[asterisk-commits] jpeeler: trunk r297076 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: jpeeler
Date: Wed Dec  1 11:53:54 2010
New Revision: 297076

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=297076
Log:
Merged revisions 297075 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r297075 | jpeeler | 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) | 37 lines
  
  Merged revisions 297073 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines
    
    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:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=297076&r1=297075&r2=297076
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Dec  1 11:53:54 2010
@@ -22424,6 +22424,9 @@
 
 	/* 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) {