Hi,
I'm trying to register SIP phone with an asterisk serve, failing miserably.
The server is sending "401 Unauthorized" responses to the registration
attempts, but every time the phone is re-REGISTERing without authorization.
I'd think this was a problem with the IP phone, except... the very same
phone registers correctly (authenticated) with another asterisk box, same brand,
similarly configured.
The phone is a Leadtek BVP 8882 videophone. The "bad" asterisk server
has the following build info, but I haven't seen any bug reports for this
problem...
Linux aadk 2.6.16.27sx00i-1.0.3.1 #2 Thu Aug 30 13:18:42 CDT 2007 blackfin
unknown
Asterisk Build:
Asterisk autotag_for_sx00i-1.0.3 (sx00i 1.0.3.1)
Asterisk GUI-version Revision: 1453 $
I'm wondering if the "401 unauthorized" response has bad
formatting. I compared the "bad" asterisk server repeated response,
with the "good" asterisk server first response (the phone includes
authorization in subsequent REGISTER for that one). The only difference I can
see, is that the "bad" asterisk responses have a blank
"Access-URL:" line before "WWW-Authenticate".
I've included log from the "bad" asterisk server. If necessary I
can provide one from the good server as well, but I've left it out for now
to avoid confusion.
Asterisk Business Edition autotag_for_sx00i-1.0.3 (sx00i 1.0.3.1), Copyright (C)
1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer
Thank you for using Business Edition. This Software is provided by Digium Inc
under license. Please refer to the license agreement provided with the Software.
==============================================================================Connected
to Asterisk autotag_for_sx00i-1.0.3 (sx00i 1.0.3.1) currently running on aadk
(pid = 304)
aadk*CLI> sip debug
aadk*CLI> SIP Debugging enabled
[Kaadk*CLI> The 'sip debug' command is deprecated and will be removed
in a future release. Please use 'sip set debug' instead.
[Kaadk*CLI> core set debug 255
aadk*CLI> Core debug was 0 and is now 255
[Kaadk*CLI> core set verbose 255
aadk*CLI> Verbosity was 0 and is now 255
[Kaadk*CLI>
<--- SIP read from 192.168.220.31:5060 --->
REGISTER sip:asterisk.foo.internal SIP/2.0
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce
Contact: sip:6001 at 192.168.220.31:5060
Max-Forwards: 70
User-Agent: LRSTD LR8882 2.5.00_99
Expires: 300
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.220.31 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001 at 192.168.224.91>
Access-URL:
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>;tag=as1aa11ae2
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL:
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="141ab0a6"
Content-Length: 0
<------------>
[Kaadk*CLI> Scheduling destruction of SIP dialog '2000fb00-4bca-c0a8efe3
at asterisk.foo.internal' in 32000 ms (Method: REGISTER)
[Kaadk*CLI>
<--- SIP read from 192.168.220.31:5060 --->
REGISTER sip:asterisk.foo.internal SIP/2.0
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce
Contact: sip:6001 at 192.168.220.31:5060
Max-Forwards: 70
User-Agent: LRSTD LR8882 2.5.00_99
Expires: 300
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.220.31 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001 at 192.168.224.91>
Access-URL:
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>;tag=as1aa11ae2
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL:
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="141ab0a6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2000fb00-4bca-c0a8efe3 at
asterisk.foo.internal' in 32000 ms (Method: REGISTER)
[Kaadk*CLI>
<--- SIP read from 192.168.220.31:5060 --->
REGISTER sip:asterisk.foo.internal SIP/2.0
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce
Contact: sip:6001 at 192.168.220.31:5060
Max-Forwards: 70
User-Agent: LRSTD LR8882 2.5.00_99
Expires: 300
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.220.31 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001 at 192.168.224.91>
Access-URL:
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>;tag=as1aa11ae2
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL:
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="141ab0a6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2000fb00-4bca-c0a8efe3 at
asterisk.foo.internal' in 32000 ms (Method: REGISTER)
[Kaadk*CLI>
<--- SIP read from 192.168.220.31:5060 --->
REGISTER sip:asterisk.foo.internal SIP/2.0
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce
Contact: sip:6001 at 192.168.220.31:5060
Max-Forwards: 70
User-Agent: LRSTD LR8882 2.5.00_99
Expires: 300
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.220.31 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001 at 192.168.224.91>
Access-URL:
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>;tag=as1aa11ae2
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL:
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="141ab0a6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2000fb00-4bca-c0a8efe3 at
asterisk.foo.internal' in 32000 ms (Method: REGISTER)
[Kaadk*CLI>
<--- SIP read from 192.168.220.31:5060 --->
REGISTER sip:asterisk.foo.internal SIP/2.0
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce
Contact: sip:6001 at 192.168.220.31:5060
Max-Forwards: 70
User-Agent: LRSTD LR8882 2.5.00_99
Expires: 300
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.220.31 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001 at 192.168.224.91>
Access-URL:
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.220.31:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31
From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9
To: 6001<sip:6001 at asterisk.foo.internal>;tag=as1aa11ae2
Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Access-URL:
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="141ab0a6"
Content-Length: 0
Any suggestions are welcome. I am frustrated as there is no reason I can tell
why the phone won't recognize need to authorize with one server, but works
fine with another.
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