[asterisk-commits] file: branch 1.6.0 r165603 - in /branches/1.6.0: ./ main/rtp.c

[SVN commits to the Asterisk project]

Author: file
Date: Thu Dec 18 11:14:27 2008
New Revision: 165603

URL: http://svn.digium.com/view/asterisk?view=rev&rev=165603
Log:
Merged revisions 165599 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines
  
  Merged revisions 165591 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
    
    Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
    (closes issue #13545)
    Reported by: davidw
  ........
................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/main/rtp.c

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

Modified: branches/1.6.0/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/main/rtp.c?view=diff&rev=165603&r1=165602&r2=165603
==============================================================================
--- branches/1.6.0/main/rtp.c (original)
+++ branches/1.6.0/main/rtp.c Thu Dec 18 11:14:27 2008
@@ -1863,10 +1863,9 @@
 	else
 		destcodec = 0;
 	/* Ensure we have at least one matching codec */
-	if (!(srccodec & destcodec)) {
+	if (srcp && !(srccodec & destcodec)) {
 		ast_channel_unlock(c0);
-		if (c1)
-			ast_channel_unlock(c1);
+		ast_channel_unlock(c1);
 		return 0;
 	}
 	/* Consider empty media as non-existent */