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

[Brian Cuthie]

Bruce,

I think this is related to your firewall. You may want to take a look a 
posting I did a few weeks ago.

http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html

Something on this topic probably belongs in the wiki.

-brian


Bruce Komito wrote:

>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
>
>
>
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