[Asterisk-Users] Grandstream Budgetone SIP registration fails

[Jean-Marc V. Liotier]

This is a very basic problem, and I feel stupid to resort to the list to
solve it, but after three hours pulling my hair trying all combinations
of a handful of parameters and getting nowhere I fail to see the path
leading to a solution.

I just got a pair of Budgetones. I have played a little with Asterisk
before, for example using Gnophone to call the talking clock, to leave
voicemail and receive it by email or to call the Digium IVR - basic
stuff but this is hust to point out that I am not completely lost with
Asterisk. Although I am new to the SIP part I have probably read all
that there is to read about configuring a SIP phone with Asterisk : it
seems like a very simple process and it makes not succeeding even more
frustrating... Whatever solution I find I will add to the wiki !

Here is the Asterisk console output of what the phone initially sends
when it attempts SIP registration :

Sip read: 
REGISTER sip:192.168.1.30 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bK4f3dc531eb4f14dc
From: <sip:6040 at 192.168.1.30>;tag=eb4f14dc11187288
To: <sip:6040 at 192.168.1.30>
Contact: *
Call-ID: 365d952bded45bf0 at 192.168.1.40
CSeq: 102 REGISTER
Expires: 0
User-Agent: Grandstream SIP UA 1.0.4.23
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0

Here is the relevant section of sip.conf :

[6040]
username=6040
secret=mysecret
type=friend
                                                                                                                                                                           The phone is on a static IP address. I have tried various possibly useful additions I picked along my readings such as :

defaultip=192.168.1.40
auth=md5
reinvite=no
nat=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw

But it makes no difference to my authentication problem, so I stuck to
the most basic possible set of parameters.

Now the worst part is that at some point I got a working setup, but I
changed something afterward to try to make it better, but I neglected
taking note of the working setup and I broke it again. I never found the
working setup again... Next time I'll make a backup copy of the working
setup before trying to ameliorate it...

On a successful attempt, here is the Asterisk console output of what the
phone initially sends when it attempts SIP registration :

Sip read: 
REGISTER sip:192.168.1.30 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bK43770e17dbdb1738
From: <sip:poste40 at 192.168.1.30>;tag=1c6ad3084cb9ac64
To: <sip:poste40 at 192.168.1.30>
Contact: <sip:poste40 at 192.168.1.40>
Proxy-Authorization: DIGEST username="poste40", realm="asterisk",
algorithm=MD5, uri="sip:192.168.1.30", nonce="4c7355bd",
response="7c3304ec9ffa7069de64ed17ef72f14d"
Call-ID: 5875e220123e627a at 192.168.1.40
CSeq: 103 REGISTER
Expires: 3600
User-Agent: Grandstream SIP UA 1.0.4.23
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0

It is different from what the phone sent before a failed attempt. The
main difference is that the 'Contact: *' line of the failed attempt is
replaced by :

Contact: <sip:poste40 at 192.168.1.40>
Proxy-Authorization: DIGEST username="poste40", realm="asterisk",
algorithm=MD5, uri="sip:192.168.1.30", nonce="4c7355bd",
respo

Since this is the initial packet in the SIP session and it is emitted by
the client, I infer that the solution of my problem certainly lies in
the configuration of the phone. Admin password, IP configuration and SIP
server address being proved correct since I access the phone's
administrative interface and the phones reaches the , the only remaining
parameters that have been changed are :

SIP User ID: 6040
Authenticate ID: 6040
Authenticate Password: mysecret

Now what ? I tried countless combinations of those parameters with the
basic configuration on the server, but I can't find the working one. The
answer is probably very simple and very obvious...

Someone on the list certainly has a working setup with Asterisk and
Grandstream Budgetone phones, I would be grateful if their SIP
configuration was posted to the list. Quite unexpectedly I found no
complete example of such working setup on the Web, maybe because it was
so simple that no one thought that posting it would be useful to anyone.
One I get mine working I shall post the parameters !