ja das Gerät läuft und ich kann es ohne Probleme anrufufen
hier einmal das sip. debug ip ....
--------
12 headers, 0 lines
Reliably Transmitting (no NAT) to 84.181.212.13:5060:
OPTIONS sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK11dcf575;rport
From: "asterisk" <sip:
[email protected]>;tag=as3468bbe4
To: <sip:
[email protected]>
Contact: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 Aug 2006 08:14:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 84.181.212.13:5060:
SIP/2.0 200 OK
Call-ID:
[email protected]
CSeq: 102 OPTIONS
From: "asterisk" <sip:
[email protected]>;tag=as3468bbe4
To: <sip:
[email protected]>;tag=a362329bcc1cc1a
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK11dcf575;rport
Content-Length: 0
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Contact: <sip:
[email protected]>
Supported: replaces
User-Agent: Brcm_Callctrl/1.5.1.0 MxSF/v3.2.6.26
--- (11 headers 0 lines)---
Destroying call '
[email protected]'
asterisk*CLI>
<-- SIP read from 84.181.212.13:5060:
REGISTER sip:voip.gk-soft.de SIP/2.0
Via: SIP/2.0/UDP 84.181.212.13;branch=z9hG4bKd17566418
Max-Forwards: 70
Content-Length: 0
To: 6722537265 <sip:
[email protected]>
From: 6722537265 <sip:
[email protected]>;tag=eba666ff605d0db
Call-ID:
[email protected]
CSeq: 1031143491 REGISTER
Contact: 6722537265 <sip:
[email protected]>;expires=240
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Authorization
igest response="8eb6bf37313cf4cac8769a4431b286a7",username="6722537265",realm="asterisk",nonce="1fea47d4",algorithm=MD5,uri="sip:voip.gk-soft.de"
User-Agent: Brcm_Callctrl/1.5.1.0 MxSF/v3.2.6.26
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 84.181.212.13 : 5060 (non-NAT)
Transmitting (no NAT) to 84.181.212.13:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.181.212.13;branch=z9hG4bKd17566418;received=84.181.212.13
From: 6722537265 <sip:
[email protected]>;tag=eba666ff605d0db
To: 6722537265 <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 1031143491 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:
[email protected]>
Content-Length: 0
---
Transmitting (no NAT) to 84.181.212.13:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 84.181.212.13;branch=z9hG4bKd17566418;received=84.181.212.13
From: 6722537265 <sip:
[email protected]>;tag=eba666ff605d0db
To: 6722537265 <sip:
[email protected]>;tag=as199221a6
Call-ID:
[email protected]
CSeq: 1031143491 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:
[email protected]>
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4629ede9"
Content-Length: 0
---
Scheduling destruction of call '
[email protected]' in 15000 ms
asterisk*CLI>
<-- SIP read from 84.181.212.13:5060:
REGISTER sip:voip.gk-soft.de SIP/2.0
Via: SIP/2.0/UDP 84.181.212.13;branch=z9hG4bK31b3d20d7
Max-Forwards: 70
Content-Length: 0
To: 6722537265 <sip:
[email protected]>
From: 6722537265 <sip:
[email protected]>;tag=eba666ff605d0db
Call-ID:
[email protected]
CSeq: 1031143492 REGISTER
Contact: 6722537265 <sip:
[email protected]>;expires=240
Allow: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE
Authorization
igest response="5b07a30d6ab93ab7bf15320ce87e4427",username="6722537265",realm="asterisk",nonce="4629ede9",algorithm=MD5,uri="sip:voip.gk-soft.de"
User-Agent: Brcm_Callctrl/1.5.1.0 MxSF/v3.2.6.26
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 84.181.212.13 : 5060 (non-NAT)
Transmitting (no NAT) to 84.181.212.13:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.181.212.13;branch=z9hG4bK31b3d20d7;received=84.181.212.13
From: 6722537265 <sip:
[email protected]>;tag=eba666ff605d0db
To: 6722537265 <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 1031143492 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:
[email protected]>
Content-Length: 0
---
Transmitting (no NAT) to 84.181.212.13:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.181.212.13;branch=z9hG4bK31b3d20d7;received=84.181.212.13
From: 6722537265 <sip:
[email protected]>;tag=eba666ff605d0db
To: 6722537265 <sip:
[email protected]>;tag=as199221a6
Call-ID:
[email protected]
CSeq: 1031143492 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 240
Contact: <sip:
[email protected]>;expires=240
Date: Mon, 14 Aug 2006 08:15:25 GMT
Content-Length: 0
---
Scheduling destruction of call '
[email protected]' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (no NAT) to 84.181.212.13:5060:
NOTIFY sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK7b64e22c;rport
From: "asterisk" <sip:
[email protected]>;tag=as5195822e
To: <sip:
[email protected]>
Contact: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: yes
Message-Account: sip:
[email protected]
Voice-Message: 4/0 (0/0)
---
Scheduling destruction of call '
[email protected]' in 15000 ms
asterisk*CLI>
<-- SIP read from 84.181.212.13:5060:
SIP/2.0 481 Call Does Not Exist
Call-ID:
[email protected]
CSeq: 102 NOTIFY
From: "asterisk" <sip:
[email protected]>;tag=as5195822e
To: <sip:
[email protected]>;tag=7a9dc365a0ce407
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK7b64e22c;rport
Content-Length: 0
Contact: <sip:
[email protected]>
User-Agent: Brcm_Callctrl/1.5.1.0 MxSF/v3.2.6.26
--- (9 headers 0 lines)---
Destroying call '
[email protected]'
Destroying call '
[email protected]'
--------
