[asterisk-commits] mmichelson: branch 1.6.0 r208589 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Fri Jul 24 13:31:35 2009
New Revision: 208589

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=208589
Log:
Merged revisions 208588 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines
  
  Merged revisions 208587 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines
    
    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/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=208589&r1=208588&r2=208589
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Fri Jul 24 13:31:35 2009
@@ -5103,7 +5103,9 @@
 				if (p->trtp)
 					textqos = ast_rtp_get_quality(p->trtp, 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 (p->do_history) {