[asterisk-users] Assigning specific RTP ports to SIP clients

[Tom Playford]

Dear All,

Is there a way of getting asterisk to assign specific UDP ports for
RTP communication between SIP clients. We have a dodgy SIP device that
will only send it's RTP audio out to port 8000.

I have tried fiddling with rtp.conf, but this is global, restricting
it to just one RTP port stops other SIP devices from working (duh).

If the above isn't possible, is there a way of finding which port has
been assigned? A small iptables script could then divert the RTP
stream to the correct port.


Thanks,

Tom