[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;
 		}
 	}
 




More information about the asterisk-commits mailing list