[asterisk-commits] file: trunk r350977 - in /trunk: ./ main/rtp_engine.c

[SVN commits to the Asterisk project]

Author: file
Date: Mon Jan 16 11:07:13 2012
New Revision: 350977

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=350977
Log:
Add missing code to set direct RTP setup information during dialing.
........

Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 350976 from http://svn.asterisk.org/svn/asterisk/branches/10

Modified:
    trunk/   (props changed)
    trunk/main/rtp_engine.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.

Modified: trunk/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/rtp_engine.c?view=diff&rev=350977&r1=350976&r2=350977
==============================================================================
--- trunk/main/rtp_engine.c (original)
+++ trunk/main/rtp_engine.c Mon Jan 16 11:07:13 2012
@@ -1504,6 +1504,10 @@
 		ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
 	}
 
+        if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
+                ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
+        }
+
 	res = 0;
 
 done: