[Asterisk-Users] SIP behavior between different providers

[snacktime]

I spent the better part of the day trying to figure out why my SIP
calls going through * were just going dead after 20 seconds.  I was
sure it was a nat issue but now I'm not so sure anymore.

I have * on a public ip and clients behind a nat.  I was using
simpletelecom to terminate my calls.  I could connect fine if I went
direct from client -> simpletelecom.  If I used * as a proxy the audio
just went dead after 20 seconds.

A few minutes ago I tested the same setup using FWD instead of
simpletelecom.  Everything works, no dead audio.

And of course iax didn't have any problems at all.

Where should I be looking for the cause?  

Chris