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authorJoshua Colp <jcolp@digium.com>2014-12-12 13:05:15 +0000
committerJoshua Colp <jcolp@digium.com>2014-12-12 13:05:15 +0000
commit0c9fbb449f272b826e022499dddda39bc12dda7e (patch)
tree395256dbbf4ad0c321fb8fd91bf77ab402342917 /UPGRADE-10.txt
parent61fe4f10d230d09b5246c97675637810ffed2fa2 (diff)
res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.
Given the scenario where a PJSIP channel is in a native RTP bridge with direct media and the channel is then hung up the code will currently re-INVITE the channel back to Asterisk and send a BYE at the same time. Many SIP implementations dislike this greatly. This change makes it so that if a re-INVITE transaction is in progress the BYE is queued to occur after the completion of the transaction (be it through normal means or a timeout). Review: https://reviewboard.asterisk.org/r/4248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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