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path: root/res/res_pjsip_pubsub.c
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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 when the subscription_terminated callback is called for subscription ↵Mark Michelson
handlers. The previous placement would result in the resubscribe() callback called instead of the subscription_terminated() callback being called when a subscription was ended via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw an assertion. ........ Merged revisions 397955 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Handle default body types for SIP event packages in res_pjsip_pubsubMark Michelson
Prior to this change, we would reject SUBSCRIBE requests that had no Accept headers. Now event package handlers that handle the default type for the event package indicate that they do so. Therefore, if we have a handler that can handle the default type, we can allow SUBSCRIBEs for the handler's event package that have no Accept headers. (closes issue ASTERISK-22067) reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/2774 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 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