thr3ads.net - asterisk users - [Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config [Sep 2004]

If this information is useful, please help other people find it:
Share via:

Dinesh Nair

2004-Sep-08 05:29 UTC

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

hey * folk,

am trying to configure a WellGate 3504A FXS SIP ATA 
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set 
up two SIP clients in sip.conf as follows:

[general]
port = 5060			; Port to bind to
bindaddr = 0.0.0.0		; Address to bind SIP channel to
context = default		; Default context for incoming calls

[1235]
host = dynamic
secret = somepass
context = default
type = friend
amaflags = billing
accountcode = SIPUSER
disallow=all
allow=ulaw
allow=alaw

[1234]
host = dynamic
secret = somepass
context = default
type = friend
amaflags = billing
accountcode = SIPUSER
disallow=all
allow=ulaw
allow=alaw

and on the wellgate i've set the following under SIP information:

Run Mode: Proxy
Primary Proxy IP Address: 192.168.0.200 <-------- this is the IP addy of 
the asterisk server
Secondary Proxy IP Address: 	
Outbound Proxy: 	
Proxy port:  5060

Line 1 Number: 1234
Line 1 Account: 1234
Line 1 Password: somepass

Line 2 Number: 1235
Line 2 Account: 1235
Line 2 Password: somepass

Line 3 Number: 1236
Line 3 Account:
Line 3 Password:

Line 4 Number: 1237
Line 4 Account:
Line 4 Password:

SIP port:  5060
RTP Port:  16384
Expire:  60

however, only Line 1 from the wellgate succesfully authenticates with 
asterisk. picking up the analog handset attached to line 1 (1234) works 
fine with calls in and out working the way it should without any 
problems. Line 2 (and subsequently Lines 3 & 4) do not work at all. i've
captured the output of 'sip debug' and have attached it below. looking 
at the output, it does seem that asterisk is rejecting the 
authentication for line 2 but it doesn't mention why. the obvious 
reasons (password mismatch et al) have been ruled out, of course.

any help would be much appreciated. i've read mailing list archive of 
others being able to use the WellGate 3502 (which is just a 2xFXS port 
version) with asterisk. however no mention of VoIP protocols was 
mentioned as the wellgates traditionally supported H.323 but with 
firmware upgrade has been able to support SIP.

(sip debug output begins)

Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1234@192.168.0.202:5060>;expires=60
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3
To: <sip:1234@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 1 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3
To: <sip:1234@192.168.0.200>;tag=as2b502f28
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3
To: <sip:1234@192.168.0.200>;tag=as2b502f28
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Proxy-Authenticate: Digest realm="asterisk",
nonce="00727441"
Content-Length: 0


  to 192.168.0.202:5060


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1235@192.168.0.202:5060>;expires=60
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-488-4bff
To: <sip:1235@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 2 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-488-4bff
To: <sip:1235@192.168.0.200>;tag=as2b502f28
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1235@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-488-4bff
To: <sip:1235@192.168.0.200>;tag=as2b502f28
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1235@192.168.0.200>
Proxy-Authenticate: Digest realm="asterisk",
nonce="7b6d7de6"
Content-Length: 0


  to 192.168.0.202:5060


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:@192.168.0.202:5060>;expires=60
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
To: <sip:@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 3 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
To: <sip:@192.168.0.200>;tag=as2b502f28
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Sep  8 20:21:48 NOTICE[135410688]: chan_sip.c:5641 handle_request: 
Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:@192.168.0.202:5060>;expires=60
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
To: <sip:@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 4 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
To: <sip:@192.168.0.200>;tag=as11d046c3
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Sep  8 20:21:48 NOTICE[135410688]: chan_sip.c:5641 handle_request: 
Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1235@192.168.0.202:5060>;expires=60
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-488-4bff
To: <sip:1235@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 2 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-488-4bff
To: <sip:1235@192.168.0.200>;tag=as61d54b6c
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1235@192.168.0.200>
Content-Length: 0

  to 192.168.0.202:5060
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-488-4bff
To: <sip:1235@192.168.0.200>;tag=as61d54b6c
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1235@192.168.0.200>
Proxy-Authenticate: Digest realm="asterisk",
nonce="66543535"
Content-Length: 0


  to 192.168.0.202:5060


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:@192.168.0.202:5060>;expires=60
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
To: <sip:@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 3 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
To: <sip:@192.168.0.200>;tag=as61d54b6c
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Sep  8 20:21:48 NOTICE[135410688]: chan_sip.c:5641 handle_request: 
Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-87a-2093
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1234@192.168.0.202:5060>;expires=60
Proxy-Authorization: Digest username="1234",
realm="asterisk",
nonce="00727441", uri="sip:192.168.0.200:5060", 
response="f45772f0d9d4532a4a30926a177a0d9d", algorithm=MD5
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-875-7182
To: <sip:1234@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 5 REGISTER
Content-Length:0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-87a-2093
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-875-7182
To: <sip:1234@192.168.0.200>;tag=as3c072c96
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 5 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-87a-2093
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-875-7182
To: <sip:1234@192.168.0.200>;tag=as3c072c96
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 5 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Proxy-Authenticate: Digest realm="asterisk",
nonce="50a7e017"
Content-Length: 0


  to 192.168.0.202:5060


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-8f7-23ce
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1235@192.168.0.202:5060>;expires=60
Proxy-Authorization: Digest username="1235",
realm="asterisk",
nonce="7b6d7de6", uri="sip:192.168.0.200:5060", 
response="e56ab60e0610055d80e6168db62830fb", algorithm=MD5
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-8f7-60c9
To: <sip:1235@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 6 REGISTER
Content-Length:0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-8f7-23ce
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-8f7-60c9
To: <sip:1235@192.168.0.200>;tag=as3c072c96
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 6 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1235@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-8f7-23ce
From: <sip:1235@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-8f7-60c9
To: <sip:1235@192.168.0.200>;tag=as3c072c96
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 6 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1235@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Sep  8 20:21:49 NOTICE[135410688]: chan_sip.c:5641 handle_request: 
Registration from '<sip:1235@192.168.0.200>' failed for
'192.168.0.202'


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:@192.168.0.202:5060>;expires=60
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
To: <sip:@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 4 REGISTER
Content-Length:0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
To: <sip:@192.168.0.200>;tag=as08287470
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Sep  8 20:21:49 NOTICE[135410688]: chan_sip.c:5641 handle_request: 
Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95d-a41-4085
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1234@192.168.0.202:5060>;expires=60
Proxy-Authorization: Digest username="1234",
realm="asterisk",
nonce="50a7e017", uri="sip:192.168.0.200:5060", 
response="68272b42c4f057fe1727443efb8fde9d", algorithm=MD5
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-a41-31fc
To: <sip:1234@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 7 REGISTER
Content-Length:0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95d-a41-4085
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-a41-31fc
To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 7 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95d-a41-4085
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-a41-31fc
To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 7 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Proxy-Authenticate: Digest realm="asterisk",
nonce="514a6d7a"
Content-Length: 0


  to 192.168.0.202:5060


Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95d-b36-71e8
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:1234@192.168.0.202:5060>;expires=60
Proxy-Authorization: Digest username="1234",
realm="asterisk",
nonce="514a6d7a", uri="sip:192.168.0.200:5060", 
response="c3ddbfbb37c112fe2046556d85d1ec48", algorithm=MD5
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-b36-f0b
To: <sip:1234@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 8 REGISTER
Content-Length:0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.202 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95d-b36-71e8
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-b36-f0b
To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 8 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234@192.168.0.200>
Content-Length: 0


  to 192.168.0.202:5060
     -- Registered SIP '1234' at 192.168.0.202 port 5060 expires 60
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95d-b36-71e8
From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-b36-f0b
To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 8 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:1234@192.168.0.200>;expires=60
Date: Wed, 08 Sep 2004 12:21:49 GMT
Content-Length: 0


  to 192.168.0.202:5060

*CLI>
*CLI>
*CLI> sip no debug



(end of sip debug)

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh@alphaque.com                (0 0)    http://www.alphaque.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do                                        |
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a
$b."  |
| done; done                                                              |
+=========================================================================+

Dinesh Nair

2004-Sep-08 09:36 UTC

head link

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

On 08/09/2004 20:29 Dinesh Nair said the following:> am trying to configure a WellGate 3504A FXS SIP ATA 
> (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set 
> up two SIP clients in sip.conf as follows:
i forgot to mention that i'm running asterisk 0.90.0 on freebsd 4.10 and 
the exact same settings (unchanged) in sip.conf work with the Xlite and 
SJPhone SIP softphones.

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh@alphaque.com                (0 0)    http://www.alphaque.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do                                        |
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a
$b."  |
| done; done                                                              |
+=========================================================================+

Danny Zak

2004-Sep-08 14:17 UTC

head link

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

Hello Dinesh,

i'm pretty sure that asterisk doesn't support multi logins for each
line!


-- 
Best regards,
 Danny                            mailto:dannyz@belgonet.com

belGOnet.com  a  Euro-pictures  division - internet solutions
place princesse elisabeth 9/11   -   1030 Brussels  - Belgium
Tel : +32-(0)2-215.67.65      -      Fax : +32-(0)2-215.66.65

domains - hosting - hardware - VoiP - consultancy - backuping
CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL


No  legal  consequences  can be derived from the contents of the email
neither  is  belGOnet.com committed to them. The content of this email
is  exclusively  intended  for  adressee(s)  and information purposes.
belGOnet.com  accepts  no  liability for any damage resulting from the
use and/or acceptation of the content of this email.


Wednesday, September 8, 2004, 2:29:20 PM, you wrote:


DN> hey * folk,

DN> am trying to configure a WellGate 3504A FXS SIP ATA 
DN> (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've
set
DN> up two SIP clients in sip.conf as follows:

DN> [general]
DN> port = 5060			; Port to bind to
DN> bindaddr = 0.0.0.0		; Address to bind SIP channel to
DN> context = default		; Default context for incoming calls

DN> [1235]
DN> host = dynamic
DN> secret = somepass
DN> context = default
DN> type = friend
DN> amaflags = billing
DN> accountcode = SIPUSER
DN> disallow=all
DN> allow=ulaw
DN> allow=alaw

DN> [1234]
DN> host = dynamic
DN> secret = somepass
DN> context = default
DN> type = friend
DN> amaflags = billing
DN> accountcode = SIPUSER
DN> disallow=all
DN> allow=ulaw
DN> allow=alaw

DN> and on the wellgate i've set the following under SIP information:

DN> Run Mode: Proxy
DN> Primary Proxy IP Address: 192.168.0.200 <-------- this is the IP addy
of
DN> the asterisk server
DN> Secondary Proxy IP Address: 	
DN> Outbound Proxy: 	
DN> Proxy port:  5060

DN> Line 1 Number: 1234
DN> Line 1 Account: 1234
DN> Line 1 Password: somepass

DN> Line 2 Number: 1235
DN> Line 2 Account: 1235
DN> Line 2 Password: somepass

DN> Line 3 Number: 1236
DN> Line 3 Account:
DN> Line 3 Password:

DN> Line 4 Number: 1237
DN> Line 4 Account:
DN> Line 4 Password:

DN> SIP port:  5060
DN> RTP Port:  16384
DN> Expire:  60

DN> however, only Line 1 from the wellgate succesfully authenticates with
DN> asterisk. picking up the analog handset attached to line 1 (1234) works
DN> fine with calls in and out working the way it should without any 
DN> problems. Line 2 (and subsequently Lines 3 & 4) do not work at all.
i've
DN> captured the output of 'sip debug' and have attached it below.
looking
DN> at the output, it does seem that asterisk is rejecting the 
DN> authentication for line 2 but it doesn't mention why. the obvious 
DN> reasons (password mismatch et al) have been ruled out, of course.

DN> any help would be much appreciated. i've read mailing list archive of
DN> others being able to use the WellGate 3502 (which is just a 2xFXS port
DN> version) with asterisk. however no mention of VoIP protocols was 
DN> mentioned as the wellgates traditionally supported H.323 but with 
DN> firmware upgrade has been able to support SIP.

DN> (sip debug output begins)

DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1234@192.168.0.202:5060>;expires=60
DN> From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3
DN> To: <sip:1234@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 1 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
DN> From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3
DN> To: <sip:1234@192.168.0.200>;tag=as2b502f28
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 1 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Transmitting (no NAT):
DN> SIP/2.0 407 Proxy Authentication Required
DN> Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
DN> From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3
DN> To: <sip:1234@192.168.0.200>;tag=as2b502f28
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 1 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Proxy-Authenticate: Digest realm="asterisk",
nonce="00727441"
DN> Content-Length: 0


DN>   to 192.168.0.202:5060


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1235@192.168.0.202:5060>;expires=60
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN> To: <sip:1235@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 2 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN> To: <sip:1235@192.168.0.200>;tag=as2b502f28
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 2 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1235@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Transmitting (no NAT):
DN> SIP/2.0 407 Proxy Authentication Required
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN> To: <sip:1235@192.168.0.200>;tag=as2b502f28
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 2 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1235@192.168.0.200>
DN> Proxy-Authenticate: Digest realm="asterisk",
nonce="7b6d7de6"
DN> Content-Length: 0


DN>   to 192.168.0.202:5060


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:@192.168.0.202:5060>;expires=60
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
DN> To: <sip:@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 3 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 401 Unauthorized
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
DN> To: <sip:@192.168.0.200>;tag=as2b502f28
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 3 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Sep  8 20:21:48 NOTICE[135410688]: chan_sip.c:5641 handle_request:
DN> Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:@192.168.0.202:5060>;expires=60
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
DN> To: <sip:@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 4 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 401 Unauthorized
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
DN> To: <sip:@192.168.0.200>;tag=as11d046c3
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 4 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Sep  8 20:21:48 NOTICE[135410688]: chan_sip.c:5641 handle_request:
DN> Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1235@192.168.0.202:5060>;expires=60
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN> To: <sip:1235@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 2 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN> To: <sip:1235@192.168.0.200>;tag=as61d54b6c
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 2 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1235@192.168.0.200>
DN> Content-Length: 0

DN>   to 192.168.0.202:5060
DN> Transmitting (no NAT):
DN> SIP/2.0 407 Proxy Authentication Required
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN> To: <sip:1235@192.168.0.200>;tag=as61d54b6c
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 2 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1235@192.168.0.200>
DN> Proxy-Authenticate: Digest realm="asterisk",
nonce="66543535"
DN> Content-Length: 0


DN>   to 192.168.0.202:5060


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:@192.168.0.202:5060>;expires=60
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
DN> To: <sip:@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 3 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 401 Unauthorized
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95b-5b9-291b
DN> To: <sip:@192.168.0.200>;tag=as61d54b6c
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 3 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Sep  8 20:21:48 NOTICE[135410688]: chan_sip.c:5641 handle_request:
DN> Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-87a-2093
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1234@192.168.0.202:5060>;expires=60
DN> Proxy-Authorization: Digest username="1234",
realm="asterisk",
DN> nonce="00727441", uri="sip:192.168.0.200:5060", 
DN> response="f45772f0d9d4532a4a30926a177a0d9d", algorithm=MD5
DN> From: <sip:1234@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95c-875-7182
DN> To: <sip:1234@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 5 REGISTER
DN> Content-Length:0


DN> 12 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-87a-2093
DN> From: <sip:1234@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95c-875-7182
DN> To: <sip:1234@192.168.0.200>;tag=as3c072c96
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 5 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Transmitting (no NAT):
DN> SIP/2.0 407 Proxy Authentication Required
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-87a-2093
DN> From: <sip:1234@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95c-875-7182
DN> To: <sip:1234@192.168.0.200>;tag=as3c072c96
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 5 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Proxy-Authenticate: Digest realm="asterisk",
nonce="50a7e017"
DN> Content-Length: 0


DN>   to 192.168.0.202:5060


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-8f7-23ce
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1235@192.168.0.202:5060>;expires=60
DN> Proxy-Authorization: Digest username="1235",
realm="asterisk",
DN> nonce="7b6d7de6", uri="sip:192.168.0.200:5060", 
DN> response="e56ab60e0610055d80e6168db62830fb", algorithm=MD5
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95c-8f7-60c9
DN> To: <sip:1235@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 6 REGISTER
DN> Content-Length:0


DN> 12 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-8f7-23ce
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95c-8f7-60c9
DN> To: <sip:1235@192.168.0.200>;tag=as3c072c96
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 6 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1235@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Transmitting (no NAT):
DN> SIP/2.0 401 Unauthorized
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-8f7-23ce
DN> From: <sip:1235@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95c-8f7-60c9
DN> To: <sip:1235@192.168.0.200>;tag=as3c072c96
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 6 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1235@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Sep  8 20:21:49 NOTICE[135410688]: chan_sip.c:5641 handle_request:
DN> Registration from '<sip:1235@192.168.0.200>' failed for
'192.168.0.202'


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:@192.168.0.202:5060>;expires=60
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
DN> To: <sip:@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 4 REGISTER
DN> Content-Length:0


DN> 11 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 401 Unauthorized
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
DN> From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
DN> To: <sip:@192.168.0.200>;tag=as08287470
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 4 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Sep  8 20:21:49 NOTICE[135410688]: chan_sip.c:5641 handle_request:
DN> Registration from '<sip:@192.168.0.200>' failed for
'192.168.0.202'


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95d-a41-4085
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1234@192.168.0.202:5060>;expires=60
DN> Proxy-Authorization: Digest username="1234",
realm="asterisk",
DN> nonce="50a7e017", uri="sip:192.168.0.200:5060", 
DN> response="68272b42c4f057fe1727443efb8fde9d", algorithm=MD5
DN> From: <sip:1234@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95d-a41-31fc
DN> To: <sip:1234@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 7 REGISTER
DN> Content-Length:0


DN> 12 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95d-a41-4085
DN> From: <sip:1234@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95d-a41-31fc
DN> To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 7 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN> Transmitting (no NAT):
DN> SIP/2.0 407 Proxy Authentication Required
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95d-a41-4085
DN> From: <sip:1234@192.168.0.200>
DN> ;tag=c0a800ca-13c4-413ef95d-a41-31fc
DN> To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 7 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Proxy-Authenticate: Digest realm="asterisk",
nonce="514a6d7a"
DN> Content-Length: 0


DN>   to 192.168.0.202:5060


DN> Sip read:
DN> REGISTER sip:192.168.0.200:5060 SIP/2.0
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95d-b36-71e8
DN> Max-Forwards: 70
DN> Supported: replaces
DN> User-Agent: FXS_GW (4asipfxs.107a)
DN> Contact: <sip:1234@192.168.0.202:5060>;expires=60
DN> Proxy-Authorization: Digest username="1234",
realm="asterisk",
DN> nonce="514a6d7a", uri="sip:192.168.0.200:5060", 
DN> response="c3ddbfbb37c112fe2046556d85d1ec48", algorithm=MD5
DN> From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-b36-f0b
DN> To: <sip:1234@192.168.0.200>
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 8 REGISTER
DN> Content-Length:0


DN> 12 headers, 0 lines
DN> Using latest request as basis request
DN> Sending to 192.168.0.202 : 5060 (non-NAT)
DN> Transmitting (no NAT):
DN> SIP/2.0 100 Trying
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95d-b36-71e8
DN> From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-b36-f0b
DN> To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 8 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Contact: <sip:1234@192.168.0.200>
DN> Content-Length: 0


DN>   to 192.168.0.202:5060
DN>      -- Registered SIP '1234' at 192.168.0.202 port 5060 expires
60
DN> Transmitting (no NAT):
DN> SIP/2.0 200 OK
DN> Via: SIP/2.0/UDP
DN> 192.168.0.202:5060;branch=z9hG4bK-413ef95d-b36-71e8
DN> From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-413ef95d-b36-f0b
DN> To: <sip:1234@192.168.0.200>;tag=as6a7dd5a5
DN> Call-ID: c0a800ca-13c4-0-334-1c34
DN> CSeq: 8 REGISTER
DN> User-Agent: Asterisk PBX
DN> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN> Expires: 60
DN> Contact: <sip:1234@192.168.0.200>;expires=60
DN> Date: Wed, 08 Sep 2004 12:21:49 GMT
DN> Content-Length: 0


DN>   to 192.168.0.202:5060

*CLI>>
*CLI>>
*CLI>> sip no debug



DN> (end of sip debug)

Olle E. Johansson

2004-Sep-08 23:12 UTC

head link

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

Dinesh Nair wrote:
> 
> hey * folk,
> 
> am trying to configure a WellGate 3504A FXS SIP ATA 
> (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set 
> up two SIP clients in sip.conf as follows:The "Unathorized" really suggests that the password is wrong for 1235.
Check that you have the same secret on both asterisk and the welltech.

Look at this packet. The welltech sends no username at all. How can
we authorize that? I don't know if it's the missing "account"
on
1236 and 1237 that makes this happen.



Sip read:
REGISTER sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (4asipfxs.107a)
Contact: <sip:@192.168.0.202:5060>;expires=60
From: <sip:@192.168.0.200> ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
To: <sip:@192.168.0.200>
Call-ID: c0a800ca-13c4-0-334-1c34
CSeq: 4 REGISTER
Content-Length:0

/Olle

Olle E. Johansson

2004-Sep-08 23:13 UTC

head link

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

Dinesh Nair wrote:
> Proxy-Authenticate: Digest realm="asterisk",
nonce="514a6d7a"
Don't forget to change your realm to something that is unique for your
setup...
realm= in sip.conf :-)

Regards,
/Olle

Apparently Analagous Threads

Search for more apparently analagous threads

asterisk users - Sep 2004 - WellGate 3504A with Asterisk SIP authentication and config

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

[Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

Apparently Analagous Threads