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2014-07-29res_pjsip_session: Fix race condition where redirecting information may not ↵Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25Add module support level to ast_module_info structure. Print it in CLI ↵Mark Michelson
"module show" . ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20media formats: re-architect handling of media for performance improvementsMatthew Jordan
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07CEL: Fix incorrect/missing extra field informationKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04res_pjsip_session: Add debug statement for session refreshesMatthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28res_pjsip_session: Fix leaked video RTP ports.Richard Mudgett
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-19Undo r414123Matthew Jordan
The Test Suite caught a few problems, undoing until those are resolved git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-18bridge_native_rtp/bridge_channel: Fix direct media issues due to frame hookMatthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-30chan_pjsip: Add support for picking up calls in the configured pickup group.Joshua Colp
AST-1363 Review: https://reviewboard.asterisk.org/r/3478/ ........ Merged revisions 413117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-15Remove unused RAII_VAR() declarations.Richard Mudgett
* 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01res_pjsip_session: Set options (100rel, timers) on incoming sessions.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-26res_pjsip_session: Be less strict with core requested outgoing capabilities.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-15PJSIP: Add Path header supportKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-09PJSIP: Add unhold on reinvite without SDP behaviorJonathan Rose
Review: https://reviewboard.asterisk.org/r/3106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-09astobj2: Correct ao2_iterator opacity violationsKinsey Moore
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-31chan_pjsip: Handle hanging up before calling.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-19res_pjsip_session: Fix SDP negotiation when resending an INVITE with ↵Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-01res_pjsip_session: Apply fromuser and fromdomain to all requests as documented.Joshua Colp
........ Merged revisions 403271 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-28res_pjsip_session: Add configurable behavior for redirects.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-21res_pjsip_session: Fix memory leak of direct media format capabilitiesMatthew Jordan
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-13Fix a race condition in res_pjsip_session with rapidly terminating the session.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04Replace the connection address at the SDP level if altering the SDP with the ↵Joshua Colp
external media address. ........ Merged revisions 400510 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Cache string values of formats on ast_format_cap() to save processing.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Fix a random one way audio issue in PJSIP.Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Retrieve and store the hostname only once so multiple threads do not ↵Joshua Colp
potentially initialize it at the same time. ........ Merged revisions 400245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-27res_pjsip: crash when using localnet and external_signaling_address optionsKevin Harwell
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-23Fix crash in res_pjsip on load if error occurs, and prevent unloading of ↵Joshua Colp
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Create more accurate Contact headers for dialogs when we are the UAS.Mark Michelson
(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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-29Fix a race condition where a canceled call was answered.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01Answer with multiple codecs if the underlying pjproject supports it.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-30The large GULP->PJSIP renaming effort.Mark Michelson
The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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