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

[WipeOut]

Jim Greenfield, Computer Troubleshooters Metro NY/NJ wrote:

> 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
>
Try adding canreinvite=no in the config of the remote phones.. This will 
force the audio path through Asterisk..

Also I would suggest that you NOT put the Asterisk server behind NAT..