28 Commits (077c4187d9789eaf585568f8178f5e3a470ab781)

Author	SHA1	Message	Date
Matthew Jordan	4f603c5da3	res_pjsip_session: Add debug statement for session refreshes ... This small patch adds a debug level 3 statement indicating how a session refresh is being sent - either as a re-INVITE or as an UPDATE - and where the session refresh is going. ........ Merged revisions 415115 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	69125a7ae2	res_pjsip_session: Fix leaked video RTP ports. ... Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak video RTP ports if the codec were not negotiated by an incoming call. * Made add_sdp_streams() associate the handler with the media stream if the handler handled the media stream. Otherwise, when the ast_sip_session_media object was destroyed it didn't know how to clean up the RTP resources. * Fixed sdp_requires_deferral() associating the handler with the media stream when deciding if the SDP processing needs to be deferred for T.38. Like the leaked video RTP ports, the T.38 handler needs to clean up allocated resources from deciding if SDP processing needs to be deffered. * Cleaned up some dead code in handle_incoming_sdp() and sdp_requires_deferral(). ASTERISK-23721 #close Reported by: cervajs Review: https://reviewboard.asterisk.org/r/3571/ ........ Merged revisions 414749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	42a1dee02d	Undo r414123 ... 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	12 years ago
Matthew Jordan	17ff4d9282	bridge_native_rtp/bridge_channel: Fix direct media issues due to frame hook ... This patch fixes issues with direct media bridges that occur after a blind transfer. These issues were caught by the (currently failing) pjsip/transfers/blind_transfer/caller_direct_media test. The test currently fails primarily for two reasons: (1) When Bob and Charlie (the transfer target and the transfer destination) enter a bridge together, the framehook remains on the transfer target channel until both channels are in the bridge. As it consumes voice frames, the initial bridge type is a simple bridge. The framehook is removed when both channels are in the bridge; however, this does not currently cause the bridging framework to re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a framehook is removed so the bridge can re-evaluate itself. (2) When a channel leaves a native RTP bridge, it may be leaving due to being hung up. Sending a re-INVITE to a channel that is about to be hung up is not nice - in fact, there's a good chance we'll send the BYE request before the channel has had a chance to send back a 200 OK. To be somewhat nicer, this patch adds a function to channel.h that allows the bridging framework to query for exactly why a channel is leaving a bridge via the channel's soft hangup flags. This allows it to only send the re-INVITE if there's a chance the channel will survive the native bridging experience. Review: https://reviewboard.asterisk.org/r/3535/ ........ Merged revisions 414122 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	56ca10c7f1	chan_pjsip: Add support for picking up calls in the configured pickup group. ... AST-1363 Review: https://reviewboard.asterisk.org/r/3478/ ........ Merged revisions 413117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	45ade68cb4	Remove unused RAII_VAR() declarations. ... * Remove unused RAII_VAR() declarations. The compiler cannot catch these because the cleanup function "references" the unused variable. Some actually allocated and released resources that were never used. * Fixed some whitespace issues in stasis_bridges.c. ........ Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	597690f363	res_pjsip_session: Set options (100rel, timers) on incoming sessions. ... This change passes options to the UAS creation function. This in turn sets up 100rel and session timer properties on the incoming session. Reported by Julian Russell on asterisk-users mailing list. ........ Merged revisions 409287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	ff455ee2aa	res_pjsip_session: Be less strict with core requested outgoing capabilities. ... The core may (depending on circumstances) request a single codec on outgoing calls. Many channel drivers ignore or treat this as a suggestion while still including configured codecs. The res_pjsip_session logic treated this as an explicit request, leaving out other configured codecs. This change makes res_pjsip_session behave like other channel driver and simply adds the requested codec to the list. (closes issue ASTERISK-23082) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3140/ ........ Merged revisions 406489 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7cbb6eab15	PJSIP: Add Path header support ... This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. Path information is stored on contacts and is enabled via Address of Record (AoRs) and Registration configuration sections. While adding path support, it became necessary to be able to add SIP supplements that handled messages outside of sessions, so a framework for handling these types of hooks was added in parallel to the already-existing session supplements and several senders of out-of-dialog requests were refactored as a result. (closes issue ASTERISK-21084) Review: https://reviewboard.asterisk.org/r/3050/ ........ Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	42b087c2df	PJSIP: Add unhold on reinvite without SDP behavior ... Review: https://reviewboard.asterisk.org/r/3106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	51901aa2ed	astobj2: Correct ao2_iterator opacity violations ... This corrects the ao2_iterator opacity violations in res_pjsip_session.c by adding a global function to get the number of elements inside the container hidden behind the iterator. (closes issue ASTERISK-23053) Review: https://reviewboard.asterisk.org/r/3111/ Reported by: Richard Mudgett ........ Merged revisions 405253 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	f720a9ac89	chan_pjsip: Handle hanging up before calling. ... Channel creation in Asterisk is broken up into two steps: requesting and calling. In some cases a channel may be requested but never called. This happens in the ChanIsAvail dialplan application for determining if something is reachable or not. The PJSIP channel driver did not take this situation into account and attempted to end a session that was never called out on. The code now checks the session state to determine if the session has been called out on and if not terminates it instead of ending it. (closes issue ASTERISK-23074) Reported by: Kilburn ........ Merged revisions 404652 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	8402cd4cd9	res_pjsip_session: Fix SDP negotiation when resending an INVITE with authentication. ... The process for resending an INVITE with authentication involves restarting the UAC session. We were incorrectly passing in that a new offer is being sent, causing the SDP negotiation to get into a (technically speaking) funky state. ........ Merged revisions 404369 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b8025e789d	res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. ... Newer versions of PJSIP have changed to using a flag for the PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a configure check to detect the presence of the flag and use it if found. ........ Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	177e7861a2	res_pjsip_session: Apply fromuser and fromdomain to all requests as documented. ... ........ 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	12 years ago
Joshua Colp	a64cd7c6bb	res_pjsip_session: Add configurable behavior for redirects. ... The action taken when a redirect occurs is now configurable on a per-endpoint basis. The redirect can either be treated as a redirect to a local extension, to a URI that is dialed through the Asterisk core, or to a URI that is dialed within PJSIP itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	92af2b2e26	res_pjsip_session: Fix memory leak of direct media format capabilities ... The direct media format capabilities are always allocated in ast_sip_session_alloc and were not freed in the session destructor. Whoops. (This being the third whoops caught by Scott and Nitesh's valgrind work for the Asterisk Test Suite. Nifty!) ........ Merged revisions 402968 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b47851264e	Fix a race condition in res_pjsip_session with rapidly terminating the session. ... The INVITE session state callback wrongly assumes that a session will always exist, but when rapidly terminating the session this assumption goes out the window. As all handler code for the INVITE session state callback requires the session it will now just exit immediately if no session exists. (closes issue ASTERISK-22668) Reported by: John Bigelow ........ Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	47da03e737	Replace the connection address at the SDP level if altering the SDP with the external media address. ... ........ Merged revisions 400510 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	ee21eee7e0	Cache string values of formats on ast_format_cap() to save processing. ... Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	424c0f2eb7	Fix a random one way audio issue in PJSIP. ... Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for the SDP to be negotiated *after* a channel was created and after it was being wait on by an application. It is only after negotiation occurs that the file descriptors for RTP are placed on the channel. Since the channel was already being waited on these file descriptors were not monitored, causing incoming media to never be read. This change wakes up any application waiting on the channel so that added file descriptors end up being monitored. (closes issue AST-1227) Reported by: John Bigelow ........ Merged revisions 400256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c33aac75e4	Retrieve and store the hostname only once so multiple threads do not potentially initialize it at the same time. ... ........ Merged revisions 400245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	d6bceb0350	res_pjsip: crash when using localnet and external_signaling_address options ... There was a collision of mod_data use on the transaction between using a nat hook and an session response callback. During state change it was assumed what was in the mod_data was nothing or the response callback. However, it was possible for it to also contain a nat hook thus resulting in a bad cast and a crash. Added the ability to store multiple data elements in mod_data via a hash table. In this instance, mod_data now stores a hash table of the two values that can be retrieved using an associated string key. (closes issue ASTERISK-22394) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2843/ ........ Merged revisions 399990 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	85d6db6cbe	Fix crash in res_pjsip on load if error occurs, and prevent unloading of res_pjsip and res_pjsip_session. ... During load time in res_pjsip if an error occurred the operation would attempt to rollback all operations done during load. This is not permitted by PJSIP as it will assert if the operation has not been done. This fix changes the code so it will only rollback what has been initialized already. Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to limitations within PJSIP itself. The library environment can only be changed to a certain extent and does not provide the ability, currently, to deinitialize certain required functionality. (closes issue ASTERISK-22474) Reported by: Corey Farrell ........ Merged revisions 399624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	9deb416397	Create more accurate Contact headers for dialogs when we are the UAS. ... (closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 ........ Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	de7ce39187	Fix a race condition where a canceled call was answered. ... RFC 5407 section 3.1.2 details a scenario where a UAC sends a CANCEL at the same time that a UAS sends a 200 OK for the INVITE that the UAC is canceling. When this occurs, it is the role of the UAC to immediately send a BYE to terminate the call. This scenario was reproducible by have a Digium phone with two lines place a call to a second phone that forwarded the call to the second line on the original phone. The Digium phone, upon realizing that it was connecting to itself, would attempt to cancel the call. The timing of this happened to trigger the aforementioned race condition about 80% of the time. Asterisk was not doing its job of sending a BYE when receiving a 200 OK on a cancelled INVITE. The result was that the ast_channel structure was destroyed but the underlying SIP session, as well as the PJSIP inv_session and dialog, were still alive. Attempting to perform an action such as a transfer, once in this state, would result in Asterisk crashing. The circumstances are now detected properly and the session is ended as recommended in RFC 5407. (closes issue AST-1209) reported by John Bigelow ........ Merged revisions 397945 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	5c13969469	Answer with multiple codecs if the underlying pjproject supports it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. ... 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	12 years ago