8 Aug
2009
8 Aug
'09
7:11 p.m.
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(a)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