[Asterisk-Users] A room full of Cisco 7960s behind NAT

[steve]

The problem is the remote server. Asterisk is able to drop the media stream and allow the SIP phones to communicate directly, which has both its drawbacks and advantages depending on how you plan to use asterisk.  For this to take place you'll need the planets to be in the proper alignment and the following scenerio:

1.) the clients need to agree on a set of codecs so asterisk doesn't have to transcode them.
2.) both clients configured as 'canreinvite=yes' and 'nat=no'
3.) asterisk doens't have to listen for additional DTMF tones

Since your phones are behind a nat using a remote asterisk server the calls will always have to route through the * box even if you were calling an associate in the cube next to you.  If you were to install a local asterisk box it could handle this problem and also connect to the remote server as well.

So a call between two SIP phones will have to go through the remote
server? Or can those two phones be aware of each other?