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2015-04-10chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices.Richard Mudgett
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/ ........ Merged revisions 434671 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10chan_sip: make progressinband default to noKevin Harwell
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/ ........ Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10res_pjsip: Add an 'auto' option for DTMF ModeMatthew Jordan
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) ........ Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10res_pjsip_config_wizard: Cleanup load unloadGeorge Joseph
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/ ........ Merged revisions 434619 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10bridge_softmix.c,channel.c: Minor code simplification and cleanup.Richard Mudgett
* 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/ ........ Merged revisions 434617 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10translate.c: Only select audio codecs to determine the best translation choice.Richard Mudgett
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 ........ Merged revisions 434615 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10res/ari: Fix model validation for ChannelHold eventMatthew Jordan
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. ........ Merged revisions 434597 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10dns: Fix build when TEST_FRAMEWORK is not defined.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10channels/chan_iax2: Improve POKE expiration time calculation for lossy networksMatthew Jordan
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 ........ Merged revisions 434565 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-10channels/chan_iax2: Add a configuration parameter for call token expirationMatthew Jordan
This patch adds a new configuration parameter, 'calltokenexpiration', that controls how long before an authentication call token is expired. The default maintains the RFC specified 10 seconds. Setting it to a higher value may be useful in lossy networks. Review: https://reviewboard.asterisk.org/r/4588 ASTERISK-24939 #close Reported by: Y Ateya patches: ctoken_configuration.diff submitted by Y Ateya (License 6693) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09res_pjsip_phoneprov_provider: Fix reference leak on unloadGeorge Joseph
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 ........ Merged revisions 434545 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09loader/main: Don't set ast_fully_booted until deferred reloads are processedGeorge Joseph
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/ ........ Merged revisions 434544 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09res_pjsip: add CLI command to show global and system configurationKevin Harwell
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/ ........ Merged revisions 434527 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09chan_iax2.c: Fix ref leak in iax2_request().Richard Mudgett
* Increased warning message format capability string buffer size in iax2_request(). Review: https://reviewboard.asterisk.org/r/4601/ ........ Merged revisions 434510 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09bridge_native_rtp.c: Defer allocation and check if it fails in ↵Richard Mudgett
native_rtp_bridge_compatible(). Review: https://reviewboard.asterisk.org/r/4601/ ........ Merged revisions 434508 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requestsMatthew Jordan
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) ........ Merged revisions 434506 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09Reduce duplication of common DNS code.Mark Michelson
The NAPTR and SRV branches were worked on independently and resulted in some code being duplicated in each. Since both have been merged into trunk now, this patch reduces the duplication by factoring out common code into its own source files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09clang compiler warnings: Fix autological comparisonsMatthew Jordan
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 ........ Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09apps/app_queue: Prevent possible crash when evaluating queue penalty rulesMatthew Jordan
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 ........ Merged revisions 434449 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08res_pjsip_t38: Fix FAX failures when using PJSIP with authenticationJonathan Rose
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/ ........ Merged revisions 434425 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08Bridging: Eliminate the unnecessary make channel compatible with bridge ↵Richard Mudgett
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/ ........ Merged revisions 434424 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08Security/tcptls: MitM Attack potential from certificate with NULL byte in CN.Jonathan Rose
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 ........ Merged revisions 434384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear().Richard Mudgett
........ Merged revisions 434357 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08chan_iax2: Fix compilation issue due to funky mergeMatthew Jordan
Don't mix declarations and code! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno]Matthew Jordan
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 ........ Merged revisions 434292 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabledMatthew Jordan
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 ........ Merged revisions 434289 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08clang compiler warnings: Fix pointer-bool-converesion warningsMatthew Jordan
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 ........ Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-08cel_pgsl: Add support for GMT timestampsMatthew Jordan
This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps to be logged in GMT. Review: https://reviewboard.asterisk.org/r/4571/ ASTERISK-23186 #close Reported by: Rodrigo Ramirez Norambuena patches: cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07pjsip: resolve compatibility problem with ast_sip_sessionScott Griepentrog
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 ........ Merged revisions 434261 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07bridge.c: Hangup attended transfer target after it has been swapped outKevin Harwell
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/ ........ Merged revisions 434240 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07Do not queue message requests that we do not respond to.Mark Michelson
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. ........ Merged revisions 434218 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07ARI: Add the ability to intercept hold and raise an eventMatthew Jordan
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 ........ Merged revisions 434216 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07clang compiler warnings: Fix sometimes-initialized warning in func_mathMatthew Jordan
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 ........ Merged revisions 434191 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07clang compiler warnings: Fix non-literal-null-conversion warningsMatthew Jordan
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 ........ Merged revisions 434188 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06Uncomment test case.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06Add missing DNS NAPTR test file.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06res_pjsip: config option 'timers' can't be set to 'no'Kevin Harwell
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/ ........ Merged revisions 434131 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06build: Fixes for gcc 5 compilationGeorge Joseph
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 ........ Merged revisions 434114 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06clang compiler warnings: Remove large chunks of unused code from extconfMatthew Jordan
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 ........ Merged revisions 434097 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06clang compiler warnings: Fix sometimes-uninitialized warning in pbx_configMatthew Jordan
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 ........ Merged revisions 434091 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06clang compiler warnings: Fix format specified in framehookMatthew Jordan
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 ........ Merged revisions 434088 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06Merge NAPTR support into trunk.Mark Michelson
This adds NAPTR record allocation and sorting, as well as unit tests that verify that NAPTR records are parsed and sorted correctly. Review: https://reviewboard.asterisk.org/r/4542 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06Ensure that a non-zero sample rate is returned for all formats.Mark Michelson
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 ........ Merged revisions 434046 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06res_pjsip_phoneprov_provider: Revert 433996 / 433997.Corey Farrell
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/ ........ Merged revisions 434025 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434026 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-06ParkedCall: Don't allow dialplan fallthrough after retrieving parked call.Mark Michelson
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) ........ Merged revisions 434022 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-05res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.Corey Farrell
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/ ........ Merged revisions 433996 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-03res_pjsip_messaging: Serialize outbound SIP MESSAGEsMark Michelson
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 ........ Merged revisions 433968 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-02pjsip: resolve compatibility problem with ast_sip_sessionScott Griepentrog
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/trunk@433945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-02Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS.Corey Farrell
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/ ........ Merged revisions 433923 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-01stasis: set a channel variable on websocket disconnect errorAshley Sanders
Resolve compile errors caused by r433863 by fixing the documentation xml to comply with the schema. ........ Merged revisions 433888 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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