diff options
| author | Joshua Colp <jcolp@digium.com> | 2014-12-12 13:06:24 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2014-12-12 13:06:24 +0000 |
| commit | 74d43977cf1ec43b9d5d8f84cc64592fdbd11115 (patch) | |
| tree | e49497e449926fdbb08abb648437bd40419c94da /res/res_pjsip_session.exports.in | |
| parent | 8d384f38254142b6b41d335deebaf03fb8a8038b (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/
........
Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_session.exports.in')
| -rw-r--r-- | res/res_pjsip_session.exports.in | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/res/res_pjsip_session.exports.in b/res/res_pjsip_session.exports.in index 07f26b7e9..5a21b1d2f 100644 --- a/res/res_pjsip_session.exports.in +++ b/res/res_pjsip_session.exports.in @@ -1,5 +1,6 @@ { global: + LINKER_SYMBOL_PREFIXast_sip_session_terminate; LINKER_SYMBOL_PREFIXast_sip_session_defer_termination; LINKER_SYMBOL_PREFIXast_sip_session_register_sdp_handler; LINKER_SYMBOL_PREFIXast_sip_session_unregister_sdp_handler; |