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

[Alex Balashov]

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 
> 
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users at lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users


-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775