Hi all,
I would like to ask you please how to configure kamailio in order to send signaling traffic to a specific gateway and rtp traffic to another gateway? Kindly note that I'm using media proxy for rtp and carrierroute for sending traffic to different gateways
Regards
michel freiha wrote:
Hi all,
I would like to ask you please how to configure kamailio in order to send signaling traffic to a specific gateway and rtp traffic to another gateway? Kindly note that I'm using media proxy for rtp and carrierroute for sending traffic to different gateways
This is something that the endpoints are expected to configure amongst themselves. Kamailio is a SIP proxy.
But inasmuch as Kamailio can mangle the SDP payload (and, since you use MediaProxy, you know that it quite clearly can), I suppose you could do a substitution operation on the SDP payload/SIP message body using the functions in textops.
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
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@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
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.
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
-
Alex Balashov -
Iñaki Baz Castillo -
michel freiha -
Raju Abhyankar