I'm using Asterisk to originate a call via Kamailio.

Request-URI in SIP INVITE coming from Asterisk looks like this

<sip:kamailio@x.x.x.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 convey

that 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 update

the Request-URI in the forwarded INVITE if I use the Route header.

Thanks,

Nitesh