asterisk

Commit Graph

Author	SHA1	Message	Date
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

16 Commits (72c282cc66d7e12cd17d0eee90e066d9ebfb1243)