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

[Brian Cuthie]

Please ignore my previous post (below), as it's not really relevant to 
your problem. I was in some kind of mindless auto-email processing mode 
and responded without fully reading your message. Too much spam, too 
little sleep. Geesh.

-brian

Brian Cuthie wrote:

>
> 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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