[asterisk-commits] twilson: branch 1.8 r351080 - /branches/1.8/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jan 16 14:06:49 CST 2012
Author: twilson
Date: Mon Jan 16 14:06:45 2012
New Revision: 351080
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=351080
Log:
Don't prematurely stop SIP session timer
When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry.
(closes issue ASTERISK-18996)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches: session_timer_fix.diff by Terry Wilson (License #5357)
based on session_timer.patch by Thomas Arimont (License #5525)
Modified:
branches/1.8/channels/chan_sip.c
Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=351080&r1=351079&r2=351080
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Mon Jan 16 14:06:45 2012
@@ -25832,6 +25832,8 @@
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(p->owner);
sip_pvt_unlock(p);
+ } else {
+ res = 1;
}
}
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