[OpenSIPS-Users] Call flow issues

[Daniel Corbe]

Hi,
I'll do my best to explain this as concisely as possible.  I have an
access-side network where most of my endpoints live out on the Internet and
they all use my OpenSIPS server as their outbound proxy and registrar;
however, the tenants do not live in OpenSIPS and I'm not using a radius
server for AAA on REGISTERs.

I am passing REGISTER requests up stream to my network core, like so:

route[2]
{
        setflag(6);
        fix_nated_register();
        fix_nated_contact();
        force_rport();

        # We must handle replies for registrations, for caching and location
        # tracking purposes.
        t_on_reply("1");
        t_on_failure("1");

        # Check for digest
        if (dummy_www_authorize(""))
        {
                xlog("L_INFO", "REGISTER: No Digest, sending challenge\n");
                www_challenge("", "0");
                exit;
        }

        # Digest was good if we get here
        save("location", "0x02");

        # Relay register to porta
        t_relay("udp:216.151.151.100:5060");
        exit;
}

The problem is that when the UA is NOT being natted and has a public
Internet address, the Contact HF is not being changed, therefore INVITE
requests destined for the UA never go through my OpenSIPS system and instead
my core tries to contact the UA directly.  fix_natted_contact() and
fix_natted_registrar() just aren't doing the job.

Is there a way to force OpenSIPS to always change the contact HF before
relaying the REGISTER request so subsequent INVITEs flow through the system?

Thanks.

-Daniel
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