[Asterisk-Users] SIP behind NAT: NAT'ted end has to talk first?

[Rich Adamson]

> I am having problems in a couple of installations where I have SIP 
> phones (both GS101 and ATA186) connecting to an asterisk box that has a 
> public IP address, where the stations are behind NAT.
> 
> I'm still testing to make sure I have all the permutations looked at, 
> but from what I can tell, what is happening is that in situations where 
> stations behind the NAT call out, no audio is passed until after the 
> party on the PUBLIC side generates some audio.

Not having any problems with a C7960 in the same type invironment. Been
stable for over a month. Running Asterisk CVS-11/11/03-13:46:29 right now.

Rich