NITESH BANSAL wrote
Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in SIP INVITE coming from Asterisk looks like this<sip:
kamailio@.x
>But my objective is to use Kamailio to forward the call to a remote endpoint. What header should I put in the SIP INVITE from Asterisk to Kamailio to conveythat Kamailio should use this 'SIP URI' to route the call onwards.I tried 'Route' header, but it doesn't seem very clean, as kamailio doesn't updatethe Request-URI in the forwarded INVITE if I use the Route header. Thanks,Nitesh
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If you need to change R-URI, than modify it in script logic (PV $ru - is read/writable) e.g. (random example, set your new_user, new_domain if want to change them. Play around with it): $ru = "sip:" + $var(new_user) + "@" $var(new_domain) + ":" + $var(new_port) + ";transport=UDP" ;
If you need to route SIP request to destination that differ from R-URI - use route header (as you already tried). Or use Destination URI PV - $du (it is set, it is used for message routing, not R-URI). Both mentioned here does not change R-URI.
Hope this helps. Cheers!
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