I am trying to set up a very basic SIP server. I can only establish external-to-internal calls using the following parameters with the default configuration files:
RTP Engine Config File interface=server_local_ip!public_domain_name
Kamailio Config File alias="public_domain_name" listen=server_local_ip:5061 adverise public_domain_name:5061
I cannot establish internal-to-external calls nor external-to-external calls with the above configuration settings. In these failed calls, the answering external callee UA sends 200-OK, but the caller UA does not ACK. In looking at the sip trace messages, no alias for the public ip address is appended to the Contact header of the external UA's 200-OK SIP message. So perhaps the caller doesn't know where to send the ACK.
The network setup has the local UAs and the Kamailio server behind a port-forwarded NAT routher. External (internet) UAs are also behind NAT routers.
Is there a change or addition to the above config snippets that would solve the problem, or do I need to add some instruction to the routing section?
Finally, does anyone sell or otherwise offer basic Kamailio/RTPEngine configuration files that work more or less "out-of-the box" for the described setup?
Thanks,
Steve Hamilton