[Asterisk-Users] Serious NAT problems: can't call between lines on sipura

[Bruce Komito]

I have a problem that is almost certainly nat-related, but I can't figure
out what's happening.

Since moving the Sipura behind a NAT server (Linksys), I am no longer able
to call between the two lines on the same Sipura.  When I dial one
extension from the other, it rings, but immediately after I pick up the
ringing phone, the call is uncerimoniously dumped.  I can tell the call
terminates immediately because I am watching the CDRs come out.  The *
server is on a public address with no firewall between it and the outside
world.

sip.conf: (both extensions have identical settings)
; Bruce
[5815]
type=friend
username=5815
secret=wpti5815
host=dynamic
mailbox=5815 at wpti
context=vpbx-wpti
qualify=3000
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
nat=yes

I'm thinking this has something to do with a setting in the Sipura, but I
don't know where to start.  I have nat keep-alive turned on, but I had to
turn stun off because it was causing a long, inexplicable delay after
dialing before the call would complete.

I'm realizing NAT with VoIP is a real problem.  Anyone have a silver
bullet they wish to share?

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115