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

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Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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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
This commit is contained in:
Joshua Colp
2012-01-16 17:07:13 +00:00
parent 382d14a214
commit 35fef9a7dc
1 changed files with 4 additions and 0 deletions
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4
main/rtp_engine.c
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@@ -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) : "<unspecified>");
}
res = 0;
done: