[asterisk-commits] mmichelson: branch 1.6.2 r189105 - in /branches/1.6.2: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Fri Apr 17 15:21:42 2009
New Revision: 189105

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=189105
Log:
Merged revisions 189097 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
  r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines
  
  Prevent a crash when SIP blonde transferring an unbridged call.
  
  If one attempts to use the attended transfer button on a SIP phone
  to transfer an unbridged call (such as a call to an IVR) but hangs
  up while the target of the transfer is still ringing, we need to not
  crash.
  
  The problem was that ast_hangup was called from outside the channel
  thread.
  
  AST-211
........

Modified:
    branches/1.6.2/   (props changed)
    branches/1.6.2/channels/chan_sip.c

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=189105&r1=189104&r2=189105
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Fri Apr 17 15:21:42 2009
@@ -19426,11 +19426,7 @@
 		append_history(transferer, "Xfer", "Refer failed");
 		if (targetcall_pvt->owner)
 			ast_channel_unlock(targetcall_pvt->owner);
-		/* Right now, we have to hangup, sorry. Bridge is destroyed */
-		if (res != -2)
-			ast_hangup(transferer->owner);
-		else
-			ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
+		ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
 	} else {
 		/* Transfer succeeded! */
 		const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");