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	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
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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: | ||||
|   | ||||
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