I’ve been searching forums & browsing the web, but not found an answer so far. In my setup, asterisk is running on the gateway machine (asus wl500gp). So there’s two IP addresses - the public one, and the local ip.
internet <----> [public ip - asterisk - private ip] <----> SIP phones
With one provider everything works fine. However with another provider (netappel), I have the “one way audio” problem - no matter what nat= settings I have tried. (from a trace it seems that netappel has the RTP packets sent to a different IP than the original SIP packets - but they get “lost in space”)
My question is; how is asterisk supposed to work with multiple IP addresses ? Really there should be no NAT involved in my setup (I think); The SIP provider directly addresses asterisk on its public IP, and the phones directly contact asterisk on the private ip. As long as asterisk doesn’t try to setup a direct audio path between SIP provider and phone (canreinvite=no), everything should be ok (should - but it isn’t…).
Has anyone tried this setup ?