Nitesh Bansal
2016-Apr-13 20:11 UTC
[asterisk-users] Using Asterisk to route call via an outbound proxy
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 <jcolp at digium.com> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160413/60d0ba7e/attachment.html>
Nitesh Bansal
2016-Apr-13 20:22 UTC
[asterisk-users] Using Asterisk to route call via an outbound proxy
I'm also using ARI to dynamically select the Asterisk peer and remote end point, concern is how to use the same peer configuration with different end points and have asterisk populate the request-uri correctly? Nitesh On Wed, Apr 13, 2016 at 10:11 PM, Nitesh Bansal <nitesh.bansal at gmail.com> wrote:> 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 <jcolp at digium.com> 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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160413/72fda53b/attachment.html>
Joshua Colp
2016-Apr-13 20:31 UTC
[asterisk-users] Using Asterisk to route call via an outbound proxy
Nitesh Bansal wrote:> I'm also using ARI to dynamically select the Asterisk peer and remote > end point, concern is how to use the same peer configuration > with different end points and have asterisk populate the request-uri > correctly?I can only speak for chan_pjsip I'm afraid, but Olle has provided information about chan_sip on your -dev cross post. -- 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