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2015-01-09AMI: Make AMI actions that generate event lists consistent.Richard Mudgett
* 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09res_fax: Add T.38 negotiation timeout optionKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-08res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdownGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-08res_pjsip_outbound_registration: Fix reference leak.George Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-08res_pjsip_outbound_registration: Fix several reload issuesGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07ARI: Allow usage of ASYNCGOTO with Stasis()Kinsey Moore
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07Add the ability to continue and originate using priority labels.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07res_pjsip_exten_state: Change 'does not exist' warning to noticeGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07res_pjsip_mwi: Change "MWI Subscription failed" message from warning to noticeGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07Fix ability to perform a remote attended transfer with PJSIP.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07Fix dev-mode build on recent gccKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06res_pjsip_mwi: Change warning to noticeGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06outbound_registration: Add 'pjsip send register' and update 'send unregister'George Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06pjsip cli: Fix sorting of contacts for 'pjsip list contacts'George Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-05pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions.Joshua Colp
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-29PJSIP: Update transport method documentationKinsey Moore
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24res_pjsip: Backport missing commits for user_eq_phoneMatthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24res_pjsip_keepalive: Add runtime configurable keepalive module for ↵Matthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when ↵Matthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-23pjsip_options: Fix continued qualifies after endpoint/aor deletionGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-22res_pjsip_phoneprovi_provider: Fix reloadGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-19res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write().Richard Mudgett
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-18chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5.Richard Mudgett
ASTERISK-24337 #close Reported by: Rusty Newton ........ Merged revisions 429804 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-18res_pjsip_sdp_rtp: wrong bridge chosen when the DTMF mode is not compatibleKevin Harwell
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-18Prevent potential infinite outbound authentication loops in registration.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-18res_pjsip_config_wizard: fix unload SEGVGeorge Joseph
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-17res_pjsip_config_wizard: Change FILEUNCHANGED config_load2 flag determinationGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-17Fix printf problems with high ascii characters after r413586 (1.8).Walter Doekes
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-16res_pjsip_config_wizard: fix test breakageGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-16res_pjsip_t38: Fix T.38 failure when peer reinvites immediately.Joshua Colp
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-15res_pjsip_config_wizard: Allow streamlined config of common pjsip scenariosGeorge Joseph
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-15Activate persistent subscriptions when they are recreated.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-12res/res_agi: Make Verbose message for 'stream file' match other playbacksMatthew Jordan
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-12Fix crash for sorcery misconfigsDavid M. Lee
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-12PJSIP: Allow use of 'inactive' streams for holdKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-12Sorcery: Log when old config remains in useKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-12res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.Joshua Colp
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-12res_pjsip_session: Fix issue where a declined media stream in a re-INVITE ↵Joshua Colp
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-10res_http_websocket: Fix crash due to double freeing memory when receiving a ↵Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-10PJSIP: Fix assert on initial mass qualifyKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-09ARI/AMI: Include language in standard channel snapshot outputKevin Harwell
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-09Direct Media calls within private network sometimes get one way audioKevin Harwell
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-09res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizardKevin Harwell
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-09ari: Add support for specifying an originator channel when originating.Joshua Colp
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-09PJSIP: Stagger outbound qualifiesKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-08Fix a crash that would occur when receiving a 491 response to a reinvite.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-08Add new AMI and ARI events for connected line changes on a channel.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-08Stasis: Fix StasisStart/End order and missing eventsKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-06res/res_monitor: Reset in/out sample counts on Monitor startMatthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-02res_pjsip_refer: Fix issue where native bridge may not occur upon completion ↵Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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