If that's really your complete Kamailio config, it's missing a number of critical checks and handlers, such as loose_route(), which handles sequential (in-dialog) requests differently from initial requests due to the presence of a Route: header and a To header tag.

You can't just stick cr_route() in the main route { ... } block without those canonical checks and expect everything to work as desired.

The reason that the proxy is not seeing the sequential requests (reinvites, BYE, etc.) is because you don't have a Record-Route header, which tells the endpoints to relay sequential requests through the proxy on the network and transport level. Add this to your initial request route prior to (cr_route())[1]:

if(!is_method("REGISTER|OPTIONS|MESSAGE")) record_route();

That should take care of it.

-- Alex

[1] And ideally, add quite a few other things as well. See the stock configuration file for more insight.

On 06/08/2010 05:13 PM, JR Richardson wrote:

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