicemail.conf when a user changes + their pin, change the generator string to be the same as the + module name when reading so that the same config_hook will be + called. Review: https://reviewboard.asterisk.org/r/3837/ ........ + Merged revisions 419284 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419285 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-23 01:28 +0000 [r419268] Corey Farrell <git@cfware.com> + + * main/manager.c, res/res_fax.c: res_fax: unregister manager + actions on unload * Unregister manager actions FAXSessions, + FAXSession and FAXStats at unload. * Update ast_manager_register2 + use ao2_t_alloc tagged with the action name. ASTERISK-24058 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3831/ + +2014-07-22 20:22 +0000 [r419222-419252] Michael L. Young <elgueromexicano@gmail.com> + + * CHANGES, main/bridge_channel.c: core/bridge_channel: Substitute + Variables In Features Application Map Say you wanted to include + variables in an application map and have those variables + substituted and passed along to the application being executed; + currently this does not happen. This patch adds this ability to + pass channel variable values to an application before being + executed. ASTERISK-22608 #close Reported by: Michael L. Young + patches: features_substitute_arguments_v2.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3819/ + + * CHANGES, apps/app_mixmonitor.c: apps/app_mixmonitor: Add Options + To Play Beep At Start Or Stop We have a new periodic beep feature + but sometimes a user needs some sort of feedback, without the + need to have a periodic beep during the recording, to let them + know that MixMonitor started recording or ended the recording. + The use case where this patch is being used is when using Dynamic + Features to start and end MixMonitor. This patch adds an option + to play a beep when MixMonitor starts and an option to play a + beep when MixMonitor ends. ASTERISK-24051 #close Reported by: + Michael L. Young patches: mixmonitor-play-beep-start-stop.diff + uploaded by Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3820/ + + * main/db.c, include/asterisk/astdb.h: core/db: Improve I/O When + Updating Rows When updating a row, we are currently doing an + INSERT OR REPLACE INTO. The downside to this is that the row is + deleted if it exists and then a new row is inserted. So, we are + hitting the disk twice. One for the deletion and one for the + insertion. This patch changes this statement to an INSERT INTO + and if the insert fails because a row with that key exists, we + will IGNORE the failure. Then we will attempt to perform an + UPDATE on the existing row if that row wasn't just INSERTed. + ASTERISK-24050 #close Reported by: Michael L. Young patches: + astdb-insert-update-io-help_trunk_v2.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3815/ + +2014-07-22 17:10 +0000 [r419206] Richard Mudgett <rmudgett@digium.com> + + * codecs/codec_speex.c: codec_speex: Fix trashing normal static + frame for AST_FRAME_CNG. Made use a local static frame to + generate the AST_FRAME_CNG frame when silence starts. I don't + think the handling of the AST_FRAME_CNG has ever really worked + because there doesn't seem to be any consumers of it. Review: + https://reviewboard.asterisk.org/r/3813/ + +2014-07-22 16:20 +0000 [r419203] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/endpoints.h, + rest-api/api-docs/applications.json, include/asterisk/xmpp.h, + main/channel_internal_api.c, channels/chan_motif.c, + include/asterisk/channel.h, res/ari/resource_applications.h, + res/res_xmpp.c, channels/chan_iax2.c, main/endpoints.c, + channels/chan_pjsip.c, main/channel.c, + res/ari/resource_endpoints.c, /, channels/chan_sip.c: ARI: Fix + endpoint/channel subscription issues; allow for subscriptions to + tech This patch serves two purposes: (1) It fixes some bugs with + endpoint subscriptions not reporting all of the channel events + (2) It serves as the preliminary work needed for ASTERISK-23692, + which allows for sending/receiving arbitrary out of call text + messages through ARI in a technology agnostic fashion. The + messaging functionality described on ASTERISK-23692 requires two + things: (1) The ability to send/receive messages associated with + an endpoint. This is relatively straight forwards with the + endpoint core in Asterisk now. (2) The ability to send/receive + messages associated with a technology and an arbitrary technology + defined URI. This is less straight forward, as endpoints are + formed from a tech + resource pair. We don't have a mechanism to + note that a technology that *may* have endpoints exists. This + patch provides such a mechanism, and fixes a few bugs along the + way. The first major bug this patch fixes is the forwarding of + channel messages to their respective endpoints. Prior to this + patch, there were two problems: (1) Channel caching messages + weren't forwarded. Thus, the endpoints missed most of the + interesting bits (such as channel creation, destruction, state + changes, etc.) (2) Channels weren't associated with their + endpoint until after creation. This resulted in endpoints missing + the channel creation message, which limited the usefulness of the + subscription in the first place (a major use case being 'tell me + when this endpoint has a channel'). Unfortunately, this meant + another parameter to ast_channel_alloc. Since not all channel + technologies support an ast_endpoint, this patch makes such a + call optional and opts for a new function, + ast_channel_alloc_with_endpoint. When endpoints are created, they + will implicitly create a technology endpoint for their technology + (if one does not already exist). A technology endpoint is special + in that it has no state, cannot have channels created for it, + cannot be created explicitly, and cannot be destroyed except on + shutdown. It does, however, have all messages from other + endpoints in its technology forwarded to it. Combined with the + bug fixes, we now have Stasis messages being properly forwarded. + Consider the following scenario: two PJSIP endpoints (foo and + bar), where bar has a single channel associated with it and foo + has two channels associated with it. The messages would be + forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint + PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / + channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the + applications resource, can: - subscribe to endpoint:PJSIP/foo and + get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and + endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get + notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - + subscribe to endpoint:PJSIP and get notifications for channels + PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints + PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, + it never has events itself. It merely provides an aggregation + point for all other endpoints in its technology (which in turn + aggregate all channel messages associated with that endpoint). + This patch also adds endpoints to res_xmpp and chan_motif, + because the actual messaging work will need it (messaging without + XMPP is just sad). Review: + https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........ + Merged revisions 419196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-22 14:36 +0000 [r419180] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: chan_iax2: Restore previous behavior of + iax2_best_codec. The iax2_best_codec function was changed to + convert the formats into a format compatibilities structure and + grab the first format from it. The resulting order differs from + the previous order of iax2_best_codec which causes unexpected + formats to get chosen (such as g723). This commit brings back the + old behavior of iax2_best_codec by having a specified preference + list. Review: https://reviewboard.asterisk.org/r/3835/ + +2014-07-22 14:22 +0000 [r419110-419175] Kinsey Moore <kmoore@digium.com> + + * addons/ooh323c/src/printHandler.c, tests/test_sorcery_realtime.c, + tests/test_json.c, addons/ooh323c/src/ooq931.c, + tests/test_astobj2_thrash.c, addons/chan_ooh323.c, /, + tests/test_optional_api.c, tests/test_abstract_jb.c, + apps/app_meetme.c, tests/test_logger.c, tests/test_event.c, + tests/test_hashtab_thrash.c, res/res_mwi_external_ami.c, + tests/test_sorcery.c, res/res_corosync.c, + tests/test_voicemail_api.c, tests/test_aoc.c, + tests/test_astobj2.c, tests/test_config.c: Fix more dev-mode + build issues ........ Merged revisions 419129 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 419162 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419163 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/dial.c: Dial API: Prevent crash on NULL cap This prevents a + crash in the Dial API triggered by use of the Page() application + where a format capability struct was used before checking whether + it was NULL. ASTERISK-24074 #close + + * channels/chan_skinny.c, tests/test_core_format.c: Fix build in + dev-mode + +2014-07-21 16:26 +0000 [r419109] Jonathan Rose <jrose@digium.com> + + * channels/chan_iax2.c: chan_iax2: Restore codec choice behavior + from media formats branch After merging the media formats branch, + chan_iax2 was discarding codec preferences for the purpose of + choosing which codec a channel would use once a call started. + This patch restores the Asterisk 1.8-12 codec choice behaviors. + ASTERISK-23958 #close Review: + https://reviewboard.asterisk.org/r/3800/ + +2014-07-21 16:09 +0000 [r419093] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: chan_iax2: Only send mini frames if the + underlying format has not changed, not if it has. ASTERISK-24072 + #close Reported by: Matt Jordan + +2014-07-21 14:49 +0000 [r419077] Sean Bright <sean@malleable.com> + + * configure, configure.ac: Fix build when pjproject is installed in + a non-standard location. When configuring Asterisk to build + against a version of pjproject installed in a non-standard + location, the checks for "PJSIP Transaction Group Lock Support" + and "PJSIP Media Stream Replacement Support" fail. This is + because these secondary checks are not taking the CFLAGS and LIBS + returned by the pkg-config check into account. Review: + https://reviewboard.asterisk.org/r/3830 + +2014-07-21 08:41 +0000 [r419060] Corey Farrell <git@cfware.com> + + * channels/sig_analog.c, res/res_smdi.c, channels/chan_motif.c, + include/asterisk/smdi.h, apps/app_voicemail.c, + channels/chan_dahdi.c: res_smdi: convert to astobj2 Remove + functions: ast_smdi_interface_unref ast_smdi_md_message_putback + ast_smdi_mwi_message_putback ast_smdi_md_message destructor + ast_smdi_mwi_message destructor Includes for astobj.h are removed + everywhere it's possible. ASTERISK-24066 #close Review: + https://reviewboard.asterisk.org/r/3758/ + +2014-07-20 22:06 +0000 [r419044] Matthew Jordan <mjordan@digium.com> + + * apps/app_confbridge.c, res/ari/resource_channels.c, + include/asterisk/rtp_engine.h, include/asterisk/slinfactory.h, + res/res_calendar.c, codecs/codec_g722.c, + include/asterisk/res_pjsip_session.h, main/frame.c, + codecs/ex_lpc10.h, apps/app_dictate.c, res/res_fax.c, + apps/app_echo.c, include/asterisk/slin.h, codecs/codec_g726.c, + formats/format_ogg_vorbis.c, codecs/codec_gsm.c, + codecs/ex_alaw.h, formats/format_wav_gsm.c, + channels/iax2/provision.c, channels/chan_iax2.c, + res/res_format_attr_h264.c, main/data.c, main/manager.c, + include/asterisk/audiohook.h, formats/format_pcm.c, + main/config_options.c, res/res_format_attr_silk.c, + main/bridge_channel.c, res/res_speech.c, channels/chan_pjsip.c, + res/res_clioriginate.c, formats/format_g729.c, + channels/chan_unistim.c, res/res_rtp_asterisk.c, + include/asterisk/smoother.h (added), main/rtp_engine.c, + addons/format_mp3.c, formats/format_wav.c, + apps/confbridge/conf_chan_record.c, include/asterisk/speech.h, + codecs/ex_adpcm.h, channels/iax2/codec_pref.c (added), + include/asterisk/codec.h (added), formats/format_siren7.c, + include/asterisk/file.h, channels/chan_dahdi.c, + include/asterisk/image.h, funcs/func_channel.c, + main/abstract_jb.c, formats/format_h263.c, codecs/codec_dahdi.c, + main/dsp.c, apps/app_voicemail.c, apps/app_jack.c, + funcs/func_talkdetect.c, channels/chan_vpb.cc, + channels/chan_sip.c, formats/format_sln.c, + tests/test_abstract_jb.c, codecs/codec_alaw.c, UPGRADE.txt, + main/smoother.c (added), codecs/ex_speex.h, + channels/chan_console.c, apps/app_talkdetect.c, + main/format_pref.c (removed), main/indications.c, + include/asterisk/format_cap.h, main/media_index.c, + apps/app_agent_pool.c, res/res_pjsip_session.c, main/cli.c, + res/res_format_attr_celt.c, channels/chan_skinny.c, + tests/test_core_format.c (added), funcs/func_frame_trace.c, + res/res_pjsip/pjsip_configuration.c, main/file.c, + include/asterisk/frame.h, formats/format_g726.c, + apps/app_mixmonitor.c, channels/chan_mgcp.c, main/sorcery.c, + codecs/ex_ilbc.h, codecs/codec_lpc10.c, tests/test_format_cache.c + (added), apps/app_meetme.c, main/translate.c, + apps/app_originate.c, res/parking/parking_applications.c, + apps/app_ices.c, channels/iax2/parser.c, res/res_rtp_multicast.c, + pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_vox.c, + main/format_cap.c, tests/test_cel.c, include/asterisk/format.h, + formats/format_h264.c, apps/app_chanspy.c, apps/app_nbscat.c, + addons/chan_ooh323.c, bridges/bridge_holding.c, + channels/iax2/include/codec_pref.h (added), codecs/codec_adpcm.c, + apps/app_waitforsilence.c, res/res_pjsip_sdp_rtp.c, + addons/chan_ooh323.h, bridges/bridge_simple.c, + apps/app_alarmreceiver.c, bridges/bridge_softmix.c, + res/res_stasis_snoop.c, main/sounds_index.c, main/core_local.c, + main/codec_builtin.c (added), include/asterisk/format_cache.h + (added), apps/app_speech_utils.c, res/res_format_attr_opus.c, + include/asterisk/abstract_jb.h, main/channel.c, + include/asterisk/format_compatibility.h (added), apps/app_mp3.c, + tests/test_voicemail_api.c, channels/chan_alsa.c, main/app.c, + formats/format_g723.c, codecs/codec_ilbc.c, tests/test_config.c, + formats/format_gsm.c, apps/app_milliwatt.c, codecs/ex_ulaw.h, + main/asterisk.c, include/asterisk/res_pjsip.h, main/format.c, + main/ccss.c, main/bridge.c, codecs/codec_speex.c, + include/asterisk/format_pref.h (removed), apps/app_record.c, + main/slinfactory.c, res/res_adsi.c, main/core_unreal.c, + res/ari/resource_bridges.c, include/asterisk/callerid.h, + channels/pjsip/dialplan_functions.c, main/dial.c, + channels/dahdi/bridge_native_dahdi.c, main/format_cache.c + (added), include/asterisk/mod_format.h, apps/app_sms.c, + codecs/codec_resample.c, main/format_compatibility.c (added), + main/audiohook.c, formats/format_jpeg.c, res/res_stasis.c, + formats/format_g719.c, include/asterisk/translate.h, + funcs/func_speex.c, codecs/codec_a_mu.c, + channels/iax2/format_compatibility.c (added), + apps/app_festival.c, main/channel_internal_api.c, + tests/test_format_api.c (removed), codecs/ex_g722.h, + main/utils.c, res/ari/resource_sounds.c, + res/res_format_attr_h263.c, codecs/ex_g726.h, + include/asterisk/_private.h, channels/chan_oss.c, + channels/chan_misdn.c, main/codec.c (added), main/callerid.c, + addons/ooh323cDriver.c, apps/app_amd.c, codecs/codec_ulaw.c, + main/image.c, channels/chan_nbs.c, bridges/bridge_native_rtp.c, + channels/iax2/include/format_compatibility.h (added), + formats/format_siren14.c, res/res_fax_spandsp.c, + addons/chan_mobile.c, addons/ooh323cDriver.h, + channels/sip/include/sip.h, tests/test_format_cap.c (added), + channels/chan_multicast_rtp.c, include/asterisk/vector.h, + channels/chan_bridge_media.c, apps/app_fax.c, + main/bridge_basic.c, apps/app_test.c, include/asterisk/channel.h, + include/asterisk/data.h, tests/test_core_codec.c (added), + res/res_musiconhold.c, codecs/ex_gsm.h, formats/format_ilbc.c, + include/asterisk/config_options.h, channels/chan_phone.c, + include/asterisk/bridge_channel.h, apps/app_dumpchan.c, + channels/chan_motif.c, res/res_agi.c: media formats: re-architect + handling of media for performance improvements In the old times + media formats were represented using a bit field. This was fast + but had a few limitations. 1. Asterisk was limited in how many + formats it could handle. 2. Formats, being a bit field, could not + include any attribute information. A format was strictly its + type, e.g., "this is ulaw". This was changed in Asterisk 10 (see + https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal + for notes on that work) which led to the creation of the + ast_format structure. This structure allowed Asterisk to handle + attributes and bundle information with a format. Additionally, + ast_format_cap was created to act as a container for multiple + formats that, together, formed the capability of some entity. + Another mechanism was added to allow logic to be registered which + performed format attribute negotiation. Everywhere throughout the + codebase Asterisk was changed to use this strategy. + Unfortunately, in software, there is no free lunch. These new + capabilities came at a cost. Performance analysis and profiling + showed that we spend an inordinate amount of time comparing, + copying, and generally manipulating formats and their related + structures. Basic prototyping has shown that a reasonably large + performance improvement could be made in this area. This patch is + the result of that project, which overhauled the media format + architecture and its usage in Asterisk to improve performance. + Generally, the new philosophy for handling formats is as follows: + * The ast_format structure is reference counted. This removed a + large amount of the memory allocations and copying that was done + in prior versions. * In order to prevent race conditions while + keeping things performant, the ast_format structure is immutable + by convention and lock-free. Violate this tenet at your peril! * + Because formats are reference counted, codecs are also reference + counted. The Asterisk core generally provides built-in codecs and + caches the ast_format structures created to represent them. + Generally, to prevent inordinate amounts of module reference + bumping, codecs and formats can be added at run-time but cannot + be removed. * All compatibility with the bit field representation + of codecs/formats has been moved to a compatibility API. The + primary user of this representation is chan_iax2, which must + continue to maintain its bit-field usage of formats for + interoperability concerns. * When a format is negotiated with + attributes, or when a format cannot be represented by one of the + cached formats, a new format object is created or cloned from an + existing format. That format may have the same codec underlying + it, but is a different format than a version of the format with + different attributes or without attributes. * While formats are + reference counted objects, the reference count maintained on the + format should be manipulated with care. Formats are generally + cached and will persist for the lifetime of Asterisk and do not + explicitly need to have their lifetime modified. An exception to + this is when the user of a format does not know where the format + came from *and* the user may outlive the provider of the format. + This occurs, for example, when a format is read from a channel: + the channel may have a format with attributes (hence, non-cached) + and the user of the format may last longer than the channel (if + the reference to the channel is released prior to the format's + reference). For more information on this work, see the API design + notes: + https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite + Finally, this work was the culmination of a large number of + developer's efforts. Extra thanks goes to Corey Farrell, who took + on a large amount of the work in the Asterisk core, chan_sip, and + was an invaluable resource in peer reviews throughout this + project. There were a substantial number of patches contributed + during this work; the following issues/patch names simply reflect + some of the work (and will cause the release scripts to give + attribution to the individuals who work on them). Reviews: + https://reviewboard.asterisk.org/r/3814 + https://reviewboard.asterisk.org/r/3808 + https://reviewboard.asterisk.org/r/3805 + https://reviewboard.asterisk.org/r/3803 + https://reviewboard.asterisk.org/r/3801 + https://reviewboard.asterisk.org/r/3798 + https://reviewboard.asterisk.org/r/3800 + https://reviewboard.asterisk.org/r/3794 + https://reviewboard.asterisk.org/r/3793 + https://reviewboard.asterisk.org/r/3792 + https://reviewboard.asterisk.org/r/3791 + https://reviewboard.asterisk.org/r/3790 + https://reviewboard.asterisk.org/r/3789 + https://reviewboard.asterisk.org/r/3788 + https://reviewboard.asterisk.org/r/3787 + https://reviewboard.asterisk.org/r/3786 + https://reviewboard.asterisk.org/r/3784 + https://reviewboard.asterisk.org/r/3783 + https://reviewboard.asterisk.org/r/3778 + https://reviewboard.asterisk.org/r/3774 + https://reviewboard.asterisk.org/r/3775 + https://reviewboard.asterisk.org/r/3772 + https://reviewboard.asterisk.org/r/3761 + https://reviewboard.asterisk.org/r/3754 + https://reviewboard.asterisk.org/r/3753 + https://reviewboard.asterisk.org/r/3751 + https://reviewboard.asterisk.org/r/3750 + https://reviewboard.asterisk.org/r/3748 + https://reviewboard.asterisk.org/r/3747 + https://reviewboard.asterisk.org/r/3746 + https://reviewboard.asterisk.org/r/3742 + https://reviewboard.asterisk.org/r/3740 + https://reviewboard.asterisk.org/r/3739 + https://reviewboard.asterisk.org/r/3738 + https://reviewboard.asterisk.org/r/3737 + https://reviewboard.asterisk.org/r/3736 + https://reviewboard.asterisk.org/r/3734 + https://reviewboard.asterisk.org/r/3722 + https://reviewboard.asterisk.org/r/3713 + https://reviewboard.asterisk.org/r/3703 + https://reviewboard.asterisk.org/r/3689 + https://reviewboard.asterisk.org/r/3687 + https://reviewboard.asterisk.org/r/3674 + https://reviewboard.asterisk.org/r/3671 + https://reviewboard.asterisk.org/r/3667 + https://reviewboard.asterisk.org/r/3665 + https://reviewboard.asterisk.org/r/3625 + https://reviewboard.asterisk.org/r/3602 + https://reviewboard.asterisk.org/r/3519 + https://reviewboard.asterisk.org/r/3518 + https://reviewboard.asterisk.org/r/3516 + https://reviewboard.asterisk.org/r/3515 + https://reviewboard.asterisk.org/r/3512 + https://reviewboard.asterisk.org/r/3506 + https://reviewboard.asterisk.org/r/3413 + https://reviewboard.asterisk.org/r/3410 + https://reviewboard.asterisk.org/r/3387 + https://reviewboard.asterisk.org/r/3388 + https://reviewboard.asterisk.org/r/3389 + https://reviewboard.asterisk.org/r/3390 + https://reviewboard.asterisk.org/r/3321 + https://reviewboard.asterisk.org/r/3320 + https://reviewboard.asterisk.org/r/3319 + https://reviewboard.asterisk.org/r/3318 + https://reviewboard.asterisk.org/r/3266 + https://reviewboard.asterisk.org/r/3265 + https://reviewboard.asterisk.org/r/3234 + https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close + Reported by: mjordan media_formats_translation_core.diff uploaded + by kharwell (License 6464) rb3506.diff uploaded by mjordan + (License 6283) media_format_app_file.diff uploaded by kharwell + (License 6464) misc-2.diff uploaded by file (License 5000) + chan_mild-3.diff uploaded by file (License 5000) + chan_obscure.diff uploaded by file (License 5000) jingle.diff + uploaded by file (License 5000) funcs.diff uploaded by file + (License 5000) formats.diff uploaded by file (License 5000) + core.diff uploaded by file (License 5000) bridges.diff uploaded + by file (License 5000) mf-codecs-2.diff uploaded by file (License + 5000) mf-app_fax.diff uploaded by file (License 5000) + mf-apps-3.diff uploaded by file (License 5000) + media-formats-3.diff uploaded by file (License 5000) + ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License + 5909) rb3689.patch uploaded by mjordan (License 6283) + ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) + mf-attributes-3.diff uploaded by file (License 5000) + ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by + coreyfarrell (License 5909) rb3800.patch uploaded by jrose + (License 6182) chan_sip.diff uploaded by mjordan (License 6283) + rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 + #close Tested by: sgriepentrog, mjordan, coreyfarrell + sip_cleanup.diff uploaded by opticron (License 6273) + chan_sip_caps.diff uploaded by mjordan (License 6283) + rb3751.patch uploaded by coreyfarrell (License 5909) + chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 + #close Tested by: opticron direct_media.diff uploaded by opticron + (License 6273) pjsip-direct-media.diff uploaded by file (License + 5000) format_cap_remove.diff uploaded by opticron (License 6273) + media_format_fixes.diff uploaded by opticron (License 6273) + chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 + #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti + (License 5621) chan_dahdi.diff uploaded by file (License 5000) + ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, + file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by + rmudgett (License 5621) moh_cleanup.diff uploaded by opticron + (License 6273) bridge_leak.diff uploaded by opticron (License + 6273) translate.diff uploaded by file (License 5000) rb3795.patch + uploaded by rmudgett (License 5621) tls_fix.diff uploaded by + mjordan (License 6283) fax-mf-fix-2.diff uploaded by file + (License 5000) rtp_transfer_stuff uploaded by mjordan (License + 6283) rb3787.patch uploaded by rmudgett (License 5621) + media-formats-explicit-translate-format-3.diff uploaded by file + (License 5000) format_cache_case_fix.diff uploaded by opticron + (License 6273) rb3774.patch uploaded by rmudgett (License 5621) + rb3775.patch uploaded by rmudgett (License 5621) + rtp_engine_fix.diff uploaded by opticron (License 6273) + rtp_crash_fix.diff uploaded by opticron (License 6273) + rb3753.patch uploaded by mjordan (License 6283) rb3750.patch + uploaded by mjordan (License 6283) rb3748.patch uploaded by + rmudgett (License 5621) media_format_fixes.diff uploaded by + opticron (License 6273) rb3740.patch uploaded by mjordan (License + 6283) rb3739.patch uploaded by mjordan (License 6283) + rb3734.patch uploaded by mjordan (License 6283) rb3689.patch + uploaded by mjordan (License 6283) rb3674.patch uploaded by + coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell + (License 5909) rb3667.patch uploaded by coreyfarrell (License + 5909) rb3665.patch uploaded by mjordan (License 6283) + rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch + uploaded by coreyfarrell (License 5909) + format_compatibility-2.diff uploaded by file (License 5000) + core.diff uploaded by file (License 5000) + +2014-07-18 21:48 +0000 [r419022] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/recordings.json, res/ari/resource_recordings.c, + res/stasis_recording/stored.c, res/res_ari_recordings.c, /, + include/asterisk/stasis_app_recording.h, + res/ari/resource_recordings.h, CHANGES: ari: Add a copy operation + for stored recordings This patch adds a new operation for stored + recordings, copy. It takes an existing stored recording and makes + a copy of it in the same directory or a relative directory under + the stored recording directory. + /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name} + This is particularly useful for voicemail-esque applications, + which may need to copy or move recordings around a directory + structure. Review: https://reviewboard.asterisk.org/r/3768/ + ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam + Galarneau ........ Merged revisions 419021 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 21:25 +0000 [r418997-419020] Corey Farrell <git@cfware.com> + + * main/stasis_message_router.c, /: stasis: fix call to ao2_t_alloc + for stasis_message_router_create This fixes a build failure + introduced by r3821. struct stasis_topic is opaque, so + topic->name is unavailable. Switch to using stasis_topic_name(). + ........ Merged revisions 419019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis.c, main/stasis_cache_pattern.c, + main/stasis_message.c, main/stasis_message_router.c, /: stasis: + use ao2_t_alloc for certain object allocators Add tags to stasis + objects using the name. This makes it easier to track the source + of certain stasis ref leaks. Review: + https://reviewboard.asterisk.org/r/3821/ ........ Merged + revisions 418996 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 19:07 +0000 [r418980] Kinsey Moore <kmoore@digium.com> + + * res/res_fax_spandsp.c: Fix build in dev-mode + +2014-07-18 17:55 +0000 [r418961-418963] Scott Griepentrog <sgriepentrog@digium.com> + + * res/res_pjsip_pubsub.c, main/astobj2.c, + include/asterisk/astobj2.h, main/logger.c, main/utils.c: astobj2: + assert on invalid ref and backtrace cleanup If a reference count + goes negative, instead of just logging that fact, be more helpful + with a backtrace and an assert that will DO_CRASH. This patch + also removes the duplicate ao2_bt() function and cleans up + extraneous usage of the ast_log_backtrace() call. Review: + https://reviewboard.asterisk.org/r/3765/ + + * /, channels/chan_sip.c: media formats: fix ref leak of peer for + mwi subscription Holding a reference to the peer during mwi + subscriptions resulted in a circular reference because the final + event message would not be sent until destruction of the peer. + Instead, pass the name of the peer to the event callback so that + it can fail gracefully after the peer has gone. ASTERISK-23959 + Review: https://reviewboard.asterisk.org/r/3754/ ........ Merged + revisions 418636 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/features_config.c: feature_config: insure featuregroups + and applicationmaps are initialized If the features.conf is + missing, the cfg->featurgroups and cfg->applicationmaps is not + initialized, resulting in assert on ao2_find of a null container. + This patch changes the initialization call and adds asserts for a + safeguard. Review: https://reviewboard.asterisk.org/r/3809/ + ........ Merged revisions 418886 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 16:47 +0000 [r418938] Richard Mudgett <rmudgett@digium.com> + + * funcs/func_audiohookinherit.c, /: func_audiohookinherit.c: Fixup + some XML documentation wording. ........ Merged revisions 418937 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 16:28 +0000 [r418911-418936] Jonathan Rose <jrose@digium.com> + + * main/channel.c, funcs/func_audiohookinherit.c, /, + include/asterisk/audiohook.h, main/framehook.c, res/res_fax.c, + main/bridge_basic.c, include/asterisk/res_fax.h, + bridges/bridge_native_rtp.c, main/audiohook.c, CHANGES, + include/asterisk/framehook.h, res/res_pjsip_refer.c: Channels: + Masquerades to automatically move frame/audio hooks Whenever + possible, audiohooks and framehooks will now be copied over to + the channel that the masquerading channel gets cloned into. This + should occur for all audiohooks and most framehooks. As a result, + in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now + deprecated and its behavior is essentially the new default for + all audiohooks, plus some additional audiohooks/framehooks. + Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged + revisions 418914 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_fax.c, include/asterisk/res_fax.h, CHANGES, + res/res_fax.exports.in, res/res_fax_spandsp.c: res_fax: Provide + AMI equivalents for fax CLI commands Specifically the following + equivalents were created: fax show session -> FAXSession fax show + sessions -> FAXSessions fax show stats -> FAXStats Review: + https://reviewboard.asterisk.org/r/3666/ + +2014-07-18 00:11 +0000 [r418893-418895] Sean Bright <sean@malleable.com> + + * config.sub, menuselect/config.guess, menuselect/config.sub, + config.guess: Update config.guess and config.sub + + * autoconf/ast_ext_tool_check.m4: Add missing file from previous + commit. + + * menuselect/aclocal.m4, menuselect/configure, + menuselect/acinclude.m4 (removed), menuselect/bootstrap.sh, + menuselect/autoconfig.h.in: Import Asterisk's autoconf magic + instead of using our own. + +2014-07-17 21:17 +0000 [r418832-418870] Matthew Jordan <mjordan@digium.com> + + * configs/samples/acl.conf.sample (added), + configs/samples/extensions.conf.sample (added), + configs/res_parking.conf.sample (removed), + configs/samples/cel_sqlite3_custom.conf.sample (added), + configs/cdr_sqlite3_custom.conf.sample (removed), + configs/modules.conf.sample (removed), + configs/samples/cli_aliases.conf.sample (added), + configs/meetme.conf.sample (removed), + configs/cdr_pgsql.conf.sample (removed), + configs/samples/extensions.ael.sample (added), + configs/samples/cdr_adaptive_odbc.conf.sample (added), + configs/samples/motif.conf.sample (added), + configs/samples/extensions_minivm.conf.sample (added), + configs/samples/res_curl.conf.sample (added), + configs/res_config_sqlite3.conf.sample (removed), + configs/mgcp.conf.sample (removed), configs/dsp.conf.sample + (removed), configs/udptl.conf.sample (removed), + configs/sip.conf.sample (removed), configs/dbsep.conf.sample + (removed), configs/queuerules.conf.sample (removed), + configs/samples/cdr_mysql.conf.sample (added), + configs/confbridge.conf.sample (removed), + configs/samples/cdr_odbc.conf.sample (added), + configs/samples/minivm.conf.sample (added), + configs/enum.conf.sample (removed), + configs/samples/codecs.conf.sample (added), + configs/samples/chan_dahdi.conf.sample (added), + configs/samples/cdr_custom.conf.sample (added), + configs/samples/res_config_mysql.conf.sample (added), + configs/samples/dundi.conf.sample (added), + configs/samples/oss.conf.sample (added), + configs/samples/app_mysql.conf.sample (added), + configs/samples/queues.conf.sample (added), + configs/samples/cdr.conf.sample (added), + configs/samples/cdr_syslog.conf.sample (added), + configs/festival.conf.sample (removed), + configs/samples/cel_pgsql.conf.sample (added), + configs/http.conf.sample (removed), configs/phoneprov.conf.sample + (removed), configs/alarmreceiver.conf.sample (removed), + configs/samples/features.conf.sample (added), + configs/cdr_tds.conf.sample (removed), + configs/func_odbc.conf.sample (removed), + configs/samples/logger.conf.sample (added), + configs/samples/res_odbc.conf.sample (added), + configs/samples/agents.conf.sample (added), + configs/res_fax.conf.sample (removed), + configs/samples/xmpp.conf.sample (added), + configs/iaxprov.conf.sample (removed), + configs/res_pgsql.conf.sample (removed), + configs/extensions.conf.sample (removed), + configs/chan_mobile.conf.sample (removed), configs/asterisk.adsi + (removed), configs/cel_sqlite3_custom.conf.sample (removed), + configs/users.conf.sample (removed), + configs/samples/res_pktccops.conf.sample (added), + configs/samples/amd.conf.sample (added), configs/rtp.conf.sample + (removed), configs/samples/res_parking.conf.sample (added), + configs/hep.conf.sample (removed), + configs/samples/modules.conf.sample (added), + configs/cel_tds.conf.sample (removed), + configs/res_curl.conf.sample (removed), + configs/samples/skinny.conf.sample (added), + configs/samples/cdr_pgsql.conf.sample (added), + configs/samples/sip_notify.conf.sample (added), + configs/samples/test_sorcery.conf.sample (added), + configs/samples/dsp.conf.sample (added), + configs/ss7.timers.sample (removed), + configs/samples/udptl.conf.sample (added), + configs/cdr_odbc.conf.sample (removed), + configs/samples/sip.conf.sample (added), + configs/minivm.conf.sample (removed), + configs/res_config_sqlite.conf.sample (removed), + configs/codecs.conf.sample (removed), configs/osp.conf.sample + (removed), configs/samples/cel_custom.conf.sample (added), + configs/samples/dbsep.conf.sample (added), + configs/samples/app_skel.conf.sample (added), + configs/console.conf.sample (removed), + configs/cdr_manager.conf.sample (removed), + configs/cdr_custom.conf.sample (removed), + configs/chan_dahdi.conf.sample (removed), + configs/res_config_mysql.conf.sample (removed), + configs/samples/statsd.conf.sample (added), + configs/cli.conf.sample (removed), configs/queues.conf.sample + (removed), configs/cdr_syslog.conf.sample (removed), UPGRADE.txt, + configs/manager.conf.sample (removed), + configs/samples/res_corosync.conf.sample (added), + configs/features.conf.sample (removed), configs/sla.conf.sample + (removed), configs/logger.conf.sample (removed), + configs/res_odbc.conf.sample (removed), + configs/agents.conf.sample (removed), + configs/samples/ooh323.conf.sample (added), Makefile, + configs/xmpp.conf.sample (removed), + configs/samples/phoneprov.conf.sample (added), + configs/samples/alarmreceiver.conf.sample (added), + configs/samples/cdr_tds.conf.sample (added), + configs/extconfig.conf.sample (removed), + configs/samples/func_odbc.conf.sample (added), + configs/samples/res_fax.conf.sample (added), + configs/samples/iaxprov.conf.sample (added), + configs/samples/res_ldap.conf.sample (added), + configs/samples/dnsmgr.conf.sample (added), + configs/res_pktccops.conf.sample (removed), + configs/cel.conf.sample (removed), + configs/samples/res_pgsql.conf.sample (added), + configs/samples/chan_mobile.conf.sample (added), + configs/samples/asterisk.adsi (added), + configs/samples/users.conf.sample (added), + configs/samples/rtp.conf.sample (added), + configs/phone.conf.sample (removed), configs/skinny.conf.sample + (removed), configs/muted.conf.sample (removed), + configs/samples/hep.conf.sample (added), configs/iax.conf.sample + (removed), configs/samples/cel_tds.conf.sample (added), + configs/sip_notify.conf.sample (removed), + configs/samples/telcordia-1.adsi (added), + configs/samples/alsa.conf.sample (added), + configs/samples/adsi.conf.sample (added), + configs/test_sorcery.conf.sample (removed), + configs/samples/followme.conf.sample (added), + configs/samples/asterisk.conf.sample (added), + configs/extensions.lua.sample (removed), configs/say.conf.sample + (removed), configs/cel_custom.conf.sample (removed), + configs/samples/ss7.timers.sample (added), + configs/samples/cel_odbc.conf.sample (added), + configs/app_skel.conf.sample (removed), + configs/samples/ccss.conf.sample (added), + configs/cli_permissions.conf.sample (removed), + configs/statsd.conf.sample (removed), + configs/samples/res_config_sqlite.conf.sample (added), + configs/config_test.conf.sample (removed), + configs/indications.conf.sample (removed), + configs/samples/osp.conf.sample (added), + configs/samples/cdr_manager.conf.sample (added), + configs/samples/console.conf.sample (added), + configs/voicemail.conf.sample (removed), + configs/res_corosync.conf.sample (removed), + configs/misdn.conf.sample (removed), + configs/samples/cli.conf.sample (added), configs/ari.conf.sample + (removed), configs/ooh323.conf.sample (removed), + configs/samples/calendar.conf.sample (added), + configs/samples/res_stun_monitor.conf.sample (added), + configs/samples/manager.conf.sample (added), + configs/samples/pjsip_notify.conf.sample (added), + configs/samples/sla.conf.sample (added), + configs/musiconhold.conf.sample (removed), + configs/pjsip.conf.sample (removed), configs/sorcery.conf.sample + (removed), configs/vpb.conf.sample (removed), + configs/unistim.conf.sample (removed), + configs/res_ldap.conf.sample (removed), + configs/dnsmgr.conf.sample (removed), + configs/samples/extconfig.conf.sample (added), + configs/samples/res_snmp.conf.sample (added), + configs/acl.conf.sample (removed), + configs/samples/smdi.conf.sample (added), + configs/samples/cel.conf.sample (added), + configs/cli_aliases.conf.sample (removed), + configs/samples/cdr_sqlite3_custom.conf.sample (added), + configs/extensions.ael.sample (removed), + configs/cdr_adaptive_odbc.conf.sample (removed), + configs/samples/phone.conf.sample (added), + configs/extensions_minivm.conf.sample (removed), + configs/motif.conf.sample (removed), configs/telcordia-1.adsi + (removed), configs/samples/meetme.conf.sample (added), + configs/adsi.conf.sample (removed), configs/alsa.conf.sample + (removed), configs/samples/muted.conf.sample (added), + configs/followme.conf.sample (removed), + configs/asterisk.conf.sample (removed), + configs/samples/iax.conf.sample (added), + configs/samples/res_config_sqlite3.conf.sample (added), + configs/samples/mgcp.conf.sample (added), + configs/cel_odbc.conf.sample (removed), configs/ccss.conf.sample + (removed), configs/cdr_mysql.conf.sample (removed), + configs/samples/extensions.lua.sample (added), + configs/samples/say.conf.sample (added), + configs/dundi.conf.sample (removed), + configs/samples/queuerules.conf.sample (added), + configs/oss.conf.sample (removed), configs/app_mysql.conf.sample + (removed), configs/samples/confbridge.conf.sample (added), + configs/samples/cli_permissions.conf.sample (added), + configs/samples/enum.conf.sample (added), + configs/samples/config_test.conf.sample (added), + configs/cdr.conf.sample (removed), + configs/samples/indications.conf.sample (added), + configs/cel_pgsql.conf.sample (removed), + configs/res_stun_monitor.conf.sample (removed), + configs/calendar.conf.sample (removed), + configs/samples/voicemail.conf.sample (added), + configs/pjsip_notify.conf.sample (removed), + configs/samples/misdn.conf.sample (added), + configs/samples/ari.conf.sample (added), + configs/samples/festival.conf.sample (added), + configs/samples/http.conf.sample (added), + configs/res_snmp.conf.sample (removed), + configs/samples/musiconhold.conf.sample (added), + configs/samples/pjsip.conf.sample (added), + configs/samples/sorcery.conf.sample (added), + configs/samples/vpb.conf.sample (added), configs/smdi.conf.sample + (removed), configs/samples/unistim.conf.sample (added), + configs/samples (added), configs/amd.conf.sample (removed): + configs: Move sample config files into a subdirectory of configs + This moves all samples configs from configs/ to configs/samples. + This allows for additional sets of sample configuration files to + be added in the future. Review: + https://reviewboard.asterisk.org/r/3804/ + + * channels/chan_sip.c, UPGRADE.txt: chan_sip: Make + progressinband=never really mean 'never' progressinband=never in + sip.conf is easily defeated if an onward trunk sends a progress + indication of its own. This is almost certain to happen if the + onward trunk is ISDN or IAX as these technologies send a progress + indication even if early media is not required. This progress + message is passed to the caller, and causes the "never" option to + be rather badly named. This patch changes the behaviour of this + setting in the following ways: 1) In sip_write(), do not pass the + media unless we have either progressed beyond INV_EARLY_MEDIA, or + we are in INV_EARLY_MEDIA state, and early media is both set-up + and wanted. This helps resolve double-ringing on some buggy + handsets. 2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, + but SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to + avoid implicitly enabling early media. Avoid sending double ring + indications. NOTE: the meaning of the SIP_PROGRESS_SENT flag + changes slightly in this patch to also encapsulate the fact that + a channel has *sent or received* a 183 Progress indication. This + makes the updated code in sip_write() much more simple. Review: + https://reviewboard.asterisk.org/r/3700 ASTERISK-23972 #close + Reported by: Steve Davies patches: + inband_never_present_early_media2 uploaded by Steve Davies + (License 5012) + + * menuselect: Add svn:ignore property + + * UPGRADE.txt, menuselect/configure, menuselect/configure.ac, + configure, configure.ac: configure: Fix libxml2 development + library dependency checking The commit that added libxml2 support + didn't fully check for the libxml2 development script in the + Asterisk configure file. As a result, Asterisk could be + configured, then fail on menuselect. This patch fixes it so that + Asterisk should detect the libxml2 dependency failure first. + + * menuselect/makeopts.in, menuselect/autoconfig.h.in, + menuselect/menuselect.h, menuselect/example_menuselect-tree, + configure, include/asterisk/autoconfig.h.in, menuselect/Makefile, + menuselect/README, menuselect/aclocal.m4, configure.ac, + UPGRADE.txt, menuselect/configure, menuselect/configure.ac, + menuselect/menuselect.c, menuselect/acinclude.m4: menuselect: Add + libxml2 support (Patch 3) This is the final patch in adding + menuselect to Asterisk. - The first patch (r418832) added + menuselect along with mxml - The second patch (r418833) removed + mxml from menuselect This patch adds support for libxml2 to + menuselect, and makes libxml2 a required library for Asterisk. + Note that the libxml2 portion of this patch was written by Sean + Bright, and was made available on a team branch: + http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/ + Review: https://reviewboard.asterisk.org/r/3773/ ASTERISK-20703 + #close patches: some_mysterious_team_branch uploaded by + seanbright (License 5060) + + * menuselect/mxml (removed): menuselect: Remove mxml from + menuselect (Patch 2) This is the second patch that adds + menuselect to Asterisk trunk. The previous commit (r418832) added + menuselect along with mxml; this patch removes mxml completely + from Menuselect. A subsequent patch will switch menuselect over + to using libxml2, and make libxml2 a required dependency for + Asterisk. ASTERISK-20703 + + * menuselect/mxml/configure.in (added), menuselect/acinclude.m4 + (added), menuselect/mxml/mxml.list.in (added), + menuselect/mxml/README (added), menuselect/linkedlists.h (added), + menuselect/mxml (added), menuselect/mxml/config.h.in (added), + menuselect/aclocal.m4 (added), menuselect/install-sh (added), + menuselect/mxml/mxml-string.c (added), + menuselect/menuselect_stub.c (added), menuselect/make_version + (added), menuselect/mxml/mxml-entity.c (added), + menuselect/bootstrap.sh (added), menuselect/makeopts.in (added), + menuselect/autoconfig.h.in (added), menuselect/config.guess + (added), menuselect/mxml/install-sh (added), + menuselect/test/build_tools/menuselect-deps (added), /, + menuselect/contrib/menuselect-dummy (added), + menuselect/config.sub (added), menuselect/mxml/configure (added), + menuselect/mxml/Makefile.in (added), menuselect (added), + menuselect/contrib (added), menuselect/mxml/mxml.pc.in (added), + menuselect/configure.ac (added), menuselect/mxml/mxml-set.c + (added), menuselect/contrib/Makefile-dummy (added), + menuselect/mxml/ANNOUNCEMENT (added), menuselect/missing (added), + menuselect/menuselect_curses.c (added), + menuselect/example_menuselect-tree (added), menuselect/Makefile + (added), menuselect/mxml/mxml-search.c (added), menuselect/test + (added), menuselect/test/menuselect-tree (added), + menuselect/mxml/mxml.h (added), menuselect/mxml/mxml-index.c + (added), menuselect/configure (added), + menuselect/menuselect_newt.c (added), menuselect/mxml/mxml-attr.c + (added), menuselect/mxml/mxml-private.c (added), + menuselect/menuselect.c (added), menuselect/mxml/CHANGES (added), + menuselect/mxml/COPYING (added), menuselect/mxml/mxml-file.c + (added), menuselect/menuselect.h (added), + menuselect/menuselect_gtk.c (added), menuselect/README (added), + menuselect/strcompat.c (added), menuselect/mxml/mxml-node.c + (added), menuselect/test/build_tools (added): menuselect: Add + menuselect to Asterisk trunk (Patch 1) This is the first patch + that adds menuselect to Asterisk trunk, and removes the + svn:externals property. This is being done for two reasons: (1) + The removal of external repositories eases a future migration to + git (2) Asterisk is now the only thing that uses menuselect; as a + result, there's little need to keep it in an external repository + Subsequent patches will remove the mxml dependency from + menuselect and tidy up the build system. ASTERISK-20703 + +2014-07-17 14:28 +0000 [r418811] Kinsey Moore <kmoore@digium.com> + + * /, main/bridge_channel.c: TEST_FRAMEWORK: Fix threewaytransfer + reporting Ensure that three-way transfers can be reported even if + featuremap is non-NULL. ........ Merged revisions 418810 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-16 23:08 +0000 [r418788] Corey Farrell <git@cfware.com> + + * /, channels/dahdi/bridge_native_dahdi.c: Remove include of + astobj.h from channels/dahdi/bridge_native_dahdi.c. The include + was unneeded, this is split off from r3758 as it applies to 12. + ........ Merged revisions 418787 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-16 14:03 +0000 [r418717-418757] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip/pjsip_configuration.c, CHANGES, res/res_pjsip.c, + channels/chan_pjsip.c, include/asterisk/res_pjsip.h, + contrib/ast-db-manage/config/versions/1d50859ed02e_create_accountcode.py + (added), /, configs/pjsip.conf.sample: res_pjsip: Support setting + a default accountcode on endpoints Most channel drivers let you + specify a default accountcode to be set on channels associated + with a particular peer/endpoint/object. Prior to this patch, + chan_pjsip/res_pjsip did not support such a setting. This patch + adds a new setting to the res_pjsip endpoint object, + 'accountcode'. When a channel is created that is associated with + an endpoint with this value set, the channel will automatically + have its accountcode property set to the value configured for the + endpoint. Review: https://reviewboard.asterisk.org/r/3724/ + ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged + revisions 418756 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * cdr/cdr_pgsql.c, CHANGES, configs/cdr_pgsql.conf.sample, + configs/res_pgsql.conf.sample, cel/cel_pgsql.c, + res/res_config_pgsql.c, configs/cel_pgsql.conf.sample: cel_pgsql, + cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name + support This patch adds support for the PostgreSQL + application_name connection setting. When the appropriate + PostgreSQL module's configuration is set with an application + name, the name will be passed to PostgreSQL on connection and + displayed in the database's pg_stat_activity view, as well as in + CSV logs. This aids in managing which applications/servers are + connected to a PostgreSQL database, as well as tracing the + activity of those connections. Review: + https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close + Reported by: Gergely Domodi patches: pgsql_application_name.patch + uploaded by Gergely Domodi (License 6610) + + * codecs/codec_adpcm.c, main/format.c: codec_adpcm: Change + description of codec "ADPCM" to "Dialogic ADPCM" Technically, + ADPCM is a method that can be applied to several codecs. + Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec. See + http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information + about said codec. Review: https://reviewboard.asterisk.org/r/3744 + patches: rb3744.patch uploaded by dennis.guse (License 6513) + + * UPGRADE.txt, main/manager.c, /: manager: Return ActionID on + nominal responses to PresenceState action When the PresenceState + action is executed, the nominal path fails to include the + ActionID in the successful response. This patch adds a call to + astman_start_ack, which guarantees that an ActionID (if provided) + will be sent back to the AMI client. Unlike the Asterisk 11 and + 12 patches, this patch also deprecates the duplicate Message key + in the response to the action, replacing it with the key + 'PresenceMessage'. Review: + https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close + ........ Merged revisions 418713 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418714 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-15 23:03 +0000 [r418716] Kinsey Moore <kmoore@digium.com> + + * /, main/bridge_channel.c: TEST_FRAMEWORK: Fix ref leak in feature + activation This fixes two reference leaks that would occur when + TEST_FRAMEWORK was enabled and features were successfully + executed. ........ Merged revisions 418715 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-15 17:57 +0000 [r418654] Jonathan Rose <jrose@digium.com> + + * funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty + strings as argument Previously these two dialplan functions would + issue warnings and return failure when an empty string is used as + the argument. Now they will not issue a warning and will + successfully return an empty string. ASTERISK-23911 #close + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3745/ ........ Merged + revisions 418641 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 418649 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-15 12:11 +0000 [r418616] Sean Bright <sean@malleable.com> + + * main/asterisk.c: Update Asterisk copyright year in + main/asterisk.c It's been 2014 for like... 6 months. + +2014-07-14 14:55 +0000 [r418566-418587] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/logger.h, /: logger.h: Extract DEBUG_ATLEAST() + to complement VERBOSITY_ATLEAST(). ........ Merged revisions + 418586 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/jabber.h (removed), include/asterisk/jingle.h + (removed), include/asterisk/frame_defs.h (removed), + configs/h323.conf.sample (removed): Actually delete the removed + files. + +2014-07-13 21:57 +0000 [r418507] Corey Farrell <git@cfware.com> + + * /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work + around REF_DEBUG race which causes out of order log entries * + Update refcounter.py to use delta's to track the current + reference count. * Use result from internal_ao2_ref to write + old_refcount to refs_log. Review: + https://reviewboard.asterisk.org/r/3756/ ........ Merged + revisions 418504 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 418505 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418506 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-13 20:08 +0000 [r418488] Scott Griepentrog <sgriepentrog@digium.com> + + * include/asterisk/astobj2.h: astobj2: correct define for + ao2_t_cleanup This change maps the ao2_t_cleanup() function to + the correct debug function so that it can be used. Review: + https://reviewboard.asterisk.org/r/3764/ + +2014-07-13 16:48 +0000 [r418448-418467] Corey Farrell <git@cfware.com> + + * main/manager.c, /, apps/app_skel.c: Fix minor reference leaks in + app_skel and TEST_FRAMEWORK * Cleanup games object in app_skel. * + Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+). + Review: https://reviewboard.asterisk.org/r/3757/ ........ Merged + revisions 418465 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418466 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/jabber.h, include/asterisk/jingle.h, + configs/h323.conf.sample: Remove files left behind on removal of + h323, jingle and jabber. This change removes h323.conf.sample, + jingle.h, jabber.h left behind by r3698. Review: + https://reviewboard.asterisk.org/r/3755/ + +2014-07-11 23:00 +0000 [r418419] Matthew Jordan <mjordan@digium.com> + + * main/astobj2.c, include/asterisk/astobj2.h: astobj2: Add tag + variants for ao2_bump, ao2_cleanup, and ao2_replace Tags are + useful in hunting down ref imbalances; this patch adds tag + variants for these commonly used macros/functions. Review: + https://reviewboard.asterisk.org/r/3750/ + +2014-07-11 21:10 +0000 [r418397] Corey Farrell <git@cfware.com> + + * /, include/asterisk/astobj2.h: astobj2: tweak ao2_replace to do + nothing when it would be a NoOp This change causes ao2_replace to + do nothing when src == dst. This avoids REF_DEBUG logging when + we're not actually doing anything. Review: + https://reviewboard.asterisk.org/r/3743/ ........ Merged + revisions 418396 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-11 16:42 +0000 [r418370] Scott Griepentrog <sgriepentrog@digium.com> + + * /, main/config.c: config: inform config hook of change when + writing file When updated configuration is written back to the + conf file - for example when a user changes their voicemail pin, + make sure that any config hook that wants to know of changes is + informed. Review: https://reviewboard.asterisk.org/r/3708/ + ........ Merged revisions 418366 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418369 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-10 15:36 +0000 [r418325] Matthew Jordan <mjordan@digium.com> + + * /, include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert + indentation to tabs This is a whitespace only change. ........ + Merged revisions 418323 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-10 01:59 +0000 [r418226-418264] Richard Mudgett <rmudgett@digium.com> + + * channels/sig_pri.c, /: chan_dahdi/sig_pri: Fix type mismatch in + the idledial feature's channel creation. Square pegs in round + holes don't work very well. ........ Merged revisions 418261 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 418262 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418263 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/stasis/stasis_bridge.h (added), main/bridge_channel.c, + res/res_stasis.c, /, res/stasis/stasis_bridge.c (added), + include/asterisk/bridge_channel.h, main/bridge_basic.c: ARI: Make + mixing bridges propagate linkedids and accountcodes. * Create a + Stasis bridge sub-class to propagate linkedids and accountcodes. + * Fixed the basic bridge sub-class to update peeraccount codes + when the number of channels in the bridge drops back down to two + parties. * Refactored ast_bridge_channel_update_accountcodes() to + handle channels joining/leaving the bridge. * Fixed the basic + bridge sub-class to not call the base bridge class pull method + twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard + Mudgett Review: https://reviewboard.asterisk.org/r/3720/ ........ + Merged revisions 418225 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-08 14:48 +0000 [r418174-418183] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json: manager/ARI: Update version to + 2.4.0/1.4.0; Update UPGRADE.txt ........ Merged revisions 418182 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix undefined + function when PJPROJECT is not installed The + dtls_perform_handshake function was mistakenly placed under the + guards for USE_PJPROJECT. If PJPROJECT was not installed, the + function would not be defined, while other functions would + attempt to still use it. This prevented res_rtp_asterisk from + being loaded. ASTERISK-24001 #close Reported by: Don Fanning + ........ Merged revisions 418172 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-07 16:08 +0000 [r418117] Joshua Colp <jcolp@digium.com> + + * include/asterisk/res_pjsip_body_generator_types.h, + res/res_pjsip_dialog_info_body_generator.c (added), + res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c, /, + include/asterisk/res_pjsip_presence_xml.h: + res_pjsip_dialog_info_body_generator: Add dialog-info+xml support + for presence. This module implements dialog-info+xml for the + purposes of presence. This means that phones such as Grandstreams + can now subscribe to receive presence information for an + extension. ASTERISK-21443 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3705/ ........ Merged + revisions 418116 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-07 02:15 +0000 [r418090] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/stasis_app.h, res/ari/resource_channels.c, + res/res_stasis.c, /, res/stasis/app.c: ARI/res_stasis: Subscribe + to both Local channel halves when originating to app This patch + fixes two bugs: 1. When originating a channel into a Stasis + application, we already create a subscription for the channel + that is going into our Stasis app. Unfortunately, when you create + a Local channel and pass it off to a Stasis app, you really + aren't creating just one channel: you're creating two. This patch + snags the second half of the Local channel pair (assuming it is a + Local channel pair, but luckily core_local is kind about such + assumptions) and subscribes to it as well. 2. Subscriptions are a + bit sticky right now. If a subscription is made, the 'interest' + count gets bumped on the Stasis subscription - but unless + something explicitly unsubscribes the channel, said subscription + sticks around. This is not much of a problem is a user is + creating the subscription - if they made it, they must want it. + However, when we are creating implicit subscriptions, we need to + make sure something clears them out. This patch takes a + pessimistic approach: it watches the cache updates coming from + Stasis and, if we notice that the cache just cleared out an + object, we delete our subscription object. This keeps our ao2 + container of Stasis forwards in an application from growing out + of hand; it also is a bit more forgiving for end users who may + not realize they were supposed to unsubscribe from that channel + that just hung up. Review: + https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close + ........ Merged revisions 418089 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-07 01:22 +0000 [r418067-418084] Kinsey Moore <kmoore@digium.com> + + * tests/test_cel.c, main/cel.c, channels/chan_pjsip.c, + res/res_pjsip_session.c, /: CEL: Fix incorrect/missing extra + field information This corrects two issues with the extra field + information in Asterisk 12+ in channel event logs. It is possible + to inject custom values into the dialstatus provided by + ast_channel_dial_type() Stasis messages that fall outside the + enumeration allowed for the DIALSTATUS channel variable. CEL now + filters for the allowed values and ignores other values. The + "hangupsource" extra field key is always blank if the far end + channel is a chan_pjsip channel. This is because the hangupsource + is never set for the pjsip channel driver. This change sets the + hangupsource whenever a hangup is queued for chan_pjsip channels. + This corrects an issue with the pjsip channel driver where the + hangupcause information was not being set properly. Review: + https://reviewboard.asterisk.org/r/3690/ ........ Merged + revisions 418071 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/http.c: HTTP: Fix build for gcc 4.10 ........ Merged + revisions 418066 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-04 15:26 +0000 [r418019-418050] Matthew Jordan <mjordan@digium.com> + + * main/Makefile: main/Makefile: fix compilation error of buildinfo + occurring on 'make install' Egads. Another bad deletion of too + much when attempting to remove h323 stuff. + + * configure.ac, build_tools/menuselect-deps.in, configure, + main/Makefile: configure: Remove last vestiges of h323; DO create + menuselect-deps The previous patch (r418034) fixed the 'glitch' + that the channels/h323 Makefile no longer existed. Unfortunately, + removing the entire line was a bit of a blunder, as it meant that + build_tools/menuselect-deps was never generated. Hilarity ensued + when actually trying to compile. But hey! At least configure + worked. This patch fixes *that* glitch, and removes some more of + the vestiges of h323. (It had tendrils in the main Makefile? + Crazy.) + + * configure.ac, configure: configure: Update script to pass if + channels/h323/Makefile.in does not exist This simply removes that + check from the configure script, as r418019 removed chan_h323. + + * apps/app_dahdibarge.c (removed), configs/gtalk.conf.sample + (removed), main/pbx.c, apps/app_readfile.c (removed), + channels/chan_sip.c, configs/jingle.conf.sample (removed), + UPGRADE.txt, res/res_musiconhold.c, channels/chan_gtalk.c + (removed), channels/Makefile, CHANGES, res/res_jabber.c + (removed), channels/h323 (removed), utils/conf2ael.c, + channels/chan_jingle.c (removed), res/ael/pval.c, + configs/jabber.conf.sample (removed), + configs/asterisk.conf.sample, res/res_agi.c, channels/chan_h323.c + (removed), addons/Makefile, pbx/pbx_realtime.c, utils/ael_main.c, + include/asterisk/options.h, main/asterisk.c, + addons/app_saycountpl.c (removed): Remove many deprecated modules + Billing records are fair, To get paid is quite bright, You should + really use ODBC; Good-bye cdr_sqlite. Microsoft did once push + H.323, Hell, we all remember NetMeeting. But try to compile + chan_h323 now And you will take quite a beating. The XMPP and SIP + war was fierce, And in the distant fray Was birthed + res_jabber/chan_jingle; But neither to stay. For everyone did + care and chase what Google professed. "Free Internet Calling" was + what devotees cried, But Google did change the specs so often + That the developers were happy the day chan_gtalk died. And then + there was that odd application Dedicated to the Polish tongue. + app_saycountpl was subsumed by Say; One could say its bell was + rung. To read and parse a file from the dialplan You could (I + guess) use an application. app_readfile did fill that purpose, + but I think A function is perhaps better in its creation. Barging + is rude, I'm not sure why we do it. Inwardly, the caller will + probably sigh. But if you really must do it, Don't use + app_dahdibarge, use ChanSpy. We all despise the sound of tinny + robots It makes our queues so cold. To control such an + abomination It's better to not use Wait/SetMusicOnHold. It's + often nice to know properties of a channel It makes our calls + right We have a nice function called CHANNEL And so SIPCHANINFO + is sent off into the night. And now things get odd; Apparently + one could delimit with a colon Properties from the SIPPEER + function! Commas are in; all others are done. Finally, a word on + pipes and commas. We're sorry. We can't say it enough. But those + compatibility options in asterisk.conf; To maintain them forever + was just too tough. This patch removes: * cdr_sqlite * chan_gtalk + * chan_jingle * chan_h323 * res_jabber * app_saycountpl * + app_readfile * app_dahdibarge It removes the following + applications/functions: * WaitMusicOnHold * SetMusicOnHold * + SIPCHANINFO It removes the colon delimiter from the SIPPEER + function. Finally, it also removes all compatibility options that + were configurable from asterisk.conf, as these all applied to + compatibility with Asterisk 1.4 systems. Review: + https://reviewboard.asterisk.org/r/3698/ + +2014-07-03 22:22 +0000 [r417933-417976] Richard Mudgett <rmudgett@digium.com> + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, /, UPGRADE.txt, + channels/sig_pri.c: chan_dahdi: Add inband_on_setup_ack + compatibility option. The new inband_on_setup_ack option causes + Asterisk to assume inband audio may be present when a + SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says + that in scenarios with overlap dialing, when a dialtone is sent + from the network side, progress indicator 8 "Inband info now + available" MAY be sent to the CPE if no digits were received with + the SETUP. It is thus implied that the ie is mandatory if digits + came with the SETUP and dialtone is needed. This option should be + enabled, when the network sends dialtone and you want to hear it, + but the network doesn't send the progress indicator when needed. + NOTE: For Q.SIG setups this option should be enabled when + outgoing overlap dialing is also enabled because Q.SIG does not + send the progress indicator with the SETUP ACK. The commit + -r413714 (AST-1338) which causes this issue was dealing with a + SIP-to-ISDN interoperability issue. This commit is a merge of the + two patches indicated below. ASTERISK-23897 #close Reported by: + Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded + by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) + patch uploaded by rmudgett Review: + https://reviewboard.asterisk.org/r/3633/ ........ Merged + revisions 417956 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417957 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417958 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_channels.c, res/res_ari.c, main/manager.c, /: + res_ari: Fix some off-nominal paths just dropping the HTTP + connection. * Removed some incorrect newlines on ast_http_error() + messages in manager.c. * Removed an incorrect newline in + res_ari_channels.c. Addendum to ASTERISK-23552 ........ Merged + revisions 417932 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-03 17:34 +0000 [r417910-417916] Jonathan Rose <jrose@digium.com> + + * CHANGES, channels/chan_dahdi.c: chan_dahdi: Add AMI commands for + controlling PRI debugging output Adds the following AMI commands: + PRIDebugSet - Set PRI debug levels for a specific span + PRIDebugFileSet - Set the file used for PRI debug message output + PRIDebugFileUnset - Disables file output for PRI debug messages + Review: https://reviewboard.asterisk.org/r/3681/ + + * CHANGES, pbx/pbx_config.c, main/pbx.c: pbx_config: Add manager + actions to add/remove extensions Adds two new manager commands to + pbx_config - DialplanExtensionAdd and DialplanExtensionRemove + which allow manager users to create and delete extensions + respectively. Review: https://reviewboard.asterisk.org/r/3650/ + +2014-07-03 17:16 +0000 [r417901] Richard Mudgett <rmudgett@digium.com> + + * res/res_phoneprov.c, main/http.c, UPGRADE.txt, + include/asterisk/tcptls.h, res/res_http_post.c, + res/res_http_websocket.c, configs/http.conf.sample, + include/asterisk/http.h, main/tcptls.c, res/res_ari.c, + main/manager.c, /: HTTP: Add persistent connection support. + Persistent HTTP connection support is needed due to the increased + usage of the Asterisk core HTTP transport and the frequency at + which REST API calls are going to be issued. * Add http.conf + session_keep_alive option to enable persistent connections. * + Parse and discard optional chunked body extension information and + trailing request headers. * Increased the maximum + application/json and application/x-www-form-urlencoded body size + allowed to 4k. The previous 1k was kind of small. * Removed a + couple inlined versions of ast_http_manid_from_vars() by calling + the function. manager.c:generic_http_callback() and + res_http_post.c:http_post_callback() * Add missing va_end() in + ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use + in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott + Griepentrog Review: https://reviewboard.asterisk.org/r/3691/ + ........ Merged revisions 417880 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-03 16:55 +0000 [r417900] Matthew Jordan <mjordan@digium.com> + + * main/tcptls.c, configure, include/asterisk/autoconfig.h.in, + configure.ac: main/tcptls: Add checks for OpenSSL Elliptic Curve + support The patch for ASTERISK-23905 that added PFS support in + Asterisk depends on the elliptic curve library support being + present in OpenSSL. As it turns out, some versions of OpenSSL + don't have this library - notably the version running on our + build agents. This patch fixes the build by providing a configure + check for the specific library calls that the PFS patch relies + on. Review: https://reviewboard.asterisk.org/r/3709/ + +2014-07-03 16:14 +0000 [r417877-417879] sgalarneau <sgalarneau@localhost>: + + * res/ari/resource_events.h, rest-api/api-docs/channels.json, + res/ari/resource_channels.h, rest-api/api-docs/events.json, /: + ARI: Improvements to body parameters documentation The variables + body parameter under the originate and originate with id + operations of the channel resource showed invalid JSON in its + description. The variables body parameter under the userEvent + operation of the event resource made no mention that the custom + key/value pairs should be wrapped in a variables key in order to + be added to the custom user event. ASTERISK-23975 #close Review: + https://reviewboard.asterisk.org/r/3692/ ........ Merged + revisions 417878 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api-templates/api.wiki.mustache, + rest-api-templates/swagger_model.py, /: api.wiki.mustache: Update + wiki template to support body parameters This patch updates the + api.wiki.mustache template and the swagger_model python script to + understand if an operation has a body parameter. If an operation + does have a body parameter, it will now be displayed in the + corresponding wiki entry. ........ Merged revisions 407389 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-03 14:08 +0000 [r417863] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * Makefile, contrib/scripts/dahdi_span_config_hook (added): + dahdi_span_config_hook: automatically register new dahdi channels + Install a hook script for DAHDI to register new spans with + Asterisk automatically by running: asterisk -rx 'dahdi create + channel FIRST LAST' Review: + https://reviewboard.asterisk.org/r/3157/ + +2014-07-03 12:10 +0000 [r417800-417803] Matthew Jordan <mjordan@digium.com> + + * main/tcptls.c, CHANGES: main/tcptls: Add support for Perfect + Forward Secrecy This patch enables Perfect Forward Secrecy (PFS) + in Asterisk's core TLS API. Modules that wish to enable PFS + should consider the following: - Ephemeral ECDH (ECDHE) is + enabled by default. To disable it, do not specify a ECDHE cipher + suite in a module's configuration, for example: + tlscipher=AES128-SHA:DES-CBC3-SHA - Ephemeral DH (DHE) is + disabled by default. To enable it, add DH parameters into the + private key file, i.e., tlsprivatekey. For an example, see the + default dh2048.pem at + http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt + - Because clients expect the server to prefer PFS, and because + OpenSSL sorts its cipher suites by bit strength, (see "openssl + ciphers -v DEFAULT") consider re-ordering your cipher suites in + the conf file. For example: + tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH + will use PFS when offered by the client. Clients which do not + offer PFS fall-back to AES-128 (or even 3DES as recommend by RFC + 3261). Review: https://reviewboard.asterisk.org/r/3647/ + ASTERISK-23905 #close Reported by: Alexander Traud patches: + tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520) + tlsPFS.patch uploaded by Alexander Traud (License 6520) + + * /, main/utils.c: main/untils: Prevent potential infinite loop in + ast_careful_fwrite A loop in ast_careful_fwrite exists that will + continually attempt to write to a file stream, even in the + presence of EAGAIN/EINTR errors. However, if a connection that + uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's + call to fflush may return EAGAIN/EINTER along with EOF. A + subsequent call to fflush will return EOF but not clear errno, + resulting in an infinite loop. This patch clears errno after it + is detected and handled the loop, such that any subsequent call + to fflush will not get erroneously stuck. Review: + https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close + Reported by: Steve Davies patches: fflush_loop_fix uploaded by + one47 (License 5012) ........ Merged revisions 417797 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417798 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417799 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-02 21:13 +0000 [r417770] Jonathan Rose <jrose@digium.com> + + * res/ari/resource_events.h, res/ari/resource_asterisk.h, + res/ari/resource_applications.h, res/ari/resource_playbacks.h, + res/ari/resource_channels.h, res/ari/resource_sounds.h, /, + res/ari/resource_bridges.h, res/ari/resource_recordings.h, + rest-api-templates/ari_resource.h.mustache, + res/ari/resource_device_states.h, res/ari/resource_endpoints.h, + res/ari/resource_mailboxes.h: ARI: Remove unnecessary \briefs + from automatically generated documentation Review: + https://reviewboard.asterisk.org/r/3440/ ........ Merged + revisions 412653 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-01 14:42 +0000 [r417679-417706] Joshua Colp <jcolp@digium.com> + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Don't leak memory or + reset state if DTLS configuration is set multiple times. ........ + Merged revisions 417705 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, + contrib/ast-db-manage/config/versions/51f8cb66540e_add_further_dtls_options.py + (added), include/asterisk/res_pjsip_session.h, main/rtp_engine.c, + /, channels/chan_sip.c, main/sdp_srtp.c, res/res_pjsip_sdp_rtp.c, + res/res_pjsip/pjsip_configuration.c, configs/sip.conf.sample, + include/asterisk/rtp_engine.h, res/res_pjsip.c, + channels/sip/include/sip.h, include/asterisk/res_pjsip.h, + include/asterisk/sdp_srtp.h: Recorded merge of revisions 417677 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS + negotiation on RTCP. This change fixes up DTLS support in + res_rtp_asterisk so it can accept and provide a SHA-256 + fingerprint, so it occurs on RTCP, and so it occurs after ICE + negotiation completes. Configuration options to chan_sip and + chan_pjsip have also been added to allow behavior to be tweaked + (such as forcing the AVP type media transports in SDP). + ASTERISK-22961 #close Reported by: Jay Jideliov Review: + https://reviewboard.asterisk.org/r/3679/ Review: + https://reviewboard.asterisk.org/r/3686/ ........ Merged + revisions 417678 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-30 18:39 +0000 [r417663] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c: Reverse logic during subscription + persistence recreation. In the abstraction effort, this bit of + logic got messed up. We want to recreate the persistence if + things go well, not if things fail. + +2014-06-30 13:02 +0000 [r417590-417649] Matthew Jordan <mjordan@digium.com> + + * apps/app_voicemail.c: apps/app_voicemail: Fix compilation error + introduced in r417591 Not sure why that change to + ast_channel_alloc was made but ... okay. + + * apps/app_voicemail.c, main/say.c, CHANGES: app_voicemail, say: + Add support for Japanese Language This patch adds support for the + Japanese language to both the say family of applications, as well + as for VoiceMail and VoiceMailMain. A new pack of language sounds + will be released at the same time as the next major version of + Asterisk to support the new language features. The language + features can be enabled using a language code of 'ja'. Review: + https://reviewboard.asterisk.org/r/3477 ASTERISK-23324 #close + Reported by: Kevin McCoy patches: + app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy + (License 6586) say.c.20140226.jb.patch uploaded by Kevin McCoy + (License 6586) + + * /, channels/chan_sip.c: chan_sip: be more tolerant of whitespace + between attributes in SDP fmtp line This patch is essentially a + backport of a small portion of r397526 from ASTERISK-21981. In + that patch, pass through support and format attribute negotiation + was added for Opus. Part of that included being more tolerant to + whitespace in the fmtp line of an SDP; that part of the patch is + being applied here. As the author of the backport pointed out, in + SDP, the fmtp line is allowed to include whitespace between + attributes. RFC 3267 chapter 8.3 (from 2001) includes an example + for this. This was not removed in the updated RFC 4867 in 2007. + Review: https://reviewboard.asterisk.org/r/3658 #ASTERISK-23916 + #close Reported by: Alexander Traud patches: + sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud + (License 6520) ........ Merged revisions 417587 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417588 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417589 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 23:21 +0000 [r417571] Richard Mudgett <rmudgett@digium.com> + + * /, main/event.c: event.c: Fix type mismatch errors in ie_maps[]. + In v12+ the type values from the table are only used by the CEL + unit tests. Since the unit tests were only comparing a generated + expected event with a real event to see if the ie contents + matched and using the same table IE_PLTYPE values to read the + event contents, the type mismatches were not detected. ........ + Merged revisions 417565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 19:27 +0000 [r417485-417511] Corey Farrell <git@cfware.com> + + * /, main/astobj2.c: Ensure REF_DEBUG records entrys for attempts + to ao2_ref an invalid object This change ensures that + __ao2_ref_debug writes to ref_log when given a non-NULL pointer + to an invalid ao2 object. This is to ensure that we record any + attempt manipulate references of already freed objects. + ASTERISK-23948 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3677/ ........ Merged + revisions 417500 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417505 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417509 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, contrib/scripts/refcounter.py: refcounter.py: prevent use of + excessive RAM with large refs logs When processing a 212MB refs + file, refcounter.py used over 3GB of RAM. This change greatly + reduces memory usage in two ways: * Saving object history in + whole lines instead of separated values. * Not saving + normal/skewed/leaked object lists unless they are requested. + ASTERISK-23921 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3668/ ........ Merged + revisions 417480 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417481 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417483 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 13:50 +0000 [r417461] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip/pjsip_configuration.c, res/res_pjsip_pubsub.c, + res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h, /, + res/res_pjsip_outbound_registration.c: res_pjsip: Add ActionID to + events created as a result of PJSIP AMI actions A number of + various PJSIP AMI actions were failing to parse out and place the + ActionID into their responses. This patch updates the various + PJSIP actions such that the passed in ActionID is emitted on any + event list complete events, as well as any intermediate events + created as a result of the action. #ASTERISK-23947 #close + Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3675/ ........ Merged + revisions 417460 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 02:04 +0000 [r417423-417447] Kinsey Moore <kmoore@digium.com> + + * tests/test_cel.c: CEL: Update unit tests for bridge tech field + Update the CEL unit tests that handle BRIDGE_ENTER and + BRIDGE_EXIT events to expect the "bridge_technology" extra field + key. + + * CHANGES: CHANGES: Add missing changes Add missing CHANGES changes + from r417361 and r417383. + +2014-06-26 18:27 +0000 [r417400-417421] Matthew Jordan <mjordan@digium.com> + + * res/res_http_websocket.exports.in, /: res_http_websocket: Export + symbol for ast_websocket_set_timeout Thanks to Sean Bright for + pointing out that this was missed in #asterisk-dev. ........ + Merged revisions 417419 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417420 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_pjsip.c, /: chan_pjsip: Add a test event for fast + picture updates This will drive the test on review r3419. Note + that the patch for this was done by Ben Ford, although it was + slightly modified for this commit. ASTERISK-23562 Reported by: + Matt Jordan ........ Merged revisions 417399 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-26 14:48 +0000 [r417361-417383] Kinsey Moore <kmoore@digium.com> + + * main/cel.c: CEL: Add bridge tech to relevant CEL records Add the + "bridge_technology" extra field key to BRIDGE_ENTER and + BRIDGE_EXIT CEL events to convey the bridge technology in use at + the time the record was generated. + + * main/bridge.c, include/asterisk/channel.h, + include/asterisk/bridge_features.h, + tests/test_channel_feature_hooks.c (added), + main/bridge_channel.c, main/channel.c: Bridging: Allow channels + to define bridging hooks This patch allows the current owner of a + channel to define various feature hooks to be made available once + the channel has entered a bridge. This includes any hooks that + are setup on the ast_bridge_features struct such as DTMF hooks, + bridge event hooks (join, leave, etc.), and interval hooks. + Review: https://reviewboard.asterisk.org/r/3649/ + +2014-06-26 12:43 +0000 [r417317-417360] Matthew Jordan <mjordan@digium.com> + + * CHANGES, apps/app_jack.c: app_jack: Support audio with a sampling + rate higher than 8kHz This patch enables the jack-audiohook to + cope with dynamic sampling rates from and to Asterisk. + Information from the channel is taken to derive the channel's + sampling rate, suiting SLINxx format and frame->datalen. There + are stil a few limitations after this patch: * Required + information is taken from the channel during initialization as + the audiohook does not provide this information. + Audiohook.internal_sampl_rate(...) is set later, but no callback + is available to inform app_jack. * Frame.datalen is computed + using "rate / 50" assuming a ptime of 20ms. There is no internal + API available to determine datalen for a SLINxx. * Ringbuffer + size is now dynamic depending on the value of frame.datalen (see + above) and the number of frames, which are in + RINGBUFFER_FRAME_CAPACITY, that need to fit. Review: + https://reviewboard.asterisk.org/r/3618 Note that the patch being + committed here is based on the patch posted on ASTERISK-23836. + However, Matthis Schmieder also provided a patch to enable this + functionality, and that patch is noted below. ASTERISK-20696 + #close Reported by: Matthis Schmieder patches: app_jack.patch + uploaded by Matthis Schmieder (License 6445) ASTERISK-23836 + #close Reported by: Dennis Guse patches: patch-app_jack.c + uploaded by Dennis Guse (License 6513) + + * main/udptl.c, /: udptl: Correct FEC to not consider negative + sequence numbers as missing When using FEC, with span=3 and + entries=4 Asterisk will attempt to repair the packet with + sequence number 5, as it will see that packet -4 is missing. The + result is Asterisk sending garbage packets that can kill a fax. + This patch adds a check to see if the sequence number is valid + before checking if the packet is missing. Review: + https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close + Reported by: Torrey Searle patches: udptl_fec.patch uploaded by + Torrey Searle (License 5334) ........ Merged revisions 417318 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 417320 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/internal.h, configs/ari.conf.sample, + res/res_http_websocket.c, res/res_pjsip.c, + configs/pjsip.conf.sample, include/asterisk/http_websocket.h, + configs/sip.conf.sample, res/res_pjsip/config_transport.c, + res/ari/ari_websockets.c, res/res_pjsip_transport_websocket.c, + res/ari/config.c, channels/sip/include/sip.h, + include/asterisk/res_pjsip.h, res/res_ari.c, /, + channels/chan_sip.c, UPGRADE.txt: res_http_websocket: Close + websocket correctly and use careful fwrite When a client takes a + long time to process information received from Asterisk, a write + operation using fwrite may fail to write all information. This + causes the underlying file stream to be in an unknown state, such + that the socket must be disconnected. Unfortunately, there are + two problems with this in Asterisk's existing websocket code: 1. + Periodically, during the read loop, Asterisk must write to the + connected websocket to respond to pings. As such, Asterisk + maintains a reference to the session during the loop. When + ast_http_websocket_write fails, it may cause the session to + decrement its ref count, but this in and of itself does not break + the read loop. The read loop's write, on the other hand, does not + break the loop if it fails. This causes the socket to get in a + 'stuck' state, preventing the client from reconnecting to the + server. 2. More importantly, however, is that the fwrite in + ast_http_websocket_write fails with a large volume of data when + the client takes awhile to process the information. When it does + fail, it fails writing only a portion of the bytes. With some + debugging, it was shown that this was failing in a similar + fashion to ASTERISK-12767. Switching this over to + ast_careful_fwrite with a long enough timeout solved the problem. + Note that this version of the patch, unlike r417310 in Asterisk + 11, exposes configuration options beyond just chan_sip's + sip.conf. Configuration options to configure the write timeout + have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 + #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3624/ ........ Merged + revisions 417310 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-26 10:06 +0000 [r417251] Corey Farrell <git@cfware.com> + + * /, channels/chan_sip.c: chan_sip: Fix handling of "From" headers + longer than 256 characters From headers were processed using a + 256 character buffer on the stack. This change replaces that with + a heap allocation by ast_strdup. ASTERISK-23790 #close Reported + by: uniken1 Tested by: uniken1 Review: + https://reviewboard.asterisk.org/r/3669/ Patches: + chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes + (license 5674) ........ Merged revisions 417248 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417249 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417250 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-25 20:57 +0000 [r417233] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pidf_body_generator.c, + res/res_pjsip_pubsub.exports.in, res/res_pjsip_mwi.c, + res/res_pjsip_xpidf_body_generator.c: Abstract PJSIP-specific + elements from the pubsub API. This helps to pave the way for RLS + work that is to come. Since this is a self-contained change and + subscription tests still pass, this work is being committed + directly to trunk instead of a working branch. ASTERISK-23865 + #close Review: https://reviewboard.asterisk.org/r/3628 + +2014-06-25 18:57 +0000 [r417213] Corey Farrell <git@cfware.com> + + * main/astobj2_container.c, /: ao2_container node object ignores + REF_DEBUG in all places except one Almost every reference + operation against container node's uses __ao2_alloc or __ao2_ref, + thereby preventing ref logging for the nodes. One node reference + is released with ao2_t_ref, causing refcounter.py to falsely + report skews and leaks for many nodes. ASTERISK-23922 #close + Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3670/ ........ Merged + revisions 417212 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-25 00:45 +0000 [r417193] Damien Wedhorn <voip@facts.com.au> + + * channels/chan_skinny.c: Skinny: cleanup some log messages around + sessions. + +2014-06-24 02:50 +0000 [r417167] Corey Farrell <git@cfware.com> + + * include/asterisk/netsock.h, main/utils.c, main/netsock.c, + include/asterisk/res_pjsip_session.h: Move eid functions to + utils.c, mark netsock.h deprecated Move eid functions from + netsock.c to utils.c. These functions were already published by + utils.h. Flag netsock.h as deprecated and switch + res_pjsip_session.h to use netsock2.h. The only code that still + uses netsock.h is chan_iax2. ASTERISK-23920 #close Reported by: + Corey Farrell Review: https://reviewboard.asterisk.org/r/3661/ + +2014-06-23 18:50 +0000 [r417143] Joshua Colp <jcolp@digium.com> + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Return the length of + data written when sending via ICE instead of 0. ASTERISK-23834 + #close Reported by: Richard Kenner ........ Merged revisions + 417141 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 417142 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-23 16:04 +0000 [r417120] Richard Mudgett <rmudgett@digium.com> + + * /, main/core_unreal.c: core_unreal: Fix off by one buffer + overwrite error. Appending the ;2 to the user supplied ;1 + uniqueid to create the ;2 version if the user did not also supply + an extra uniqueid for the ;2 channel resulted in allocating a + buffer that was one byte too small. * Fix off by one error in + ast_unreal_new_channels() when generating the ;2 uniqueid from + the user suppled ;1 version. * Pulled some long assignment lines + from if tests to improve line break readability in + ast_unreal_new_channels(). ........ Merged revisions 417119 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-23 07:44 +0000 [r417059] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c: + suspended destructions of pri spans on events If a DAHDI span + disappears, we wish for its representation in Asterisk to be + destroyed as well. The information about the span's removal may + come from several paths: 1. DAHDI sends DAHDI_EVENT_REMOVE on + every channel. 2. An extra DAHDI_EVENT_REMOVED is sent on every + subsequent call to DAHDI_GET_EVENT. 3. Every read (including the + internal one by libpri on the D-channel) returns -ENODEV. + Asterisk responsds to DAHDI_EVENT_REMOVE on a channel by + destroying it. Destroying a channel requires holding the channel + list lock (iflock). Destroying a channel that is part of a span + requires holding the span's lock. Destroying a channel from a + context that holds the span lock, while at the same time another + channel is destroyed directly, leads to a deadlock. Solution: + don't destroy span while holding the channels list lock. Thus + changes in this patch: * Deferring removal of PRI spans in + response to events: doomed spans are collected on a list. * + Doomed spans are removed periodically by the monitor thread. * + ENODEV reads from the D-channel will warant the same deferred + removal. Review: https://reviewboard.asterisk.org/r/3548/ + +2014-06-22 18:53 +0000 [r416996] George Joseph <george.joseph@fairview5.com> + + * include/asterisk/astobj2.h, Makefile.rules, Makefile, /: astobj2: + Add an ao2_replace macro to astobj2.h This macro replaces one + object reference with another cleaning up the original. param dst + Pointer to the object that will be cleaned up. param src Pointer + to the object replacing it. src's ref count is bumped if it's + non-NULL. dst's ref count is decremented if it's non-NULL. src is + assigned to dst, This patch was reviewed on IRC by coreyfarrell + and mjordan. Tested by: George Joseph ........ Merged revisions + 416995 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-20 23:18 +0000 [r416872-416935] George Joseph <george.joseph@fairview5.com> + + * /, configure, include/asterisk/autoconfig.h.in: build: Allow + autoconf/ast_ext_tool_check to handle cross-compiling better. + ast_ext_tool_check.m4 isn't handling cases where a path to a + package is provided (E.G. --with-mysqlclient=/some/sysroot) and + the package has a config tool (E.G. mysql_config) and the package + has its own subdirectories in include or lib. For example, + mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but + ast_ext_tool_check sets MYSQLCLIENT_LIB to + ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its + includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not + directly in ${LIBXML2_DIR}/usr/include. Both cause configure to + fail and there are others in the same boat. The problem is caused + by logic in ast_ext_tool_check that overrides the result of the + config tool's --cflags and --libs options if package_DIR is set. + This patch prepends package_DIR (if specified) to the -L and -I + results from the package's config tool instead of overriding + them. A regenerated ./configure and + include/asterisk/autoconfig.h.in are included but can be + regenerated by running ./bootstrap.sh at any time. Tested by: + George Joseph Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3550/ ........ Merged + revisions 416929 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416930 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416931 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * autoconf/ast_ext_tool_check.m4, /: build: Allow + autoconf/ast_ext_tool_check to handle cross-compiling better. + ast_ext_tool_check.m4 isn't handling cases where a path to a + package is provided (E.G. --with-mysqlclient=/some/sysroot) and + the package has a config tool (E.G. mysql_config) and the package + has its own subdirectories in include or lib. For example, + mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but + ast_ext_tool_check sets MYSQLCLIENT_LIB to + ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its + includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not + directly in ${LIBXML2_DIR}/usr/include. Both cause configure to + fail and there are others in the same boat. The problem is caused + by logic in ast_ext_tool_check that overrides the result of the + config tool's --cflags and --libs options if package_DIR is set. + This patch prepends package_DIR (if specified) to the -L and -I + results from the package's config tool instead of overriding + them. Tested by: George Joseph Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3550/ ........ Merged + revisions 416870 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416871 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-20 20:57 +0000 [r416848-416850] Jonathan Rose <jrose@digium.com> + + * res/parking/parking_manager.c, /: res_parking: Make manager + commands register with module information Previously module + information was not included due to an oversight. Review: + https://reviewboard.asterisk.org/r/3626/ ........ Merged + revisions 416849 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/logger.c, CHANGES, include/asterisk/logger.h, + main/manager.c: Logger: Add manager command 'LoggerRotate' to + rotate logger Part of a series of AMI command equivalents to + existing CLI commands Review: + https://reviewboard.asterisk.org/r/3651/ + +2014-06-20 17:06 +0000 [r416830] Richard Mudgett <rmudgett@digium.com> + + * apps/app_voicemail.c, include/asterisk/app.h, main/app.c, + apps/app_directory.c, apps/app_chanspy.c: voicemail API + callbacks: Extract the sayname API call to its own registerd + callback. * Extract the sayname API call to its own registerd + callback. This allows the app_directory and app_chanspy + applications to say a mailbox owner's name using an alternate + provider when app_voicemail is not available because you are + using res_mwi_external. app_directory still uses the + voicemail.conf file. AFS-64 #close Reported by: Mark Michelson + +2014-06-20 15:27 +0000 [r416738-416807] George Joseph <george.joseph@fairview5.com> + + * main/astobj2_private.h, main/astobj2_container_private.h, + main/astobj2_container.c, main/astobj2_hash.c, + main/astobj2_rbtree.c, build_tools/cflags.xml, /, + tests/test_astobj2.c: astobj2: Additional refactoring to push + impl specific code down into the impls. Move some implementation + specific code from astobj2_container.c into astobj2_hash.c and + astobj2_rbtree.c. This completely removes the need for + astobj2_container to switch on RTTI and it no longer has any + knowledge of the implementation details. Also adds AO2_DEBUG as a + new compile option in menuselect which controls astobj2 debugging + independently of AST_DEVMODE and REF_DEBUG. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3593/ ........ + Merged revisions 416806 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, + include/asterisk/netsock2.h, include/asterisk/acl.h, + main/netsock2.c: pjsip cli: Change Identify to show CIDR notation + instead of netmasks. * Added ast_sockaddr_cidr_bits() to count + the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which + uses ast_sockaddr_cidr_bits() for the netmask instead of + ast_sockaddr_stringify_addr. * Changed + res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() + instead of ast_ha_join() for the CLI output. This is a CLI change + only. AMI was not affected. Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3652/ ........ Merged + revisions 416737 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-19 19:40 +0000 [r416736] Kinsey Moore <kmoore@digium.com> + + * /, main/bridge.c, res/parking/parking_tests.c, + channels/sip/reqresp_parser.c, main/logger.c, main/test.c: Fix + build warnings with TEST_FRAMEWORK enabled ........ Merged + revisions 416732 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416733 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416734 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-19 16:04 +0000 [r416589-416670] George Joseph <george.joseph@fairview5.com> + + * pbx/pbx_lua.c, /: Remove the problematic and unneeded + AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c + AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be + incorrectly loaded before pbx_config. pbx_config was therefore + blowing away contexts that were created by pbx_lua. With + AST_MODFLAG_DEFAULT the load order is now correct and contexs are + being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed + anyway since no other modules needed its global symbols that + early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by: + Dennis Guse Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3629/ ........ Merged + revisions 416668 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416669 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/extensions.lua.sample, /: Update extensions.lua.sample + with naming conflict guidance. The sample extensions.lua was + causing pbx_lua to fail to load when parsing 'app.goto("default", + "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This + patch adds guidance to extensions.lua.sample and changed + 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", + 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by: + gtjoseph Review: https://reviewboard.asterisk.org/r/3627/ + ........ Merged revisions 416581 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-18 04:22 +0000 [r416561] Matthew Jordan <mjordan@digium.com> + + * /, main/stasis_channels.c: stasis_channels: Update the stasis + cache if manager variables are needed In r416211, the publishing + of variable changes was modified such that a cached channel + snapshot was used if manager variables were not requested with + each AMI event. This was done to reduce the amount of channel + snapshots created. However, an assumption was made that + generating a channel snapshot and publishing the snapshot to the + channel topic was sufficient to ensure that the cache would be + updated; this is not the case. The channel snapshot type must be + used to force a snapshot update. This patch updates the + publication of channel variables such that the cache is updated + prior to publication of the channel variable message if manager + variables are in use. This ensures that all AMI events receive + the variable update when they are supposed to. Note that this + issue was caught by the Asterisk Test Suite (go go testing) + ........ Merged revisions 416557 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-17 18:45 +0000 [r416444-416503] Mark Michelson <mmichelson@digium.com> + + * /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to + set inheritable channel variables. ........ Merged revisions + 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 416501 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416502 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pidf_body_generator.c, /, + res/res_pjsip_xpidf_body_generator.c: Fix string growth algorithm + for XML presence bodies. pjpidf_print() does not return < 0 if + there is not enough room for the document to be printed. Rather, + it returns 39, the length of the XML prolog. The algorithm also + had a bug in that it would return if it attempted to grow the + string larger. ........ Merged revisions 416442 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-17 16:33 +0000 [r416443] Kinsey Moore <kmoore@digium.com> + + * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated + start calls Currently, music on hold will stop and then start + again from the beginning if ast_moh_start() is called multiple + times. This can happen if a call is put on hold repeatedly (the + channel receives multiple HOLD control frames) and can be + triggered from ARI by starting MoH on a channel multiple times. + This is fairly jarring/annoying to users. This change prevents + MoH from being restarted if the requested music class is the same + as the one currently playing. This includes an extra check to + prevent the errors previously experienced in the testsuite and + has 100+ test runs behind it. Review: + https://reviewboard.asterisk.org/r/3615/ ........ Merged + revisions 416439 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416440 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416441 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-16 18:27 +0000 [r416416] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/sig_ss7.h, configure, channels/chan_dahdi.h, + configure.ac, UPGRADE.txt, configs/ss7.timers.sample (added), + CHANGES, channels/sig_ss7.c: chan_dahdi: Adds support for major + update to libss7. * SS7 support now requires libss7 v2.0 or + later. The new libss7 is not backwards compatible. * Added SS7 + support for connected line and redirecting. * Most SS7 CLI + commands are reworked as well as new SS7 commands added. See + online CLI help. * Added several SS7 config option parameters + described in chan_dahdi.conf.sample. * ISUP timer support + reworked and now requires explicit configuration. See + ss7.timers.sample. Special thanks to Kaloyan Kovachev for his + support and persistence in getting the original patch by adomjan + updated and ready for release. SS7-27 #close Reported by: adomjan + +2014-06-16 16:22 +0000 [r416394] Kevin Harwell <kharwell@digium.com> + + * include/asterisk/http_websocket.h, tests/test_websocket_client.c, + res/res_http_websocket.c: res_http_websocket: read/write string + fixup There was a problem when reading a string from the + websocket. It assumed the received data had a null terminator and + tried to write the data to an ast_str. This of course could/would + read past the end of the given buffer while writing the data to + the internal buffer of ast_str. Modified the the code to + correctly place a null terminator on the result string. + +2014-06-16 09:04 +0000 [r416339] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * cel/cel_sqlite3_custom.c, main/db.c, res/res_config_sqlite3.c, + cdr/cdr_sqlite3_custom.c, /: We have faced situation when using + CDR and CEL by sqlite3 modules. With system having high load + (~100 concurrent calls created by sipp) we found many cdr and cel + records missed. There is special finction in sqlite3, that make + able to fix this situation - sqlite3_wait_timeout, that also can + replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this + function can be used for aastdb and res_config_sqlite3 to avoid + missed writes to sqlite db. #ASTERISK-23766 #close Reported by: + Igor Goncharovsky Review: + https://reviewboard.asterisk.org/r/3559/ ........ Merged + revisions 416336 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416337 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416338 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-16 02:40 +0000 [r416267-416319] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: channels/chan_sip: Forbid remote bridging + if T.38 is negotiated When a framehook is removed - such as the + fax gateway framehook - the bridge framework will re-evaluate the + bridge mixing technologies to see if it can improve the bridging. + When this occurs, get_rtp_info will be called to determine if + local or remote bridging can be used. Using remote bridging will + cause a fax to fail, as direct media negotiation will cause some + small number of packets to not arrive at the remote endpoint. + This patch forces local native bridging if T.38 negotiation is in + progress or has been established. ........ Merged revisions + 416318 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/channel_internal_api.c: channel_internal_api: Publish a + snapshot change when linkedids change Snapshots are now not + published *quite* as much as they used to. One instance where + they are not published any longer is during bridge enter and exit + - the state of the channel doesn't change, the bridge does. + However, channels are changed when a linkedid is propagated; + previously, the channel's state would be updated and published + during the bridge enter event. Now this must be explicitly done. + ........ Merged revisions 416300 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_stasis_endpoints.c: test_stasis_endpoints: Remove + expected channel snapshot We no longer publish a channel snapshot + when it is associated with an endpoint; after all, the channel + itself hasn't changed - the endpoint state has changed. This + updates the channel_messages unit test accordingly. ........ + Merged revisions 416298 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This + patch reverts r416150. When the comparison between mohclass->name + and state->class->name is made, you are not guaranteed that (a) + state->class is non-NULL or that state or state->class are in a + safe state. Crashes caught by the bridges/transfer_capabilities + test. ........ Merged revisions 416251 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416252 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416255 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-14 19:26 +0000 [r416237] Corey Farrell <git@cfware.com> + + * res/res_manager_devicestate.c, res/res_manager_presencestate.c: + res_manager_devicestate and res_manager_presencestate missing + support level Add MODULEINFO comment block to define support + level core for these new modules. Review: + https://reviewboard.asterisk.org/r/3620/ + +2014-06-13 18:24 +0000 [r416216] Matthew Jordan <mjordan@digium.com> + + * res/res_agi.c, res/res_pjsip/pjsip_configuration.c, + main/stasis_channels.c, res/ari/resource_channels.c, + main/bridge_channel.c, main/pbx.c, main/stasis_cache.c, /, + apps/app_meetme.c, main/pickup.c, main/channel_internal_api.c, + include/asterisk/channel.h, main/core_local.c, main/aoc.c, + main/endpoints.c, main/cel.c, apps/app_queue.c, + main/stasis_bridges.c, apps/app_agent_pool.c, main/cli.c, + main/channel.c, main/dial.c, main/manager.c, + include/asterisk/stasis_channels.h: stasis: Reduce creation of + channel snapshots to improve performance During some performance + testing of Asterisk with AGI, ARI, and lots of Local channels, we + noticed that there's quite a hit in performance during channel + creation and releasing to the dialplan (ARI continue). After + investigating the performance spike that occurs during channel + creation, we discovered that we create a lot of channel snapshots + that are technically unnecessary. This includes creating + snapshots during: * AGI execution * Returning objects for ARI + commands * During some Local channel operations * During some + dialling operations * During variable setting * During some + bridging operations And more. This patch does the following: - It + removes a number of fields from channel snapshots. These fields + were rarely used, were expensive to have on the snapshot, and + hurt performance. This included formats, translation paths, Log + Call ID, callgroup, pickup group, and all channel variables. As a + result, AMI Status, "core show channel", "core show channelvar", + and "pjsip show channel" were modified to either hit the live + channel or not show certain pieces of data. While this is + unfortunate, the performance gain from this patch is worth the + loss in behaviour. - It adds a mechanism to publish a cached + snapshot + blob. A large number of publications were changed to + use this, including: - During Dial begin - During Variable + assignment (if no AMI variables are emitted - if AMI variables + are set, we have to make snapshots when a variable is changed) - + During channel pickup - When a channel is put on hold/unhold - + When a DTMF digit is begun/ended - When creating a bridge + snapshot - When an AOC event is raised - During Local channel + optimization/Local bridging - When endpoint snapshots are + generated - All AGI events - All ARI responses that return a + channel - Events in the AgentPool, MeetMe, and some in Queue - + Additionally, some extraneous channel snapshots were being made + that were unnecessary. These were removed. - The result of + ast_hashtab_hash_string is now cached in stasis_cache. This + reduces a large number of calls to ast_hashtab_hash_string, which + reduced the amount of time spent in this function in gprof by + around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged + revisions 416211 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-13 13:11 +0000 [r416149-416153] Kinsey Moore <kmoore@digium.com> + + * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated + start calls Currently, music on hold will stop and then start + again from the beginning if ast_moh_start() is called multiple + times. This can happen if a call is put on hold repeatedly (the + channel receives multiple HOLD control frames) and can be + triggered from ARI by starting MoH on a channel multiple times. + This is fairly jarring/annoying to users. This change prevents + MoH from being restarted if the requested music class is the same + as the one currently playing. Review: + https://reviewboard.asterisk.org/r/3615/ ........ Merged + revisions 416150 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416151 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cel.c, /: CEL: Expose parking retreiver in extra field This + exposes the retreiver of a parked call under the "retreiver" key + of the extra field when this information is available. Review: + https://reviewboard.asterisk.org/r/3608/ ........ Merged + revisions 416148 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-13 05:16 +0000 [r416071] Richard Mudgett <rmudgett@digium.com> + + * main/http.c, include/asterisk/tcptls.h, main/tcptls.c, + main/manager.c, /, channels/chan_sip.c: AST-2014-007: Fix of fix + to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close + Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/3617/ ........ Merged + revisions 416066 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416067 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416070 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 21:27 +0000 [r416024] Rusty Newton <rnewton@digium.com> + + * main/pbx.c: main/pbx - documentation - enhance 'core show hints' + and 'core show hint' help text Adds descriptive help text to + 'core show hints' and 'core show hint'. The text describes the + various columns for the sake of clarity. It takes into account + recent changes to the content displayed by the commands + https://reviewboard.asterisk.org/r/3604/ and + https://reviewboard.asterisk.org/r/3611/. ASTERISK-23764 Review: + https://reviewboard.asterisk.org/r/3610/ + +2014-06-12 20:17 +0000 [r415982] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip_pubsub.c, /: Fix build in devmode for GCC 4.10 + ........ Merged revisions 415980 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 17:00 +0000 [r415907] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/utils.h, main/tcptls.c, main/manager.c, /, + channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c, + include/asterisk/tcptls.h, res/res_http_websocket.c, + configs/http.conf.sample: AST-2014-007: Fix DOS by consuming the + number of allowed HTTP connections. Simply establishing a TCP + connection and never sending anything to the configured HTTP port + in http.conf will tie up a HTTP connection. Since there is a + maximum number of open HTTP sessions allowed at a time you can + block legitimate connections. A similar problem exists if a HTTP + request is started but never finished. * Added http.conf + session_inactivity timer option to close HTTP connections that + aren't doing anything. Defaults to 30000 ms. * Removed the + undocumented manager.conf block-sockets option. It interferes + with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections + now have better authentication timeout protection. Though I + didn't remove the bizzare TLS timeout polling code from chan_sip. + * chan_sip can now handle SSL certificate renegotiations in the + middle of a session. It couldn't do that before because the + socket was non-blocking and the SSL calls were not restarted as + documented by the OpenSSL documentation. * Fixed an off nominal + leak of the ssl struct in handle_tcptls_connection() if the FILE + stream failed to open and the SSL certificate negotiations + failed. The patch creates a custom FILE stream handler to give + the created FILE streams inactivity timeout and timeout after a + specific moment in time capability. This approach eliminates the + need for code using the FILE stream to be redesigned to deal with + the timeouts. This patch indirectly fixes most of ASTERISK-18345 + by fixing the usage of the SSL_read/SSL_write operations. + ASTERISK-23673 #close Reported by: Richard Mudgett ........ + Merged revisions 415841 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415854 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415896 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 15:50 +0000 [r415839] Scott Griepentrog <sgriepentrog@digium.com> + + * /, apps/app_queue.c: app_queue: delayed state can cause early + leavewhenempty ringing In app_queue, device state changes arrive + in event messages and update the queue member status value. That + value is checked in get_member_status() to decide that the caller + should leave when there are no available members. Although event + messages can be delayed by other activity, there is no adverse + affect by lagged status except in one specific case: there is + only one available member, it was just rung, and leavewhenempty + is enabled set for ringing members. This change adds a direct + check of the device state only under this condition where the + caller may be dropped incorrectly, resolving this issue without + affecting performance of app_queue normally. AST-1248 #close + Review: https://reviewboard.asterisk.org/r/3595/ Reported by: + Thomas Arimont ........ Merged revisions 415833 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415835 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415836 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 15:39 +0000 [r415834] Jonathan Rose <jrose@digium.com> + + * apps/app_mixmonitor.c, /, UPGRADE.txt: MixMontior: Add class + authorization requirements to MixMonitor AMI commands MixMonitor + AMI commands StartMixMonitor and StopMixMonitor lacked class + authorization. StopMixMonitor now requires that the manager user + either have the call or system class authorization. + StartMixMonitor is a slightly larger issue since it can execute + shell commands if the right arguments are passed into it, and we + consider this a permission escalation. A security release will be + issued for problem this shortly. ASTERISK-23609 #close Reported + by: Corey Farrell ........ Merged revisions 415825 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415832 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 14:39 +0000 [r415813] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: unauthenticated + remote crash in PJSIP pub/sub framework A remotely exploitable + crash vulnerability exists in the PJSIP channel driver's pub/sub + framework. If an attempt is made to unsubscribe when not + currently subscribed and the endpoint's "sub_min_expiry" is set + to zero, Asterisk tries to create an expiration timer with zero + seconds, which is not allowed, so an assertion raised. The fix + was to reject a subscription that is attempting to unsubscribe + when not being already subscribed. Asterisk now checks for this + situation appropriately and responds with a 400 instead of + crashing. AST-2014-005 ASTERISK-23489 #close ........ Merged + revisions 415812 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 14:15 +0000 [r415795] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip.c, /: Fix potential deadlock situation in + res_pjsip. SIP transaction timeouts are handled in the PJSIP + monitor thread. When this happens on a subscription, and the + subscription is destroyed, the subscription destruction is + dispatched synchronously to the threadpool. The issue is that the + PJSIP dialog is locked by the monitor thread, and then the + dispatched task attempts to lock the dialog. This leads to a + deadlock that causes SIP traffic to no longer be accepted on the + Asterisk server. The fix here is to treat the monitor thread as + if it were a threadpool thread when it attempts to dispatch + synchronous tasks. This way, the dispatched task turns into a + simple function call within the same thread, and the locking + issue is averted. AST-2014-008 ASTERISK-23802 #close ........ + Merged revisions 415794 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 11:34 +0000 [r415767] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, + res/res_pjsip_exten_state.c, include/asterisk/res_pjsip.h, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pubsub.exports.in, /, + contrib/ast-db-manage/config/versions/c6d929b23a8_create_pjsip_subscription_persistence_.py + (added), res/res_pjsip_mwi.c: res_pjsip_pubsub: Persist + subscriptions in sorcery so they are recreated on startup. This + change makes res_pjsip_pubsub persist inbound subscriptions in + sorcery. By default this uses the local astdb but it can also be + configured to store within an outside database. When Asterisk is + started these subscriptions are recreated if they have not + expired. Notifications are sent to the devices which have + subscribed and they are none the wiser that the system has + restarted. Review: https://reviewboard.asterisk.org/r/3598/ + ........ Merged revisions 415766 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 07:52 +0000 [r415749] Walter Doekes <walter+asterisk@wjd.nu> + + * UPGRADE.txt, contrib/scripts/safe_asterisk, Makefile, /: + safe_asterisk: Overwrite old safe_asterisk on make install. From + now on, make install will overwrite safe_asterisk with the latest + version. You need to move any local modifications to files inside + /etc/asterisk/startup.d, if you have any. See also commits + r394939 and r397938. ASTERISK-21965 #close Patches: + safe_asterisk.patch uploaded by jkister (License 6232, modified + by me) ........ Merged revisions 415748 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-11 23:01 +0000 [r415730] Richard Mudgett <rmudgett@digium.com> + + * main/format.c, /: format.c: Fix misuse of hash container + function. The supplied hash function to a container must be + idempotent given the object's key value to figure out which + container bucket the object belongs in. Returning a random number + or the current container count is not idempotent. The "computed + hash" value doesn't help find the object later in those cases. * + Fixed the format_list container to actually be a list since that + is how the container is used. Conceptually, if more than 283 + formats were added to the format_list then odd things may have + happened before the fix. ........ Merged revisions 415728 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415729 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-11 20:22 +0000 [r415698-415715] Scott Griepentrog <sgriepentrog@digium.com> + + * main/pbx.c: CLI: correct presence information on core show hints + Adds presence to core show hint and changes presence string + conversion to use the correct function. ASTERISK-23858 #close + Review: https://reviewboard.asterisk.org/r/3611/ + + * main/pbx.c: CLI: add presence information to core show hints Adds + presence state value to output of core show hints. Also reformats + the output slightly so it doesn't use as much space as it would + otherwise. Was: 1000@demo : SIP/1000 State:Unavailable Watchers 0 + Now: 1000@demo : SIP/1000 State:Unavailable Presence:Idle + Watchers 0 AFS-53 #close Review: + https://reviewboard.asterisk.org/r/3604/ + +2014-06-10 18:32 +0000 [r415679] Kinsey Moore <kmoore@digium.com> + + * main/channel.c, /: Fix build in dev mode due to signed/unsigned + mismatch ........ Merged revisions 415678 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-10 16:06 +0000 [r415659] Jonathan Rose <jrose@digium.com> + + * main/message.c, /, res/res_pjsip_notify.c: PJSIP: PJSIPNotify - + Strip content-length headers and add documentation Documentation + for how to add custom headers/content to notifies created with + the PJSIPNotify manager action was a little sparse and it also + wasn't vetting application of Content-length headers like its + chan_sip equivalent was (so two Content-length headers could be + applied... and PJSIP determines the content length anyway, so it + just opens people up for error). This patch also flips the + variable order so that the variables are interpreted in the same + order as they are put in the AMI action. Review: + https://reviewboard.asterisk.org/r/3587/ ........ Merged + revisions 415658 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-10 09:28 +0000 [r415630] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c, /: chan_ooh323: fix loading module failure + if there no accessible h323_log or ooh323 config file change + return 1 to return AST_MODULE_LOAD_FAILURE on module load routine + few cosmetic changes ASTERISK-23814 #close (closes issue + ASTERISK-23814) Reported by: Igor Goncharovsky Patches: + ASTERISK-23814-ast11.patch ........ Merged revisions 415599 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415602 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-09 20:21 +0000 [r415580] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_header_funcs.c, /: chan_pjsip: Fix bug where custom + SIP headers could be duplicated on outgoing INVITEs. When using + PJSIP_HEADER() to add custom headers to outgoing INVITE requests, + certain situations could result in the headers being duplicated. + For instance, if the request were retransmitted, or if the INVITE + were re-sent with authentication credentials, the custom headers + would be re-added to the request. The fix here is to, after + adding the custom headers to the outbound INVITE, remove the + datastore that holds the custom headers to add. This way, there + is no risk in accidentally adding them if the session supplement + is called into a second or third time. ........ Merged revisions + 415579 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-09 12:12 +0000 [r415524] Walter Doekes <walter+asterisk@wjd.nu> + + * /, UPGRADE.txt, contrib/scripts/safe_asterisk: safe_asterisk: + Cleanup additions to r415132. * Replaced a stray echo that + should've been a message call in safe_asterisk. This replaces a + conditional log message by a slightly different message. Please + update your log parsing scripts. * Made the $NOTIFY mail Subject + more verbose by adding the machine name and exitstatus. (Note + that a 'make install' still won't overwrite your old + safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492 + #close ........ Merged revisions 415521 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415522 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415523 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-09 03:50 +0000 [r415466] Corey Farrell <git@cfware.com> + + * /, main/autoservice.c: autoservice: stop thread on graceful + shutdown This change adds thread shutdown to autoservice for + graceful shutdowns only. ast_register_cleanup is backported to + 1.8 to allow this. The logger callid is also released on shutdown + in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3594/ ........ Merged + revisions 415463 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415464 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-08 18:12 +0000 [r415444] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + main/bridge_channel.c, main/channel.c, main/pbx.c, /, + main/framehook.c, main/bridge_after.c: bridges/bridge_native_rtp: + Reconfigure bridge on removal of framehook This patch is a re-do + of r414122. When r414122 was merged, a major problem with it was + uncovered. UNBRIDGE soft hangup flags have a catastrophic effect + on the pbx core if they leak out from the bridge layer: the + channel gets hung up. With the number of threads involved in a + blind transfer, and with the initial patch, it was likely that + this would occur. This caused a large number of test failures + This patch is nearly identical with the one proposed in r414122, + save for the following changes: - We explicitly clear the + UNBRIDGE flag when setting an after goto on a channel in a bridge + - Defensively, if we encounter an UNBRIDGE flag in the pbx core, + we handle it https://reviewboard.asterisk.org/r/3585/ ........ + Merged revisions 415443 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-07 00:42 +0000 [r415428] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/bridge.h, /: bridge.h: Remove redundant struct + ast_bridge_channel forward declaration. ........ Merged revisions + 415427 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 21:44 +0000 [r415411] Jonathan Rose <jrose@digium.com> + + * include/asterisk/manager.h, main/config.c, main/manager.c, /, + channels/chan_sip.c, include/asterisk/config.h: chan_sip: Fix + order of variables specified in SIPNotify action Prior to this + patch, sequential variables would be ordered in reverse from the + order specified in the manager action. Review: + https://reviewboard.asterisk.org/r/3588/ ........ Merged + revisions 415359 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415390 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415410 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 20:45 +0000 [r415358] Kevin Harwell <kharwell@digium.com> + + * main/uri.c, tests/test_websocket_client.c: core uri: Custom uri + parsing error when no query parameters If using the custom URI + parsing code (not external uriparser lib) and there was no query + parameters the resulting pointer would be NULL and then an + attempt was made to subtract from it. The pointer is now set to a + valid value if there is no query parameter(s). Also, in the + 'ast_uri_make_host_with_port' function when setting the + terminator on the resulting string it was writing it one past the + end of allocated memory. It now writes the string terminator + appropriately. + +2014-06-06 19:13 +0000 [r415343] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip_sdp_rtp.c: PJSIP: Remove premature write of raw + formats Currently, there are situations that can occur when using + chan_pjsip and certain dialplan applications (notably ChanSpy()) + that can cause the channel to get no audio with scrolling + warnings about format mismatches. This is caused by a failure to + update translation paths on a mid-call native format update since + the raw formats have already been updated by res_pjsip_sdp_rtp.c + in set_caps(). Removing the premature raw format updates allows + the translation paths to be setup correctly and the raw read and + write formats with them. AFS-63 #close ........ Merged revisions + 415342 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 14:12 +0000 [r415319] George Joseph <george.joseph@fairview5.com> + + * tests/test_astobj2.c, main/astobj2_private.h (added), + main/astobj2.c, main/astobj2_container_private.h (added), + main/astobj2_container.c (added), main/astobj2_hash.c (added), + main/astobj2_rbtree.c (added), /, include/asterisk/astobj2.h: + Split astobj2.c into more maintainable components. Split + astobj2.c into the following files to improve maintainability. + astobj2.c - object primitives, object primitive misc and + initialization code. astobj2_private.h - internal object + declarations needed by the containers. astobj2_container.c - + generic conainer and container misc code. + astobj2_container_hash.c - hash container specific code. + astobj2_container_rbtree.c - rbtree container specific code. + astobj2_container_private.h - generic container definitions and + rtti prototypes. https://reviewboard.asterisk.org/r/3576/ + ........ Merged revisions 415317 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 12:49 +0000 [r415302] Rusty Newton <rnewton@digium.com> + + * /, configs/cli_aliases.conf.sample: configs/cli_aliases.conf: Two + new aliases, plus enhancements for context names. Changed naming + of included alias templates to avoid confusion between version + names. For example, asterisk12 was for asterisk 1.2, so I changed + it to asterisk_1dot2, so that later we can use asterisk_12 for + Asterisk 12. Added alias for "features reload" to the template + for Asterisk 11 style syntax template, as features reload was + removed in 12, but you can still do "module reload features" + Added alias for "pjsip reload" to the friendly template. It is + shorter than "module reload res_pjsip.so" and if some are like + me; I constantly forget that reloading chan_pjsip doesn't parse + config. Remembering "pjsip reload" is just easier. ASTERISK-23654 + #close ASTERISK-23654 #comment Fixed by adding two new aliases + and enhancements for context names. Review: + https://reviewboard.asterisk.org/r/3572/ ........ Merged + revisions 415301 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-05 19:04 +0000 [r415231-415288] Richard Mudgett <rmudgett@digium.com> + + * main/config.c: config: Fix indentation and missing curlies in + config_text_file_load(). + + * main/config.c, /: config: Fix config files not reloading when + only an included file changes. The twisted logic determining if a + config file should be reloaded was mostly broken and disabled. + The incorrect test that ASTERISK-23383 fixed actually reenabled + the broken logic. The incorrect test was causing the timestamp to + always be cleared which caused config files with includes to + always be reloaded. * Made wildcard includes always cause a + reload. Determining if a file was deleted cannot be determined + without restructuring the cache to determine if any files are + missing from the last files actually loaded. Also without + refactoring config_text_file_load(), the glob loop couldn't check + more than one file for changes anyway. * Made remove the cache + entry if the file no longer exists when trying to get its + timestamp or it is no longer a regular file. This fixes the + corner case where the file was loaded, then deleted, then the + config reloaded, then the file restored with the same timestamp, + and then the config reloaded again. * Made remove the cache entry + include list when actually loading the file. This gets rid of any + stale includes the file had from the last time the file was + loaded. ASTERISK-23683 #close Reported by: tootai Review: + https://reviewboard.asterisk.org/r/3575/ ........ Merged + revisions 415225 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415229 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415230 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-05 17:22 +0000 [r415223] Kevin Harwell <kharwell@digium.com> + + * tests/test_uri.c (added), include/asterisk/http_websocket.h, + main/http.c, main/uri.c (added), tests/test_websocket_client.c + (added), res/res_http_websocket.c, include/asterisk/http.h, + include/asterisk/uri.h (added), + res/res_http_websocket.exports.in: res_http_websocket: Create a + websocket client Added a websocket server client in Asterisk. + Asterisk has a websocket server, but not a client. The ability to + have Asterisk be able to connect to a websocket server can + potentially be useful for future work (for instance this could + allow ARI to connect back to some external system, although more + work would be needed in order to incorporate that). Also a couple + of things to note - proxy connection support has not been + implemented and there is limited http response code handling + (basically, it is connect or not). Also added an initial new URI + handling mechanism to core. Internet type URI's are parsed into a + data structure that contains pointers to the various parts of the + URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell + Review: https://reviewboard.asterisk.org/r/3541/ + +2014-06-05 14:49 +0000 [r415208] Matthew Jordan <mjordan@digium.com> + + * /, apps/app_confbridge.c: app_confbridge: Allow muting of users + waiting to enter a ConfBridge Prior to this patch, users waiting + to enter a ConfBridge were not considered when muted via the CLI + or via AMI. Instead, a confusing message would be emitted stating + that the channel did not exist. This patch allows a user to be + muted when waiting to enter a ConfBridge conference. This is + equivalent to start when muted, only toggled via the CLI or AMI. + Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824 + #close patches: rb3582.patch uploaded by tm1000 (License 6524) + ........ Merged revisions 415206 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415207 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-05 11:59 +0000 [r415192] Kinsey Moore <kmoore@digium.com> + + * /, channels/chan_pjsip.c: PJSIP: Send initial connected line + information This makes chan_pjsip send connected line information + when it is called so that connected line information is available + on the connected channel. (closes issue DPMA-442) Reported by: + John Bigelow Review: https://reviewboard.asterisk.org/r/3584/ + ........ Merged revisions 415191 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-04 20:16 +0000 [r415173] Walter Doekes <walter+asterisk@wjd.nu> + + * /, contrib/scripts/safe_asterisk: safe_asterisk: Cleanup and + debian compatibility. Cleans up the safe_asterisk script and adds + the ASTSAFE_FOREGROUND option that allows the debian asterisk + init script to capture the right pid. * Drop the vim #modeline + which wasn't used. Use test consistently without the odd + configure xno syntax. Double quote all paths. General cleanup. * + Don't output message()s to the console but only to TTY if set. * + Allow TTY to be "no" as well as empty (debian compatibility with + debian/patches/safe_asterisk-config). * Add option to export + ASTSAFE_FOREGROUND=1 from the init script that calls this to + disable backgrounding. Debian uses a similar method in + debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review: + https://reviewboard.asterisk.org/r/3574/ ........ Merged + revisions 415132 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415171 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415172 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-04 14:13 +0000 [r415116-415118] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_pjsip.c: chan_pjsip: Add debug in RTP Engine + glue callback This patch adds some debug statements that aid with + determining why a direct media request may or may not be + initiated. ........ Merged revisions 415117 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: res_pjsip_session: Add debug + statement for session refreshes This small patch adds a debug + level 3 statement indicating how a session refresh is being sent + - either as a re-INVITE or as an UPDATE - and where the session + refresh is going. ........ Merged revisions 415115 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-04 07:27 +0000 [r415080] Corey Farrell <git@cfware.com> + + * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c: + app_confbridge: Correct verification of conference name length + Conference names were not checked for maximum length, allowing + unexpected behaviour. This change adds checking to ensure the + maximum length is not exceeded. The maximum length is also + changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close + Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: + confbridge-enforce_max-1.8.patch uploaded by coreyfarrell + (license 5909) confbridge-enforce_max-11up.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 415060 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415066 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415078 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-03 07:36 +0000 [r415000] Walter Doekes <walter+asterisk@wjd.nu> + + * /, funcs/func_odbc.c: func_odbc: Fix fixed size buffers fix + (r414968). The change that removed the fixed size buffers in + odbc-related code -- removing arbitrary column width limits -- + was incomplete. This change adds: no segfault on writesql without + insertsql and return value checks after strdup. While I was in + the vicinity I cleaned up the linefeeds in the odbc function + descriptions, moved some code for clarity, removed some blobs and + noted (but didn't fix) that the 'odbc write ... exec' CLI command + doesn't behave as the dialplan equivalent when insertsql= is + used. ASTERISK-23582 #close Review: + https://reviewboard.asterisk.org/r/3579/ ........ Merged + revisions 414997 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414998 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-01 15:32 +0000 [r414976] Joshua Colp <jcolp@digium.com> + + * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Take the + bridge type choice of both channels into account. The + bridge_native_rtp module currently uses the bridge result of the + first channel that joins a bridge as the ultimate result. This + means that if the first channel has direct media enabled but the + second does not a direct media bridge will still occur. This + change makes it so that both sides are taken into account. If + either side forbids the bridge or responds with a local bridge + result then either a generic or local bridge occurs. + ASTERISK-23541 #close Reported by: Justin E Review: + https://reviewboard.asterisk.org/r/3577/ ........ Merged + revisions 414975 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-30 14:53 +0000 [r414949] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip_refer.c, /: PJSIP: Prevent crash on blind transfer + Blind transfers don't go too well with NULL channels which can + occur if the channel has already been transferred away. (closes + issue ASTERISK-23718) Reported by: Jonathan Rose ........ Merged + revisions 414948 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-30 12:42 +0000 [r414883-414935] Matthew Jordan <mjordan@digium.com> + + * main/audiohook.c, CHANGES, res/ari/ari_model_validators.c, + res/ari/ari_model_validators.h, funcs/func_talkdetect.c (added), + include/asterisk/stasis_channels.h, + rest-api/api-docs/events.json, /, main/stasis_channels.c: + TALK_DETECT: A channel function that raises events when talking + is detected This patch adds a new channel function TALK_DETECT + that, when set on a channel, causes events indicating the + start/stop of talking on a channel to be emitted to both AMI and + ARI clients. The function allows setting both the silence + threshold (the length of silence after which we decide no one is + talking) as well as the talking threshold (the amount of energy + that counts as talking). Parameters can be updated on a channel + after talk detection has been enabled, and talk detection can be + removed at any time. The events raised by the function use a + nomenclature similar to existing AMI/ARI events. For AMI: + ChannelTalkingStart/ChannelTalkingStop For ARI: + ChannelTalkingStarted/ChannelTalkingFinished Review: + https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close + Reported by: Matt Jordan ........ Merged revisions 414934 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, /: main/config.c: AMI action UpdateConfig EmptyCat + clears all categories When invoking UpdateConfig AMI action with + Action set to EmptyCat, Asterisk will make all categories empty + in the config but the one requested with a Cat variable. This is + due to a bug in ast_category_empty (main/config.c) that makes an + incorrect comparison for a category name. This patch corrects the + comparison such that only the requested category is cleared. + Review: https://reviewboard.asterisk.org/r/3573/ #ASTERISK-23803 + #close Reported by: zvision patches: manager.c.diff uploaded by + zvision (License 5755) ........ Merged revisions 414880 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414881 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-29 18:51 +0000 [r414861] Kinsey Moore <kmoore@digium.com> + + * main/pbx.c, /: PBX: Prevent incorrect hint parsing Dynamic and + pattern matching hints should not be checked for their last known + state until they are instantiated by subscribers. (closes issue + AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted + by Matt Jordan (license 6283) ........ Merged revisions 414813 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 414859 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414860 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 22:54 +0000 [r414798] Matthew Jordan <mjordan@digium.com> + + * main/loader.c, include/asterisk/logger.h, res/res_config_curl.c, + cel/cel_odbc.c, res/res_config_odbc.c, + bridges/bridge_builtin_features.c, main/optional_api.c, + main/logger.c, main/config_options.c, cdr/cdr_odbc.c, + apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c, + main/xmldoc.c, apps/app_voicemail.c, cel/cel_pgsql.c, + channels/chan_unistim.c, res/res_config_pgsql.c, main/pbx.c, + cdr/cdr_sqlite3_custom.c, res/res_fax.c, main/bridge.c, + apps/app_waitforsilence.c, cdr/cdr_adaptive_odbc.c, + res/parking/parking_applications.c, cdr/cdr_pgsql.c, + res/res_jabber.c: Logger/CLI/etc.: Fix some aesthetic issues; + reduce chatty verbose messages This patch addresses some + aesthetic issues in Asterisk. These are all just minor tweaks to + improve the look of the CLI when used in a variety of settings. + Specifically: * A number of chatty verbose messages were removed + or demoted to DEBUG messages. Verbose messages with a verbosity + level of 5 or higher were - if kept as verbose messages - demoted + to level 4. Several messages that were emitted at verbose level 3 + were demoted to 4, as announcement of dialplan applications being + executed occur at level 3 (and so the effects of those + applications should generally be less). * Some verbose messages + that only appear when their respective 'debug' options are + enabled were bumped up to always be displayed. * + Prefix/timestamping of verbose messages were moved to the + verboser handlers. This was done to prevent duplication of + prefixes when the timestamp option (-T) is used with the CLI. * + Verbose magic is removed from messages before being emitted to + non-verboser handlers. This prevents the magic in multi-line + verbose messages (such as SIP debug traces or the output of + DumpChan) from being written to files. * _Slightly_ better + support for the "light background" option (-W) was added. This + includes using ast_term_quit in the output of XML documentation + help, as well as changing the "Asterisk Ready" prompt to bright + green on the default background (which stands a better chance of + being displayed properly than bright white). Review: + https://reviewboard.asterisk.org/r/3547/ + +2014-05-28 20:53 +0000 [r414781] Rusty Newton <rnewton@digium.com> + + * /, configs/pjsip.conf.sample: pjsip.conf: privkey_file should be + priv_key_file, mediaencryption=yes should be mediaencryption=sdes + privkey_file was missed in the snake case update. An example + included an invalid value for the mediaencryption option. + ........ Merged revisions 414780 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 17:46 +0000 [r414764-414766] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json: AMI/ARI: Update version + numbers Update the semantic versioning of ARI to 1.3.0 and AMI to + 2.3.0 to account for backwards compatible changes going from + 12.2.0 to 12.3.0. ........ Merged revisions 414765 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/ast-db-manage/cdr/env.py, /: ast-db-manage/cdr/env.py: + Don't fail if a config file can't be loaded When generating SQL + files via the repotools alembic_creator.py script, a + configuration object is used programatically with SQLAlechemy, as + opposed to a configuration file. This patch ignores failures to + interpret a config file, as ... there isn't one in this case. + ........ Merged revisions 414763 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 16:56 +0000 [r414748-414750] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h, /, + res/res_pjsip_t38.c: res_pjsip_session: Fix leaked video RTP + ports. Simply enabling PJSIP to negotiage a video codec (e.g., + h264) would leak video RTP ports if the codec were not negotiated + by an incoming call. * Made add_sdp_streams() associate the + handler with the media stream if the handler handled the media + stream. Otherwise, when the ast_sip_session_media object was + destroyed it didn't know how to clean up the RTP resources. * + Fixed sdp_requires_deferral() associating the handler with the + media stream when deciding if the SDP processing needs to be + deferred for T.38. Like the leaked video RTP ports, the T.38 + handler needs to clean up allocated resources from deciding if + SDP processing needs to be deffered. * Cleaned up some dead code + in handle_incoming_sdp() and sdp_requires_deferral(). + ASTERISK-23721 #close Reported by: cervajs Review: + https://reviewboard.asterisk.org/r/3571/ ........ Merged + revisions 414749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, CHANGES, apps/app_agent_pool.c: app_agent_pool: Return to + dialplan if the agent fails to ack the call. Improvements to the + agent pool functionality. * AgentRequest no longer hangs up the + caller if the agent fails to connect with the caller. It now + continues in the dialplan. * AgentRequest returns AGENT_STATUS + set to NOT_CONNECTED if the agent failed to connect with the + call. Most likely because the agent did not acknowledge the call + in time or got disconnected. * The agent alerting play file + configured by the agent.conf custom_beep option can now be + disabled by setting the option to an empty string. The agent is + effectively alerted to a call presence when MOH stops. * Fixed + bridge reference leak when the agent connects with a caller. + ASTERISK-23499 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3551/ ........ Merged + revisions 414747 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 11:37 +0000 [r414696] Joshua Colp <jcolp@digium.com> + + * res/res_config_odbc.c, /, funcs/func_odbc.c: res_config_odbc: Use + dynamically sized buffers to store row data so values do not get + truncated. ASTERISK-23582 #close ASTERISk-23582 #comment Reported + by: Walter Doekes Review: + https://reviewboard.asterisk.org/r/3557/ ........ Merged + revisions 414693 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414694 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414695 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 09:43 +0000 [r414567-414679] Walter Doekes <walter+asterisk@wjd.nu> + + * /, channels/chan_unistim.c: chan_unistim: Unlock mutex in rare + OOM condition. #ASTERISK-23792 #close Reported by: Peter Whisker + Review: https://reviewboard.asterisk.org/r/3567/ ........ Merged + revisions 414677 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414678 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Start session timer at 200, not + at INVITE. Asterisk started counting the session timer at INVITE + while the other end correctly started at 200. This meant that for + short session-expiries (90 seconds) combined with long ringing + times (e.g. 30 seconds), asterisk would wrongly assume that the + timer was hit before the other end thought it was time to send a + session refresh. This resulted in prematurely ended calls. This + changes the session timer to start counting first at 200 like RFC + says it should. (Also removed a few excess NULL checks that would + never hit, because if they did, asterisk would have crashed + already.) ASTERISK-22551 #close Reported by: i2045 Review: + https://reviewboard.asterisk.org/r/3562/ ........ Merged + revisions 414620 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414628 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414636 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_config_odbc.c, /: res_config_odbc: Fix old and new + ast_string_field memory leaks. The ODBC realtime driver uses ^NN + parameter encoding to cope with the special meaning of the + semi-colon. A semi-colon in a field is interpreted as if the key + was supplied twice, something which isn't otherwise possible with + fixed database columns. E.g. allow=alaw;ulaw is parsed as + allow=alaw and allow=ulaw. A literal semi-colon is rewritten to + ^3B when stored in the database. The module uses a stringfield to + efficiently store the encoded parameters. However, this + stringfield wasn't always freed in some off-nominal cases. Commit + r413241 fixed initialization so the encoding for INSERT and + DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked + apparently.) But that commit forgot the frees. This change cleans + that up. Review: https://reviewboard.asterisk.org/r/3555/ + ........ Merged revisions 414564 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414565 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414566 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-25 02:37 +0000 [r414543] Matthew Jordan <mjordan@digium.com> + + * /, main/core_unreal.c: core_unreal: Prevent double free of + core_unreal pvt When a channel is destroyed (such as via + ast_channel_release in off nominal paths in core_unreal), it will + attempt to free (via ast_free) the channel tech pvt. This is + problematic for a few reasons: 1. The channel tech pvt is an ao2 + object in core_unreal. Free'ing the pvt directly is no good. 2. + The channel tech pvt's reference count is dropped just prior to + calling ast_channel_release, resulting in the pvt's destruction. + Hence, the channel destructor is free'ing an invalid pointer. + This patch keeps the dropping of the reference count, but sets + the pvt to NULL on the channel prior to releasing it. This models + what would occur if the channel was hung up directly. ........ + Merged revisions 414542 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-23 17:36 +0000 [r414529] Matthew Jordan <mjordan@digium.com> + + * tests/test_cel.c, /: test_cel: Fix unit tests broken due to event + def changes from res_corosync This patch instructs test_cel to + skip any IE types it doesn't care about. The addition of the raw + and bitfield types caused the tests to fail. ........ Merged + revisions 414528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-23 14:36 +0000 [r414475] Kinsey Moore <kmoore@digium.com> + + * main/event.c, /: Fix signed/unsigned build warnings ........ + Merged revisions 414474 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 16:19 +0000 [r414417] Richard Mudgett <rmudgett@digium.com> + + * /, apps/app_meetme.c: app_meetme: Don't interrupt MOH for + waitmarked users. Occasionally, when the last marked user leaves + the conference, waitmarked users don't get MOH if MOH is supposed + to be played while a waitmarked user is waiting for another + marked user. * Made not interrupt MOH when the user is a + waitmarked user. The waitmarked user doesn't need to hear any + leave announcements from the conference as the user would have + already heard different leave announcements if they were enabled. + Apparently DAHDI occasionally sends unending non-silent streams + to these users or a normal user still in the conference has + continuous high background noise. These non-silent streams cause + MOH to be suspended while the never ending "announcement" is + played. Issue caused by ASTERISK-13680. AST-1349 #close Reported + by: Tyler Stewart Review: + https://reviewboard.asterisk.org/r/3543/ ........ Merged + revisions 414401 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414402 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414404 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 16:09 +0000 [r414406] Scott Griepentrog <sgriepentrog@digium.com> + + * rest-api/api-docs/events.json, /, res/stasis/app.c, + res/ari/resource_events.c, include/asterisk/stasis_app.h, + include/asterisk/stasis.h, apps/app_userevent.c, + res/ari/resource_events.h, res/ari/ari_model_validators.c, + CHANGES, main/stasis.c, res/ari/ari_model_validators.h, + include/asterisk/stasis_channels.h, res/res_ari_events.c, + main/stasis_channels.c, res/res_stasis.c, + main/manager_channels.c, main/stasis_endpoints.c: ARI: Add + ability to raise arbitrary User Events User events can now be + generated from ARI. Events can be signalled with arbitrary json + variables, and include one or more of channel, bridge, or + endpoint snapshots. An application must be specified which will + receive the event message (other applications can subscribe to + it). The message will also be delivered via AMI provided a + channel is attached. Dialplan generated user event messages are + still transmitted via the channel, and will only be received by a + stasis application they are attached to or if the channel is + subscribed to. This change also introduces the multi object blob + mechanism used to send multiple snapshot types in a single + message. The dialplan app UserEvent was also changed to use multi + object blob, and a new stasis message type created to handle + them. ASTERISK-22697 #close Review: + https://reviewboard.asterisk.org/r/3494/ ........ Merged + revisions 414405 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 15:52 +0000 [r414403] Jonathan Rose <jrose@digium.com> + + * include/asterisk/bridge.h, res/parking/parking_bridge_features.c, + channels/chan_mgcp.c, res/res_pjsip_refer.c, + channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, main/parking.c, main/bridge.c, + main/bridge_basic.c, res/parking/parking_applications.c, + include/asterisk/parking.h: res_pjsip_refer: Fix bugs involving + Parking/PJSIP/transfers PJSIP would never send the final 200 + Notify for a blind transfer when transferring to parking. This + patch fixes that. In addition, it fixes a reference leak when + performing blind transfers to non-bridging extensions. Review: + https://reviewboard.asterisk.org/r/3485/ ........ Merged + revisions 414400 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 14:02 +0000 [r414331-414348] Matthew Jordan <mjordan@digium.com> + + * /, UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag ........ + Merged revisions 414345 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414346 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414347 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_corosync.c, include/asterisk/stasis.h, main/app.c, + main/devicestate.c, main/event.c, main/stasis.c, + include/asterisk/devicestate.h, include/asterisk/event.h, + main/stasis_message.c, /, include/asterisk/event_defs.h: + res_corosync: Update module to work with Stasis (and compile) + This patch fixes res_corosync such that it works with Asterisk + 12. This restores the functionality that was present in previous + versions of Asterisk, and ensures compatibility with those + versions by restoring the binary message format needed to pass + information from/to them. The following changes were made in the + core to support this: * The event system has been partially + restored. All event definition and event types in this patch were + pulled from Asterisk 11. Previously, we had hoped that this + information would live in res_corosync; however, the approach in + this patch seems to be better for a few reasons: (1) + Theoretically, ast_events can be used by any module as a binary + representation of a Stasis message. Given the structure of an + ast_event object, that information has to live in the core to be + used universally. For example, defining the payload of a device + state ast_event in res_corosync could result in an incompatible + device state representation in another module. (2) Much of this + representation already lived in the core, and was not easily + extensible. (3) The code already existed. :-) * Stasis message + types now have a message formatter that converts their payload to + an ast_event object. * Stasis message forwarders now handle + forwarding to themselves. Previously this would result in an + infinite recursive call. Now, this simply creates a new + forwarding object with no forwards set up (as it is the thing it + is forwarding to). This is advantageous for res_corosync, as + returning NULL would also imply an unrecoverable error. Returning + a subscription in this case allows for easier handling of message + types that are published directly to an aggregate topic that has + forwarders. Review: https://reviewboard.asterisk.org/r/3486/ + ASTERISK-22912 #close ASTERISK-22372 #close ........ Merged + revisions 414330 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-21 22:24 +0000 [r414297] Richard Mudgett <rmudgett@digium.com> + + * /, main/core_unreal.c: core_unreal: Only block media frames when + a generator is on both ends of an unreal channel. The fix for + ASTERISK-12292 was a bit too aggressive. You could have + generators pointed at each other on local channels but need to + get other kinds of frames such as DTMF or CONNECTED_LINE frames + accross. ........ Merged revisions 414269 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414270 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414272 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-21 19:08 +0000 [r414217] Scott Griepentrog <sgriepentrog@digium.com> + + * /, funcs/func_strings.c: pbx.c: prevent potential crash from + recursive replace() Recurisve usage of replace() resulted in + corruption of the temporary string storage and potential crash. + By changing the string to be allocated separtely per instance, + this is eliminated. ASTERISK-23650 #comment Reported by: Roel van + Meer ASTERISK-23650 #close Review: + https://reviewboard.asterisk.org/r/3539/ ........ Merged + revisions 414214 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414215 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414216 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-19 19:52 +0000 [r414196] Paul Belanger <paul.belanger@polybeacon.com> + + * res/res_stasis_answer.c, /: Replace __ast_answer with + ast_raw_answer in app_control_answer While load testing an ARI + application, I noticed asterisk was returning HTTP 500 internal + server errors on channels/:id/answer. After talking to + #asterisk-dev, the issue appeared to be a lack of media flowing + after __ast_answer() was called. So now, we call ast_raw_answer + instead and no longer wait for media. ASTERISK-23758 #close + Review: https://reviewboard.asterisk.org/r/3549/ ........ Merged + revisions 414195 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-19 01:10 +0000 [r414123-414138] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + main/bridge_channel.c, res/res_pjsip_refer.c, + res/res_pjsip_session.c, main/channel.c, /, main/framehook.c: + Undo r414123 The Test Suite caught a few problems, undoing until + those are resolved + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + main/bridge_channel.c, res/res_pjsip_session.c, main/channel.c, + /, main/framehook.c: bridge_native_rtp/bridge_channel: Fix direct + media issues due to frame hook This patch fixes issues with + direct media bridges that occur after a blind transfer. These + issues were caught by the (currently failing) + pjsip/transfers/blind_transfer/caller_direct_media test. The test + currently fails primarily for two reasons: (1) When Bob and + Charlie (the transfer target and the transfer destination) enter + a bridge together, the framehook remains on the transfer target + channel until both channels are in the bridge. As it consumes + voice frames, the initial bridge type is a simple bridge. The + framehook is removed when both channels are in the bridge; + however, this does not currently cause the bridging framework to + re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE + poke to the transfer target channel when a framehook is removed + so the bridge can re-evaluate itself. (2) When a channel leaves a + native RTP bridge, it may be leaving due to being hung up. + Sending a re-INVITE to a channel that is about to be hung up is + not nice - in fact, there's a good chance we'll send the BYE + request before the channel has had a chance to send back a 200 + OK. To be somewhat nicer, this patch adds a function to channel.h + that allows the bridging framework to query for exactly why a + channel is leaving a bridge via the channel's soft hangup flags. + This allows it to only send the re-INVITE if there's a chance the + channel will survive the native bridging experience. Review: + https://reviewboard.asterisk.org/r/3535/ ........ Merged + revisions 414122 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-16 20:06 +0000 [r413994-414070] Richard Mudgett <rmudgett@digium.com> + + * /, channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone + detection. * Check if waitingfordt (waitfordialtone) is enabled + in dahdi_read() to allow the DSP to operate early enough to + detect dialtone. * Made use the correct variable in + my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve + Davies Patches: dialtone_detect_fix (license #5012) patch + uploaded by Steve Davies Review: + https://reviewboard.asterisk.org/r/3534/ ........ Merged + revisions 414067 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414068 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414069 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sig_pri.c, /: sig_pri.c: Pull the pri_dchannel() + PRI_EVENT_RING case into its own function. * Populate the + CALLERID(ani2) value (and the special CALLINGANI2 channel + variable) with the ANI2 value in addition to the PRI specific + ANI2 channel variable. * Made complete snapshot staging with the + channel lock held. All channel snapshots need to be done while + the channel lock is held. ........ Merged revisions 414050 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414051 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI + conference data structure. Starting a conference recording using + the admin menu overwrites the DAHDI conference data structure + used to modify the admin user's conference mute mode. * Made no + longer pass the user's DAHDI conference data structure into the + menu functions. The menu now uses its own DAHDI conference data + structure to start the recording channel. * Moved the unlock + conf->playlock to before playing the conf-full message. No sense + keeping the lock while that prompt is playing. The user is never + going to get into the conference at that point. ........ Merged + revisions 413991 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413992 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413993 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-14 15:41 +0000 [r413897] Walter Doekes <walter+asterisk@wjd.nu> + + * /, res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a + few free()'s that should be ast_free()'s. Reverted an old + workaround that isn't necessary. Reorder a tiny bit of code. + Remove a bit of commented-out code. Review: + https://reviewboard.asterisk.org/r/3536/ ........ Merged + revisions 413894 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413895 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413896 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-13 18:09 +0000 [r413878] Jonathan Rose <jrose@digium.com> + + * main/netsock2.c, /, channels/chan_sip.c, + include/asterisk/netsock2.h: chan_sip: Add TLS and SRTP status to + CLI command 'sip show channel' ASTERISK-23564 #close Reported by: + Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ + ........ Merged revisions 413876 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413877 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-13 13:53 +0000 [r413790-413793] Walter Doekes <walter+asterisk@wjd.nu> + + * res/res_format_attr_h264.c, /: h264: Fix H264 SDP payload format. + https://tools.ietf.org/html/rfc3984#section-8.1 says + profile-level-id takes 3 bytes in base16 (6 hex digits). This + fixes video setup in certain cases. ASTERISK-23664 #close + ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume + Maudoux. Review: https://reviewboard.asterisk.org/r/3530/ + ........ Merged revisions 413791 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413792 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/rtp_engine.c: rtp: Fix case typo in H263+ mime. + http://tools.ietf.org/html/rfc3555#section-4.2.6 says the + canonical mime subtype is "H263-1998", not "h263-1998". Original + code was added in r183101 on 2009-03-19 02:26:50 +0100. This + fixes issues with Polycom phones. ASTERISK-23665 #close + ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume + Maudoux, backported by me. Review: + https://reviewboard.asterisk.org/r/3529/ ........ Merged + revisions 413787 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413788 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413789 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-13 00:35 +0000 [r413770-413772] Richard Mudgett <rmudgett@digium.com> + + * configure.ac, channels/sig_pri.c, /, configure, + include/asterisk/autoconfig.h.in: chan_dahdi/sig_pri: Prevent + unnecessary PROGRESS events when overlap dialing is enabled. When + overlap dialing is enabled, the lack of inband audio available + information in the SETUP_ACKNOWLEDGE events causes an + interoperability problem with SIP. sig_pri doesn't know if there + is dialtone present when a SETUP_ACKNOWLEDGE is received so it + assumes it is there and posts an AST_CONTROL_PROGRESS frame. The + SIP channel driver then sends out a 183 Session Progress and + blocks the desired 180 Ringing message when the ALERTING message + comes in. * Made the configure script detect if the installed + version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * + Using the new API, made generate an AST_CONTROL_PROGRESS frame on + an incoming SETUP_ACKNOWLEDGE message when the message indicates + inband audio is present instead of assuming that dialtone is + present. * Using the new API, made SETUP_ACKNOWLEDGE send out an + inband audio available indication only if dialtone is expected. + The change also makes the fallback behaviour of sending the + PROGRESS message better by sending it only if dialtone is + expected. * Changed receiving a PROCEEDING message to not + generate an AST_CONTROL_PROGRESS frame if the progress indication + ie indicates non-end-to-end-ISDN. This helps interoperability + with SIP. * Changed sending a PROCEEDING message in response to + an AST_CONTROL_PROCEEDING frame to not indicate inband audio + available. It was silly to do so anyway because the channel + driver doesn't know if inband audio is even available. This helps + interoperability with SIP. This patch and a corresponding change + in libpri work together to allow Asterisk to control the inband + audio available progress indication ie on the SETUP_ACKNOWLEDGE + message when dialtone is present. AST-1338 #close Reported by: + Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ + ........ Merged revisions 413714 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413765 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413771 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/sig_pri.c: Fix compiler warning from GCC 4.10 fixup. + ........ Merged revisions 413766 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-12 22:33 +0000 [r413713] Jonathan Rose <jrose@digium.com> + + * apps/app_chanspy.c, /: app_chanspy: Fix a test that was failing + on account of r413551 ASTERISK-23381 #close ASTERISK-23381 + #comment Reported by: Robert Moss Review: + https://reviewboard.asterisk.org/r/3505/ ........ Merged + revisions 413710 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413712 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-11 02:09 +0000 [r413651-413682] Joshua Colp <jcolp@digium.com> + + * main/bridge_basic.c, include/asterisk/channel.h, + bridges/bridge_native_rtp.c, include/asterisk/framehook.h, + main/channel.c, /, main/framehook.c: framehooks: Add callback for + determining if a hook is consuming frames of a specific type. In + the past framehooks have had no capability to determine what + frame types a hook is actually interested in consuming. This has + meant that code has had to assume they want all frames, thus + preventing native bridging. This change adds a callback which + allows a framehook to be queried for whether it is consuming a + frame of a specific type. The native RTP bridging module has also + been updated to take advantange of this, allowing native bridging + to occur when previously it would not. ASTERISK-23497 #comment + Reported by: Etienne Lessard ASTERISK-23497 #close Review: + https://reviewboard.asterisk.org/r/3522/ ........ Merged + revisions 413681 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + include/asterisk/framehook.h, main/channel.c, /, + main/framehook.c, main/bridge_basic.c: Undoing framehook support. + Issues were uncovered by Bamboo. + + * /, main/framehook.c, main/bridge_basic.c, + include/asterisk/channel.h, bridges/bridge_native_rtp.c, + include/asterisk/framehook.h, main/channel.c: framehooks: Add + callback for determining if a hook is consuming frames of a + specific type. In the past framehooks have had no capability to + determine what frame types a hook is actually interested in + consuming. This has meant that code has had to assume they want + all frames, thus preventing native bridging. This change adds a + callback which allows a framehook to be queried for whether it is + consuming a frame of a specific type. The native RTP bridging + module has also been updated to take advantange of this, allowing + native bridging to occur when previously it would not. + ASTERISK-23497 #comment Reported by: Etienne Lessard + ASTERISK-23497 #close Review: + https://reviewboard.asterisk.org/r/3522/ ........ Merged + revisions 413650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-09 23:18 +0000 [r413589-413599] Kinsey Moore <kmoore@digium.com> + + * /, funcs/func_env.c: Fix 32bit build for func_env ........ Merged + revisions 413592 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413595 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413597 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_festival.c, pbx/dundi-parser.c, apps/app_getcpeid.c, + main/netsock.c, funcs/func_channel.c, main/audiohook.c, + pbx/pbx_config.c, res/res_pjsip_registrar.c, main/xmldoc.c, + channels/iax2/firmware.c, apps/app_voicemail.c, main/format.c, + cel/cel_pgsql.c, main/rtp_engine.c, main/parking.c, + main/bridge.c, res/res_jabber.c, res/res_http_websocket.c, + main/config.c, res/res_format_attr_opus.c, main/loader.c, + res/parking/parking_bridge.c, main/cdr.c, main/manager.c, + include/asterisk/astobj.h, main/bucket.c, apps/app_dumpchan.c, + main/app.c, res/res_pjsip/config_transport.c, + res/res_pjsip_refer.c, channels/chan_mgcp.c, + res/res_rtp_asterisk.c, main/slinfactory.c, main/core_unreal.c, + res/res_pjsip_sdp_rtp.c, res/res_crypto.c, main/acl.c, + channels/sig_pri.c, res/res_monitor.c, res/res_srtp.c, + main/data.c, res/res_corosync.c, channels/sip/config_parser.c, + res/res_fax_spandsp.c, apps/app_stack.c, main/asterisk.c, + main/udptl.c, res/res_sorcery_config.c, main/security_events.c, + res/res_timing_dahdi.c, res/res_pjsip_t38.c, + res/res_musiconhold.c, main/taskprocessor.c, + res/res_format_attr_h263.c, res/res_xmpp.c, res/res_pktccops.c, + funcs/func_hangupcause.c, channels/chan_phone.c, + main/manager_bridges.c, cel/cel_odbc.c, channels/chan_skinny.c, + channels/chan_motif.c, res/res_agi.c, main/logger.c, + funcs/func_srv.c, channels/chan_alsa.c, apps/app_confbridge.c, + res/res_pjsip_pubsub.c, channels/sip/include/sip.h, main/sched.c, + apps/app_adsiprog.c, main/pbx.c, channels/chan_sip.c, + res/res_fax.c, main/aoc.c, res/res_calendar_ews.c, + res/parking/parking_bridge_features.c, channels/iax2/parser.c, + main/callerid.c, main/file.c, + res/res_pjsip/pjsip_configuration.c, main/adsi.c, + main/config_options.c, pbx/pbx_dundi.c, funcs/func_iconv.c, + main/bridge_channel.c, res/res_odbc.c, channels/chan_pjsip.c, + res/parking/parking_manager.c, res/res_calendar.c, /, + funcs/func_sysinfo.c, main/utils.c, cdr/cdr_adaptive_odbc.c, + res/res_calendar_caldav.c, res/res_stasis_snoop.c, + res/res_format_attr_h264.c, main/channel.c, res/ael/pval.c, + res/res_ari_model.c, channels/chan_dahdi.c, + channels/sig_analog.c, funcs/func_frame_trace.c, + res/res_format_attr_silk.c, main/manager_channels.c, + apps/app_dial.c, res/res_calendar_icalendar.c, main/translate.c, + apps/app_queue.c, channels/chan_jingle.c, res/res_stun_monitor.c, + main/abstract_jb.c, res/res_stasis_recording.c, apps/app_sms.c, + main/event.c, apps/app_verbose.c, main/dsp.c, + channels/chan_unistim.c, main/frame.c, res/res_stasis_playback.c, + main/ccss.c, funcs/func_env.c, main/devicestate.c, + bridges/bridge_softmix.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/enum.c, main/cli.c, + res/res_format_attr_celt.c, apps/confbridge/conf_config_parser.c, + main/io.c, channels/pjsip/dialplan_functions.c, + res/res_config_odbc.c, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, formats/format_pcm.c, + apps/app_minivm.c, main/stdtime/localtime.c, main/stun.c: Allow + Asterisk to compile under GCC 4.10 This resolves a large number + of compiler warnings from GCC 4.10 which cause the build to fail + under dev mode. The vast majority are signed/unsigned mismatches + in printf-style format strings. ........ Merged revisions 413586 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 413587 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413588 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-09 18:15 +0000 [r413572] Richard Mudgett <rmudgett@digium.com> + + * main/http.c: http.c: Remove dead code. + +2014-05-09 17:03 +0000 [r413557] Jonathan Rose <jrose@digium.com> + + * apps/app_chanspy.c, /: app_chanspy: Fix a bug where Barge mode + could fail If the barge audiohook was attached prior to the spyee + and its peer actually being bridged, the audiohook would not be + applied and the connected peer would not be able to hear audio + from the spy when the spy is in barge mode. (closes issue + ASTERISK-23381) Reported by: Robert Moss Review: + https://reviewboard.asterisk.org/r/3505/ ........ Merged + revisions 413551 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413556 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-08 00:36 +0000 [r413488] Joshua Colp <jcolp@digium.com> + + * apps/app_queue.c, main/manager.c, /: app_queue: Extend + documentation for various Manager actions and events. ........ + Merged revisions 413485 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413486 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413487 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-07 21:58 +0000 [r413469] Mark Michelson <mmichelson@digium.com> + + * funcs/func_presencestate.c: Ensure that presence state is decoded + properly on Asterisk startup. The CustomPresence provider + callback will automatically base64 decode stored data if the 'e' + option was present when the state was set. However, since the + provider callback was bypassed on Asterisk startup, encoded + presence subtypes and messages were being sent instead. This fix + makes it so the provider callback is always used when providing + presence state updates. + +2014-05-07 20:59 +0000 [r413453-413455] Richard Mudgett <rmudgett@digium.com> + + * apps/app_confbridge.c, /: app_confbridge: Fixed "CBAnn" channels + not going away. Fixed a ref leak in conf_handle_talker_cb() + everytime the conference bridge was found to report a channel's + talker status change. The resulting leak caused the "CBAnn" + channels and the conference bridge to never be destroyed. Thanks + to Richard Kenner on the asterisk-user's list for locating the + problem. Reported by: Richard Kenner ........ Merged revisions + 413454 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_confbridge.c, /: app_confbridge: Fix ref leak in CLI + "confbridge kick" command. Fixed ref leak in the CLI "confbridge + kick" command when the channel to be kicked was not in the + conference. ........ Merged revisions 413451 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413452 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-07 17:56 +0000 [r413307-413399] Mark Michelson <mmichelson@digium.com> + + * res/res_config_odbc.c, /: Fix encoding of custom prepare extra + data. Patches: res_config_odbc-take2.patch by John Hardin + (License #6512) ........ Merged revisions 413396 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413397 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/presence_xml.c, /, + res/res_pjsip_pidf_digium_body_supplement.c: Improve XML + sanitization in NOTIFYs, especially for presence subtypes and + messages. Embedded carriage return line feed combinations may + appear in presence subtypes and messages since they may be + derived from user input in an instant messenger client. As such, + they need to be properly escaped so that XML parsers do not vomit + when the messages are received. ........ Merged revisions 413372 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_registrar.c, /: Check for an act on failures to + update contacts during registration. There was an underlying + issue in a realtime backend where database updates would fail. + Since we were not checking for failure, we would end up in a + strange state where the old database entry was still present but + Asterisk thought that it had been updated. Now when an entry + fails to update, we print a warning and delete the old contact + from sorcery so there is no mismatch between foreground and + backend state. Patches: res_pjsip_registrar.patch by John Hardin + (License #6512) ........ Merged revisions 413358 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_config_odbc.c, /: Ensure that all parts of SQL UPDATEs + and DELETEs are encoded. Patches: res_config_odbc.patch by John + Hardin (License #6512) ........ Merged revisions 413304 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413305 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-02 20:28 +0000 [r413227-413263] Mark Michelson <mmichelson@digium.com> + + * /, res/res_config_odbc.c: Prevent crashes in res_config_odbc due + to uninitialized string fields. Patches: odbc-crash.patch by John + Hardin (License #6512) ........ Merged revisions 413241 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413251 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413258 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_config_pgsql.c, /: Return the number of rows affected by + a SQL insert, rather than an object ID. The realtime API + specifies that the store callback is supposed to return the + number of rows affected. res_config_pgsql was instead returning + an Oid cast as an int, which during any nominal execution would + be cast to 0. Returning 0 when more than 0 rows were inserted + causes problems to the function's callers. To give an idea of how + strange code can be, this is the necessary code change to fix a + device state issue reported against chan_pjsip in Asterisk 12+. + The issue was that the registrar would attempt to insert contacts + into the database. Because of the 0 return from res_config_pgsql, + the registrar would think that the contact was not successfully + inserted, even though it actually was. As such, even though the + contact was query-able and it was possible to call the endpoint, + Asterisk would "think" the endpoint was unregistered, meaning it + would report the device state as UNAVAILABLE instead of + NOT_INUSE. The necessary fix applies to all versions of Asterisk, + so even though the bug reported only applies to Asterisk 12+, the + code correction is being inserted into 1.8+. Closes issue + ASTERISK-23707 Reported by Mark Michelson ........ Merged + revisions 413224 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413225 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-02 16:39 +0000 [r413211] Richard Mudgett <rmudgett@digium.com> + + * UPGRADE.txt, res/res_pjsip_refer.c, /, channels/chan_sip.c: + res_pjsip_refer: Add Referred-By header on INVITE for blind + transfers. Per rfc3892, the Referred-By header in a REFER must be + copied into the referenced request (IE. The outgoing INVITE to + the transfer target). * Automatically put the Referred-By header + in the outgoing INVITE message if the SIPREFERREDBYHDR channel + variable is defined with a value. * Made + chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance + so chan_pjsip has a better chance to interoperate. * Fixed + refer_blind_callback() and refer_incoming_refer_request() to not + modify the data in the pointer returned by + pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data + since the calling routine doesn't own the buffer. ASTERISK-23501 + #close Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/3514/ ........ Merged + revisions 413210 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-02 16:06 +0000 [r413197] Jonathan Rose <jrose@digium.com> + + * res/parking/res_parking.h, /, CHANGES, + res/parking/parking_bridge_features.c, + res/parking/parking_manager.c: Parking: Add 'AnnounceChannel' + argument to manager action 'Park' (closes ASTERISK-23397) + Reported by: Denis Review: + https://reviewboard.asterisk.org/r/3446/ ........ Merged + revisions 413196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-01 16:21 +0000 [r413174-413183] Mark Michelson <mmichelson@digium.com> + + * funcs/func_presencestate.c: Make behavior of the PRESENCE_STATE + 'e' option more consistent. When writing presence state, if 'e' + is specified, then the presence state will be stored in the astdb + encoded. However, consumers of presence state events or those + that query for the presence state will be given decoded + information. If base64 encoding is desired for consumers, then + the information can be base64-encoded manually and the 'e' option + can be omitted. closes issue ASTERISK-23671 Reported by Mark + Michelson Review: https://reviewboard.asterisk.org/r/3482 + + * res/res_pjsip_exten_state.c, /: Remove unnecessary repetition + checks from res_pjsip_exten_state The PBX core already takes care + of ensuring that repeated state changes are not communicated to + exten state consumers. Because the check in res_pjsip_exten_state + was incomplete, it was causing valid presence state changes not + to be sent out. For instance, if the presence state did not + change but the message or subtype did, then no presence-related + NOTIFY request would be sent out. closes issue ASTERISK-23672 + Reported by Mark Michelson ........ Merged revisions 413173 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-01 12:31 +0000 [r413160] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip/config_transport.c, /: res_pjsip: Add the ability + to configure ciphers based on name. Previously this code would + only accept the OpenSSL identifier instead of the documented + name. ASTERISK-23498 #close ASTERISK-23498 #comment Reported by: + Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/ + ........ Merged revisions 413159 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 21:03 +0000 [r413144] Richard Mudgett <rmudgett@digium.com> + + * main/message.c, /, channels/chan_sip.c, + include/asterisk/message.h, res/res_pjsip_messaging.c: + chan_sip.c: Fixed off-nominal message iterator ref count and + alloc fail issues. * Fixed early exit in sip_msg_send() not + destroying the message iterator. * Made + ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() + tolerant of a NULL iter parameter in case + ast_msg_var_iterator_init() fails. * Made + ast_msg_var_iterator_destroy() clean up any current message data + ref. * Made struct ast_msg_var_iterator, + ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), + ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() + use iter instead of i. * Eliminated RAII_VAR usage in + res_pjsip_messaging.c:vars_to_headers(). ........ Merged + revisions 413139 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413142 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 20:39 +0000 [r413141] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_pjsip.c: chan_pjsip: Fix deadlock when + retrieving call-id of channel. If a task was in-flight which + required the channel or bridge lock it was possible for the + synchronous task retrieving the call-id to deadlock as it holds + those locks. After discussing with Mark Michelson the synchronous + task was removed and the call-id accessed directly. This should + be safe as each object involved is guaranteed to exist and the + call-id will never change. ........ Merged revisions 413140 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 13:08 +0000 [r413125] Kinsey Moore <kmoore@digium.com> + + * res/res_http_websocket.c, /: Websocket: Add session locking and + delay close This resolves a race condition where data could be + written to a NULL FILE pointer causing a crash as a websocket + connection was in the process of shutting down by adding locking + to websocket session writes and by deferring session teardown + until session destruction. (closes issue ASTERISK-23605) Review: + https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan + ........ Merged revisions 413123 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413124 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 12:42 +0000 [r413118-413122] Joshua Colp <jcolp@digium.com> + + * /, res/stasis/control.c: res_stasis: Add progress indications to + operations which perform media. This change fixes operations + which did not account for the fact that they may be executed on + channels which have not been answered. These operations will now + indicate progress when invoked. ASTERISK-23560 #close + ASTERISk-23560 #comment Reported by: Jan Svoboda Review: + https://reviewboard.asterisk.org/r/3495/ ........ Merged + revisions 413121 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix issue where + sending a hold SDP twice could cause an unhold. This change fixes + a bug where if an SDP with media address and sendonly was + received twice the underlying call would go off hold, instead of + remaining on hold. This occured because the code did not properly + take into account that the SDP may contain both a valid media + address and the sendonly attribute. The code now examines the + sendonly attribute and media address first, so if the SDP is + received again no change will occur. ASTERISK-23558 #comment + Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/3472/ ........ Merged + revisions 413119 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip: + Add support for picking up calls in the configured pickup group. + AST-1363 Review: https://reviewboard.asterisk.org/r/3478/ + ........ Merged revisions 413117 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-29 15:10 +0000 [r413103] George Joseph <george.joseph@fairview5.com> + + * /, include/asterisk/spinlock.h: Add "destroy" implementation for + spinlock. The original commit for spinlock was missing "destroy" + implementations. Most of them are no-ops but phtread_spin and + pthread_mutex do need their locks destroyed. ........ Merged + revisions 413102 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-29 11:27 +0000 [r413089] Joshua Colp <jcolp@digium.com> + + * channels/chan_pjsip.c, /: chan_pjsip: Implement core ability to + get Call-ID of a channel. This changes implement the + "get_pvt_uniqueid" which is used to return the technology + specific unique identifier. In the case of SIP this is the + Call-ID of the dialog. Review: + https://reviewboard.asterisk.org/r/3480/ ........ Merged + revisions 413088 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-28 20:07 +0000 [r413074] Kinsey Moore <kmoore@digium.com> + + * /, main/bridge.c, main/bridge_basic.c: Bridging: Don't lock NULL + bridges When bridge locking was added for bridge snapshot + creation, some locations where bridge locking was added were not + guaranteed to actually have a bridge and locking NULL AO2 objects + tends to cause segfaults. This ensures that NULL bridges aren't + locked. ........ Merged revisions 413073 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-28 14:40 +0000 [r413060] Mark Michelson <mmichelson@digium.com> + + * res/res_manager_presencestate.c (added), main/devicestate.c, + CHANGES, main/presencestate.c, res/res_manager_devicestate.c + (added): Add DeviceStateChanged and PresenceStateChanged AMI + events. These events are controlled by two new modules, + res_manager_devicestate and res_manager_presencestate. Review: + https://reviewboard.asterisk.org/r/3417 + +2014-04-28 07:43 +0000 [r413048] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * UPGRADE.txt, CHANGES, channels/chan_unistim.c, + configs/unistim.conf.sample: Introducing changes proposed to + chan_unistim driver: 1) Added the unistim.conf variable + dtmf_duration which can select the DTMF playback duration from + 0ms to 150ms (0 is off and is the new default) 2) Enabled the + transmission of month names, which are sent with the date and + changed the dateformat variable to accept the values 0-3 as per + the UNISTIM standard (2 & 3 match the previous 1 & 2 formats). 3) + Enabled the "Mute" packet so muting microphone works as expected + and microphone muted for all calls while LED light on 4) Changed + Duree to Timer on i2004 display (closes issue ASTERISK-23592) + +2014-04-27 19:29 +0000 [r413036] Olle Johansson <oej@edvina.net> + + * main/tcptls.c: tcptls.c : Log errors as ERROR, not warning or + something else. + +2014-04-25 19:26 +0000 [r413012] Matthew Jordan <mjordan@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Add support for DTLS + handshake retransmissions On congested networks, it is possible + for the DTLS handshake messages to get lost. This patch adds a + timer to res_rtp_asterisk that will periodically check to see if + the handshake has succeeded. If not, it will retransmit the DTLS + handshake. Review: https://reviewboard.asterisk.org/r/3337 + ASTERISK-23649 #close Reported by: Nitesh Bansal patches: + dtls_retransmission.patch uploaded by Nitesh Bansal (License + 6418) ........ Merged revisions 413008 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413009 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-24 14:37 +0000 [r412993] Kevin Harwell <kharwell@digium.com> + + * /, + contrib/ast-db-manage/config/versions/e96a0b8071c_increase_pjsip_column_size.py + (added): pjsip realtime: increase the size of some columns The + string lengths on certain columns created through alembic for + PJSIP were too short. For instance, columns containing URIs are + currently set to 40 characters, but this can be too small and + result in truncated values. Added an alembic migration script + that increases the size of these columns and a few others to 255. + ASTERISK-23639 #close Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3475/ ........ Merged + revisions 412992 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-23 20:13 +0000 [r412977] George Joseph <george.joseph@fairview5.com> + + * include/asterisk/spinlock.h (added), /, configure, + include/asterisk/autoconfig.h.in, configure.ac: This patch adds + support for spinlocks in Asterisk. There are cases in Asterisk + where it might be desirable to lock a short critical code section + but not incur the context switch and yield penalty of a mutex or + rwlock. The primary spinlock implementations execute exclusively + in userspace and therefore don't incur those penalties. Spinlocks + are NOT meant to be a general replacement for mutexes. They + should be used only for protecting short blocks of critical code + such as simple compares and assignments. Operations that may + block, hold a lock, or cause the thread to give up it's timeslice + should NEVER be attempted in a spinlock. The first use case for + spinlocks is in astobj2 - internal_ao2_ref. Currently the + manipulation of the reference counter is done with an + ast_atomic_fetchadd_int which works fine. When weak reference + containers are introduced however, there's an additional + comparison and assignment that'll need to be done while the lock + is held. A mutex would be way too expensive here, hence the + spinlock. Given that lock contention in this situation would be + infrequent, the overhead of the spinlock is only a few more + machine instructions than the current ast_atomic_fetchadd_int + call. ASTERISK-23553 #close Review: + https://reviewboard.asterisk.org/r/3405/ ........ Merged + revisions 412976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-23 18:03 +0000 [r412925] Richard Mudgett <rmudgett@digium.com> + + * /, main/http.c: http: Fix spurious ERROR message in responses + with no content. Backport -r411687 and fix the fix because + content_length is the length of out plus the length of the file + controlled by fd. When a response has an out content length of 0, + fwrite would be called to write a buffer with no data in it. This + resulted in the following classic error message: [Apr 3 11:49:17] + ERROR[26421] http.c: fwrite() failed: Success This patch makes it + so that we only attempt to write the content of out if the out + string is non-zero. ........ Merged revisions 412922 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412923 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-23 15:02 +0000 [r412910] Russell Bryant <russell@russellbryant.com> + + * res/res_monitor.c, funcs/func_periodic_hook.exports.in (added), + main/asterisk.dynamics, funcs/func_periodic_hook.c: Fix error + loading res_monitor. For some odd reason, loading app_mixmonitor + was fine, but res_monitor was not. This patch fixes a set of + issues related to func_periodic_hook exporting the beep functions + that gets res_monitor working again. + +2014-04-22 10:09 +0000 [r412883] Joshua Colp <jcolp@digium.com> + + * /, res/stasis/app.c: res_stasis: Fix crash when handling a failed + blind transfer message. This changes fixes a crash that occurs + when stasis determines if it should send a message out to an + application or not. The code incorrectly assumed that a bridge + snapshot would always be present when in reality for failure + cases it may not be. ASTERISK-23573 #close ........ Merged + revisions 412882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-21 17:56 +0000 [r412759-412824] Jonathan Rose <jrose@digium.com> + + * CHANGES, /: chan_sip: trust_id_outbound CHANGES message + improvement (closes issue AST-1301) (closes issue ASTERISK-19465) + Reported by: Krzysztof Chmielewski ........ Merged revisions + 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 412822 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412823 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: chan_sip: Add sendrpid trust options + In r411189, some behavior was changed which made sendrpid + behavior act in a more trusting manner by sending full user data + for peers set with private caller presence in P-Asserted-Identity + headers. Since this changed long time expected behaviors, we + decided to pull that patch when that was pointed out by the + community. Instead, this patch provides a trust_id_outbound + setting which will expose the data per RFC-3325 if set to 'yes' + and simply not send the PAI/RPID headers at all if set to 'no'. + By default trust_id_outbound will be set to 'legacy' which will + preserve the behavior prior to these patches. Extra special + thanks to Walter Doekes for providing advice and feedback. + (closes issue AST-1301) (closes issue ASTERISK-19465) Reported + by: Krzysztof Chmielewski Review: + https://reviewboard.asterisk.org/r/3447/ ........ Merged + revisions 412744 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412746 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412747 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-21 16:16 +0000 [r412729-412750] Kinsey Moore <kmoore@digium.com> + + * main/http.c, main/manager.c, /: HTTP: Add TCP_NODELAY to accepted + connections This adds the TCP_NODELAY option to accepted + connections on the HTTP server built into Asterisk. This option + disables the Nagle algorithm which controls queueing of outbound + data and in some cases can cause delays on receipt of response by + the client due to how the Nagle algorithm interacts with TCP + delayed ACK. This option is already set on all non-HTTP AMI + connections and this change would cover standard HTTP requests, + manager HTTP connections, and ARI HTTP requests and websockets in + Asterisk 12+ along with any future use of the HTTP server. + Review: https://reviewboard.asterisk.org/r/3466/ ........ Merged + revisions 412745 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412748 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_confbridge.c, /: Confbridge: Fix ConfbridgeKick AMI + documentation This adds documentation for the "all" channel + option for the ConfbridgeKick AMI action and adjusts AMI + responses accordingly. (issue ASTERISK-23282) Reported by: Dorian + Logan ........ Merged revisions 412730 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_confbridge.c: Confbridge: Add references for kick all + option After the ability to kick all attendees from a conference + was added, a rework removed the comment about that feature from + the CLI documentation. This adds that documentation and adds + "all" to the participant tab completion list for the confbridge + kick command. (closes issue ASTERISK-23282) Reported by: Dorian + Logan ........ Merged revisions 412728 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-21 08:36 +0000 [r412714] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * /, channels/chan_unistim.c: Fix wrong dialtone. The "modulation" + should not be referenced for tone+tone as it refers to the on-off + characteristic - this often resulted in a single tone rather than + the multitone as in the UK. ........ Merged revisions 412712 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-19 02:14 +0000 [r412697-412699] Matthew Jordan <mjordan@digium.com> + + * /, main/asterisk.c: main/asterisk: Fix startup sequence for + realtime features When ASTERISK-23265/ASTERISK-23320 was fixed, + it inadvertently led to realtime features breaking. This was due + to features loading prior to realtime. This patch fixes this by + loading features after loading dynamic modules. ASTERISK-23487 + #close Reported by: Denis Tested by: Denis ........ Merged + revisions 412698 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_sms.c: app_sms: Fix uninitialized values; hangup + channel when REL is sent successfully This patch fixes two issues + in app_sms: (1) Firstly, the 'flags' field on the stack in + sms_exec() is uninitialised, causing it to use the wrong protocol + in some cases. This patch correctly initializes the flags fields. + (2) Secondly, when disconnect supervision is not working or + inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was + failing to terminate the call after it sent the REL(ease) message + and the peer stopped talking to it. This patch fixes the code to + handle the 'bad stop bit' message more gracefully in that case, + and hang up the call. Review: + https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close + Reported by: David Woodhouse patches: asterisk-fix-sms.patch + uploaded by David Woodhouse (License 5754) ........ Merged + revisions 412655 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412656 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412657 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 20:09 +0000 [r412641] Jonathan Rose <jrose@digium.com> + + * /, res/ari/resource_bridges.h, res/stasis/control.c, + include/asterisk/stasis_app.h, res/stasis/control.h, + res/ari/resource_channels.c, CHANGES, res/res_stasis.c, + rest-api/api-docs/bridges.json, res/ari/resource_bridges.c, + res/res_ari_bridges.c, res/res_stasis_playback.c: ARI: Make + bridges/{bridgeID}/play queue sound files Previously multiple + play actions against a bridge at one time would cause the sounds + to play simultaneously on the bridge. Now if a sound is already + playing, the play action will queue playback to occur after the + completion of other sounds currently on the queue. (closes issue + ASTERISK-22677) Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/3379/ ........ Merged + revisions 412639 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 17:17 +0000 [r412589] Rusty Newton <rnewton@digium.com> + + * sounds/sounds.xml, sounds/Makefile, /: sounds: Fix Sounds + Makefile and XML that didn't support new sound prompt sets In + sounds/Makefile 1 Adds and moves some lines necessary for the + en_GB core set. I'm just following how the other sets are defined + here. 2 removes the ES extra sounds related lines as we don't + have ES extra sound sets. In sounds/sounds.xml 3 Adds member + definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in + extra sound sets ASTERISK-23550 #close Review: + https://reviewboard.asterisk.org/r/3464/ ........ Merged + revisions 412586 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412587 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 17:02 +0000 [r412584] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip/location.c: Allow for multiple contacts to be + configured in a single contact= line. This is useful for + configuring multiple permanent contacts for an AOR when using + realtime AORs. Review: https://reviewboard.asterisk.org/r/3462 + ........ Merged revisions 412582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 16:44 +0000 [r412580-412583] Richard Mudgett <rmudgett@digium.com> + + * main/dial.c, main/pbx.c, /, apps/app_originate.c, + include/asterisk/pbx.h: Originated calls: Fix several originate + call problems. * Restore the reason value set by + pbx_outgoing_attempt() to use AST_CONTROL_xxx values as all the + consumers were expecting rather than cause codes. * Fixed the + dial routines to set cause codes for more than just ast_request() + so pbx_outgoing_attempt() reason codes will function. * Fix + inconsistent locked_channel return status in + pbx_outgoing_attempt(). The chanel may not have been locked or + the channel may have been a stale pointer. * Fixed the + OutgoingSpoolFailed channel to run dialplan whenever the dialing + fails for an originate exten and 1 < synchronous. * Fix incorrect + ast_cond_wait() usage in pbx_outgoing_attempt(). Indroduced by + issue ASTERISK-22212 patch. * Made struct pbx_outgoing use the + ao2 lock instead of its own lock for the cond wait mutex. No + sense in having two locks associated with the same struct when + only one is needed. Review: + https://reviewboard.asterisk.org/r/3421/ ........ Merged + revisions 412581 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, apps/app_queue.c, apps/app_dial.c, /: + app_dial and app_queue: Make lock the forwarding channel while + taking the channel snapshot. * Fixed + ast_channel_publish_dial_forward() not locking the forwarded + channel when taking the channel snapshot. * Fixed + app_dial.c:do_forward() using the wrong channel to get the + original call forwarding string. * Removed unnecessary locking + when calling ast_channel_publish_dial() and + ast_channel_publish_dial_forward() in app_dial and app_queue. + Holding channel locks when calling + ast_channel_publish_dial_forward() with a forwarded channel could + result in pausing the system while the stasis bus completes + processsing a forwarded channel subscription. Review: + https://reviewboard.asterisk.org/r/3451/ ........ Merged + revisions 412579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 14:25 +0000 [r412566] Kinsey Moore <kmoore@digium.com> + + * res/ari/ari_websockets.c, res/res_ari.c, main/manager.c, /: ARI: + Add debug logging for events and responses This adds DEBUG level + logging for ARI websocket events and HTTP responses similar to + what is available for AMI. Logging for ARI HTTP requests is + already adequate for debugging purposes. ........ Merged + revisions 412565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-17 22:50 +0000 [r412552] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/location.c, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + res/res_pjsip_registrar.c: res_pjsip: Handle reloading when + permanent contacts exist and qualify is configured. This change + fixes a problem where permanent contacts being qualified were not + being updated. This was caused by the permanent contacts getting + a uuid and not a known identifier, causing an inability to look + them up when updating in the qualify code. A bug also existed + where the new configuration may not be available immediately when + updating qualifies. (closes issue ASTERISK-23514) Reported by: + Richard Mudgett Review: https://reviewboard.asterisk.org/r/3448/ + ........ Merged revisions 412551 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-17 22:42 +0000 [r412536-412550] Jonathan Rose <jrose@digium.com> + + * /, main/app.c: Fix a silly shadowed variable mistake that was + missed from play tones patch ........ Merged revisions 412549 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/ari/resource_bridges.h, main/app.c, + rest-api/api-docs/channels.json, CHANGES, + rest-api/api-docs/bridges.json, res/ari/resource_channels.h, + include/asterisk/app.h, res/res_stasis_playback.c: ARI: Add tones + playback resource Adds a tones URI type to the playback resource. + The tone can be specified by name (from indications.conf) or by a + tone pattern. In addition, tonezone can be specified in the URI + (by appending ;tonezone=<zone>). Tones must be stopped manually + in order for a stasis control to move on from playback of the + tone. Tones may be paused, resumed, restarted, and stopped. They + may not be rewound or fast forwarded (tones can't be controlled + in a way that lets you skip around from note to note and pausing + and resuming will also restart the tone from the beginning). + Tests are currently in development for this feature + (https://reviewboard.asterisk.org/r/3428/). (closes issue + ASTERISK-23433) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3427/ ........ Merged + revisions 412535 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-17 20:25 +0000 [r412467-412484] Matthew Jordan <mjordan@digium.com> + + * channels/chan_oss.c, /, main/Makefile: main/Makefile: Fix build + failure on SmartOS/Illumos/SunOS This patch fixes two issues when + building on SmartOS: - channels/chan_oss.c: it makes sure + soundcard.h is found - main/Makefile: only use + "-Wl,--version-script" when GNU LD is used as the Sun Linker + doesn't support that. Similar checks are already used elswhere in + the Makefile Review: https://reviewboard.asterisk.org/r/3426 + ASTERISK-23576 #close Reported by: Sebastian Wiedenroth patches: + fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597) + ........ Merged revisions 412468 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412483 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sip/include/sip.h, channels/chan_sip.c, CHANGES: + chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL + URIs This patch is a continuation of + https://reviewboard.asterisk.org/r/3349/, committed in r412303. + It resolves a finding oej had that the phone-context be available + in a channel variable separate from SIPDOMAIN. This patch adds + that variable as SIPURIPHONECONTEXT. It also allows a local + number (or global number specified in the TEL URI) to be used to + look up as a peer. (issue ASTERISK-17179) Review: + https://reviewboard.asterisk.org/r/3349/ + +2014-04-17 15:17 +0000 [r412454] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_refer.c, /: res_pjsip_refer: Channel variable + SIPREFERTOHDR not being set during blind transfer The + SIPREFERTOHDR channel variable is not being set on any channel + when performing a blind transfer using PJSIP. The + 'refer->refer_to' was not being set during a blind transfer. + Updated so the 'refer_to' is set to the target uri on a blind + transfer. (closes issue ASTERISK-23502) Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/3445/ ........ Merged + revisions 412453 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-16 19:14 +0000 [r412440] Kinsey Moore <kmoore@digium.com> + + * /, include/asterisk/stasis_app.h: Stasis: Add a usage note on + stasis_app_get_bridge This function returns an ast_bridge without + a refcount bump and the caller must increment the count if it + intends to hold the pointer. (closes issue ASTERISK-23588) + Review: https://reviewboard.asterisk.org/r/3450/ Reported by: + Matt Jordan ........ Merged revisions 412439 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-15 23:21 +0000 [r412427] Russell Bryant <russell@russellbryant.com> + + * bridges/bridge_builtin_features.c, include/asterisk/monitor.h, + CHANGES, apps/app_queue.c, funcs/func_periodic_hook.c, + apps/app_mixmonitor.c, include/asterisk/beep.h (added), + res/res_monitor.c: (mix)monitor: Add options to enable a periodic + beep Add an option to enable a periodic beep to be played into a + call if it is being recorded. If enabled, it uses the + PERIODIC_HOOK() function internally to play the 'beep' prompt + into the call at a specified interval. This option is provided + for both Monitor() and MixMonitor(). Review: + https://reviewboard.asterisk.org/r/3424/ + +2014-04-15 18:30 +0000 [r412384-412414] Richard Mudgett <rmudgett@digium.com> + + * main/stasis_channels.c, main/features_config.c, + res/res_parking.c, main/rtp_engine.c, /: Eliminate some more + unnecessary RAII_VAR() uses. RAII_VAR() is not a hammer + appropriate to pound all nails. ........ Merged revisions 412413 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /, res/stasis/app.c, res/res_fax.c, + res/res_pjsip/security_events.c, + res/parking/parking_applications.c, channels/chan_oss.c, + main/stasis_bridges.c, res/res_pjsip_session.c, + res/stasis_recording/stored.c, main/cdr.c, res/res_parking.c, + channels/chan_skinny.c, res/res_pjsip/location.c, + res/res_stasis_recording.c, main/stasis_channels.c, + res/ari/resource_channels.c, res/parking/parking_manager.c, + res/ari/resource_recordings.c, res/res_pjsip_refer.c, + res/res_ari.c, main/pbx.c: Remove unused RAII_VAR() declarations. + * Remove unused RAII_VAR() declarations. The compiler cannot + catch these because the cleanup function "references" the unused + variable. Some actually allocated and released resources that + were never used. * Fixed some whitespace issues in + stasis_bridges.c. ........ Merged revisions 412399 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/rtp_engine.h, main/rtp_engine.c, /, + channels/chan_sip.c: chan_sip.c: Fix channel staging assertion + failure. The failing assertion ensures that the final snapshot + gets generated so CDR records can get finalized. The only place + where a channel staging snapshot flag could be left set is in + chan_sip.c:handle_request_bye(). The function could return before + clearing the flag because the channel could dissappear while the + function had to have the channel unlocked. * Fixed + handle_request_bye() channel snapshot staging coverage area to + not have a return in the middle of it and be unable to clear the + staging flag. * Pushed the channel snapshot staging coverage area + into ast_rtp_instance_set_stats_vars() to ensure that the staging + is not interrutped. * Made callers of + ast_rtp_instance_set_stats_vars() not call it with any channels + or channel driver private locks held to eliminate the deadlock + potential. The callers must hold references to the passed in + channel and rtp objects. * Eliminated sip_hangup() trying to get + the bridge peer. It is futile at this point because the channel + could never be in a bridge. Review: + https://reviewboard.asterisk.org/r/3431/ ........ Merged + revisions 412385 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip.c: Moved some sip_pvt unrefs + after their last use. * Moved sip_pvt unref in ast_hangup() and + handle_request_do() to the end of the function. The unref needs + to happen after the last use of the pointer. ........ Merged + revisions 412348 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-15 16:13 +0000 [r412331] Jonathan Rose <jrose@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Reverting + r411189 so that it can be put up for public review --- r411189 | + jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines + chan_sip: Send real CallerID information with + P-Assserted-Identity (RFC-3325) Prior to this patch, the + P-Asserted-Identity header would include anonymous caller id + information which seems to go against the point of the + P-Asserted-Identity header. Now the real caller ID information + will be included in this header. Also, no privacy header would be + included. This patch adds 'Privacy: id' to outgoing SIP messages + that include the P-Asserted-Identity header. (closes issue + AST-1301) --- ........ Merged revisions 412328 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412329 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412330 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-14 15:54 +0000 [r412307] Corey Farrell <git@cfware.com> + + * main/autoservice.c, /: autoservice: fix reference leak of logger + callid. autoservice acquires a local reference to the logger + callid of each channel in a loop. This local reference was not + released, causing the callid of every channel in autoservice to + leak. This change moves the callid unref inside the loop. + ASTERISK-23616 #close Reported by: ibercom ........ Merged + revisions 412305 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-12 02:27 +0000 [r412292] Matthew Jordan <mjordan@digium.com> + + * channels/sip/reqresp_parser.c, CHANGES, channels/chan_sip.c: + chan_sip: Support RFC-3966 TEL URIs in inbound INVITE requests + This patch adds support for handling TEL URIs in inbound INVITE + requests. This includes the Request URI and the From URI. The + number specified in the Request URI will be the destination of + the inbound channel in the dialplan. The phone-context specified + in the Request URI will be stored in the TELPHONECONTEXT channel + variable. Review: https://reviewboard.asterisk.org/r/3349 + ASTERISK-17179 #close Reported by: Geert Van Pamel Tested by: + Geert Van Pamel patches: + asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van + Pamel (License 6140) + asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by + Geert Van Pamel (License 6140) + +2014-04-12 01:35 +0000 [r412279-412280] Russell Bryant <russell@russellbryant.com> + + * funcs/func_periodic_hook.c: func_periodic_hook: move module ref + The previous code left one error path where the module would be + unref'd twice instead of once. It was done once in the error + handling block, and again inside of datastore destruction. Now + the module ref is only released in the datastore destructor and + only acquired when the datastore has been successfully allocated. + + * funcs/func_periodic_hook.c: func_periodic_hook: add module ref + counting This module lacked necessary module ref count + incrementing and decrementing when used. This patch adds it. + There's already a datastore used, so doing the ref counting along + with the lifetime of the datastore provides a convenient place to + do it. + +2014-04-11 21:43 +0000 [r412213-412228] Richard Mudgett <rmudgett@digium.com> + + * apps/app_stack.c, /: app_stack: Add missing unlock in off-nominal + path of STACK_PEEK function. ASTERISK-23620 #close Reported by: + Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch + (license #5021) patch uploaded by Bradley Watkins ........ Merged + revisions 412225 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412226 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412227 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * utils/Makefile, utils: utils dir: Remove no longer needed traces + of refcounter except in the clean make target. * Removed no + longer needed files from the svn:ignore property to make them + visible. + +2014-04-11 12:43 +0000 [r412194] Kinsey Moore <kmoore@digium.com> + + * /, main/bridge.c, main/bridge_basic.c, + include/asterisk/stasis_bridges.h, tests/test_cel.c, + apps/app_confbridge.c, res/ari/resource_bridges.c: bridging: + Ensure locking during snapshot creation While the vast majority + of bridge snapshot creation is locked properly, there are + currently some instances that are not. This adds the missing + locking to ensure bridge state is not malleable during snapshot + creation. (closes issue ASTERISK-22904) Review: + https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan + ........ Merged revisions 412193 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-11 08:28 +0000 [r412168-412180] Olle Johansson <oej@edvina.net> + + * main/audiohook.c: Formatting: Remove invisible characters + + * main/audiohook.c: Formatting only. + +2014-04-11 02:59 +0000 [r412154] Matthew Jordan <mjordan@digium.com> + + * main/astobj2.c, contrib/scripts/refcounter.py (added), + main/asterisk.c, utils/refcounter.c (removed), + build_tools/cflags.xml, utils/utils.xml, /, channels/chan_sip.c, + channels/sip/security_events.c, include/asterisk/astobj2.h, + UPGRADE.txt: main/astobj2: Make REF_DEBUG a menuselect item; + improve REF_DEBUG output This patch does the following: (1) It + makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables + REF_DEBUG globally throughout Asterisk. (2) The ref debug log + file is now created in the AST_LOG_DIR directory. Every run will + now blow away the previous run (as large ref files sometimes + caused issues). We now also no longer open/close the file on each + write, instead relying on fflush to make sure data gets written + to the file (in case the ao2 call being performed is about to + cause a crash) (3) It goes with a comma delineated format for the + ref debug file. This makes parsing much easier. This also now + includes the thread ID of the thread that caused ref change. (4) + A new python script instead for refcounting has been added in the + contrib/scripts folder. (5) The old refcounter implementation in + utils/ has been removed. Review: + https://reviewboard.asterisk.org/r/3377/ ........ Merged + revisions 412114 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412115 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412153 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-11 01:12 +0000 [r412102] Russell Bryant <russell@russellbryant.com> + + * res/res_monitor.c: monitor: use app options parsing helper code + This app is pretty ancient, so it was never converted to use the + option parsing helper code. I'd like to add an option to this app + that takes an argument, and that's a pain to do when not using + this helper, so start by doing this conversion. Review: + https://reviewboard.asterisk.org/r/3429/ + +2014-04-10 21:28 +0000 [r412089] Matthew Jordan <mjordan@digium.com> + + * /, res/res_hep_pjsip.c: res_hep_pjsip: Use the channel name + instead of the call ID when it is available During discussions + with Alexandr Dubovikov at Kamailio World, it became apparent + that while the SIP call ID is a useful identifier prior to an + Asterisk channel being created, it is far more preferable to use + the channel name (or some channel based identifier) when the + channel is available. Homer is smart enough to tie the various + messages together. This patch opts to use the channel name when + it is available, falling back to the call ID otherwise. ........ + Merged revisions 412088 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-10 21:10 +0000 [r412075] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Set the body + generation result to 0 for a valid path The result of the + "ast_sip_pubsub_generate_body_content" was not set/initialized. + Consequently, the nominal path potentially returned an invalid + value, thus not sending mwi notifications. ........ Merged + revisions 412074 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-09 21:43 +0000 [r412050] Mark Michelson <mmichelson@digium.com> + + * /, CHANGES, apps/app_mixmonitor.c: Add a Command header to the + AMI Mixmonitor action. This fixes a parsing error that occurred + during the processing of the AMI action. The error did not result + in MixMonitor itself misbehaving, but it could result in the AMI + response not giving correct information back. The new header + allows for one to specify a post-process command to run when + recording finishes. Previously, in order to do this, the + post-process command would have to be placed at the end of the + Options: header. Patches: mixmonitor_command_2.patch by jhardin + (License #6512) ........ Merged revisions 412048 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-09 18:17 +0000 [r412035] Kinsey Moore <kmoore@digium.com> + + * /, res/res_stasis_answer.c: res_stasis_answer: Add missing + newlines ........ Merged revisions 412034 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-08 21:25 +0000 [r411946-411990] Richard Mudgett <rmudgett@digium.com> + + * /, main/asterisk.c: Internal timing: Add notice that the -I and + internal_timing option are no longer needed. Add notice messages + during execution that the -I command line option and the + astersik.conf internal_timing option are no longer needed. The + internal timing functionality is now always enabled if there is a + timing module loaded. NOTE: Since the command line options and + the asterisk.conf config file are processed before the logging + system is initialized, the messages are output to stderr. Change + requested as a result of asterisk-dev list comments about the + commit for ASTERISK-22846 that removed the -I and internal_timing + options. Review: https://reviewboard.asterisk.org/r/3423/ + ........ Merged revisions 411964 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411974 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411985 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, /: config: Fix CB_ADD_LEN() to work as originally + intended. Fix a long standing bug in CB_ADD_LEN() behaving like + CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes + ........ Merged revisions 411960 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411961 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411962 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix + confbridge.conf dsp_talking_threshold option setting wrong + parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported + by: John Knott ........ Merged revisions 411944 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411945 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-08 14:49 +0000 [r411928] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip.c: res_pjsip: Ignore explicit transport + configuration if a WebSocket transport is specified. This change + makes it so if a transport is configured on an endpoint that is a + WebSocket type the option will be ignored. In practice this is + fine because the WebSocket transport can not create outgoing + connections, it can only reuse existing ones. By ignoring the + option the existing PJSIP logic for using the existing connection + will be invoked and stuff will proceed. (closes issue + ASTERISK-23584) Reported by: Rusty Newton ........ Merged + revisions 411927 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-08 00:26 +0000 [r411897] Russell Bryant <russell@russellbryant.com> + + * funcs/func_periodic_hook.c: func_periodic_hook: List more modules + as dependencies This module makes use of some existing Asterisk + components. app_chanspy was already listed as a dependency. There + are a few function modules used, as well, so list them. + +2014-04-07 20:41 +0000 [r411884] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip_pubsub.c: PJSIP: Ensure test event has new state + The change that fixed the pubsub test event's use of a dangling + pointer also changed when it was processed relative to the pjsip + subscription state change processing. This change corrects the + order of events while holding a reference to the pointer that was + previously dangling. ........ Merged revisions 411883 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-07 16:15 +0000 [r411870] Jonathan Rose <jrose@digium.com> + + * main/manager_channels.c, /: AGI/Manager: Prevent multiple + NewExten events during AGI application changes AGI applications + would trigger NewExten events every time the state of the AGI + application changed. This has historically not been the behavior + and this behavior was introduced with a CDR patch. This patch + corrects that. (closes issue ASTERISK-23390) Reported by: + Benjamin Keith Ford Review: + https://reviewboard.asterisk.org/r/3406/ ........ Merged + revisions 411868 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-07 14:57 +0000 [r411812] Walter Doekes <walter+asterisk@wjd.nu> + + * apps/app_queue.c, /: app_queue: Re-add HoldTime to + QueueCallerAbandon event (simple typo during ast12 refactor). + Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........ + Merged revisions 411811 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-07 14:29 +0000 [r411791-411806] Kinsey Moore <kmoore@digium.com> + + * /, res/res_stasis.c: Stasis: Fix Stasis() bridge refcount issue + The Stasis() dialplan application monitors what bridge a channel + is in and so necessarily holds on to a bridge pointer. This + change ensures that it also holds on to a reference for that + bridge to prevent the bridge pointer from becoming a dangling + pointer. ........ Merged revisions 411804 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c, /: PJSIP: Fix crash introduced in r411671 + The test event introduced in revision 411671 uses a dangling + pointer to access information about pubsub state changes. This + moves the event to within the lifetime of the pointer. ........ + Merged revisions 411790 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-05 13:06 +0000 [r411768] Russell Bryant <russell@russellbryant.com> + + * CHANGES, funcs/func_periodic_hook.c (added): func_periodic_hook: + New function for periodic hooks. This commit introduces a new + dialplan function, PERIODIC_HOOK(). It allows you run to a + dialplan hook on a channel periodically. The original use case + that inspired this was the ability to play a beep periodically + into a call being recorded. The implementation is much more + generic though and could be used for many other things. The + implementation makes heavy use of existing Asterisk components. + It uses a combination of Local channels and ChanSpy() to run some + custom dialplan and inject any audio it generates into an active + call. The other important bit of the implementation is how it + figures out when to trigger the beep playback. This + implementation uses the audiohook API, even though it's not + actually touching the audio in any way. It's a convenient way to + get a callback and check if it's time to kick off another beep. + It would be nice if this was timer event based instead of polling + based, but unfortunately I don't see a way to do it that won't + interfere with other things. Review: + https://reviewboard.asterisk.org/r/3362/ + +2014-04-04 19:19 +0000 [r411702-411724] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/options.h, main/asterisk.c, main/channel.c, /, + channels/chan_sip.c, configs/asterisk.conf.sample, UPGRADE.txt, + include/asterisk/channel.h, utils/extconf.c: internal_timing: + Remove the option and always make it enabled if a timing module + is loaded. The masquerade supertest frequently fails because + either the local channel chain doesn't completely optimize out or + the DTMF handshake doesn't completely get accross. Local channel + optimization requires frames flowing to trigger when optimization + can happen. When optimization happens the media frame that + triggered the optimization is dropped. Sending DTMF requires + frames to flow in the other direction for timing purposes while + sending nothing. If internal timing is not enabled when MOH is + playing, Asterisk switches to received timing when an audio frame + is received. With optimization dropping media frames and MOH not + sending frames unless it receives frames, occasionaly there are + no more frames being passed and the test fails. * The asterisk + command line -I option and the asterisk.conf internal_timing + option are removed. Asterisk now always uses internal timing when + needed if any timing module is loaded. The issue ASTERISK-14861 + did this quite awhile ago in v1.4 but effectively is broken if + other internal timing modules besides DAHDI are used. The + ast_read_generator_actions() now only does received timing if it + has no choice for frame generators like MOH, silence, and + playback streaming. * Cleaned up some code dealing with frame + generators in ast_deactivate_generator(), + generator_write_format_change(), ast_activate_generator(), and + ast_channel_stop_silence_generator(). * Removed + ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and + ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ + Merged revisions 411715 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411716 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411717 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/utils.c, res/res_musiconhold.c, main/channel.c, + main/stasis_cache.c, /: Add some asserts that were handy when + looking for a stasis cache problem. * Assert if a channel is + destroyed but has the snapshot staging flag set. In this case the + final channel destruction snapshot would never get taken. * + Assert if what we just got out of the stasis cache is not what we + were looking for. This assert would have saved several days + searching for a bug and a lot of my hair. * Assert if the music + on hold message posts could not find the associated channel. A + crash will happen later when manager tries to send the MOH AMI + message. This assert catches the problem when the stasis message + is posted instead of by the thread processing the defective + message. * Always generate a backtrace when an ast_assert() + fails. Review: https://reviewboard.asterisk.org/r/3411/ ........ + Merged revisions 411701 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-04 15:13 +0000 [r411688] Matthew Jordan <mjordan@digium.com> + + * /, main/http.c: http: Fix spurious ERROR message in responses + with no content When a response has a content length of 0, fwrite + would be called to write a buffer with no data in it. This + resulted in the following classic error message: [Apr 3 11:49:17] + ERROR[26421] http.c: fwrite() failed: Success This patch makes it + so that we only attempt to write out the content if the + calculated content_length is non-zero. ........ Merged revisions + 411687 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-03 12:06 +0000 [r411671] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Add test event for + state change This adds a test event when subscription state + changes so that integration tests may trigger new actions at the + appropriate times. Review: + https://reviewboard.asterisk.org/r/3383/ ........ Merged + revisions 411670 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-03 11:47 +0000 [r411669] Matthew Jordan <mjordan@digium.com> + + * res/res_hep.c, /: res_hep: Fix crash when hep.conf not available + Parts of res_hep properly checked for a valid configuration + object before attempting to access the configuration. A check, + however, was missed when a packet is sent. This patch fixes the + crash caused by not checking if the configuration object is + valid. ........ Merged revisions 411668 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-02 18:57 +0000 [r411656] Mark Michelson <mmichelson@digium.com> + + * main/sorcery.c, /, res/res_mwi_external.c, + res/res_pjsip/config_system.c, configs/sorcery.conf.sample, + main/bucket.c, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c, + tests/test_sorcery.c, tests/test_sorcery_realtime.c: Prevent + duplicate sorcery wizards from being applied to sorcery object + types. This commit contains several changes to sorcery: 1) + Application of sorcery configuration based on module name is + automatically performed when sorcery is opened for a module. 2) + Sorcery will not attempt to apply the same wizard to an object + type more than once. 3) Sorcery gives more exact results when + attempting to apply a wizard, whether as the default or based on + configuration. Sorcery unit tests still pass for me after making + these changes. Review: https://reviewboard.asterisk.org/r/3326 + ........ Merged revisions 411159 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-01 22:42 +0000 [r411637-411639] Richard Mudgett <rmudgett@digium.com> + + * res/parking/parking_bridge.c, /: res_parking: Minor tweaks. * Use + ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of + ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables. + * Use ast_copy_string() instead of inlining it. * Remove an + already done TODO comment. * Some whitespace tweaks. ........ + Merged revisions 411638 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, /: stasis_channels.c: Eliminate another + overuse of RAII_VAR(). ........ Merged revisions 411636 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-01 16:52 +0000 [r411587] Joshua Colp <jcolp@digium.com> + + * /, apps/app_queue.c: app_queue: Fix a bug where realtime members + would be deleted during reload causing waiting callers to get + ejected. This patch causes realtime queue members to remain in + queues during the reload process. Previously these members would + be removed causing any waiting callers to be ejected from the + queue with a reason of "EXITEMPTY". ASTERISK-23547 #close + ASTERISK-23547 #comment Patch + app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo + Rossi (license 6409) Review: + https://reviewboard.asterisk.org/r/3404/ ........ Merged + revisions 411584 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411585 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411586 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 18:32 +0000 [r411556] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added), + res/res_hep.exports.in (added), configs/hep.conf.sample (added), + CHANGES, res/res_hep.c (added), /: res_hep/res_hep_pjsip: Add a + HEPv3 capture agent module and a logger for PJSIP This patch adds + the following: (1) A new module, res_hep, which implements a + generic packet capture agent for the Homer Encapsulation Protocol + (HEP) version 3. Note that this code is based on a patch provided + by Alexandr Dubovikov; I basically just wrapped it up, added + configuration via the configuration framework, and threw in a + taskprocessor. (2) A new module, res_hep_pjsip, which forwards + all SIP message traffic that passes through the res_pjsip stack + over to res_hep for encapsulation and transmission to a HEPv3 + capture server. Much thanks to Alexandr for his Asterisk patch + for this code and for a *lot* of patience waiting for me to port + it to 12/trunk. Due to some dithering on my part, this has taken + the better part of a year to port forward (I still blame CDRs for + the delay). ASTERISK-23557 #close Review: + https://reviewboard.asterisk.org/r/3207/ ........ Merged + revisions 411534 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 18:00 +0000 [r411533] Alexandr Anikin <may@telecom-service.ru> + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, + addons/chan_ooh323.c, /, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c: + process stack command even if gatekeeper client isn't register + don't destroy gatekeeper client if it is not started don't + destroy gatekeeper client in some sort of gatekeeper errors + signal rtp create condition when call cleared before rtp + structure created (closes issue ASTERISK-23460) Reported by: + Dmitry Melekhov Patches: ASTERISK-23460-2.patch Tested by: Dmitry + Melekhov ........ Merged revisions 411531 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411532 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 17:41 +0000 [r411515-411530] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/channels.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/playbacks.json, UPGRADE.txt, + rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES, + include/asterisk/manager.h, rest-api/api-docs/bridges.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/mailboxes.json, + rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json: Update API versions and + UPGRADE/CHANGES for 12.2.0 This patch does the following: * It + updates the AMI version to 2.2.0 to indicate backwards compatible + changes have been made since the last release * It updates the + ARI version to 1.2.0 to indicate backwards compatible changes + have been made since the last release * It updates the + UPGRADE/CHANGES files with changes that were not mentioned + ........ Merged revisions 411529 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * UPGRADE.txt, res/res_config_odbc.c: res_config_odbc: Fix for + nullable integer columns and keyfield existence check in + update_odbc. This patch fixes setting nullable integer columns to + NULL instead of an empty string, which fails for PostgreSQL, for + example. The current code is supposed to do so, but the check is + broken. The patch also allows the first column in the list to be + a nullable integer. Also, the check for existence of a mandatory + column checked for the first column in the list instead of the + key field lookup column. This patch fixes that issue as well. + Finally, the compatibility option allow_empty_string_in_nontext, + which was added to previous revisions to allow for some database + backends with certain schemas to function, has been removed. + Review: https://reviewboard.asterisk.org/r/3335 ASTERISK-23459 + #close ASTERISK-23351 #close (closes issue ASTERISK-23459) + Reported by: zvision patches: res_config_odbc.diff uploaded by + zvision (License 5755) + +2014-03-28 16:18 +0000 [r411469] Scott Griepentrog <sgriepentrog@digium.com> + + * main/tcptls.c, main/manager.c, /, main/http.c: http: response + body often missing after specific request This patch works around + a problem with the HTTP body being dropped from the response to a + specific client and under specific circumstances: a) Client + request comes from node.js user agent "Shred" via use of + swagger-client library. b) Asterisk and Client are *not* on the + same host or TCP/IP stack In testing this problem, it has been + determined that the write of the HTTP body is lost, even if the + data is written using low level write function. The only solution + found is to instruct the TCP stack with the shutdown function to + flush the last write and finish the transmission. See review for + more details. ASTERISK-23548 #close (closes issue ASTERISK-23548) + Reported by: Sam Galarneau Review: + https://reviewboard.asterisk.org/r/3402/ ........ Merged + revisions 411462 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411463 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 15:48 +0000 [r411375-411460] Matthew Jordan <mjordan@digium.com> + + * UPGRADE.txt, /: UPGRADE: Note IAX2 compatibility issue between + 1.4 and 1.8+ systems. ........ Merged revisions 411457 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411458 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411459 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/realtime/mysql/voicemail_messages.sql (removed), + contrib/realtime/postgresql/realtime.sql (removed), + contrib/realtime/mysql/voicemail_data.sql (removed), + contrib/realtime/mysql/musiconhold.sql (removed), + contrib/realtime/mysql/queue_log.sql (removed), + contrib/realtime/mysql/voicemail.sql (removed), + contrib/realtime/mysql/sippeers.sql (removed), /, + contrib/realtime/mysql/iaxfriends.sql (removed), + contrib/realtime/mysql/meetme.sql (removed): contrib/realtime: + Remove empty SQL script files Since the relatime scripts are now + managed by Alembic, the previous realtime scripts were previously + removed. However, the removal process messed up, as the files + were still in the repository. The contents were just empty. This + removes the files from the tree. ........ Merged revisions 411442 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/sip/include/sip.h: chan_sip: Add MESSAGE request to + allowed methods The allowed methods advertised by chan_sip did + not previously note the MESSAGE request. Even in Asterisk 1.8, we + do accept in-dialog MESSAGE requests; we should advertise that we + support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504 + #comment Reported by: Martin Kontsek ASTERISK-23504 #comment + Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587) + Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged + revisions 411372 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411373 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411374 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-27 19:21 +0000 [r411312-411328] Corey Farrell <git@cfware.com> + + * funcs/func_global.c, apps/app_speech_utils.c, + apps/confbridge/conf_config_parser.c, + funcs/func_callcompletion.c, funcs/func_frame_trace.c, + funcs/func_callerid.c, main/message.c, /, res/res_mutestream.c, + channels/pjsip/dialplan_functions.c, + res/res_pjsip_header_funcs.c, funcs/func_pitchshift.c, + funcs/func_groupcount.c, funcs/func_volume.c, funcs/func_odbc.c, + funcs/func_channel.c, funcs/func_cdr.c, funcs/func_blacklist.c, + apps/app_stack.c, apps/app_voicemail.c, res/res_calendar.c, + apps/app_jack.c, funcs/func_dialplan.c, funcs/func_speex.c, + channels/chan_sip.c, funcs/func_math.c, funcs/func_strings.c, + funcs/func_jitterbuffer.c, res/res_xmpp.c, channels/chan_iax2.c, + main/features_config.c, res/res_jabber.c: Fix dialplan function + NULL channel safety issues (closes issue ASTERISK-23391) Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3386/ ........ Merged + revisions 411313 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411314 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411315 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/format.c, include/asterisk.h, /: main/formats: Fix crash in + ast_format_cmp during non-clean shutdown. * Update asterisk.h to + reflect availability of ast_register_cleanup in 11.9. * Use + ast_register_cleanup for format_attr_shutdown. (closes issue + ASTERISK-23103) Reported by: JoshE ........ Merged revisions + 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 411311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-27 14:21 +0000 [r411296] Mark Michelson <mmichelson@digium.com> + + * main/sorcery.c, /: Give sorcery instances a reference to their + wizards. On graceful shutdown, sorcery wizards are all killed + off, but it is possible for sorcery instances to still have + dangling pointers after this, possibly causing a crash. Giving + the sorcery instances a reference to their wizards ensures that + the wizard reference will remain valid for the lifetime of the + sorcery instance. Review: https://reviewboard.asterisk.org/r/3401 + ........ Merged revisions 411295 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-26 22:45 +0000 [r411246] Joshua Colp <jcolp@digium.com> + + * /, main/say.c: say: Fix a bug where SayNumber in Polish tries to + play incorrect sound. This change fixes a bug where calling + SayNumber with a number divisible by 100 using the Polish + language would cause the code to attempt to play a sound file + with an empty name. (closes issue ASTERISK-23509) Reported by: + zvision Review: https://reviewboard.asterisk.org/r/3378/ ........ + Merged revisions 411243 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411244 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411245 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-26 16:15 +0000 [r411194] Jonathan Rose <jrose@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send + real CallerID information with P-Assserted-Identity (RFC-3325) + Prior too this patch, the P-Asserted-Identity header would + include anonymous caller id information which seems to go against + the point of the P-Asserted-Identity header. Now the real caller + ID information will be included in this header. Also, no privacy + header would be included. This patch adds 'Privacy: id' to + outgoing SIP messages that include the P-Asserted-Identity + header. (closes issue AST-1301) ........ Merged revisions 411189 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 411190 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411193 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-26 16:05 +0000 [r411192] Richard Mudgett <rmudgett@digium.com> + + * /, + contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py: + Fix 'alembic branches' merge conflict as described by the web + page. ........ Merged revisions 411191 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 18:44 +0000 [r411174] Sean Bright <sean@malleable.com> + + * /, res/ari/config.c: ARI: Don't complain about missing ARI users + when we aren't enabled Currently, if ARI is not enabled it will + still complain that there are no configured users. This patch + checks to see if ARI is enabled before logging and error or + iterating the container to validate the users. Review: + https://reviewboard.asterisk.org/r/3391/ ........ Merged + revisions 411173 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 17:40 +0000 [r411158] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip/pjsip_configuration.c, UPGRADE.txt, + res/res_pjsip_messaging.c, res/res_pjsip.c, + include/asterisk/res_pjsip.h: Add a "message_context" option for + PJSIP endpoints. ........ Merged revisions 411157 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 16:57 +0000 [r411142] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + include/asterisk/res_pjsip.h, /: res_pjsip: Fix contact + authenticate_qualify endpoint lookup when qualifing a contact. * + Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of + find_endpoints() with find_an_endpoint() since only the first + found endpoint is ever needed. * Fixed qualify_contact_cb() to + update the contact with the aor authenticate_qualify setting. + Otherwise, permanent contacts in the aor type sections would have + a config line order dependancy. * Fixed off nominal path contact + ref leak in qualify_contact(). The comment saying the unref is + not needed was wrong. * Fixed off nominal path use of the + endpoint parameter if it is NULL in send_out_of_dialog_request(). + * Added missing off nominal path unref of pjsip tdata in + send_out_of_dialog_request(). * Fixed off nominal path failing to + call the callback in send_request_cb() when the request is + challenged for authentication. * Eliminated silly RAII_VAR() use + in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen + to better reflect reality. (closes issue ASTERISK-23254) Reported + by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/ + ........ Merged revisions 411141 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 16:06 +0000 [r411092] Kinsey Moore <kmoore@digium.com> + + * /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If + update_provisional_keepalive() is called while + send_provisional_keepalive_full() is waiting on the PVT lock, + then pvt->provisional_keepalive_sched_id will be changed to a new + sched_id value by update_provisional_keepalive(), but that new + sched_id then may be overwritten with -1 by + send_provisional_keepalive_full(), killing the pvt's reference to + a schedule and "leaking" the reference. (closes issue + ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ + Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies + Patches: provisional_keepalive_fix.diff uploaded by Steve Davies + (license 5012) ........ Merged revisions 411088 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411089 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411091 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 15:56 +0000 [r411090] Jonathan Rose <jrose@digium.com> + + * /, res/res_stasis.c: ARI: Resolve a subscription leak against + implicit bridge subscriptions When a channel in a stasis + application is joined to a bridge, a subscription for that bridge + is created implicitly for the stasis application serving the + channel. Prior to this patch, subsequent removals of the channel + from the bridge would leave the subscription open. Review: + https://reviewboard.asterisk.org/r/3380/ ........ Merged + revisions 411086 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 15:47 +0000 [r411073-411087] Richard Mudgett <rmudgett@digium.com> + + * utils/conf2ael.c, main/lock.c, utils/ael_main.c: Revert -r411073. + It didn't help and blew up the system. + + * utils/ael_main.c, utils/conf2ael.c, main/lock.c: locking: Add + temporary sanity checks. Add some temporary sanity checks to hunt + for locking problems with the masquerade supertest. + +2014-03-24 21:39 +0000 [r411024] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: chan_sip: Always use fromdomain if set + for domain, even if callerid is set to restricted. (closes issue + ASTERISK-20841) Reported by: Kelly Goedert ........ Merged + revisions 411021 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411022 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411023 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-21 16:04 +0000 [r410996] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_pjsip_registrar.c: res_pjsip_registrar.c: + Miscellaneous cleanup in rx_task(). * Fix variable shadowing of + 'updated' by renaming it to 'contact_update'. * Checked + 'contact_update' for ast_sorcery_copy() failure. * Removed silly + use of RAII_VAR() for 'contact_update'. ........ Merged revisions + 410995 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-21 15:50 +0000 [r410981-410994] Sean Bright <sean@malleable.com> + + * res/ael/ael.flex, utils/Makefile, pbx/pbx_ael.c, + res/ael/ael_lex.c: Make the AEL load process less chatty. + Switched a bunch of LOG_NOTICEs to ast_debug. This time without + breaking the build. + + * pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Revert + r410981. aelparse blew up. + + * main/config.c: Remove a LOG_NOTICE from + ast_config_engine_register. There is enough indication from the + CLI that we are loading a realtime engine as it is. + + * pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Make the AEL + load process less chatty. Switched a bunch of LOG_NOTICEs to + ast_debug. + +2014-03-20 23:02 +0000 [r410967] Jonathan Rose <jrose@digium.com> + + * apps/app_confbridge.c, /: app_confbridge: Fix bug - users with + startmuted set don't start muted (closes issue ASTERISK-23461) + Reported by: Chico Manobela Review: + https://reviewboard.asterisk.org/r/3373/ ........ Merged + revisions 410965 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410966 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-20 16:35 +0000 [r410950] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/rtp_engine.h, main/dial.c, main/manager.c, /, + main/channel_internal_api.c, main/core_unreal.c, + include/asterisk/channel.h, res/ari/resource_channels.c, + res/res_stasis_snoop.c: assigned-uniqueids: Miscellaneous cleanup + and fixes. * Fix memory leak in ast_unreal_new_channels(). Made + it generate the ;2 uniqueid on a stack variable instead of + mallocing it. * Made send error response to ARI and AMI requests + instead of just logging excessive uniqueid length and allowing + truncation. action_originate() and + ari_channels_handle_originate_with_id(). * Fixed minor truncating + uniqueid hole when generating the ;2 uniqueid string length. + Created public and internal lengths of uniqueid. The internal + length can handle a max public uniqueid plus an appended ;2. * + free() and ast_free() are NULL tolerant so they don't need a NULL + test before calling. * Made use better struct initialization + format instead of the position dependent initialization format. + Also anything not explicitly initialized in the struct is + initialized to zero by the compiler. * Made + ast_channel_internal_set_fake_ids() use the safer + ast_copy_string() instead of strncpy(). Review: + https://reviewboard.asterisk.org/r/3371/ ........ Merged + revisions 410949 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-19 17:27 +0000 [r410934] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_endpoint_identifier_ip.c: PJSIP: Allow for + identify sections to be specified in sorcery.conf. "identify" is + a special type of configuration object in PJSIP because unlike + the other objects, it is not provided by the base res_pjsip + module. Instead, it is provided by the + res_pjsip_endpoint_identifier_ip module. If using the default + sorcery wizard (config,criteria=type=identify) then things work + because the module that applies the default wizard is the correct + module. However, if attempting to use sorcery.conf to apply an + alternate wizard, it was not possible. If you attempted to + specify the identify object type in the res_pjsip section, then + the object could not be registered since the object was + undocumented for the res_pjsip module. There was no alternate + configuration section defined for it, so you were out of luck if + you wanted to override the default wizard. With this change, the + identify section will properly have a sorcery.conf-based wizard + applied when the identify definition is within the + res_pjsip_endpoint_identifier_ip section. ........ Merged + revisions 410933 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-19 14:25 +0000 [r410905-410919] Joshua Colp <jcolp@digium.com> + + * res/res_stasis.c, /: res_stasis: Fix a bug where the default + bridge type was not set. ........ Merged revisions 410918 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * CHANGES, res/res_stasis.c, rest-api/api-docs/bridges.json, /, + res/ari/resource_bridges.h: res_stasis: Extend bridge type to be + a comma separated list of bridge attributes. This change turns + the bridge type field into a comma separated list of attributes. + These attributes include: mixing, holding, dtmf_events, and + proxy_media. By setting the various attributes a user can control + the type of bridge created with the behavior they need for their + application. (closes issue ASTERISK-23437) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........ + Merged revisions 410904 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-19 02:33 +0000 [r410891] Matthew Jordan <mjordan@digium.com> + + * res/res_ari.c, /: res_ari: Fix documentation schema error + ........ Merged revisions 410890 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 23:32 +0000 [r410877] Rusty Newton <rnewton@digium.com> + + * res/res_ari.c, /: res_ari: Add notes about Asterisk HTTP server + to the "enabled" config option for the res_ari general section + Added note and see-also reminding user to enable the HTTP server. + (closes issue ASTERISK-22499) Reported by: Rusty Newton ........ + Merged revisions 410876 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 15:45 +0000 [r410863] Scott Griepentrog <sgriepentrog@digium.com> + + * /, main/http.c: ARI: allow json content type with zero length + body When a request was received with a Content-type of json, the + body was sent for json parsing - even if it was zero length. This + resulted in ARI requests failing that were valid, such as a + channel DELETE with no parameters. The code has now been changed + to skip json parsing with zero content length. (closes issue + SWP-6748) Reported by: Samuel Galarneau Review: + https://reviewboard.asterisk.org/r/3360/ ........ Merged + revisions 410858 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 15:28 +0000 [r410862] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /: cdr: Add asserts for when we don't know about a + CDR for a channel In the CDR core, every channel should either be + filtered out (due to being an 'internal' channel used as an + implementation detail, such as playing media back into a bridge) + or it should get a CDR. Even if that CDR ends up being discarded, + we still give the channel a CDR in case we end up needing it. If + we hit a situation where a channel does not have a CDR, we should + blow up in -dev-mode. Asserts are appropriate for that. This + patch adds those asserts, as they would have quickly caught the + error fixed by r410814. ........ Merged revisions 410861 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 12:45 +0000 [r410845] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/config_system.c: res_pjsip: Fix memory leak of + nameservers in off-nominal resolver creation failure. Thanks + Walter Doekes! ........ Merged revisions 410844 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 11:52 +0000 [r410831] Sean Bright <sean@malleable.com> + + * res/res_fax_spandsp.c, /: res_fax_spandsp: Use g711_free() when + available. Per Johann Steinwendtner on the asterisk-dev mailing + list: + http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html + g711_free() was introduced in spandsp 0.0.6pre4 and + g711_release() became a noop. I opted not to remove the call to + g711_release() since it is harmless and to call g711_free() if we + have a sufficiently recent version of spandsp. (issue + ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged + revisions 410829 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410830 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 02:09 +0000 [r410814] Richard Mudgett <rmudgett@digium.com> + + * main/stasis_cache.c, /: stasis_cache: Use the right variable in + the cache entry ao2 cmp function. ........ Merged revisions + 410813 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-17 22:54 +0000 [r410794-410796] Joshua Colp <jcolp@digium.com> + + * include/asterisk/dns.h, CHANGES, + res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c, + main/dns.c, /, res/res_pjsip/config_system.c: res_pjsip: Enable + PJSIP DNS client support. This change enables DNS client support + within PJSIP. System nameservers are automatically discovered + using res_init or res_ninit. If this fails then PJSIP will resort + to using gethostbyname for resolution. By enabling this support + we gain SRV support, failover, and weight support. (closes issue + ASTERISK-23435) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3343/ ........ Merged + revisions 410795 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Make address + replacement less aggressive. This change makes the + res_pjsip_multihomed module less aggressive when changing the + address in messages. It will now only occur if the transport in + use is bound to the any address OR if the system determined + source address matches the bound address of the transport in use. + Review: https://reviewboard.asterisk.org/r/3369/ ........ Merged + revisions 410793 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-17 22:24 +0000 [r410775] Russ Meyerriecks <rmeyerreicks@digium.com> + + * /, main/callerid.c: callerid: Logic error in checksum processing + Callerid checksum-ing was being handled incorrectly here. When + the checksum is calculated to be 0x00, it will perform 0x100-0x00 + which results in 0x100. This value will then fail the otherwise + correct callerid message. This patch changes the logic to simply + add the calculated checksum to the transmitted 2's compliment + checksum. Review: https://reviewboard.asterisk.org/r/3356/ + (closes issue ASTERISK-23488) ........ This is a merge of merged + revisions 410750 410747 from + http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a + broken patch to be comitted to trunk so I pre-merge merged them. + +2014-03-17 19:35 +0000 [r410684-410699] Mark Michelson <mmichelson@digium.com> + + * res/res_mwi_external.c, res/res_pjsip/config_system.c, + configs/sorcery.conf.sample, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c, + tests/test_sorcery.c, tests/test_sorcery_realtime.c, + main/sorcery.c, /: Revert changes to sorcery that accidentally + got committed. These changes were still up for review and have + not been approved yet. I must have had the changes in my working + copy when making a different change. ........ Merged revisions + 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * bridges/bridge_softmix.c, tests/test_sorcery.c, main/channel.c, + res/res_pjsip/config_system.c, res/res_mwi_external.c, + include/asterisk/bridge_channel.h, funcs/func_frame_trace.c, + configs/sorcery.conf.sample, res/res_pjsip/pjsip_configuration.c, + include/asterisk/sorcery.h, tests/test_sorcery_astdb.c, + include/asterisk/frame.h, main/bridge_channel.c, + tests/test_sorcery_realtime.c, main/sorcery.c, + res/res_stasis_playback.c, main/frame.c, /: Fix stuck channel in + ARI through the introduction of synchronous bridge actions. + Playing back a file to a channel in an ARI bridge would attempt + to wait until the playback concluded before returning. The method + used involved signaling the waiting thread in the ARI custom + playback function. The problem with this is that there were some + corner cases that were not accounted for: * If a bridge channel + could not be found, then we never would attempt the playback but + would still attempt to wait for the playback to complete. * If + the bridge playfile action failed to queue, we would still + attempt to wait for the playback to complete. * If the bridge + playfile action were queued but some circumstance caused the + playback not to occur (the bridge dies, the channel is removed + from the bridge), then we would never be notified. The solution + to this is to move the waiting logic into the bridge code. A new + bridge API function is added to queue a synchronous action on a + bridge. The waiting thread is notified when the queued frame has + been freed, either due to an error occurring or due to successful + playback. As a failsafe, the waiting thread has a 10 minute + timeout just in case there is a frame leak somewhere. Review: + https://reviewboard.asterisk.org/r/3338 ........ Merged revisions + 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-17 16:48 +0000 [r410672] Richard Mudgett <rmudgett@digium.com> + + * /, apps/confbridge/conf_chan_announce.c: app_confbridge: Add + missing destructor call to announcer channel destructor. ........ + Merged revisions 410671 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-16 20:27 +0000 [r410651] Matthew Jordan <mjordan@digium.com> + + * /, res/stasis/app.c: stasis/app.c: Add some extra debugging for + subscription counts Events are sent to a connected ARI + application based on the things that ARI application cares about. + These subscriptions can be set up implicitly - such as when that + ARI application creates a new object - or explicitly, via the + application resource's subscription operations. Debugging *why* + something was being sent to an application - or why something was + not being sent to an application - was a bit tricky, as there was + no debug information for the subscriptions. This patch adds some + debug level 3 statements that show the subscription counts for + applications. (Level 3 was chosen as it matches the verbose level + 3 statements elsewhere) ........ Merged revisions 410650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-15 15:24 +0000 [r410639] Russell Bryant <russell@russellbryant.com> + + * include/asterisk/framehook.h: framehook.h: Fix some doc typos. + There were a number of instances in this header file where + "function all" was intended to be "function call". This patch + fixes that up. + +2014-03-14 21:56 +0000 [r410626] Mark Michelson <mmichelson@digium.com> + + * /, tests/test_sorcery_realtime.c: Fix failing realtime sorcery + tests. The store realtime callback needs to return a positive + value for sorcery to treat the store as a success. ........ + Merged revisions 410625 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 21:36 +0000 [r410624] Jonathan Rose <jrose@digium.com> + + * main/manager.c, /: manager: fix memory leak in manager_add_filter + function (closes issue ASTERISK-23420) Reported by: Etienne + Lessard Patches: manager_eventfilter_leak uploaded by Etienne + Lessard (license 6394) ........ Merged revisions 410609 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410623 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 20:55 +0000 [r410591-410608] Mark Michelson <mmichelson@digium.com> + + * /, main/db.c: Remove an extra ast_cond_wait() that slipped + through the patch. ........ Merged revisions 410606 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410607 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/config.c, res/res_sorcery_realtime.c: Handle the return + values of realtime updates and stores more accurately. Realtime + backends' update and store callbacks return the number of rows + affected, or -1 if there was a failure. There were a couple of + issues: * The config API was treating 0 as a successful return, + and positive values as a failure. Now the config API treats + anything >= 0 as a success. * res_sorcery_realtime was treating 0 + as a successful return from the store procedure, and any positive + values as a failure. Now sorcery treats anything > 0 as a + success. It still considers 0 a "failure" since there is no + change to report to observers. Review: + https://reviewboard.asterisk.org/r/3341 ........ Merged revisions + 410592 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c: Prevent conflicts regarding unsolicited + and solicited MWI to an endpoint. If an endpoint is receiving + unsolicited MWI for a mailbox and then attempts to subscribe to + an AOR that provides MWI for the same mailbox, then the SUBSCRIBE + is rejected with a 500 response. Review: + https://reviewboard.asterisk.org/r/3345 ........ Merged revisions + 410590 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 17:56 +0000 [r410589] Scott Griepentrog <sgriepentrog@digium.com> + + * /, CHANGES: uniqueid: Update CHANGES to reflect new features Note + the new features provided by uniqueid in the CHANGES file. (issue + ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/ + ........ Merged revisions 410588 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:42 +0000 [r410575] Jonathan Rose <jrose@digium.com> + + * /, main/acl.c, res/res_pjsip/pjsip_configuration.c, + contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py, + CHANGES, res/res_pjsip/config_transport.c, + include/asterisk/acl.h: PJSIP: TOS values should be represented + as decimals in sorcery objects (closes issue ASTERISK-23235) + Reported by: George Joseph Review: + https://reviewboard.asterisk.org/r/3324/ ........ Merged + revisions 410574 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:19 +0000 [r410567] Mark Michelson <mmichelson@digium.com> + + * /, main/db.c: Prevent delayed astdb syncs. The syncing thread + sleeps for a second before waiting to be told to attempt to sync + again. If a signal were sent during this sleeping period, we + would end up having to wait until the next sync signal occurred + in order to sync up the astdb. This code rearrangement also + ensures that any pending transactions will be synced prior to + Asterisk shutting down. Patches: db_sync.patch by John Hardin + (License #6512) ........ Merged revisions 410556 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410559 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:17 +0000 [r410560] Jonathan Rose <jrose@digium.com> + + * res/ari/resource_bridges.c, /: ARI/bridges: Forward + Playback/Recording Started/Finished to bridge topic (closes issue + ASTERISK-23444) Reported by: Ben Merrills Review: + https://reviewboard.asterisk.org/r/3340/ ........ Merged + revisions 410558 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:01 +0000 [r410542-410557] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/app.h, /, res/res_mwi_external.c, main/app.c: + res_mwi_external: Clear the stasis cache entry when the external + MWI is deleted. One of the things missing when external MWI + support was added was the ability to clear the stasis cache entry + of deleted external MWI mailboxes. Review: + https://reviewboard.asterisk.org/r/3325/ ........ Merged + revisions 410555 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/cdr.c: cdr.c: Add missing aow_unlock(cdr) in off nominal + path of handle_dial_message(). * Trivial common code hoisting in + handle_bridge_leave_message(). * Some whitespace fixing. ........ + Merged revisions 410541 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-13 19:33 +0000 [r410528] Kinsey Moore <kmoore@digium.com> + + * res/stasis/control.h, res/res_stasis.c, /, res/stasis/control.c: + ARI: Ensure managing application receives ChannelEnteredBridge + messages This fixes an issue where a Stasis application running + over ARI and subscribed to ari/events could miss the + ChannelEnteredBridge event because it did not subscribe to the + new bridge fast enough. To accomplish this, it subscribes the + application controlling the channel to the new bridge before + adding it to that bridge which required the stasis_app_control + structure to maintain a reference to the stasis_app. (closes + issue ASTERISK-23295) Review: + https://reviewboard.asterisk.org/r/3336/ ........ Merged + revisions 410527 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-13 13:25 +0000 [r410511] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_multihomed.c, /: Multiple revisions 410509-410510 + ........ r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar + 2014) | 2 lines res_pjsip_multihomed: Fix a bug where the 200 OK + for a REGISTER would contain the wrong contact. ........ r410510 + | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines + res_pjsip_multihomed: Remove change for testing fix. ........ + Merged revisions 410509-410510 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-12 19:06 +0000 [r410492-410494] Richard Mudgett <rmudgett@digium.com> + + * res/res_musiconhold.c, main/channel.c, /: res_musiconhold.c: + Generate MOH start/stop events whenever the MOH stream is + started/stopped. * Made res_musiconhold.c always post the + MusicOnHoldStart/MusicOnHoldStop events when it actually + starts/stops the music streams. This allows the events to always + happen when MOH starts/stops. The event posting code was moved to + the MOH alloc/release routines. * Made channel_do_masquerade() + stop any MOH on the original channel before masquerading so the + original channel will get a stop event with correct information. + * Cleaned up a couple odd codings in moh_files_alloc() and + moh_alloc() dealing with the music state variable. (issue + ASTERISK-23311) Reported by: Benjamin Keith Ford Review: + https://reviewboard.asterisk.org/r/3306/ ........ Merged + revisions 410493 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_state.c, + apps/confbridge/conf_state_single.c, + apps/confbridge/conf_state_inactive.c, + apps/confbridge/conf_state_single_marked.c, /: app_confbridge: + Make explicitly stop MOH if a user is kicked or hangs up while + MOH is playing. When MOH is playing to a user in a conference and + the user is kicked or hangs up from the conference then the AMI + MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event: + MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported + by: Benjamin Keith Ford Review: + https://reviewboard.asterisk.org/r/3306/ ........ Merged + revisions 410490 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410491 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-12 12:51 +0000 [r410452-410472] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Fix a bug + where outgoing messages for TCP would go out using UDP. This + change fixes a bug where the code which changes the transport did + not check whether the message is going out over UDP or not before + changing it. For TCP and TLS transports we don't need to change + the transport as the correct one is already chosen. ........ + Merged revisions 410471 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_multihomed.c (added), /: res_pjsip_multihomed: Add + module which places the correct address within messages. Due to + how messages are handled within PJSIP it is not until a message + is actually sent that the destination is reliably known. This + means that the addresses placed within the message may not be of + the interface the message is being sent out on. This module + determines what interface a message is being sent on and updates + the message to contain the correct address if applicable. This + module was tested by myself in a virtualized environment with + multiple interfaces and also by Kinsey Moore in the following + configuration: Networks: * 10.24.16.0/21 ** hard phone ** default + gateway * 10.24.64.0/21 ** softphone with pjsip-based stack + Transport details: bind address: 0.0.0.0 protocol: UDP All + endpoints were tested with explicitly configured transports and + unconfigured transports. This was tested with inbound and + outbound calls, both of which were experiencing detrimental + effects from incorrect IP addresses in SIP messages. These + effects were only experienced by the soft phone on the 10.24.64.0 + network since the messages to the hard phone on the 10.24.16.0 + network had the correct IP address. (closes issue ASTERISK-23020) + Reported by: xrobau Review: + https://reviewboard.asterisk.org/r/3102/ ........ Merged + revisions 410451 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 17:21 +0000 [r410395] Richard Mudgett <rmudgett@digium.com> + + * /, main/http.c: AST-2014-001: Stack overflow in HTTP processing + of Cookie headers. Sending a HTTP request that is handled by + Asterisk with a large number of Cookie headers could overflow the + stack. Another vulnerability along similar lines is any HTTP + request with a ridiculous number of headers in the request could + exhaust system memory. (closes issue ASTERISK-23340) Reported by: + Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. + Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions + 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 410381 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 16:33 +0000 [r410369] Scott Griepentrog <sgriepentrog@digium.com> + + * res/ari/resource_channels.c, main/manager.c, /: unqiueid: correct + max uniqueid length test This patch adds null string test prior + to checking for a max uniqueid value that was added in r410157. + ........ Merged revisions 410368 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 13:30 +0000 [r410346] Kinsey Moore <kmoore@digium.com> + + * /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad + session timers request This change allows chan_sip to avoid + creation of the channel and consumption of associated file + descriptors altogether if the inbound request is going to be + rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey + Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey + Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by + Corey Farrell (license 5909) ........ Merged revisions 410308 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 410311 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410329 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 12:53 +0000 [r410307] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c: AST-2014-003: + res_pjsip: When handling 401/407 responses don't assume a request + will have an endpoint. This change removes the assumption that an + outgoing request will always have an endpoint and makes the + authenticate_qualify option work once again. (closes issue + ASTERISK-23210) Reported by: Joshua Colp ........ Merged + revisions 410306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-08 16:50 +0000 [r410288] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip_endpoint_identifier_ip.c, + include/asterisk/res_pjsip_cli.h, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_cli.c, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/config_transport.c, main/sorcery.c, + include/asterisk/res_pjsip.h: pjsip_cli: Create pjsip show + channel and contact, and general cli code cleanup. Created the + 'pjsip show channel' and 'pjsip show contact' commands. + Refactored out the hated ast_hashtab. Replaced with + ao2_container. Cleaned up function naming. Internal only, no + public name changes. Cleaned up whitespace and brace formatting + in cli code. Changed some NULL checking from "if"s to + ast_asserts. Fixed some register/unregister ordering to reduce + deadlock potential. Fixed ast_sip_location_add_contact where the + 'name' buffer was too short. Fixed some self-assignment issues in + res_pjsip_outbound_registration. (closes issue ASTERISK-23276) + Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged + revisions 410287 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-08 15:45 +0000 [r410275] Matthew Jordan <mjordan@digium.com> + + * /, res/ari/resource_channels.c: resource_channels: Check if a + passed in ID is NULL before checking its length Calling strlen on + a NULL string is explosive. This patch checks whether or not the + passed in string is NULL or zero length before checking to see if + the string is too long. ........ Merged revisions 410274 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 22:56 +0000 [r410227] Corey Farrell <git@cfware.com> + + * /, channels/chan_sip.c: chan_sip: Fix deadlock of monlock between + unload_module and do_monitor Release monlock before calling + pthread_join. This ensures do_monitor cannot freeze by locking + monlock during module unload. (closes issue ASTERISK-21406) + Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3284/ ........ Merged + revisions 410224 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 410225 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 22:08 +0000 [r410212] Scott Griepentrog <sgriepentrog@digium.com> + + * /, include/asterisk/sorcery.h: sorcery: correct field register + argument list This fixes mistakes I previously made in merging + gtjoseph's changes with mine. ........ Merged revisions 410211 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 21:54 +0000 [r410208-410210] Matthew Jordan <mjordan@digium.com> + + * /, main/config_options.c: config_options: Display the see-also + information for CLI config option help The config option help + information has always parsed the <see-also> tags in the XML + documentation. Unfortunately, it just never bothered displaying + them on the CLI. With this patch, when you execute 'config show + help [module] [obj] [option]', it will display what other options + are useful to you. (closes issue ASTERISK-22008) Reported by: + Richard Mudgett ........ Merged revisions 410209 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip.c, /: res_pjsip: Fix documentation for one touch + recording see-also links The one touch recording options have + several see-also links between the various configuration options. + These were 'broken' by the snake casing of those options. This + patch corrects the see-also links such that they reference the + correct option names. ........ Merged revisions 410194 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 21:23 +0000 [r410207] Mark Michelson <mmichelson@digium.com> + + * main/sorcery.c, res/res_sorcery_realtime.c, /, + include/asterisk/sorcery.h, tests/test_sorcery_realtime.c: Make + res_sorcery_realtime filter unknown retrieved results. When + retrieving data from a database or other realtime backend, it's + quite possible to retrieve variables that Asterisk does not care + about but that are legitimate to exist. Asterisk does not need to + throw a hissy fit when these variables are encountered but rather + just filter them out. Review: + https://reviewboard.asterisk.org/r/3305 ........ Merged revisions + 410187 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 21:11 +0000 [r410191] Scott Griepentrog <sgriepentrog@digium.com> + + * main/sorcery.c, /, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c: pjsip: allow and disallow + show same codecs In order to prevent confusion over the allow and + disallow list of codecs being the same an option for registering + a field as an alias is added. The alias field will be read from + the configuration file, but afterwards is not listed as a known + field. With disallow set as an alias, the CLI command pjsip show + endpoint # will list the allow= field, but not the disallow + field. (closes issue ASTERISK-23092) Review: + https://reviewboard.asterisk.org/r/3193/ ........ Merged + revisions 410190 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 20:41 +0000 [r410174-410185] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/devicestate.h, main/stasis_cache.c, + main/stasis_message.c, /, tests/test_devicestate.c, + include/asterisk/stasis.h, main/app.c, main/devicestate.c, + tests/test_stasis.c: stasis cache: Enhance to keep track of an + item from different entities. A stasis cache entry now contains + more than a single message/snapshot. It contains + messages/snapshots for the local entity as well as any remote + entities that post to the cached item. In addition callbacks can + be supplied when the cache is created to compute and post the + aggregate message/snapshot representing all entities stored in + the cache entry. * All stasis messages now have an eid to + indicate what entity posted it. * The stasis cache enhancements + allow device state to cache and aggregate the device states from + local and remote entities in a single operation. The cached + aggregate device state is available immediately after it is + posted to the stasis bus. This improves performance by + eliminating a cache dump and associated ao2 container traversals + to calculate the aggregate state. (closes issue ASTERISK-23204) + Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3281/ ........ Merged + revisions 410184 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c, channels/sig_pri.c, channels/sig_ss7.c, + include/asterisk/bridge.h, tests/test_cdr.c, channels/sig_pri.h, + channels/chan_dahdi.c, channels/sig_ss7.h, /: uniqueid: Fix + chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler + errors. (issue ASTERISK-23120) ........ Merged revisions 410171 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 15:47 +0000 [r410158] Scott Griepentrog <sgriepentrog@digium.com> + + * tests/test_cdr.c, res/res_clioriginate.c, res/res_ari_bridges.c, + tests/test_substitution.c, res/res_stasis_playback.c, + channels/chan_multicast_rtp.c, apps/app_meetme.c, /, + main/bridge_basic.c, include/asterisk/channel_internal.h, + tests/test_app.c, apps/confbridge/conf_chan_record.c, + main/core_unreal.c, channels/chan_gtalk.c, + include/asterisk/stasis_app_playback.h, + res/ari/resource_bridges.c, channels/chan_jingle.c, + channels/chan_phone.c, pbx/pbx_spool.c, + res/ari/resource_bridges.h, res/parking/parking_tests.c, + channels/chan_motif.c, apps/app_confbridge.c, + res/ari/resource_channels.c, include/asterisk/pbx.h, + res/res_stasis.c, include/asterisk/bridge.h, + apps/app_voicemail.c, res/ari/resource_channels.h, + apps/app_dial.c, res/res_calendar_exchange.c, + channels/chan_vpb.cc, apps/app_page.c, apps/app_chanisavail.c, + include/asterisk/dial.h, main/core_local.c, + res/parking/parking_bridge_features.c, + tests/test_stasis_endpoints.c, res/parking/parking_bridge.c, + channels/chan_skinny.c, include/asterisk/stasis_app_snoop.h, + addons/chan_mobile.c, main/bridge_channel.c, + channels/chan_pjsip.c, channels/chan_mgcp.c, + channels/chan_unistim.c, main/pbx.c, + res/res_calendar_icalendar.c, main/ccss.c, + channels/chan_bridge_media.c, main/bridge.c, + tests/test_stasis_channels.c, apps/app_bridgewait.c, + apps/app_originate.c, res/res_calendar_caldav.c, + include/asterisk/channel.h, res/parking/parking_applications.c, + apps/app_followme.c, main/cel.c, apps/app_queue.c, + res/res_ari_channels.c, res/res_calendar_ews.c, + rest-api/api-docs/bridges.json, main/dial.c, + channels/chan_dahdi.c, channels/chan_h323.c, tests/test_cel.c, + rest-api/api-docs/channels.json, + include/asterisk/bridge_internal.h, + apps/confbridge/conf_chan_announce.c, res/res_calendar.c, + include/asterisk/core_unreal.h, addons/chan_ooh323.c, + res/stasis/control.c, channels/chan_sip.c, + main/channel_internal_api.c, include/asterisk/stasis_app.h, + res/res_stasis_snoop.c, channels/chan_console.c, + channels/chan_iax2.c, channels/chan_oss.c, apps/app_agent_pool.c, + main/channel.c, main/manager.c, channels/chan_misdn.c, + tests/test_voicemail_api.c, channels/chan_alsa.c, + channels/chan_nbs.c, main/message.c: uniqueid: channel linkedid, + ami, ari object creation with id's Much needed was a way to + assign id to objects on creation, and much change was necessary + to accomplish it. Channel uniqueids and linkedids are split into + separate string and creation time components without breaking + linkedid propgation. This allowed the uniqueid to be specified by + the user interface - and those values are now carried through to + channel creation, adding the assignedids value to every function + in the chain including the channel drivers. For local channels, + the second channel can be specified or left to default to a ;2 + suffix of first. In ARI, bridge, playback, and snoop objects can + also be created with a specified uniqueid. Along the way, the + args order to allocating channels was fixed in chan_mgcp and + chan_gtalk, and linkedid is no longer lost as masquerade occurs. + (closes issue ASTERISK-23120) Review: + https://reviewboard.asterisk.org/r/3191/ ........ Merged + revisions 410157 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 05:04 +0000 [r410108] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: chan_sip: Allow static realtime members + to be qualified during module load. When a static realtime peer + with qualify=yes is loaded, Asterisk will fail to send an OPTIONS + request due to the lastms being equal to 0. This results in the + peer being unable to receive calls from Asterisk because the + status is permanently UNKNOWN. This patch allows an OPTIONS + request to be sent during module load by ignoring the lastms + value on startup only. Review: + https://reviewboard.asterisk.org/r/3294/ (closes issue + ASTERISK-17523) Reported by: Maciej Krajewski Tested by: + wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor + Peirce (license 6112) ........ Merged revisions 410105 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 410106 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410107 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 23:47 +0000 [r410092] Richard Mudgett <rmudgett@digium.com> + + * main/sorcery.c, /: sorcery.c: Fix off-nominal path ref and memory + leak in ast_sorcery_objectset_json_create(). * Made exit a loop + early on error in ast_sorcery_objectset_json_create(). * Removed + some dead code in ast_sorcery_objectset_create2(). ........ + Merged revisions 410089 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 23:43 +0000 [r410091] Russell Bryant <russell@russellbryant.com> + + * /, res/res_musiconhold.c: moh: fix a refcount error with realtime + MOH I observed a crash in res_musiconhold on an Asterisk 11 + system using realtime MOH. Investigation of the backtrace showed + a corrupt mohclass, implying that it got destroyed before the + code expected it to. I went looking for reference counting errors + that could have caused this crash and this patch this result. It + contains 2 changes. 1) Remove a usless block of code that was + impossible to reach. There was even a comment indicating that it + was impossible to reach. The conditional includes + "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's + inside of an if block with the opposite check + "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no + good reason to keep it around. 2) A similar block to #1 contained + a reference counting error. It stores state->class in the local + variable mohclass without increasing its reference count. The + reference count on mohclass is decremented at the end of the + function. This block of code probably very rarely runs, which + would help explain why this system was working fine for many + months before experiencing a crash. Review: + https://reviewboard.asterisk.org/r/3282/ ........ Merged + revisions 410043 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 410044 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410090 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 22:39 +0000 [r410042] George Joseph <george.joseph@fairview5.com> + + * res/res_pjsip/config_auth.c, funcs/func_sorcery.c (added), + res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c, + main/bucket.c, res/res_pjsip_endpoint_identifier_ip.c, + include/asterisk/config.h, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c, res/res_pjsip_acl.c, + CHANGES, tests/test_sorcery.c, res/res_pjsip/config_transport.c, + main/config.c, main/sorcery.c: sorcery: Create AST_SORCERY + dialplan function. This patch creates the AST_SORCERY dialplan + function which allows someone to retrieve any value from a + sorcery-based config file. It's similar to AST_CONFIG. The + creation of the function itself was fairly straightforward but it + required changes to the underlying sorcery infrastructure that + rippled into individual sorcery objects. The changes stemmed from + inconsistencies in how sorcery created ast_variable objectsets + from sorcery objects and the inconsistency in how individual + objects used that feature especially when it came to parameters + that can be specified multiple times like contact in aor and + match in identify. You can read more here... + http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html + So, what this patch does, besides actually creating the + AST_SORCERY function, is the following... * Creates + ast_variable_list_append which is a helper to append one + ast_variable list to another. * Modifies the + ast_sorcery_object_field_register functions to accept the + already-defined sorcery_fields_handler callback. * Modifies + ast_sorcery_objectset_create to accept a parameter indicating + return type preference...a single ast_variable with all values + concatenated or an ast_variable list with multiple entries. Also + fixed a few bugs. * Modifies individual sorcery object + implementations to use the new function definition of the + ast_sorcery_object_field_register functions. * Modifies + location.c and res_pjsip_endpoint_identifier_ip.c to implement + sorcery_fields_handler handlers so they return multiple + occurrences as an ast_variable_list. * Added a whole bunch of + tests to test_sorcery. (closes issue ASTERISK-22537) Review: + http://reviewboard.asterisk.org/r/3254/ + +2014-03-06 19:04 +0000 [r410029] Jonathan Rose <jrose@digium.com> + + * include/asterisk/acl.h, /, main/acl.c, + res/res_pjsip/pjsip_configuration.c, UPGRADE.txt, + contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py + (added), res/res_pjsip/config_transport.c: pjsip configuration: + Make transport TOS values consistent with endpoints Transport TOS + values were interpreted as DSCP values without being documented + as such. Endpoint TOS values (tos_audio/tos_video) behaved + normally as TOS values have historically. This patch makes the + transport TOS values behave as TOS values and makes all TOS + values readable as string values (e.g. AF11). In addition, + alembic scripts have been updated to use the proper field types + for all TOS/COS values. (issue ASTERISK-23235) Reported by: + George Joseph Review: https://reviewboard.asterisk.org/r/3304/ + ........ Merged revisions 410028 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 18:20 +0000 [r410027] Joshua Colp <jcolp@digium.com> + + * res/ari/resource_channels.c, CHANGES, + res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, res/ari/resource_bridges.c, + res/ari/ari_model_validators.h, /, + include/asterisk/stasis_app_recording.h, + res/res_stasis_recording.c: res_stasis_recording: Add a + "target_uri" field to recording events. This change adds a + target_uri field to the live recording object. It contains the + URI of what is being recorded. (closes issue ASTERISK-23258) + Reported by: Ben Merrills Review: + https://reviewboard.asterisk.org/r/3299/ ........ Merged + revisions 410025 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 15:58 +0000 [r410012] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_mwi.c, /: Don't attempt to link in an aggregate MWI + subscription if an endpoint does not aggregate MWI. Attempting to + link a NULL object into an ao2 container had been benign + previously, but since enabling DO_CRASH in the testsuite, this is + now causing a crash. It's better to be right here anyway. + ........ Merged revisions 410011 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 02:22 +0000 [r409996] Matthew Jordan <mjordan@digium.com> + + * res/res_fax_spandsp.c, /: res_fax_spandsp: Fix crash when passing + ulaw/alaw data to spandsp When acting as a T.38 fax gateway, + res_fax_spandsp would at times cause a crash in libspandsp. This + would occur when, during fax tone detection, a ulaw/alaw frame + would be passed to modem_connect_tones_rx. That particular + routine expects the data to be in slin format. This patch looks + at the frame type and, if the data is ulaw/alaw, converts the + format to slin before passing it to modem_connect_tones_rx. + Review: https://reviewboard.asterisk.org/r/3296 (closes issue + ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal + Rybarik patches: spandsp_g711decode.diff uploaded by Michal + Rybarik (license 6578) ........ Merged revisions 409990 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409991 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 00:33 +0000 [r409970-409977] Richard Mudgett <rmudgett@digium.com> + + * apps/confbridge/conf_state_multi.c, + apps/confbridge/conf_state_inactive.c, /: app_confbridge: Remove + some noop code. ........ Merged revisions 409976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_musiconhold.c: res_musiconhold.c: Remove some + unnecessary RAII_VAR() usage. * Made the moh_register() define + use useful parameter names. ........ Merged revisions 409967 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 20:41 +0000 [r409904-409919] Kinsey Moore <kmoore@digium.com> + + * main/config.c, /: config: Fix inverted test The test of the + result of the stat() call was inverted such that its output was + only used if the call failed. This inverts the test so that the + output of stat() is used correctly. This was causing full reloads + on unchanged files. (closes issue ASTERISK-23383) Reported by: + David Woolley ........ Merged revisions 409916 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409917 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409918 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * bridges/bridge_native_rtp.c, /: bridge_native_rtp: Fix crash + involving masquerade It is possible for a channel to be + masqueraded out of a bridge which means it may no longer have RTP + glue to check upon leaving said bridge. If this situation + occurred (it's possible at least during dial and call pickup) + then Asterisk would crash. This change makes sure the glue is + checked before use. (closes issue AST-1290) Reported by: John + Bigelow ........ Merged revisions 409900 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 18:51 +0000 [r409889] Richard Mudgett <rmudgett@digium.com> + + * contrib/ast-db-manage/cdr/versions, + contrib/ast-db-manage/cdr/versions/210693f3123d_create_cdr_table.py, + /, + contrib/ast-db-manage/config/versions/28887f25a46f_create_queue_tables.py + (added), contrib/ast-db-manage/cdr.ini.sample (added), + contrib/ast-db-manage/cdr/env.py, contrib/ast-db-manage/cdr + (added), contrib/ast-db-manage/cdr/script.py.mako: alembic: Add + missing queue and CDR table creation scripts. * Added the queues + and queue_members tables to the config alembic scripts. * Added + the CDR table alembic creation script. The CDR table is more of + an example for new setups since the actual table can be fully + customized in cdr_adaptive_odbc.conf. (closes issue + ASTERISK-23233) Reported by: jmls Review: + https://reviewboard.asterisk.org/r/3227/ ........ Merged + revisions 409885 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 18:47 +0000 [r409888] Mark Michelson <mmichelson@digium.com> + + * funcs/func_presencestate.c, /: Fix documentation for + PRESENCE_STATE to properly illustrate how to create a presence + hint. There was a missing comma. This was discovered by Dan + Kaplan. ........ Merged revisions 409886 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409887 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 16:58 +0000 [r409836] David M. Lee <dlee@digium.com> + + * main/config.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac: Corrected cross-platform stat nanosecond code When + nanosecond time resolution was added for identifying config file + changes, it didn't cover all of the myriad of ways that one might + obtain nanosecond time resolution off of struct stat. Rather than + complicate the #if even further figuring out one system from the + next, this patch directly tests for the three struct members I + know about today, and #ifdef's accordingly. Review: + https://reviewboard.asterisk.org/r/3273/ ........ Merged + revisions 409833 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409834 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409835 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 16:26 +0000 [r409831-409832] Moises Silva <moises.silva@gmail.com> + + * res/res_http_websocket.c: Fix res/res_http_websocket.c build + failure in 32bit due to incorrect print format for uint64_t + + * res/res_http_websocket.c, /: Fix WebRTC over WSS not working + Several fixes for the WebSockets implementation in + res/res_http_websocket.c * Flush the websocket session FILE* as + fwrite() may not actually guarantee sending the data to the + network. If we do not flush, it seems that buffering on the SSL + socket for outbound messages causes issues * Refactored + ast_websocket_read to take into account that SSL file descriptors + may be ready to read via fread() but poll() will not actually say + so because the data was already read from the network buffers and + is now in the libc buffers (closes issue ASTERISK-23099) (closes + issue ASTERISK-21930) Review: + https://reviewboard.asterisk.org/r/3248/ ........ Merged + revisions 409681 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409697 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 12:06 +0000 [r409780] Sean Bright <sean@malleable.com> + + * contrib/scripts/astgenkey, contrib/scripts/astgenkey.8, /: Fix + references to 'keys' CLI commands in astgenkey ........ Merged + revisions 409777 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409778 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409779 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 06:17 +0000 [r409747] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * channels/chan_unistim.c: Add update_peer function to + unistim_rtp_glue, improve other unistim_rtp_glue functions + conforming to other channel drivers. Do not forget auto-detected + and user-selected phone settings on 'unistim reload' ........ + Merged revisions 409705 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409745 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-03-05 01:05 +0000 [r409683] Richard Mudgett <rmudgett@digium.com> + + * /, include/asterisk/stasis_internal.h: stasis: Made + internal_stasis_subscribe() prototype and definition match + exactly. ........ Merged revisions 409682 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 19:34 +0000 [r409627] Michael L. Young <elgueromexicano@gmail.com> + + * funcs/func_audiohookinherit.c, /: func_audiohookinheritance: + Check If A Channel Was Specified This patch prevents a crash when + using the function audiohookinheritance without setting the + channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal + Tested by: Joel Vandal Patches: + asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3272/ ........ Merged + revisions 409623 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409625 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409626 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 17:22 +0000 [r409587] Jonathan Rose <jrose@digium.com> + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix one way audio + problems with hold/unhold when using ICE ICE sessions will now be + restarted if sessions are changed to use new sets of remote + candidates. (closes issue ASTERISK-22911) Reported by: Vytis + Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ + ........ Merged revisions 409565 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409570 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 16:55 +0000 [r409569] Kinsey Moore <kmoore@digium.com> + + * /, main/astobj2.c: AO2: Add an assert for bad objects This adds + an assert that will only be active if Asterisk is compiled with + DO_CRASH and allows the testsuite to fail tests that would + otherwise require log file parsing. ........ Merged revisions + 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 409567 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409568 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 14:55 +0000 [r409475] Sean Bright <sean@malleable.com> + + * /, channels/chan_sip.c: Minor whitespace change to 'sip show + peers' output. (closes issue ASTERISK-23406) Reported by: ibercom + Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom + ........ Merged revisions 409472 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409473 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409474 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-03 19:44 +0000 [r409423] Joshua Colp <jcolp@digium.com> + + * /, res/res_stasis_recording.c: res_stasis_recording: Fix memory + leak of the absolute name. ........ Merged revisions 409422 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-03 02:08 +0000 [r409364] Matthew Jordan <mjordan@digium.com> + + * main/asterisk.c, /: doxygen: Tweak the link back to ye olde + Digium website ........ Merged revisions 409361 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409362 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409363 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-02 17:03 +0000 [r409350] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * /, Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a + legal option of gcc. Unofficially gcc considers it to be + equivalent of -O3. clang chalks on it, though. This commit sets + the default optimization flag to be -O3, like gcc actually + considered it. Review: https://reviewboard.asterisk.org/r/3280/ + ........ Merged revisions 409308 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409344 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409346 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-01 20:28 +0000 [r409288] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, /: res_pjsip_session: Set options + (100rel, timers) on incoming sessions. This change passes options + to the UAS creation function. This in turn sets up 100rel and + session timer properties on the incoming session. Reported by + Julian Russell on asterisk-users mailing list. ........ Merged + revisions 409287 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-01 00:05 +0000 [r409257-409275] Richard Mudgett <rmudgett@digium.com> + + * /, main/devicestate.c: devicestate.c: Simplified some logic in + _ast_device_state(). ........ Merged revisions 409274 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_cache.c, /: stasis_cache.c: Remove some unnecessary + RAII_VAR() usage. ........ Merged revisions 409272 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis.c, /: stasis.c: Misc code cleanups. * Remove some + unnecessary RAII_VAR() usage. * Made the struct + stasis_subscription ao2 object use the ao2 lock instead of a + redundant join_lock in the struct for ast_cond_wait(). * Removed + locks on some ao2 objects that don't need the lock. * Made the + topic pool entries container use the ao2 template functions. * + Add some missing allocation failure checks. * Add missing cleanup + in off nominal path of dispatch_message(). ........ Merged + revisions 409270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Add precautionary p->owner + checks. * Add precautionary p->owner checks in sip_hangup(), + get_refer_info(), get_also_info(), and + interpret_t38_parameters(). * Simplify some tangled logic in + get_refer_info(), get_also_info(), and add_rpid(). * Removed some + dead code in handle_request_invite(). (closes issue + ASTERISK-23323) Reported by: Walter Doekes Patches: + issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) + uploaded by wdoekes (modified) + issueA23323-more_p_owner_checks-11.x.patch (license #5674) + uploaded by wdoekes (modified) + issueA23323-more_p_owner_checks-12.x.patch (license #5674) + uploaded by wdoekes (modified) + issueA23323-more_p_owner_checks-trunk.patch (license #5674) + uploaded by wdoekes (modified) ........ Merged revisions 409207 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 409255 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409256 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-28 21:24 +0000 [r409237] Kinsey Moore <kmoore@digium.com> + + * apps/app_queue.c, /: app_queue: Fix documented AMI event name + During the rewrite of AMI events to use the Stasis bus, the name + of the QueueMemberPaused event was changed to QueueMemberPause. + This corrects documentation to reflect that. ........ Merged + revisions 409234 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-28 18:03 +0000 [r409159] Richard Mudgett <rmudgett@digium.com> + + * /, channels/chan_sip.c: chan_sip: Fix crash in + ast_channel_hangupcause_set(). * Fix crash in + ast_channel_hangupcause_set() because p->owner not checked before + calling. Regression introduced by the fix for ASTERISK-22621. + (closes issue ASTERISK-23135) Reported by: OK (issue + ASTERISK-23323) Reported by: Walter Doekes ........ Merged + revisions 409156 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409157 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409158 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 19:54 +0000 [r409132] Jonathan Rose <jrose@digium.com> + + * res/res_rtp_asterisk.c, /: Multiple revisions 409129-409130 + ........ r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb + 2014) | 15 lines res_rtp_asterisk: Fix checklist creating + problems in ICE sessions Prior to this patch, local candidate + lists including SRFLX would fail to start properly when building + ICE candidate check lists. This patch fixes that problem by + making sure that each SRFLX candidate is associated with the + proper base address so that the check list can create matches + properly. This patch was written by jcolp. The issue will be left + open to await testing by the issue participants. (issue + ASTERISK-23213) Reported by: Andrea Suisani Review: + https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose + | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines + res_rtp_asterisk: correct build error from r409129 Accidentally + placed a declaration below functional code (issue ASTERISK-23213) + Reported by: Andrea Suisani Review: + https://reviewboard.asterisk.org/r/3256/ ........ Merged + revisions 409129-409130 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409131 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 16:26 +0000 [r409091] David M. Lee <dlee@digium.com> + + * utils/astman.c, /: Fix memory stomping bug in astman. This memset + complained in dev mod on my Ubuntu box. The memset is both + unnecessary and dangerous. At this point, m hasn't been + initialized yet, so the memset will write off to whatever address + happens to be on the stack at the time. ........ Merged revisions + 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 409083 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409087 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 16:08 +0000 [r409055] Corey Farrell <git@cfware.com> + + * /, configs/res_fax.conf.sample: res_fax: Comment out default + settings from res_fax.conf. Comment out many settings in + res_fax.conf.sample. The defaults are set in res_fax.c, so + setting the same value in sample config does nothing but make the + sample config more fragile. (closes issue ASTERISK-23231) + Reported by: David Brillert Review: + https://reviewboard.asterisk.org/r/3261/ ........ Merged + revisions 409052 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409053 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409054 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 12:29 +0000 [r409000] Matthew Jordan <mjordan@digium.com> + + * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Apply + packetization rules on inbound SDP handling The setting + 'use_ptime' is supposed to tell Asterisk to honour the ptime + attribute in an offer, preferring it to whatever packetization + preferences have been set internally. Currently, however, + something rather quirky will happen: (1) The SDP answer will be + constructed in create_outgoing_sdp_stream. This will use the + preferences from the endpoint, such that the 200 OK response will + add the packetization preferences from the endpoint, and not what + was offered. (2) When the 200 response is issued, + apply_negotiated_sdp_stream is called. This will call + apply_packetization, which will use the ptime attribute from the + offer internally. We end up telling the offerer to use the + internal ptime attribute, but we end up using the offered ptime + attribute. Hilarity ensues. This patch modifies the behaviour by + calling apply_packetization from negotiate_incoming_sdp_stream, + which is called prior to create_outgoing_sdp_stream. This causes + the format preferences on the session's media object to be set to + the inbound ptime value (if 'use_ptime' is enabled), such that + the construction of the answer gets the right value immediately. + Review: https://reviewboard.asterisk.org/r/3244/ ........ Merged + revisions 408999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 23:35 +0000 [r408984] Richard Mudgett <rmudgett@digium.com> + + * /, tests/test_stasis.c: test_stasis.c: Misc cleanups. * Make the + consumer ao2 object use the ao2 lock instead of a redundant lock + in the struct for ast_cond_wait(). * Fixed some curly brace + placements. * Fixed use of malloc(0). malloc(0) has variant + behavior. It is up to the implementation to determine if it + returns NULL or a valid pointer that can be later passed to + free(). ........ Merged revisions 408983 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 19:00 +0000 [r408971] Scott Griepentrog <sgriepentrog@digium.com> + + * channels/chan_pjsip.c, /: pjsip: avoid edge case potential crash + in answer() When accidentally compiling against a wrong version + of pjsip headers with a different pjsip_inv_session size, the + invite_tsx structure could be null in the answer() function. This + led to a crash because it attempted to send the session response + with an uninitialized packet pointer. This patch presets packet + to null and adds a diagnostic log message to explain why the call + fails. Review: https://reviewboard.asterisk.org/r/3267/ ........ + Merged revisions 408970 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 17:04 +0000 [r408958] Joshua Colp <jcolp@digium.com> + + * res/res_ari.c, /: res_ari: Make some additional error responses + consistent with the rest of the system. This change makes some + error cases use ast_ari_response_error to construct their error + responses instead of manually doing it. This ensures they are + consistent with the other error responses. Based on the original + patch as done by Paul Belanger on the associated review. Review: + https://reviewboard.asterisk.org/r/2904/ ........ Merged + revisions 408957 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 13:47 +0000 [r408942-408944] Kinsey Moore <kmoore@digium.com> + + * include/asterisk/res_pjsip_session.h, /: PJSIP: Fix some bad + spacing ........ Merged revisions 408943 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_refer.c: PJSIP: Prevent crash if channel has + gone away It is currently possible for an ast_sip_session to + exist without an associated channel as is the case when a new + invite is coming in or just after a hangup is issued on a + chan_pjsip channel. Part of the attended transfer code assumed + the channel would be non-NULL and used it as such causing a + crash. This bug was exposed thanks to the attended transfer ARI + test in the test suite. (closes issue ASTERISK-23287) Reported + by: Matt Jordan ........ Merged revisions 408941 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 08:57 +0000 [r408932] Igor Goncharovskiy <igor.goncharovsky@gmail.com> + + * channels/chan_unistim.c: Implement functions handling keypress, + display icons and text for i2004 KEM support. + +2014-02-25 17:51 +0000 [r408881-408883] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_exten_state.c, /, + res/res_pjsip_pidf_digium_body_supplement.c (added), + include/asterisk/res_pjsip_body_generator_types.h: + res_pjsip_exten_state: Presence for digium phones Added presence + support for digium phones. Review: + https://reviewboard.asterisk.org/r/3239/ ........ Merged + revisions 408882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_send_to_voicemail.c (added), + res/res_pjsip_header_funcs.c: res_pjsip_send_to_voicemail: + transferring to voicemail for digium phones Added the ability for + transferring directly to voicemail on digium phones. Added a new + module that checks for the presence of a custom header and/or + diversion header within a sip REFER. If either is found and they + specify a sending to voicemail action then variables are added to + the channel allowing the user access to them in the dialplan. + Dialplan can then be written that branches based upon these + values allowing, for instace, for a single number to be used for + dialing and/or accessing voicemail directly. Also fixed a problem + where the PJSIP_HEADER function was allowing non pjsip channels + through (checked to make sure it has the correct channel type + before proceeding). Review: + https://reviewboard.asterisk.org/r/3245/ ........ Merged + revisions 408880 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-25 17:44 +0000 [r408879] Rusty Newton <rnewton@digium.com> + + * configs/voicemail.conf.sample, /: configs/voicemail.conf.sample - + Make mailcmd sample text more explicit Made the wording a bit + more explicit. Didn't really change the meaning. ........ Merged + revisions 408876 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408877 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408878 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-22 23:31 +0000 [r408859] Matthew Jordan <mjordan@digium.com> + + * /, main/asterisk.c: main: Initialize dialplan providing core + components prior to module pre-load It is possible to pre-load + pbx_config. As a result, pbx_config - which will load and parse + the dialplan - will attempt to use various dialplan components, + such as device state providers and presence state providers, + prior to them being initialized by the core. This would lead to a + crash, as the components had not created their Stasis cache + entries. This patch moves a number of core component + initializations before the module pre-load. This guarantees that + if someone does pre-load pbx_config - or other pbx modules - that + the Stasis caches for the various core components are created. + (closes issue ASTERISK-23320) Reported by: xrobau (closes issue + ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy, + Rusty Newton ........ Merged revisions 408855 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-22 18:01 +0000 [r408840] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c, /: ignore AST_CONTROL_PVT_CAUSE_CODE + without any messages (closes issue ASTERISK-23336) Reported by: + Alexander Semych ........ Merged revisions 408838 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408839 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-22 02:31 +0000 [r408788] Corey Farrell <git@cfware.com> + + * /, utils/extconf.c, utils/conf2ael.c, res/ael/pval.c, main/pbx.c: + Remove extra defines of AST_PBX_MAX_STACK. * Ensure + AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix + incorrect function parameters in utils/extconf.c. (closes issue + ASTERISK-23141) Reported by: Maxim Review: + https://reviewboard.asterisk.org/r/3241/ ........ Merged + revisions 408785 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408786 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408787 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 18:37 +0000 [r408731] Kevin Harwell <kharwell@digium.com> + + * main/rtp_engine.c, /: rtp_engine: Dynamic payload change in rtp + mapping not supported Asterisk didn't support the dynamic payload + change in rtp mapping in the 200 OK response. Scenario: Asterisk + sends the INVITE proposing alaw and telephone-event, it proposes + rtpmap:101 for telephone-event. Peer responds with 2xx, it + answers with alaw and telephone-event also, but it proposes a + different rtpmap number (rtpmap:103) for telephone-event. + Expected Behaviour: Asterisk should honour the rtpmapping in the + response and send DTMF packets using 103 as payload type for + DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload + type 101. With this patch asterisk now supports changes that can + occur in the rtp mapping in the response. (closes issue + ASTERISK-23279) Reported by: NITESH BANSAL Review: + https://reviewboard.asterisk.org/r/3225/ Patches: + dynamic_payload_change.patch uploaded by nbansal (license 6418) + ........ Merged revisions 408729 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408730 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 18:19 +0000 [r408712-408723] Richard Mudgett <rmudgett@digium.com> + + * main/manager.c, /: manager: Fix AMI Status action of a single + channel. Fixed use of uninitialized ao2 container iterator in an + off-nominal condition. Either a memory allocation error or the + requested channel is an internal channel not exposed to the + outside. ........ Merged revisions 408715 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/sorcery.c, res/ari/resource_endpoints.c, /, + apps/app_meetme.c, res/res_fax.c, res/res_stasis_recording.c, + main/stasis_channels.c, res/res_sorcery_astdb.c, + include/asterisk/json.h: json: Fix off-nominal json ref counting + issues. * Fixed off-nominal json ref counting issue with using + the following API calls: ast_json_object_set() and + ast_json_array_append(). * Fixed off-nominal error reporting in + ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal + json ref counting issues in report_receive_fax_status() and + dial_to_json(). ........ Merged revisions 408713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/json.c, /: json: Fix json API wrapper code for json library + versions earlier than 2.3.0. * Fixed json ref counting issue with + json API wrapper code for ast_json_object_update_existing() and + ast_json_object_update_missing() when the json library is earlier + than version 2.3.0. ........ Merged revisions 408711 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 16:49 +0000 [r408699] Corey Farrell <git@cfware.com> + + * channels/chan_sip.c: chan_sip: prevent add_route from adding + empty header. Fix regression caused by ASTERISK-22582. Empty + Route headers were added when the route had a single strict hop. + (closes issue ASTERISK-23306) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3236/ + +2014-02-21 16:27 +0000 [r408645-408652] Kevin Harwell <kharwell@digium.com> + + * main/rtp_engine.c, /: rtp_engine: Output mixup in + ${CHANNEL(rtpqos,audio,all)} Fixed the output of + CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter. + (closes issue ASTERISK-23261) Reported by: rsw686 Patches: + rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged + revisions 408646 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408647 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408649 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel.c, /: channel.c: MOH is not working for transferee + after attended transfer Updated the code to check to see if MOH + is playing on the transferor and if so then start it on the + channel that replaces it during a masquerade. Example scenario of + the problem: Alice calls Bob and then Bob begins the attended + transfer process into a queue. Upon going on hold Alice hears + music and so does Bob once he is in the queue. Bob then transfers + Alice into the queue and then music for Alice stops even though + she should be hearing it since has now replaced Bob in the queue. + The problem that was occurring is that once the channel was + masqueraded the app (queues, confbridge, etc...) had no way of + knowing that the channel had just been swapped out thus it did + not start music for the present channel. Credit to Olle Johansson + for pointing me in the right direction on this issue. (closes + issue ASTERISK-19499) Reported by: Timo Teräs Review: + https://reviewboard.asterisk.org/r/3226/ ........ Merged + revisions 408642 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408643 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408644 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 10:45 +0000 [r408592] Alexandr Anikin <may@telecom-service.ru> + + * /, addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay + variables ........ Merged revisions 408589 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408590 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408591 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 00:50 +0000 [r408539] Michael L. Young <elgueromexicano@gmail.com> + + * /, apps/app_chanspy.c: app_chanspy: Documentation Update To + Clarify "x" Option When using the "x" option (specify a DTMF + digit to exit the application), it is not obvious in the + documentation that this only works when spying on a channel. If a + channel being used to spy on other channels is waiting to connect + to a channel or is no longer attached to a channel, the DTMF is + ignored. As noted on the issue tracker, since there are + workarounds available and this is a rarely used option we are + opting for a documentation change here. (closes issue + ASTERISK-22661) Reported by: Chris Hillman Patches: + asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2990/ ........ Merged + revisions 408536 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408537 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408538 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-20 21:12 +0000 [r408519-408523] George Joseph <george.joseph@fairview5.com> + + * /, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c: pjsip_cli: Add pjsip + commands 'show registrations' and 'show contacts'. Added 'show + registrations' and 'show contacts' to pjsip cli to make things a + little more consistent. The output is exactly the same as the + list command. Just needed to add entries to their respective + ast_cli_entry structures. (closes issue ASTERISK-23275) Review: + http://reviewboard.asterisk.org/r/3210/ ........ Merged revisions + 408522 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_cli.c, main/config.c: pjsip_cli: Fix + memory leak in ast_sip_cli_print_sorcery_objectset. Fixed memory + leaks in ast_sip_cli_print_sorcery_objectset and + ast_variable_list_sort. (closes issue ASTERISK-23266) Review: + http://reviewboard.asterisk.org/r/3200/ ........ Merged revisions + 408520 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/sorcery.h, + res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c, + tests/test_sorcery.c, main/sorcery.c, /, + res/res_pjsip/config_system.c: sorcery: Create sorcery instance + registry. In order to retrieve an arbitrary sorcery instance from + a dialplan function (or any place else) there needs to be a + registry of sorcery instances. ast_sorcery_init now creates a + hashtab as a registry. ast_sorcery_open now checks the hashtab + for an existing sorcery instance matching the caller's module + name. If it finds one, it bumps the refcount and returns it. If + not, it creates a new sorcery instance, adds it to the hashtab, + then returns it. ast_sorcery_retrieve_by_module_name is a new + function that does a hashtab lookup by module name. It can be + called by the future dialplan function. res_pjsip/config_system + needed a small change to share the main res_pjsip sorcery + instance. tests/test_sorcery was updated to include a test for + the registry. (closes issue ASTERISK-22537) Review: + http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions + 408518 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-20 19:02 +0000 [r408503] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip.c, /: res_pjsip: Update documentation for + 'use_avpf' option When 'use_avpf' is set to True, inbound offers + must use the AVPF/SAVPF RTP profile. However, when 'use_avpf' is + set to False, Asterisk will accept both AVP/SAVP or AVPF/SAVPF + RTP profiles in inbound offers. The documentation previously + implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was + set to False and a UA offered said profile in an INVITE request. + ........ Merged revisions 408502 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-20 02:44 +0000 [r408450] Rusty Newton <rnewton@digium.com> + + * /, apps/app_queue.c: apps/app_queue - Fix incorrect Macro + parameter documentation Macro is executed on the called channel, + not the calling channel. (closes issue ASTERISK-23069) Reported + By: Bryan Anderson ........ Merged revisions 408447 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408448 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408449 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-19 19:09 +0000 [r408386-408390] Richard Mudgett <rmudgett@digium.com> + + * /, main/config.c: config: Add file size and nanosecond resolution + fields to the cached modified config file information. Repeatedly + modifying config files and reloading too fast sometimes fails to + reload the configuration because the cached modification + timestamp has one second resolution. * Added file size and + nanosecond resolution fields to the cached config file + modification timestamp information. Now if the file size changes + or the file system supports nanosecond resolution the modified + file has a better chance of being detected for reload. * Added a + missing unlock in an off-nominal code path. (closes issue + AST-1303) Review: https://reviewboard.asterisk.org/r/3235/ + ........ Merged revisions 408387 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408388 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408389 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix regex + handling and keep simple prefix matching performance. The sorcery + astDB wizzard does not handle regex correctly if the pattern + begins with an anchor character. This patch attempts to convert + the anchored regex pattern to a prefix pattern supported by astDB + for performance reasons. If it is not able to convert the pattern + it falls back to getting all astDB members of the family and + doing a normal regex pattern matching on the retrieved records. + Review: https://reviewboard.asterisk.org/r/3161/ ........ Merged + revisions 408385 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-19 12:04 +0000 [r408315-408332] Alexandr Anikin <may@telecom-service.ru> + + * addons/ooh323c/src/ooCapability.c, /, + addons/ooh323c/src/ooh245.c: process receiveAndTransmit user + input remote caps instead of receive only send receiveAndTransmit + user input our caps instead of receive only ........ Merged + revisions 408328 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408330 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408331 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * addons/ooh323c/src/ooh323.c, /: Allow different socket and + signalling ip on h.323 connection if gk mode is active Reported + by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by: + Gabriele Odone (closes issue ASTERISK-22738) ........ Merged + revisions 408312 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408314 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-18 19:19 +0000 [r408299] Richard Mudgett <rmudgett@digium.com> + + * contrib/ast-db-manage/config/env.py, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py, + contrib/ast-db-manage/config, + contrib/ast-db-manage/voicemail/env.py, + contrib/ast-db-manage/voicemail, + contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py, + contrib/ast-db-manage/config/versions, + contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py, + contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py, + contrib/ast-db-manage/voicemail/versions, contrib/ast-db-manage, + /: alembic: Add svn:ignore *.pyc to directories and + svn:executable to *.py files. ........ Merged revisions 408297 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-17 15:36 +0000 [r408272] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip/location.c, UPGRADE.txt, res/res_pjsip.c, + res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h: Store + SIP User-Agent information in contacts. When an endpoint sends a + REGISTER request to Asterisk, we now will associate the + User-Agent header with all contacts that were bound in that + REGISTER request. ........ Merged revisions 408270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-16 03:25 +0000 [r408199-408227] Matthew Jordan <mjordan@digium.com> + + * /, main/pbx.c: pbx: Handle a completely empty dialplan during a + context merge It is highly unlikely, but - at least in Asterisk + 12 - theoretically possible to load Asterisk with no dialplan + whatsoever. If that occurs, and some other module (that is not a + pbx module) attempts to merge its contexts into the dialplan, the + existing merge routine will crash. This is because it is not + insane, and rightly believes that you provided some sort of + dialplan, somewhere. This patch will gracefully merge the + contexts in such a case. Note that this is highly unlikely to + occur in 1.8/11, as features will most likely provide some + dialplan via parking. However, in Asterisk 12, parking is now + provided by res_parking, and hence may create its dialplan later. + (closes issue ASTERISK-23297) Reported by: CJ Oster Review: + https://reviewboard.asterisk.org/r/3222 ........ Merged revisions + 408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 408201 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408220 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, Makefile: buildsystem: Unbreak the build (infloop) on Asterisk + 11+ Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/ + ) broke the build. This patch fixes it by ignoring the .lastclean + dependencies if the MENUSELECT_EMBED variable is not defined. + patches: tmp.diff uploaded by wdoekes (License 5674) Review: + https://reviewboard.asterisk.org/r/3228/ ........ Merged + revisions 408193 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408194 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 21:44 +0000 [r408139-408141] Scott Griepentrog <sgriepentrog@digium.com> + + * main/stasis_endpoints.c, /: ARI: correct upper/lower case URI + discrepancies URI's are supposed to be case sensitive and all + lower case. In practice some portions of URI's in ARI are case + insensitive and others are not, such as TECH, which in one + instance would match a lower case name and in another would not. + In this patch, the ast_endpoint_lastest_snapshot() function is + modified to change the TECH portion to full upper case before + lookup. This resolves the discrepancy noted by the reporter. + However I chose to avoid forcing the /ari prefix of the URI's to + be lower case for now. Except for the two cases here, all URI's + should be lower case, unless they are part of a resource name or + id. Review: https://reviewboard.asterisk.org/r/3211/ Reported by: + Zane Conkle (closes issue ASTERISK-23125) ........ Merged + revisions 408140 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/format.c, /: format.c: correct possible null pointer + dereference In ast_format_sdp_parse and ast_format_sdp_generate + the check checks for a valid interface and function were + potentially confusing, and hid an error in the test of the + presence of the function that is called later. This patch clears + up and corrects the test. Review: + https://reviewboard.asterisk.org/r/3208/ (closes issue + ASTERISK-23098) Reported by: marcelloceschia Patches: + main_format.patch uploaded by marcelloceschia (license 6036) + ASTERISK-23098.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 408137 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408138 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 13:31 +0000 [r408086] Walter Doekes <walter+asterisk@wjd.nu> + + * Makefile, /: buildsystem: Don't force main to depend on + everything else. Directory 'main' only needs to depend on + embedded modules. If no module embedding is selected, the + dependency is dropped. Review: + https://reviewboard.asterisk.org/r/3212/ ........ Merged + revisions 408083 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408084 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408085 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 12:41 +0000 [r408070] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER + prior to calling bridge blind transfer This patch moves setting + SIP_DEFER_BY_ON_TRANSFER prior to calling + ast_bridge_transfer_blind. This prevents a BYE from being sent + prior to the NOTIFY request that informs the transferor if the + transfer succeeded or failed. This patch also clears said flag + from the off nominal NOTIFY paths in the local_attended_transfer + code, as once we've sent the NOTIFY request it is safe to send by + the BYE request. This was caught by the + blind-transfer-accountcode test in the Asterisk Test Suite. + (closes issue ASTERISK-23290) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3214/ ........ Merged + revisions 408069 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 08:52 +0000 [r408059] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * Makefile, build_tools/install_subst (added): install_subst: + helper script for installing with path substitution A helper + script to copy a source file substituting any + __ASTERISK_<foo>_DIR__ with the content of $AST<foo>DIR. Review: + https://reviewboard.asterisk.org/r/3202/ + +2014-02-13 18:52 +0000 [r407990-408006] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pubsub.c, /, res/res_pjsip_mwi.c: Remove all PJSIP + MWI-specific use from our MWI code. PJSIP has built-in MWI code + that could be useful to some degree, but our utilization of the + API actually made our code a bit more cluttered since we had to + have special cases peppered throughout. With this change, we move + to using the pjsip_evsub API instead, which streamlines the code + by removing special cases. Review: + https://reviewboard.asterisk.org/r/3205 ........ Merged revisions + 408005 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/location.c: Fix crash in AMI PJSIPShowEndpoint + action. If an AOR has no permanent contacts, then the + permanent_contacts container is never allocated. This makes the + code safe in the face of NULLs. I also changed the variable that + counts contacts from "num" to "total_contacts" since there are + now two variables that are indicate numbers of things. ........ + Merged revisions 407988 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-13 15:51 +0000 [r407989] Kinsey Moore <kmoore@digium.com> + + * main/logger.c, CHANGES: Logger: Add dynamic logger channels This + adds the ability to dynamically add and remove logger channels + from Asterisk via the CLI. (closes issue AST-1150) Review: + https://reviewboard.asterisk.org/r/3185/ + +2014-02-12 08:25 +0000 [r407970] Walter Doekes <walter+asterisk@wjd.nu> + + * /, main/config.c: realtime: Fix ast_update2_realtime() on + raspberry pi. The old code depended on undefined va_arg + behaviour: calling a function twice with the same va_list + parameter and expecting it to continue where it left off. The + changed code behaves like the manpage says it should. Also added + a bunch of early returns to trap errors (e.g. OOM) instead of + crashing. The problem was found by Julian Lyndon-Smith. The + deviant behaviour on the raspberry PI also uncovered another bug + (fixed in r407875) in the res_config_pgsql.so driver. Reported + by: jmls Tested by: jmls Review: + https://reviewboard.asterisk.org/r/3201/ ........ Merged + revisions 407968 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-11 20:17 +0000 [r407958] Joshua Colp <jcolp@digium.com> + + * main/sched.c: scheduler: Remove hashtab usage. This is a first + stab at tweaking the performance profile of the scheduler. + Removing the hashtab usage removes an extra memory allocation + when scheduling something and makes it so rescheduling does not + incur any memory allocation at all. Review: + https://reviewboard.asterisk.org/r/3199/ + +2014-02-11 03:18 +0000 [r407940] Matthew Jordan <mjordan@digium.com> + + * res/ari/resource_channels.c, /: ari/resource_channels: Add + channel variables earlier in the creation process This patch + tweaks the behaviour of POST /channels with channel variables + such that the variables are passed into the pbx.c routines that + perform the origination. This allows the variables to be assigned + to the newly created channels immediately upon their + construction, as opposed to be assigned after the originate has + completed. The upshot of this is that the variables are available + on the channels if they execute in the dialplan, as opposed to + only being available once the channels are answered. Review: + https://reviewboard.asterisk.org/r/3183/ ........ Merged + revisions 407937 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-10 18:28 +0000 [r407926] Corey Farrell <git@cfware.com> + + * channels/sip/include/reqresp_parser.h, + channels/sip/include/route.h (added), channels/chan_sip.c, + channels/sip/route.c (added), channels/sip/include/sip.h: + chan_sip: Isolate code that manages struct sip_route. * Move + route code to sip/route.c + sip/include/route.h * Rename + functions to sip_route_* * Replace ad-hoc list code with macro's + from linkedlists.h * Create sip_route_process_header() to + processes Path and Record-Route headers (previously done with + different code in build_route and build_path) * Add use of const + where possible * Move struct uriparams, struct contact and + contactliststruct from sip.h to reqresp_parser.h. sip/route.c + uses reqresp_parser.h but not sip.h, this was a problem. These + moved declares are not used outside of reqresp_parser. * While + modifying reqprep() the lack of {} caused me trouble. I added + them. * Code outside route.c treats sip_route as an opaque + structure, using macro's or procedures for all access. (closes + issue ASTERISK-22582) Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3173/ + +2014-02-10 16:49 +0000 [r407876] Walter Doekes <walter+asterisk@wjd.nu> + + * res/res_config_pgsql.c, /: res_config_pgsql: Fix + ast_update2_realtime calls. Fix so multiple updates from a single + call works (add missing ','). Remove bogus ast_free's that + weren't supposed to be there. Moved a few spaces for readability. + Review: https://reviewboard.asterisk.org/r/3194/ ........ Merged + revisions 407873 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407874 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407875 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-10 16:01 +0000 [r407859] Kinsey Moore <kmoore@digium.com> + + * apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c, + apps/confbridge/conf_state_empty.c, + apps/confbridge/conf_config_parser.c, + configs/confbridge.conf.sample, /, + apps/confbridge/include/confbridge.h, UPGRADE.txt: ConfBridge: + Correct prompt playback target Currently, when the first marked + user enters the conference that contains waitmarked users, a + prompt is played indicating that the user is being placed into + the conference. Unfortunately, this prompt is played to the + marked user and not the waitmarked users which is not very + helpful. This patch changes that behavior to play a prompt + stating "The conference will now begin" to the entire conference + after adding and unmuting the waitmarked users since the design + of confbridge is not conducive to playing a prompt to a subset of + users in a conference in an asynchronous manner. (closes issue + PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/ + Reported by: Steve Pitts ........ Merged revisions 407857 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407858 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 20:52 +0000 [r407767] Richard Mudgett <rmudgett@digium.com> + + * /, channels/chan_iax2.c: chan_iax2: Add some more iaxs[] NULL + checks to a routine already full of them. ........ Merged + revisions 407764 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407765 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407766 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 20:17 +0000 [r407752] Matthew Jordan <mjordan@digium.com> + + * /, main/security_events.c: security_events: Fix assertion failure + in dev-mode on optional IE parsing When formatting an optional + IE, the value is, of course, optional. As such, it is entirely + appropriate for ast_json_object_get to return NULL. If that + occurs, we now simply skip the IE that was requested, as it was + not provided by the entity that raised the event. Thanks to + George Joseph (gtjoseph) for catching this and reporting it in + #asterisk-dev ........ Merged revisions 407750 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 20:01 +0000 [r407749] Joshua Colp <jcolp@digium.com> + + * main/timing.c, res/res_timing_pthread.c, res/res_timing_dahdi.c, + res/res_timing_timerfd.c, include/asterisk/timing.h, + res/res_timing_kqueue.c: timing: Improve performance for most + timing implementations. This change allows timing implementation + data to be stored directly on the timer itself thus removing the + requirement for many implementations to do a container lookup for + the same information. This means that API calls into timing + implementations can directly access the information they need + instead of having to find it. Review: + https://reviewboard.asterisk.org/r/3175/ + +2014-02-07 19:40 +0000 [r407748] Matthew Jordan <mjordan@digium.com> + + * /, funcs/func_cdr.c: funcs/func_cdr: Handle empty time values + when extracting parsed values When extracting timestamps that are + parsed, time stamp values that are not set (time values of + 0.000000) should not actually result in a parsed string. The + value should be skipped, and the result of the CDR function + should be an empty string. Prior to this patch, the result was + fed to the time formatting, which would result in an output of a + date/time in 1969. ........ Merged revisions 407747 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 18:29 +0000 [r407731] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_iax2.c, include/asterisk/frame.h, + configs/iax.conf.sample, /: chan_iax2: Block unnecessary control + frames to/from the wire. Establishing an IAX2 call between + Asterisk v1.4 and v1.8 (or later) results in an unexpected call + disconnect. The problem happens because newer values in the enum + ast_control_frame_type are not consistent between the branch + versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) + using IAX2 2) v1.8 answers and sends a connected line update + control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 + receives the control frame as an end-of-q (on v1.4 + AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the + receive queue becomes empty. Several things are done by this + patch to fix the problem and attempt to prevent it from happening + again in the future: * Added a warning at the definition of enum + ast_control_frame_type about how to add new control frame values. + * Made block sending and receiving control frames that have no + reason to go over the wire. * Extended the connectedline iax.conf + parameter to also include the redirecting information updates. * + Updated the connectedline iax.conf parameter documentation to + include a notice that the parameter must be "no" when the peer is + an Asterisk v1.4 instance. (closes issue AST-1302) Review: + https://reviewboard.asterisk.org/r/3174/ ........ Merged + revisions 407678 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407727 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407729 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 16:47 +0000 [r407677] Matthew Jordan <mjordan@digium.com> + + * /, main/security_events.c: security_events: Fix error caused by + DTD validation error The appdocsxml.dtd specifies that a + "required" attribute in a parameter may have a value of yes, no, + true, or false. On some systems, specifying "False" instead of + "false" would cause a validation error. This patch fixes the + casing to explicitly match the DTD. ........ Merged revisions + 407676 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 13:15 +0000 [r407625] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * /, configs/indications.conf.sample: indications.conf: add stutter + tone; end properly * If the "stutter" (voicemail indication) tone + is indeed a stutter tone, and it ends with a constant tone, make + sure that it is the dial tone. This was done for India (in), + Mexico (mx) and the Philippines (ph). * If no "stutter" tone + exists for a country, provide one. This was done for Spain (es), + Malaysia (my) and Venezuela (ve). Review: + https://reviewboard.asterisk.org/r/3158/ ........ Merged + revisions 407622 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407623 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 21:24 +0000 [r407602] Matthew Jordan <mjordan@digium.com> + + * /, main/security_events.c, UPGRADE.txt, CHANGES: security_events: + Add AMI documentation; output optional fields This patch adds + documentation for the Security Events that are emited over AMI. + It also notes these events in the UPGRADE/CHANGES file. ........ + Merged revisions 407589 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 19:58 +0000 [r407588] Rusty Newton <rnewton@digium.com> + + * /, configs/pjsip.conf.sample: configs/pjsip.conf.sample: + Configuration section naming in pjsip.conf.sample needs a little + clarification There is a bit of nuance to how you name things in + pjsip.conf. This is a documentation patch to at least clear it up + a little for users. Review: + https://reviewboard.asterisk.org/r/3180/ ........ Merged + revisions 407587 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 18:11 +0000 [r407574] Kevin Harwell <kharwell@digium.com> + + * /, + contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py: + pjsip realtime: already created enum failure for postgresql If an + enum had been previously created the alembic script would attempt + to re-create it and an error would be generated while running + migrations for a postgresql server. The work around for this is + to use the ENUM object type for postgres as opposed to the + generic enum type used by sqlalchemy. Using this type in the + script seems to work properly for both postgres and mysql. + ........ Merged revisions 407572 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 17:55 +0000 [r407573] Richard Mudgett <rmudgett@digium.com> + + * res/res_pjsip_logger.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip_endpoint_identifier_ip.c, + include/asterisk/res_pjsip_cli.h, res/res_pjsip/pjsip_cli.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/config_domain_aliases.c: res_pjsip: Updates and + adds more PJSIP CLI commands. * Adds identify, transport, and + registration support to the PJSIP CLI. * Creates three additional + callbacks, one for an iterator, one for a comparator, and one for + a container. This eliminates the link dependency from higher + level modules to lower level ones. * Eliminates duplicate sorting + in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. * + Pushes CLI command registration down to the implementing source + file. * Adds several ast_sip_destroy_sorcery functions to + complement existing ast_sip_sorcery_initialize functions. The + destroy functions unregister PJSIP CLI commands and PJSIP CLI + formatters. Reported by: George Joseph Review: + https://reviewboard.asterisk.org/r/3104/ ........ Merged + revisions 407568 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 23:04 +0000 [r407514] Rusty Newton <rnewton@digium.com> + + * /, formats/format_wav.c: formats/format_wav: enhancing log + message "Not a wav file" to be clear on what is supported + Modifying the log message to be more specific as to what is + supported. Specifically it seems format_wav supports only PCM + encoded versions with a lower-case '.wav' extension. (closes + issues ASTERISK-22310) Reported by: Jim Credland Review: + https://reviewboard.asterisk.org/r/3188/ ........ Merged + revisions 407511 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407512 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407513 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 20:56 +0000 [r407462] Jonathan Rose <jrose@digium.com> + + * CHANGES, /: CHANGES: Improved description of Name/Creator changes + to bridge ARI, adds AMI The changes log was written with language + that was a little too internal Asterisk specific, so it's been + changed to be more in the frame of reference of an ARI user. + Also, previously the AMI event changes were omitted from the + change log as well as the ability to include a bridge name in the + ARI post bridges command. ........ Merged revisions 407461 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 20:43 +0000 [r407459] Kinsey Moore <kmoore@digium.com> + + * main/logger.c, /: Logger: Fix handling of absolute paths This + fixes path handling for log files so that an extra / is not + appended to the file path when the path is absolute (begins with + /). This would previously result in different but functionally + equivalent paths in the output of 'logger show channels'. + ........ Merged revisions 407455 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407456 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 19:42 +0000 [r407443] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip/config_global.c, /: res_pjsip: When no global type + the debug option defaults to "yes" If the global section was not + specified in pjsip.conf then the configuration object does not + exist in sorcery so when retrieving "debug" option it would + return NULL. Then the NULL result was passed to ast_false utils + function which would return false because it wasn't set to some + representation of false, thus enabling sip debug logging. Made it + so if the global config object does not exist then it will return + a default of "no" for sip debugging. (issue ASTERISK-23038) + Reported by: Rusty Newton ........ Merged revisions 407442 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 17:42 +0000 [r407422-407425] Jonathan Rose <jrose@digium.com> + + * CHANGES: CHANGES: Update changes log to include r403414 entry + Adds note of additional 0 for operator option on app_record + + * CHANGES, /: CHANGES: Update changes log to include new bridge + fields added in r404042 ........ Merged revisions 407419 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 15:29 +0000 [r407407] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/playbacks.json, UPGRADE.txt, + rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES, + include/asterisk/manager.h, rest-api/api-docs/bridges.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/mailboxes.json, + rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/channels.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + /: ARI/AMI: Update versions; update UPGRADE/CHANGES notes for + 12.1.0 changes Due to backwards compatible changes made to + AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0, + respectively. ........ Merged revisions 407402 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 20:15 +0000 [r407275-407340] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/devicestate.h, /, main/devicestate.c: + devicestate: Make ast_devstate_changed_literal() return value and + doxygen consistent. Nothing actually cares about the value + anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose + ........ Merged revisions 407337 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407338 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407339 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix assertion + for pjsip.conf authorization list options. (closes issue + ASTERISK-23168) Reported by: George Joseph Review: + https://reviewboard.asterisk.org/r/3143/ ........ Merged + revisions 407324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/sip.conf.sample, main/tcptls.c, /: tcptls.c: Made TLS + handle a certificate chain file. Thanks to Guillaume Martres for + doing the necessary research to validate the change. (closes + issue ASTERISK-17727) Reported by: LN Patches: + use_certificate_chain.patch (license #5864) patch uploaded by st + documente_certificate_chain.patch (license #6576) patch uploaded + by Guillaume Martres ........ Merged revisions 407272 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407273 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407274 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 16:55 +0000 [r407260] Matthew Jordan <mjordan@digium.com> + + * /, funcs/func_cdr.c: funcs/func_cdr: Fix non-epoch timestamps + broken by improper char array deref Thanks to snuffy for pointing + this issue out and fixing it. (closes issue ASTERISK-23250) + Reported by: snuffy patches: func_cdr-fix.diff uploaded by snuffy + (License 5024) ........ Merged revisions 407259 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 02:22 +0000 [r407217] Joshua Colp <jcolp@digium.com> + + * res/res_clialiases.c, /: res_clialiases: Fix crash when reloading + and re-aliasing an alias that is in use. The code assumed that + unregistering the alias would always succeed while in practice + this is not actually true. A common case is the "reload" command + itself. If the cli_aliases.conf configuration file was changed + and reload executed the command would fail to unregister and + ultimately point to freed memory. The reload process now checks + whether unregistering succeeded or not and if not the old CLI + alias is retained. (closes issue ASTERISK-19773) Reported by: + Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth + Blades ........ Merged revisions 407205 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407210 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407213 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 02:07 +0000 [r407198] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Skinny - Fix deadlock when pickup of + no call. Locking issues in skinny when picking up a call that + doesn't exist. Cleaned up sub locking by fully removing and using + the chan lock instead. Also changed ast_call_pickup to check + whether chan was masq'd. (closes issue ASTERISK-23249) Reported + by: wedhorn Tested by: snuffy, myself Patches: + skinny-locking01.diff uploaded by wedhorn (license 5019) ........ + Merged revisions 407197 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-03 01:31 +0000 [r407169] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /: cdrs: Check for applications to lock onto during + dial begin handling This patch brings CDR processing further in + line with r407085. During some dial operations, the application + would not be locked to the Dial application and would instead + continue to show the previously known application. In particular, + this would occur when a Parked call would time out. This was due + to a previous snapshot already locking the application to Park - + processing this in a Dial Begin allows the Dial application to + reassert its rightful place. (CDRs. Ugh.) But hooray for the + Parked Call tests for catching this in the Asterisk Test Suite. + ........ Merged revisions 407166 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-01 16:26 +0000 [r407154] Joshua Colp <jcolp@digium.com> + + * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /, + res/stasis/app.c, res/ari/ari_model_validators.c, + res/res_stasis.c, main/stasis_bridges.c: res_stasis: Enable + transfers and provide events when they occur. This change enables + transfers within ARI created bridges and adds events for when + they occur. Unlike other events these will be received if *any* + subscribed object is involved in the transfer. (closes issue + ASTERISK-22984) Reported by: David M. Lee Review: + https://reviewboard.asterisk.org/r/3120/ ........ Merged + revisions 407153 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-01 00:25 +0000 [r407105] Corey Farrell <git@cfware.com> + + * apps/app_stack.c, /: app_stack: protect against missing + parameters to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2 + parameters and LOCAL_PEEK requires 1 parameter. This protects + against situations where those parameters are blank or missing by + logging an error and returning. (closes issue ASTERISK-23220) + Reported by: James Sharp ........ Merged revisions 407100 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407103 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407104 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 23:40 +0000 [r407083-407085] Matthew Jordan <mjordan@digium.com> + + * apps/app_dial.c, main/cdr.c, main/pbx.c, /, main/bridge_after.c, + UPGRADE.txt, main/manager_channels.c: CDRs: fix a variety of dial + status problems, h/hangup handler creating CDRs This patch fixes + a number of small-ish problems that were noticed when witnessing + the records that the FreePBX dialplan produces: (1) Mid-call + events (as well as privacy options) have the ability to change + the overall state of the Dial operation after the called party + answers. This means that publishing the DialEnd event when the + called party is premature; we have to wait for the execution of + these subroutines to complete before we can signal the overall + status of the DialEnd. This patch moves that publication and adds + handlers for the mid-call events. (2) The AST_FLAG_OUTGOING + channel flag is cleared if an after bridge goto datastore is + detected. This flag was preventing CDRs from being recorded for + all outbound channels that had a 'continue' option enabled on + them by the Dial application. (3) The CDR engine now locks the + 'Dial' application as being the CDR application if it detects + that the current CDR has entered that app. This is similar to the + logic that is done for Parking. In general, if we entered into + Dial, then we want that CDR to record the application as such - + this prevents pre-dial handlers, mid-call handlers, and other + shenaniganry from changing the application value. (4) The CDR + engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more + places to determine if the channel is in hangup logic or dead. In + either case, we don't want to record changes in the channel. (5) + The default option for "endbeforehexten" has been changed to + "yes". In general, you don't want to see CDRs in the 'h' exten or + in hangup logic. Since the semantics of that option changed in + 12, it made sense to update the default value as well. (6) + Finally, because we now have the ability to synchronize on the + messages published to the CDR topic, on shutdown the CDR engine + will now synchronize to the messages currently in flight. This + helps to ensure that all in-flight CDRs are written before + shutting down. (closes issue ASTERISK-23164) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3154 ........ + Merged revisions 407084 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_dial.c, /: app_dial: Allow macro/gosub pre-bridge + execution to occur on priorities The parsing for the destination + of the macro/gosub uses the '^' character to separate out + context, extension, and priority. However, the logic for the + macro/gosub execution was written such that it would only do the + actual macro/gosub jump if a '^' character existed. This doesn't + apply when the macro/gosub jump occurs in a priority/priority + label. This patch changes the logic so that the parsing still + occurs, but the jump will occur even for priorities/priority + labels. (issue ASTERISK-23164) Review: + https://reviewboard.asterisk.org/r/3154 ........ Merged revisions + 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 407074 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407082 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 23:15 +0000 [r407035-407037] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_logger.c, CHANGES, res/res_pjsip.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py + (added), /, configs/pjsip.conf.sample, UPGRADE.txt: res_pjsip: + Config option to enable PJSIP logger at load time. Added a + "debug" configuration option for res_pjsip that when set to "yes" + enables SIP messages to be logged. It is specified under the + "system" type. Also added an alembic script to add the option to + realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton + Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged + revisions 407036 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_exten_state.c, /: res_pjsip_exten_state: Exporting + global symbols caused load order issues Removed the exportation + of global symbols from the module as it is no longer needed and + it could potentially cause load problems as on some systems it + would try to load before res_pjsip_pubsub ........ Merged + revisions 407034 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 23:04 +0000 [r407033] Richard Mudgett <rmudgett@digium.com> + + * CHANGES, apps/app_chanspy.c: ChanSpy: Add ability to specify + channel uniqueids as well as channel names. * Made ChanSpy accept + a channel uniqueid or a fully specified channel name as the + chanprefix parameter if the 'u' option is specified. (closes + issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/ + +2014-01-31 22:39 +0000 [r407030-407032] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/res_pjsip_presence_xml.h (added), /: Add file + that apparently got missed in the merge. ........ Merged + revisions 407031 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pidf_body_generator.c (added), + include/asterisk/res_pjsip_exten_state.h (removed), + res/res_pjsip_pubsub.exports.in, /, + include/asterisk/res_pjsip_body_generator_types.h (added), + res/res_pjsip_mwi.c, res/res_pjsip_xpidf_body_generator.c + (added), res/res_pjsip_mwi_body_generator.c (added), + res/res_pjsip_pubsub.c, res/res_pjsip_pidf.c (removed), + res/res_pjsip_pidf_eyebeam_body_supplement.c (added), + res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c + (added), include/asterisk/res_pjsip_pubsub.h: Decouple + subscription handling from NOTIFY/PUBLISH body generation. When + the PJSIP pubsub framework was created, subscription handlers + were required to state what event they handled along with what + body types they knew how to generate. While this serves well when + implementing a base RFC, it has problems when trying to extend + the body to support non-standard or proprietary body elements. + The code also was NOTIFY-specific, meaning that when the time + comes that we start writing code to send out PUBLISH requests + with MWI or presence bodies, we would likely find ourselves + duplicating code that had previously been written. This changeset + introduces the concept of body generators and body supplements. A + body generator is responsible for allocating a native structure + for a given body type, providing the primary body content, + converting the native structure to a string, and deallocating + resources. A body supplement takes the primary body content (the + native structure, not a string) generated by the body generator + and adds nonstandard elements to the body. With these elements + living in their own module, it becomes easy to extend our support + for body types and to re-use resources when sending a PUBLISH + request. Body generators and body supplements register themselves + with the pubsub core, similar to how subscription and publish + handlers had done. Now, subscription handlers do not need to know + what type of body content they generate, but they still need to + inform the pubsub core about what the default body type for a + given event package is. The pubsub core keeps track of what body + generators and body supplements have been registered. When a + SUBSCRIBE arrives, the pubsub core will check that there is a + subscription handler for the event in the SUBSCRIBE, then it will + check that there is a body generator that can provide the content + specified in the Accept header(s). Because of the nature of body + generators and supplements, it means res_pjsip_exten_state and + res_pjsip_mwi have been completely gutted. They no longer worry + about body types, instead calling + ast_sip_pubsub_generate_body_content() when they need to generate + a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150 + ........ Merged revisions 407016 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 22:23 +0000 [r407015-407029] Kevin Harwell <kharwell@digium.com> + + * contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py, + contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py, + /, UPGRADE.txt: alembic: script modifications due to errors A + couple of the scripts had errors that would not allow a full + migration to take place. The extensions table needed to make its + 'id' column a primary key in order to work with mysql. The other + script ...add_endpoints... was missing tables that it was trying + to add columns to. Added the primary key on id for extensions and + added the tables in for the missing pjsip configuration options. + While it is not ideal to modify already released scripts this was + a case where it had to be done due to errors in the script and + lacking a better alternative. Review: + https://reviewboard.asterisk.org/r/3167/ ........ Merged + revisions 407019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c: res_pjsip_mwi: Subscribe fails when + missing aor name When subscribing to MWI (res_pjsip_mwi) and the + sip uri did not contain a name (ex: sip:<ip address>) then the + subscription would fail since it would be unable to locate an + associated aor. This patch makes it so that when a subscribe + comes with no aor name then it will subscribe to all aors on the + located endpoint. (closes issue ASTERISK-23072) Reported by: Bob + M Review: https://reviewboard.asterisk.org/r/3164/ ........ + Merged revisions 407014 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 15:08 +0000 [r407001] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip_nat.c, /: PJSIP: Fix address for ACK in NAT + situations In NAT scenarios where a call is placed to a + Grandstream phone, res_pjsip will sometimes send the ACK to a 200 + OK to the private address of the device behind the NAT instead of + the address of the NAT device. This corrects that behavior by + rewriting the address in the Contact header in the incoming 200 + OK and the dialog's target address if necessary (since it has + already been rewritten to the incorrect private address). (closes + issue ASTERISK-23106) Review: + https://reviewboard.asterisk.org/r/3168/ Reported by: Matt Jordan + ........ Merged revisions 407000 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 05:31 +0000 [r406988] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Skinny: fix up possible double unlock + of chan. Return before chan is possibly unlocked a second time + when hanging up a channel in SUBSTATE_OFFHOOK. ........ Merged + revisions 406987 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-30 20:36 +0000 [r406936] Corey Farrell <git@cfware.com> + + * main/udptl.c, res/res_rtp_asterisk.c, /: res_rtp_asterisk & + udptl: fix port selection to work with SELinux restrictions + ast_bind to a port reserved for another program by SELinux causes + errno == EACCES. This caused random failures when binding rtp or + udptl sockets. Treat EACCES as a non-fatal error, try next port. + (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ + Merged revisions 406933 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406934 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406935 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-30 17:35 +0000 [r406920] Sean Bright <sean@malleable.com> + + * main/manager.c, /: Make a NOTICE about an invalid channel name + more useful. ........ Merged revisions 406918 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406919 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-29 00:44 +0000 [r406863] Russell Bryant <russell@russellbryant.com> + + * /, configs/queues.conf.sample: queues.conf.sample Fix documented + default for persistentmembers Closes issue ASTERISK-22662 + ........ Merged revisions 406860 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406861 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406862 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 23:40 +0000 [r406789-406848] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: potential crash on + timeout What seems to be happening is if a subscription has been + terminated and the subscription timeout/expires is less than the + time it takes for all pending transactions (currently on the + subscription) to end then the subscription timer will not have + been canceled yet and sub will be null. Since the subscription + has already been canceled nothing needs to be done so a null + check in the asterisk code is sufficient in working around this + problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins + ........ Merged revisions 406847 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * cdr/cdr_radius.c, cel/cel_radius.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: cdr_radius, + cel_radius: build agains libfreeradius-client Asterisk's RADIUS + module currently build against libradiusclient-ng, but this + project has been superseeded by libfreeradius-client. The API is + 99% compatible except that the header name has changed, the + library name has changed, and the configuration file location has + changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé + Patches: freeradius-client.patch uploaded by sharky (license + 6561) ........ Merged revisions 406801 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406802 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406803 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/include/res_pjsip_private.h, /, + include/asterisk/compat.h: res_pjsip,compat: INFINITY and NAN + undefined On some systems the values for INFINITY and NAN are not + defined thus causing a build error on those systems. Added + definitions for those if they had not previously been defined. + (closes issue ASTERISK-23056) Reported by: capouch Patches: + inf-nan-patch.txt uploaded by capouch (license 6564) ........ + Merged revisions 406788 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 19:19 +0000 [r406778] Kinsey Moore <kmoore@digium.com> + + * /, res/res_stasis_device_state.c: ARI: Make double subscribe + respond with success Currently, attempting to subscribe an + application to a device state that it has already subscribed to + will generate a 500 error response. This will now be treated as a + subscription refresh even though ARI subscriptions don't + currently support lifetimes and will respond with the normal + response for a successful subscription (200 OK). (closes issue + ASTERISK-23143) Reported by: Matt Jordan ........ Merged + revisions 406775 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 16:43 +0000 [r406724] Scott Griepentrog <sgriepentrog@digium.com> + + * main/rtp_engine.c, /: rtp_engine: improved handling of + get_rtp_info failure In ast_rtp_instance_make_compatible(), after + a failure of channel tech call get_rtp_info() to return + peer_instance, the null pointer would be passed to ao2_ref, + producing an error that looked like a refernce counting problem + but is not. This patch corrects that and adds helpful LOG_ERROR + messages to indicate which failure path occurred. (issue + AST-1276) Review: https://reviewboard.asterisk.org/r/3156/ + ........ Merged revisions 406721 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406722 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406723 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 00:20 +0000 [r406710] Richard Mudgett <rmudgett@digium.com> + + * /, tests/test_cel.c, tests/test_cdr.c: test_cdr.c, test_cel.c: + Correctly destroy created bridges. * Fixed the + test_cel_attended_transfer_bridges_link unit test to also account + for the local channel link being destroyed now that the bridges + are actually destroyed. * Made CDR unit test use its own version + of do_sleep() from the CEL unit tests. ........ Merged revisions + 406707 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-27 22:54 +0000 [r406647-406696] Kevin Harwell <kharwell@digium.com> + + * CHANGES: manager: ExtensionStatus event status human readable + Added a note in the changes file about the new 'StatusText' field + that was added to the 'ExtensionStatus' event. (issue + ASTERISK-23154) Reported by: Jonathan Rose + + * main/manager.c: manager: ExtensionStatus event status human + readable When an 'ExtensionStatus' event was raised it included + the status as a numerical value, but did not include a text + description of the status. Added a 'StatusText' field to the + event which is a string representation of the extension status. + Also added this to the 'Extension State' command response. + (closes issue ASTERISK-23154) Reported by: Jonathan Rose + +2014-01-27 20:38 +0000 [r406646] Russell Bryant <russell@russellbryant.com> + + * main/config.c, /: Allow nested #includes in extconfig.conf + extconfig.conf was hard-coded to not allow nested includes for + some reason. The code has been this way since a patch was merged + for ASTERISK-3333 (revision 4889), which was a significant update + to this code ("Merge config updates"). I can't figure out any + good reason why this should be limited. This patch just removes + the limit and uses the default nesting depth limit. Closes issue + ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/ + ........ Merged revisions 406643 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406644 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406645 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-27 08:17 +0000 [r406618] Walter Doekes <walter+asterisk@wjd.nu> + + * main/manager.c, UPGRADE.txt, configs/manager.conf.sample: + manager: The eventfilter= option now takes an extended regex. In + pre-trunk versions (...12) it accepts a basic regex, which is + confusing because all other regexes in asterisk are of the + extended kind. Review: https://reviewboard.asterisk.org/r/3147/ + +2014-01-27 01:25 +0000 [r406595] Russell Bryant <russell@russellbryant.com> + + * main/file.c, include/asterisk/channel.h, main/channel.c, /: + Protect ast_filestream object when on a channel The + ast_filestream object gets tacked on to a channel via + chan->timingdata. It's a reference counted object, but the + reference count isn't used when putting it on a channel. It's + theoretically possible for another thread to interfere with the + channel while it's unlocked and cause the filestream to get + destroyed. Use the astobj2 reference count to make sure that as + long as this code path is holding on the ast_filestream and + passing it into the file.c playback code, that it knows it's + valid. Bug reported by Leif Madsen. Review: + https://reviewboard.asterisk.org/r/3135/ ........ Merged + revisions 406566 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406567 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406574 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-26 23:04 +0000 [r406517] Richard Mudgett <rmudgett@digium.com> + + * /, main/tcptls.c: tcptls.c: Add missing cleanup on off nominal + path. ........ Merged revisions 406514 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406515 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406516 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-26 14:19 +0000 [r406503] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * contrib/scripts/live_ast: live_ast: run wrapped programs with + exec live_ast can be used as a wrapper script to run asterisk, + gdb or valgrind. In those cases it runs them and returns the + result. It is more useful to use 'exec' to avoid having another + odd process in the chain. Review: + https://reviewboard.asterisk.org/r/3110/ + +2014-01-26 02:11 +0000 [r406490] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, /: res_pjsip_session: Be less strict + with core requested outgoing capabilities. The core may + (depending on circumstances) request a single codec on outgoing + calls. Many channel drivers ignore or treat this as a suggestion + while still including configured codecs. The res_pjsip_session + logic treated this as an explicit request, leaving out other + configured codecs. This change makes res_pjsip_session behave + like other channel driver and simply adds the requested codec to + the list. (closes issue ASTERISK-23082) Reported by: xrobau + Review: https://reviewboard.asterisk.org/r/3140/ ........ Merged + revisions 406489 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 23:33 +0000 [r406466] Richard Mudgett <rmudgett@digium.com> + + * /, main/cel.c: CEL: Protect data structures during reload and + shutdown. The CEL data structures need to be protected during a + configuration reload and shutdown. Asterisk crashed during a + shutdown because CEL events were still in flight and the CEL data + structures were already destroyed. * Protected the cel_backends, + cel_dialstatus_store, and cel_linkedids ao2 containers with a + global ao2 object wrapper. * Added NULL checks before use of the + cel_backends, cel_dialstatus_store, and cel_linkedids ao2 + containers in case the CEL module is already shutdown. * Fixed + overloading of the cel_linkedids held objects reference count. + During shutdown any held objects would be leaked. * Fixed memory + leak of cel_linkedids held objects if the LINKEDID_END is not + being tracked. The objects in the cel_linkedids container were + not removed if the LINKEDID_END event is not used. * Added access + protection to the cel_backends container during the CLI "cel show + status" command. * Made cel_backends, cel_dialstatus_store, and + cel_linkedids use the standard ao2 callback templates for the + hash and cmp functions. * Eliminated unnecessary uses of + RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated + resources on failure. (closes issue AST-1253) Reported by: + Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3128/ ........ Merged + revisions 406417 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406418 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 22:34 +0000 [r406416] Jonathan Rose <jrose@digium.com> + + * main/utils.c, CHANGES: Thread Debugging: Add LWP to core show + locks output This patch adds the LWP to core show locks output if + it is available. Review: https://reviewboard.asterisk.org/r/3142/ + +2014-01-24 22:18 +0000 [r406407] Richard Mudgett <rmudgett@digium.com> + + * main/manager.c, /: manager: Register atexit shutdown routine only + once. * Made register atexit shutdown routine only once in + __init_manager(). * Fixed some initial load failure conditions in + __init_manager(). * Made reset options to defaults on reload when + the reload will actually happen. * Removed unnecessary container + traversals of the white/black filters during manager_free_user(). + * ast_free() does not need a NULL check before calling. ........ + Merged revisions 406359 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406400 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406401 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 21:46 +0000 [r406399] Jonathan Rose <jrose@digium.com> + + * res/res_config_pgsql.c, /: res_config_pgsql: Fix a memory leak + and use RAII_VAR for cleanup when practical Review: + https://reviewboard.asterisk.org/r/3141/ ........ Merged + revisions 406360 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406361 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406389 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 18:13 +0000 [r406343] Richard Mudgett <rmudgett@digium.com> + + * main/manager.c, /: manager: Protect data structures during + shutdown. Occasionally, the manager module would get an + "INTERNAL_OBJ: bad magic number" error on a "core restart + gracefully" command if an AMI connection is established. * Added + ao2_global_obj protection to the sessions global container. * + Fixed the order of unreferencing a session object in + session_destroy(). * Removed unnecessary container traversals of + the white/black filters during session_destructor(). (closes + issue AST-1242) Reported by: Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3144/ ........ Merged + revisions 406341 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406342 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-23 23:43 +0000 [r406328] Mark Michelson <mmichelson@digium.com> + + * /: Today is not my day for writing code that compiles. ........ + Merged revisions 406327 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-23 22:56 +0000 [r406312] Michael L. Young <elgueromexicano@gmail.com> + + * /, addons/res_config_mysql.c: res_config_mysql: Fix Setting The + Column Name Incorrectly When support for a realtime sorcery + module was added in revision 386731, the wrong property was + accidentally used for setting the column name to be updated in + the database table. This patch fixes the typo. (closes issue + ASTERISK-23177) Reported by: Denis Tested by: Denis Patches: + asterisk-23177-use-field-name.diff by Michael L. Young (license + 5026) ........ Merged revisions 406311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-23 21:18 +0000 [r406298] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pidf.c, /: Multiple revisions 406294-406295 + ........ r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu, + 23 Jan 2014) | 11 lines Fix presence body errors found during + testing: * PIDF bodies were reporting an "open" state in many + cases where it should have been reporting "closed" * XPIDF bodies + had XML nodes placed incorrectly within the hierarchy. * SIP URIs + in XPIDF bodies did not go through XML sanitization * XML + sanitization had some errors: * Right angle bracket was being + replaced with "&rt;" instead of ">" * Double quote, + apostrophe, and ampersand were not being escaped. ........ + r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan + 2014) | 11 lines Fix presence body errors found during testing: * + PIDF bodies were reporting an "open" state in many cases where it + should have been reporting "closed" * XPIDF bodies had XML nodes + placed incorrectly within the hierarchy. * SIP URIs in XPIDF + bodies did not go through XML sanitization * XML sanitization had + some errors: * Right angle bracket was being replaced with "&rt;" + instead of ">" * Double quote, apostrophe, and ampersand were + not being escaped. ........ Merged revisions 406294-406295 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-22 22:24 +0000 [r406269] Scott Griepentrog <sgriepentrog@digium.com> + + * main/pbx.c, /, utils/extconf.c: pbx.c: Pre-initialize timezone to + avoid crash on destroy In ast_build_timing, initialize the + timezone value to NULL in order to avoid deferencing an + uninitialized value later when calling ast_destroy_timing. The + timezone value could be uninitialized if ast_build_timing were to + fail due to a zero length time string. (closes issue + ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review: + https://reviewboard.asterisk.org/r/3134/ Patches: + ast_build_timing-initialize-timezone.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 406241 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406245 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406264 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-22 19:36 +0000 [r406153-406224] Kinsey Moore <kmoore@digium.com> + + * /, apps/app_confbridge.c: ConfBridge: Fix channel parameter + documentation Confbridge AMI and CLI commands for mute, unmute, + and setting the single video source can accept channel prefixes + in lieu of a full channel name, but documentation states only + that it is required and is a channel name. This corrects the + documentation. (closes issue PQ-1397) Reported by: Steve Pitts + ........ Merged revisions 406217 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406223 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Decline image streams on + unsupported transports This change allows chan_sip to decline + individual image streams over unsupported transports in the SDP + of the 200 response. Previously, an image stream offer with + RTP/AVP as the transport would cause chan_sip to respond with a + 488. (closes issue ASTERISK-22988) Reported by: adomjan Original + patch by: adomjan ........ Merged revisions 406170 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406171 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406172 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /: res_stasis_playback: Correct error + argument order Several of the playback error messages for invalid + media input in res_stasis_playback.c had the media name and + channel name reversed. They now correctly identify the channel + name and media name. Reported by: skrusty ........ Merged + revisions 406152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-21 21:48 +0000 [r406134] Rusty Newton <rnewton@digium.com> + + * /, res/res_pjsip.c: res_pjsip: Documentation improvement for + Endpoint and AOR mailbox options. Making the help text for both + more explicit regarding the format of mailbox identifiers. i.e. + clarifying the format for app_voicemail mailboxes vs mailboxes + from external MWI sources through modules such as + res_external_mwi. ........ Merged revisions 406133 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-21 21:08 +0000 [r406082] Walter Doekes <walter+asterisk@wjd.nu> + + * main/manager.c, /, configs/manager.conf.sample: manager: Clarify + eventfilter documentation. Textual changes only. Review: + https://reviewboard.asterisk.org/r/3133/ ........ Merged + revisions 406079 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406080 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406081 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-21 20:28 +0000 [r406006-406078] Kinsey Moore <kmoore@digium.com> + + * channels/chan_mgcp.c, /: chan_mgcp: Enforce locking for oseq This + restricts direct usage of global oseq so that all accesses are + locked and threads are not racing to get oseq values that they + did not claim. This also fixes a build error in res_pktccops + under dev mode. (closes issue ASTERISK-23100) Reported by: + adomjan Patch by: adomjan ........ Merged revisions 406037 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406038 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406049 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c, res/res_pjsip.c: PJSIP: + Handle headers in a list appropriately The PJSIP header parsing + function (pjsip_parse_hdr) can generate more than one header + instance from a single header field. These header instances exist + as a list attached to the returned header and must be handled + appropriately when they are added to a message or else only the + first header instance will be used. This changes the linked list + functions used in outbound proxy code to merge the lists + properly. ........ Merged revisions 406020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_sounds.h, res/ari/resource_bridges.h, + res/ari/resource_device_states.h, res/ari/resource_mailboxes.h, + res/ari/resource_asterisk.h, rest-api/api-docs/channels.json, + res/ari/resource_applications.h, res/ari/resource_channels.c, + res/res_ari_playbacks.c, res/res_ari_sounds.c, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, res/res_ari_bridges.c, /, + res/res_ari_device_states.c, + rest-api-templates/ari_resource.h.mustache, + res/res_ari_mailboxes.c, res/res_ari_asterisk.c, + res/res_ari_applications.c, + rest-api-templates/res_ari_resource.c.mustache, + rest-api-templates/body_parsing.mustache (added), + res/res_ari_channels.c, res/ari/resource_playbacks.h, + rest-api-templates/param_parsing.mustache: ARI: Support channel + variables in originate This adds back in support for specifying + channel variables during an originate without compromising the + ability to specify query parameters in the JSON body. This was + accomplished by generating the body-parsing code in a separate + function instead of being integrated with the URI query parameter + parsing code such that it could be called by paths with body + parameters. This is transparent to the user of the API and + prevents manual duplication of code or data structures. (closes + issue ASTERISK-23051) Review: + https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan + ........ Merged revisions 406003 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 23:25 +0000 [r405985] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Skinny: fix up handling of fragmented + packets. Bad offset in reading second or more fragment of skinny + packets. Fixed to offset by char (single byte) rather than size + of req. ........ Merged revisions 405982 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 22:23 +0000 [r405947] Richard Mudgett <rmudgett@digium.com> + + * channels/sig_pri.c, /: chan_dahdi/PRI: Suppress CONNECTED_LINE + updates when nothing in the udpate is valid. * Also simplified + some subddress handling code. (closes issue ASTERISK-23008) + Reported by: Michael Cargile ........ Merged revisions 405926 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 405927 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405928 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 21:56 +0000 [r405925] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Skinny: fix up session logging. + Logging from the skinny session loop was providing some incorrect + reasons for exiting the loop. Cleaned up messages and handling so + correct reason displayed. ........ Merged revisions 405924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 18:18 +0000 [r405910] Jonathan Rose <jrose@digium.com> + + * channels/chan_pjsip.c, /: chan_pjsip: Provide a means for + tracking device state when holding/unholding Previously PJSIP did + not track hold/unhold and it would always simply be 'inuse'. This + patch fixes that. review: + https://reviewboard.asterisk.org/r/3129/ ........ Merged + revisions 405908 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-19 00:01 +0000 [r405894] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Skinny: fix reversed device reset from + CLI. Existing code would do a full device restart when "skinny + reset device" was entered at the CLI and do a reset when "skinny + reset device restart" entered. ........ Merged revisions 405893 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 22:09 +0000 [r405878] Sean Bright <sean@malleable.com> + + * /, channels/chan_sip.c: Make sure the maxptime attribute is added + to the correct offers. ........ Merged revisions 405877 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 21:33 +0000 [r405862-405876] Scott Griepentrog <sgriepentrog@digium.com> + + * main/format_pref.c, main/sorcery.c, main/frame.c, /, + include/asterisk/format_pref.h, res/res_pjsip_sdp_rtp.c: pjsip: + fix support for allow=all This change adds improvements to + support for allow=all in pjsip.conf so that it functions as + intended. Previously, the allow/disallow socery configuration + would set & clear codecs from the media.codecs and media.prefs + list, but if all was specified the prefs list was not updated. + Then a call would fail when create_outgoing_sdp_stream() created + an SDP with no audio codecs. A new function + ast_codec_pref_append_all() is provided to add all codecs to the + prefs list - only those not already on the list. This enables the + configuration to specify a codec preference, but still add all + codecs, and even then remove some codecs, as shown in this + example: allow = ulaw, alaw, all, !g729, !g723 Also, the display + order of allow in cli output is updated to match the + configuration by using prefs instead of caps when generating a + human readable string. Finally, a change to + create_outgoing_sdp_stream() skips a codec when it does not have + a payload code instead of the call failing. (closes issue + ASTERISK-23018) Reported by: xrobau Review: + https://reviewboard.asterisk.org/r/3131/ ........ Merged + revisions 405875 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/http.c: http: supported chunked Transfer-Encoding This + change implements support for HTTP Transfer-Encoding chunked in + both JSON and Form (post vars) body content. A new function + ast_http_get_contents() handles both regular and chunked mode + body, returning after the entire body is received. (closes issue + ASTERISK-23068) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3125/ ........ Merged + revisions 405861 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 18:55 +0000 [r405778-405844] Rusty Newton <rnewton@digium.com> + + * res/res_pjsip.c, /: Fixing some XML syntax issues with my + previous commit at r405777 for ASTERISK-23071 ........ Merged + revisions 405843 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c, doc/asterisk.8, main/features.c, + configs/sip.conf.sample, apps/app_queue.c, apps/app_transfer.c, + channels/chan_iax2.c: Documentation: doc fixes across various + parts of the code for ASTERISK issues 23061,23028,23046,23027 + Fixes typos of "transfered" instead of "transferred" in various + code. Fixes incorrect gosub param help text for app_queue. Fixes + Asterisk man pages containing unquoted minus signs. Adds note + about the "textsupport" option in sip.conf.sample. (issue + ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) + (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes + issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue + ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis + Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine + (license 6561) hyphen.patch uploaded by Jeremy Laine (license + 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) + ........ Merged revisions 405791 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405792 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405829 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip.c, /: res_pjsip: enhance documentation for + mailboxes options, for both endpoints and aors Made documentation + more explicit as to the use of the both options. (issue + ASTERISK-23071) (closes issue ASTERISK-23071) Reported by: Matt + Jordan ........ Merged revisions 405777 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 14:17 +0000 [r405766] Walter Doekes <walter+asterisk@wjd.nu> + + * res/res_musiconhold.c, CHANGES: Enable wide band audio in + musiconhold streams. Review: + https://reviewboard.asterisk.org/r/3112/ + +2014-01-16 20:06 +0000 [r405747-405749] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip/pjsip_options.c, /: res_pjsip: AOR option + qualify_frequency not respected on startup If an endpoint had + previously dynamically registered a contact and the contact + information was successfully stored in astdb then upon restart + the qualify notifications would not be sent out if the + qualify_frequency was set. This was due to the fact that only + permanent contacts were being checked and scheduled for qualifies + on startup. Modified the code to check and schedule all + registered contacts at startup. (closes issue ASTERISK-23062) + Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3124/ ........ Merged + revisions 405748 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Originate doesn't abort on failed + format_cap allocation action_originate responds to the remote + system with an error when cap==NULL, but doesn't return (abort + the originate). Patched to return. (closes issue ASTERISK-23034) + Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded + by coreyfarrell (license 5909) ........ Merged revisions 405745 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + Merged revisions 405746 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-16 19:33 +0000 [r405744] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip.c: PJSIP: Fix outbound OPTIONS support When path + support was added and contacts were made available during request + creation and transmission, the code path used by outbound qualify + support was not modified correctly and was causing request + creation to fail. This ensures that outbound request creation + with only a contact and no dialog, endpoint, or uri can succeed + which restores qualify support. Reported by: gtjoseph Reported + by: kharwell ........ Merged revisions 405743 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-16 19:13 +0000 [r405644-405695] Kevin Harwell <kharwell@digium.com> + + * /, res/res_fax.c, configs/res_fax.conf.sample: res_fax: + check_modem_rate() returned incorrect rate for V.27 According to + the new standard for V.27 and V.32 they are able to transmit at a + bit rate of 4,800 or 9,600. The check_mode_rate function needed + to be updated to reflect this. Also, because of this change the + default 'minrate' value was updated to be 4800. (closes issue + ASTERISK-22790) Reported by: Paolo Compagnini Patches: + res_fax.txt uploaded by looserouting (license 6548) ........ + Merged revisions 405656 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405693 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405694 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_pjsip.c: chan_pjsip: initial device state on + endpoints is INVALID When endpoints get loaded their device state + gets set to 'INVALID' because the channel driver has not been + loaded yet. Fixed by updating the device state for every endpoint + upon load of the channel driver. (closes issue ASTERISK-23065) + Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3123/ ........ Merged + revisions 405643 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-15 16:51 +0000 [r405586-405589] Jonathan Rose <jrose@digium.com> + + * CHANGES: Make 12 - 12.1 CHANGES log the same as in 12 + + * CHANGES, /: Include CHANGES info for r405553 ........ Merged + revisions 405585 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-15 16:36 +0000 [r405584] Joshua Colp <jcolp@digium.com> + + * /, cel/cel_manager.c: cel_manager: Don't crash if configuration + file is invalid. The cel_manager module did not properly handle + the case where the configuration file was invalid. The module + will now output a warning message and disable itself if this + occurs. Reported by: Bryan Walters ........ Merged revisions + 405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 405582 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405583 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-15 13:16 +0000 [r405566] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c, + res/res_pjsip_path.c (added), res/res_pjsip_mwi.c, + res/res_pjsip/pjsip_distributor.c, res/res_pjsip_diversion.c, + channels/chan_pjsip.c, res/res_pjsip_registrar.c, + res/res_pjsip_refer.c, include/asterisk/res_pjsip.h, + include/asterisk/res_pjsip_session.h, res/res_pjsip_notify.c, /, + res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c, + res/res_pjsip_t38.c, res/res_pjsip.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip_nat.c, + res/res_pjsip_session.c, + contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py + (added), res/res_pjsip_header_funcs.c: PJSIP: Add Path header + support This adds Path support to chan_pjsip in res_pjsip_path.c + with minimal additions in res_pjsip_registrar.c to store the path + and additions in res_pjsip_outbound_registration.c to enable + advertisement of path support to registrars and intervening + proxies. Path information is stored on contacts and is enabled + via Address of Record (AoRs) and Registration configuration + sections. While adding path support, it became necessary to be + able to add SIP supplements that handled messages outside of + sessions, so a framework for handling these types of hooks was + added in parallel to the already-existing session supplements and + several senders of out-of-dialog requests were refactored as a + result. (closes issue ASTERISK-21084) Review: + https://reviewboard.asterisk.org/r/3050/ ........ Merged + revisions 405565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 23:44 +0000 [r405554] Jonathan Rose <jrose@digium.com> + + * res/res_stasis_mailbox.exports.in (added), + res/ari/ari_model_validators.h, rest-api/api-docs/mailboxes.json + (added), include/asterisk/stasis_app_mailbox.h (added), + res/ari/resource_mailboxes.c (added), /, res/ari.make, + res/res_ari_mailboxes.c (added), res/ari/resource_mailboxes.h + (added), res/res_stasis_mailbox.c (added), + rest-api/resources.json, res/ari/ari_model_validators.c: ARI: Add + mailboxes resource for controlling and polling external MWI Adds + the following AMI commands: PUT mailboxes/mailboxName modifies + mailbox state and implicitly creates new mailboxes GET + mailboxes/mailboxName retrieves a JSON representation of a single + mailbox if it exists GET mailboxes retrieves a JSON array of all + mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that + res_mwi_external must be loaded for these functions to actually + do anything. Review: https://reviewboard.asterisk.org/r/3117/ + ........ Merged revisions 405553 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 21:46 +0000 [r405542] Richard Mudgett <rmudgett@digium.com> + + * main/strings.c, /: string container: Remove unnecessary RAII_VAR + usage and string object lock. ........ Merged revisions 405541 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 18:15 +0000 [r405437] Scott Griepentrog <sgriepentrog@digium.com> + + * /, channels/chan_sip.c: chan_sip: fix Local From tag on outbound + register regression In ASTERISK-12117, an improvement to insure + consistant local from tags on outbound registrations resulted in + an undesirable behavior - caused by leftover unexpired sip_pvt + dialogs (with the previous cseq number), resulting in many + uncessary REGISTER requests. Instead of significant rework of + transmit_register(), this change deletes the dialogs after a 200 + OK response indiciating a successful registration, keeping the + old dialogs from interfering with normal operation. (closes issue + ASTERISK-22946) Reported by: Stephan Eisvogel Review: + https://reviewboard.asterisk.org/r/3109/ ........ Merged + revisions 405433 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405434 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405435 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 18:14 +0000 [r405436] Richard Mudgett <rmudgett@digium.com> + + * apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample, + main/cli.c, include/asterisk/logger.h, main/pbx.c, + main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c, + main/logger.c, UPGRADE.txt: verbosity: Fix performance of console + verbose messages. The per console verbose level feature as + previously implemented caused a large performance penalty. The + fix required some minor incompatibilities if the new rasterisk is + used to connect to an earlier version. If the new rasterisk + connects to an older Asterisk version then the root console + verbose level is always affected by the "core set verbose" + command of the remote console even though it may appear to only + affect the current console. If an older version of rasterisk + connects to the new version then the "core set verbose" command + will have no effect. * Fixed the verbose performance by not + generating a verbose message if nothing is going to use it and + then filtered any generated verbose messages before actually + sending them to the remote consoles. * Split the "core set debug" + and "core set verbose" CLI commands to remove the per module + verbose support that cannot work with the per console verbose + level. * Added a silent option to the "core set verbose" command. + * Fixed "core set debug off" tab completion. * Made "core show + settings" list the current console verbosity in addition to the + root console verbosity. * Changed the default verbose level of + the 'verbose' setting in the logger.conf [logfiles] section. The + default is now to once again follow the current root console + level. As a result, using the AMI Command action with "core set + verbose" could again set the root console verbose level and + affect the verbose level logged. (closes issue AST-1252) Reported + by: Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3114/ ........ Merged + revisions 405431 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405432 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 16:43 +0000 [r405420] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip/pjsip_distributor.c: Fix erroneous behavior when + sending auth rejection to artificial endpoint. We were not + including an authentication challenge when sending a 401 response + to unmatched endpoints. This was due to the conversion to use a + vector for authentication section names on an endpoint. The + vector for artificial endpoints was empty, resulting in the + challenge being sent back containing no challenges. This is + worked around by placing a bogus value in the artificial + endpoint's auth vector. This value is never looked up by + anything, since they instead will directly call + ast_sip_get_artificial_auth(). + +2014-01-14 03:27 +0000 [r405369] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Skinny: do not add call to missed + calls list if answered elsewhere. Patch updates skinny devices + with a SKINNY_CONNECTED callstate if an inbound ringing or + callwaiting call is answered elsewhere. ........ Merged revisions + 405367 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-13 13:34 +0000 [r405339] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip/pjsip_cli.c: res_pjsip: Fix CLI tab completion + issues This fixes several issues with the new res_pjsip CLI tab + completion such as output of headers during tab completion and + being able to tab-complete more items than the code actually + handled (further items would simply be ignored). (closes issue + ASTERISK-23081) Review: https://reviewboard.asterisk.org/r/3115/ + Reported by: xrobau ........ Merged revisions 405338 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-12 22:24 +0000 [r405326] Joshua Colp <jcolp@digium.com> + + * res/ari/resource_playbacks.c, res/ari/resource_channels.c, + include/asterisk/ari.h, res/ari/resource_bridges.c, + res/ari/resource_recordings.c, res/ari/resource_device_states.c, + res/res_ari.c, res/ari/resource_endpoints.c, /, + res/ari/resource_applications.c: res_ari: Fix various memory + leaks. This change fixes a few memory leaks that were found based + on a mailing list post. 1. Some JSON response messages were never + freed. This was caused by the documentation stating that message + references were stolen when in reality they were not. The code + now follows the documentation and usage has been updated. 2. HTTP + response headers were never freed. 3. The variable list for + wildcards paths was never freed. (closes issue ASTERISK-23128) + Reported by: Kenneth Watson (on list) Review: + https://reviewboard.asterisk.org/r/3119/ ........ Merged + revisions 405325 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-12 22:13 +0000 [r405313-405314] Matthew Jordan <mjordan@digium.com> + + * apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h, + apps/app_cdr.c, main/cdr.c: CDRs: Synchronize dialplan + applications that manipulate CDRs with the engine In + https://reviewboard.asterisk.org/r/3057/, applications and + functions that manipulate CDRs were made to interact over Stasis. + This was done to synchronize manipulations of CDRs from the + dialplan with the updates the engine itself receives over the + message bus. This change rested on a faulty premise: that + messages published to the CDR topic or to a topic that forwards + to the CDR topic are synchronized with the messages handled by + the CDR topic subscription in the CDR engine. This is not the + case. There is no ordering guaranteed for two messages published + to the same topic; ordering is only guaranteed if a message is + published to the same subscriber. Stasis was modified in r405311 + to allow a publisher to synchronize on the subscriber. This patch + uses that API to synchronize the CDR publishers with the CDR + engine message router, which maintains the overall topic + subscription. (closes issue ASTERISK-22884) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ + Merged revisions 405312 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis.c, main/stasis_message_router.c, /, + include/asterisk/stasis.h, + include/asterisk/stasis_message_router.h, tests/test_stasis.c: + stasis: Add methods to allow for synchronous publishing to + subscriber This patch adds an API call to Stasis that allows a + publisher to publish a stasis message that will not return until + a specific subscriber handles the message. Since a subscriber can + have their own forwarding topic which orders messages from many + topics, this allows a publisher who knows of that subscriber to + synchronize to that subscriber regardless of the forwarding + relationships between topics. This is of particular use for + dialplan applications that need to synchronize on a particular + subscriber's handling of a message. (issue ASTERISK-22884) + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3099/ ........ Merged + revisions 405311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-10 20:00 +0000 [r405299] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip/security_events.c: Print "<unknown>" for + artificial endpoint in PJSIP security events. Previously, this + printed a UUID, which was not very clear when dealing with an + artificial endpoint. Review: + https://reviewboard.asterisk.org/r/3113 ........ Merged revisions + 405298 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-10 18:17 +0000 [r405284] Richard Mudgett <rmudgett@digium.com> + + * /, main/logger.c: Logging callid: Fix some sizeof() references + per coding guidelines. ........ Merged revisions 405281 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405282 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 23:52 +0000 [r405270] Jonathan Rose <jrose@digium.com> + + * res/res_pjsip_session.c: PJSIP: Add unhold on reinvite without + SDP behavior Review: https://reviewboard.asterisk.org/r/3106/ + +2014-01-09 23:50 +0000 [r405269] Damien Wedhorn <voip@facts.com.au> + + * channels/chan_dahdi.c, /: Fix chan_dahdi copile issue in + dev-mode. Error "unused variable i in dahdi_create_channel_range" + when compiling in dev-mode. Small restructure to + dahdi_create_channel_range to move the for(x) loop and int i,x to + a block within the IFDEF. ........ Merged revisions 405268 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 23:39 +0000 [r405267] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip.c, /, res/res_pjsip_messaging.c: + res_pjsip_messaging: potential for field values in from/to + headers to be missing Added in ability to specify display name + format ("name" <sip:name@ipaddr:port>) for a given URI and made + sure it was fully propagated to the outgoing message. Also made + it so outoing messages in res_pjsip always send as "sip:". + (closes issue ASTERISK-22924) Reported by: Anthony Messina + Review: https://reviewboard.asterisk.org/r/3094/ ........ Merged + revisions 405266 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 20:34 +0000 [r405254] Kinsey Moore <kmoore@digium.com> + + * main/astobj2.c, res/res_pjsip_session.c, /, + include/asterisk/astobj2.h: astobj2: Correct ao2_iterator opacity + violations This corrects the ao2_iterator opacity violations in + res_pjsip_session.c by adding a global function to get the number + of elements inside the container hidden behind the iterator. + (closes issue ASTERISK-23053) Review: + https://reviewboard.asterisk.org/r/3111/ Reported by: Richard + Mudgett ........ Merged revisions 405253 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 16:52 +0000 [r405236] Kevin Harwell <kharwell@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fails to resume + WebRTC call from hold In ast_rtp_ice_start if the ice session + create check list failed, start check was never initiated and + ice_started was never set to true. Upon re-entering the function + (for instance, [un]hold) it would try to create the check list + again with duplicate remote candidates. Fixed so that if the + create check list fails the necessary data structures are + properly re-initialized for any subsequent retries. Note, it was + decided to not stop ice support (by calling ast_rtp_ice_stop) on + a check list failure because it possible things might still work. + However, a debug message was added to help with any future + troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis + Valentinavičius Patches: works_on_my_machine.patch uploaded by + xytis (license 6558) ........ Merged revisions 405234 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405235 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 15:50 +0000 [r405217] Matthew Jordan <mjordan@digium.com> + + * /, apps/app_confbridge.c, + apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix + crash caused when waitmarked/marked users leave together When + waitmarked users join a ConfBridge, the conference state is + transitioned from EMPTY -> INACTIVE. In this state, the users are + maintined in a waiting users list. When a marked user joins, the + ConfBridge conference transitions from INACTIVE -> MULTI_MARKED, + and all users are put onto the active list of users. This process + works correctly. When the marked user leaves, if they are the + last marked user, the MULTI_MARKED state does the following: (1) + It plays back a message to the bridge stating that the leader has + left the conference. This requires an unlocking of the bridge. + (2) It moves waitmarked users back to the waiting list (3) It + transitions to the appropriate state: in this case, INACTIVE + However, because it plays the prompt back to the bridge before + moving the users and before finishing the state transition, this + creates a race condition: with the bridge unlocked, waitmarked + users who leave the conference (or are kicked from it) can cause + a state transition of the bridge to another state before the + conference is transitioned to the INACTIVE state. This causes the + state machine to get a bit wonky, often leading to a crash when + the MULTI_MARKED state attempts to conclude its processing. This + patch fixes this problem: (1) It prevents kicked users from being + kicked again. That's just a nicety. (2) More importantly, it + fixes the race condition by only playing the prompt once the + state has transitioned correctly to INACTIVE. If waitmarked users + sneak out during the prompt being played, no harm no foul. + Review: https://reviewboard.asterisk.org/r/3108/ Note that the + patch committed here is essentially the same as uploaded by Simon + Moxon on ASTERISK-22740, with the addition of the double kick + prevention. (closes issue AST-1258) Reported by: Steve Pitts + (closes issue ASTERISK-22740) Reported by: Simon Moxon patches: + ASTERISK-22740.diff uploaded by Simon Moxon (license 6546) + ........ Merged revisions 405215 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405216 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 14:15 +0000 [r405163] Walter Doekes <walter+asterisk@wjd.nu> + + * /, apps/app_dumpchan.c: "Minimun" typo. ........ Merged revisions + 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 405161 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405162 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-08 17:23 +0000 [r405144] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip/security_events.c: Use proper case for checking + if digest authentication is used. ........ Merged revisions + 405131 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-08 16:34 +0000 [r405129-405130] Kinsey Moore <kmoore@digium.com> + + * /, configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support + for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is + available on newer operating systems. (closes issue + ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/ + Reported by: George Joseph Patch by: George Joseph ........ + Merged revisions 405090 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405091 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405124 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: Add the missing part of r400140 When the + patch to add retry-on-forbidden-response was committed, part of + the patch for chan_sip was not committed which caused the feature + to be entirely nonfunctional. This corrects the code in question. + (closes issue ASTERISK-17138) Review: + https://reviewboard.asterisk.org/r/2874 ........ Merged revisions + 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 405081 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405083 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-07 19:56 +0000 [r405020-405035] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_acl.c: res_pjsip_acl: Fix another case of + assuming a contact will always contain a URI. ........ Merged + revisions 405034 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_nat.c: res_pjsip_nat: Don't assume a Contact + header will always contain a URI. If the 'rewrite_contact' option + was enabled and a Contact header was received which contained a + '*' a crash would occur. This change makes the res_pjsip_nat + module ignore the Contact header if it contains only a '*'. + (closes issue ASTERISK-23101) Reported by: Matt Jordan ........ + Merged revisions 405019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-06 21:55 +0000 [r404953-405007] Richard Mudgett <rmudgett@digium.com> + + * apps/app_voicemail.c, /: app_voicemail: Explicitly set + defaultenabled=yes ........ Merged revisions 405006 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_mwi_external_ami.c (added): External MWI AMI support. + The external MWI AMI interface provides a thin wrapper around the + core external MWI resource. The resource adds the following AMI + actions: MWIGet, MWIDelete, and MWIUpdate. (closes issue AFS-46) + Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged + revisions 404954 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_mwi_external.c (added), configs/sorcery.conf.sample, + include/asterisk/res_mwi_external.h (added), + res/res_mwi_external.exports.in (added), apps/app_voicemail.c: + External MWI core support. * The core external MWI resource + provides for MWI message counts persistence using sorcery. With + sorcery, the user is able to configure which sorcery wizzard + backend to use if the default astdb is not desired. * The core + external MWI resoruce provides some debugging CLI commands + enabled by defining MWI_DEBUG_CLI. The debugging CLI commands + are: "mwi delete all", "mwi delete like <regex>", "mwi delete + mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi + show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new> + [<old>]]". (closes issue AFS-43) Review: + https://reviewboard.asterisk.org/r/3061/ ........ Merged + revisions 404952 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-05 16:01 +0000 [r404924-404936] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Don't assume that a registration + client will always exist. ........ Merged revisions 404935 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Create registration client in pj + thread. Depending on which threading was loading the outbound + registration it was possible for the registration client to be + allocated outside of a pj thread. This change moves the creation + inside the synchronous task where it is guaranteed it will occur + in a pj thread. Reported by: Rob Thomas ........ Merged revisions + 404923 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-04 10:52 +0000 [r404912] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * main/asterisk.c, /: asterisk.c: suppress live_dangerously warning + on rasterisk Even since the fixes of AST-2013-007, Asterisk + prints the following warning on startup if the user decided to + live dangerously: Privilege escalation protection disabled! See + https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This + message is intended for the logs and interactive startup. No need + for it to appear on a remote console. This commit removes it from + there. (closes issue ASTERISK-23084) Review: + https://reviewboard.asterisk.org/r/3101/ ........ Merged + revisions 404861 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404888 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404911 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 22:00 +0000 [r404860] Kevin Harwell <kharwell@digium.com> + + * cel/cel_pgsql.c, /: cel_pgsql: module not correctly reloading + Upon reload the module unconditionally "unloaded" the module + (freeing memory and setting pointers to NULL) and then when + attempting a "load" if the config file had not changed then + nothing would be reinitialized. By moving the "unload" to occur + conditionally (reload only) after an attempted configuration + load, but before module "loading" alleviates the issue. The + module now loads/unloads/reloads correctly. (closes issue + ASTERISK-22871) Reported by: Matteo ........ Merged revisions + 404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 404858 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404859 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 21:45 +0000 [r404844-404856] Matthew Jordan <mjordan@digium.com> + + * /, res/res_pjsip_logger.c: res_pjsip_logger: Add the + ASTERISK_FILE_VERSION macro Registering yourself with the + Asterisk core is the nice thing to do, even when you're a logging + module. ........ Merged revisions 404855 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_authenticator_digest.c, tests/test_utils.c: + res_pjsip_authenticator_digest: Fix md5 hash buffer An md5 hash + is 32 bytes long. The char buffer must be at least 33 bytes to + avoid clobbering of the stack. This patch also fixes a potential + clobbering in test_utils.c. Thanks to Andrew Nagy for reporting + and testing this out in #asterisk-dev Reported by: Andrew Nagy + Tested by: Andrew Nagy ........ Merged revisions 404843 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 20:02 +0000 [r404787-404832] Kevin Harwell <kharwell@digium.com> + + * main/manager.c: manager: UserEvent including action on output AMI + action UserEvent event response would include the action header + in its keyvalue pairs list. Adjusted the start of the header loop + to skip over the action part. (closes issue ASTERISK-22899) + Reported by: outtolunc Patches: + svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license + 5198) + + * channels/chan_dahdi.c, /: chan_dahdi: dahdi show channels slices + PRI channel dnid on output dahdi show channels output slices the + callerid (which is dnid copied over on PRI channels). If the + channel naming structures look like: 'DAHDI/i1/1408409XXXX-6' + then the output slices 1408409XXXX down to 1408409XXX. This patch + just opens it up to 15 chars so you can see the whole thing. + (closes issue ASTERISK-22918) Reported by: outtolunc Patches: + svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc + (license 5198) ........ Merged revisions 404784 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404785 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404786 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 18:33 +0000 [r404783] Richard Mudgett <rmudgett@digium.com> + + * tests/test_stasis.c, /: test_stasis.c: Fix ref leak in normal + execution path. ........ Merged revisions 404764 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 18:31 +0000 [r404782] Kevin Harwell <kharwell@digium.com> + + * /, apps/app_meetme.c: app_meetme: compiler warning Fixed a + compiler warning (errors in 'dev-mode') given by gcc version + 4.8.1. The one in app_meetme involved the + 'sizeof-pointer-memaccess' (see: + http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it + would no longer issue a warning and can compile again in + 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/ + ........ Merged revisions 404742 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404773 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404781 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 17:27 +0000 [r404726-404738] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip/pjsip_configuration.c, /, res/res_pjsip/location.c: + res_pjsip: Ensure more URI validation happens in pj threads. + ........ Merged revisions 404737 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Ensure URI validation happens in + a pjlib thread. This change moves outbound registration URI + validation into the task executed within a pjlib thread. Reported + by: Andrew Nagy ........ Merged revisions 404725 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-02 19:38 +0000 [r404677] Scott Griepentrog <sgriepentrog@digium.com> + + * /, funcs/func_strings.c: func_strings: use memmove to prevent + overlapping memory on strcpy When calling REPLACE() with an empty + replace-char argument, strcpy is used to overwrite the the + matching <find-char>. However as the src and dest arguments to + strcpy must not overlap, it causes other parts of the string to + be overwritten with adjacent characters and the result is + mangled. Patch replaces call to strcpy with memmove and adds a + test suite case for REPLACE. (closes issue ASTERISK-22910) + Reported by: Gareth Palmer Review: + https://reviewboard.asterisk.org/r/3083/ Patches: + func_strings.patch uploaded by Gareth Palmer (license 5169) + ........ Merged revisions 404674 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404675 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404676 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-02 19:08 +0000 [r404664] Kevin Harwell <kharwell@digium.com> + + * channels/chan_pjsip.c, include/asterisk/res_pjsip.h, /, + configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c, + CHANGES, res/res_pjsip.c: res_pjsip: add 'set_var' support on + endpoints Added a new 'set_var' option for ast_sip_endpoint(s). + For each variable specified that variable gets set upon creation + of a pjsip channel involving the endpoint. (closes issue + ASTERISK-22868) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/3095/ ........ Merged + revisions 404663 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-31 22:51 +0000 [r404620-404653] Joshua Colp <jcolp@digium.com> + + * channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip: + Handle hanging up before calling. Channel creation in Asterisk is + broken up into two steps: requesting and calling. In some cases a + channel may be requested but never called. This happens in the + ChanIsAvail dialplan application for determining if something is + reachable or not. The PJSIP channel driver did not take this + situation into account and attempted to end a session that was + never called out on. The code now checks the session state to + determine if the session has been called out on and if not + terminates it instead of ending it. (closes issue ASTERISK-23074) + Reported by: Kilburn ........ Merged revisions 404652 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_endpoint_identifier_ip.c: + res_pjsip_endpoint_identifier_ip: Accept hostnames in the 'match' + field. Hostnames specified in the 'match' field will be resolved + and all addresses returned. Each address will be added to the + endpoint identifier for the matching process. Reported by: Rob + Thomas ........ Merged revisions 404613 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-31 21:39 +0000 [r404606] Kevin Harwell <kharwell@digium.com> + + * cel/cel_pgsql.c, /: cel_pgsql: deadlock on unload and + core_event_dispatcher A deadlock can happen between a thread + unloading or reloading the cel_pgsql module and the + core_event_dispatcher taskprocessor thread. Description of what + is happening: Thread 1 (for example, a netconsole thread): a + "module reload cel_pgsql" is launched the thread enter the + "my_unload_module" function (cel_pgsql.c) the thread acquire the + write lock on psql_columns the thread enter the + "ast_event_unsubscribe" function (event.c) the thread try to + acquire the write lock on ast_event_subs[sub->type] Thread 2 + (core_event_dispatcher taskprocessor thread): the taskprocessor + pop a CEL event the thread enter the "handle_event" function + (event.c) the thread acquire the read lock on + ast_event_subs[sub->type] the thread callback the "pgsql_log" + function (cel_pgsql.c), since it's a subscriber of CEL events the + thread try to acquire a read lock on psql_columns (closes issue + ASTERISK-22854) Reported by: Etienne Lessard Patches: + cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license + 6394) ........ Merged revisions 404603 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404604 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404605 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-31 20:27 +0000 [r404593] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_outbound_registration.c, /: + res_pjsip_outbound_registration: Add validation for 'server_uri' + and 'client_uri'. When applying configuration for outbound + registrations the 'server_uri' and 'client_uri' fields were not + validated. The code will now confirm that they exist and that + they contain parseable SIP URIs. Reported by: Andrew Nagy + ........ Merged revisions 404592 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-30 23:25 +0000 [r404582] Kevin Harwell <kharwell@digium.com> + + * main/channel.c, /: channels.c: core show channeltypes slicing + 'core show channeltypes' type column is being sliced, resulting + in incomplete type names. (closes issue ASTERISK-22919) Reported + by: outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded + by outtolunc (license 5198) ........ Merged revisions 404579 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404581 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-24 17:12 +0000 [r404567-404569] David M. Lee <dlee@digium.com> + + * UPGRADE-12.txt, /: Added note to UPGRADE.txt about the default + value of live_dangerously changing ........ Merged revisions + 404568 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/http.c: http: Properly reject requests with + Transfer-Encoding set Asterisk does not support any of the + transfer encodings specified in HTTP/1.1, other than the default + "identity" encoding. According to RFC 2616: A server which + receives an entity-body with a transfer-coding it does not + understand SHOULD return 501 (Unimplemented), and close the + connection. A server MUST NOT send transfer-codings to an + HTTP/1.0 client. This patch adds the 501 Unimplemented response, + instead of the hard work of actually implementing other + recordings. This behavior is especially problematic for Node.js + clients, which use chunked encoding by default. (closes issue + ASTERISK-22486) Review: https://reviewboard.asterisk.org/r/3092/ + ........ Merged revisions 404565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-24 02:20 +0000 [r404554] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Ensure dialog + manipulation happens on proper thread. When destroying a + subscription we remove the serializer from its dialog and + decrease its reference count. Depending on which thread dropped + the subscription reference count to 0 it was possible for this to + occur in a thread where it is not possible. (closes issue + ASTERISK-22952) Reported by: Matt Jordan ........ Merged + revisions 404553 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-23 16:38 +0000 [r404542] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + UPGRADE-12.txt: chan_dahdi: enable ignore_failed_channels by + default If ignore_failed_channels is set to "true" for a channel, + the channel will continue to be configured even if configuring it + has failed. This allows Asterisk to start before all the DAHDI + initialization is done and thus not force the starting order + dahdi -> asterisk. Review: + https://reviewboard.asterisk.org/r/3063/ + +2013-12-21 03:35 +0000 [r404532] Matthew Jordan <mjordan@digium.com> + + * /, res/res_pjsip/pjsip_cli.c: res_pjsip/pjsip_cli: fix + compilation error caused by passing ast_free When wanting to pass + *free as a function pointer, ast_free_ptr has to be used instead + of ast_free. This allows it to be compiled with MALLOC_DEBUG + enabled. ........ Merged revisions 404531 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 22:04 +0000 [r404511-404512] David M. Lee <dlee@digium.com> + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /, + rest-api/api-docs/applications.json: ari: Remove support for + specifying channel vars during origination. When we added support + for specifying channel variables for an origination, we didn't + consider how that would interact with another feature, namely + specifying request parameters in a JSON request body. The method + of specifying channel variables (as a flat JSON object passed in + the JSON body) interferes with parsing parameters out of the + request body. Unfortunately, fixing this would be a backward + incompatible change. In the interest of keeping the API sane and + keeping our release schedule, we're dropping the feature for + specifying channel variables in the origination request. We will + bring the feature back soon, as a backward compatible addition to + the API. (closes issue ASTERISK-23051) Review: + https://reviewboard.asterisk.org/r/3088 ........ Merged revisions + 404509 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Remove automerge properties ........ Merged revisions 404488 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 21:32 +0000 [r404507] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/config.h, main/config.c, main/channel.c, + res/res_pjsip/location.c, include/asterisk/res_pjsip_cli.h + (added), res/res_pjsip/pjsip_cli.c (added), + include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip_registrar.c, main/sorcery.c, + include/asterisk/res_pjsip.h, CREDITS, + res/res_pjsip/config_auth.c, /, + res/res_pjsip_endpoint_identifier_ip.c: res_pjsip: Add PJSIP CLI + commands Implements the following cli commands: pjsip list aors + pjsip list auths pjsip list channels pjsip list contacts pjsip + list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show + channels pjsip show endpoint(s) Also... Minor modifications made + to the AMI command implementations to facilitate reuse. New + function ast_variable_list_sort added to config.c and config.h to + implement variable list sorting. (issue ASTERISK-22610) patches: + pjsip_cli_v2.patch uploaded by george.joseph (License 6322) + ........ Merged revisions 404480 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 21:18 +0000 [r404461] Scott Griepentrog <sgriepentrog@digium.com> + + * /, main/say.c: say.c: correct time for polish In + ast_say_date_with_format_pl(), change ast_say_number() to use + tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported + by: Robert Mordec Review: + https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch + uploaded by veilen (license 6555) ........ Merged revisions + 404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 404457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 20:28 +0000 [r404452] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_refer.c: Fix issue where PJSIP blind transferer + dialog may not complete as planned. When transferring to a + dialplan extension that will not place any outbound calls, the + only control frames that the PJSIP REFER framehook will receive + are inconsequential (such as unhold or srcchange). As such, we + shouldn't allow for the reception of those types of frames + prevent us from signaling to the transferring party that the + transfer has completed successfully once voice frames are read. + Thanks to Jonathan Rose for pointing this out. ........ Merged + revisions 404439 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 20:05 +0000 [r404438] Matthew Jordan <mjordan@digium.com> + + * /, res/ari/resource_applications.h, + res/res_stasis_device_state.c: res_stasis_device_state: Set + resource type for subscriptions to deviceState The documentation + for ARI already specifies that the device state resource when + used for subscribing for events is "deviceState", not + "device_state". The code, however, used "device_state"; although + this was inconsistent as well in doxygen comments in + resource_applications. Because the actual resource being + subscribed to is /deviceStates/{device}/, it makes sense for the + resource type specifier to be deviceState. Note that the key + value in the events is still "device_state". ........ Merged + revisions 404437 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 20:00 +0000 [r404436] Richard Mudgett <rmudgett@digium.com> + + * res/ari/resource_channels.c, tests/test_scoped_lock.c, + tests/test_stasis.c, res/parking/parking_manager.c, + res/ari/resource_bridges.c, res/ari/resource_endpoints.c, /, + res/res_pjsip/location.c, tests/test_cel.c: ao2_iterator: + Mini-audit of the ao2_iterator loops in the new code files. * + Fixed several places where ao2_iterator_destroy() was not called. + * Fixed several iterator loop object variable reference problems. + * Fixed res_parking AMI actions returning non-zero. Only the AMI + logoff action can return non-zero. Review: + https://reviewboard.asterisk.org/r/3087/ ........ Merged + revisions 404434 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 19:25 +0000 [r404433] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/manager.h, /: manager: bump version to 2.0.0 AMI + has received substantial updates over the past year. Not only has + the syntax been vastly improved and made consistent (which + entails many event changes), but the underlying things that those + events convey have changed substantially as well. After some + conversation in #asterisk-dev, it was agreed that this is a good + time to jump to 2. At the same time, since ARI will most likely + use semantic versioning, we might as well use that for AMI as + well. That also affords us greater meaning for the AMI version. + ........ Merged revisions 404421 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 19:06 +0000 [r404420] Richard Mudgett <rmudgett@digium.com> + + * /, main/sounds_index.c: Whitespace fixes. ........ Merged + revisions 404419 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 17:22 +0000 [r404406] Rusty Newton <rnewton@digium.com> + + * /, configs/pjsip.conf.sample: Documentation: Updates for info + about NAT-related settings and fixes for pjsip.conf.sample Added + another NAT example to pjsip.conf.sample. We had a few mentions + of NAT configuration throughout the sample, but I added another + for a little bit more clarity. Additionally many pjsip options + were affected by the change to snake case, so I fixed any + instances of those options in pjsip.conf. I regenerated the + config option list (at the bottom of the file) from a new xml + config doc dump, so all the snake case changes should be + reflected there, as well as any other changes to those options. + (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/ + ........ Merged revisions 404405 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 20:48 +0000 [r404387] Scott Griepentrog <sgriepentrog@digium.com> + + * main/security_events.c: security_events: log events with + descriptive names This patch updates the log messages to include + descriptive names for event types. This is an improvement over + having only cryptic type numbers. (closes issue ASTERISK-22909) + Reported by: outtolunc Review: + https://reviewboard.asterisk.org/r/3081/ Patches: + svn_security_events.c.names.diff.txt uploaded by outtolunc + (license 5198) + +2013-12-19 18:16 +0000 [r404376] Richard Mudgett <rmudgett@digium.com> + + * /, CHANGES: Put notice in CHANGES as well as UPGRADE.txt. + ........ Merged revisions 404375 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 18:00 +0000 [r404370-404372] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip/pjsip_outbound_auth.c, /: res_pjsip: Ignore 401/407 + responses for transactions and dialogs we don't know about. Under + normal conditions it is unlikely we will ever receive a response + for a transaction or dialog we don't know about but if any are + received ignore them. ........ Merged revisions 404371 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_session.c: res_pjsip_session: Fix SDP + negotiation when resending an INVITE with authentication. The + process for resending an INVITE with authentication involves + restarting the UAC session. We were incorrectly passing in that a + new offer is being sent, causing the SDP negotiation to get into + a (technically speaking) funky state. ........ Merged revisions + 404369 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 17:45 +0000 [r404368] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/channel.h, res/res_pjsip.c, main/channel.c, /, + include/asterisk/autochan.h: Fix a deadlock that occurred due to + a conflict of masquerades. For the explanation, here is a + copy-paste of the review board explanation: Initially, it was + discovered that performing an attended transfer of a multiparty + bridge with a PJSIP channel would cause a deadlock. A PBX thread + started a masquerade and reached the point where it was calling + the fixup() callback on the "original" channel. For chan_pjsip, + this involves pushing a synchronous task to the session's + serializer. The problem was that a task ahead of the fixup task + was also attempting to perform a channel masquerade. However, + since masquerades are designed in a way to only allow for one to + occur at a time, the task ahead of the fixup could not continue + until the masquerade already in progress had completed. And of + course, the masquerade in progress could not complete until the + task ahead of the fixup task had completed. Deadlock. The initial + fix was to change the fixup task to be asynchronous. While this + prevented the deadlock from occurring, it had the frightful side + effect of potentially allowing for tasks in the session's + serializer to operate on a zombie channel. Taking a step back + from this particular deadlock, it became clear that the problem + was not really this one particular issue but that masquerades + themselves needed to be addressed. A PJSIP attended transfer + operation calls ast_channel_move(), which attempts to both set up + and execute a masquerade. The problem was that after it had set + up the masquerade, the PBX thread had swooped in and tried to + actually perform the masquerade. Looking at changes that had been + made to Asterisk 12, it became clear that there never is any time + now that anyone ever wants to set up a masquerade and allow for + the channel thread to actually perform the masquerade. Everyone + always is calling ast_channel_move(), performs the masquerade + itself before returning. In this patch, I have removed all blocks + of code from channel.c that will attempt to perform a masquerade + if ast_channel_masq() returns true. Now, there is no distinction + between setting up a masquerade and performing the masquerade. It + is one operation. The only remaining checks for + ast_channel_masq() and ast_channel_masqr() are in ast_hangup() + since we do not want to interrupt a masquerade by hanging up the + channel. Instead, now ast_hangup() will wait for a masquerade to + complete before moving forward with its operation. The + ast_channel_move() function has been modified to basically + in-line the logic that used to be in ast_channel_masquerade(). + ast_channel_masquerade() has been killed off for real. + ast_channel_move() now has a lock associated with it that is used + to prevent any simultaneous moves from occurring at once. This + means there is no need to make sure that ast_channel_masq() or + ast_channel_masqr() are already set on a channel when + ast_channel_move() is called. It also means the channel container + lock is not pulling double duty by both keeping the container + locked and preventing multiple masquerades from occurring + simultaneously. The ast_do_masquerade() function has been renamed + to do_channel_masquerade() and is now internal to channel.c. The + function now takes explicit arguments of which channels are + involved in the masquerade instead of a single channel. While it + probably is possible to do some further refactoring of this + method, I feel that I would be treading dangerously. Instead, all + I did was change some comments that no longer are true after this + changeset. The other more minor change introduced in this patch + is to res_pjsip.c to make ast_sip_push_task_synchronous() run the + task in-place if we are already a SIP servant thread. This is + related to this patch because even when we isolate the channel + masquerade to only running in the SIP servant thread, we would + still deadlock when the fixup() callback is reached since we + would essentially be waiting forever for ourselves to finish + before actually running the fixup. This makes it so the fixup is + run without having to push a task into a serializer at all. + (closes issue ASTERISK-22936) Reported by Jonathan Rose Review: + https://reviewboard.asterisk.org/r/3069 ........ Merged revisions + 404356 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 17:13 +0000 [r404355] Richard Mudgett <rmudgett@digium.com> + + * main/udptl.c, addons/chan_ooh323.c, /, channels/chan_sip.c, + include/asterisk/udptl.h: udptl: Dead code elimination. + ast_udptl_bridge was not used. Removing dead code starting with + ast_udptl_bridge() eliminated the code in this change. Note: This + code has actually been dead since Asterisk v1.4 when it was first + put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ + Merged revisions 404354 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 17:03 +0000 [r404353] Scott Griepentrog <sgriepentrog@digium.com> + + * /, res/res_fax.c: res_fax.c: crash on framehook with no dsp in + fax detect In fax_detect_framehook() a null pointer reference can + occur where a voice frame is processed but no dsp is attached to + the fax detection structure. The code block that rejects frames + that detection cannot be processed on is checking for dsp but + falls through when it should instead return, as this change + implements. (closes issue ASTERISK-22942) Reported by: adomjan + Review: https://reviewboard.asterisk.org/r/3076/ ........ Merged + revisions 404351 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404352 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 16:52 +0000 [r404350] Richard Mudgett <rmudgett@digium.com> + + * configs/skinny.conf.sample, res/res_xmpp.c, res/res_jabber.c, + CHANGES, channels/chan_iax2.c, channels/sig_pri.c, + channels/h323/chan_h323.h, configs/iax.conf.sample, + channels/sig_pri.h, channels/chan_dahdi.c, + include/asterisk/app.h, channels/chan_skinny.c, + channels/chan_dahdi.h, channels/chan_h323.c, main/app.c, + UPGRADE-12.txt, configs/sip.conf.sample, + channels/sip/include/sip.h, channels/chan_mgcp.c, + apps/app_voicemail.c, channels/chan_unistim.c, + configs/chan_dahdi.conf.sample, /, channels/chan_sip.c, + configs/voicemail.conf.sample, funcs/func_vmcount.c: Voicemail: + Remove mailbox identifier format (box@context) assumptions in the + system. This change is in preparation for external MWI support. + Removed code from the system for normal mailbox handling that + appends @default to the mailbox identifier if it does not have a + context. The only exception is the legacy hasvoicemail users.conf + option. The legacy option will only work for app_voicemail + mailboxes. The system cannot make any assumptions about the + format of the mailbox identifer used by app_voicemail. chan_sip + and chan_dahdi/sig_pri had the most changes because they both + tried to interpret the mailbox identifier. chan_sip just stored + and compared the two components. chan_dahdi actually used the box + information. The ISDN MWI support configuration options had to be + reworked because chan_dahdi was parsing the box@context format to + get the box number. As a result the mwi_vm_boxes chan_dahdi.conf + option was added and is documented in the chan_dahdi.conf.sample + file. Review: https://reviewboard.asterisk.org/r/3072/ ........ + Merged revisions 404348 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 16:33 +0000 [r404346] Scott Griepentrog <sgriepentrog@digium.com> + + * main/db.c, /: astdb: crash in sqlite3 during shutdown When + Asterisk is shut down, the astdb_atexit() function releases + (finalize) the previously initiated (prepared) SQL statements in + sqlite3. Another thread making a subsequent request can cause a + crash in sqlite3. This patch eliminates that issue by resetting + the statement pointer after it is released/cleared. The sqlite3 + code detects the null pointer, and aborts the operation cleanly. + (closes issue AST-1265) Reported by: Alexander Hömig (closes + issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter + Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged + revisions 404344 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404345 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 12:18 +0000 [r404333] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: channel: Add a missing ast_channel_unlock when + allocating a Surrogate channel. ........ Merged revisions 404332 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 08:35 +0000 [r404321] Alexandr Anikin <may@telecom-service.ru> + + * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooGkClient.c, + addons/chan_ooh323.c, /, addons/ooh323c/src/ooGkClient.h: Handle + temporary failures on gk registration Introduce new 'stopped' + state for gk client and restart gk client on failures Remove + ooh323 stack command lock as it is not need now. (closes issue + ASTERISK-21960) Reported by: Dmitry Melekhov Patches: + ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested + by: Dmitry Melekhov ........ Merged revisions 404318 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404320 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 02:59 +0000 [r404307] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Fixup some skinny bugs causing Fracks + and ao2 cleanup issues. Moved channel locking into setsubstate so + that a process can complete working on a sub before another + starts changing it. The existing code was causing some Fracks + with schedule deletion. Removed multiple rtp cleanup. Now only + cleansup up once, fixing ao2 object cleanup issues. ........ + Merged revisions 404306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 00:50 +0000 [r404295] Matthew Jordan <mjordan@digium.com> + + * include/asterisk/cdr.h, CHANGES, apps/app_cdr.c, main/cdr.c, + apps/app_forkcdr.c, main/pbx.c, /, funcs/func_cdr.c, + apps/app_disa.c, UPGRADE-12.txt: app_cdr,app_forkcdr,func_cdr: + Synchronize with engine when manipulating state When doing the + rework of the CDR engine that pushed all of the logic into cdr.c + and made it respond to changes in channel state over Stasis, we + knew that accessing the CDR engine from the dialplan would be + "slightly" non-deterministic. Dialplan threads would be accessing + CDRs while Stasis threads would be updating the state of said + CDRs - whereas in the past, everything happened on the dialplan + threads. Tests have shown that "slightly" is in reality "very". + This patch synchronizes things by making the dialplan + applications/functions that manipulate CDRs do so over Stasis. + ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to + send their requests over to the CDR engine, and synchronize on + the channel Stasis topic via a subscription so that they return + their values/control to the dialplan at the appropriate time. + While going through this, the following changes were also made: * + DISA, which can reset the CDR when a user successfully + authenticates, now just uses the ResetCDR app to do this. This + prevents having to duplicate the same Stasis synchronization + logic in that application. * Answer no longer disables CDRs. It + actually didn't work anyway - calling DISABLE on the channel's + CDR doesn't stop the CDR from getting the Answer time - it just + kills all CDRs on that channel, which isn't what the caller would + intend. (closes issue ASTERISK-22884) (closes issue + ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/ + ........ Merged revisions 404294 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 00:32 +0000 [r404293] Damien Wedhorn <voip@facts.com.au> + + * /, channels/chan_skinny.c: Fixup skinny registration following + network issues. On session registration, if device is already + reporting that it is connected to a device, an innocuous packet + (update time) is sent to the already connected device. If the tcp + connection is down, the device will be unregistered and the new + connection allowed. Without this patch, network issues can see a + situation where a device can not reregister until after + 3*timeout. ........ Merged revisions 404292 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 23:00 +0000 [r404280] Jason Parker <jparker@digium.com> + + * main/manager.c, /: Add AMI event for presence state. Review: + https://reviewboard.asterisk.org/r/3039/ ........ Merged + revisions 404275 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404279 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 21:12 +0000 [r404264] Richard Mudgett <rmudgett@digium.com> + + * addons/ooh323c/src/ooTimer.c, /: ooh323c: Fix gcc 4.6.3 compiler + warnings. ........ Merged revisions 404212 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404219 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404263 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 20:48 +0000 [r404260-404262] Kevin Harwell <kharwell@digium.com> + + * channels/chan_oss.c, /: chan_oss.c: channel being locked twice + and unlocked once Removed channel lock as it is now being down in + ast_channel_alloc ........ Merged revisions 404261 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c, + addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c, + channels/chan_pjsip.c, res/parking/parking_manager.c, + channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c, + funcs/func_timeout.c, /, apps/app_meetme.c, main/bridge.c, + tests/test_stasis_channels.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/sig_pri.c, apps/app_queue.c, + main/cel.c, main/stasis_bridges.c, channels/chan_jingle.c, + channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c, + channels/sig_analog.c, include/asterisk/stasis_channels.h, + res/res_agi.c, channels/chan_motif.c, tests/test_cel.c, + apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c, + apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc, + addons/chan_ooh323.c, main/pickup.c, include/asterisk/aoc.h, + include/asterisk/stasis_bridges.h, apps/app_userevent.c, + apps/app_disa.c, channels/chan_console.c, + include/asterisk/channelstate.h, main/core_local.c, + channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c, + res/parking/parking_bridge_features.c, apps/app_agent_pool.c, + main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c: + channel locking: Add locking for channel snapshot creation + Original commit message by mmichelson (asterisk 12 r403311): + "This adds channel locks around calls to create channel snapshots + as well as other functions which operate on a channel and then + end up creating a channel snapshot. Functions that expect the + channel to be locked prior to being called have had their + documentation updated to indicate such." The above was initially + committed and then reverted at r403398. The problem was found to + be in core_local.c in the publish_local_bridge_message function. + The ast_unreal_lock_all function locks and adds a reference to + the returned channels and while they were being unlocked they + were not being unreffed when no longer needed. Fixed by unreffing + the channels. Also in bridge.c a lock was obtained on + "other->chan", but then an attempt was made to unlock "other" and + not the previously locked channel. Fixed by unlocking + "other->chan" (closes issue ASTERISK-22709) Reported by: John + Bigelow ........ Merged revisions 404237 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 19:36 +0000 [r404211] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c, configs/ooh323.conf.sample: Introduce new + config option 'aniasdni'. If yes then asterisk set dialed number + as own id back to the caller on incoming h.323 calls. Option can + be set globally or per user section. (closes issue + ASTERISK-22020) Reported by: Ross Beer + +2013-12-18 19:28 +0000 [r404210] Joshua Colp <jcolp@digium.com> + + * channels/chan_mgcp.c, main/pbx.c, channels/chan_sip.c, + apps/confbridge/conf_chan_record.c, tests/test_app.c, + tests/test_stasis_channels.c, main/core_unreal.c, + include/asterisk/channel.h, channels/chan_console.c, + channels/chan_oss.c, channels/chan_jingle.c, + channels/chan_misdn.c, channels/chan_h323.c, tests/test_cel.c, + channels/chan_nbs.c, channels/chan_pjsip.c, res/res_calendar.c, + apps/app_voicemail.c, channels/chan_unistim.c, + tests/test_substitution.c, channels/chan_vpb.cc, + addons/chan_ooh323.c, channels/chan_multicast_rtp.c, /, + apps/app_meetme.c, res/res_stasis_snoop.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c, + channels/chan_phone.c, channels/chan_skinny.c, + res/parking/parking_tests.c, channels/chan_motif.c, + tests/test_voicemail_api.c, channels/chan_alsa.c, main/message.c, + addons/chan_mobile.c, tests/test_cdr.c: channels: Return + allocated channels locked. This change makes ast_channel_alloc + return allocated channels locked. By doing so no other thread can + acquire, lock, and manipulate the channel before it is completely + set up. (closes issue AST-1256) Review: + https://reviewboard.asterisk.org/r/3067/ ........ Merged + revisions 404204 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 19:10 +0000 [r404198] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c: Implement module reload command for + chan_ooh323 (close issue ASTERISK-22817) Patches: + ooh323_module_reload.patch + +2013-12-18 12:46 +0000 [r404185] Matthew Jordan <mjordan@digium.com> + + * rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json: ari: Bump the version of ARI + to 1.0.0 (closes issue ASTERISK-23007) ........ Merged revisions + 404184 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 12:01 +0000 [r404138] Joshua Colp <jcolp@digium.com> + + * res/res_calendar.c, /: res_calendar: Protect channel when adding + datastore. This change adds a missing channel lock when adding a + datastore to a channel. ........ Merged revisions 404135 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404136 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404137 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 00:36 +0000 [r404100] Rusty Newton <rnewton@digium.com> + + * /, funcs/func_strings.c: func_strings: Documentation fix for + QUOTE() Example output was inaccurate. (issue ASTERISK-22970) + (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches: + func_strings.patch uploaded by Gareth Palmer (license 5169) + ........ Merged revisions 404081 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404087 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404099 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 00:17 +0000 [r404051] Matthew Jordan <mjordan@digium.com> + + * /, LICENSE: LICENSE: Update language to include ARI ........ + Merged revisions 404050 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 23:57 +0000 [r404049] Jonathan Rose <jrose@digium.com> + + * /, tests/test_cel.c, tests/test_cdr.c: tests: fix + ast_bridge_base_new calls not using the additional arguments + r404042 gave ast_bridge_base_new two new arguments for setting a + bridge creator and name. Unfortunately since a couple test + modules aren't compiled by default, I missed the fact that this + change impacted those tests and caused compilation failures + against them. ........ Merged revisions 404048 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 23:38 +0000 [r404047] Rusty Newton <rnewton@digium.com> + + * include/asterisk/test.h, main/channel.c, main/rtp_engine.c, /, + channels/chan_iax2.c, apps/app_chanspy.c, apps/app_mixmonitor.c: + Several components: fixing Typos in comments and code, + "avaliable" instead of "available" (issue ASTERISK-23021) (closes + issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty + Newton Patches: available.patch uploaded by Jeremy Lainé (license + 6561) ........ Merged revisions 404046 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 23:25 +0000 [r404043] Jonathan Rose <jrose@digium.com> + + * apps/app_bridgewait.c, res/ari/ari_model_validators.c, + doc/appdocsxml.xslt, main/stasis_bridges.c, + rest-api/api-docs/bridges.json, res/ari/resource_bridges.c, + apps/app_agent_pool.c, res/parking/parking_bridge.c, + res/ari/ari_model_validators.h, main/manager_bridges.c, + res/ari/resource_bridges.h, include/asterisk/bridge_internal.h, + apps/app_confbridge.c, res/res_stasis.c, + include/asterisk/bridge.h, res/res_ari_bridges.c, /, + main/bridge.c, main/bridge_basic.c, + include/asterisk/stasis_bridges.h, include/asterisk/stasis_app.h: + bridging: Give bridges a name and a known creator Bridges have + two new optional properties, a creator and a name. Certain + consumers of bridges will automatically provide bridges that they + create with these properties. Examples include app_bridgewait, + res_parking, app_confbridge, and app_agent_pool. In addition, a + name may now be provided as an argument to the POST function for + creating new bridges via ARI. (closes issue AFS-47) Review: + https://reviewboard.asterisk.org/r/3070/ ........ Merged + revisions 404042 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 18:35 +0000 [r404028-404030] Joshua Colp <jcolp@digium.com> + + * res/res_sorcery_config.c, /: res_sorcery_config: Output an error + message when an object can't be created. If object creation fails + an error message will now be output with the id, type, and + configuration file. ........ Merged revisions 404029 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/framehook.c: framehooks: Re-iterate if framehook provides + different frame. Framehooks can be used in a reactive manner to + execute specific logic when a frame is received with a certain + type and payload. Since it is possible for framehooks to provide + frames it was possible for this reactive framehook to be unaware + of frames it is looking for. This change makes it so that when + framehooks return a modified frame the code will now re-iterate + (from the beginning) and call any previous framehooks that have + not provided a modified frame themselves. Review: + https://reviewboard.asterisk.org/r/3046/ ........ Merged + revisions 404027 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 14:41 +0000 [r404008-404009] David M. Lee <dlee@digium.com> + + * /, configs/asterisk.conf.sample, main/asterisk.c: Changed the + default for live_dangerously to no ........ Merged revisions + 404006 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/pjsip, /: Setting svn:ignore ........ Merged revisions + 403748 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 12:59 +0000 [r403994] Matthew Jordan <mjordan@digium.com> + + * /, res/ari/resource_channels.c: ari/resource_channels: When + creating a channel, specify a default format (SLIN) When creating + channels via ARI, the current code fails to provide any default + format capabilities. For non-virtual channels this isn't really a + problem - the channels typically receive their capabilities as a + result of the underlying channel driver configuration. For + virtual channels (such as Local channels), the lack of any format + capabilities causes the Asterisk core to make some 'odd' choices + with respect to the translation paths. The issue reporter had + some paths that had 3 hops on each channel leg, causing multiple + transcodings and some really crappy audio/performance. By + specifying a baseline of SLIN, we prevent that from occurring. + Note that this is what AMI does when it performs an Originate, as + does res_clioriginate. Review: + https://reviewboard.asterisk.org/r/3068/ (issue ASTERISK-22962) + Reported by: Matt DiMeo ........ Merged revisions 403993 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-16 19:11 +0000 [r403960] David M. Lee <dlee@digium.com> + + * include/asterisk/pbx.h, main/asterisk.c, funcs/func_realtime.c, + main/pbx.c, main/tcptls.c, funcs/func_db.c, /, + README-SERIOUSLY.bestpractices.txt, configs/asterisk.conf.sample, + funcs/func_shell.c, funcs/func_env.c, funcs/func_lock.c, + UPGRADE-12.txt: security: Inhibit execution of privilege + escalating functions This patch allows individual dialplan + functions to be marked as 'dangerous', to inhibit their execution + from external sources. A 'dangerous' function is one which + results in a privilege escalation. For example, if one were to + read the channel variable SHELL(rm -rf /) Bad Things(TM) could + happen; even if the external source has only read permissions. + Execution from external sources may be enabled by setting + 'live_dangerously' to 'yes' in the [options] section of + asterisk.conf. Although doing so is not recommended. Also, the + ABI was changed to something more reasonable, since Asterisk 12 + does not yet have a public release. (closes issue ASTERISK-22905) + Review: http://reviewboard.digium.internal/r/432/ ........ Merged + revisions 403913 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403917 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403959 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-16 18:31 +0000 [r403958] Jonathan Rose <jrose@digium.com> + + * /, main/bridge.c: transfers: Fix bug setting both BLINDTRANSFER + and ATTENDEDTRANSFER The ast_bridge_set_transfer_variables + function is supposed to wipe whichever variable isn't being set. + Instead it was setting both to the new value. Oops. (issue + AFS-24) ........ Merged revisions 403957 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-16 16:12 +0000 [r403857-403865] Scott Griepentrog <sgriepentrog@digium.com> + + * main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to + prevent memory corruption During dialplan execution in + pbx_extension_helper(), the contexts global read lock prevents + link list corruption, but was released with a pointer to the + ast_exten and data later used in variable substitution. Instead, + this patch removes pbx_substitute_variables() and locates a copy + of the ast_exten data on the stack before releasing the lock, + where ast_exten could get free'd by another thread performing a + module reload. (issue AST-1179) Reported by: Thomas Arimont + (issue AST-1246) Reported by: Alexander Hömig Review: + https://reviewboard.asterisk.org/r/3055/ ........ Merged + revisions 403862 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403863 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403864 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_sms.c: app_sms: BufferOverflow when receiving odd + length 16 bit message This patch prevents an infinite loop + overwriting memory when a message is received into the + unpacksms16() function, where the length of the message is an odd + number of bytes. (closes issue ASTERISK-22590) Reported by: Jan + Juergens Tested by: Jan Juergens ........ Merged revisions 403856 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-15 01:39 +0000 [r403824] Matthew Jordan <mjordan@digium.com> + + * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions: + Use the right buffer length when printing URIs While + entertaining, sizeof(buflen) is not the same as buflen. Doh. + ........ Merged revisions 403823 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-14 17:28 +0000 [r403810-403812] Joshua Colp <jcolp@digium.com> + + * include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c: res_pjsip: Apply + outbound proxy to all SIP requests. Objects which are involved in + SIP request creation and sending now allow an outbound proxy to + be specified. For cases where an endpoint is used the outbound + proxy specified there will be applied. (closes issue + ASTERISK-22673) Reported by: Antti Yrjola Review: + https://reviewboard.asterisk.org/r/3022/ ........ Merged + revisions 403811 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, apps/app_queue.c, + res/ari/ari_model_validators.c, apps/app_dial.c, + res/ari/ari_model_validators.h, main/dial.c, + include/asterisk/stasis_channels.h, + rest-api/api-docs/events.json, /, res/stasis/app.c: res_stasis: + Expose event for call forwarding and follow forwarded channel. + This change adds an event for when an originated call is + redirected to another target. This event contains the original + channel and the newly created channel. If a stasis subscription + exists on the original originated channel for a stasis + application then a new subscription will also be created on the + stasis application to the redirected channel. This allows the + application to follow the call path completely. (closes issue + ASTERISK-22719) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/3054/ ........ Merged + revisions 403808 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 21:35 +0000 [r403797] Jonathan Rose <jrose@digium.com> + + * /, res/res_pjsip_messaging.c, main/message.c: documentation: Add + PJSIP technology to messaging documentation ........ Merged + revisions 403796 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 20:17 +0000 [r403784] Richard Mudgett <rmudgett@digium.com> + + * /, main/test.c: test.c: Fix too sticky unit test failed status. + Rerunning a failed unit test after loading any required modules + should allow the test to report a pass status if it now passes. + ........ Merged revisions 403782 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 20:13 +0000 [r403783] Jonathan Rose <jrose@digium.com> + + * /, main/bridge.c, main/bridge_basic.c, include/asterisk/bridge.h, + res/parking/parking_bridge_features.c, + res/parking/parking_manager.c: Transfers: Make Asterisk set + ATTENDEDTRANSFER/BLINDTRANSFER more reliably There were still a + few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be + set on channels involved with blind and attended transfers. This + would happen with features that were initialized by channel + driver specific mechanisms in multiparty calls. This patch + resolves those cases while attempted to keep the behavior for + setting those variables as consistent as possible. (closes issue + AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........ + Merged revisions 403781 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 18:33 +0000 [r403750-403768] Kevin Harwell <kharwell@digium.com> + + * main/channel.c, /, channels/chan_sip.c, + include/asterisk/channel.h, bridges/bridge_native_rtp.c, + channels/chan_pjsip.c: bridge_native_rtp: Deadlock during 4-way + conference creation The change contains a slightly adjusted patch + that was on the issue (submitted by kmoore). A fix was made by + adding in a bridge lock while calling bridge_start/stop from the + framehook callback. Since the framehook callback is not called + from the bridging core the bridge is not locked, but needs to be + before calling bridge_start. (closes issue ASTERISK-22749) + Reported by: Kinsey Moore Review: + https://reviewboard.asterisk.org/r/3066/ Patches: + lock_inversion.diff uploaded by kmoore (license 6273) ........ + Merged revisions 403767 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /, + main/http.c: ARI: Allow specifying channel variables during a + POST /channels Added the ability to specify channel variables + when creating/originating a channel in ARI. The variables are + sent in the body of the request and should be formatted as a + single level JSON object. No nested objects allowed. For example: + {"variable1": "foo", "variable2": "bar"}. (closes issue + ASTERISK-22872) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3052/ ........ Merged + revisions 403752 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_answer.c, rest-api/api-docs/bridges.json, + res/ari/resource_bridges.c, res/res_ari_bridges.c, + res/stasis/command.c, res/res_stasis_playback.c, /, + res/stasis/control.c, res/stasis/command.h, + include/asterisk/stasis_app.h, + include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c: + ARI: Adding a channel to a bridge while a live recording is + active blocks Added the ability to have rules that are checked + when adding and/or removing channels to/from a bridge. In this + case, if a channel is currently recording and someone attempts to + add it to a bridge an "is recording" rule is checked, fails, and + a 409 conflict is returned. Also command functions now return an + integer value that can be descriptive of what kind of problems, + if any, occurred before or during execution. (closes issue + ASTERISK-22624) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2947/ ........ Merged + revisions 403749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 05:00 +0000 [r403737] Matthew Jordan <mjordan@digium.com> + + * /, channels/Makefile: channels/Makefile: clean pjsip directory + ........ Merged revisions 403736 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 00:40 +0000 [r403726] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/app.h, tests/test_voicemail_api.c, main/app.c: + test_voicemail_api: Add check for a registered voicemail provider + before tests. It is much nicer diagnosing a test failure if + app_voicemail is actually loaded. + +2013-12-12 19:46 +0000 [r403714] Scott Griepentrog <sgriepentrog@digium.com> + + * contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py + (added), /: realtime: Create extensions in alembic ast-db-manage + contribution When the alembic scripts were written for creating + Asterisk realtime databases the extensions table for dialplan + wasn't included. This update creates the extensions table. + (closes issue ASTERISK-22815) Reported by: Zone Conkle Review: + https://reviewboard.asterisk.org/r/3064/ ........ Merged + revisions 403713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-12 19:18 +0000 [r403707] Jonathan Rose <jrose@digium.com> + + * /, channels/chan_pjsip.c: chan_pjsip: Revert r403587 This patch + was intended to eliminate a deadlock that occurs when masquerades + occur in pjsip channels, but has some potential side effects. + Mark Michelson is currently working on addressing this problem + from another angle. (issue ASTERISK-22936) Reported by: Jonathan + Rose ........ Merged revisions 403705 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-11 20:24 +0000 [r403687] Kevin Harwell <kharwell@digium.com> + + * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, /, + configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip_messaging.c, + res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c: + res_pjsip_messaging: send message to a default outbound endpoint + In some cases messages need to be sent to a direct URI (sip:<ip + address>). This patch adds in that support by using a default + outbound endpoint. When sending messages, if no endpoint can be + found then the default one is used. To facilitate this a new + default_outbound_endpoint option was added to the globals section + for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ + ........ Merged revisions 403680 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-11 19:22 +0000 [r403652] Russell Bryant <russell@russellbryant.com> + + * /, channels/chan_sip.c: Reset peer outboundproxy on sip.conf + reload If you set a peer's outboundproxy and then removed it from + the config, this would not get picked up in a config reload. This + patch fixes that by resetting it in set_peer_defaults(). Closes + ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/ + ........ Merged revisions 403634 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403635 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403639 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-11 19:19 +0000 [r403643] Richard Mudgett <rmudgett@digium.com> + + * apps/app_voicemail.c, include/asterisk/app.h, + include/asterisk/doxyref.h, main/app.c: app_voicemail: Voicemail + callback registration/unregistration function improvements. * The + voicemail registration/unregistration functions now take a struct + of callbacks instead of a lengthy parameter list of callbacks. * + The voicemail registration/unregistration functions now prevent a + competing module from interfering with an already registered + callback supplying module. + +2013-12-11 13:06 +0000 [r403617-403619] Matthew Jordan <mjordan@digium.com> + + * channels/pjsip/dialplan_functions.c, + include/asterisk/res_pjsip_session.h, channels/pjsip (added), /, + funcs/func_channel.c, channels/pjsip/include, + channels/pjsip/include/dialplan_functions.h, res/res_pjsip_t38.c, + channels/pjsip/include/chan_pjsip.h, channels/Makefile, + channels/chan_pjsip.c, main/xmldoc.c: func_channel, chan_pjsip: + Add CHANNEL read function support for chan_pjsip This patch adds + CHANNEL read support for chan_pjsip. This allows the dialplan to + use the CHANNEL function on a chan_pjsip channel to obtain + run-time information about the channel from the PJSIP channel + driver and the PJSIP stack. This includes: * RTP information, + including source/destination media addresses, whether or not the + media is secure, held, and other properties. * RTCP information. + This includes sets of parseable information, as well as + individual statistic attriutes. * PJSIP information. This + includes URIs, local/remote signalling addresses, whether or not + the signalling is secure, and other properties. * The endpoint + name. This can be used in conjunction with the PJSIP_ENDPOINT + function to obtain more detailed endpoint information. Review: + https://reviewboard.asterisk.org/r/3038/ ........ Merged + revisions 403618 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * Makefile, funcs/func_pjsip_endpoint.c (added), doc/snapshots.xslt + (removed), /, doc/appdocsxml.xslt (added), doc/appdocsxml.dtd, + main/sorcery.c: func_pjsip_endpoint: Add PJSIP_ENDPOINT function + for querying endpoint details This patch adds a new function, + PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint, + any property configured on an endpoint. This function is a + companion to the CHANNEL function, which can be used to extract + the endpoint name for a channel. Review: + https://reviewboard.asterisk.org/r/3035 ........ Merged revisions + 403616 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-10 15:15 +0000 [r403605] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_authenticator_digest.c: Fix correct authentication + behavior for artificial endpoint. When switching to using a + vector for authentication, I initialized the vector for the + artificial endpoint to be of size 1. However, this does not + result in AST_VECTOR_SIZE() returning 1 since there isn't + actually anything in the vector. Rather than trifle with the + vector by putting unnecessary elements in, I simply changed the + callback in res_pjsip_authenticator_digest.c to explicitly report + that the artificial endpoint requires authentication. Thanks to + Joshua Colp for pointing this out. + +2013-12-09 22:59 +0000 [r403576-403588] Jonathan Rose <jrose@digium.com> + + * /, channels/chan_pjsip.c: chan_pjsip: Fix a sticking channel lock + caused by channel masquerades (closes issue ASTERISK-22936) + Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/3042/ ........ Merged + revisions 403587 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * CHANGES, main/dial.c, apps/app_page.c, include/asterisk/dial.h: + app_page: Add predial handlers for app_page. (closes issue + AFS-14) Review: https://reviewboard.asterisk.org/r/3045/ + +2013-12-09 19:24 +0000 [r403544-403560] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_sorcery_astdb.c: Reverting regex part of -r403545 at + request of file. res_sorcery_astdb.c: Fix get multiple records by + regex. * Fix sorcery_astdb_retrieve_regex() pattern matching. Let + the regexec() function match the stored key values instead of + having astdb prefilter them. Previoiusly you could only use a + simple regex pattern when the pattern began with '^'. ........ + Merged revisions 403559 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix get multiple + records by regex. * Fix sorcery_astdb_retrieve_regex() pattern + matching. Let the regexec() function match the stored key values + instead of having astdb prefilter them. Previoiusly you could + only use a simple regex pattern when the pattern began with '^'. + * Fix off nominal memory leak in sorcery_astdb_retrieve_regex(). + ........ Merged revisions 403545 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/sorcery.c, /: sorcery: Eliminate shadowing a varaible that + caused confusion. * Eliminated shadowing of the + __ast_sorcery_apply_config() name parameter causing confusion. * + Fix potential crash from sorcery.conf user input in + __ast_sorcery_apply_config() if the user supplied a malformed + config line that is missing the sorcery object type name. * + Remove redundant test in __ast_sorcery_apply_config(). !config + and config == CONFIGS_STATUS_FILEMISSING are identical. ........ + Merged revisions 403541 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 18:32 +0000 [r403543] Joshua Colp <jcolp@digium.com> + + * /, main/endpoints.c: endpoints: Keep a reference to channel ids + when creating snapshot. The snapshot process for endpoints uses + the channel ids present on the endpoint itself. Without keeping a + reference it was possible for the strings to be freed underneath + any consumer of an endpoint snapshot. A reference is now held by + the snapshot to the channel ids and released when the snapshot is + destroyed. (issue ASTERISK-22801) Reported by: Matt Jordan + ........ Merged revisions 403542 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 18:14 +0000 [r403528] Richard Mudgett <rmudgett@digium.com> + + * main/sorcery.c, /: sorcery: Whitespace You would think that a new + file would start off without any whitespace oddities. ........ + Merged revisions 403527 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 17:29 +0000 [r403512-403526] Mark Michelson <mmichelson@digium.com> + + * apps/app_confbridge.c, CHANGES, + apps/confbridge/conf_state_multi_marked.c: Add a + CONFBRIDGE_RESULT channel variable to discern why a channel left + a ConfBridge. Review: https://reviewboard.asterisk.org/r/3009 + + * CHANGES, apps/app_mixmonitor.c: Create function for retrieving + Mixmonitor instance data. For the time, this is only useful for + retrieving the filename. The purpose of this function is to + better facilitate multiple mixmonitors per channel. Setting a + MIXMONITOR_FILENAME channel variable is not conducive to such + behavior, so allowing finer grained access to individual + mixmonitor properties improves the situation. The + MIXMONITOR_FILENAME channel variable is still set, though, so + there is no worry about backwards compatibility. Review: + https://reviewboard.asterisk.org/r/3023 + +2013-12-09 16:41 +0000 [r403511] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_nat.c, /: res_pjsip_nat: Add NAT module to session + dialogs. Due to the way pjproject internally works it was + possible for the NAT module to not be invoked on messages with-in + a session dialog. This means that the various parts of the + message would not get rewritten with the source IP address and + port. This change uses a session supplement to add the NAT module + to the dialog on the first incoming or outgoing INVITE. (closes + issue ASTERISK-22941) Reported by: Leif Madsen ........ Merged + revisions 403510 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 16:10 +0000 [r403499] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip/config_auth.c, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_pjsip_authenticator_digest.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_distributor.c, res/res_pjsip.c, + include/asterisk/res_pjsip.h: Switch PJSIP auth to use a vector. + Since Asterisk has a vector API now, places where arrays are + manually resized don't really make sense any more. Since the auth + work in PJSIP was freshly-written, it was easy to reform it to + use a vector. Review: https://reviewboard.asterisk.org/r/3044 + +2013-12-09 03:21 +0000 [r403436-403466] Matthew Jordan <mjordan@digium.com> + + * /, res/res_fax_spandsp.c: res_fax_spandsp: Always init T.38 + session to avoid crashes during state change Prior to this patch, + res_fax_spandsp was conservative with how it initialized the + spandsp T.38 context. It would only initialize it if the driver + thought the current state was a T.38 fax. While this works fine + in nominal situations, in certain off nominal situations, + res_fax_spandsp can believe that a T.38 fax will not occur when + in fact one has started. In particular, this was discovered when + res_fax would fall back to audio after timing out on a T.38 + upgrade. The SIP channel driver would continue to retry the + re-INVITE and - if the remote end responded after res_fax timed + out with a 200 OK - a T.38 frame would be delivered to the + res_fax stack when it no longer expected it. As it turns out, + there does not appear to be any downside to always initializing + the T.38 context, other than the actual memory allocation. Since + that avoids this off nominal situation (and others which are + equally likely hard to predict), this is the safest way to avoid + this problem. Much thanks to Torrey as well for providing a + scenario that reproduces this issue. (closes issue + ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey + Searle patches: always-init-t38.patch uploaded by awinters + (License 6477) A_PARTY.xml uploaded by tsearle (License 5334) + ........ Merged revisions 403449 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403450 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_config_sqlite.c: res_config_sqlite: Check for CDR + unregistration failures If the CDR unregistration fails due to an + inflight CDR, the res_config_sqlite module needs to bail on + unloading itself. Otherwise, the config could be unloaded + (including the CDR table name) while the CDR engine posts a CDR + to the still registered backend, resulting in a crash. ........ + Merged revisions 403435 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-05 23:40 +0000 [r403414] Jonathan Rose <jrose@digium.com> + + * apps/app_record.c: app_record: Add an option that allows DTMF '0' + to act as an additional terminator Using this terminator when + active results in ${RECORD_STATUS} being set to 'OPERATOR' + instead of 'DTMF' (closes issue AFS-7) Review: + https://reviewboard.asterisk.org/r/3041/ + +2013-12-05 22:10 +0000 [r403402-403404] David M. Lee <dlee@digium.com> + + * addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c, + channels/chan_pjsip.c, res/parking/parking_manager.c, + channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c, /, + apps/app_meetme.c, funcs/func_timeout.c, main/bridge.c, + tests/test_stasis_channels.c, main/core_unreal.c, + include/asterisk/channel.h, channels/chan_gtalk.c, main/cel.c, + apps/app_queue.c, channels/sig_pri.c, main/stasis_bridges.c, + channels/chan_jingle.c, channels/chan_phone.c, + channels/chan_dahdi.c, main/dial.c, channels/sig_analog.c, + include/asterisk/stasis_channels.h, res/res_agi.c, + channels/chan_motif.c, channels/chan_h323.c, tests/test_cel.c, + apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c, + apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc, + addons/chan_ooh323.c, channels/chan_sip.c, main/pickup.c, + include/asterisk/aoc.h, include/asterisk/stasis_bridges.h, + apps/app_userevent.c, apps/app_disa.c, main/core_local.c, + include/asterisk/channelstate.h, channels/chan_console.c, + channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c, + res/parking/parking_bridge_features.c, apps/app_agent_pool.c, + main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, + pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c, + channels/chan_nbs.c: Reverting r403311. It's causing ARI tests to + hang. ........ Merged revisions 403398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c: ari: Fix deadlock problem with functions + that use autoservice. The code for getting channel variables from + ARI assumed that you needed to lock the channel in order to + properly execute functions and read channel variables. + Apparently, this is not the case, since any dialplan function + that puts the channel into autoservice deadlocks when attempting + to remove the channel from autoservice. ........ Merged revisions + 403342 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Multiple revisions 403304,403310 ........ r403304 | dlee | + 2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line Fixed the + filename for the ari.conf docs ........ r403310 | file | + 2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines Revert + revision 403304: Fixed the filename for the ari.conf docs The + changed value refers to the name of the module. The name of the + configuration file is specified in the configFile section. + ........ Merged revisions 403304,403310 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-04 21:42 +0000 [r403378] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip_registrar.c: res_pjsip_registrar: undefined + function pointer symbol Used a static wrapper around the + offending function to alleviate the issue. Reported by: rmudgett + ........ Merged revisions 403377 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-04 20:54 +0000 [r403365] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_t38.c, /: res_pjsip_t38: Don't pass T.38 control + frames through to other hooks. This crept up during gateway + testing where the gateway would receive the request to negotiate + and assume it came from the remote side, causing the gateway + state machine to go a little, to a use a technical term, "wonky". + ........ Merged revisions 403364 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-04 18:41 +0000 [r403350] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip.c: Initialize the hash value argument to + pj_hash_get() to 0. Passing a non-zero value causes PJLIB to use + the given input as the hash value. Passing zero causes the + parameter to become an output parameter that receives the hash + value that was computed based on the given key. This change + essentially makes ast_sip_dict_get() properly retrieve the + desired value. ........ Merged revisions 403349 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 18:01 +0000 [r403330] Joshua Colp <jcolp@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_pjsip_session.c: res_pjsip_session: Add support for + PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. Newer versions of PJSIP + have changed to using a flag for the + PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds + a configure check to detect the presence of the flag and use it + if found. ........ Merged revisions 403329 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 17:35 +0000 [r403327] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip_registrar_expire.c, res/res_pjsip/pjsip_options.c, + tests/test_sorcery.c, include/asterisk/bucket.h, main/sorcery.c, + /, main/bucket.c: sorcery, bucket: Change observer remove calls + to take const callbacks struct. * Make + ast_sorcery_observer_remove() accept a const callbacks struct. * + Make ast_sorcery_observer_remove() tolerant of the sorcery + parameter being NULL. Now it can be called within a module unload + routine if the sorcery initialization fails. * Fix + ast_sorcery_observer_add() to fail if the container link fails. + ........ Merged revisions 403324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 17:07 +0000 [r403314] Mark Michelson <mmichelson@digium.com> + + * channels/chan_nbs.c, main/bridge_channel.c, res/res_stasis.c, + channels/chan_pjsip.c, res/parking/parking_manager.c, + apps/app_voicemail.c, channels/chan_unistim.c, + channels/chan_vpb.cc, addons/chan_ooh323.c, /, + include/asterisk/aoc.h, apps/app_meetme.c, main/bridge.c, + apps/app_userevent.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/endpoints.c, main/stasis_bridges.c, + main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, + main/dial.c, channels/sig_analog.c, channels/chan_skinny.c, + res/res_agi.c, channels/chan_motif.c, pbx/pbx_realtime.c, + channels/chan_alsa.c, main/stasis_channels.c, + apps/app_confbridge.c, addons/chan_mobile.c, tests/test_cdr.c, + res/res_pjsip_refer.c, channels/chan_mgcp.c, apps/app_dial.c, + main/pbx.c, channels/chan_sip.c, main/pickup.c, + funcs/func_timeout.c, tests/test_stasis_channels.c, + main/core_unreal.c, include/asterisk/stasis_bridges.h, + apps/app_disa.c, include/asterisk/channel.h, main/core_local.c, + include/asterisk/channelstate.h, channels/chan_console.c, + main/cel.c, apps/app_queue.c, channels/sig_pri.c, + channels/chan_oss.c, res/parking/parking_bridge_features.c, + apps/app_agent_pool.c, channels/chan_jingle.c, + channels/chan_misdn.c, include/asterisk/stasis_channels.h, + channels/chan_h323.c, tests/test_cel.c: Add channel locking for + channel snapshot creation. This adds channel locks around calls + to create channel snapshots as well as other functions which + operate on a channel and then end up creating a channel snapshot. + Functions that expect the channel to be locked prior to being + called have had their documentation updated to indicate such. + ........ Merged revisions 403311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 16:39 +0000 [r403313] Joshua Colp <jcolp@digium.com> + + * main/media_index.c, /: media_index: Make media indexing tolerable + of bad symlinks. Media indexing will now skip over files and + directories that stat will not return information about. This can + occur under normal conditions when a symbolic link points to a + location that no longer exists. ........ Merged revisions 403312 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-02 18:12 +0000 [r403292] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c, /: Check and reject non-digits e164 values + on peers and general sections in ooh323.conf Regenerate e164 + endpoint list on reload ooh323 (issue ASTERISK-22901) Reported + by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch ........ + Merged revisions 403288 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403290 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-01 21:13 +0000 [r403257-403272] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_session.c: res_pjsip_session: Apply fromuser and + fromdomain to all requests as documented. ........ Merged + revisions 403271 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_t38.c, /: res_pjsip_t38: Add the framehook to the + channel only on first INVITE. The check for determining whether + the T.38 framehook should be added to the channel or not has now + been changed to guarantee adding only occurs on the first + incoming or outgoing INVITE. ........ Merged revisions 403258 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/security_events.c, res/res_pjsip/pjsip_options.c, + res/res_pjsip.c, res/res_pjsip_transport_websocket.c, + include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c: + res_pjsip_transport_websocket: Fix security events and simplify + implementation. Transport type determination for security events + has been simplified to use the type present on the message itself + instead of searching through configured transports to find the + transport used. The actual WebSocket transport has also been + simplified. It now leverages the existing PJSIP transport manager + for finding the active WebSocket transport for outgoing messages. + This removes the need for res_pjsip_transport_websocket to store + a mapping itself. (closes issue ASTERISK-22897) Reported by: Max + E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/ + ........ Merged revisions 403256 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-30 14:12 +0000 [r403241] Joshua Colp <jcolp@digium.com> + + * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /, + res/ari/ari_model_validators.c: res_ari: Add Recording events to + the validator. ........ Merged revisions 403240 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-28 02:12 +0000 [r403208-403224] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't produce an + invalid media stream with no formats. Depending on configuration + it was possible for a media stream to be created without any + media formats. The produced SDP would fail internal validation + and cause a crash. The code will now no longer add media streams + with no formats to the SDP, allowing it to pass validation and + work. (closes issue ASTERISK-22858) Reported by: Anthony Messina + ........ Merged revisions 403223 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_header_funcs.c, /: res_pjsip_header_funcs: Don't + add headers to re-INVITEs. When sending a re-INVITE to an + endpoint it was possible for received headers to be added as well + (since they are stored for retrieval using the PJSIP_HEADER + dialplan function). This caused a broken (and potentially large) + SIP INVITE to be produced and sent. This changes the module so it + will no longer add headers to re-INVITEs. (closes issue + ASTERISK-22882) Reported by: David M. Lee ........ Merged + revisions 403221 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /: res_stasis_playback: Add 'number', + 'digits', and 'characters' URI scheme implementations. This + change adds new URI scheme implementations for playing numbers, + digits, and characters. This is done as part of the normal + playback mechanism and can be used with queueing to create a + combined sentence. Review: + https://reviewboard.asterisk.org/r/3028/ ........ Merged + revisions 403209 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c, + res/res_pjsip_session.c, include/asterisk/res_pjsip.h: + res_pjsip_session: Add configurable behavior for redirects. The + action taken when a redirect occurs is now configurable on a + per-endpoint basis. The redirect can either be treated as a + redirect to a local extension, to a URI that is dialed through + the Asterisk core, or to a URI that is dialed within PJSIP + itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged + revisions 403207 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-27 17:32 +0000 [r403192] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/astdb.h: astdb: Tweak some doxygen comments. + +2013-11-27 16:12 +0000 [r403180] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix crash when + reloading certain configurations. Certain options available that + specify a SIP URI perform validation on the provided URI using + the PJSIP URI parser. This operation requires that the thread + executing it be registered with the PJLIB library. During reloads + this was done on a thread which was NOT registered with it. This + fixes the problem by creating a task which reloads the + configuration on a PJSIP thread. (closes issue ASTERISK-22923) + Reported by: Anthony Messina ........ Merged revisions 403179 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-27 15:48 +0000 [r403177] David M. Lee <dlee@digium.com> + + * res/res_ari_channels.c, include/asterisk/ari.h, + rest-api-templates/param_parsing.mustache, + include/asterisk/http.h, res/res_ari_recordings.c, + res/res_ari_endpoints.c, main/http.c, + rest-api-templates/swagger_model.py, res/res_ari_playbacks.c, + res/res_ari_sounds.c, rest-api-templates/asterisk_processor.py, + res/res_ari_bridges.c, tests/test_ari.c, res/res_ari.c, /, + res/res_ari_device_states.c, res/res_ari_asterisk.c, + rest-api-templates/res_ari_resource.c.mustache, + res/res_ari_applications.c: ari:Add application/json parameter + support The patch allows ARI to parse request parameters from an + incoming JSON request body, instead of requiring the request to + come in as query parameters (which is just weird for POST and + DELETE) or form parameters (which is okay, but a bit asymmetric + given that all of our responses are JSON). For any operation that + does _not_ have a parameter defined of type body (i.e. + "paramType": "body" in the API declaration), if a request + provides a request body with a Content type of + "application/json", the provided JSON document is parsed and + searched for parameters. The expected fields in the provided JSON + document should match the query parameters defined for the + operation. If the parameter has 'allowMultiple' set, then the + field in the JSON document may optionally be an array of values. + (closes issue ASTERISK-22685) Review: + https://reviewboard.asterisk.org/r/2994/ + +2013-11-27 15:31 +0000 [r403161-403174] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Update + handling of some options to work with new option names. Some + options (such as call_group and pickup_group) share the same + configuration handler and decide what logic to use based on the + name of the option. These handlers were not updated to check for + the new option names and were treating the options as invalid. + This change simply updates the handlers with the proper names of + the options. (closes issue ASTERISK-22922) Reported by: Anthony + Messina ........ Merged revisions 403173 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Fix + a configure issue with PJSIP transaction group lock detection. + The configure check did not use the provided paths for pjproject + if provided when looking for transaction group lock support. + ........ Merged revisions 403160 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-23 17:48 +0000 [r403133-403135] Kevin Harwell <kharwell@digium.com> + + * res/ari.make, rest-api/api-docs/applications.json, + res/ari/resource_device_states.h (added), + include/asterisk/stasis_app_device_state.h (added), + res/ari/resource_applications.h, res/res_stasis.c, + include/asterisk/devicestate.h, rest-api/api-docs/events.json, + res/res_stasis_device_state.exports.in (added), res/stasis/app.c, + res/res_ari_device_states.c (added), /, + include/asterisk/stasis_app.h, main/devicestate.c, + res/stasis/app.h, rest-api/resources.json, + res/res_stasis_device_state.c (added), + res/ari/ari_model_validators.c, res/ari/ari_model_validators.h, + res/ari/resource_device_states.c (added), + rest-api/api-docs/deviceStates.json (added), + rest-api-templates/ari.make.mustache: ARI: Implement device state + API Created a data model and implemented functionality for an ARI + device state resource. The following operations have been added + that allow a user to manipulate an ARI controlled device: + Create/Change the state of an ARI controlled device PUT + /deviceStates/{deviceName}&{deviceState} Retrieve all ARI + controlled devices GET /deviceStates Retrieve the current state + of a device GET /deviceStates/{deviceName} Destroy a device-state + controlled by ARI DELETE /deviceStates/{deviceName} The ARI + controlled device must begin with 'Stasis:'. An example + controlled device name would be Stasis:Example. A + 'DeviceStateChanged' event has also been added so that an + application can subscribe and receive device change events. Any + device state, ARI controlled or not, can be subscribed to. While + adding the event, the underlying subscription control mechanism + was refactored so that all current and future resource + subscriptions would be the same. Each event resource must now + register itself in order to be able to properly handle + [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/3025/ ........ Merged + revisions 403134 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_registrar.c, main/sorcery.c, + include/asterisk/res_pjsip.h, include/asterisk/acl.h, + res/res_pjsip/config_auth.c, include/asterisk/utils.h, + res/res_pjsip.exports.in, /, + res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, main/utils.c, + res/res_pjsip.c, res/res_pjsip_exten_state.c, + include/asterisk/res_pjsip_pubsub.h, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, res/res_pjsip_mwi.c, + res/res_pjsip/pjsip_configuration.c, include/asterisk/sorcery.h, + include/asterisk/strings.h, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip_pubsub.c, res/res_pjsip/config_transport.c: + res_pjsip: AMI commands and events. Created the following AMI + commands and corresponding events for res_pjsip: + PJSIPShowEndpoints - Provides a listing of all pjsip endpoints + and a few select attributes on each. Events: EndpointList - for + each endpoint a few attributes. EndpointlistComplete - after all + endpoints have been listed. PJSIPShowEndpoint - Provides a detail + list of attributes for a specified endpoint. Events: + EndpointDetail - attributes on an endpoint. AorDetail - raised + for each AOR on an endpoint. AuthDetail - raised for each + associated inbound and outbound auth TransportDetail - transport + attributes. IdentifyDetail - attributes for the identify object + associated with the endpoint. EndpointDetailComplete - last event + raised after all detail events. PJSIPShowRegistrationsInbound - + Provides a detail listing of all inbound registrations. Events: + InboundRegistrationDetail - inbound registration attributes for + each registration. InboundRegistrationDetailComplete - raised + after all detail records have been listed. + PJSIPShowRegistrationsOutbound - Provides a detail listing of all + outbound registrations. Events: OutboundRegistrationDetail - + outbound registration attributes for each registration. + OutboundRegistrationDetailComplete - raised after all detail + records have been listed. PJSIPShowSubscriptionsInbound - A + detail listing of all inbound subscriptions and their attributes. + Events: SubscriptionDetail - on each subscription detailed + attributes SubscriptionDetailComplete - raised after all detail + records have been listed. PJSIPShowSubscriptionsOutbound - A + detail listing of all outboundbound subscriptions and their + attributes. Events: SubscriptionDetail - on each subscription + detailed attributes SubscriptionDetailComplete - raised after all + detail records have been listed. (issue ASTERISK-22609) Reported + by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ + ........ Merged revisions 403131 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-23 12:52 +0000 [r403118-403120] Joshua Colp <jcolp@digium.com> + + * res/res_stasis_playback.c, rest-api/api-docs/events.json, /, + res/res_stasis_recording.c, res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, + res/ari/ari_model_validators.h: ari: Add events for playback and + recording. While there were events defined for playback and + recording these were not actually sent. This change implements + the to_json handlers which produces them. (closes issue + ASTERISK-22710) Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/3026/ ........ Merged + revisions 403119 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_snoop.exports.in (added), /, + include/asterisk/stasis_app_snoop.h (added), + rest-api/api-docs/channels.json, res/res_stasis_snoop.c (added), + main/audiohook.c, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h: ari: Add + Snoop operation for spying/whispering on channels. The Snoop + operation can be invoked on a channel to spy or whisper on it. It + returns a channel that any channel operations can then be invoked + on (such as record to do monitoring). (closes issue + ASTERISK-22780) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3003/ ........ Merged + revisions 403117 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-23 00:22 +0000 [r403106] Rusty Newton <rnewton@digium.com> + + * apps/app_voicemail.c: app_voicemail: when forwarding a message, + play vm-msgforwarded instead of vm-msgsaved In the last release + of sounds, 1.4.25 we added a vm-msgforwarded prompt for various + core languages. Now we use that prompt. (issue ASTERISK-21413) + (closes issue ASTERISK-21413) Reported by: netwrkr Tested by: + newtonr + +2013-11-22 23:57 +0000 [r403095] Kinsey Moore <kmoore@digium.com> + + * tests/test_stasis.c, /, tests/test_stasis_channels.c: Make sure + unit tests compile This fixes the unit tests that were broken by + r403069 and several functions requiring a new parameter for + sanitization of JSON messages generated from object snapshots. + ........ Merged revisions 403094 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 22:37 +0000 [r403083] Kevin Harwell <kharwell@digium.com> + + * /, + contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py, + res/res_pjsip/pjsip_configuration.c: res_pjsip: convert + configuration settings names to snake case some more Updated the + alembic script for pjsip. Also, the dtls config parsing stuff was + expecting strings with no underscores, so removed the underscores + from the option name before passing it to the parser. ........ + Merged revisions 403082 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 20:10 +0000 [r403070] Kinsey Moore <kmoore@digium.com> + + * res/res_stasis.c, main/stasis_endpoints.c, + res/ari/resource_endpoints.c, main/rtp_engine.c, /, + res/stasis/app.c, include/asterisk/stasis_bridges.h, + include/asterisk/stasis.h, include/asterisk/stasis_app.h, + main/stasis_bridges.c, res/ari/resource_bridges.c, main/json.c, + main/stasis_message.c, include/asterisk/stasis_channels.h, + main/stasis_channels.c, res/ari/resource_channels.c, + include/asterisk/stasis_endpoints.h: ARI: Don't leak + implementation details This change prevents channels used as + implementation details from leaking out to ARI. It does this by + preventing creation of JSON blobs of channel snapshots created + from those channels and sanitizing JSON blobs of bridge snapshots + as they are created. This introduces a framework for excluding + information from output targeted at Stasis applications on a + consumer-by-consumer basis using channel sanitization callbacks + which could be extended to bridges or endpoints if necessary. + This prevents unhelpful error messages from being generated by + ast_json_pack. This also corrects a bug where BridgeCreated + events would not be created. (closes issue ASTERISK-22744) + Review: https://reviewboard.asterisk.org/r/2987/ Reported by: + David M. Lee ........ Merged revisions 403069 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 17:27 +0000 [r403051] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_acl.c, res/res_pjsip.c, + res/res_pjsip/config_transport.c, res/res_pjsip/config_global.c, + /, configs/pjsip.conf.sample, res/res_pjsip/config_system.c, + contrib/scripts/sip_to_pjsip/sip_to_pjsip.py, + res/res_pjsip/pjsip_configuration.c: res_pjsip: convert + configuration settings names to snake case Renamed, where + appropriate, the configuration options for chan/res_pjsip to use + snake case (compound words separated by an underscore). For + example, faxdetect will become fax_detect, recordofffeature will + become record_off_feature, etc... Review: + https://reviewboard.asterisk.org/r/3002/ ........ Merged + revisions 403022 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 17:12 +0000 [r403017] Joshua Colp <jcolp@digium.com> + + * /, main/translate.c: translate: Move freeing of frame to after it + is used. When translating from one format to another it is + possible to inform the translation function that the source frame + should be freed. This was previously done immediately but shortly + afterwards the frame that was freed was accessed and used again. + This change moves code around a bit so that the frame is now + freed after it has been completely used. (closes issue + ASTERISK-22788) Reported by: Corey Farrell Patches: + translate-access-after-free-11up.patch uploaded by coreyfarrell + (license 5909) translate-access-after-free-1.8.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 403014 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403015 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403016 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 16:43 +0000 [r403013] Richard Mudgett <rmudgett@digium.com> + + * apps/app_directed_pickup.c, CHANGES: PickupChan: Add ability to + specify channel uniqueids as well as channel names. * Made + PickupChan() search by channel uniqueids if the search could not + find a channel by name. * Ensured PickupChan() never considers + the picking channel for pickup. * Made PickupChan() option p use + a common search by name routine. The original search was + erroneously case sensitive. (issue AFS-42) Review: + https://reviewboard.asterisk.org/r/3017/ + +2013-11-21 22:38 +0000 [r402995] Jonathan Rose <jrose@digium.com> + + * CHANGES, apps/app_directory.c: app_directory: Set variable + indicating reason directory exited By the time the directory + application exits, a channel variable DIRECTORY_RESULT will be + set for the channel that invoked it which can be used to + determine the reason for exit. The changes log and the + app_directory documentation contain specific details about each + of the possible values for DIRECTORY_RESULT. Review: + https://reviewboard.asterisk.org/r/3016/ + +2013-11-21 22:36 +0000 [r402982-402994] David M. Lee <dlee@digium.com> + + * rest-api-templates/ari_resource.c.mustache, /, + rest-api-templates/res_ari_resource.c.mustache: ari: Fix #include + to match generated headers for snakeCase resource files ........ + Merged revisions 402993 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api-templates/make_ari_stubs.py, /: ari: Fix generators for + resources with camelCase names. For the new deviceState resource, + we need to properly generate device_state.[ch] files. ........ + Merged revisions 402981 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 19:22 +0000 [r402969] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_session.c, /: res_pjsip_session: Fix memory leak of + direct media format capabilities The direct media format + capabilities are always allocated in ast_sip_session_alloc and + were not freed in the session destructor. Whoops. (This being the + third whoops caught by Scott and Nitesh's valgrind work for the + Asterisk Test Suite. Nifty!) ........ Merged revisions 402968 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 19:09 +0000 [r402945-402957] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/app.h, /: voicemail: Fixup some doxygen + comments. ........ Merged revisions 402956 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/bucket.c: bucket: Fix scheme ref leak in + __ast_bucket_scheme_register(). ........ Merged revisions 402944 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 17:53 +0000 [r402942-402943] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix use of + uninitialized value in PJSIP In PJMEDIA, + pjmedia_sdp_rtpmap_to_attr will attempt to use the string + rtpmap.param regardless of its length value. Simply setting the + length to 0 does not prevent the garbage on the stack in + rtpmap.param.ptr from being formatted in a sprintf call. This + patch initializes the string to NULL so that at the very least, + something is provided to the function that is predictable. + ........ Merged revisions 402941 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c: res_pjsip_mwi: Fix memory leak of MWI + subscriptions container This patch fixes a reference counting + memory leak on the ao2_container created as part of + create_mwi_subscriptions. When we create the container in this + routine, the intent is to hand lifetime ownership over to the + global container unsolicited_mwi. When + ao2_global_obj_replace_unref is called, the reference count on + mwi_subscriptions (the container) will be bumped by 1; however, + the function does not decrement the reference count on + mwi_subscriptions when this occurs. This will prevent the + container from being fully disposed of when Asterisk exits (or on + any subsequent call to this operation, such as during a reload). + ........ Merged revisions 402940 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 15:57 +0000 [r402928-402929] David M. Lee <dlee@digium.com> + + * res/res_stasis.c, /: stasis: Fixed scoping problem with bridge + tracking. ........ Merged revisions 402817 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_channels.c, res/res_ari_channels.c, + res/ari/resource_channels.h, /, res/stasis/control.c, + include/asterisk/stasis_app.h, rest-api/api-docs/channels.json: + ari: Add silence generator controls This patch adds the ability + to start a silence generator on a channel via ARI. This generator + will play silence on the channel (avoiding audio timeouts on the + peer) until it is stopped, or some other media operation is + started (like playing media, starting music on hold, etc.). + (closes issue ASTERISK-22514) Review: + https://reviewboard.asterisk.org/r/3019/ ........ Merged + revisions 402926 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-19 23:17 +0000 [r402892] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_caller_id.c: res_pjsip_caller_id: Don't + overwrite user portion of the From header when fromuser is set. + The fromuser option is used to explicitly set the user within the + From header. The res_pjsip_caller_id module did not take this + setting into account when determining if the From header could be + modified or not. (closes issue ASTERISK-22866) Reported by: + Anthony Messina ........ Merged revisions 402891 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-16 13:51 +0000 [r402865] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip/pjsip_distributor.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: res_pjsip: Add + support for building against pjproject with SIP transaction group + lock support. SIP transaction group lock support has been + backported into our pjproject. Since the code now internally uses + a group lock the code is now changed to unlock it if present. + Note that the act of finding the transaction is what actually + returns it locked. For further information about group locks + check out the wiki page at: + http://trac.pjsip.org/repos/wiki/Group_Lock (issue + ASTERISK-22818) Reported by: Matt Jordan ........ Merged + revisions 402864 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-15 22:38 +0000 [r402854] Jonathan Rose <jrose@digium.com> + + * apps/app_confbridge.c, CHANGES, + apps/confbridge/conf_config_parser.c, + configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h: Confbridge: Add option to + review the recording similar to announce_join_leave Review: + https://reviewboard.asterisk.org/r/3008/ + +2013-11-15 14:37 +0000 [r402839] Kinsey Moore <kmoore@digium.com> + + * /, main/cel.c: CEL: Fix crash when using CELGenUserEvent This + fixes a crash when CELGenUserEvent is called from the dialplan + while CEL is disabled. Currently, CEL does not create its topics + and forwards if it is not enabled and external entities may + depend on these topics blindly since they should always be + available. This patch breaks up route creation and topic/forward + creation such that the CEL topics and forwards will always exist + while the router and its associated routes will be torn down and + recreated as necessary. (closes issue ASTERISK-22799) Review: + https://reviewboard.asterisk.org/r/3010/ Reported by: Matt Jordan + ........ Merged revisions 402838 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-14 21:36 +0000 [r402820-402829] Richard Mudgett <rmudgett@digium.com> + + * apps/app_directed_pickup.c: Pickup: Pickup() and PickupChan() + parameter parsing improvements. * Made Pickup() and PickupChan() + tollerate empty pickup values. i.e., You can now have + Pickup(&&exten@context). * Made PickupChan() use the standard + option flag parsing code. + + * apps/app_directed_pickup.c: Pickup: Ensure using PICKUPMARK never + considers the picking channel. + +2013-11-14 20:32 +0000 [r402819] Jonathan Rose <jrose@digium.com> + + * CHANGES, main/pbx.c, apps/app_sayunixtime.c: Say: If + SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF + Similar to how background works, if a say application is called + with this variable set to 'true', 'yes', 'on', etc. then using + DTMF while the say action is in progress will result in the + channel jumping to that extension in the dialplan. Review: + https://reviewboard.asterisk.org/r/3011/ + +2013-11-13 23:11 +0000 [r402805] Joshua Colp <jcolp@digium.com> + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /, + res/stasis/control.c, include/asterisk/stasis_app.h: + res_ari_channels: Add the ability to stop locally generated + ringing on a channel. Using the 'ring' operation it is possible + to start locally generated ringback if the channel is answered. + This change adds the ability to stop it by using DELETE. ........ + Merged revisions 402804 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 23:17 +0000 [r402788-402795] Kevin Harwell <kharwell@digium.com> + + * res/ari/resource_endpoints.c, /: ari endpoints: GET + /ari/endpoints/{invalid-tech} should return a 404 Was returning a + 404 on a valid technology with an empty list of endpoints. Now + checking against the channel tech to make sure the tech itself is + valid and not just an empty list of endpoints. (issue + ASTERISK-22803) Reported by: David M. Lee ........ Merged + revisions 402793 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c, + /, res/res_ari_endpoints.c: ari endpoints: GET + /ari/endpoints/{invalid-tech} should return a 404 Implementation + listing endpoints by technology returned an empty array if no + matching endpoints were found. Fixed so a "404 Not Found" will be + returned instead. (closes issue ASTERISK-22803) Reported by: + David M. Lee ........ Merged revisions 402787 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 19:38 +0000 [r402768-402778] Mark Michelson <mmichelson@digium.com> + + * /, main/channel.c: Switch to a scoped lock to avoid missing + unlocks in failure returns. ........ Merged revisions 402769 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel.c, /: Move a NULL check to a place that makes more + sense. Two variables were being checked for NULLity immediately + after being declared NULL. I moved the NULL check until after the + variables are allocated. This allows for the "channelvars" option + in manager.conf to work as intended again. ........ Merged + revisions 402767 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 16:49 +0000 [r402758] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_messaging.c, res/res_pjsip_header_funcs.c, /: + pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer + dereferences Both res_pjsip_messaging and res_pjsip_header_funcs + were causing asterisk to crash because they were trying to + dereference a NULL pointer. In the case of res_pjsip_messaging it + was attempting to "print" a contact header that did not exist. In + fact contact headers should not be part of a SIP MESSAGE, so the + offending code was simply removed. In the case of + res_pjsip_header_funcs a null private channel tech was being + passed to the function and then later dereferenced. Added null + checks (and error logging) to the read/write function handlers to + guard against crashing. (closes issue ASTERISK-22821) Reported + by: Anthony Messina ........ Merged revisions 402757 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 16:34 +0000 [r402756] Kinsey Moore <kmoore@digium.com> + + * /, apps/app_celgenuserevent.c: CELGenUserEvent: Fix error message + from ast_json_pack This prevents NULL from being passed into an + ast_json_pack call when no extra information is passed to the + application which prevents an error message about NULL arguments + from being generated. ........ Merged revisions 402755 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 15:27 +0000 [r402741] David M. Lee <dlee@digium.com> + + * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /: + Fixed a typ. ........ Merged revisions 402738 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 15:03 +0000 [r402711] Kinsey Moore <kmoore@digium.com> + + * channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID + read Asterisk will sometimes core dump during caller id read on + analog channels due to a negative return value from the read() in + my_get_callerid that slips through as a negative length argument + to callerid_feed() if the errno returned by DAHDI is ELAST. This + change ensures that the negative return is treated properly even + when it is ELAST. (closes issue ASTERISK-22746) Reported by: + Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch + uploaded by Michael Walton (License 6502) ........ Merged + revisions 402708 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402709 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402710 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-11 20:28 +0000 [r402698] Jonathan Rose <jrose@digium.com> + + * apps/app_confbridge.c: Confbridge: add test events for dynamic + menus test Adds a couple of test events for conference menu + actions so that it's easy to discern when those menu actions have + been triggered. (issue ASTERISK-22760) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2999/ + +2013-11-11 19:31 +0000 [r402688] Mark Michelson <mmichelson@digium.com> + + * apps/app_confbridge.c, /: Get rid of some inaccurate comments. + I'm doing some unrelated work in app_confbridge and finding these + "invalid pin" comments to be annoying. Get out! ........ Merged + revisions 402686 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402687 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-11 15:37 +0000 [r402648] Kinsey Moore <kmoore@digium.com> + + * /, apps/app_queue.c: app_queue: Honor penalty limits of 0 In the + current app_queue code from 1.8 up to trunk the upper and lower + penalties can be set to 0 but the value is interpreted to be + disabled instead of actually setting limits. This is especially + evident if min and max limits are set to 0 and members with + penalties of 0 and 1 are in the queue since the member with + penalty 1 will still receive calls. This patch adjusts the + special disabled value to be INT_MAX instead of 0. (closes issue + ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/ + Reported by: Schmooze Com ........ Merged revisions 402645 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402646 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402647 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 23:07 +0000 [r402607] Scott Griepentrog <sgriepentrog@digium.com> + + * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: + keep same local (from) tag for outgoing register requests For + outbound register requests the tag on the From line was updated + every 20 seconds prior to a successful registration and also once + for each registration renewal. That behavior can possibly cause + the registration to be denied because of the different tag, and + is not aligned with the intention of RFC 3261 8.1.3.5 "... + request constitutes a new transaction and SHOULD have the same + value of the Call-ID, To, and From of the previous request...". + This updates chan_sip to have a field to keep the local tag in + the registration structure and use that tag for registration + requests where the callid is also unchanged. (closes issue + ASTERISK-12117) Reported by: Pawel Pierscionek Review: + https://reviewboard.asterisk.org/r/2988/ ........ Merged + revisions 402604 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402605 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402606 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 20:37 +0000 [r402595] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_stasis.c: res_stasis.c: Fix locking issues with the + app_bridge_moh container. * Fix unlinking from the + app_bridges_moh container in remove_bridge_moh() without a lock + under normal circumstances. * Made check + ast_bridge_set_after_callback() return value in + bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK() + locking over too much scope in stasis_app_bridge_moh_channel() + and stasis_app_bridge_moh_stop(). * Fixed unusual usage of + ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge + from off nominal path in stasis_app_bridge_create(). * Fixed + strange construct in stasis_app_unsubscribe(). From a bad merge? + * Made load_module() cleanup on failure. Review: + https://reviewboard.asterisk.org/r/2962/ ........ Merged + revisions 402593 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 19:33 +0000 [r402585] Jonathan Rose <jrose@digium.com> + + * /, main/security_events.c, configs/manager.conf.sample, CHANGES, + include/asterisk/manager.h, main/manager.c: security_events: Push + out security events over AMI events Security Events will now be + written to any listener of the new 'security' class Review: + https://reviewboard.asterisk.org/r/2998/ ........ Merged + revisions 402584 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 19:22 +0000 [r402583] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip.c, /: Clarify an ambiguous error message. ........ + Merged revisions 402582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 18:53 +0000 [r402571-402572] David M. Lee <dlee@digium.com> + + * /, res/res_pjsip/config_system.c: res_pjsip: Print a helpful + error message if sorcery registration fails ........ Merged + revisions 402570 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_playbacks.h, /: Changes from make ari-stubs + after r402560 ........ Merged revisions 402561 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 17:59 +0000 [r402562] Kevin Harwell <kharwell@digium.com> + + * rest-api/resources.json, res/ari/resource_playback.h (removed), + res/res_ari_playbacks.c (added), res/ari/resource_playbacks.h + (added), /, res/ari.make, rest-api/api-docs/playback.json + (removed), res/ari/resource_playback.c (removed), + res/res_ari_playback.c (removed), + rest-api/api-docs/playbacks.json (added), + res/ari/resource_playbacks.c (added): ARI playback: Rename ARI + Playback to Playbacks Before playback was the only non plural + resource. It has been renamed to playbacks for consistency. + (closes issue ASTERISK-22737) Reported by: Paul Belanger ........ + Merged revisions 402560 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 17:29 +0000 [r402557] David M. Lee <dlee@digium.com> + + * res/res_ari.c, main/manager.c, /, main/http.c: ari: Add + application/x-www-form-urlencoded parameter support ARI POST + calls only accept parameters via the URL's query string. While + this works, it's atypical for HTTP API's in general, and + specifically frowned upon with RESTful API's. This patch adds + parsing for application/x-www-form-urlencoded request bodies if + they are sent in with the request. Any variables parsed this way + are prepended to the variable list supplied by the query string. + (closes issue ASTERISK-22743) Review: + https://reviewboard.asterisk.org/r/2986/ ........ Merged + revisions 402555 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 14:58 +0000 [r402546] Kevin Harwell <kharwell@digium.com> + + * apps/app_dahdiras.c, utils/extconf.c, main/asterisk.c: + app_dahdiras: Use waitpid instead of wait4. Several places in the + code were using wait4 while other places were using waitpid. This + change makes all places use waitpid in order to make things more + consistent and since the 'rusage' object passed in/out of wait4 + was never used. (closes issue ASTERISK-22557) Reported by: + YvesGael Patches: asterisk-11.5.1-wait4.patch uploaded by hurdman + (license 6537) + +2013-11-07 23:42 +0000 [r402538] Jonathan Rose <jrose@digium.com> + + * res/res_pjsip_authenticator_digest.c, /: PJSIP: Improve error + handling in digest authenticator Previously, regardless of + whether failure to authenticate was due to lacking any + authentication or actually failing authentication, the Digest + Authenticator would simply return that a challenge was still + needed. It will continue to do that when no authentication + information is in the received SIP digest, but when + authentication information is present and does not pass + authentication, that will be treated as an authentication error. + This is to ensure that PJSIP will issue security events indicated + failed auths. ........ Merged revisions 402537 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-07 21:10 +0000 [r402529] David M. Lee <dlee@digium.com> + + * res/ari/resource_applications.c, res/ari/resource_playback.c, + rest-api/api-docs/channels.json, res/ari/resource_applications.h, + res/ari/resource_channels.c, res/ari/resource_playback.h, + rest-api/api-docs/recordings.json, res/ari/resource_recordings.c, + rest-api-templates/ari_resource.c.mustache, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, rest-api/api-docs/endpoints.json, + res/ari/resource_endpoints.c, res/ari/resource_recordings.h, + res/ari/resource_events.c, res/res_ari_playback.c, + res/res_ari_applications.c, res/ari/resource_endpoints.h, + res/ari/resource_events.h, rest-api/api-docs/sounds.json, + res/ari/resource_sounds.c, res/res_ari_channels.c, + rest-api/api-docs/bridges.json, res/ari/resource_bridges.c, + res/ari/resource_sounds.h, res/res_ari_recordings.c, + res/ari/resource_bridges.h, rest-api/api-docs/asterisk.json, + res/ari/resource_asterisk.c, res/res_ari_endpoints.c, + rest-api/api-docs/applications.json, + rest-api/api-docs/playback.json, res/res_ari_events.c, + res/ari/resource_asterisk.h, rest-api-templates/swagger_model.py, + res/res_ari_sounds.c, res/res_ari_bridges.c, /, + rest-api-templates/ari_resource.h.mustache, + rest-api-templates/rest_handler.mustache, res/res_ari_asterisk.c, + rest-api-templates/res_ari_resource.c.mustache: ari: User better + nicknames for ARI operations While working on building client + libraries from the Swagger API, I noticed a problem with the + nicknames. channel.deleteChannel() channel.answerChannel() + channel.muteChannel() Etc. We put the object name in the nickname + (since we were generating C code), but it makes OO generators + redundant. This patch makes the nicknames more OO friendly. This + resulted in a lot of name changing within the res_ari_*.so + modules, but not much else. There were a couple of other fixed I + made in the process. * When reversible operations (POST /hold, + POST /unhold) were made more RESTful (POST /hold, DELETE + /unhold), the path for the second operation was left in the API + declaration. This worked, but really the two operations should + have been on the same API. * The POST /unmute operation had still + not been REST-ified. Review: + https://reviewboard.asterisk.org/r/2940/ ........ Merged + revisions 402528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-06 21:58 +0000 [r402518] Kevin Harwell <kharwell@digium.com> + + * /, apps/app_queue.c: app_queue: crash if first agent is "busy" If + the first agent/member (via CLI "queue show") in a queue is + "busy" (dnd, circuit busy, etc...) and no agents answered then + app_queue would crash. This occurred because while the calling of + agent(s) remained valid the channel on "busy" agent would be set + to NULL and then later dereferenced upon a second "rna" function + call. The original intention of the code is to have only valid + "call attempt" objects (channels != NULL) checked while + attempting to call agent(s). It does this by building a + "call_next" list of valid "call attempt" objects. In the case of + the "busy" agent subsequent builds of the valid "call attempt" + list would sometimes include (the case mentioned above) an + invalid "call attempt" object. The fix was to make sure the "call + attempt" list was appropriately built on every iteration. A NULL + sanity check was also added at the original offending spot of the + crash just in case another one slipped by somehow. (closes issue + ASTERISK-22644) Reported by: Marco Signorini Review: + https://reviewboard.asterisk.org/r/2983/ ........ Merged + revisions 402517 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-05 21:17 +0000 [r402502-402508] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: chan_sip: Use AST_AF* defined constant + when calling ast_get_ip While the structure passed to ast_get_ip + should be set memset to 0, thus initializing the ss_family member + to 0, explicitly setting it to AST_AF_UNSPEC is more portable. + ........ Merged revisions 402507 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix incorrect usage of + ast_get_ip involving uninitialized struct This started off as a + fix for the failing IAX2 acl_call test in the Asterisk Test + Suite. When inspecting why that test was failing, it became clear + that all attempts to bind to any local loopback address was + failing: [Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding + IAX2 to address 127.0.0.1:4569 [Nov 2 15:56:28] DEBUG[15787] + netsock2.c: Splitting '127.0.0.1' into... [Nov 2 15:56:28] + DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''. [Nov 2 + 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", + "(null)", ...): ai_family not supported [Nov 2 15:56:28] + WARNING[15787] acl.c: Unable to lookup '127.0.0.1' While there's + conceivably other ways for getaddrino to return EAI_FAMILY, the + most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not + provided as the desired family. The culprit was the call to + ast_get_ip, defined in acl.h. This function uses the family from + the passed in addr object (which it will also populate when it + returns!) when it eventually calls getaddrinfo. This patch fixes + the use of ast_get_ip that were not specifying the family in + chan_iax2. This prevents uninitialized use of the structure, so + that the addresses resolve correctly. Review: + https://reviewboard.asterisk.org/r/2991 ........ Merged revisions + 402505 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/acl.h, /, include/asterisk/netsock2.h: netsock2: + Define AST_AF_* enum constants to their AF_* equivalents This + patch explicitly defines AST_AF_* enum constants to their + sys/socket.h defined equivalents. It is certainly unclear why + these constants actually have to exist, given that netsock2.h + includes sys/socket.h; however, since the code base is already + liberally sprinkled with the usage of AST_AF_* (as well as with + direct calls to AF_*), this will at least keep the semantics + consistent between their usage across systems. ........ Merged + revisions 402503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, /: stasis_channels: Don't give preference + to ANI info in channel snapshots When publishing channel + snapshots, we currently compute the caller ID name and number by + giving preference first to ani.{name|number}, then to + id.{name|number}. However, when a channel driver (such as + chan_sip) updates the caller ID, it typically only updates the + caller ID stored in id.{name|number}. This means that we are + currently giving preference to stale information. When looking at + the rest of the code base, the only other place where we appear + to use this same logic is in app_amd. Everywhere else, we treat + the party information in ani as being separate to the party + information in id. This patch publishes only the caller ID name + and number in the snapshot field for caller_name and caller_num. + Note that the information in ANI is still available in + caller_ani. Review: https://reviewboard.asterisk.org/r/2992/ + ........ Merged revisions 402501 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-04 21:02 +0000 [r402453] Kevin Harwell <kharwell@digium.com> + + * /, channels/chan_sip.c: chan_sip: notify dialog info ignores + presentation indicator in callerid The presentation indicator in + a callerid (e.g. set by dialplan function + Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog + Info Notifies are generated during extension monitoring. Added a + check to make sure the name and/or number presentations on the + callee (remote identity) are set to allow. If they are restricted + then "anonymous" is used instead. (closes issue AST-1175) + Reported by: Thomas Arimont Review: + https://reviewboard.asterisk.org/r/2976/ ........ Merged + revisions 402450 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402452 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-02 04:30 +0000 [r402406-402439] Richard Mudgett <rmudgett@digium.com> + + * main/stasis.c, main/stasis_message_router.c, /, + include/asterisk/vector.h: vector: Uppercase API to follow C + convention. C does not support templates like C++. ........ + Merged revisions 402438 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/lock.h, main/stasis.c, + main/stasis_message_router.c, /, include/asterisk/vector.h: + vector: Update API to be more flexible. Made the vector macro API + be more like linked lists. 1) Added a name parameter to + ast_vector() to name the vector struct. 2) Made the API take a + pointer to the vector struct instead of the struct itself. 3) + Added an element cleanup macro/function parameter when removing + an element from the vector for ast_vector_remove_cmp_unordered() + and ast_vector_remove_elem_unordered(). 4) Added + ast_vector_get_addr() in case the vector element is not a simple + pointer. * Converted an inline vector usage in + stasis_message_router to use the vector API. It needed the API + improvements so it could be converted. * Fixed topic reference + leak in router_dtor() when the stasis_message_router is + destroyed. * Fixed deadlock potential in stasis_forward_all() and + stasis_forward_cancel(). Locking two topics at the same time + requires deadlock avoidance. * Made internal_stasis_subscribe() + tolerant of a NULL topic. * Made stasis_message_router_add(), + stasis_message_router_add_cache_update(), + stasis_message_router_remove(), and + stasis_message_router_remove_cache_update() tolerant of a NULL + message_type. * Promoted a LOG_DEBUG message to LOG_ERROR as + intended in dispatch_message(). Review: + https://reviewboard.asterisk.org/r/2903/ ........ Merged + revisions 402429 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_state_single.c, + apps/confbridge/conf_state_inactive.c, + apps/confbridge/conf_state_single_marked.c, /, + apps/confbridge/include/confbridge.h, + apps/confbridge/conf_state_multi.c, apps/app_confbridge.c, + apps/confbridge/conf_state_multi_marked.c, + apps/confbridge/conf_state.c: confbridge: Separate user muting + from system muting overrides. The system overrides the user + muting requests when MOH is playing or a waitmarked user is + waiting for a marked user to join. System muting overrides + interfere with what the user may wish the muting to be when the + system override ends. * User muting requests are now independent + of the system muting overrides. The effective muting is now the + logical or of the user request and system override. * Added a + Muted flag to the CLI "confbridge list <conference>" command. * + Added a Muted header to the AMI ConfbridgeList action + ConfbridgeList event. (closes issue AST-1102) Reported by: John + Bigelow Review: https://reviewboard.asterisk.org/r/2960/ ........ + Merged revisions 402425 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402427 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, apps/confbridge/conf_config_parser.c, + configs/confbridge.conf.sample, /: config: Allow ConfBridge DTMF + menus to have '#' as the first digit. ConfBridge allows custom + DTMF menus to be created in the confbridge.conf file by assigning + a DTMF key sequence to a sequence of actions as follows: + DTMF-sequence = action,action... Unfortunately, the normal config + file processing code interprets an initial '#' character as + starting a directive such as #include. * Add the ability to + escape the first non-blank character in a config line so the '#' + character can be used without triggering the directive processing + code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported + by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch + (license #5621) patch uploaded by rmudgett (modified) Review: + https://reviewboard.asterisk.org/r/2969/ ........ Merged + revisions 402407 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402416 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/app.h, /, main/app.c: voicemail: Simplify + callback pointer declarations and add doxygen. * Typedefed and + added doxegen for the voicemail callback functions. * Simplified + the prototypes for ast_install_vm_functions() and + ast_install_vm_test_functions() to use the new function typedefs. + * Simplified the voicemail callback function pointer variable + declarations to use the new function typedefs. ........ Merged + revisions 402398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 22:48 +0000 [r402397] Jonathan Rose <jrose@digium.com> + + * apps/confbridge/conf_config_parser.c, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + CHANGES: app_confbridge: Make the CONFBRIDGE function be able to + create dynamic menus Also adds the ability to clear all profile + items and makes behavior more consistent with documentation as + when choosing whether to use CONFBRIDGE datastore profiles or the + application arguments to the confbridge application. (closes + issue ASTERISK-22760) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2971/ + +2013-11-01 21:51 +0000 [r402388] Scott Griepentrog <sgriepentrog@digium.com> + + * main/manager_bridges.c, /, main/bridge.c, + include/asterisk/bridge.h: Manager: Add equivalent AMI actions + for the bridge CLI commands. Adds the following AMI events, + closely following their CLI counterparts: BridgeDestroy + BridgeKick BridgeTechnologyList BridgeTechnologySuspend + BridgeTechnologyUnsuspend BridgeDestroy kicks an entire bridge, + where BridgeKick kicks just one channel off the bridge. When + kicking a channel, specifying the bridge also (optional) insures + it is not removed from the wrong bridge. The BridgeTechnology + events allow viewing and changing suspension status, which + affects only subsequent not active bridging. (closes + ASTERISK-22356) Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/2973/ ........ Merged + revisions 402387 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 16:31 +0000 [r402368] David M. Lee <dlee@digium.com> + + * /, rest-api-templates/api.wiki.mustache: ari wiki docs: add notes + about allowMultiple parameters. This patch adds a note to any + parameter that has 'allowMultiple' set in the Swagger + documentation. (closes issue ASTERISK-22704) ........ Merged + revisions 402367 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 14:38 +0000 [r402359] Joshua Colp <jcolp@digium.com> + + * include/asterisk/stasis_app.h, rest-api/api-docs/channels.json, + res/ari/resource_channels.c, res/res_ari_channels.c, + res/ari/resource_channels.h, res/res_stasis_playback.c, /, + res/stasis/control.c: res_ari_channels: Add ring operation, dtmf + operation, hangup reasons, and tweak early media. The ring + operation sends ringing to the specified channel it is invoked + on. The dtmf operation can be used to send DTMF digits to the + specified channel of a specific length with a wait time in + between. Finally hangup reasons allow you to specify why a + channel is being hung up (busy, congestion). Early media behavior + has also been tweaked slightly. When playing media to a channel + it will no longer automatically answer. If it has not been + answered a progress indication is sent instead. (closes issue + ASTERISK-22701) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2916/ ........ Merged + revisions 402358 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 12:40 +0000 [r402349] Kinsey Moore <kmoore@digium.com> + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c, + include/asterisk/rtp_engine.h: chan_sip: Fix RTCP port for SRFLX + ICE candidates This corrects one-way audio between Asterisk and + Chrome/jssip as a result of Asterisk inserting the incorrect RTCP + port into RTCP SRFLX ICE candidates. This also exposes an ICE + component enumeration to extract further details from candidates. + (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: + https://reviewboard.asterisk.org/r/2967/ ........ Merged + revisions 402345 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402348 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 12:33 +0000 [r402337-402347] Joshua Colp <jcolp@digium.com> + + * /, include/asterisk/stasis_app.h, res/ari/resource_channels.c: + res_ari_channels: Fix a deadlock when originating multiple + channels close to eachother. If a Stasis application is specified + an implicit subscription is done on the originated channel. This + was previously done with the channel lock held which is dangerous + as the underlying code locks the container and iterates items. + This change releases the lock on the originated channel before + subscribing occurs. (closes issue ASTERISK-22768) Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/2979/ + ........ Merged revisions 402346 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c: res_stasis: Ensure the channel is always + departed from the bridge when it leaves. This change adds a + command to the command queue to explicitly depart the channel + from the bridge when it is told it has left. If the channel has + already been departed or has entered a different bridge this + command will become a no-op. (closes issue ASTERISK-22703) + Reported by: John Bigelow (closes issue ASTERISK-22634) Reported + by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2965/ ........ Merged + revisions 402336 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-31 22:09 +0000 [r402328] Mark Michelson <mmichelson@digium.com> + + * /, contrib/scripts/sip_to_pjsip/sip_to_pjsip.py, + contrib/scripts/sip_to_res_sip (removed), + contrib/scripts/sip_to_pjsip (added), + contrib/scripts/sip_to_pjsip/astconfigparser.py, + contrib/scripts/sip_to_pjsip/astdicts.py: Update the conversion + script from sip.conf to pjsip.conf (closes issue ASTERISK-22374) + Reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/2846 ........ Merged revisions + 402327 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-31 16:06 +0000 [r402286-402290] Matthew Jordan <mjordan@digium.com> + + * main/loader.c, /: core/loader: Don't call dlclose in a while loop + For awhile now, we've noticed continuous integration builds + hanging on CentOS 6 64-bit build agents. After resolving a number + of problems with symbols, strange locks, and other shenanigans, + the problem has persisted. In all cases, gdb shows the Asterisk + process stuck in loader.c on one of the infinite while loops that + calls dlclose repeatedly until success. The documentation of + dlclose states that it returns 0 on success; any other value on + error. It does not state that repeatedly calling it will + eventually clear those errors. Most likely, the repeated calls to + dlclose was to force a close by exhausting the references on the + library; however, that will never succeed if: (a) There is some + fundamental error at work in the loaded library that precludes + unloading it (b) Some other loaded module is referencing a symbol + in the currently loaded module This results in Asterisk sitting + forever. Since we have matching pairs of dlopen/dlclose, this + patch opts to only call dlclose once, and log out as an ERROR if + dlclose fails to return success. If nothing else, this might help + to determine why on the CentOS 6 64-bit build agent things are + not closing successfully. Review: + https://reviewboard.asterisk.org/r/2970 ........ Merged revisions + 402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 402288 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402289 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/media_index.c, /: medix_index: Display errors when library + calls fail Based on feedback from ipengineer in #asterisk, when + the media indexer cannot access a sound file on the system (or + otherwise fails) Asterisk displays a "Cannot frob file" error but + fails to tell you why. This is especially problematic as the + media_indexer failing will rpevent Asterisk from starting, as it + is in the core. We now display the errno error messages so folks + can figure out what they've done wrong. ........ Merged revisions + 402285 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-31 14:45 +0000 [r402277] David M. Lee <dlee@digium.com> + + * /, res/stasis/app.c: stasis: add functions embarrassingly missing + from r400522 I neglected to implement two of the endpoint + subscription functions when I did the work. Normally, you'll only + hit that when you unsubscribe from a specific endpoint. ........ + Merged revisions 402276 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-30 17:54 +0000 [r402266] Kevin Harwell <kharwell@digium.com> + + * channels/chan_pjsip.c, /, res/res_pjsip_messaging.c: + pjsip_messaging: Added debug for in dialog messaging (issue + ASTERISK-22777) Reported by: Matt Jordan ........ Merged + revisions 402265 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 23:43 +0000 [r402227] Rusty Newton <rnewton@digium.com> + + * /, sounds/Makefile: Updates for 1.4.25 core sounds and 1.4.14 + extra sounds, plus new en_GB language set The new sound packages + relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, + ASTERISK-20782 Modified sounds/Makefile for the new sound + versions and to account for the new en_GB language set. (issue + ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue + ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged + revisions 402224 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402225 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 12:57 +0000 [r402155] Matthew Jordan <mjordan@digium.com> + + * main/xmldoc.c, main/channel.c, main/pbx.c, /, main/translate.c: + Remove some spammy debug messages; improve clarity of others + Debug messages aren't free. Even when the debug level is + sufficiently low such that the messages are never evaluated, + there is a cost to having to parse Asterisk logs that contain + debug messages that (a) fail to convey sufficient information or + (b) occur so frequently as to be next to meaningless. Based on + having to stare at lots of DEBUG messages, this patch makes the + following changes: * channel.c: When copying variables from a + parent channel to a child channel, specify the channels involved. + Do not log anything for a variable that is not inherited; the + fact that it doesn't have an _ or __ already signifies that it + won't be inherited. * pbx.c: Specify what function evaluation has + occurred that created the result. * translate.c: Bump up the + translator path messages to 10. I've never once had to use these + debug messages, and for each format that is registered (on + startup) and unregistered (on shutdown) the entire f^2 matrix is + logged out. For short tests in the Asterisk Test Suite, this + should make finding the actual test much easier. * xmldoc.c: The + debug message that 'blah' is not found in the tree is expected. + Often, description elements - which are not required - are not + provided. This debug message adds no additional value, as it is + not indicative of an error or helpful in debugging which element + did not contain a 'blah' element as a child. If an element is + supposed to contain a child element, then that XML tree should + have failed validation in the first place. Review: + https://reviewboard.asterisk.org/r/2966/ ........ Merged + revisions 402150 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402151 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402154 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 12:51 +0000 [r402149-402153] Kinsey Moore <kmoore@digium.com> + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /: ARI: + Remove channels/{channelId}/dial This removes the + /ari/channels/{channelId}/dial URI since it is redundant, overly + complex, is likely to become more externally complex over time, + and is too high-level compared with other ARI operations. See the + following for further information: + http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html + (closes issue ASTERISK-22784) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2968/ ........ Merged + revisions 402152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * bridges/bridge_native_rtp.c, /: bridge_native_rtp: Ensure bridge + is torn down When a bridge transitions away from one tech to + another, the tech going away is provided a dummy bridge with no + channels in it to tear down. Currently this means that the + teardown code exits prematurely and does not tear anything down. + This change tears down RTP bridging for the channel provided in + the leave bridge tech callback. This also reverts the majority of + r400403 since it is now redundant. (closes issue ASTERISK-22628) + (closes issue ASTERISK-22676) Reported by: John Bigelow Reported + by: Kevin Harwell Tested by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2905/ Patches: + native_rtp_fix.diff uploaded by Kinsey Moore (License 6273) + ........ Merged revisions 402148 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 11:15 +0000 [r402140] Joshua Colp <jcolp@digium.com> + + * /, rest-api/api-docs/playback.json, res/res_ari_playback.c: + res_ari_playback: Add missing 404 error response for GET and + DELETE. (closes issue ASTERISK-22722) Reported by: Richard + Mudgett ........ Merged revisions 402139 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-28 22:10 +0000 [r402128-402130] David M. Lee <dlee@digium.com> + + * /, doc: Ignore full docs ........ Merged revisions 402127 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Put back several merge revisions that were lost in r402054 + + * /: Put back several merge revisions that were lost in r401962 + +2013-10-28 15:08 +0000 [r402113-402117] Michael L. Young <elgueromexicano@gmail.com> + + * /, UPGRADE-11.txt, UPGRADE-12.txt: Fix UPGRADE.txt Due To Merging + From Branch 11 When merging in the patch for ASTERISK-22728, the + UPGRADE.txt file was changed incorrectly. That change should have + gone into ASTERISK-11.txt. This commit is to fix that. Also, + another comment in the UPGRADE-11.txt was missing and this commit + adds that as well. ........ Merged revisions 402115 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c, UPGRADE-12.txt: chan_sip: Clarify + 'Forcerport' Setting Displayed When Running "sip show peers" + While looking at ASTERISK-22236, Walter Doekes pointed out that + when running "sip show peers", the setting being displayed can be + confusing. The display of "N" used to mean NAT (i.e. yes). The + NAT setting has gone through many different changes resulting in + the display of different characters to try and convey what the + current setting is for 'Forcerport' (A for Auto and Forcerport is + currently on, a for Auto but Forcerport is off, Y for yes, and N + for no). During the initial code review to try and clarify these + settings (especially since "N" no longer meant what it used to + mean in prior versions of Asterisk), Mark Michelson suggested + using the full space available to display the settings which + helped to make the settings very clear. That was a great + suggestion. Therefore, this patch does the following: * The + column for 'Forcerport' now will show: Auto (Yes), Auto (No), + Yes, or No. * A column for the 'Comedia' setting has been added. + It too will display the setting in a non-cryptic way: Auto (Yes), + Auto (No), Yes, or No. * UPGRADE.txt has been updated to document + this change. (closes issue ASTERISK-22728) Reported by: Walter + Doekes Tested by: Michael L. Young Patches: + asterisk-forcerport-display-clarification_v3.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2941 ........ Merged revisions + 402111 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 402112 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-27 23:22 +0000 [r402073-402091] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /: Filter out internal channels from dial message + handling Surrogate channels would pop up from time to time in + dial message handling. This would cause a WARNING message to + appear, indicating that the Surrogate channel had no CDR. This + patch filters out those channels that have the internal + implementation flag set, such that the WARNING message isn't + displayed. ........ Merged revisions 402090 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * cdr/cdr_sqlite3_custom.c, /, cdr/cdr_syslog.c, cdr/cdr_sqlite.c, + cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, + include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, + cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, + cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c: Prevent CDR backends + from unregistering while billing data is in flight This patch + makes it so that CDR backends cannot be unregistered while active + CDR records exist. This helps to prevent billing data from being + lost during restarts and shutdowns. Review: + https://reviewboard.asterisk.org/r/2880/ ........ Merged + revisions 402081 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, contrib/ast-db-manage/config/env.py, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py, + contrib/ast-db-manage/voicemail/env.py: Update Alembic database + scripts for external scripting and PostgreSQL, Oracle This patch + does the following: 1) The env scripts have been updated to be + tolerant of a NULL configuration file. This occurs when + configuration is provided by an external script, such that the + actual config.ini file is not used. 2) Enum types have all been + given names. This is needed for PostgreSQL script generation. 3) + The identifier meetme_confno_starttime_endtime is greater than 30 + characters, and hence invalid for Oracle databases. This has been + truncated down to meetme_confno_start_end. ........ Merged + revisions 400383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-26 12:56 +0000 [r402065] Joshua Colp <jcolp@digium.com> + + * channels/chan_pjsip.c, include/asterisk/res_pjsip_session.h, /: + chan_pjsip: Fix a crash when direct media is enabled and an ACK + is received after the channel is hung up. (closes issue + ASTERISK-22731) Reported by: Kinsey Moore ........ Merged + revisions 402064 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-26 00:36 +0000 [r402056] Richard Mudgett <rmudgett@digium.com> + + * res/res_stasis.c, /: res_stasis.c: Made use the ao2_container + callback templates. * Made res_stasis.c use the OBJ_SEARCH_XXX + defines. ........ Merged revisions 402055 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-26 00:27 +0000 [r402054] Scott Griepentrog <sgriepentrog@digium.com> + + * main/rtp_engine.c, /, include/asterisk/rtp_engine.h: rtp_engine: + fix rtp payloads copy and improve argument names In function + ast_rtp_instance_early _bridge_make_compatible the use of + instance 0/1 as arguments doesn't clearly communicate a direction + that the copying of payloads from the source channel to the + destination channel will occur, making it more probable to have + the arguments to ast_rtp_codecs_payloads_copy() put in the + reverse order. This patch renames the arguments with _dst and + _src suffixes and corrects the copy direction. (closes issue + ASTERISK-21464) Reported by: Kevin Stewart Review: + https://reviewboard.asterisk.org/r/2894/ ........ Merged + revisions 402000 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows + rtpmap:119 being copied per this change, but is not in sip invite + ........ Merged revisions 402042 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402043 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 23:58 +0000 [r402004-402045] Richard Mudgett <rmudgett@digium.com> + + * /, main/taskprocessor.c: taskprocessor: Made use pthread_equal() + to compare thread ids. * Removed another silly use of RAII_VAR(). + RAII_VAR() and SCOPED_LOCK() are not silver bullets that allow + you to turn off your brain. ........ Merged revisions 402044 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/app.c: You'd think that new files would be free of + whitespace issues. But you would be wrong. ........ Merged + revisions 402003 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 22:01 +0000 [r401999-402002] Jonathan Rose <jrose@digium.com> + + * res/ari/resource_bridges.c, res/res_ari_bridges.c, /, + rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, rest-api/api-docs/bridges.json: ARI: + channel/bridge recording errors when invalid format specified + Asterisk will now issue 422 if recording is requested against + channels or bridges with an unknown format (closes issue + ASTERISK-22626) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2939/ ........ Merged + revisions 402001 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_recording.c, rest-api/api-docs/channels.json, + res/ari/resource_channels.c, res/ari/ari_model_validators.c, + res/res_ari_channels.c, rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, res/ari/resource_bridges.c, + res/ari/ari_model_validators.h, res/res_ari_bridges.c, + rest-api/api-docs/events.json, /: ARI recordings: Issue HTTP + failures for recording requests with file conflicts If a file + already exists in the recordings directory with the same name as + what we would record, issue a 422 instead of relying on the + internal failure and issuing success. (closes issue + ASTERISK-22623) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2922/ ........ Merged + revisions 401973 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 20:51 +0000 [r401962] Scott Griepentrog <sgriepentrog@digium.com> + + * include/asterisk/pbx.h, main/pbx.c, /: pbx.c: fix confused match + caller id that deleted exten still in hash This fixes a bug where + a zero length callerid match adjacent to a no match callerid + extension entry would be deleted together, which then resulted in + hashtable references to free'd memory. A third state of the + matchcid value has been added to indicate match to any extension + which allows enforcing comparison of matchcid on/off without + errors. (closes issue AST-1235) Reported by: Guenther Kelleter + Review: https://reviewboard.asterisk.org/r/2930/ ........ Merged + revisions 401959 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401960 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401961 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 17:41 +0000 [r401898-401939] Jonathan Rose <jrose@digium.com> + + * /, res/res_pjsip/pjsip_distributor.c, + res/res_pjsip_endpoint_identifier_user.c: PJSIP: Add log messages + when requests are received for non-existent endpoints (closes + issue ASTERISK-22552) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2934/ ........ Merged + revisions 401938 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * utils/clicompat.c, utils/refcounter.c, /: Put clicompat-r2.patch + back in We've figured out how to resolve the problems this was + causing in 12/trunk, so this can go back in now. (issue + ASTERISK-22467) Reported by: Corey Farrell Patches: + clicompat-r2.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 401914 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401935 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401936 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, utils/clicompat.c: revert clicompat-r2.patch from r401704 + Patch caused the following build errors against testsuite + https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244 + (issue ASTERISK-22467) Reported by: Corey Farrell ........ Merged + revisions 401895 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401896 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401897 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 16:09 +0000 [r401886] Kevin Harwell <kharwell@digium.com> + + * /, channels/chan_sip.c: chan_sip: Allow a sip peer to accept both + AVP and AVPF calls Adapts the behaviour of avpf to only impact + the format of outgoing calls. For inbound calls, both AVP and + AVPF calls will be accepted regardless of the value of avpf in + the configuration. (closes issue ASTERISK-22005) Reported by: + Torrey Searle Patches: optional_avpf_trunk.patch uploaded by + tsearle (license 5334) ........ Merged revisions 401884 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401885 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 13:49 +0000 [r401873] David M. Lee <dlee@digium.com> + + * tests/test_json.c, /: test_json: Fix deprecation warnings After a + series of upgrades over recent weeks, I've discovered that + test_json.c won't compile in dev mode any more for me. One of + gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate + tempnam. Which, in general, is a good thing. But for test code + that just needs a temporary file, it's just annoying. This patch + replaces usage of tempname with mkstemp, avoiding the deprecation + warning. It also removes the temporary files when the test is + complete, which apparently we weren't doing before (oops). + Review: https://reviewboard.asterisk.org/r/2957/ ........ Merged + revisions 401872 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-24 21:06 +0000 [r401836] Kevin Harwell <kharwell@digium.com> + + * /, main/logger.c: Logging: Logging types ignored after specifying + a verbose level If one specified a verbose level within a logging + facility in logger.conf then any component after it was ignored. + Fixed so all values are correctly read. (closes issue + ASTERISK-22456) Reported by: Kevin Harwell ........ Merged + revisions 401833 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401835 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-24 20:48 +0000 [r401834] David M. Lee <dlee@digium.com> + + * rest-api-templates/models.wiki.mustache, + rest-api/api-docs/events.json, /, + rest-api-templates/swagger_model.py, + rest-api-templates/ari_model_validators.c.mustache: The Swagger + 1.2 specification for type extension ended up being slightly + different than my proposal. Instead of putting an 'extends' field + on the subtype, the base type has a 'subTypes' field, which is a + list of the subTypes. Given that its a messaging model and not an + object model, kinda makes sense. This patch changes the + events.json api-doc, and the python translators to take the new + format into account. Other changes that are in Swagger 1.2 were + not adopted, since the spec is still in flux, and could change + before it's finalized. A summary of changes to the Swagger-1.2 + spec can be found at + https://github.com/wordnik/swagger-core/wiki/1.2-transition. + (closes issue ASTERISK-22440) Review: + https://reviewboard.asterisk.org/r/2909/ ........ Merged + revisions 401701 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-24 20:34 +0000 [r401622-401832] Jonathan Rose <jrose@digium.com> + + * /, main/utils.c: utils: Fix memory leaks and missed + unregistration of CLI commands on shutdown Final set of patches + in a series of memory leak/cleanup patches by Corey Farrell + (closes issue ASTERISK-22467) Reported by: Corey Farrell Patches: + main-utils-1.8.patch uploaded by coreyfarrell (license 5909) + main-utils-11.patch uploaded by coreyfarrell (license 5909) + main-utils-12up.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 401829 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401830 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401831 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_linkedlists.c: test_linkedlists: Fix memory leak + (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + test_linkedlists-1.8.patch uploaded by coreyfarrell (license + 5909) test_linkedlists-11up.patch uploaded by coreyfarrell + (license 5909) ........ Merged revisions 401790 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401791 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401792 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer + reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + jitterbuf-jb_reset-leak-1.8.patch + jitterbuf-jb_reset-leak-11up.patch ........ Merged revisions + 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 401787 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401788 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/astobj2.c, /: astobj2: Unregister debug CLI commands at exit + (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell + (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 401781 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401783 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401784 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_voicemail.c, /: app_voicemail: Memory Leaks against + tests (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909) + app_voicemail-11up.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 401743 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401744 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401745 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/app.c, main/asterisk.c, utils/clicompat.c, + channels/chan_dahdi.c, codecs/ilbc/doCPLC.c, main/data.c, /: + memory leaks: Memory leak cleanup patch by Corey Farrell (second + set) Also covers ast_app_parse_timelen-fail-zero-length.patch, + but the patch was replaced with one of my own. (issue + ASTERISK-22467) Reported by: Corey Farrell Patches: + chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license + 5909) clicompat-r2.patch uploaded by coreyfarrell (license 5909) + codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909) + data-cleanup-test-registration.patch uploaded by coreyfarrell + (license 5909) main-asterisk-kill-listener.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 401704 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401705 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401706 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_dlinklists.c, funcs/func_math.c, + channels/sip/reqresp_parser.c, main/test.c, + main/editline/readline.c: memory leaks: Memory leak cleanup patch + by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by: + Corey Farrell Patches: + chan_sip-parse_contact_header_test-free-contacts.patch uploaded + by coreyfarrell (license 5909) cli-filename-completion-leak.patch + uploaded by coreyfarrell (license 5909) func_math.patch uploaded + by corefarrell (license 5909) main-test-cleanup.patch uploaded by + coreyfarrell (license 5909) test_dlinklists.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 401660 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401661 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401662 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk: + Address jittery DTMF events in RTP streams (closes issue + ASTERISK-21170) Reported by: NITESH BANSAL Patches: + dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418) + Review: https://reviewboard.asterisk.org/r/2938/ ........ Merged + revisions 401619 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401620 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401621 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 16:52 +0000 [r401582] Richard Mudgett <rmudgett@digium.com> + + * /, cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a + filter when the CDR value is empty. Extra CDR records are written + if a filtered CDR value is empty because the filter is not + checked. (closes issue ASTERISK-22272) Reported by: Jordi Llull + Chavarria ........ Merged revisions 401577 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401579 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401581 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 16:48 +0000 [r401580] John Bigelow <jbigelow@digium.com> + + * /, main/bridge_channel.c: Add a test suite event to indicate when + the atxfer 3-way feature is detected This adds a test suite event + that indicates to tests when the attended transfer three-way call + feature is detected. Review: + https://reviewboard.asterisk.org/r/2912/ ........ Merged + revisions 401578 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 15:23 +0000 [r401540] Kinsey Moore <kmoore@digium.com> + + * channels/chan_mgcp.c, /: chan_mgcp: Properly handle malformed + media lines This corrects a situation in which a media line was + not parsed properly and resulted in a crash. (closes issue + ASTERISK-21190) Reported by: adomjan Patches: + chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448) + ........ Merged revisions 401537 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401538 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401539 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 11:16 +0000 [r401500] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: chan_sip: Fix an issue where an + incompatible audio format may be added to SDP. If preferred + codecs included any non-audio format the code would mistakenly + add the audio format, even if it was not a joint capability with + the remote side. (closes issue ASTERISK-21131) Reported by: + nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by + nbougues (license 6470) ........ Merged revisions 401497 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401498 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401499 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 02:36 +0000 [r401489] Michael L. Young <elgueromexicano@gmail.com> + + * channels/chan_iax2.c, configs/iax.conf.sample, /: chan_iax2: Fix + Binding To Multiple Addresses Again When reworking chan_iax2 for + IPv6, the ability to bind to multiple addresses was removed by + mistake. This patch restores this functionality and adds notes + about IPv6 addresses in the sample config. (closes issue + ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L. + Young Patches: asterisk-22741-fix-binding-multiple-addr.diff + uploaded by Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2945/ ........ Merged + revisions 401488 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-22 23:10 +0000 [r401450] Matthew Jordan <mjordan@digium.com> + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP + is not available during SSRC change In r400089, a patch was put + in to correct erroneous RTCP statistic resets. Unfortunately, + ast_rtp_read can be called on an RTP instance that does not have + RTCP information. This patch prevents that crash by only + resetting the statistics if we do actually have an RTCP instance. + (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John + Bigelow ........ Merged revisions 401445 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401446 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401447 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-22 19:04 +0000 [r401421-401435] Richard Mudgett <rmudgett@digium.com> + + * apps/app_queue.c, /: app_queue: Fix CLI "queue remove member" + queue_log entry. The queue_log entry resulting from CLI "queue + remove member" when log_membername_as_agent is enabled is wrong. + It always uses the interface name instead of the member name in + the queue_log entry. * Get the queue member before removing it + from the queue so the member name is available for the queue_log + entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve + Patches: fix_membername.diff (license #6505) patch uploaded by + Oscar Esteve (modified to fix potential ref leak) ........ Merged + revisions 401433 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401434 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/bridge_channel.c, + include/asterisk/bridge_channel_internal.h, /, main/bridge.c: + Bridging: Fix orphaned bridge if neither of the joining channels + can join. The original issue noted that the bridge is orphaned + when res_parking.so is not loaded and a call uses the dial kK + flags. A similar issue happens when only one of the park flags is + used. In this case you have the bridge with one or the other + channel left in it. The channel and bridge will stay around until + the channel hangs up. * Fixed the initial bridge channel push + failure to act as if the channel were kicked out of the bridge. + The bridge then decides if it needs to be dissolved. (closes + issue ASTERISK-22629) Reported by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2928/ ........ Merged + revisions 401424 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/parking/parking_bridge_features.c, + res/parking/parking_bridge.c, /: res_parking: Give parking + timeout comebacktoorigin channel DTMF features. Parking timeouts + did not set any DTMF features for the channel calling the parker + back. * Added code to set the parkedcalltransfers, + parkedcallreparking, parkedcallhangup, and parkedcallrecording + options appropriately for the channels when a parking timeout + occurs. The recall channel DTMF options are set using the + BRIDGE_FEATURES channel variable to allow the other timeout + options to have the DTMF features available. (closes issue + ASTERISK-22630) Reported by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2942/ ........ Merged + revisions 401422 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_parking.c: res_parking: Update XML documention for + DTMF features after parking timeout. * Updated the XML + documentation to indicate that the parkedcalltransfers, + parkedcallreparking, parkedcallhangup, and parkedcallrecording + configuration options also apply to parking timeouts. (issue + ASTERISK-22630) Reported by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2942/ ........ Merged + revisions 401420 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-22 15:17 +0000 [r401411] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c: Add an 'R' option to Dial which sends ringing + until early media has been received. (closes issue + ASTERISK-10487) Reported by: Gaspar Zoltan Patches: 10487.patch + uploaded by n8ideas (license 6075) + +2013-10-21 21:06 +0000 [r401365] Mark Michelson <mmichelson@digium.com> + + * /, main/bridge_channel.c: Remove a noisy debug message from + bridging code. This particular debug message, during a stress + test, was logged so often that it appeared that there may be a + memory leak in the logger code. In actuality, there was no memory + leak, but the logger thread was having a hard time keeping up + with the demands of the rest of the system. Since this debug + message has no value at all, the best way to fix the problem was + to just remove the message. (closes issue AST-1225) reported by + John Bigelow Patches: spammy_log.diff uploaded by Mark Michelson + (License #5049) ........ Merged revisions 401364 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-21 19:50 +0000 [r401328] Kevin Harwell <kharwell@digium.com> + + * /, main/editline/term.c: Segfault in LIBEDIT_INTERNAL after + tgetstr(), when libncurses5-dev isn't installed Include the + appropriate declarations when not using termcap, but term+curses + and [n]curses do not exist. (closes issue ASTERISK-22351) + Reported by: A. Iglesias Patches: + issueA22351_libedit_internal_without_ncurses_dev.patch uploaded + by wdoekes (license 5674) ........ Merged revisions 401325 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401326 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401327 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-21 18:59 +0000 [r401316-401317] David M. Lee <dlee@digium.com> + + * rest-api/api-docs/channels.json, /: Fixing r401281; the model + name is Channel, with a capital C ........ Merged revisions + 401315 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_ari.c, /: Fixed malformed Access-Control-Allow-Methods + header. Was causing Safari to barf on POST and DELETE. ........ + Merged revisions 401106 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-19 21:57 +0000 [r401292] Kinsey Moore <kmoore@digium.com> + + * /, channels/chan_iax2.c: Fix IAX2 incoming call address lookups + This fixes address lookup for incoming calls without a peer + definition. The address family was unset instead of being set to + AST_AF_UNSPEC which was causing lookup failures on "127.0.0.1". + This is one of the causes of the current failure of the app_page + integration test. Review: + https://reviewboard.asterisk.org/r/2933/ ........ Merged + revisions 401291 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-19 14:45 +0000 [r401282] Joshua Colp <jcolp@digium.com> + + * res/ari/resource_channels.h, main/pbx.c, /, + rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c: Return a channel snapshot when + originating using ARI, and subscribe the Stasis application to + it. This change allows a user of ARI to know what channel it has + originated and also follow any progress. If a Stasis application + is provided it will be automatically subscribed to the originated + channel immediately. (closes issue ASTERISK-22485) Reported by: + David Lee Review: https://reviewboard.asterisk.org/r/2910/ + ........ Merged revisions 401281 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 22:52 +0000 [r401272] Richard Mudgett <rmudgett@digium.com> + + * /, res/parking/parking_controller.c: res_parking: Remove setting + useless flag. ........ Merged revisions 401271 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 21:51 +0000 [r401263] David M. Lee <dlee@digium.com> + + * contrib/scripts/get_swagger_ui.sh (added), Makefile, /, + static-http: This is just a quick script for dumping swagger-ui + into static-http, so that it can be served by the Asterisk web + server. I had to change the Makefile in order to recursively + install content from the static-http directory, hence the code + review instead of just putting it in. Review: + https://reviewboard.asterisk.org/r/2924/ ........ Merged + revisions 401261 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 18:44 +0000 [r401249] Mark Michelson <mmichelson@digium.com> + + * main/sorcery.c, main/cli.c, main/manager.c, /, main/bridge.c, + main/bucket.c: Resolve some memory leaks due to incorrect for + loop / ao2 ref usage. A common idiom in Asterisk is to due + something like: for (ao2_obj = list_beginning; ao2_obj = + next_item; ao2_ref(ao2_obj, -1)) { ...do stuff... } This is nice + because it automatically takes care of the object references for + you. However, there is a pitfall here. If a break statement is in + the for loop, then the current reference is not cleaned up. In + some cases, this is on purpose, but in others there is a leak. + This commit fixes the leak cases. ........ Merged revisions + 401248 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 16:59 +0000 [r401233-401240] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_fax.c, include/asterisk/channel.h, apps/app_dial.c, + main/channel.c: Add channel lock protection around translation + path setup. Most callers of ast_channel_make_compatible() happen + before the channels enter a two party bridge. With the new + bridging framework, two party bridging technologies may also call + ast_channel_make_compatible() when there is more than one thread + involved with the two channels. * Added channel lock protection + in set_format() and ast_channel_make_compatible_helper() when + dealing with the channel's native formats while setting up a + translation path. * Fixed best_src_fmt and best_dst_fmt usage + consistency in ast_channel_make_compatible_helper(). The call to + ast_translator_best_choice() got them backwards. * Updated some + callers of ast_channel_make_compatible() and the function + documentation. There is actually a difference between the two + channels passed in. * Fixed the deadlock potential in res_fax.c + dealing with ast_channel_make_compatible(). The deadlock + potential was already there anyway because res_fax called + ast_channel_make_compatible() with chan locked. (closes issue + ASTERISK-22542) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2915/ ........ Merged + revisions 401239 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, include/asterisk/bridge.h: Tweak ast_bridge_depart() doxygen. + ........ Merged revisions 401232 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 16:06 +0000 [r401216-401224] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/bridge.h, /: Remove the bit about requiring + ast_bridge_depart() to be called before ast_bridge_destroy(). + ........ Merged revisions 401223 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/bridge.h, /: Clarify in ast_bridge_destroy() + about how departable channels must be handled. ........ Merged + revisions 401212 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 15:14 +0000 [r401184] Michael L. Young <elgueromexicano@gmail.com> + + * /, channels/chan_sip.c: Remove Port Restriction When Checking For + NAT When trying to determine if a peer is behind NAT, we should + not be using the ports when comparing addresses. This patch + removes the port from being checked and just useds the addresses + now. (closes issue ASTERISK-22729) Reported by: Michael L. Young + Tested by: Michael L. Young Patches: + asterisk-remove-using-port-for-nat-check.diff uploaded by Michael + L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2927/ ........ Merged + revisions 401182 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401183 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 14:50 +0000 [r401181] Walter Doekes <walter+asterisk@wjd.nu> + + * main/channel.c, /: Properly copy/remove the device state cache + flag over a masquerade. In r378303 the + AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the + devstate system to not cache states for non-real devices. + However, when optimizing away channels (ast_do_masquerade), that + flag wasn't copied. In my case, using Local devices as queue + members created a situation where the endpoint was considered in + use, but the state change of the device being available again was + ignored (not cached). The endpoint channel was optimized into the + (previously) Local channel, but kept the do-not-cache flag. The + end result being that the queue member apparently stayed in use + forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes + Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged + revisions 401178 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401179 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401180 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-17 20:39 +0000 [r401169] Michael L. Young <elgueromexicano@gmail.com> + + * /, channels/chan_sip.c: Fix Setting A chan_sip Dialog's + SIP_NAT_FORCE_RPORT Flag A condition was added in a commit to fix + ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was + set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the + dialog. This condition should not have been there since it + assumed that if Asterisk is in an environment where NAT is + involved, that the auto_* nat settings or force_rport setting + would be on in the global settings. If the nat setting in the + global setting is set to 'nat=no' and then turned on for peers + (which is not quite the recommended way, although it is allowed) + this flag is never copied to the dialog resulting in problems + like, REGISTER replies going to the wrong port. This patch + removes this conditional check and will now always use the peer's + flag which by this point in the code the checks on whether the + peer is behind NAT or not (if using auto_force_rport) have + already been run. (closes issue ASTERISK-22236) Reported by: + Filip Frank Tested by: Michael L. Young Patches: + asterisk-2236-always-set-rport.diff uploaded by Michael L. Young + (license 5026) Review: https://reviewboard.asterisk.org/r/2919/ + ........ Merged revisions 401167 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401168 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-17 18:25 +0000 [r401159] Jonathan Rose <jrose@digium.com> + + * res/res_parking.c, /: res_parking: Fix bug where reloading + immediately wipes new parkpos extensions (closes issue + ASTERISK-22631) Reported by: Kevin Harwell ........ Merged + revisions 401158 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-17 15:41 +0000 [r401122] Kinsey Moore <kmoore@digium.com> + + * /, res/res_xmpp.c, res/res_jabber.c: Reduce log level of a + non-pubsub error message Drop an error log message to debug level + 1 since distributed device state functions correctly when + receiving this message and it spams the logs. (closes issue + ASTERISK-22410) Reported by: abelbeck Patches: + asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch + uploaded by abelbeck (License 5903) + asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded + by abelbeck (License 5903) ........ Merged revisions 401119 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401120 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401121 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 21:22 +0000 [r401108] Richard Mudgett <rmudgett@digium.com> + + * /, res/ari/resource_playback.c: ARI: Fix crash when POST + /playback/{id}/control does not have an operation parameter. + (closes issue ASTERISK-22680) Reported by: John Bigelow ........ + Merged revisions 401107 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 17:01 +0000 [r401097] David M. Lee <dlee@digium.com> + + * rest-api/resources.json, /: Oops. Leftover /stasis reference + ........ Merged revisions 401096 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 14:02 +0000 [r401088] Kinsey Moore <kmoore@digium.com> + + * rest-api/api-docs/bridges.json, res/ari/resource_channels.h, /, + res/ari/resource_bridges.h, rest-api/api-docs/channels.json: + Clarify documentation for channel and bridge list This makes it + clear that the ARI API calls for listing channels and bridges + will list all channels or bridges in the system and not just + those that are in or are controlled by a Stasis application. + (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........ + Merged revisions 401087 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 12:19 +0000 [r401079] Walter Doekes <walter+asterisk@wjd.nu> + + * /, apps/app_queue.c: Don't check all realtime queues when doing + "queue show some_queue". When using realtime queues, queues have + to be fetched from the database every now and then to see if any + info has been changed or to see if the queue has been removed. + When fetching info for an individual queue, the pruning of other + queues is unnecessarily costly. Review: + https://reviewboard.asterisk.org/r/2907/ ........ Merged + revisions 401049 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401076 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401077 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 00:12 +0000 [r401041] Paul Belanger <paul.belanger@polybeacon.com> + + * /, rest-api/api-docs/bridges.json, res/res_ari_bridges.c: Use + POST / DELETE to toggle ARI bridge moh Review: + https://reviewboard.asterisk.org/r/2911/ ........ Merged + revisions 401040 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 23:44 +0000 [r401020-401039] Richard Mudgett <rmudgett@digium.com> + + * main/translate.c: translate.c: Some minor code tweaks. * + Consistently compare format2index() return value so matrix_get() + cannot get passed negative values. * Optimize + ast_translator_best_choice() to defer initializing things until + needed. Also cached the matrix_get() return value rather than + repeatedly calling it. + + * /, channels/dahdi/bridge_native_dahdi.c: bridge_native_dahdi: + Return channel join failure if could not make the channels + compatible. ........ Merged revisions 401030 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_iax2.c: chan_iax2: Fix channel left locked in + off nominal code path. ........ Merged revisions 401016 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401017 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 20:03 +0000 [r401019] Kinsey Moore <kmoore@digium.com> + + * rest-api/api-docs/bridges.json, res/res_ari_bridges.c, /: Ensure + bridge record error responses validate This adds the list of + expected errors to the /bridges/{bridgeId}/record ARI + documentation so that outbound 4xx errors validate properly. + Previously, this would result in a response validation failure. + (closes issue ASTERISK-22627) Reported by: Joshua Colp ........ + Merged revisions 401018 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 15:30 +0000 [r401007] Paul Belanger <paul.belanger@polybeacon.com> + + * rest-api/api-docs/channels.json, res/res_ari_channels.c, /: Use + POST / DELETE to toggle hold / moh for ARI channels This change + updates how we handle toggle events, rather then create two + different function names, we'll just use POST / DELETE from HTTP + to handle it. Review: https://reviewboard.asterisk.org/r/2906/ + ........ Merged revisions 400999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 15:26 +0000 [r400998] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Prevent chan_sip from sending duplicate + BYEs. When a 200 OK for an initial INVITE is received, we were + doing the right thing by ACKing and sending an immediate BYE. + However, we also were doing the wrong thing and queuing an answer + frame, thus causing the call to be answered. This would cause the + call to be hung up by the channel thread, thus resulting in a + second BYE being sent out. In this fix, I also have set the + hangupcause to be correct since the initial BYE being sent by + Asterisk had an unknown hangup cause. I have changed to using + "Bearer capabilty not available" since the call was hung up due + to an SDP offer/answer error. (closes issue ASTERISK-22621) + reported by Kinsey Moore ........ Merged revisions 400970 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400971 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400984 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 13:44 +0000 [r400959] David M. Lee <dlee@digium.com> + + * /, rest-api-templates/asterisk_processor.py: My doc correction in + r400842 had a silly bug. Because I added a wiki_description to + models and not their properties, the rendered wiki page had the + model description instead of the property descriptions, which + looks very silly indeed. (closes issue ASTERISK-22705) ........ + Merged revisions 400958 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 22:52 +0000 [r400913-400950] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/chan_dahdi.h: chan_dahdi: Add config support for hwgain + settings. * Add hwtxgain and hwrxgain config options to + chan_dahdi.conf with documentation in chan_dahdi.conf.sample. + (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches: + jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch + uploaded by rmudgett + + * channels/chan_dahdi.c, /, channels/chan_dahdi.h: chan_dahdi: + Reflect the set software gain in the CLI "dahdi show channel" + output. * Remember the swgain setting from CLI "dahdi set swgain" + command so the CLI "dahdi show channel" output will reflect the + current setting. * Updated CLI "dahdi set hwgain" and "dahdi set + swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco + Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621) + patch uploaded by rmudgett ........ Merged revisions 400907 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400909 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400911 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 22:03 +0000 [r400912] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: chan_sip: Do not increment the SDP + version between 183 and 200 responses. Bumping the SDP version + number can cause interoperability problems since receivers of the + responses will expect that a 200 SDP will be identical to a + previous 183 SDP. (closes issue ASTERISK-21204) reported by + NITESH BANSAL Patches: + dont-increment-session-version-in-2xx-after-183.patch uploaded by + NITESH BANSAL (License #6418) ........ Merged revisions 400906 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 400908 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400910 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 15:54 +0000 [r400891] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip_outbound_registration.c: pjsip outbound + registration: Log message says received a 408 when we didn't If + the server didn't exist that we are trying to register to the log + message would say that a 408 was received from that server when + in reality one wasn't. Added log messages stating no response was + received if the response does not exist. (closes issue + ASTERISK-22554) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2893/ ........ Merged + revisions 400890 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 15:01 +0000 [r400882] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_mwi.c, /: Remove duplicate module info block The + module info block was repeated twice. Once is sufficient. + ........ Merged revisions 400881 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-13 15:42 +0000 [r400873] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_session.c, /: Fix a race condition in + res_pjsip_session with rapidly terminating the session. The + INVITE session state callback wrongly assumes that a session will + always exist, but when rapidly terminating the session this + assumption goes out the window. As all handler code for the + INVITE session state callback requires the session it will now + just exit immediately if no session exists. (closes issue + ASTERISK-22668) Reported by: John Bigelow ........ Merged + revisions 400872 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-12 16:53 +0000 [r400864] Kinsey Moore <kmoore@digium.com> + + * /, res/res_pjsip_outbound_authenticator_digest.c: Fix realm + comparison for outbound auth When generating the list of + authentication credentials to pass to PJSIP, Asterisk was using + the raw pointer of a pj_str_t which is not always + NULL-terminated. This sometimes resulted in incorrect text for + the realm and a failure to match the realm for authentication + purposes which was causing the outbound nominal auth pjsip basic + call test to bounce. This now uses the pj_str_t that contains the + realm instead of generating a new one. Thanks to John Bigelow for + helping to narrow this down. ........ Merged revisions 400863 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-11 17:05 +0000 [r400855] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/channel.h, /: channel.h: whitespace changes. + ........ Merged revisions 400854 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-11 16:36 +0000 [r400851-400852] David M. Lee <dlee@digium.com> + + * /, res/ari/resource_bridges.h, rest-api/api-docs/playback.json, + rest-api-templates/api.wiki.mustache, res/res_ari_playback.c, + rest-api/api-docs/channels.json, res/ari/resource_playback.h, + rest-api/api-docs/bridges.json, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, + rest-api-templates/models.wiki.mustache: Multiple revisions + 400508,400842-400843,400848 ........ r400508 | dlee | 2013-10-03 + 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line Corrected response + class for stopPlayback ........ r400842 | dlee | 2013-10-10 + 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line Correct some ARI wiki + rendering errors ........ r400843 | dlee | 2013-10-10 14:26:19 + -0500 (Thu, 10 Oct 2013) | 1 line Updated /play resource docs. + The playback of http: resources isn't implemented... yet ........ + r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 + lines Fix a stupid copy/paste error in ARI docs. Patches: + ari-doc-patch.txt uploaded by jbigelow (license 5091) ........ + Merged revisions 400508,400842-400843,400848 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Fixed merge tracking for r400360, which was somehow lost + +2013-10-11 16:28 +0000 [r400850] Richard Mudgett <rmudgett@digium.com> + + * bridges/bridge_softmix.c, /: Softmix: Fix crash when switching + from softmix to another bridge technology. The crash is caused by + a race condition when switching between native RTP and softmix + bridging technologies. In this situation, the bridging technology + is switched from native RTP to softmix, and then back to native + RTP fast enough that the softmix private data gets destroyed + before the softmix mixing thread gets started. Thanks to Kinsey + Moore for the crash analysis. * Fix race condition when starting + the softmix mixing thread and switching to another bridge + technology. (closes issue ASTERISK-22678) Reported by: John + Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621) + patch uploaded by rmudgett Tested by: John Bigelow ........ + Merged revisions 400849 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-10 18:21 +0000 [r400825-400834] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip/location.c: Perform validation of permanent + contacts on AORs in res_pjsip. ........ Merged revisions 400833 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c: Fix an + assertion in res_pjsip when specifying an invalid outbound proxy. + This change fixes two issues when setting an outbound proxy: 1. + The outbound proxy URI was not parsed and validated during + configuration. 2. If an outgoing dialog was created and the + outbound proxy could not be set an assertion would occur because + the usage count on the dialog was not decremented. The + documentation has also been updated to specify that a full URI + must be specified for the outbound proxy. (closes issue + ASTERISK-22672) Reported by: Antti Yrjola ........ Merged + revisions 400824 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-09 11:02 +0000 [r400772-400813] Matthew Jordan <mjordan@digium.com> + + * res/res_pjsip_header_funcs.c, /: Use 'z' as the format specifier + for size_t Using 'lu' will produce a compiler warning for some + versions of gcc and on some architectures. 'z' should be portable + as a format specifier for size_t. ........ Merged revisions + 400812 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_header_funcs.c (added), /: Add PJSIP_HEADER + function for manipulation of SIP headers in the PJSIP stack This + patch adds support to the PJSIP stack in Asterisk for SIP header + manipulation. Note that this is analagous to + SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming + supplemental session callback is registered that takes the + pjsip_hdrs from the incoming session and stores them in a linked + list in the session datastore. Calls to PJSIP_HEADER traverse + over the list and return the nth matching header where 'n' is the + 'number' argument to the function. When adding a header, the + first call creates a datastore and linked list and adds the + datastore to the session. The header is then created as a + pjsip_hdr and added to the list. An outgoing supplemental session + callback then traverses the list and adds the headers to the + outgoing pjsip_msg. When removing a header, the list created with + PJSIP_HEADER(add,...) is traversed and all matching entries are + removed. (closes issue ASTERISK-22498) Reported by: George Joseph + patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph + (License 6322) ........ Merged revisions 400771 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 22:33 +0000 [r400770] Kinsey Moore <kmoore@digium.com> + + * /, configure, configure.ac: Add warning when compiling with iODBC + support When running configure, libiodbc2 development headers + will fulfill the requirement for ODBC development headers, but + will not function properly. This adds a warning when libiodbc2 + development headers are detected instead of unixodbc development + headers. (closes issue ASTERISK-22459) Reported by: Patrick + Maille Tested by: Walter Doekes Patches: + issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes + (License 5674) ........ Merged revisions 400767 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400768 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400769 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 21:20 +0000 [r400759] Richard Mudgett <rmudgett@digium.com> + + * apps/app_agent_pool.c, /: app_agent_pool: Fix AMI/CLI AgentLogoff + soft preventing agents from logging back in. * Clear the + deferred_logoff flag when an agent logs in. (closes issue + ASTERISK-22669) Reported by: John Bigelow ........ Merged + revisions 400754 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 20:52 +0000 [r400750] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip.c, res/res_pjsip/config_transport.c: Switch from + using pjsip_strerror to pj_strerror. pjsip_strerror is only aware + of PJSIP-specific error codes. pj_strerror() is aware of all + PJProject error codes and OS-specific error codes. This + specifically fixes an oft-seen error in transport configuration + code where EADDRINUSE would result in "Unknown PJSIP error + 120098" instead of a useful message. ........ Merged revisions + 400749 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 20:18 +0000 [r400728-400744] Richard Mudgett <rmudgett@digium.com> + + * configs/confbridge.conf.sample, /, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + CHANGES, apps/confbridge/conf_config_parser.c: app_confbridge: + Can now set the language used for announcements to the + conference. ConfBridge now has the ability to set the language of + announcements to the conference. The language can be set on a + bridge profile in confbridge.conf or by the dialplan function + CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) + Reported by: Jonathan White Patches: M19983_rev2.diff (license + #5138) patch uploaded by junky (modified) Tested by: rmudgett + ........ Merged revisions 400741 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400742 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix + duplicate default_user profile. * Fixed looking in the wrong + profiles container to see if the default_user profile is already + created in verify_default_profiles(). The bridge profile + container is never going to hold user profiles. :) ........ + Merged revisions 400723 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400724 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 18:19 +0000 [r400684-400704] Kinsey Moore <kmoore@digium.com> + + * funcs/func_config.c, /: Fix func_config list entry allocation The + AST_CONFIG dialplan function defined in func_config.c allocates + its config file list entries using ast_malloc. List entry + allocations destined for use with Asterisk's linked list API must + be ast_calloc()d or otherwise initialized so that list pointers + are set to NULL. These uses of ast_malloc have been replaced by + ast_calloc to prevent dereferencing of uninitialized pointer + values when traversing the list. (closes issue ASTERISK-22483) + Reported by: Brian Scott ........ Merged revisions 400694 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400697 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400701 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: Fix STUN crash when using IPv6 any + address Ensure that when chan_sip binds to the IPv6 any address + ([::]), IPv4 candidates are also added. (closes issue + ASTERISK-21917) Reported by: Torrey Searle Patches: + 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License + 5334) ........ Merged revisions 400681 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400682 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 15:44 +0000 [r400683] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip/pjsip_options.c, /: Push CLI qualify into the + threadpool. If you run Asterisk in the background and then + connect to it through a separate console, the thread that runs + CLI commands is not registered with PJLIB. Thus PJLIB does not + like it when you attempt to send OPTIONS requests from that + thread. So now we push the task into the threadpool, which we + know to be registered with PJLIB. Thanks to Antti Yrjola for + reporting this. ........ Merged revisions 400680 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 15:12 +0000 [r400662-400672] Richard Mudgett <rmudgett@digium.com> + + * /, res/res_agi.c, apps/app_queue.c: Make app_queue and res_agi + independent of AMI being enabled. The + https://reviewboard.asterisk.org/r/2888/ review changes manager + to not subscribe to stasis when it is disabled for performance + reasons. When manager is disabled app_queue and res_agi decline + to load and fail to clean up what they have already allocated. * + Made app_queue and res_agi clean up allocated resources when they + decline to load. * Made app_queue and res_agi use their own + subscriptions to the stasis topics instead of borrowing manager's + message router structure inappropriately. (closes issue + ASTERISK-22604) Reported by: rmudgett Review: + https://reviewboard.asterisk.org/r/2902/ ........ Merged + revisions 400671 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, include/asterisk/stasis.h, apps/app_queue.c, + include/asterisk/manager.h: Miscellaneous stand alone comment + cleanups. ........ Merged revisions 400661 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-06 17:13 +0000 [r400625] Michael L. Young <elgueromexicano@gmail.com> + + * /, apps/app_queue.c: app_queue: Fix Queuelog EXITWITHKEY only + logging two of four fields Commit r62462 added two extra fields + for logging "the original position the caller entered the queue + at, and the amount of time the caller was waiting in the queue." + But when r75969 was merged from 1.4 into trunk (r75977), these + two fields disappeared. Those two extra fields were not logged in + 1.4 and when the patch was merged, those fields went away. + Therefore, this is a regression and was caught by the reporter + because he was reading the awesome "Asterisk: The Definitive + Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M. + Tested by: Dalius M. Patches: + asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2901/ ........ Merged + revisions 400622 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400623 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-05 00:59 +0000 [r400593] Richard Mudgett <rmudgett@digium.com> + + * /, channels/iax2/include/parser.h: chan_iax2: Fix compile error. + ........ Merged revisions 400588 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 21:41 +0000 [r400568] Michael L. Young <elgueromexicano@gmail.com> + + * main/acl.c, include/asterisk/netsock2.h, CHANGES, + channels/chan_iax2.c, channels/iax2/parser.c, main/netsock.c, + main/netsock2.c, /, channels/iax2/include/parser.h: Add IPv6 + Support To chan_iax2 This patch adds IPv6 support to chan_iax2. + Yay! (closes issue ASTERISK-22025) Patches: + iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) + Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged + revisions 400567 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 19:32 +0000 [r400553] David M. Lee <dlee@digium.com> + + * rest-api/api-docs/applications.json (added), /: Added missing + file from r400522 ........ Merged revisions 400552 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 19:11 +0000 [r400533-400543] Jonathan Rose <jrose@digium.com> + + * res/res_pjsip_logger.c, /: chan_pjsip: Make logger togglable + without loading/unloading This patch makes the res_pjsip_logger + do a few things... First, it will be built and installed by + default now, so end users won't need to enable it in menuselect. + Second, while it is loaded, it no longer will immediately issue + log messages. Upon loading, it is in the disabled state and must + be turned on with the new CLI command. The CLI command 'pjsip set + logger <on/off/host> has been added and can be used to do the + following: pjsip set logger on: Enables logger for all PJSIP + traffic pjsip set logger off: Disables logger for all PJSIP + traffic pjsip set logger host <host>: Enables logger for the + specific host Review: https://reviewboard.asterisk.org/r/2900/ + ........ Merged revisions 400542 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, + contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py + (added), configs/extconfig.conf.sample, + configs/sorcery.conf.sample, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py: + chan_pjsip: Add alembic scripts for generating db tables for + PJSIP Also updates sample configurations for sorcery and + extconfig to demonstrate how to use databases created by that + alembic script. (closes issue ASTERISK-22133) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........ + Merged revisions 400532 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 16:01 +0000 [r400523] Matthew Jordan <mjordan@digium.com> + + * res/res_stasis.c, main/asterisk.c, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + res/stasis/app.c, /, + rest-api-templates/ari_model_validators.h.mustache, + include/asterisk/endpoints.h, res/res_ari_applications.c (added), + res/ari/resource_endpoints.h, include/asterisk/stasis_app.h, + res/stasis/app.h, rest-api/resources.json, + include/asterisk/_private.h, res/ari/ari_model_validators.c, + main/endpoints.c, res/ari/ari_model_validators.h, main/json.c, + res/res_ari_model.c, res/ari.make, + res/ari/resource_applications.c (added), + res/ari/resource_applications.h (added): ARI: Add subscription + support This patch adds an /applications API to ARI, allowing + explicit management of Stasis applications. * GET /applications - + list current applications * GET /applications/{applicationName} - + get details of a specific application * POST + /applications/{applicationName}/subscription - explicitly + subscribe to a channel, bridge or endpoint * DELETE + /applications/{applicationName}/subscription - explicitly + unsubscribe from a channel, bridge or endpoint Subscriptions work + by a reference counting mechanism: if you subscript to an event + source X number of times, you must unsubscribe X number of times + to stop receiveing events for that event source. Review: + https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) + Reported by: Matt Jordan ........ Merged revisions 400522 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 15:49 +0000 [r400511-400521] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip.c: Enclose the To URI and update its user + portion if a request user has been specified. ........ Merged + revisions 400520 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: Replace the connection address at the + SDP level if altering the SDP with the external media address. + ........ Merged revisions 400510 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 23:20 +0000 [r400482] Jonathan Rose <jrose@digium.com> + + * /, channels/chan_sip.c: chan_sip: Don't ignore expires value in + contact header if it lacks semicolon (closes issue + ASTERISK-22574) Reported by: Filip Jenicek Patches: + chan_sip_expires.patch uploaded by Filip Jenicek (license 6277) + ........ Merged revisions 400469 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400470 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400471 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 21:46 +0000 [r400461] Matthew Jordan <mjordan@digium.com> + + * /, main/channel_internal_api.c: Remove publication of a channel + snapshot when the technology is set This patch removes said + publication for a few reasons: (1) It is unnecessary. Association + of the channel technology with a specific channel is an + implementation detail that should be assumed to "just happen", + and consumers of Stasis don't need to be informed about it. (2) + Publication of said message can now cause crashes, as the actual + creation of a channel in normal locations now stages its + messages. As a result, things that create dummy channels (such as + the SIP RTP QOS unit test) and associate them with a channel + technology were now crashing, as the channel itself was not known + by Stasis. ........ Merged revisions 400460 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 20:22 +0000 [r400452] Mark Michelson <mmichelson@digium.com> + + * bridges/bridge_native_rtp.c, /, + include/asterisk/bridge_technology.h: Fix assumption in + bridge_native_rtp.c regarding number of participants in a bridge. + When a party leaves a bridge, there may be more participants in + the bridge than expected. As such, it is important not to make + assumptions regarding the list of channels in a bridge. This + change makes it so that when a party leaves a native RTP bridge, + we unbridge it and the party it was bridged with. Previously, the + first and last channels in the list were unbridged since it was + assumed that these were the two channels that had been bridged. + As previously stated, a new party had been inserted into the + bridge, so this logic did not work properly. (closes issue + ASTERISK-22615) reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/2899 ........ Merged revisions + 400403 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 19:32 +0000 [r400443] Joshua Colp <jcolp@digium.com> + + * /, main/cdr.c: When serializing CDR variables (like for "core + show channels") don't output an error if CDRs aren't enabled. + ........ Merged revisions 400442 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 19:30 +0000 [r400441] Kinsey Moore <kmoore@digium.com> + + * /, main/security_events.c: Fix security events for AMI invalid + password In r337595, additional security events were added for + chan_sip authentication failures. The new IEs added to the + existing invalid password event were defined as required IEs, but + existing users of the event did not set the new IEs and could not + since they didn't apply to existing uses. They are now marked as + optional IEs. (closes issue ASTERISK-22578) Reported by: Matt + Jordan ........ Merged revisions 400421 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400440 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 19:06 +0000 [r400402] Joshua Colp <jcolp@digium.com> + + * res/ari/resource_channels.c, /: Fix a crash caused by muting and + unmuting a channel in ARI without specifying a direction. (closes + issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by + Matt Jordan, whose office I have taken over in the name of + Canada. ........ Merged revisions 400401 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 18:51 +0000 [r400399] Richard Mudgett <rmudgett@digium.com> + + * /, main/cel.c: cel: Some whitespace cleanups ........ Merged + revisions 400398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 18:32 +0000 [r400385-400397] Kinsey Moore <kmoore@digium.com> + + * res/res_rtp_multicast.c, /: res_rtp_multicast: Ensure SSRC is set + properly This fixes a bug where the SSRC field on multicast RTP + can be stuck at 0 which can cause problems for endpoints trying + to make sense of incoming streams. (closes issue ASTERISK-22567) + Reported by: Simone Camporeale Patches: + 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale + (License 6536) ........ Merged revisions 400393 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400394 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400395 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/xml.c: Detect and use xsltCleanupGlobals when available This + introduces usage of an additional libxslt cleanup function, + xsltCleanupGlobals, when the configure script detects that it is + available. Early versions of the library did not include this + function. (closes issue ASTERISK-22570) Reported by: Corey + Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey + Farrell (License 5909) ........ Merged revisions 400384 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 16:28 +0000 [r400374] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_vpb.cc, /: chan_vpb: Make compile again. ........ + Merged revisions 400373 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 14:59 +0000 [r400363-400364] Mark Michelson <mmichelson@digium.com> + + * tests/test_cel.c, /: Get rid of uses of stasis_topic_wait() + ........ Merged revisions 400362 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * pbx/pbx_spool.c, main/manager.c, main/format_cap.c, + channels/chan_skinny.c, res/res_agi.c, channels/chan_motif.c, + channels/chan_alsa.c, apps/app_confbridge.c, + addons/chan_mobile.c, channels/chan_mgcp.c, + res/res_clioriginate.c, channels/chan_bridge_media.c, + channels/chan_sip.c, tests/test_format_api.c, + res/res_pjsip_sdp_rtp.c, bridges/bridge_simple.c, + apps/app_originate.c, res/parking/parking_applications.c, + main/core_local.c, channels/chan_console.c, channels/chan_oss.c, + include/asterisk/format_cap.h, res/res_pjsip_session.c, + res/ari/resource_bridges.c, channels/chan_jingle.c, + channels/chan_misdn.c, channels/dahdi/bridge_native_dahdi.c, + res/res_pjsip/pjsip_configuration.c, main/file.c, + channels/chan_h323.c, channels/chan_nbs.c, + bridges/bridge_native_rtp.c, tests/test_config.c, + res/res_stasis.c, channels/chan_pjsip.c, channels/chan_unistim.c, + channels/chan_multicast_rtp.c, addons/chan_ooh323.c, + main/rtp_engine.c, /, main/ccss.c, apps/app_meetme.c, + bridges/bridge_holding.c, main/bridge_basic.c, + bridges/bridge_softmix.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/media_index.c, main/channel.c, + channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c: Cache + string values of formats on ast_format_cap() to save processing. + Channel snapshots have string representations of the channel's + native formats. Prior to this change, the format strings were + re-created on ever channel snapshot creation. Since channel + native formats rarely change, this was very wasteful. Now, string + representations of formats may optionally be stored on the + ast_format_cap for cases where string representations may be + requested frequently. When formats are altered, the string cache + is marked as invalid. When strings are requested, the cache + validity is checked. If the cache is valid, then the cached + strings are copied. If the cache is invalid, then the string + cache is rebuilt and copied, and the cache is marked as being + valid again. Review: https://reviewboard.asterisk.org/r/2879 + ........ Merged revisions 400356 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 14:52 +0000 [r400361] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c, /: Fix crashes in + res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and + external_media_address is set. The callback function for changing + the media address in streams wrongly assumes that a connection + line will always be present. This is false as no line is present + if a stream has been rejected. (closes issue ASTERISK-22645) + Reported by: Rusty Newton ........ Merged revisions 400360 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 22:22 +0000 [r400335] Mark Michelson <mmichelson@digium.com> + + * main/stasis_wait.c (removed), res/ari/resource_endpoints.c, /, + include/asterisk/stasis.h, tests/test_cel.c, + include/asterisk/stasis_endpoints.h, channels/chan_pjsip.c, + main/stasis.c, main/stasis_endpoints.c: Multiple revisions + 400318-400319 ........ r400318 | mmichelson | 2013-10-02 17:08:49 + -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from + stasis. Since caches are updated on publisher threads, there is + no need to wait for the cache updates to occur after a stasis + message is published. In the case of chan_pjsip device state + changes, this set of changes caused an improvement to + performance. Review: https://reviewboard.asterisk.org/r/2890 + ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, + 02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........ + Merged revisions 400318-400319 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 21:33 +0000 [r400317] Michael L. Young <elgueromexicano@gmail.com> + + * channels/chan_iax2.c, /: Cast Integer Argument To Unsigned Char + The member reg in the peercnt structure is an unsigned char and + peercnt_modify() is expecting an unsigned char argument which + gets assigned to peercnt->reg. This patch fixes that by casting + the integer argument being passed to peercnt_modify to unsigned + char. ........ Merged revisions 400314 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400315 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400316 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 21:26 +0000 [r400313] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, main/manager.c, /, main/cel.c: Only create Stasis + subscriptions when enabled Subscribing to Stasis isn't free. As + such, this patch makes AMI, CDR, and CEL - the "big 3" - only + subscribe when enabled. Toggling their availability via a .conf + file will unsubscribe/subscribe as appropriate. Review: + https://reviewboard.asterisk.org/r/2888/ ........ Merged + revisions 400312 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 20:31 +0000 [r400304] Richard Mudgett <rmudgett@digium.com> + + * main/pbx.c, /: Originate: Make setting caller id on outgoing call + use either name or number. Previous code was requiring both name + and number to be available. Also restored a comment block on why + caller id is also set on an outgoing call leg in addition to + connected line from earlier versions of Asterisk. ........ Merged + revisions 400303 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 19:20 +0000 [r400295] Kinsey Moore <kmoore@digium.com> + + * /, rest-api/api-docs/asterisk.json: Correct allowable values for + ARI general information filter ........ Merged revisions 400291 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 19:17 +0000 [r400287] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /: Fix the CDR CLI command 'cdr show active + {channel}' When the switch from channel names to channel unique + IDs happened, the poor CLI command got left in the dust. This + fixes the command so that users can once again see how Asterisk + is messing up your billing information. ........ Merged revisions + 400286 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 18:44 +0000 [r400285] Joshua Colp <jcolp@digium.com> + + * /, res/res_pjsip_t38.c: Fix a crash in res_pjsip_t38 caused by + the wrong assumption that a session will always have a channel. + When starting up or shutting down this assumption is false. + ........ Merged revisions 400284 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 18:28 +0000 [r400282] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * Makefile, doc/astdb2sqlite3.8 (added), /, doc/astdb2bdb.8 + (added): man pages for astdb2bdb and astdb2sqlite3 Review: + https://reviewboard.asterisk.org/r/2898/ ........ Merged + revisions 400279 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400281 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 17:12 +0000 [r400269-400271] Richard Mudgett <rmudgett@digium.com> + + * apps/app_stack.c, res/stasis_recording/stored.c, main/json.c, + main/stasis_cache.c, res/res_ari.c, /, main/utils.c: + MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is + enabled. * There were several places in ARI where an external + library was mallocing memory that must always be released with + free(). When MALLOC_DEBUG is enabled, free() is redirected to the + MALLOC_DEBUG version. Since the external library call still uses + the normal malloc(), MALLOC_DEBUG complains that the freed memory + block is not registered and will not free it. These cases must + use ast_std_free(). * Changed calls to asprintf() and vasprintf() + to the equivalent ast_asprintf() and ast_vasprintf() versions + respectively. ........ Merged revisions 400270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sig_ss7.c, /: sig_ss7: Fix compiler warnings. ........ + Merged revisions 400268 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 16:23 +0000 [r400246-400266] Joshua Colp <jcolp@digium.com> + + * channels/chan_alsa.c, main/stasis_channels.c, channels/sig_ss7.c, + channels/chan_pjsip.c, channels/chan_mgcp.c, + channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, /, + channels/chan_sip.c, main/bridge.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/chan_console.c, + channels/sig_pri.c, channels/chan_iax2.c, channels/chan_jingle.c, + main/channel.c, channels/chan_dahdi.c, main/dial.c, + include/asterisk/stasis_channels.h, channels/chan_skinny.c, + channels/chan_motif.c: Reduce channel snapshot creation and + publishing by up to 50%. This change introduces the ability to + stage channel snapshot creation and publishing by suppressing the + implicit creation and publishing that some functions have. Once + all operations are executed the staging is marked as done and a + single snapshot is created and published. Review: + https://reviewboard.asterisk.org/r/2889/ ........ Merged + revisions 400265 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: Fix a random one way audio issue in + PJSIP. Due to the asynchronous design of the PJMEDIA SDP + negotiator it was possible for the SDP to be negotiated *after* a + channel was created and after it was being wait on by an + application. It is only after negotiation occurs that the file + descriptors for RTP are placed on the channel. Since the channel + was already being waited on these file descriptors were not + monitored, causing incoming media to never be read. This change + wakes up any application waiting on the channel so that added + file descriptors end up being monitored. (closes issue AST-1227) + Reported by: John Bigelow ........ Merged revisions 400256 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c, include/asterisk/stasis_app.h, + res/ari/resource_channels.c: Allow specifying a channel to dial + an extension and context in an ARI dial operation. (issue + ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged + revisions 400254 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_session.c: Retrieve and store the hostname only + once so multiple threads do not potentially initialize it at the + same time. ........ Merged revisions 400245 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-01 21:19 +0000 [r400228-400237] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, channels/sig_analog.c, /: chan_dahdi: Fix + analog parking using flash-hook. Transferring an analog call + using a flash-hook to parking would fail to park the call and + result in an invalid ao2 object unref. * Park the correct bridged + channel. ........ Merged revisions 400236 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/features_config.c, /: Features: Rearm the parking config + options have moved warning for each reload. ........ Merged + revisions 400227 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-01 15:54 +0000 [r400218] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /: Filter out internal channels for bridge leave + messages and parked call messages Granted, if you manage to park + a Conference announcer channel, something has gone horrifically + wrong. ........ Merged revisions 400217 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 21:40 +0000 [r400206] Jonathan Rose <jrose@digium.com> + + * configs/features.conf.sample, /, configs/res_parking.conf.sample: + configuration samples: Pull all parking related stuff out of + features.conf This patch also adds documentation for parking from + features.conf to res_parking.conf ........ Merged revisions + 400205 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 19:58 +0000 [r400195-400197] Matthew Jordan <mjordan@digium.com> + + * /, funcs/func_cdr.c: Parse arguments passed to the CDR_PROP + function correctly I can only blame this on a bad merge, because + this in no way worked properly the way it was written. Mea culpa. + The function should now parse its arguments correctly and + function properly. (Note that the API used by the CDR_PROP + function has working unit tests... this was merely bad coding of + the actual registered function) (closes issue ASTERISK-22613) + Reported by: Private Name ........ Merged revisions 400196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Remove spurious event raised when CDRs are + reloaded The Reload event is now raised by the module loading + core. As such, the Reload event in the CDR engine was a duplicate + and not needed. ........ Merged revisions 400194 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 18:55 +0000 [r400186] David M. Lee <dlee@digium.com> + + * tests/test_devicestate.c, include/asterisk/sem.h (added), + tests/test_taskprocessor.c, res/res_pjsip_mwi.c, + res/res_pjsip/include/res_pjsip_private.h, tests/test_stasis.c, + res/parking/parking_manager.c, res/res_security_log.c, + channels/chan_mgcp.c, main/stasis_cache_pattern.c, main/pbx.c, + include/asterisk/vector.h (added), /, main/ccss.c, + apps/app_meetme.c, include/asterisk/taskprocessor.h, + configs/stasis.conf.sample (removed), configure.ac, + res/parking/parking_applications.c, channels/sig_pri.c, + apps/app_queue.c, main/cel.c, main/stasis.c, + channels/chan_dahdi.c, funcs/func_presencestate.c, + main/stasis_message_router.c, configure, + apps/confbridge/confbridge_manager.c, res/res_agi.c, + main/manager_system.c, res/res_stasis_test.c, main/sem.c (added), + main/manager_channels.c, res/res_pjsip_refer.c, + main/manager_mwi.c, apps/app_voicemail.c, main/stasis_cache.c, + main/stasis_wait.c, main/stasis_config.c (removed), + include/asterisk/stasis_internal.h, res/stasis/app.c, + channels/chan_sip.c, include/asterisk/autoconfig.h.in, + main/manager_endpoints.c, main/channel_internal_api.c, + include/asterisk/stasis.h, main/devicestate.c, + main/taskprocessor.c, res/res_xmpp.c, main/sounds_index.c, + include/asterisk/stasis_message_router.h, channels/chan_iax2.c, + res/res_jabber.c, main/endpoints.c, main/astobj2.c, + res/res_chan_stats.c, res/parking/parking_bridge_features.c, + tests/test_stasis_endpoints.c, main/cdr.c, main/channel.c, + main/manager_bridges.c, main/manager.c, channels/chan_skinny.c: + Multiple revisions 399887,400138,400178,400180-400181 ........ + r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 + line Minor performance bump by not allocate manager variable + struct if we don't need it ........ r400138 | dlee | 2013-09-30 + 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance + improvements This patch addresses several performance problems + that were found in the initial performance testing of Asterisk + 12. The Stasis dispatch object was allocated as an AO2 object, + even though it has a very confined lifecycle. This was replaced + with a straight ast_malloc(). The Stasis message router was + spending an inordinate amount of time searching hash tables. In + this case, most of our routers had 6 or fewer routes in them to + begin with. This was replaced with an array that's searched + linearly for the route. We more heavily rely on AO2 objects in + Asterisk 12, and the memset() in ao2_ref() actually became + noticeable on the profile. This was #ifdef'ed to only run when + AO2_DEBUG was enabled. After being misled by an erroneous comment + in taskprocessor.c during profiling, the wrong comment was + removed. Review: https://reviewboard.asterisk.org/r/2873/ + ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep + 2013) | 24 lines Taskprocessor optimization; switch Stasis to use + taskprocessors This patch optimizes taskprocessor to use a + semaphore for signaling, which the OS can do a better job at + managing contention and waiting that we can with a mutex and + condition. The taskprocessor execution was also slightly + optimized to reduce the number of locks taken. The only + observable difference in the taskprocessor implementation is that + when the final reference to the taskprocessor goes away, it will + execute all tasks to completion instead of discarding the + unexecuted tasks. For systems where unnamed semaphores are not + supported, a really simple semaphore implementation is provided. + (Which gives identical performance as the original taskprocessor + implementation). The way we ended up implementing Stasis caused + the threadpool to be a burden instead of a boost to performance. + This was switched to just use taskprocessors directly for + subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ + ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep + 2013) | 28 lines Optimize how Stasis forwards are dispatched This + patch optimizes how forwards are dispatched in Stasis. + Originally, forwards were dispatched as subscriptions that are + invoked on the publishing thread. This did not account for the + vast number of forwards we would end up having in the system, and + the amount of work it would take to walk though the forward + subscriptions. This patch modifies Stasis so that rather than + walking the tree of forwards on every dispatch, when forwards and + subscriptions are changed, the subscriber list for every topic in + the tree is changed. This has a couple of benefits. First, this + reduces the workload of dispatching messages. It also reduces + contention when dispatching to different topics that happen to + forward to the same aggregation topic (as happens with all of the + channel, bridge and endpoint topics). Since forwards are no + longer subscriptions, the bulk of this patch is simply changing + stasis_subscription objects to stasis_forward objects (which, + admittedly, I should have done in the first place.) Since this + required me to yet again put in a growing array, I finally + abstracted that out into a set of ast_vector macros in + asterisk/vector.h. Review: + https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee + | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove + dispatch object allocation from Stasis publishing While looking + for areas for performance improvement, I realized that an unused + feature in Stasis was negatively impacting performance. When a + message is sent to a subscriber, a dispatch object is allocated + for the dispatch, containing the topic the message was published + to, the subscriber the message is being sent to, and the message + itself. The topic is actually unused by any subscriber in + Asterisk today. And the subscriber is associated with the + taskprocessor the message is being dispatched to. First, this + patch removes the unused topic parameter from Stasis subscription + callbacks. Second, this patch introduces the concept of + taskprocessor local data, data that may be set on a taskprocessor + and provided along with the data pointer when a task is pushed + using the ast_taskprocessor_push_local() call. This allows the + task to have both data specific to that taskprocessor, in + addition to data specific to that invocation. With those two + changes, the dispatch object can be removed completely, and the + message is simply refcounted and sent directly to the + taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ + ........ Merged revisions 399887,400138,400178,400180-400181 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 15:57 +0000 [r400142] Kinsey Moore <kmoore@digium.com> + + * /, channels/chan_sip.c, configs/pjsip.conf.sample, + res/res_pjsip_outbound_registration.c, configs/sip.conf.sample, + CHANGES: chan_sip: Allow Asterisk to retry after 403 on register + This adds a global option in chan_sip to allow it to continue + attempting registration if a 403 is received, clearing the cached + nonce and treating it as a non-fatal response. Normally, this + would cause registration attempts to that endpoint to stop. This + also adds a similar per-outbound-registration option to + chan_pjsip which allows the retry interval to be altered for 403 + responses to REGISTER requests. (closes issue ASTERISK-17138) + Review: https://reviewboard.asterisk.org/r/2874/ Reported by: + Rudi ........ Merged revisions 400137 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400140 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400141 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-28 22:57 +0000 [r400059-400122] Matthew Jordan <mjordan@digium.com> + + * /, res/res_pjsip_notify.c, configs/pjsip_notify.conf.sample + (added): res_pjsip_notify: Add documentation We forgot to add + documentation for res_pjsip_notify, which would prevent it from + being loaded. Whoops. This patch also updates res_pjsip_notify to + use pjsip_notify.conf, which now has its own sample file in the + configs directory as well. Review: + https://reviewboard.asterisk.org/r/2835/ ........ Merged + revisions 400121 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Correct erroneous + lost packet information in RTCP reports RTCP's calculation of the + number of lost packets in an RTP stream is based on that stream's + sequence number count, the number of received packets, and how + many packets we expect to receive. When the SSRC for an RTP + stream changes, there can - and almost always will be - a large + jump in the next packet's timestamp and sequence number. If we + don't reset the number of received packets, sequence number + count, and other metrics used by RTCP, the next RR/SR report will + use the previous SSRC's values to calculate the lost packet count + for the new SSRC - resulting in a very large number of lost + packets. This patch modifies res_rtp_asterisk such that, if it + detects a SSRC change, it will reset the various values used by + the RTCP calculations. From the perspective of RTCP, this appears + as a new media stream - which is what it is. Review: + https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) + Reported by: Thomas Arimont ........ Merged revisions 400089 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400093 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400108 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, configure.ac: Add check for openSUSE when detecting + bfd library In ASTERISK-17842, some additional library checks + were added to the configure script so that the bfd library could + be found on CentOS and Fedora systems. As it turns out, openSUSE + requires an additional library. This patch adds another check to + the configure script for openSUSE that will add that library. + Review: https://reviewboard.asterisk.org/r/2885/ (closes issue + AST-1169) Reported by: Guenther Kelleter ........ Merged + revisions 400073 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400075 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400077 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: CDR: Improve handling of parking; resolve + assertion when originating into park This patch covers two + problems: 1) Currently, when a call is transferred into a parking + lot from a bridge (using either the blind transfer or one touch + parking mechanisms), the application fails to be set to "Park" in + the resulting CDR record for the parked channel. This is due to + the ParkedCall message arriving before the BridgeEnter for the + channel entering the parking bridge. The ParkedCall message isn't + handled as the CDR for the channel has already been finalized + (due to the channel having left its two party bridge), and the + BridgeEnter - which creates the new CDR - doesn't have the + parking information. This patch modifies the behavior so that + reception of a ParkedCall message will - if not handled by a CDR + chain - cause a new CDR to be created and put into the Parking + state. 2) It fixes a FRACK that occurred when a channel is + originated into a parking space. The DialedPending state - which + occurs for both Dialed and Originated channels - assumed that it + couldn't handle the parking transitions due to it having a Party + B; however, Originated channels don't have a Party B. As such, + the existing CDR needs to transition into the parking state - + this patch does that. Review: + https://reviewboard.asterisk.org/r/2877/ (closes issue + ASTERISK-22482) Reported by: Richard Mudgett ........ Merged + revisions 400062 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_queue.c: app_queue: Make manager events tolerant of + Local channel shenanigans app_queue currently attempts to handle + Local channel optimizations in an effort to provide accurate + information in Stasis messages (and their corresponding AMI + events) as well as the Queue log. Sometimes, however, things + don't go as planned. Consider the following scenario: SIP/foo <-> + L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local + channel optimization. app_queue will normally do the following: * + Listen for the Local optimization events and update our agent + accordingly to SIP/agent in the queue log and messages * When we + get a hangup, publish the AgentComplete event based on our + information (SIP/foo and SIP/agent) However, as with all things + that depend on sanity from something as capricious as Local + channels, things can go wrong: (1) SIP/agent immediately hangs up + upon answering. This triggers a race condition between + termination messages coming from SIP/agent and the ongoing Local + channel optimization messages. (Note that this can also occur + with SIP/foo) (2) In a race condition, Asterisk can (rarely) + deliver the hangup messages prior to the Local channel + optimization. In that case, the messages *may* arrive to + app_queue in the following order: * Hangup SIP/Agent * Hangup + SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When + app_queue receives the hangup of the agent or the caller, it will + attempt to publish the AgentComplete event. However, it now has a + problem - it thinks its agent is the ;1 side of the Local + channel, as it never received the optimization event. At the same + time, that channel is already gone. This results in getting NULL + from the Stasis cache. What's more, we can't really wait for the + optimization message, as we are currently handling the hangup of + the channel that the optimization event would tell us to use. + This patch modifies the behavior in app_queue such that, since we + still have a lot of pertinent queue information (interface, queue + name, etc.), we now raise the event with what information we + know. The channels involved now may or may not be present. Users + will still at least get the "AgentComplete" event, which + "completes" the known Agent information. Review: + https://reviewboard.asterisk.org/r/2878/ (closes issue + ASTERISK-22507) Reported by: Richard Mudgett ........ Merged + revisions 400060 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Fix crash when appending a manager + channel variable In r399887, a minor performance improvement was + introduced by not allocating the manager variable struct if it + wasn't used. Unfortunately, when directly accessing an + ast_channel struct, manager assumed that the struct was always + allocated. Since this was no longer the case, things got a bit + crashy. This fixes that problem by simply bypassing appending + variables if the manager channel variable struct isn't there. + ........ Merged revisions 400058 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 21:58 +0000 [r400016-400021] Richard Mudgett <rmudgett@digium.com> + + * apps/app_cdr.c, res/res_parking.c, /: app_cdr and res_parking: + Fix some resource leaks. * app_cdr left the ResetCDR application + registered. * res_parking leaked a ref to config global. (closes + issue ASTERISK-22566) Reported by: Corey Farrell Patches: + ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey + Farrell ........ Merged revisions 400020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sip/reqresp_parser.c, /, channels/chan_sip.c: chan_sip: + Increase some scratch buffer sizes dealing with caller id. * + Eliminated an unnecessary initialization in check_user_full(). + (closes issue ASTERISK-22477) Reported by: Michael Shepelev + ........ Merged revisions 400013 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400014 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400015 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 19:18 +0000 [r400000] Sean Bright <sean@malleable.com> + + * configs/sip.conf.sample: Remove some trailing whitespace and + steal revision 400000. + +2013-09-27 18:28 +0000 [r399991] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip.c, res/res_pjsip_session.c, + include/asterisk/res_pjsip.h, res/res_pjsip.exports.in: + res_pjsip: crash when using localnet and + external_signaling_address options There was a collision of + mod_data use on the transaction between using a nat hook and an + session response callback. During state change it was assumed + what was in the mod_data was nothing or the response callback. + However, it was possible for it to also contain a nat hook thus + resulting in a bad cast and a crash. Added the ability to store + multiple data elements in mod_data via a hash table. In this + instance, mod_data now stores a hash table of the two values that + can be retrieved using an associated string key. (closes issue + ASTERISK-22394) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2843/ ........ Merged + revisions 399990 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 17:46 +0000 [r399978] Jonathan Rose <jrose@digium.com> + + * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip: + Reject calls on 200 OKs if no SDP has been received When Asterisk + receives a 200 OK in response to an invite, that peer should have + sent an SDP at some point by then. If the channel has never + received an SDP, media won't have been set and the remote address + won't be known. Endpoints in general should not be doing this. + This patch makes it so that Asterisk will simply hang up a call + if it sends a 200 OK at this point. So far this odd behavior for + endpoints has only been observed in tests which involved manually + created SIP transactions in SIPp. (closes issue ASTERISK-22424) + Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/2827/ ........ Merged + revisions 399939 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399962 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 17:11 +0000 [r399938] Richard Mudgett <rmudgett@digium.com> + + * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c, + /: astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a + strange feature that came into the world under suspicious + circumstances to support an abuse of the ao2_container by + chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is + safe to remove it. The simplified code should help performance + slightly and make understanding the code easier. Review: + https://reviewboard.asterisk.org/r/2887/ ........ Merged + revisions 399937 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 14:35 +0000 [r399925] Mark Michelson <mmichelson@digium.com> + + * /, bridges/bridge_native_rtp.c: Fix refleaks of ast_rtp_instance + structures. These refleaks were causing bridged calls not to + close their RTP ports. Thus a call would leave open 4 ports (RTP + for party A, RTCP for party A, RTP for party B, and RTCP for + party B). This led to an eventual depletion of available RTP + ports. ........ Merged revisions 399924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 14:08 +0000 [r399913] Kinsey Moore <kmoore@digium.com> + + * tests/test_cel.c, main/cel.c, /, include/asterisk/cel.h: Restore + usefulness of the CEL Peer field This change makes the CEL peer + field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and + fills the field with a comma-separated list of all channels in + the bridge other than the channel that is entering or exiting the + bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes + issue ASTERISK-22393) ........ Merged revisions 399912 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-26 18:51 +0000 [r399898] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h, + res/res_pjsip.exports.in, /, res/res_pjsip/security_events.c: + pjsip: race condition in registrar While handling a registration + request a race condition could occur if/when two+ clients + registered at the same time. This happened when one request + obtained a copy of the current contacts for an AOR and another + request did the same before the first request updated. Thus the + second would update and overwrite the first (or vice-versa + depending on which actually updated first). In the case of it + being the same contact two "add" events would be raised. pjsip + registration handling is now serialized to alleviate this issue. + (closes issue AST-1213) Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2860/ ........ Merged + revisions 399897 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-26 14:13 +0000 [r399875] Rusty Newton <rnewton@digium.com> + + * /, apps/app_dial.c: Adding a few words to the Dial option 'r' + help text to clarify its tone argument description ........ + Merged revisions 399874 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 20:38 +0000 [r399844] Richard Mudgett <rmudgett@digium.com> + + * channels/sig_ss7.c, channels/chan_dahdi.c, /: chan_dahdi: CLI + "core stop gracefully" has needless delay for PRI and SS7. The + PRI and SS7 link control threads are not stopped correctly when + the chan_dahdi.so module is unloaded. The link control threads + pri_dchannel() and ss7_linkset() are not awakened from a poll() + to cancel the thread. * Added a SIGURG signal after requesting + the thread cancel to break the link control thread poll() + immediately. For SS7 it was slightly worse, the link poll() + timeout would always be whatever was the last libss7 scheduled + event time used. If no libss7 scheduled event was pending, the + thread could run more often than necessary. * Set nextms to 60 + seconds for the ss7_linkset() poll() if there is no other libss7 + scheduled event. ........ Merged revisions 399818 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399834 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399842 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 19:43 +0000 [r399799] Rusty Newton <rnewton@digium.com> + + * /, res/res_pjsip.c: Broke the build - Fixing XML DTD violation + added in r399782, missing <para> tags inside a <note> ........ + Merged revisions 399798 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 19:29 +0000 [r399797] Michael L. Young <elgueromexicano@gmail.com> + + * /, channels/chan_sip.c: chan_sip: Fix Realtime Peer Update + Problem When Un-registering And Expires Header In 200ok 1st Issue + When a realtime peer sends an un-REGISTER request, Asterisk + un-registers the peer but the database table record still has + regseconds and fullcontact for the peer. This results in calls + attempting to be routed to the peer which is no longer + registered. The expected behavior is to get busy/congested when + attempting to call an un-registered peer through the dialplan. + What was discovered is that we are clearing out the peer's + registration in the database in parse_register_contact() when + calling expire_register() but then upon returning from + parse_register_contact(), update_peer() is run which stores back + in the database table regseconds and fullcontact. 2nd Issue The + reporter pointed out that the 200 ok being returned by Asterisk + after un-registering a peer contains a Contact header with + ;expires= and the Expires header is not set to 0. This is + actually a regression. Tests were created for this second issue + (ASTERISK-22548). The tests have been reviewed and a Ship It! was + received on those tests. This patch does the following: * Do not + ignore the Expires header value even when it is set to 0. The + patch sets the pvt->expiry earlier on in the function so that it + is set properly and used. * If pvt->expiry is 0, do not call + update_peer since that means the peer has already been + un-registered and there is no need to update the database record + again since nothing has changed. (closes issue ASTERISK-22428) + Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L. + Young Patches: + asterisk-22428-rt-peer-update-and-expires-header.diff by Michael + L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2869/ ........ Merged + revisions 399794 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399795 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399796 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 18:38 +0000 [r399782] Rusty Newton <rnewton@digium.com> + + * /, res/res_pjsip.c: Fixing documentation for the configOption + "external_media_address" of both Endpoints and Transports + Re-using some of Mark Michelson's text from an E-mail discussion + for: * Modifying synopsis for both options * Adding description + to both options * Changing name of "external_media_address" for + Endpoint configuration to "media_address" in anticipation of the + option name being changed. (As it is not really specific to + external destinations) (issue ASTERISK-22405) (closes issue + ASTERISK-22405) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2850/ ........ Merged + revisions 399781 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-24 22:55 +0000 [r399737-399750] Richard Mudgett <rmudgett@digium.com> + + * /, main/astobj2.c: astobj2: Made use OBJ_SEARCH_xxx identifiers + as field enum values internally. * Made ao2_unlink to protect + itself from stray OBJ_SEARCH_xxx values passed in. ........ + Merged revisions 399749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Prevent some needless + breaking of the native IAX2 bridge. * Clean up some twisted code + in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and + AST_CONTROL_SRCCHANGE to a list of frames to prevent the native + bridge loop from breaking. * Passing the + AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a + native IAX2 bridge. (issue ABE-2912) Review: + https://reviewboard.asterisk.org/r/2870/ ........ Merged + revisions 399697 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399708 from + http://svn.asterisk.org/svn/asterisk/branches/11 For v12 and + above this is really just documentation until IAX2 native + bridging is restored. ........ Merged revisions 399736 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-24 19:22 +0000 [r399667-399696] Matthew Jordan <mjordan@digium.com> + + * apps/app_queue.c, /: app_queue: Don't be quite so aggressive in + initializing the array We only need the first character. ........ + Merged revisions 399695 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_queue.c, /: app_queue: Initialize array holding + MixMonitor exec options If the channel variable MONITOR_EXEC is + set, app_queue will pass the specified execution parameters to + the MixMonitor application when a queue is recorded. If that + channel variable is not set, the buffer that holds the escaped + value was not being initialized to NULL, and so would be passed + to the MixMonitor application with garbage. Hilarity ensued as + app_mixmonitor attempted to execute gobeldy-gook. ........ Merged + revisions 399681 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_bridges.c, tests/test_cdr.c, main/cdr.c, /: Fix a + performance problem CDRs There is a large performance price + currently in the CDR engine. We currently perform two + ao2_callback calls on a container that has an entry for every + channel in the system. This is done to create matching pairs + between channels in a bridge. As such, the portion of the CDR + logic that this patch deals with is how we make pairings when a + channel enters a mixing bridge. In general, when a channel enters + such a bridge, we need to do two things: (1) Figure out if anyone + in the bridge can be this channel's Party B. (2) Make pairings + with every other channel in the bridge that is not already our + Party B. This is a two step process. In the first step, we look + through everyone in the bridge and see if they can be our Party B + (single_state_process_bridge_enter). If they can - yay! We mark + our CDR as having gotten a Party B. If not, we keep searching. If + we don't find one, we wait until someone joins who can be our + Party B. Step 2 is where we changed the logic + (handle_bridge_pairings and bridge_candidate_process). + Previously, we would first find candidates - those channels in + the bridge with us - from the active_cdrs_by_channel container. + Because a channel could be a candidate if it was Party B to an + item in the container, the code implemented multiple + ao2_container callbacks to get all the candidates. We also had to + store them in another container with some other meta information. + This was rather complex and costly, particularly if you have 300 + Local channels (600 channels!) going at once. Luckily, none of it + is needed: when a channel enters a bridge (which is when we're + figuring all this stuff out), the bridge snapshot tells us the + unique IDs of everyone already in the bridge. All we need to do + is: For all channels in the bridge: If the channel is us or our + Party B that we got in step 1, skip it Compare us and the + candidate to figure out who is Party A (based on some specific + rules) If we are Party A: Make a new CDR for us, append it to our + chain, and set the candidate as Party B If they are Party A: If + they don't have a Party B: Make a new CDR for them, append us to + their chain, and us as Party B Otherwise: Copy us over as Party B + on their existing CDR. This patch does that. Because we now use + channel unique IDs to find the candidates during bridging, + active_cdrs_by_channel now looks up things using uniqueid instead + of channel name. This makes the more complex code simpler; it + does, however, have the drawback that dialplan applications and + functions will be slightly slower as they have to iterate through + the container looking for the CDR by name. That's a small price + to pay however as the bridging code will be called a lot more + often. This patch also does two other minor changes: (1) It + reduces the container size of the channels in a bridge snapshot + to 1. In order to be predictable for multi-party bridges, the + order of the channels in the container must be stable; that is, + it must always devolve to a linked list. (2) CDRs and the + multi-party test was updated to show the relationship between two + dialed channels. You still want to know if they talked - + previously, dialed channels were always ignored, which is wrong + when they have managed to get a Party B. (closes issue + ASTERISK-22488) Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/2861/ ........ Merged + revisions 399666 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-23 12:03 +0000 [r399625] Joshua Colp <jcolp@digium.com> + + * res/res_pjsip.c, res/res_pjsip_session.c, /: Fix crash in + res_pjsip on load if error occurs, and prevent unloading of + res_pjsip and res_pjsip_session. During load time in res_pjsip if + an error occurred the operation would attempt to rollback all + operations done during load. This is not permitted by PJSIP as it + will assert if the operation has not been done. This fix changes + the code so it will only rollback what has been initialized + already. Further changes also prevent res_pjsip and + res_pjsip_session from being unloaded. This is due to limitations + within PJSIP itself. The library environment can only be changed + to a certain extent and does not provide the ability, currently, + to deinitialize certain required functionality. (closes issue + ASTERISK-22474) Reported by: Corey Farrell ........ Merged + revisions 399624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-21 04:49 +0000 [r399578-399608] Richard Mudgett <rmudgett@digium.com> + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix ref leaks in + ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the + loop so it is unref'ed after every loop. Moved message_blob to + loop and switched it to a regular variable. The regular variable + was used since message_blob is used in a very contained way. + (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches: + rtcp_report-leak.patch (license #5909) patch uploaded by Corey + Farrell Tested by: Corey Farrell ........ Merged revisions 399607 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/media_index.c: media_index: Fix + process_description_file() memory leak of file_id_persist. + ........ Merged revisions 399596 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/features_config.c: features_config: Fix config ref leak + of parkinglots. This leak happend for just about every channel + created. ........ Merged revisions 399585 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_queue.c: app_queue: Fix json blob ref leak. The json + ref from queue_member_blob_create() was never released. ........ + Merged revisions 399583 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/json.c, /: json: Make it obvious that ast_json_unref() is + NULL safe. It looked like the safety check was done after the + NULL pointer was used. ........ Merged revisions 399576 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 22:44 +0000 [r399566] Kinsey Moore <kmoore@digium.com> + + * main/config_options.c, /: Ensure global types in the config + framework are initialized If a config object was allocated but + one of its global objects was never encountered, then the global + object's defaults were never applied. Ensure that global objects + are initialized properly upon allocation instead of on + configuration. Review: https://reviewboard.asterisk.org/r/2866/ + ........ Merged revisions 399564 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 22:06 +0000 [r399554] Jonathan Rose <jrose@digium.com> + + * main/dial.c, /: originate/call forwarding: Fix a crash when + forwarding a call from originate (closes issue ASTERISK-22487) + Reported by: David M. Lee Review: + https://reviewboard.asterisk.org/r/2868/ ........ Merged + revisions 399553 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 16:18 +0000 [r399533] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_pjsip.c: Add a missing session supplement + unregistration in chan_pjsip for ACKs. (closes issue + ASTERISK-22453) Reported by: Corey Farrell Patches: + chan_pjsip_session_unregister_supplement.patch uploaded by Corey + Farrell (license 5909) ........ Merged revisions 399531 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 14:26 +0000 [r399515] Kevin Harwell <kharwell@digium.com> + + * /, main/logger.c: Fix memory leak in logger. Fixed a memory leak + discovered in the logger where a temporary string buffer was not + being freed. (closes issue ASTERISK-22540) Reported by: John + Hardin ........ Merged revisions 399513 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399514 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-19 23:20 +0000 [r399503] Richard Mudgett <rmudgett@digium.com> + + * /, main/optional_api.c: optional_api: Make always use the + standard malloc functions even with MALLOC_DEBUG. ........ Merged + revisions 399501 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-19 17:01 +0000 [r399459] Jonathan Rose <jrose@digium.com> + + * /, channels/chan_sip.c: chan_sip: Make direct media reinvites for + T38 put Asterisk in the media path Prior to this patch, Asterisk + would incorrectly use the previous endpoint addresses in SDP in + spite of providing its own port. T38 is never meant to be done + through directmedia and Asterisk should always be in the media + path for these streams. (closes issue ASTERISK-17273) Reported + by: Kevin Stewart (closes issue ASTERISK-18706) Reported by: + Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/ + ........ Merged revisions 399456 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 20:04 +0000 [r399405] Kinsey Moore <kmoore@digium.com> + + * /, main/abstract_jb.c: Fix jitter buffer log file creation This + adjusts '/'-to-'#' replacement to replace all instances of '/' + instead of just the first to ensure that the jitter buffer log + file gets the correct name as per Richard Kenner's suggestion. + (closes issue ASTERISK-21036) Reported by: Richard Kenner + ........ Merged revisions 399402 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399403 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399404 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 17:23 +0000 [r399368-399378] Matthew Jordan <mjordan@digium.com> + + * /, build_tools/prep_tarball: Update prep_tarball with new + documentation files on the Asterisk wiki This will now pull both + a command reference for the version being prepared, as well as an + Admin Guide that applies to all versions of Asterisk. (issue + ASTERISK-22439) Reported by: Olle Johansson ........ Merged + revisions 399351 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399373 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399376 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when + a timing module isn't loaded If bridge_softmix fails to be + created because no timing source is present in Asterisk, this + will currently fail gracefully but with (most likely) a generic + error message by whatever module tried to create the softmix + bridge. This patch adds a more explicit warning so you can + actually diagnose and fix the problem. Review: + https://reviewboard.asterisk.org/r/2857/ ........ Merged + revisions 399353 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399365 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 17:15 +0000 [r399352] Richard Mudgett <rmudgett@digium.com> + + * main/config_options.c: Make config framework able to reload + module configs with multiple config files. The config framework + is supposed to be able to load configs that come from multiple + config files. The principle example is chan_sip's sip.conf and + users.conf. Unfortunately, it only does this correctly on initial + load. This patch causes the module's config to be reloaded + entirely if any of the config files change. (closes issue + ASTERISK-22009) Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/2859/ + +2013-09-18 14:56 +0000 [r399340] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_messaging.c, /: res_pjsip_messaging: Register + message technology as pjsip pjsip's message technology was being + registered as 'sip', which was causing it to not load due it + conflicting with chan_sip's registered 'sip' technology for + messaging. It now registers as 'pjsip'. However, due to this + change the "to" field for outgoing pjsip messages need to be + prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to + res_pjsip_messaging will automatically have their "to" fields + altered in order to accommodate the change. Outgoing messages + also handle changing it back to 'sip' before being sent so the + pjsip library will properly handle it. (closes issue + ASTERISK-22445) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2833/ ........ Merged + revisions 399339 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 00:13 +0000 [r399295] Michael L. Young <elgueromexicano@gmail.com> + + * /, main/features_config.c: Fix Segfault In features-config.c When + Application Has No Arguments Some applications do not require + arguments. Therefore, when parsing application maps in + features.conf, it is possible that app_data will be set to NULL. + * This patch sets app_data to "" if it is NULL. Review: + https://reviewboard.asterisk.org/r/2804 ........ Merged revisions + 399294 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 23:10 +0000 [r399284] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip_t38.c, include/asterisk/res_pjsip.h, /: Change the + "external_media_address" PJSIP endpoint option to + "media_address". The endpoint option does not apply to + communication with external entities. Rather, the option is + applied to all communications with the endpoint. The + external_media_address transport configuration option may + override the endpoint option if it turns out that we are going to + be communicating with an external entity. Two things of note: 1) + I have not updated the XML documentation. This is being taken + care of by Rusty as part of his work on issue ASTERISK-22405 2) + This commit is likely to cause testsuite failures since there are + tests that use the external_media_address endpoint option, and + they will need to be changed over. Well, I'm planning to get that + updated ASAP after this commit. (closes issue ASTERISK-22528) + reported by Rusty Newton ........ Merged revisions 399283 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 18:44 +0000 [r399269] Kevin Harwell <kharwell@digium.com> + + * main/logger.c, main/asterisk.c, /: Remote console: more output + discrepancies The remote console continued to have issues with + its output. In this case CLI command output would either not show + up (if verbose level = 0) or would contain verbose prefixes (if + verbose level > 0) once log messages were sent to the remote + console. The fix now now adds verbose prefix data to all new + lines contained in a verbose log string. (closes issue + ASTERISK-22450) Reported by: David Brillert (closes issue + AST-1193) Reported by: Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/2825/ ........ Merged + revisions 399267 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399268 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 17:55 +0000 [r399258] Richard Mudgett <rmudgett@digium.com> + + * /, include/asterisk/features_config.h: Fix doxygen to use correct + units of features.conf options. ........ Merged revisions 399257 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 17:10 +0000 [r399238-399248] Mark Michelson <mmichelson@digium.com> + + * main/bridge_basic.c, main/features_config.c, /: Fix other + timeouts (atxferloopdelay and atxfernoanswertimeout) to use + seconds instead of milliseconds. Thanks to Richard Mudgett for + pointing this out. ........ Merged revisions 399247 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/features_config.c, /, include/asterisk/features_config.h, + main/bridge_basic.c: Switch transferdigittimeout to be configured + as seconds instead of milliseconds. This was an unintentional + consequence of the update of features.conf to use the config + framework in Asterisk 12. Thanks to Marco Signorini on the + Asterisk developers list for pointing out the problem. ........ + Merged revisions 399237 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 14:58 +0000 [r399226] Kevin Harwell <kharwell@digium.com> + + * apps/confbridge/conf_state_multi_marked.c, /: Confbridge: empty + conference not being torn down Confbridge would not properly tear + down an empty conference bridge when all users were kicked via + end_marked=yes and at least one user was also set to wait_marked. + This occurred because while end_marked users were being kicked + and at least one was also set to wait_marked then the leave + wait_marked handler would be called on that user, but there would + be no waiting user (still considered active). The waiting users + would decrement and now be negative. The conference would remain, + but be put into an inactive state. The solution was to move from + the active list to the wait list, those users with wait_marked + set right before kicking. This allows both the active and wait + users to decrement correctly and the confbridge to tear down + properly. A crashed also occurred when trying to list the + specific conference from the CLI. This happened because the + conference specified was invalid. Since the conference properly + tears down now there is no way to reference it thus alleviating + the crash as well. (closes issue ASTERISK-21859) Reported by: + Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/ + ........ Merged revisions 399222 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399225 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-16 18:36 +0000 [r399161-399208] Richard Mudgett <rmudgett@digium.com> + + * tests/test_ari_model.c, /: Fix module load errors for + test_ari_model.so. You cannot use a function pointer variable + with an external function from another dynamically loaded module + because data variables are always resolved even with RTLD_LAZY. * + Added wrapper functions for ast_ari_validate_int() and + ast_ari_validate_string() to use instead for the function pointer + variable. (closes issue ASTERISK-22457) Reported by: David M. Lee + ........ Merged revisions 399207 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_speech_utils.c, /, res/res_speech.exports.in: + app_speech_utils: Fix unresolved symbol ast_speech_get_setting(). + Fixes regression introduced by -r374096. * Made + res_speech.export.in export ast_* symbols instead of specific + functions. * Made app_speech_utils.c declare that it is dependent + upon res_speech. (issue ASTERISK-17136) Reported by: Richard + Kenner ........ Merged revisions 399197 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix saving the wrong expiry + time in astdb. When a new IAX2 client registers, the astdb + database is updated with the value of minregexpire defined in + iax.conf instead of using the expiry time that is provided by the + client. The provided expiry time of the client is updated after + inserting the astdb entry. As a consequence, restarting or + reloading asterisk creates clients whose registration may expire + before they reregister. The clients are therefore unavailable + after minregexpire seconds until they reregister. * Move updating + of the expiry time to before inserting into the astdb. (closes + issue ASTERISK-22504) Reported by: Stefan Wachtler Patches: + chan_iax2.c.patch (license #6533) patch uploaded by Stefan + Wachtler ........ Merged revisions 399158 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399159 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399160 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-16 02:37 +0000 [r399147] Matthew Jordan <mjordan@digium.com> + + * main/cdr.c, /: Filter internal channels out of bridge enter/leave + message handling Some channels exist merely as an implementation + detail in Asterisk, such as ConfBridge's announcer/recorder + channels. These channels should never be exposed to the outside + world, or to interfaces that report on Asterisk. We already + filter out such channels in snapshot processing; however, we + failed to filter out bridge related messages that involved these + channels. This patch filters out bridge related messages that are + for such channels. This prevents a spurious WARNING message from + being displayed when those channels move in and out of bridges. + ........ Merged revisions 399146 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 22:19 +0000 [r399138] Richard Mudgett <rmudgett@digium.com> + + * res/parking/parking_bridge_features.c, apps/app_agent_pool.c, + include/asterisk/features.h, main/channel.c, + res/parking/parking_tests.c, include/asterisk/bridge_channel.h, + main/features.c, tests/test_cel.c, main/bridge_channel.c, + tests/test_cdr.c, apps/confbridge/conf_chan_announce.c, + include/asterisk/bridge.h, res/res_pjsip_refer.c, /, + channels/chan_sip.c, res/stasis/control.c, main/bridge.c, + main/bridge_basic.c, main/core_unreal.c, + res/parking/parking_applications.c, main/core_local.c: Restore + Dial, Queue, and FollowMe 'I' option support. The Dial, Queue, + and FollowMe applications need to inhibit the bridging initial + connected line exchange in order to support the 'I' option. * + Replaced the pass_reference flag on ast_bridge_join() with a + flags parameter to pass other flags defined by enum + ast_bridge_join_flags. * Replaced the independent flag on + ast_bridge_impart() with a flags parameter to pass other flags + defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, + and FollowMe applications are now the only callers of + ast_bridge_call() and ast_bridge_call_with_flags(), changed the + calling contract to require the initial COLP exchange to already + have been done by the caller. * Made all callers of + ast_bridge_impart() check the return value. It is important. As a + precaution, I also made the compiler complain now if it is not + checked. * Did some cleanup in parking_tests.c as a result of + checking the ast_bridge_impart() return value. An independent, + but associated change is: * Reduce stack usage in + ast_indicate_data() and add a dropping redundant connected line + verbose message. (closes issue ASTERISK-22072) Reported by: + Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ + ........ Merged revisions 399136 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 20:55 +0000 [r399101] David M. Lee <dlee@digium.com> + + * /, main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not + defined. If MALLOC_DEBUG is enabled, then the debug destructor + for the container is used, which would erroneously write to + /tmp/refs. This patch only uses the debug destructor if ref_debug + is used. (closes issue ASTERISK-22536) ........ Merged revisions + 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 399099 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399100 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:50 +0000 [r399082-399084] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c, + include/asterisk/res_pjsip.h, res/res_pjsip.exports.in, /: Create + more accurate Contact headers for dialogs when we are the UAS. + (closes issue AST-1207) reported by John Bigelow Review: + https://reviewboard.asterisk.org/r/2842 ........ Merged revisions + 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/config_auth.c, /, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_pjsip_authenticator_digest.c: Change how realms are + handled for outbound authentication. With this change, if no + realm is specified in an outbound auth section, then we will + simply match the realm that was present in the 401/407 challenge. + (closes issue ASTERISK-22471) Reported by George Joseph (closes + issue ASTERISK-22386) Reported by Rusty Newton Patches: + outbound_auth_realm_v4.patch uploaded by George Joseph (License + #6322) ........ Merged revisions 399059 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:43 +0000 [r399080-399081] David M. Lee <dlee@digium.com> + + * /: Recorded merge of revisions 399035,399049 from + http://svn.asterisk.org/svn/asterisk/branches/12 These were lost + in r399071 + + * /: Put merge tracking for r399039 back. + +2013-09-13 14:27 +0000 [r399071] Rusty Newton <rnewton@digium.com> + + * /, res/res_pjsip_endpoint_identifier_ip.c: Broke the build! + Forgot para tags within my description. + https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304 + ........ Merged revisions 399064 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:22 +0000 [r399042-399051] David M. Lee <dlee@digium.com> + + * res/res_pjsip_log_forwarder.c (added), res/res_pjsip_logger.c, + res/res_rtp_asterisk.c, /: res_pjsip: Forward PJSIP logging to + Asterisk logging This patch uses PJSIP's pj_log_set_log_func() to + forward PJSIP's log messages to Asterisk's logger. This is done + in a new module: res_pjsip_log_forwarder.so. This patch sets + defaultenabled on the existing res_pjsip_logger.so to no, since + logging every SIP packet seems a bit odd to do by default, and is + (hopefully) less necessary with regular PJSIP logging. It also + removes res_rtp_asterisk's disabling of PJSIP logging. (closes + issue ASTERISK-22360) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2830/ ........ Merged + revisions 399049 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_http_websocket.c: ARI: Fix WebSocket response when + subprotocol isn't specified When I moved the ARI WebSocket from + /ws to /ari/events, I added code to allow a WebSocket to connect + without specifying the subprotocol if there's only one + subprotocol handler registered for the WebSocket. Naively, I + coded it to always respond with the subprotocol in use. + Unfortunately, according to RFC 6455, if the server's response + includes a subprotocol header field that "indicates the use of a + subprotocol that was not present in the client's handshake [...], + the client MUST _Fail the WebSocket Connection_.", emphasis + theirs. This patch correctly omits the Sec-WebSocket-Protocol if + one is not specified by the client. (closes issue ASTERISK-22441) + Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged + revisions 399039 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:17 +0000 [r399036] Kinsey Moore <kmoore@digium.com> + + * /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This + change ensures that MeetMeAdmin commands requiring a user + actually get a user and fixes another issue where an extra + dereference could occur for a last-entered user being ejected if + a user identifier was also provided. (closes issue + ASTERISK-21907) Reported by: Alex Epshteyn Review: + https://reviewboard.asterisk.org/r/2844/ ........ Merged + revisions 399033 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399034 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399035 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 13:28 +0000 [r399032] Rusty Newton <rnewton@digium.com> + + * /, res/res_pjsip_endpoint_identifier_ip.c: 'identify' + configObject doesn't have a synopsis Add a straightforward + synopsis and description to the identify config object in XML + documentation. (issue ASTERISK-22311) (closes issue + ASTERISK-22311) Reported By: Rusty Newton ........ Merged + revisions 399031 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 23:42 +0000 [r399020-399022] Richard Mudgett <rmudgett@digium.com> + + * /, main/bridge.c: CLI bridge: Fix "bridge destroy <id>" and + "bridge kick <id> <chan>" tab completion. These two commands must + deal with the live bridges container for tab completion and not + the stasis cache. ........ Merged revisions 399021 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/bridge.c, /: astobj2: Register the bridges container for + debug inspection. ........ Merged revisions 399019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 23:23 +0000 [r399018] Rusty Newton <rnewton@digium.com> + + * /, res/res_pjsip_acl.c: Documentation fix and improvements to XML + configuration help res_pjsip_acl * One bug fix. Made the synopsis + for "type" to accurate. * changing the usage of "IP-domains" to + "IP addresses" * clarifying the usage for the options, by adding + a relevant description for each * modified other areas of the XML + help for clarity, such as the module description and a few + synopsis changes here and there. See the patch. (issue + ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty + Newton Review: https://reviewboard.asterisk.org/r/2823/ ........ + Merged revisions 399017 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 20:27 +0000 [r399006] Jonathan Rose <jrose@digium.com> + + * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip: + Revert r398835 due to failing tests involving originate (issue + ASTERISK-22424) Reported by: Jonathan Rose ........ Merged + revisions 398977 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398986 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398991 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 16:44 +0000 [r398939] Richard Mudgett <rmudgett@digium.com> + + * main/core_unreal.c, /: core_local: Fix memory corruption race + condition. The masquerade super test is failing on v12 with high + fence violations and crashing. The fence violations are showing + that party id allocated memory strings are somehow getting + corrupted in the bridge_reconfigured_connected_line_update() + function. The invalid string values happen to be the freed memory + fill pattern. After much puzzling, I deduced that the + bridge_reconfigured_connected_line_update() is copying a string + out of the source channel's caller party id struct just as + another thread is updating it with a new value. The copying + thread is using the old string pointer being freed by the + updating thread. A search of the code found the + unreal_colp_redirect_indicate() routine updating the caller party + id's without holding the channel lock. A latent bug in v1.8 and + v11 hatched in v12 because of the bridging and connected line + changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2839/ ........ Merged + revisions 398938 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 15:23 +0000 [r398928] David M. Lee <dlee@digium.com> + + * res/res_pjsip.c, /: Fix symbol collision with pjsua. We shouldn't + be exporting any symbols that start with pjsip_. ........ Merged + revisions 398927 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 00:04 +0000 [r398883-398887] Rusty Newton <rnewton@digium.com> + + * /, apps/app_queue.c: 'queue add member' help text correction You + are adding dial strings to the queue, not channels. An aribitrary + string could be used, but you are typically referencing a + channel. Correcting the command help text. (issue ASTERISK-22263) + (closes issue ASTERISK-22263) Reported By: Rusty Newton ........ + Merged revisions 398884 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398885 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398886 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/chan_dahdi.conf.sample, /: Documentation fix - + waitfordialtone is not boolean, it's time in milliseconds + Changing text in chan_dahdi.conf sample to be accurate. (issue + ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: + Malcolm Davenport ........ Merged revisions 398880 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398881 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-11 20:03 +0000 [r398838] Jonathan Rose <jrose@digium.com> + + * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: + Reject calls without prior SDP on 200 OK If we receive a 200 OK + without SDP, we will now check to see if the remote address has + been established for that channel's RTP session and if the to tag + for that channel has changed from the most recent to tag in a + response less than 200. If either a change has been made since + the last to-tag was received or the remote address is unset, then + we will drop the call. (closes issue ASTERISK-22424) Reported by: + Jonathan Rose Review: + https://reviewboard.asterisk.org/r/2827/diff/#index_header + ........ Merged revisions 398835 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398836 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398837 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-11 18:03 +0000 [r398822] Russell Bryant <russell@russellbryant.com> + + * configs/confbridge.conf.sample, /: Fix typo in + confbridge.conf.sample The denoise filter requires func_speex, + not codec_speex. Fix this in the description of the denoise=yes + option in confbridge.conf. ........ Merged revisions 398820 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398821 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-11 14:23 +0000 [r398808] Kevin Harwell <kharwell@digium.com> + + * res/res_pjsip_caller_id.c, channels/chan_pjsip.c, /: pjsip: + reinvite for connected line updates occurs when it should not + Connected line updates are now only sent out if an actual update + needs to occur. This happens under the following conditions: 1. + The endpoint we are sending to is trusted. 2. Either a + P-Asserted-Identity or Remote Party-ID header needs to be + added/sent. 3. The connected id's number and name are valid. Also + added an SDP when an update is sent out. (closes issue AST-1212) + Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2831/ ........ Merged + revisions 398806 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-10 18:05 +0000 [r398760] Richard Mudgett <rmudgett@digium.com> + + * main/event.c, res/res_musiconhold.c, main/indications.c, + main/asterisk.c, main/xmldoc.c, main/cli.c, /, + funcs/func_dialgroup.c, main/heap.c, + res/res_pjsip/pjsip_configuration.c: Fix incorrect usages of + ast_realloc(). There are several locations in the code base where + this is done: buf = ast_realloc(buf, new_size); This is going to + leak the original buf contents if the realloc fails. Review: + https://reviewboard.asterisk.org/r/2832/ ........ Merged + revisions 398757 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398758 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398759 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-10 17:50 +0000 [r398751-398755] David M. Lee <dlee@digium.com> + + * utils/check_expr.c, /: Fixed utils directory breakage from + r398748, this time with extra hate. ........ Merged revisions + 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398753 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398754 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * utils/check_expr.c, /, utils/ael_main.c, utils/conf2ael.c: Fixed + utils directory breakage from r398648 ........ Merged revisions + 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398749 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398750 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-09 23:29 +0000 [r398732] Richard Mudgett <rmudgett@digium.com> + + * main/astmm.c, /: MALLOC_DEBUG: Change fence magic number to be + completely different from the freed magic number. Race conditions + between freeing a nul terminated string and ast_strdup()'ing it + are more likely to be detected if the fence and freed magic + numbers are completely different. ........ Merged revisions + 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398721 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398726 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-09 22:00 +0000 [r398695] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_endpoint_identifier_ip.c, /: Add extra debugging to + res_pjsip_endpoint_identifier_ip ........ Merged revisions 398694 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-09 20:13 +0000 [r398641-398652] David M. Lee <dlee@digium.com> + + * /, main/utils.c, include/asterisk/lock.h, main/lock.c: Fix + DEBUG_THREADS when lock is acquired in __constructor__ This patch + fixes some long-standing bugs in debug threads that were + exacerbated with recent Optional API work in Asterisk 12. With + debug threads enabled, on some systems, there's a lock ordering + problem between our mutex and glibc's mutex protecting its module + list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one + thread, the module list will be locked before acquiring our + mutex. In another thread, our mutex will be locked before locking + the module list (which happens in the depths of calling + backtrace()). This patch fixes this issue by moving backtrace() + calls outside of critical sections that have the mutex acquired. + The bigger change was to reentrancy tracking for + ast_cond_{timed,}wait, which wrongly assumed that waiting on the + mutex was equivalent to a single unlock (it actually suspends all + recursive locks on the mutex). (closes issue ASTERISK-22455) + Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged + revisions 398648 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398649 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398651 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_channels.h, /, rest-api/api-docs/channels.json: + Multiple revisions 398638-398639 ........ r398638 | dlee | + 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note + about expected behavior of originate ........ r398639 | dlee | + 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note + about expected behavior of originate (the rest of the commit) + ........ Merged revisions 398638-398639 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-08 23:30 +0000 [r398629] Matthew Jordan <mjordan@digium.com> + + * tests/test_cdr.c, /: Update CDR Unit tests to reflect container + changes in r398579 When a channel joins a multi-party bridge, the + ordering of the CDRs that is created is determined by the + ordering of the channels who happen to be in that bridge. When + r398579 changed the number of buckets in the container to + something sensible, it changed the ordering that the CDRs was + created in, causing one of the multiparty tests to fail. This + fixes the test with the now expected ordering. ........ Merged + revisions 398628 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-07 01:03 +0000 [r398603-398620] Kinsey Moore <kmoore@digium.com> + + * /, res/res_xmpp.c: Prevent XMPP timeout on blank responses + Sometimes the Google Voice servers have a bad habit of sending + out 1 byte replies to the xmpp resource. When a blank 1 byte + reply is received from the socket the buffer attempts to wait + (endlessly) for the rest of the reply from google which + effectively blocks the socket and google voice calls will no + longer come into the server. This patch allows the xmpp module to + correctly detect empty packets and send out ping replies to + google. It also sets a socket timeout on the default socket which + prevents the xmpp socket from closing and preventing future + google voice calls from coming into the server. Furthermore + instead of sending an empty reply back to google we send a proper + xmpp ping reply back. This also adds several more socket + messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy + Review: https://reviewboard.asterisk.org/r/2771 Patches: + xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........ + Merged revisions 398618 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398619 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_xmpp.c, res/res_jabber.c: Multiple revisions + 398558,398577 ........ r398558 | kmoore | 2013-09-06 14:28:16 + -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed + MWI The mailbox and context are swapped on the receiving end for + all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and + all more recent versions. This swaps those values to be correct + when publishing to the internal event system from Jabber/XMPP + distributed MWI state. (closes issue ASTERISK-22435) Reported by: + abelbeck Tested by: Michael Keuter Patches: + asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by + abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch + uploaded by abelbeck ........ Merged revisions 398523 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | + 10 lines Commit the remainder of r398523 This is a missing part + of the commit in revision 398523 that corrects the name of a + variable. (issue ASTERISK-22435) ........ Merged revisions 398576 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 398558,398577 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398580 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-06 21:17 +0000 [r398557-398583] Richard Mudgett <rmudgett@digium.com> + + * main/cdr.c, /: cdr: Change the number of container buckets to be + similar to the channels container. * Fix the temporary cdr + candidate containers to use a prime number of buckets. ........ + Merged revisions 398579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/core_local.c, /: core_local: Fix LocalOptimizationBegin AMI + event missing Source channel snapshot. * Fix the + LocalOptimizationBegin AMI event by eliminating an artificial + buffer size limitation that is too small anyway. ........ Merged + revisions 398572 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: cdr: Fix some ref leaks. * Added missing + unregister of the cdr container in cdr_engine_shutdown(). * Fixed + ref leak in off nominal path of cdr_object_alloc(). * Removed + some unnecessary NULL checks in cdr_object_dtor(). ........ + Merged revisions 398562 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/astobj2.h, main/cel.c, main/features_config.c, + apps/app_agent_pool.c, main/cdr.c, main/udptl.c, /, + main/parking.c, main/stasis_config.c: astobj2: Add warn unused + attribute to some functions. * Fixed resulting warnings with + improper use of ao2_global_obj_replace(). * Made a couple uses of + ao2_global_obj_replace_unref(x, NULL) into the equivalent and + more appropriate ao2_global_obj_release() call. ........ Merged + revisions 398533 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-06 18:53 +0000 [r398512-398522] Kinsey Moore <kmoore@digium.com> + + * main/http.c, /, res/stasis/app.c: Fix build warnings When + AST_DEVMODE is not defined, ast_asserts are not compiled into the + binary. In some cases, this means variables are not referenced or + are set but unused which causes warnings to show up. (closes + issue ASTERISK-22446) Reported by: Jason Parker (qwell) ........ + Merged revisions 398521 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_h323.c: Fix chan_h323 compilation This fixes the + things in chan_h323 that were missed or ignored in the great + channel opaquification and gets chan_h323 back into a compiling + state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov + Patches: chan_h323.patch uploaded by Dmitry Melekhov ........ + Merged revisions 398510 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398511 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-05 21:48 +0000 [r398384-398499] Richard Mudgett <rmudgett@digium.com> + + * /, main/astobj2.c: astobj2: Only define ao2_bt() once. * Make + ao2_bt() not use single char variable names. * Fix ao2_bt() + formatting. ........ Merged revisions 398498 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Reduce indentation in + __attempt_transmit(). * Reduce indentation in + __attempt_transmit(). * Don't update the static last error time + variable every time in __schedule_action() and socket_read(). + ........ Merged revisions 398456 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker + thread idle_list. * Fix stray reference to idle_list in + cleanup_thread_list(). This may be the reason for the note in + iax2_process_thread() about threads not being removed from the + task lists. * Move cleanup_thread_list(&idle_list) to after the + other lists are cleaned up. ........ Merged revisions 398416 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398417 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398418 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock + avoidance. * Fix bridgecallno deadlock avoidance. When doing + deadlock avoidance, you need to retest the status of values for + each loop to see if you still need the lock for bridgecallno. * + As a safety check, after acquiring the bridgecallno lock you + should check if iaxs[bridgecallno] is NULL just like the current + callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE + to after processing any deferred frames to ensure that the + iostate is IDLE when it is placed back into the idle list. + defer_full_frame() tries to ensure iax2_process_thread() wakes up + to process the frame. ........ Merged revisions 398379 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398380 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398381 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-05 14:10 +0000 [r398369] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_outbound_registration.c: Clarify server_uri and + client_uri registration settings. Used some of Rusty's suggested + language plus also included more SIPesque descriptions of where + the URIs are actually used in an outgoing REGISTER. (closes issue + ASTERISK-22390) reported by Rusty Newton ........ Merged + revisions 398368 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 23:07 +0000 [r398304] Richard Mudgett <rmudgett@digium.com> + + * channels/iax2/parser.c, /: chan_iax2: Add missing control frame + names to debug frame decode output. ........ Merged revisions + 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398302 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398303 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 22:49 +0000 [r398300] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_outbound_authenticator_digest.c: Give more + detail regarding failures to create request with auth + credentials. (issue ASTERISK-22386) ........ Merged revisions + 398299 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 21:37 +0000 [r398284-398287] Jonathan Rose <jrose@digium.com> + + * /, tests/test_voicemail_api.c: unit tests: test_voicemail_api + leaks stringfields from snapshots (closes issue ASTERISK-22414) + Reported by: Corey Farrell Patches: + test_voicemail_api-leaks-11.patch uploaded by coreyfarrell + (license 5909) ........ Merged revisions 398285 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398286 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_voicemail.c, /: app_voicemail: Fix leaking config + objects when msg_id doesn't match (issues ASTERISK-22414) + Reported by: Corey Farrell Patch: + test_voicemail_api-leaks-11.patch uploaded by coreyfarrell + (license 5909) ........ Merged revisions 398281 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398283 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 16:03 +0000 [r398238] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output + printed with arbitrary verbose levels. Fix the misdn debug output + to remote consoles. chan_misdn uses ast_console_puts() which + doesn't know about verbose levels. Better to use ast_verbose() + instead. Without this patch the misdn debug messages are appended + to the verbose level which ever was set by the message sent to + the console before, i.e. any undefined level. (closes issue + AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch + (license #6372) patch uploaded by Guenther Kelleter ........ + Merged revisions 398235 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398236 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398237 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 14:32 +0000 [r398227] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip_outbound_registration.c: Debug messages for + pjsip outbound registration Added debug messages indicating that + an outbound registration attempt was made and it was successful + in pjsip. (closes issue ASTERISK-22388) Reported by: Rusty Newton + ........ Merged revisions 398226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 20:28 +0000 [r398217] Alexandr Anikin <may@telecom-service.ru> + + * /, addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling + on empty tcs received ........ Merged revisions 398214 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398215 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 18:09 +0000 [r398207] Kinsey Moore <kmoore@digium.com> + + * res/res_pjsip_dtmf_info.c, /: Prevent a crash in + res_pjsip_dtmf_info.c This change makes sure that a content type + header exists before checking the contents of the header against + known SIP INFO DTMF content types. ........ Merged revisions + 398206 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 17:19 +0000 [r398205] David M. Lee <dlee@digium.com> + + * Makefile, /: Fixed 'make clean' for wiki docs ........ Merged + revisions 398198 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 14:29 +0000 [r398197] Walter Doekes <walter+asterisk@wjd.nu> + + * /, cel/cel_custom.c: Be a little more verbose when loading + cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/ + ........ Merged revisions 398167 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398168 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 20:58 +0000 [r398150] David M. Lee <dlee@digium.com> + + * main/asterisk.c, include/asterisk/optional_api.h, /, + main/optional_api.c: Fix graceful shutdown crash. The cleanup + code for optional_api needs to happen after all of the optional + API users and providers have unused/unprovided. Unfortunately, + regsitering the atexit() handler at the beginning of main() isn't + soon enough, since module destructors run after that. ........ + Merged revisions 398149 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 20:37 +0000 [r398148] Rusty Newton <rnewton@digium.com> + + * /, configs/pjsip.conf.sample: New pjsip.conf.sample (issue + ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt + Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........ + Merged revisions 398147 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 19:55 +0000 [r398124-398140] Kevin Harwell <kharwell@digium.com> + + * /, res/res_pjsip_outbound_registration.c, + include/asterisk/sorcery.h, res/res_pjsip.c, + res/res_pjsip/config_transport.c, main/sorcery.c: Add a + reloadable option for sorcery type objects Some configuration + objects currently won't place nice if reloaded. Specifically, in + this case the pjsip transport objects. Now when registering an + object in sorcery one may specify that the object is allowed to + be reloaded or not. If the object is set to not reload then upon + reloading of the configuration the objects of that type will not + be reloaded. The initially loaded objects of that type however + will remain. While the transport objects will not longer be + reloaded it is still possible for a user to configure an endpoint + to an invalid transport. A couple of log messages were added to + help diagnose this problem if it occurs. (closes issue + ASTERISK-22382) Reported by: Rusty Newton (closes issue + ASTERISK-22384) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2807/ ........ Merged + revisions 398139 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, res/res_security_log.c, /, channels/chan_sip.c, + main/translate.c, main/named_acl.c, main/indications.c: Fix + various memory leaks main/config.c - cleanup cache fie includes + res/res_security_log.c - unregister logger level + channesl/chan_sip.c - cleanup io context and notify_types + main/translator.c - cleanup at shutdown main/named_acl.c - + cleanup cli commands main/indications.c - + ast_get_indication_tone() unref default_tone_zone if used (closes + issues ASTERISK-22378) Reported by: Corey Farrell Patches: + config_shutdown.patch uploaded by coreyfarrell (license 5909) + res_security_log.patch uploaded by coreyfarrell (license 5909) + chan_sip-11.patch uploaded by coreyfarrell (license 5909) + indications_refleak.patch uploaded by coreyfarrell (license 5909) + named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license + 5909) translate_shutdown.patch uploaded by coreyfarrell (license + 5909) ........ Merged revisions 398102 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398103 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398116 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 18:38 +0000 [r398101] Matthew Jordan <mjordan@digium.com> + + * /, UPGRADE-12.txt (added), UPGRADE.txt: Update UPGRADE.txt file + for Asterisk 12 This simply pulls in the changes that were + breaking from the CHANGES file and updates a few other areas + accordingly. It also removes the 10 => 11 notes, which are + traditionally removed from each major version and stored in the + appropriate UPGRADE-X.txt file. ........ Merged revisions 398100 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 18:30 +0000 [r398064-398099] Jonathan Rose <jrose@digium.com> + + * main/features_config.c, /, main/config_options.c: + features_config: Ignore parkinglots in features.conf instead of + failing to load Parkinglots are defined in res_features.conf now, + but this patch fixes features_config so that features don't fail + to load when parkinglots are present in features.conf Review: + https://reviewboard.asterisk.org/r/2801/ ........ Merged + revisions 398068 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/features_config.c, main/udptl.c, /: features_config: Don't + require features.conf to be present for Asterisk to load (closes + issue ASTERISK-22426) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2806/ ........ Merged + revisions 398020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 17:59 +0000 [r398063] Kevin Harwell <kharwell@digium.com> + + * main/manager.c, /, res/res_agi.c: Memory leak fix + ast_xmldoc_printable returns an allocated block that must be + freed by the caller. Fixed manager.c and res_agi.c to stop + leaking these results. (closes issue ASTERISK-22395) Reported by: + Corey Farrell Patches: manager-leaks-12.patch uploaded by + coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded + by coreyfarrell (license 5909) ........ Merged revisions 398060 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 398061 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398062 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 17:11 +0000 [r398024-398026] Richard Mudgett <rmudgett@digium.com> + + * tests/test_substitution.c, /: test_substitution: Fix failing + test. Revert the -r392190 change. The original test was correct. + The CDR code was actually returning an unititialized buffer. + ........ Merged revisions 398025 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_substitution.c, /: test_substituition: Fix failed test + reporting to actually report failure. You cannot put the "Testing + <blah> pass/fail" on a single line before actually performing the + test. Now any additional failure information is logged before the + test pass/fail announcement. * Added an additional CDR(answer,u) + test. ........ Merged revisions 398018 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398019 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398023 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 16:27 +0000 [r398003-398017] Kevin Harwell <kharwell@digium.com> + + * /, apps/app_mixmonitor.c: Fix memory leaks (closes issue + ASTERISK-22368) Reported by: Corey Farrell Patches: + issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes + (license 5674) ........ Merged revisions 398004 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged + revisions 398011 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398016 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/asterisk.c, /: Check return value on fwrite ........ Merged + revisions 398000 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398002 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 13:40 +0000 [r397987-397990] David M. Lee <dlee@digium.com> + + * rest-api-templates/swagger_model.py, res/ari/ari_websockets.c, + channels/sip/include/sip.h, main/asterisk.c, res/res_ari.c, + tests/test_optional_api.c (added), /, channels/chan_sip.c, + include/asterisk/autoconfig.h.in, configure.ac, + rest-api-templates/res_ari_resource.c.mustache, + res/ari/internal.h, res/res_http_websocket.c, CHANGES, + include/asterisk/compiler.h, include/asterisk/ari.h, + main/loader.c, include/asterisk/optional_api.h, + build_tools/cflags.xml, configure, res/res_ari_events.c, + include/asterisk/http_websocket.h, main/optional_api.c (added): + optional_api: Fix linking problems between modules that export + global symbols With the new work in Asterisk 12, there are some + uses of the optional_api that are prone to failure. The details + are rather involved, and captured on [the wiki][1]. This patch + addresses the issue by removing almost all of the magic from the + optional API implementation. Instead of relying on weak symbol + resolution, a new optional_api.c module was added to Asterisk + core. For modules providing an optional API, the pointer to the + implementation function is registered with the core. For modules + that use an optional API, a pointer to a stub function, along + with a optional_ref function pointer are registered with the + core. The optional_ref function pointers is set to the + implementation function when it's provided, or the stub function + when it's now. Since the implementation no longer relies on + magic, it is now supported on all platforms. In the spirit of + choice, an OPTIONAL_API flag was added, so we can disable the + optional_api if needed (maybe it's buggy on some bizarre platform + I haven't tested on) The AST_OPTIONAL_API*() macros themselves + remained unchanged, so existing code could remain unchanged. But + to help with debugging the optional_api, the patch limits the + #include of optional API's to just the modules using the API. + This also reduces resource waste maintaining optional_ref + pointers that aren't used. Other changes made as a part of this + patch: * The stubs for http_websocket that wrap system calls set + errno to ENOSYS. * res_http_websocket now properly increments + module use count. * In loader.c, the while() wrappers around + dlclose() were removed. The while(!dlclose()) is actually an + anti-pattern, which can lead to infinite loops if the module + you're attempting to unload exports a symbol that was directly + linked to. * The special handling of nonoptreq on systems without + weak symbol support was removed, since we no longer rely on weak + symbols for optional_api. [1]: + https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue + ASTERISK-22296) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2797/ ........ Merged + revisions 397989 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /, + include/asterisk/stasis_app_recording.h, + res/ari/resource_recordings.h, res/res_stasis_recording.c, + res/Makefile, res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, res/stasis_recording (added), + res/ari/resource_recordings.c, res/ari/ari_model_validators.h, + res/res_ari_recordings.c: ARI: Implement /recordings/stored API's + his patch implements the ARI API's for stored recordings. While + the original task only specified deleting a recording, it was + simple enough to implement the GET for all recordings, and for an + individual recording. The recording playback operation was + modified to use the same code for accessing the recording as the + REST API, so that they will behave consistently. There were + several problems with the api-docs that were also fixed, bringing + the ARI spec in line with the implementation. There were some + 'wishful thinking' fields on the stored recording model (duration + and timestamp) that were removed, because I ended up not + implementing a metadata file to go along with the recording to + store such information. The GET /recordings/live operation was + removed, since it's not really that useful to get a list of all + recordings that are currently going on in the system. (At least, + if we did that, we'd probably want to also list all of the + current playbacks. Which seems weird.) (closes issue + ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/ + ........ Merged revisions 397985 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Multiple revisions 397975-397976 ........ r397975 | rmudgett | + 2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line pbx.c: Make + ast_str_substitute_variables_full() not mask variables. ........ + r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013) + | 1 line Revert last commit. ........ Merged revisions + 397975-397976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 01:20 +0000 [r397978] Richard Mudgett <rmudgett@digium.com> + + * main/pbx.c, /: pbx.c: Make pbx_substitute_variables_helper_full() + not mask variables. ........ Merged revisions 397977 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 00:11 +0000 [r397962-397969] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip_pidf.c, /: Sanitize XML output for PIDF bodies. + PJSIP's PIDF API does not replace angle brackets with their + appropriate counterparts for XML. So we have to do it ourself. In + this particular case, the problem had to do with attempting to + place an unsanitized SIP URI into an XML node. Now we don't get a + 488 from recipients of our PIDF NOTIFYs. ........ Merged + revisions 397968 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pidf.c, /: Fix method for creating activities + string in PIDF bodies. The previous method did not allocate + enough space to create the entire string, but adjusted the + string's slen value to be larger than the actual allocation. This + resulted in garbled text in NOTIFY requests from Asterisk. This + method allocates the proper amount of space first and then writes + the content into the buffer. ........ Merged revisions 397960 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 22:49 +0000 [r397959] Kevin Harwell <kharwell@digium.com> + + * apps/app_dumpchan.c, main/logger.c, apps/app_verbose.c, + main/asterisk.c, channels/chan_misdn.c, /: Verbose logging + discrepancies Refactored cases where a combination of + ast_verbose/options_verbose were present. Also in general tried + to eliminate, in as many places as possible, where the + options_verbose global variable was being used. Refactored the + way local and remote consoles handle verbose message logging in + an attempt to solve the various discrepancies that sometimes + would show between the two. (closes issue AST-1193) Reported by: + Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/2798/ ........ Merged + revisions 397948 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397958 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 22:26 +0000 [r397956-397957] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_pubsub.c: Fix when the subscription_terminated + callback is called for subscription handlers. The previous + placement would result in the resubscribe() callback called + instead of the subscription_terminated() callback being called + when a subscription was ended via a SUBSCRIBE request. This would + result in confusing PJSIP and having it throw an assertion. + ........ Merged revisions 397955 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: Fix a race condition where a canceled + call was answered. RFC 5407 section 3.1.2 details a scenario + where a UAC sends a CANCEL at the same time that a UAS sends a + 200 OK for the INVITE that the UAC is canceling. When this + occurs, it is the role of the UAC to immediately send a BYE to + terminate the call. This scenario was reproducible by have a + Digium phone with two lines place a call to a second phone that + forwarded the call to the second line on the original phone. The + Digium phone, upon realizing that it was connecting to itself, + would attempt to cancel the call. The timing of this happened to + trigger the aforementioned race condition about 80% of the time. + Asterisk was not doing its job of sending a BYE when receiving a + 200 OK on a cancelled INVITE. The result was that the ast_channel + structure was destroyed but the underlying SIP session, as well + as the PJSIP inv_session and dialog, were still alive. Attempting + to perform an action such as a transfer, once in this state, + would result in Asterisk crashing. The circumstances are now + detected properly and the session is ended as recommended in RFC + 5407. (closes issue AST-1209) reported by John Bigelow ........ + Merged revisions 397945 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 21:37 +0000 [r397947] Kevin Harwell <kharwell@digium.com> + + * main/file.c, main/app.c, main/config_options.c, main/cel.c, + main/asterisk.c, main/cdr.c, main/manager.c, /, + main/stasis_config.c: Memory leaks fix (closes ASTERISK-22376) + Reported by: John Hardin Patches: memleak.patch uploaded by + jhardin (license 6512) memleak2.patch uploaded by jhardin + (license 6512) ........ Merged revisions 397946 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 20:22 +0000 [r397939] Matthew Jordan <mjordan@digium.com> + + * configs/safe_asterisk.conf.sample (removed), /, CHANGES, + contrib/scripts/safe_asterisk, Makefile: Revert r394939 due to + (numerous) objections The patch from ASTERISK-21965 was committed + perhaps a bit too hastily. Walter and Tzafrir have pointed out + numerous issues with the approach and have propsed an alternative + in r/2757. Since it's not a time critical issue and is not worth + holding up the release of 12 for it, I've gone ahead and reverted + r394939 from 12/trunk and re-opened ASTERISK-21965. ........ + Merged revisions 397938 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 16:21 +0000 [r397932] David M. Lee <dlee@digium.com> + + * rest-api-templates/make_ari_stubs.py, /, + rest-api-templates/api.wiki.mustache, + rest-api-templates/asterisk_processor.py: Account for {} in + Swagger notes ........ Merged revisions 397927 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 16:05 +0000 [r397925] Matthew Jordan <mjordan@digium.com> + + * Makefile, /: Recursively search for '.c' files when making + documentation with 'make full' Without this, documentation + defined in sub-folders is ignored. Since having properly + generated documentation is especially important in Asterisk 12 - + not having it can cause a module to not load - 'make full' needs + to look in all .c files. ........ Merged revisions 397924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 15:43 +0000 [r397923] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c, main/cel.c, main/stasis_bridges.c: Multiple + revisions 397921-397922 ........ r397921 | mmichelson | + 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines Resolve + assumptions that bridge snapshots would be non-NULL for transfer + stasis events. Attempting to transfer an unbridged call would + result in crashes in either CEL code or in the conversion to AMI + messages. ........ r397922 | mmichelson | 2013-08-29 10:42:29 + -0500 (Thu, 29 Aug 2013) | 3 lines Remove extra debug message. + ........ Merged revisions 397921-397922 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 12:30 +0000 [r397912] Matthew Jordan <mjordan@digium.com> + + * contrib/ast-db-manage/config, + contrib/ast-db-manage/config/script.py.mako, + contrib/ast-db-manage/voicemail.ini.sample, + contrib/ast-db-manage/voicemail/env.py, + contrib/ast-db-manage/voicemail, + contrib/ast-db-manage/voicemail/script.py.mako, + contrib/ast-db-manage/README.md, + contrib/ast-db-manage/config/versions, + contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py, + contrib/ast-db-manage (added), + contrib/ast-db-manage/voicemail/versions, /, + contrib/ast-db-manage/config.ini.sample, + contrib/ast-db-manage/config/env.py, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py: + Actually *add* the database schema management utilities In + r397874, the scripts were removed... but not replaced. Thanks to + Michael Young for noticing this! ........ Merged revisions 397911 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 23:15 +0000 [r397886-397903] Richard Mudgett <rmudgett@digium.com> + + * main/cdr.c, /, funcs/func_cdr.c, main/stdtime/localtime.c: Fix + some uninitialized buffers for CDR handling valgrind found. * + Made ast_strftime_locale() ensure that the output buffer is + initialized. The std library strftime() returns 0 and does not + touch the buffer if it has an error. However, the function can + also return 0 without an error. (closes issue ASTERISK-22412) + Reported by: rmudgett ........ Merged revisions 397902 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Fixed problems with ast_cdr_serialize_variables(). + * Fixed return value of ast_cdr_serialize_variables() on error. + It needs to return 0 indicating no CDR variables found. * Made + ast_cdr_serialize_variables() check the return value of + cdr_object_format_property() and assert if nonzero. A member of + the cdr_readonly_vars[] was not handled. * Removed unused + elements from cdr_readonly_vars[]: total_duration, total_billsec, + first_start, and first_answer. ........ Merged revisions 397900 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Made the on/off in CLI "cdr set debug [on|off]" + case insensitive. ........ Merged revisions 397898 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Make CDR variable name chandling consistently case + insensitive. ........ Merged revisions 397896 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/cdr.c: Make CDR code deal with channel names case + insensitively. ........ Merged revisions 397894 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, funcs/func_cdr.c, main/cdr.c: Some CDR code optimization. + ........ Merged revisions 397892 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, funcs/func_cdr.c: Whitespace and curly braces. ........ Merged + revisions 397885 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 21:09 +0000 [r397877] Mark Michelson <mmichelson@digium.com> + + * /, res/res_pjsip_refer.c: Improve detection of answer on SIP + blind transfer. A problem encountered during testing was that + res_pjsip_refer would not ever send a NOTIFY with a 200 OK + sipfrag. This is because the framehook that was supposed to send + the NOTIFY would never be told that an answer had occurred. This + happened for two reasons: 1) The transferee channel on which the + framehook was on was already up. 2) Answers are rarely if ever + written to channels. Rather, the ast_answer() or ast_raw_answer() + function is used to answer channels. Thanks to a suggestion by + Matt Jordan, the best way to detect that the call had been + answered was to find out when the transferee channel joined a + bridge. With stasis this is an easy task. So now, in addition to + the framehook logic, there is a stasis subscription used to + determine when the transferee has entered a bridge. Once it has + entered, an appropriate NOTIFY is sent. ........ Merged revisions + 397876 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 20:55 +0000 [r397872-397875] Matthew Jordan <mjordan@digium.com> + + * contrib/realtime/mysql/queue_log.sql, + contrib/realtime/mysql/voicemail.sql, + contrib/realtime/mysql/sippeers.sql, /, + contrib/realtime/mysql/iaxfriends.sql, + contrib/realtime/mysql/meetme.sql, + contrib/realtime/mysql/voicemail_messages.sql, + contrib/realtime/postgresql/realtime.sql, + contrib/realtime/mysql/voicemail_data.sql, CHANGES, + contrib/realtime/mysql/musiconhold.sql: Add database schema + management using Alembic This patch replaces contrib/realtime/ + with a new setup for managing the database schema required for + database integration with Asterisk. In addition to initializing a + database with the proper schema, alembic can do a database + migration to assist with upgrading Asterisk in the future. + Hopefully this helps make setting up and operating Asterisk with + a database easier. With this the schema only needs to be + maintained in one place instead of once per database. The schemas + I have added here have a bit of improvement over the examples + that were there before (some added consistency and added some + missing indexes). Managing the schema in one place here also + applies to all databases supported by SQLAlchemy. See + contrib/ast-db-manage/README.md for more details. Review: + https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant + (license 6300) ........ Merged revisions 397874 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * CHANGES, /: Update CHANGES file for Asterisk 12 This updates the + Asterisk 12 CHANGES file with the things that were missed during + the development cycle. Review: + https://reviewboard.asterisk.org/r/2795/ ........ Merged + revisions 397870 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 16:13 +0000 [r397857-397860] Richard Mudgett <rmudgett@digium.com> + + * /, main/pbx.c: pbx.c: Make ast_str_substitute_variables_full() + not mask variables. ........ Merged revisions 397859 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/chanvars.c: ast_free() is null tollerant. + + * include/asterisk/threadstorage.h, /: Match use of ast_free() with + ast_calloc() and add some curly braces. ........ Merged revisions + 397856 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 15:43 +0000 [r397855] Mark Michelson <mmichelson@digium.com> + + * res/res_pjsip/pjsip_distributor.c, /: Fix dialog matching in the + SIP distributor. Dialog matching is performed in the distributor + for the sole purpose of retrieving an associated serializer so + the request may be serialized. This patch fixes two problems. + First, incoming CANCEL requests that had no to-tag (which really + should be *all* CANCEL requests) would not match with a dialog. + An earlier bug fix to deal with early CANCEL requests would + result in the CANCEL being replied to with a 481. The fix for + this is to find the matching INVITE transaction and get the + dialog from that transaction. Second, no SIP responses were + matching dialogs. This is because we were inverting the tags that + we were passing into PJSIP's dialog finding function. This logic + has been corrected by setting local and remote tag variables + based on whether the incoming message is a request or response. + ........ Merged revisions 397854 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 19:19 +0000 [r397820] David M. Lee <dlee@digium.com> + + * rest-api-templates/param_parsing.mustache, res/res_ari_bridges.c, + /, res/stasis/app.c, res/res_ari_events.c, + res/res_ari_asterisk.c, + rest-api-templates/res_ari_resource.c.mustache, res/stasis/app.h, + res/res_stasis.c, main/stasis_bridges.c: ARI: WebSocket event + cleanup Stasis events (which get distributed over the ARI + WebSocket) are created by subscribing to the channel_all_cached + and bridge_all_cached topics, filtering out events for + channels/bridges currently subscribed to. There are two issues + with that. First was a race condition, where messages in-flight + to the master subscribe-to-all-things topic would get sent out, + even though the events happened before the channel was put into + Stasis. Secondly, as the number of channels and bridges grow in + the system, the work spent filtering messages becomes excessive. + Since r395954, individual channels and bridges have caching + topics, and can be subscribed to individually. This patch takes + advantage, so that channels and bridges are subscribed to on + demand, instead of filtering the global topics. The one case + where filtering is still required is handling BridgeMerge + messages, which are published directly to the bridge_all topic. + Other than the change to how subscriptions work, this patch + mostly just moves code around. Most of the work generating JSON + objects from messages was moved to .to_json handlers on the + message types. The callback functions handling app subscriptions + were moved from res_stasis (b/c they were global to the model) to + stasis/app.c (b/c they are local to the app now). (closes issue + ASTERISK-21969) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2754/ ........ Merged + revisions 397816 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 18:52 +0000 [r397811] Richard Mudgett <rmudgett@digium.com> + + * /, main/astmm.c: Made MALLOC_DEBUG less CPU intensive by default. + Storing a backtrace for each allocation in anticipation of a + memory management problem is very CPU intensive. * Added the CLI + "memory backtrace {on|off}" command to request that the backtrace + be gathered only on request. The backtrace is off by default. + (issue ASTERISK-22221) Reported by: Matt Jordan ........ Merged + revisions 397809 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 18:10 +0000 [r397753-397760] Matthew Jordan <mjordan@digium.com> + + * /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid + SDP If the SIP channel driver processes an invalid SDP that + defines media descriptions before connection information, it may + attempt to reference the socket address information even though + that information has not yet been set. This will cause a crash. + This patch adds checks when handling the various media + descriptions that ensures the media descriptions are handled only + if we have connection information suitable for that media. Thanks + to Walter Doekes, OSSO B.V., for reporting, testing, and + providing the solution to this problem. (closes issue + ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches: + issueA22007_sdp_without_c_death.patch uploaded by wdoekes + (License 5674) ........ Merged revisions 397756 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 397757 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged + revisions 397758 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397759 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK + on dialog that has no channel A remote exploitable crash + vulnerability exists in the SIP channel driver if an ACK with SDP + is received after the channel has been terminated. The handling + code incorrectly assumed that the channel would always be + present. This patch adds a check such that the SDP will only be + parsed and applied if Asterisk has a channel present that is + associated with the dialog. Note that the patch being applied was + modified only slightly from the patch provided by Walter Doekes + of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin + Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches: + issueA21064_fix.patch uploaded by wdoekes (License 5674) ........ + Merged revisions 397710 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 397711 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged + revisions 397712 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 16:51 +0000 [r397746] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c, + channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, channels/chan_motif.c: Fix uninitialized + value in struct ast_control_pvt_cause_code usage. ........ Merged + revisions 397744 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397745 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-26 23:48 +0000 [r397691] Matthew Jordan <mjordan@digium.com> + + * /, main/bridge_channel.c: Better handle clearing the OUTGOING + flag when a channel leaves a bridge When a channel with the + OUTGOING flag leaves a bridge, and it will survive being pulled + from the bridge (either because it will execute dialplan, go into + another bridge, or live in a friendly autoloop), we have to clear + the OUTGOING flag. This is the signal to the CDR engine that this + channel is no longer a second class citizen, i.e., it is not + "dialed". The soft hangup flags are only half the picture. If a + channel is being moved from one bridge to another, the soft + hangup flags aren't set; however, the state of the bridge_channel + will not be hung up. Since the channel does not have one of the + two hang up states, that implies that the channel is still + technically alive. This patch modifies the check so that it + checks both the soft hangup flags as well as the bridge_channel + state. If either suggests that the channel is going to persist, + we clear the OUTGOING flag. ........ Merged revisions 397690 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-26 21:32 +0000 [r397674] David M. Lee <dlee@digium.com> + + * /, main/bucket.c: Fixed bucket.c for systems where tv_usec is not + an unsigned long. ........ Merged revisions 397673 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-26 16:25 +0000 [r397644-397651] Richard Mudgett <rmudgett@digium.com> + + * /, include/asterisk/bridge_channel.h, main/bridge_channel.c: + bridging: Fix a livelock with local channel optimization. Use a + better means of waking up the bridge channel thread. ........ + Merged revisions 397650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/Makefile, /: chan_dahdi: Add some missing build cleanup. + ........ Merged revisions 397643 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-25 18:12 +0000 [r397622-397631] Matthew Jordan <mjordan@digium.com> + + * tests/test_bucket.c, /: Fix bucket unit tests After the review + for buckets was completed (r2715), the handling of names in the + bucket core was deferred to the wizards. As such, the bucket unit + tests cannot expect that passing a URI with a scheme specified + but no actual resource name will automatically fail. The tests + have been updated to not make this check. ........ Merged + revisions 397630 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/config_options.h, /, main/config_options.c, + tests/test_config.c: Fix the config_options_test The config + options test requires the entire configuration item to be + transparent from the documentation system. So we let it do that + too. As an aside, please do not use this power for evil. + Documentation is your friend, and you really should document your + configurations. Hiding your module's configuration information + from the system attempting to enforce some sanity in the universe + is something only a Bond villain would contemplate. ........ + Merged revisions 397628 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c: Add rtpengine + configuration parameter The rtpengine configuration parameter was + documented in the XML documentation, but it was not actually + registered with the sorcery object. This adds the parameter with + a default of "asterisk", such that res_rtp_asterisk is chosen as + the default RTP implementation. (closes issue ASTERISK-22380) + Reported by: Rusty Newton Tested by: Rusty Newton ........ Merged + revisions 397621 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-23 22:40 +0000 [r397615] Matthew Jordan <mjordan@digium.com> + + * /: Set new merge properties on 12 + +2013-08-23 22:20 +0000 [r397613] Joshua Colp <jcolp@digium.com> + + * main/bucket.c: Fix building of trunk. Note: This is why I commit + on the weekend. + |