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

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 4f28bb9dab5181dbc0d89e292068cc9fb63caf6a..e002746561bdbdc2b720bc2b0d727b95d6924f29 100644 (file)
--- 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: