Add missing code to set direct RTP setup information during dialing.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350975 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index aa54388227fcf6840f3ff25ad86aad689093501c..2543a54f567007dee9dfe5b34d3763b864b9f626 100644 (file)
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1447,6 +1447,10 @@ 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, 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: