[Kamailio-Users] Send all traffic including SIP to SIP to Asterisk or PSTN Gateway

[Raju Abhyankar]

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 at aliax.net> wrote:

> From: Iñaki Baz Castillo <ibc at aliax.net>
> Subject: Re: [Kamailio-Users] Send all traffic including SIP to SIP to Asterisk or PSTN Gateway
> To: 
> Cc: users at lists.kamailio.org
> Date: Sunday, August 9, 2009, 9:31 AM
> 2009/8/8 Raju Abhyankar <kf6rzt at 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 at 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 at aliax.net>
> 
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