[Asterisk-Users] SIP Registration issues

[Andy Powell]

Hi,

I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. 

I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself!

Anyone got any ideas?

Thanks

Andy



11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.nixhelp.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8
From: <sip:andy at 192.168.1.2>;tag=as72c0d7da
To: <sip:andy at 192.168.1.2>
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Expires: 3600
Contact: <sip:1000 at 192.168.1.2>
Event: registration
Content-Length: 0

 (no NAT) to 192.168.1.2:5060


Sip read:
REGISTER sip:sip.nixhelp.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8
From: <sip:andy at 192.168.1.2>;tag=as72c0d7da
To: <sip:andy at 192.168.1.2>
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Expires: 3600
Contact: <sip:1000 at 192.168.1.2>
Event: registration
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.2 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8
From: <sip:andy at 192.168.1.2>;tag=as72c0d7da
To: <sip:andy at 192.168.1.2>;tag=as72c0d7da
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:andy at 192.168.1.2>
ontent-Length: 0


 to 192.168.1.2:5060
Jul 20 23:46:40 NOTICE[81930]: chan_sip.c:7320 handle_request: Registration from '<sip:andy at 192.168.1.2>' failed for '192.168.1.2'
Scheduling destruction of call '3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1' in 15000 ms


Sip read:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8
From: <sip:andy at 192.168.1.2>;tag=as72c0d7da
To: <sip:andy at 192.168.1.2>;tag=as72c0d7da
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:andy at 192.168.1.2>
Content-Length: 0


10 headers, 0 lines
    -- Got SIP response 403 "Forbidden" back from 192.168.1.2
Destroying call '3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1'