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 _______________________________________________ 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
