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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2012-01-16 17:04:44 +00:00
parent 2263f9d10e
commit 91e4333579
1 changed files with 4 additions and 0 deletions
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4
main/rtp_engine.c
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@@ -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: