[Asterisk-Users] SIP + NAT Howto?

[Stephen Varga]

I am new to * and I have been attempting to solve this same issue, but
have come to the conclusion that they only way to make it work is for *
to have a real reachable IP address or place another * box at the second
site and use IAX trunking. This second * box, unfortunately is
unsuitable for my scenario, but it may work in yours.

The issue is created by the fact that the *'s real ip address is in the
SDP information in the INVITE, usually this address is not directly
reachable from the second site.

When the XTEN actually tries to send the RTP data to this address it
either dies in the network or gets an ICMP Destination Unreachable
message, either way you don't have a two conversation.

Side note: If you put the * on the "outside" the XTEN phones will have
to have different RTP ports to avoid call conflicts.

HTH and maybe somebody has away for this to actually work.

Steve

On Fri, 2003-09-19 at 16:11, C. Johnson wrote:
> Hello Folks-
> 
> Pretty new to the list here, got a lot of reading to do.. Does anyone
> know where I can find a decent HOWTO or set of instructions for
> running
> Asterisk and SIP clients thru firewall/NAT systems?
> 
> I have a Asterisk box sitting behind a linux firewall at a remote
> location
> and have the 5060 and etc ports open as well at 16381-16391 UDP open
> and
> routed to the Asterisk box as well. I have a bunch of clients at
> another
> location which are also sitting behind a Linux ipchains/tables
> firewall
> 
> 
> So far, I'm able to get the clients (Xten Lite) to ring each other,
> but they
> ring, and one will say it's connected, while the other one just hangs
> up.
> 
> 
> -cj
> 
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