[asterisk-commits] file: branch 1.4 r165591 - /branches/1.4/main/rtp.c

[SVN commits to the Asterisk project]

Author: file
Date: Thu Dec 18 11:11:42 2008
New Revision: 165591

URL: http://svn.digium.com/view/asterisk?view=rev&rev=165591
Log:
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/1.4/main/rtp.c

Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=165591&r1=165590&r2=165591
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Thu Dec 18 11:11:42 2008
@@ -1540,10 +1540,9 @@
 	else
 		destcodec = 0;
 	/* Ensure we have at least one matching codec */
-	if (!(srccodec & destcodec)) {
+	if (srcp && !(srccodec & destcodec)) {
 		ast_channel_unlock(dest);
-		if (src)
-			ast_channel_unlock(src);
+		ast_channel_unlock(src);
 		return 0;
 	}
 	/* Consider empty media as non-existant */