[Asterisk-Users] NAT setup

[Tom Rymes]

On Nov 14, 2005, at 11:57 AM, Andre Courchesne - Consultant wrote:

> Hi all,
>
>  I am setting up a a proof on concept where a SIP phone sits on the  
> internet and connects to a * behing a NAT.
>
>  Right now the SIP phone connects to the * box just fine, I can  
> dial and I see the commands being executed on the * box, but I  
> don't have any audio on the SIP phone. Any idas/pointers?

I would recommend that you do a little research on google, voip- 
info.org, and the list archives.

To connect to an Asterisk box that sits behind NAT, you need to  
forward ports 5060 and 10000-20000 too the asterisk box, and you need  
to configure the externip, localnet, and nat variables in sip.conf.  
audio problems are almost always due to the RTP stream (ports  
10000-20000) not being forwarded properly, either due to the port  
forwarding setup or the sip.conf settings.

Tom

----------------------------------------------------------
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Technology solutions for small and medium sized businesses.