[Asterisk-Users] Uniden uip200

[Lyle Giese]

I moved the phone to the same subnet as the * server and I got a bit further
as you indicate is the way it needs to be for now.  It's giving me a #3
registration error.

Could still use a couple of pointers on the uniden*.txt files as to what
they really need in there.  I still have something wrong in there.  I have a
GS 101 working, so I am not completely lost, but the lack of error
messages...

I turned on Sip debug and it looks like I get a lot of empty sip messages
when I have the UIP200 turned on and don't really see any traffic from it in
sip debug.  It can pickup an ip address from a dhcp server of course and
does pull down the .txt files from the tftp service I have running so it can
communicate with the network.

Lyle

----- Original Message -----
From: "Ryan Courtnage" <ryan at voxbox.ca>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Tuesday, September 21, 2004 11:44 PM
Subject: Re: [Asterisk-Users] Uniden uip200


> Lyle,
>
> If you are behind NAT, and * isn't, I'm afraid I have some bad news for
you.
>
> According to Uniden, STUN support is a "Feature Under Development".
>
> To furthur complicate things for you, the UIP200 currently does not
> respond (at all) to an INVITE that has 'rport' in the SIP Via field.  In
> other words, unless you want to tweak * source code, you have to use
> nat=never in your sip.conf.
> More info here:
> http://bugs.digium.com/bug_view_page.php?bug_id=0001935
>
> BS4.59a is the latest firmware.
>
> Your best bet is to call Uniden support and open a ticket with them.  I
> think i heard that the next firmware version is coming out mid-Oct ...
> if your lucky, that firmware will better support your environment.
>
> Ryan
>
>
> Lyle Giese wrote:
> > I got a Uniden UIP200 and started to configure it and I am lost....
> >
> > I have a tftp server setup on my * server and have the files
unidencom.txt
> > and uniden<mac>.txt there.  But it doesn't quite work yet.  It registers
as
> > a sip  phone (sip show peers), but I cann't dial it and the display
shows #1
> > disconnected all the time. It has firmware version BS4.59a in it.
> >
> > I have no idea if I have the configuration files on the tftp server
setup
> > correctly or not.  Where does one put in a STUN server?  What do they
mean
> > by proxy server?
> >
> > I tried to dial 124 and it just dropped into voicemail...
> >
> > Any ideas?
> >
> > Thanks,
> > Lyle
> >
> > sip conf
> >
> > ;uip200 1
> > [124]
> > type=friend
> > context=local
> > callerid="Lyle" <124>
> > username=124
> > secret=********
> > host=dynamic
> > nat=yes
> > canreinvite=no
> > dtmfmode=rfc2833
> > ;outgoinglimit=1
> > ;incominglimit=1
> > mailbox=101
> > disallow=all
> > ;allow=gsm
> > allow=ulaw
> > allow=alaw
> > ;allow=g723.1
> >
> > Extensions.conf
> >
> > exten => 124,1,Dial(SIP/124,24,Ttr)
> > exten => 124,2,VoiceMail(u101)
> > exten => 124,3,Hangup
> >
> > _______________________________________________
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>
> --
> Ryan Courtnage
> Director & CTO
> Coalescent Systems Inc
> 403.244.8089
> www.voxbox.ca
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