[Asterisk-Users] AudioCodes MP-104 - help !

[Dawid Mielnik]

I have really got things upto a point where I have no clue why Asterisk
doesnt authorise the audiocodes fxs box. I can not find anything in the
archives - some posts that could actually be useful are already deleted. Can
anyone please help me out ?!

My setup:

PSTN ----------------- Asterisk -----------------------router/nat------
Audiocodes ------------- POTS
                      xxx.xxx.xxx.xxx              80.54.223.79
					192.168.0.1        192.168.0.249

Debug Log:

Sip read:
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs
From: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=1c20095
To: <sip:mp_104_test at xxx.xxx.xxx.xxx>
Call-ID: 234102023420234yXVW at 80.54.223.79
CSeq: 16212 REGISTER
Expires: 3600
Contact: <sip:mp_104_test at 80.54.223.79;user=phone>;expires=3600
Content-Length: 0


9 headers, 0 lines
Using latest request as basis request
Sending to 80.54.223.79 : 5060 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs;received=80.54.223.79
From: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=1c20095
To: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=as0b288949
Call-ID: 234102023420234yXVW at 80.54.223.79
CSeq: 16212 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:mp_104_test at xxx.xxx.xxx.xxx>
Content-Length: 0


 to 80.54.223.79:1025
Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs;received=80.54.223.79
From: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=1c20095
To: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=as0b288949
Call-ID: 234102023420234yXVW at 80.54.223.79
CSeq: 16212 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:mp_104_test at xxx.xxx.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="3465f8ca"
Content-Length: 0


 to 80.54.223.79:1025
asterisk*CLI>

Sip read:
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj
From: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=1c20095
To: <sip:mp_104_test at xxx.xxx.xxx.xxx>
Call-ID: 234102023420234yXVW at 80.54.223.79
CSeq: 16213 REGISTER
Contact: <sip:mp_104_test at 80.54.223.79;user=phone>;expires=3600
Proxy-Authorization:Digest
username="mp_104_test",realm="asterisk",nonce="3465f8ca",uri="sip:xxx.xxx.xx
x.xxx",Algorithm="MD5",response="a41319cc5c8f9ddab6be04b2afe3d0ba"
Supported: em,timer,100rel
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway/MP-104 FXS/v.4.20.299.410
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 80.54.223.79 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj;received=80.54.223.79
From: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=1c20095
To: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=as0b288949
Call-ID: 234102023420234yXVW at 80.54.223.79
CSeq: 16213 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:mp_104_test at xxx.xxx.xxx.xxx>
Content-Length: 0


 to 80.54.223.79:1025
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj;received=80.54.223.79
From: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=1c20095
To: <sip:mp_104_test at xxx.xxx.xxx.xxx>;tag=as0b288949
Call-ID: 234102023420234yXVW at 80.54.223.79
CSeq: 16213 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:mp_104_test at xxx.xxx.xxx.xxx>
Content-Length: 0


 to 80.54.223.79:1025
Feb 14 12:39:49 NOTICE[1133718080]: chan_sip.c:5405 handle_request:
Registration from '<sip:mp_104_test at xxx.xxx.xxx.xxx>' failed for
'80.54.223.79'


in sip.conf

[mp_104_test]
type=friend
username=mp_104_test
secret=mp_104_test
auth=md5
disallow=all
allow=g729
allow=alaw
host=dynamic
nat=yes
qualify=200
dtmftone=rfc2833
context=default

On the AudioCodes gateway I use authentication per endpoint.

Thanks - I would really appreciate any help what so ever !!!!

Dave

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