Hi Inaki Baz and others...
That sure did work. Thanks for all the suggestions. This really helped.
Best Regards,
Raju
--- On Sun, 8/9/09, Iñaki Baz Castillo ibc@aliax.net wrote:
From: Iñaki Baz Castillo ibc@aliax.net Subject: Re: [Kamailio-Users] Send all traffic including SIP to SIP to Asterisk or PSTN Gateway To: Cc: users@lists.kamailio.org Date: Sunday, August 9, 2009, 9:31 AM 2009/8/8 Raju Abhyankar kf6rzt@yahoo.com:
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.
"I would like to turn off the call look up on Kamailio (loose route?)"
¿?¿
If you want to pass all the traffic to Asterisk it to send it back to kamailio:
- Call from alice to bob.
- Kamailio checks if bob exists => does_uri_exist()
function.
- If true, route the call to Asterisk (without changing the
entire RURI or keeping the RURI username).
- Asterisk generates a call to "sip:bob@kamailio_IP" and
sends it to Kamailio.
- Kamailio does lookup("location").
Note: This wouldn't work in a multidomain scenario as Asterisk doesn't support real multidomain.
-- Iñaki Baz Castillo ibc@aliax.net
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users