[Asterisk-Users] SIP, * and clients behind NAT

[John Huang]

I am new to VOIP, Linux and Asterisk.  Through a lot of reading (this
list, voip-info.org, documentation, etc.), I successfully installed FC3
and * on a new Dell SC420 with two X100P connecting to two PSTN lines at
my office.  I've also installed AMP to help me configure IVRs, call
groups, extensions, etc.

I use a Handytone-286 ATA and x-lite clients on the internal network and
all works fine.

I would like to connect to * as an extension from home, from client
sites, from hotels, etc.  Most of these places will be behind some type
of NAT and/or firewall.  At my home, for example, I have a consumer
grade firewall/NAT.  I cannot get the Handytone-286 to work properly
from there.  I connect to the * server and register, I can call out and
incoming calls ring in, but there is no audio sent nor received
regardless of whether dialing out or calling in.

I suspect this has to do with RTP and how my home firewall/NAT handles
RTP.  Is my thinking correct here?  What's frustrating is that I can't
get it to work even if I put the Handytone-286 in a DMZ.  Maybe the
firewall/NAT is still processing and malforming the RTP packets?

Even if I do get the ATA working fine behind my home NAT, I would have
to do some reconfiguration most likely anywhere else I try plugging it
in, right?  And, if I wanted to add another ATA at home connected to the
same remote * server, it's most like not going to work without custom
RTP port forwards, etc., right?

Thanks,

John

John Huang