[asterisk-commits] jpeeler: branch 1.6.2 r297073 - in /branches/1.6.2: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: jpeeler
Date: Wed Dec  1 11:52:46 2010
New Revision: 297073

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=297073
Log:
Merged revisions 297072 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines
  
  Fix not stopping MOH when transfered local channel queue member is answered.
  
  The problem here is only present when local channels are used with the MOH
  passthru option as well as no optimization (/nm). I will describe the slightly
  bizarre scenario that was used to test, where phones B and C are queue members:
  
  Phone A dials into a queue with two members using local channels and the above
  options. Phone B answers. Phone A blind transfers phone B into the same queue.
  Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH.
  
  In this scenario, the unhold frame that should have gotten to phone B never
  arrived due to the masquerade from the blind transfer. This is usually fine
  since app_queue manages the starting and stopping of MOH. However, with the
  passthrough option enabled when app_queue attempts to stop MOH it tries to do
  so on the local channel rather than the real channel. The easiest solution
  was to just make sure to send an unhold frame during the transfer since it
  wouldn't make sense to have MOH playing after a transfer anyway. This only
  modifies SIP transfers, but the other transfers did not seem to be a problem.
  If DTMF based transfers were a problem it might be okay to add ast_moh_stop
  to finishup, but I didn't want to have to add that unless required.
  
  ABE-2624
........

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

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=297073&r1=297072&r2=297073
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Wed Dec  1 11:52:46 2010
@@ -21048,6 +21048,9 @@
 
 	/* For blind transfers, move the call to the new extensions. For attended transfers on multiple
 	   servers - generate an INVITE with Replaces. Either way, let the dial plan decided  */
+	/* indicate before masquerade so the indication actually makes it to the real channel
+	   when using local channels with MOH passthru */
+	ast_indicate(current.chan2, AST_CONTROL_UNHOLD);
 	res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1);
 
 	if (!res) {