[Asterisk-Users] SIP behind NAT: NAT'ted end has to talk first?
radamson at routers.com
Tue Dec 2 12:38:05 MST 2003
> 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.
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