[Asterisk-Users] NAT/SIP solution?

[Leif Madsen]

Uriel Carrasquilla wrote:
> Leif:
> I take it that the * on the GW machine remains in the middle of the call
> between SIP and the Nat'ed *, correct?
> I also suspect that you will be using IAX over Ethernet (to avoid
> compression/decompression delays), correct?
> Great work!

You'd be correct.  IAX is over Ethernet, and the GW machine which runs 
asterisk is the in-between man which talks to the remote end, and talks 
to the other NAT'd asterisk box.

I have this working great now, all my voicemail is implemented again, 
FWD works fine, FWD Welcome line works again, IAXtel in and out, 
basically everything I had before.

I figured out what my problem was before, the hostnames weren't 
resolving correctly, simply changed everything to standard IP address, 
and it all worked... sheesh!  I hate when stupid things like that get in 
the way :)  Now, off to fix the hostname issue.

Thanks,
Leif Madsen.