[asterisk-commits] file: branch 1.8 r350975 - /branches/1.8/main/rtp_engine.c

[SVN commits to the Asterisk project]

Author: file
Date: Mon Jan 16 11:04:44 2012
New Revision: 350975

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=350975
Log:
Add missing code to set direct RTP setup information during dialing.

Modified:
    branches/1.8/main/rtp_engine.c

Modified: branches/1.8/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/main/rtp_engine.c?view=diff&rev=350975&r1=350974&r2=350975
==============================================================================
--- branches/1.8/main/rtp_engine.c (original)
+++ branches/1.8/main/rtp_engine.c Mon Jan 16 11:04:44 2012
@@ -1447,6 +1447,10 @@
 		ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
 	}
 
+        if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) {
+                ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+        }
+
 	res = 0;
 
 done: