Hi,

For voice communication I use a SIP+RTP proxy together with SER.
For PSTN calls SER routes the INVITE messages to the RTP proxy.

Everything works fine there, but when the call is ended by one client (which is connected to SER) or the PSTN user, the BYE message does not arrive to SER. If our RTP proxy does not see the message coming from SER, call doesn't end.

That's why I need all of the BYE messages to go through SER.
How can I do that ?
In the default configuration there is a record route process for every message other than REGISTER.
But still BYE and ACK messages are not sent to SER.

Should I change the contact header of the messages arriving from RTP proxy ?
If so, how can I do it ?

What does following configuration block do ?

        if (loose_route()) {
                # mark routing logic in request
                append_hf("P-hint: rr-enforced\r\n");
                route(1);
                break;
        };

Thanks,
ilker

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