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Mike Miller

2005-Mar-28 20:18 UTC

[Asterisk-Users] Problem with 401 Unauthorized

Hi,

I'm trying to set up asterisk, and I'm having some problems with a
simple register.

I'm not sure where to start even -- It seems that the problem is with
the response to the digest authentication, but I'm not sure how to fix
that. The log below is from linphone, but I see the exact same thing
with kphone and xten from a indows box as well.

I've tried changing the realm to the default "asterisk", or to my
IP
address, with the same results.

Log below:

*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:
REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
Contact: <sip:203@192.168.1.100>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.100 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
Content-Length: 0

to 192.168.1.100:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
WWW-Authenticate: Digest realm="mikeage.net",
nonce="4a322405"
Content-Length: 0

to 192.168.1.100:5060
Scheduling destruction of call '2259600998@192.168.1.100' in 15000 ms

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
Content-Length: 0

10 headers, 0 lines
Message is REGISTER

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
WWW-Authenticate: Digest realm="mikeage.net",
nonce="4a322405"
Content-Length: 0

11 headers, 0 lines
Message is REGISTER

Sip read:
REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
Contact: <sip:203@192.168.1.100>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.100 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
Content-Length: 0

to 192.168.1.100:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
WWW-Authenticate: Digest realm="mikeage.net",
nonce="0c54f346"
Content-Length: 0

to 192.168.1.100:5060
Scheduling destruction of call '2259600998@192.168.1.100' in 15000 ms

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
Content-Length: 0

10 headers, 0 lines
Message is REGISTER

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
WWW-Authenticate: Digest realm="mikeage.net",
nonce="0c54f346"
Content-Length: 0

11 headers, 0 lines
Message is REGISTER

Sip read:
REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
Contact: <sip:203@192.168.1.100>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.100 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
Content-Length: 0

to 192.168.1.100:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
WWW-Authenticate: Digest realm="mikeage.net",
nonce="4c62ddf6"
Content-Length: 0

to 192.168.1.100:5060
Scheduling destruction of call '2259600998@192.168.1.100' in 15000 ms

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
Content-Length: 0

10 headers, 0 lines
Message is REGISTER

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832
From: <sip:203@192.168.1.100>;tag=1543062237
To: <sip:203@192.168.1.100>;tag=1543062237
Call-ID: 2259600998@192.168.1.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:203@192.168.1.100>
WWW-Authenticate: Digest realm="mikeage.net",
nonce="4c62ddf6"
Content-Length: 0

11 headers, 0 lines
Message is REGISTER

Kevin P. Fleming

2005-Mar-29 07:37 UTC

head link

[Asterisk-Users] Problem with 401 Unauthorized

Mike Miller wrote:
> I'm not sure where to start even -- It seems that the problem is with
> the response to the digest authentication, but I'm not sure how to fix
> that. The log below is from linphone, but I see the exact same thing
> with kphone and xten from a indows box as well.
You are right (obviously), Asterisk is rejecting the REGISTER requests. 
However, without seeing the relevant portions of your sip.conf file, we 
cannot begin to tell you what is wrong, since we are not telepathic (as 
much as we may wish to be!).

Some things that can cause this:

- no type=peer entry that matches
- the peer entry is not host=dynamic
- the peer entry has permit/deny that disallow this device's IP address

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