From ca3a020b0a716a974e4d2fb188db4d0fdfdb63fa Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Mon, 16 Jan 2012 17:06:05 +0000
Subject: [PATCH] Add missing code to set direct RTP setup information during
 dialing. ........

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@350976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 main/rtp_engine.c | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 4f28bb9dab..e002746561 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1504,6 +1504,10 @@ void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struc
 		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", c0->name, c1 ? c1->name : "<unspecified>");
+        }
+
 	res = 0;
 
 done: