I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problem is that Request-URI isn't populated correctly. I'm using Asterisk 13.
Thanks, Nitesh On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <[email protected]> wrote: > Nitesh Bansal wrote: > >> Hello, >> >> I want to use Asterisk to use Kamailio as an outbound proxy for routing >> calls to remote SIP end points, one option could be to use a default >> peer, but in my case, my outbound proxy can change >> based on the remote end point, so this option doesn't work. >> And another problem is that I don't know how to configure Asterisk to >> prepare the Request-URI >> based on the remote end point and not based on the outbound proxy address? >> >> What is the best way to do it? >> > > You'll have to be specific with the channel driver in use. Speaking from > chan_pjsip it does not have a mechanism to set the outbound proxy on a > per-call basis, it's strictly controlled by the endpoint. You'd need > multiple or construct endpoints dynamically (for example using the ARI push > configuration). As for not rewriting the request URI you need to use loose > routing by specifying ;lr in the outbound proxy URI. > > Example: > > sip:example.com;lr > > If used in a configuration file: > > sip:example.com\;lr > > The '\' is so the configuration parser does not treat it as a comment. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > 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
