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" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> 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 > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Ryan Courtnage > Director & CTO > Coalescent Systems Inc > 403.244.8089 > www.voxbox.ca > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
