[SR-Users] 2 Asterisk servers re-inviting around kamailio even with t_relay being used?

Hi All,

I'm using kamailio with carrierroute to load balance calls to other servers.

I have 2 testing scenarios set up:

sipp><kamailio><asterisk server

In the above scenario all SIP transactions, dialogs go through kamailio, INVITE, Trying, ACK, OK, BYE, OK

Here is another one:

phone><asterisk><kamailio><asterisk server

In this scenario only the initial INVITE, ACK and OK go through kamailio, then the 2 asterisk servers finish the session directly to each other with ACK, BYE and OK

In both asterisk servers, canreinvite=no is set in peers and general section in sip.conf. The kamailio cfg is using t_relay. No errors are coming up anywhere.

Here is a pastebin of the 2 asterisk traces and the kamailio config.

http://pastebin.com/Uk9qVhX2

Any ideas on why the call is not maintaining SIP session through the proxy?

Thanks.

JR