[Asterisk-Users] Known SIP - NAT Solutions?

[Wade Weppler]

> There is now (thanks, Mark!) an addition in sip.conf called "nat=1"
> that can flag a sip user/peer/friend as being behind a NAT address
> translator.  The good news is that the REGISTER and INVITE requests
> seem to work on the ATA-186 from which I was experimenting.

I just tried the "nat=1" option with Windows Messenger, and it registers
now!  Woohoo!  But, I have the same problem.  Audio is only passing (as far
as I can tell) Messenger -> Asterisk but not the other way around.

Getting closer.  Great work Mark!

-wade