[Asterisk-Users] NAT router and off-premise SIP audio problem

[Jim Greenfield, Computer Troubleshooters Metro NY/NJ]

Our network is connected to a cablemodem using a dynamic DNS service to
resolve our address. The Asterisk server has been alternately set up behind
a NAT router and without a NAT router -- that is, with two NICs, one of
which is providing NAT to the rest of the network; the office SIPs are
behind that with static private IP addresses.

Off-premise SIPs are all behind simple NAT routers.

Off-premise SIPs have been able to receive calls from and make calls through
the PSTN. No problem. Calls between on-premise SIPs, not a problem. Calls
between off-premise SIPs and any other SIPs connected to the server are a
problem... they ring up but no audio is passed in either direction.

SIP.CONF has NAT=YES.

We presume that a dedicated IP address for the Asterisk server would resolve
this but we would like to avoid the extra expense.

What are we missing? TIA.

Jim Greenfield
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