[Asterisk-Users] One-way SIP question

[Jay Milk]

Ok, here's a good one -- I've tried a lot of different things, searched
the archives, etc.

I've signed up with a VOIP provider and coaxed the SIP settings out of
them.

I have username (=phone), password, sip.provider.com and
proxy.provider.com.  Using these settings, in Asterisk or X-lite, I can
place outgoing calls, but do not receive incoming calls.  With the same
settings on a Sipura SPA-2000 I can receive incoming calls, but not make
outgoing calls.

Any easy place to look?

Sipura and X-Lite both run behind a NAT firewall.  The Asterisk server
is also behind the firewall, but on a static-route (I have five static
IP addresses, * occupies one of those, and all "non-static" machines --
2 Vonage ATAs, the Sipura, several computers -- all NAT out on one
address)

Should I syslog the sipura to see what's wrong with it?  Not sure I'll
be able to read those logs.

Thanks
-- J