[Asterisk-Users] Uniden uip200
Devon Stephens
asterisk at yahwey.com
Tue Sep 28 10:37:30 MST 2004
I had my uip200 connected on the same network as the * server using
nat=never, and it worked fine. I just found the nat=route option, so I
upgraded to the latest CVS 2004-09-27. sip show peers shows it
registed, but I couldn't make any calls from the UIP.
I downgraded to the CVS mentioned in the UIP200 wiki page, 2004-08-27,
and now it works fine. It appears that something has been broken in
later releases.
I also am using 4.59a
Curt Moore wrote:
>Try nat=route to correct the rport issue mentioned earlier. I'm told
>by sources within Uniden that the firmware supporting STUN will be
>released soon.
>
>-Curt
>
>On Tue, 21 Sep 2004 22:44:34 -0600, Ryan Courtnage <ryan at voxbox.ca> wrote:
>
>
>>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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