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.Try adding canreinvite=no in the config of the remote phones.. This will force the audio path through Asterisk..
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
Also I would suggest that you NOT put the Asterisk server behind NAT..
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