[Asterisk-Users] Calling SIP Address From Behind NAT

[Daryll Strauss]

My asterisk box is behind a NAT firewall. I have friends that are on
Earthlink, Vonage, etc.

I'd like to make VOIP calls directly to them rather than going through the PSTN.

With Earthlink, I can make this work through FWD peeting numbers, but
that's sort of a waste of FWD bandwidth.

WIth Vonage, it doesn't work. I suspect this is because of the
breakage between FWD and Vonage that I saw mentioned on this list.

But going through FWD seems like a hack. I'd like to contact them
directly using SIP. Obviously this is difficult because of the NAT
firewall.

I'm running asterisk 1.0.2. In my sip.conf I've got localnet,
localmask, and externip defined. If I turn on sip debug, it looks like
the packets are getting rewritten correctly.

My entry for vonage looks like this:
[vonage]
type=peer
host=sip.vonage.net
context=default
canreinvite=no
dtmfmode=rfc2833
insecure=very

I tried telling my firewall to port forward all 5060 and 10000-11000
(my RTP range) to my asterisk box, but that doesn't seem to make any
difference. Is it necessary?

The bottom line is that outgoing doesn't work and I haven't gotten to
testing incoming calls yet. Any idea?

Thanks
- |Daryll