On 09/27/2010 11:02 AM, Eugene Oden wrote: > is there a trick to get asterisk (1.6.2.13) to propagate > codec-changing sip reinvites when directrtpsetup=yes? > > i'm trying to route calls to a gateway without keeping asterisk in the > rtp stream.
You are looking for a SIP proxy; Asterisk is not a SIP proxy, and no amount of configuration will convince it to act like one. > the gateway is first routing the call to a media server. when > connecting the call to the downstream carrier a different codec is > selected. > > the reinvite makes it to asterisk but asterisk isn't sending it along > to the originator so the transmit/receive codecs are mismatched > causing one-way audio. Asterisk never "sends along" re-INVITEs, because the two channels involved in an Asterisk "call" are distinct. If the codecs are mismatched between the two call legs, Asterisk will try to transcode them. The 'directrtpsetup' feature is still marked *experimental*, and that is primarily because it defeats much of Asterisk's normal behavior; in addition, there a quite a few normal, working call scenarios for which it will fail... so it's there, but if you use it, you can expect difficulties. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
