loose_route() has nothing to do with "call lookup" whatsoever.
Just do rewritehostport() on initial requests, and subsequent in-dialog messages will flow to the right place. For the most part. ACK and CANCEL require special handling. See stock example config file for details.
Raju Abhyankar wrote:
Hello,
I have setup Kamailio and Asterisk. Currently all PSTN traffic is forwarded to Asterisk which then terminates the call. What I would like to do is forward all SIP to SIP calls also to Asterisk? This implies I would like to turn off the call look up on Kamailio (loose route?) and blindly forward to Asterisk. Can some one suggest how this could be done.
Thanks,
Raju
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