Make sure we stop session timers as soon as we start hanging up an active call.

May fix issue 12919.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127791 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -5111,6 +5111,10 @@ static int sip_hangup(struct ast_channel *ast)
 				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 = "";