[Asterisk-Users] Asterisk and Tecom IP2005 phone, problems :(

[Mike Dent]

Hi,
I'm having terrible trouble getting a Tecom IP2005 Sip phone working
with Asterisk 1.0

I installed Asterisk couple weeks ago, then installed a X100P card and
tested with X-Link
softphone, all seemed well.

So I thought I would buy a Sip phone from a UK company.
However I cannot seem to get it to authorise with Asterisk.

This is a link to the mfcr website :-
http://www.tecomproduct.com/IP2005.htm

And a link to the UK suppliers site:-
http://www.solwise.co.uk/voip-phones-ip2005.htm

Now with sip debug on I see messages like this:

Sip read:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft
Max-Forwards: 70
User-Agent: Centrality PA1688
From: home <sip:home at 192.168.1.2>;tag=yoyzIb5v2ZNzx08i
To: home <sip:home at 192.168.1.2>
Call-ID: kv3Hc37gOQL6pI4k
CSeq: 17455 REGISTER
Contact: <sip:home at 192.168.1.245:5060>
Expires: 360
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.245 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft
From: home <sip:home at 192.168.1.2>;tag=yoyzIb5v2ZNzx08i
To: home <sip:home at 192.168.1.2>;tag=as249efa19
Call-ID: kv3Hc37gOQL6pI4k
CSeq: 17455 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:home at 192.168.1.2>
Content-Length: 0


 to 192.168.1.245:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft
From: home <sip:home at 192.168.1.2>;tag=yoyzIb5v2ZNzx08i
To: home <sip:home at 192.168.1.2>;tag=as249efa19
Call-ID: kv3Hc37gOQL6pI4k
CSeq: 17455 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:home at 192.168.1.2>
WWW-Authenticate: Digest realm="asterisk", nonce="4532aca5"
Content-Length: 0


 to 192.168.1.245:5060
Scheduling destruction of call 'kv3Hc37gOQL6pI4k' in 15000 ms
splat*CLI>


The phone itself just displays "Failed login" message.

The phone did come with some firmware which is supposed to give it SIP
functionality, I've loaded this on and configured the sip server
192.168.1.2 in the
phone. 
The phone IP is 192.168.1.245.

Here is the section from sip.conf

[home]
type=friend
username=home
secret=secret
callerid="home1" <14>
;host=dynamic
port=5060
defaultip=192.168.1.245
nat=no
dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
canreinvite=yes                ; Typically set to NO if behind NAT
disallow=all
allow=g723.1
allow=gsm                     ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
context=sip
mailbox=2002

I'd appreciate any help!
Many thanks,

Mike