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@192.168.1.2>;tag=as72c0d7da
To: <sip:andy@192.168.1.2>
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Expires: 3600
Contact: <sip:1000@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@192.168.1.2>;tag=as72c0d7da
To: <sip:andy@192.168.1.2>
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Expires: 3600
Contact: <sip:1000@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@192.168.1.2>;tag=as72c0d7da
To: <sip:andy@192.168.1.2>;tag=as72c0d7da
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:andy@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@192.168.1.2>' failed for '192.168.1.2'
Scheduling destruction of call
'3d1b58ba507ed7ab2eb141f241b71efb@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@192.168.1.2>;tag=as72c0d7da
To: <sip:andy@192.168.1.2>;tag=as72c0d7da
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:andy@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@127.0.0.1'
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell <andy@beagles-den.demon.co.uk> wrote:> 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'm using * with an IX66 and no issues, with CVS head I suggest you have a configuration error somewhere it looks like the IX66 is trying to authorise the clients, and no * have you set the IX66 to forward all sip requests for your domain to * ? Jason
Hello. Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP registration issues. My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing registration so calls to them go direct to VM although calling to other phones from them works fine. The logs show 'Transmitting (no NAT): SIP/2.0 403 Forbidden' which doesn't occur when they miraculously start working/registering. Asterisk seems to lose the user. Sep 9 11:47:36 VERBOSE[2444]: 12 headers, 0 lines Sep 9 11:47:36 VERBOSE[2444]: Using latest request as basis request Sep 9 11:47:36 VERBOSE[2444]: Sending to 192.168.1.100 : 5060 (non-NAT) Sep 9 11:47:36 VERBOSE[2444]: Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK289a5fe76 From: Martin <sip:207@192.168.1.50:5060>;tag=d6d383eca9b6910 To: Martin <sip:207@192.168.1.50:5060>;tag=as3c7c47f1 Call-ID: a7d9b00fac17fcfd05b2ccb6525a0d99@192.168.1.100 CSeq: 54943697 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:207@192.168.1.50> Content-Length: 0 to 192.168.1.100:5060 Sep 9 11:47:36 NOTICE[2444]: Registration from 'Martin <sip:207@192.168.1.50:5060>' failed for '192.168.1.100' Sep 9 11:47:36 VERBOSE[2444]: Scheduling destruction of call 'a7d9b00fac17fcfd05b2ccb6525a0d99@192.168.1.100' in 15000 ms Sep 9 11:47:36 VERBOSE[2444]: Sip read: REGISTER sip:192.168.1.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKd88070866 Max-Forwards: 70 Content-Length: 0 To: No User <sip:No%20User@192.168.1.50:5060> From: No User <sip:No%20User@192.168.1.50:5060>;tag=0e8bc4f3c760bc2 Call-ID: c682d5ee2c0a50fd8c239cf7bf254b29@192.168.1.100 CSeq: 535959059 REGISTER Contact: No User <sip:No%20User@192.168.1.100> Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 But then, some period of time later, they will start working at random times with no changes. Regards...Martin