asterisk users - Jan 2016 - Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

Hi!
I wish you all e Happy New Year first!

Allthough, I'm relative new to Asterisk, I got our server up and
Running, Softphones, ISDN, and a brand new Snom 821 are working
flawlessly. :)
Platform is Debian 8/Asterisk Packages (11) from Debian Repo.

But I am running into problems setting up 2 older Hardphones, Thomson
2030S. :(


with in my sip.conf, I have got for this hardphone:
[...]
[hard1]
        username=hard1
        secret=correct-and-three-times-checked-4-digit-pin
        allow=ulaw
        allow=alaw
        context=internal
        type=friend
        nat=force_rport,comedia
        qualify=yes
        host=dynamic
[...]

The SIP debug Output from Asteris CLI :
gw*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Using SIP CoS mark 4


Here Rebooting/Power up Thomson 2030S on 192.168.11.72

<--- SIP read from UDP:192.168.11.72:5060 --->
REGISTER sip:192.168.11.251;user=phone SIP/2.0
v: SIP/2.0/UDP 192.168.11.72:5060;branch=z9hG4bK8030819752698542643-155044
f: <sip:41 at 192.168.11.251:5060;user=phone>;tag=c0a80101-25da4
t: <sip:41 at 192.168.11.251:5060;user=phone>
i: 5789b-c0a80101-5-4 at 192.168.11.72
CSeq: 1 REGISTER
Route: <sip:192.168.11.251:5060;lr>
Max-Forwards: 70
Expires: 60
m: <sip:41 at 192.168.11.72:5060;user=phone>
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-5A-AA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
l: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.11.72:5060 (no NAT)
Sending to 192.168.11.72:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.72:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.11.72:5060;branch=z9hG4bK8030819752698542643-155044;received=192.168.11.72
From: <sip:41 at 192.168.11.251:5060;user=phone>;tag=c0a80101-25da4
To: <sip:41 at 192.168.11.251:5060;user=phone>;tag=as2220827f
Call-ID: 5789b-c0a80101-5-4 at 192.168.11.72
CSeq: 1 REGISTER
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="gw",
nonce="47287142"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5789b-c0a80101-5-4 at
192.168.11.72'
in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '5789b-c0a80101-5-4 at
192.168.11.72'
in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.11.72:5060 --->
REGISTER sip:192.168.11.251;user=phone SIP/2.0
v: SIP/2.0/UDP 192.168.11.72:5060;branch=z9hG4bK1414692036426970487-155047
f: <sip:41 at 192.168.11.251:5060;user=phone>;tag=c0a80101-25da4
t: <sip:41 at 192.168.11.251:5060;user=phone>
i: 5789b-c0a80101-5-4 at 192.168.11.72
CSeq: 2 REGISTER
Route: <sip:192.168.11.251:5060;lr>
Max-Forwards: 70
Expires: 60
m: <sip:41 at 192.168.11.72:5060;user=phone>
Authorization: Digest username="hard1", realm="gw",
nonce="47287142",
uri="sip:192.168.11.251",
response="f57f93a5a59e8f72aad5a5a43d19d3bf",
algorithm=MD5
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-5A-AA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
l: 0


<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.11.72:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.72:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.11.72:5060;branch=z9hG4bK1414692036426970487-155047;received=192.168.11.72
From: <sip:41 at 192.168.11.251:5060;user=phone>;tag=c0a80101-25da4
To: <sip:41 at 192.168.11.251:5060;user=phone>;tag=as2220827f
Call-ID: 5789b-c0a80101-5-4 at 192.168.11.72
CSeq: 2 REGISTER
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5789b-c0a80101-5-4 at
192.168.11.72'
in 32000 ms (Method: REGISTER)


Any Ideas to get those Phone working?

TIA / with kind Regards from an Sowy Norther Germany

mit freundlichen Gr??en
J?rgen Sauer
-- 
J?rgen Sauer - automatiX GmbH,
+49-4209-4699, juergen.sauer at automatix.de
Gesch?ftsf?hrer: J?rgen Sauer,
Gerichtstand: Amtsgericht Walsrode ? HRB 120986
Ust-Id: DE191468481 ? St.Nr.: 36/211/08000
GPG Public Key zur Signaturpr?fung:
http://www.automatix.de/juergen_sauer_publickey.gpg

On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote:
> with in my sip.conf, I have got for this hardphone:
> [...]
> [hard1]
>         username=hard1
>         secret=correct-and-three-times-checked-4-digit-pin
In most cases, there is no need to set the "username=" option. The
name
of the device is the name within the square brackets above the
configuration section.
Delete the "username=hard1" and reload sip.conf.

Am 07.01.2016 um 10:55 schrieb Frank:
> On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote:
Thx, 4answer. :)
>> with in my sip.conf, I have got for this hardphone:
>> [...]
>> [hard1]
>>         username=hard1
>>         secret=correct-and-three-times-checked-4-digit-pin
> 
> In most cases, there is no need to set the "username=" option.
The name
> of the device is the name within the square brackets above the
> configuration section.
> Delete the "username=hard1" and reload sip.conf.
Should be so, agreed. But it worked quite a long time not this way.
:(

Got now up. Why? I do not know. This Hard phone is really needing an
full expert".

Now this piece of antique hardware it does recognize calls, which
asterisk sends.
Calling out, works, asterisk sees the device as "hard1". Calling
"hard1"
shows up, "not avaible"... Same Setup on Snom 821 works perfectly.


mit freundlichen Gr??en
J?rgen Sauer
-- 
J?rgen Sauer - automatiX GmbH,
+49-4209-4699, juergen.sauer at automatix.de
Gesch?ftsf?hrer: J?rgen Sauer,
Gerichtstand: Amtsgericht Walsrode ? HRB 120986
Ust-Id: DE191468481 ? St.Nr.: 36/211/08000
GPG Public Key zur Signaturpr?fung:
http://www.automatix.de/juergen_sauer_publickey.gpg