asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	153095dd91	res_pjsip_rfc3326: Prefer Q.850 cause code over SIP. Resolves: #116	2 years ago
Igor Goncharovsky	410150235a	res_pjsip_rfc3326: Add SIP causes support for RFC3326 Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	3 years ago
Josh Soref	9ae9893c63	res: Spelling fixes Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda	4 years ago
Alexander Traud	008f46bf1e	res_pjsip: Sync load- and build-time deps. MODULEINFO is checked while buidling/linking the module. AST_MODULE_INFO is checked while loading/running the module. ASTERISK-28838 Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042	6 years ago
George Joseph	8f42447c68	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	7 years ago
Ross Beer	92158b7f37	res_pjsip_rfc3326.c: Account for more than one 'Reason' header ASTERISK-27741 Change-Id: I0aa59a54735c6d20b95c54db1bd095dbf93e7adf	8 years ago
Ross Beer	88cef40f6e	res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints ASTERISK-27554 Change-Id: If61c7faab7d2fa1031c056ed6268fe928e2391cf	8 years ago
Corey Farrell	527cf5a570	Remove redundant module checks and references. This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e	8 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	8 years ago
Richard Mudgett	436023a322	res_pjsip_rfc3326.c: Fix crash when channel goes away. If an authenticated incoming caller does not respond to our 200 OK INVITE response with an ACK then PJSIP will hangup the call. Unfortunately, there is a chance that the session's channel will go away between one use of the channel pointer and another when building the BYE request because the BYE is being built by the monitor thread and not the call's serializer thread. * Added a check to ensure that the thread trying to add the Reason header is the call's serializer thread. This ensures that the channel will not go away on us. Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89	10 years ago
Rodrigo Ramírez Norambuena	eec010829a	AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723	11 years ago
Kinsey Moore	86a4ce4957	PJSIP: Enforce module load dependencies This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "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	11 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

14 Commits (6fb811a5905a54a07a901b25f8f3af0ad9d3f7c9)