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<body class='hmmessage'><div dir='ltr'>I just noticed that Kamailio adds a route header of its own when forwarding the call, can't understand why it is doing it?<div>My Jitsi client doesn't like it 'preloaded route set'?</div><div>Any idea why would Kamailio add a Route header, i'm doing Record-routing, so it should add only</div><div>Record-Route header?</div><div><br></div><div>Thanks,</div><div>Nitesh<br><br><div><hr id="stopSpelling">From: nitesh.bansal@outlook.com<br>To: sr-users@lists.sip-router.org<br>Date: Tue, 5 Apr 2016 09:53:35 +0200<br>Subject: Re: [SR-Users] Kamailio setting the reqeust URI<br><br>
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<div dir="ltr">I got it working using Route header without 'lr' parameter.<div>In the absence of 'lr' parameter, Kamailio treated my request</div><div>as 'strict router' and updated the Reqeust-URI too.</div><div><br></div><div>Nitesh<br><br><div>> Date: Mon, 4 Apr 2016 08:24:42 -0700<br>> From: vancecezar@gmail.com<br>> To: sr-users@lists.sip-router.org<br>> Subject: Re: [SR-Users] Kamailio setting the reqeust URI<br>> <br>> NITESH BANSAL wrote<br>> > Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in<br>> > SIP INVITE coming from Asterisk looks like this<sip:<br>> <br>> > kamailio@.x<br>> <br>> > >But my objective is to use Kamailio to forward the call to a remote<br>> > endpoint.<br>> > What header should I put in the SIP INVITE from Asterisk to Kamailio to<br>> > conveythat Kamailio should use this 'SIP URI' to route the call onwards.I<br>> > tried 'Route' header, but it doesn't seem very clean, as kamailio doesn't<br>> > updatethe Request-URI in the forwarded INVITE if I use the Route header.<br>> > Thanks,Nitesh<br>> > <br>> > <br>> > _______________________________________________<br>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> <br>> > sr-users@.sip-router<br>> <br>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>> <br>> If you need to change R-URI, than modify it in script logic (PV $ru - is<br>> read/writable)<br>> e.g. (random example, set your new_user, new_domain if want to change them.<br>> Play around with it):<br>> $ru = "sip:" + $var(new_user) + "@" $var(new_domain) + ":" + $var(new_port)<br>> + ";transport=UDP" ;<br>> <br>> If you need to route SIP request to destination that differ from R-URI - use<br>> route header (as you already tried). Or use Destination URI PV - $du (it is<br>> set, it is used for message routing, not R-URI). Both mentioned here does<br>> not change R-URI.<br>> <br>> Hope this helps.<br>> Cheers!<br>> <br>> <br>> <br>> --<br>> View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-setting-the-reqeust-URI-tp147243p147247.html<br>> Sent from the Users mailing list archive at Nabble.com.<br>> <br>> _______________________________________________<br>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> sr-users@lists.sip-router.org<br>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br></div></div> </div>
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