[asterisk-commits] mmichelson: branch 1.4 r110635 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Tue Mar 25 10:40:33 2008
New Revision: 110635

URL: http://svn.digium.com/view/asterisk?view=rev&rev=110635
Log:
When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=110635&r1=110634&r2=110635
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Mar 25 10:40:33 2008
@@ -14655,7 +14655,9 @@
 		update_call_counter(p, DEC_CALL_LIMIT);
 
 	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
-
+	if (p->owner)
+		ast_queue_hangup(p->owner);
+	else
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 	if (p->initreq.len > 0) {
 		transmit_response_reliable(p, "487 Request Terminated", &p->initreq);