ok, thanks. i was beginning to suspect as much but was hoping to limit the number of components in our configuration.
thanks, gene On Mon, Sep 27, 2010 at 11:13 AM, Kevin P. Fleming <[email protected]> wrote: > 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 > -- _____________________________________________________________________ -- 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
