----- Original Message ----- From: "Anton Krall" <[EMAIL PROTECTED]>
would like to hear some actual setups and how people are solving the nat
issue within scenarios like:

Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones


I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities:

1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs.

2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs.

3 & 4.  Like #2 but no X100p.

All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 & 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out).

NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer.

Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special....

regards,

Paul


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