[Asterisk-Users] Grandstream Budgetone-102 client cannot register

Michael Cheedle michael at wildgate.com
Thu Aug 12 08:32:23 MST 2004 

Thanks for the reply Steve.  Changing the conf file to 'info' looks like a
good idea, but this one client still cannot register.

I've pasted a debug for an authentication attempt in case it could provide
more info:

*******
*******
Sip read:
REGISTER sip:65.126.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP 67.22.zzz.zzz:64436;branch=z9hG4bK9b6974bdc1f0be25
From: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=9a97737530f385b5
To: <sip:5611234567 at 65.126.yyy.yyy;user=phone>
Contact: <sip:5611234567 at 67.22.zzz.zzz:64436;user=phone>
Call-ID: ca3bcc17c83819b9 at 192.168.xxx.xxx
CSeq: 5874 REGISTER
Expires: 3600
User-Agent: Grandstream BT100 1.0.5.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 67.22.zzz.zzz : 64436 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
67.22.zzz.zzz:64436;branch=z9hG4bK9b6974bdc1f0be25;received=67.22.zzz.zzz;rp
ort=64436
From: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=9a97737530f385b5
To: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=as2fbab6ae
Call-ID: ca3bcc17c83819b9 at 192.168.xxx.xxx
CSeq: 5874 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5611234567 at 65.126.yyy.yyy>
Content-Length: 0


 to 67.22.zzz.zzz:64436
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
67.22.zzz.zzz:64436;branch=z9hG4bK9b6974bdc1f0be25;received=67.22.zzz.zzz;rp
ort=64436
From: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=9a97737530f385b5
To: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=as2fbab6ae
Call-ID: ca3bcc17c83819b9 at 192.168.xxx.xxx
CSeq: 5874 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5611234567 at 65.126.yyy.yyy>
WWW-Authenticate: Digest realm="sip.wildgate.com", nonce="1a8295de"
Content-Length: 0


 to 67.22.zzz.zzz:64436
Scheduling destruction of call 'ca3bcc17c83819b9 at 192.168.xxx.xxx' in 15000
ms
LNPBX02*CLI>

Sip read:
REGISTER sip:65.126.yyy.yyy SIP/2.0
Via: SIP/2.0/UDP 67.22.zzz.zzz:64436;branch=z9hG4bK7668ebac62f7bc24
From: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=9a97737530f385b5
To: <sip:5611234567 at 65.126.yyy.yyy;user=phone>
Contact: <sip:5611234567 at 67.22.zzz.zzz:64436;user=phone>
Authorization: DIGEST username="5611234567 ", realm="sip.wildgate.com",
algorithm=MD5, uri="sip:65.126.yyy.yyy", nonce="1a8295de",
response="3210e29268f1c403a5b620caade66b3e"
Call-ID: ca3bcc17c83819b9 at 192.168.xxx.xxx
CSeq: 5875 REGISTER
Expires: 3600
User-Agent: Grandstream BT100 1.0.5.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


13 headers, 0 lines
Using latest request as basis request
Sending to 67.22.zzz.zzz : 64436 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
67.22.zzz.zzz:64436;branch=z9hG4bK7668ebac62f7bc24;received=67.22.zzz.zzz;rp
ort=64436
From: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=9a97737530f385b5
To: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=as2fbab6ae
Call-ID: ca3bcc17c83819b9 at 192.168.xxx.xxx
CSeq: 5875 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5611234567 at 65.126.yyy.yyy>
Content-Length: 0


 to 67.22.zzz.zzz:64436
Transmitting (NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
67.22.zzz.zzz:64436;branch=z9hG4bK7668ebac62f7bc24;received=67.22.zzz.zzz;rp
ort=64436
From: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=9a97737530f385b5
To: <sip:5611234567 at 65.126.yyy.yyy;user=phone>;tag=as2fbab6ae
Call-ID: ca3bcc17c83819b9 at 192.168.xxx.xxx
CSeq: 5875 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5611234567 at 65.126.yyy.yyy>
Content-Length: 0
*******
*******

Best regards,

Michael

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Steve Szmidt
Sent: Wednesday, August 11, 2004 4:41 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Grandstream Budgetone-102 client cannot
register


-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On Wednesday 11 August 2004 02:39 pm, Michael Cheedle wrote:
> I have a client using a Grandstream Budgetone 102, but he is unable to
> register to my Asterisk server.  About every 20 seconds, I get the
> following messages:
>
> Aug 11 11:27:17 DEBUG[1087740720]: chan_sip.c:748 __sip_autodestruct: Auto
> destroying call '3b4b68ec48200ab9 at 192.168.xxx.xxx'
> Aug 11 11:27:19 NOTICE[1087740720]: chan_sip.c:7336 handle_request:
> Registration from '<sip:5611234567 at 65.126.yyy.yyy;user=phone>' failed for
> '67.22.zzz.zzz'
>
> I've gone through the suggested conviguration on http://www.voip-info.org,
> but it hasn't made a difference.  Any suggestions?

I got the idea from somewhere that it prefers inband or info signalling
(dtmfmode=info). This is what I use w no problems.

> Here are the details from my sip.conf:
>
> type=friend
> secret=####
> nat=yes
> canreinvite=no
> context=from-sip
> disallow=all
> allow=ulaw
> allow=alaw
> allow=ilbc
> dtmfmode=rfc2833
> host=dynamic
>
> Best regards,
>
> Michael
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

- --
Steve

"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
                                Benjamin Franklin

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