[svn-commits] mmichelson: branch 1.4 r208587 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: mmichelson
Date: Fri Jul 24 13:26:50 2009
New Revision: 208587

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=208587
Log:
Only send a BYE when hanging up a channel that is up.

For cases where Asterisk sends an INVITE and receives a non 2XX final
response, Asterisk would follow the INVITE transaction by immediately
sending a BYE, which was unnecessary.

(closes issue #14575)
Reported by: chris-mac


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=208587&r1=208586&r2=208587
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Jul 24 13:26:50 2009
@@ -3726,7 +3726,9 @@
 				if (p->vrtp)
 					videoqos = ast_rtp_get_quality(p->vrtp, NULL);
 				/* Send a hangup */
-				transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+				if (oldowner->_state == AST_STATE_UP) {
+					transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+				}
 
 				/* Get RTCP quality before end of call */
 				if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {