[Asterisk-Users] Zultys Zip 4x4

Mike Roberts mroberts at hcs.net
Thu Jul 29 13:34:51 MST 2004


I tried that, and it still doesn't work.  On your Zultys 4x4, what SIP
parameters other than these did you configure:

Outbound proxy = IP of Asterisk 
Registrar Server = IP of Asterisk
Proxy Password = same password used in sip.conf

Thanks,
Mike

-----Original Message-----
From: Bruce Komito [mailto:brucek at bagel.com] 
Sent: Thursday, July 29, 2004 4:06 PM
To: Mike Roberts
Cc: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Zultys Zip 4x4

I have * working with a 4x4.  The only difference I can see is that you
don't have a secret configured.  You might try that and see if it makes a
difference.  BTW, don't even think of putting the 4x4 behind a NAT server.
It won't work.

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


On Thu, 29 Jul 2004, Mike Roberts wrote:

> Is anyone successfully using one of these with Asterisk?  I cannot get 
> the phone to register, this message keeps coming up on the Asterisk
console:
>
> Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request:
> Registration from '"000BEA801CA6" <sip:000BEA801CA6 at hcs.net:5060>' 
> failed for '204.194.36.138'
>
> The telephone LCD says "SIP registation rejected".
>
> My sip.conf file looks like this for the ZIP 4x4
>
> [2153]
> type=friend                   ; either "friend" (peer+user), "peer" or
> "user"
> context=sip-phones
> username=000BEA801CA6           ; usually matches the [section] title
> callerid=Zultys <2153>
> host=dynamic             ; we have a dynamic IP address
> ;nat=no                        ; there is not NAT between phone and
Asterisk
> canreinvite=yes               ; allow RTP voice traffic to bypass Asterisk
> dtmfmode=rfc2833                 ; either RFC2833 or INFO for the
BudgeTone
> ;outgoinglimit=1               ; disable callwaiting signal (2nd call to
> phone)
> ;incominglimit=1               ; permit only 1 outgoing call at a time
> mailbox=2153 at default  ; mailbox 1234 in voicemail context "default"
> disallow=all                  ; need to disallow=all before we can use
> allow=
> allow=ulaw                    ; Note: In user sections the order of codecs
>                                ; listed with allow= does NOT matter!
> ;allow=alaw
> ;allow=g723.1                  ; Asterisk only supports g723.1 pass-thru!
> ;allow=g729                    ; Pass-thru only unless g729 license
obtained
>
> Thanks in advance
>
> Mike Roberts
>
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