From 35fef9a7dcad2e7c7659245511a579d2066c894b Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Mon, 16 Jan 2012 17:07:13 +0000 Subject: Add missing code to set direct RTP setup information during dialing. ........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350976 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350977 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 c5540b168..0637f5338 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", ast_channel_name(c0), c1 ? ast_channel_name(c1) : ""); + } + res = 0; done: -- cgit v1.2.3