[Asterisk-Users] SIP auth failed "wrong pw" but pw is correct

[Mark Drayton]

Hi

We've got an Asterisk 1.2.0 (planning to upgrade when I can) which is 
having trouble registering another Asterisk system as a client. We have 
the client in a realtime DB, our client has us configured as a friend and 
also has a register => username:password at our.host/username line in his 
sip.conf.

When he starts his system he gets an auth failed log line and we get:

Jun 12 14:42:21 NOTICE[1548]: chan_sip.c:10817 handle_request_register: 
Registration from '<sip:user at fqdn.our.host>' failed for '81.187.x.x' - 
Wrong password

The client has the right username and password.

One difference I've noticed is that the log error says "Registration from 
'<sip:user at fqdn.our.host>' failed for '81.187.x.x'" -- every other 
registration line quotes only the SIP account name, not the whole SIP URI.

This problem does look similar to bug #5103 but the fix should be 
incorporated in Asterisk 1.2.0, I think.

SIP trace:

Sending to 81.187.x.x : 5060 (non-NAT)
Transmitting (NAT) to 81.187.x.x:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
81.187.x.x:5060;branch=z9hG4bK26bc6cc8;received=81.187.x.x;rport=5060
From: <sip:username at fdqn.our.host>;tag=as48322a50
To: <sip:username at fdqn.our.host>
Call-ID: 6b6b0c51122cc44729ad525b47928c0e at 81.187.x.x
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:username at 146.101.our.ip>
Content-Length: 0


---
Transmitting (NAT) to 81.187.x.x:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
81.187.x.x:5060;branch=z9hG4bK26bc6cc8;received=81.187.x.x;rport=5060
From: <sip:username at fdqn.our.host>;tag=as48322a50
To: <sip:username at fdqn.our.host>;tag=as7592c16c
Call-ID: 6b6b0c51122cc44729ad525b47928c0e at 81.187.x.x
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:username at 146.101.our.ip>
WWW-Authenticate: Digest realm="asterisk", nonce="63563ef1"
Content-Length: 0


---
Scheduling destruction of call 
'6b6b0c51122cc44729ad525b47928c0e at 81.187.x.x' in 15000 ms
sip2*CLI>
<-- SIP read from 81.187.x.x:5060:
REGISTER sip:fdqn.our.host SIP/2.0
Via: SIP/2.0/UDP 81.187.x.x:5060;branch=z9hG4bK06e5ad1f;rport
From: <sip:username at fdqn.our.host>;tag=as3c9883b7
To: <sip:username at fdqn.our.host>
Call-ID: 6b6b0c51122cc44729ad525b47928c0e at 81.187.x.x
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="username", realm="asterisk", 
algorithm=MD5, uri="sip:fdqn.our.host", nonce="63563ef1", 
response="d6daaafb77c11357bda9b912b8ce6e02", opaque=""
Expires: 18000
Contact: <sip:username at 81.187.x.x>
Event: registration
Content-Length: 0


--- (13 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 81.187.x.x : 5060 (NAT)
Transmitting (NAT) to 81.187.x.x:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
81.187.x.x:5060;branch=z9hG4bK06e5ad1f;received=81.187.x.x;rport=5060
From: <sip:username at fdqn.our.host>;tag=as3c9883b7
To: <sip:username at fdqn.our.host>
Call-ID: 6b6b0c51122cc44729ad525b47928c0e at 81.187.x.x
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:username at 146.101.our.ip>
Content-Length: 0


---
Transmitting (NAT) to 81.187.x.x:5060:
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 
81.187.x.x:5060;branch=z9hG4bK06e5ad1f;received=81.187.x.x;rport=5060
From: <sip:username at fdqn.our.host>;tag=as3c9883b7
To: <sip:username at fdqn.our.host>;tag=as7592c16c
Call-ID: 6b6b0c51122cc44729ad525b47928c0e at 81.187.x.x
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:username at 146.101.our.ip>
Content-Length: 0

Any ideas?

Cheers,

Mark

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