Hi Daniel,
I tried to add the new uri from carrierroute as a special header.
But I can't figure it out how to do it. Do you have an example or can you point me where I could find the information about it?
Now I forward the invite to the gateway IP address, but I'm not sure if the new assigned uri by cr is on the header. How can I see that?
The address 50.118 is my x-lite, 50.93 is Kamailio and 50.114 is my gateway
U 192.168.50.118:60560 -> 192.168.50.93:5060 INVITE sip:456789@192.168.50.93 SIP/2.0. Via: SIP/2.0/UDP 192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport. Max-Forwards: 70. Contact: sip:1002@192.168.50.118:60560. To: "456789"sip:456789@192.168.50.93. From: "1002"sip:1002@192.168.50.93;tag=2a692168.
U 192.168.50.93:5060 -> 192.168.50.118:60560 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport=60560. To: "456789"sip:456789@192.168.50.93. From: "1002"sip:1002@192.168.50.93;tag=2a692168.
U 192.168.50.93:5060 -> 192.158.50.114:5060 INVITE sip:456789@192.168.50.93 SIP/2.0. Via: SIP/2.0/UDP 192.168.50.93;branch=z9hG4bK81cd.0e7f7944.0. Via: SIP/2.0/UDP 192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport=60560. Max-Forwards: 69. Contact: sip:1002@192.168.50.118:60560. To: "456789"sip:456789@192.168.50.93. From: "1002"sip:1002@192.168.50.93;tag=2a692168.
This is my log:
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: time [Fri Feb 20 14:47:57 2009] method (INVITE) r-uri (sip:456789@192.168.50.93) 2nd via (<null>) Feb 20 14:47:57 localhost /sbin/kamailio[18371]: INFO:carrierroute:cr_do_route: uri 456789 was rewritten to sip:456789@host5.dk-ws Feb 20 14:47:57 localhost /sbin/kamailio[18371]: ERROR:tm:t_forward_nonack: no branch for forwarding Feb 20 14:47:57 localhost /sbin/kamailio[18371]: ERROR:tm:w_t_relay: t_forward_nonack failed Feb 20 14:47:57 localhost /sbin/kamailio[18371]: time [Fri Feb 20 14:47:57 2009] method (INVITE) r-uri (sip:456789@host5.dk-ws) 2nd via (<null>) Feb 20 14:48:27 localhost /sbin/kamailio[18377]: CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 408
Cheers,
Juan.-
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the R-URI so you just use:
t_relay("udp:asteriskip:asteriskport");
and in asterisk you dial the address in the r-uri.
If carrierroute is setting the dst_uri, then add it as special header, use same to forward to asterisk, but there use the special header to dial out.
Cheers, Daniel