[Serusers] No audio when terminating on the PSTN

[John A. Hull]

I'm a SIP/SER newbie, so please forgive me if the answer to this is
obvious to someone experienced:

I currently have SER running as a registration and outbound proxy
server, and I am able to forward calls to a PSTN gateway. Right now
SIP-to-PSTN calls are establishing sessions, being neither side is able
to hear the other talking. The softphone user is behind a
restricted-cone NAT, for which I'm using a STUN server to traverse. I
have surmised that this is an issue of the RTP packets not being able to
reach the client through the firewall, but I would expect the PSTN phone
to be able to hear the audio from the client. Is there a simple cause of
and solution to this problem that I'm overlooking?

	John