asterisk

Commit Graph

Author	SHA1	Message	Date
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
George Joseph	747beb1ed1	modules: change module LOAD_FAILUREs to LOAD_DECLINES In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	9 years ago
Joshua Colp	8140d7a8ef	Merge "res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason."	10 years ago
Richard Mudgett	8c8ef4efb0	res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd	10 years ago
Richard Mudgett	75ec137e91	res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. * Fix double unref of other_party channel in off nominal path. * This is unlikely to be a real problem. However, for safety, in handle_incoming_request() keep the datastore ref with the other_party channel ref until we are finished with the other_party channel. Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821	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
Richard Mudgett	09bfe4b208	res_pjsip_refer: Fix crash from a REFER and BYE collision. Analyzing a one-off crash on a busy system showed that processing a REFER request had a NULL session channel pointer. The only way I can think of that could cause this is if an outgoing BYE transaction overlapped the incoming REFER transaction in a collision. Asterisk sends a BYE while the phone sends a REFER to complete an attended transfer. * Made check the session channel pointer before processing an incoming REFER request in res_pjsip_refer. * Fixed similar crash potential for res_pjsip supplement incoming request processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER messages. * Made res_pjsip_messaging respond to a message body too large with a 413 instead of ignoring it. ASTERISK-24700 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4417/ ........ Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	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
Kevin Harwell	eee4313fe8	res_pjsip_send_to_voicemail: transferring to voicemail for digium phones Added the ability for transferring directly to voicemail on digium phones. Added a new module that checks for the presence of a custom header and/or diversion header within a sip REFER. If either is found and they specify a sending to voicemail action then variables are added to the channel allowing the user access to them in the dialplan. Dialplan can then be written that branches based upon these values allowing, for instace, for a single number to be used for dialing and/or accessing voicemail directly. Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip channels through (checked to make sure it has the correct channel type before proceeding). Review: https://reviewboard.asterisk.org/r/3245/ ........ Merged revisions 408880 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

11 Commits (03e08d5c33c60b3e099feb322161bec85c157413)