[asterisk-commits] oej: trunk r127791 - /trunk/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Thu Jul  3 11:48:23 2008
New Revision: 127791

URL: http://svn.digium.com/view/asterisk?view=rev&rev=127791
Log:
Make sure we stop session timers as soon as we start hanging up an active call.

May fix issue 12919.


Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=127791&r1=127790&r2=127791
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Jul  3 11:48:23 2008
@@ -5111,6 +5111,10 @@
 				p->invitestate = INV_TERMINATED;
 			}
 		} else {	/* Call is in UP state, send BYE */
+			if (p->stimer->st_active == TRUE) {
+				stop_session_timer(p);
+			}
+
 			if (!p->pendinginvite) {
 				struct ast_channel *bridge = ast_bridged_channel(oldowner);
 				char *audioqos = "";