Aastra SIP Telefon 6755i register failed

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phil_discount

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Hallo Leute,

ich habe ein Problem ein Aastra 6755i Telefon an meiner asterisk 1.4 zu registrieren.
Das Telefon kann angerufen werden, jedoch nicht abgehend telefonieren, weil die Authentifizierung fehlschlägt.
Ich habe bereits einen Forumeintrag bei Aastra gemacht, jedoch ohne Erfolg: http://forum.aastra.de/viewtopic.php?f=4&t=1891.
Ich hoffe ihr könnt mir helfen.

Auf meiner Trixbox mit asterisk 1.6 funktioniert das Telefon bestens, deshalb habe ich die Konfiguration von SIP verglichen (Telefone sind identisch eingerichtet)
Die linke Konfiguration funktioniert (asterisk 1.6)
http://www.***************de/show_img.php?img=a46458-1299767363.jpg&size=original
Der Bilderlink auf bilder-space funktioniert wohl nicht, das Bild ist aber auch im vorletzten Forumbeitrag auf aastra als Link enthalten.

Alle anderen Telefone Grandstream/Zoiper funktionieren einwandfrei auf der asterisk 1.4.

Bin für jeden Tipp dankbar.

Grüsse
Code: 
ka-vici-demo*CLI> 
[Mar 9 10:35:48] Reliably Transmitting (NAT) to 192.168.2.170:5060: 
OPTIONS sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK20676eb6;rport 
From: "asterisk" <sip:[email protected]>;tag=as7ec02d16 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
Contact: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
User-Agent: Asterisk PBX 
Max-Forwards: 70 
Date: Wed, 09 Mar 2011 09:35:48 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


--- 
[Mar 9 10:35:49] Retransmitting #1 (NAT) to 192.168.2.170:5060: 
OPTIONS sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK20676eb6;rport 
From: "asterisk" <sip:[email protected]>;tag=as7ec02d16 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
Contact: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
User-Agent: Asterisk PBX 
Max-Forwards: 70 
Date: Wed, 09 Mar 2011 09:35:48 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


--- 
[Mar 9 10:35:50] Retransmitting #2 (NAT) to 192.168.2.170:5060: 
OPTIONS sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK20676eb6;rport 
From: "asterisk" <sip:[email protected]>;tag=as7ec02d16 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
Contact: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
User-Agent: Asterisk PBX 
Max-Forwards: 70 
Date: Wed, 09 Mar 2011 09:35:48 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


--- 
[Mar 9 10:35:51] Retransmitting #3 (NAT) to 192.168.2.170:5060: 
OPTIONS sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK20676eb6;rport 
From: "asterisk" <sip:[email protected]>;tag=as7ec02d16 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
Contact: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
User-Agent: Asterisk PBX 
Max-Forwards: 70 
Date: Wed, 09 Mar 2011 09:35:48 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


--- 
[Mar 9 10:35:52] Retransmitting #4 (NAT) to 192.168.2.170:5060: 
OPTIONS sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK20676eb6;rport 
From: "asterisk" <sip:[email protected]>;tag=as7ec02d16 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
Contact: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
User-Agent: Asterisk PBX 
Max-Forwards: 70 
Date: Wed, 09 Mar 2011 09:35:48 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


--- 
[Mar 9 10:35:52] NOTICE[2560]: chan_sip.c:17097 sip_poke_noanswer: Peer '100' is now UNREACHABLE! Last qualify: 71 
[Mar 9 10:35:52] Really destroying SIP dialog '[email protected]' Method: OPTIONS 
[Mar 9 10:35:53] 
<--- SIP read from 192.168.2.170:5060 ---> 
SIP/2.0 404 Not Found 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK20676eb6;rport=5060;received=192.168.203.35 
From: "asterisk" <sip:[email protected]>;tag=as7ec02d16 
To: <sip:[email protected]:5060;transport=udp;cpd=on>;tag=1443086594 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
Server: Aastra 55i/3.2.0.70 
Content-Length: 0 


<-------------> 
[Mar 9 10:35:53] --- (8 headers 0 lines) --- 
[Mar 9 10:35:55] 
<--- SIP read from 192.168.2.170:5060 ---> 
REGISTER sip:192.168.203.35:5060 SIP/2.0 
Via: SIP/2.0/UDP 192.168.2.170;branch=z9hG4bKe5f39a7833dffa903 
Max-Forwards: 70 
From: "100" <sip:[email protected]:5060>;tag=2e731562c3 
To: "100" <sip:[email protected]:5060> 
Call-ID: c043f3910f009620 
CSeq: 4628 REGISTER 
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO 
Allow-Events: talk, hold, conference, LocalModeStatus 
Contact: "100" <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D264F2E>";expires=360 
Supported: path, gruu 
User-Agent: Aastra 55i/3.2.0.70 
Content-Length: 0 


<-------------> 
[Mar 9 10:35:55] --- (13 headers 0 lines) --- 
[Mar 9 10:35:55] Using latest REGISTER request as basis request 
[Mar 9 10:35:55] Sending to 192.168.2.170 : 5060 (NAT) 
[Mar 9 10:35:55] 
<--- Transmitting (NAT) to 192.168.2.170:5060 ---> 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.2.170;branch=z9hG4bKe5f39a7833dffa903;received=192.168.2.170 
From: "100" <sip:[email protected]:5060>;tag=2e731562c3 
To: "100" <sip:[email protected]:5060> 
Call-ID: c043f3910f009620 
CSeq: 4628 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


<------------> 
[Mar 9 10:35:55] 
<--- Transmitting (NAT) to 192.168.2.170:5060 ---> 
SIP/2.0 401 Unauthorized 
Via: SIP/2.0/UDP 192.168.2.170;branch=z9hG4bKe5f39a7833dffa903;received=192.168.2.170 
From: "100" <sip:[email protected]:5060>;tag=2e731562c3 
To: "100" <sip:[email protected]:5060>;tag=as4f3fd042 
Call-ID: c043f3910f009620 
CSeq: 4628 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65e8f147" 
Content-Length: 0 


<------------> 
[Mar 9 10:35:55] Scheduling destruction of SIP dialog 'c043f3910f009620' in 32000 ms (Method: REGISTER) 
[Mar 9 10:35:55] 
<--- SIP read from 192.168.2.170:5060 ---> 
REGISTER sip:192.168.203.35:5060 SIP/2.0 
Via: SIP/2.0/UDP 192.168.2.170;branch=z9hG4bKa21197f9dd328e24f 
Max-Forwards: 70 
From: "100" <sip:[email protected]:5060>;tag=2e731562c3 
To: "100" <sip:[email protected]:5060> 
Call-ID: c043f3910f009620 
CSeq: 4629 REGISTER 
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO 
Allow-Events: talk, hold, conference, LocalModeStatus 
Authorization: Digest username="100",realm="asterisk",nonce="65e8f147",uri="sip:192.168.203.35:5060",response="5216ea54b201df149183205453cbda13",algorithm=MD5 
Contact: "100" <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D264F2E>";expires=360 
Supported: path, gruu 
User-Agent: Aastra 55i/3.2.0.70 
Content-Length: 0 


<-------------> 
[Mar 9 10:35:55] --- (14 headers 0 lines) --- 
[Mar 9 10:35:55] Using latest REGISTER request as basis request 
[Mar 9 10:35:55] Sending to 192.168.2.170 : 5060 (NAT) 
[Mar 9 10:35:55] 
<--- Transmitting (NAT) to 192.168.2.170:5060 ---> 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.2.170;branch=z9hG4bKa21197f9dd328e24f;received=192.168.2.170 
From: "100" <sip:[email protected]:5060>;tag=2e731562c3 
To: "100" <sip:[email protected]:5060> 
Call-ID: c043f3910f009620 
CSeq: 4629 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


<------------> 
[Mar 9 10:35:55] Reliably Transmitting (NAT) to 192.168.2.170:5060: 
OPTIONS sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK6019865a;rport 
From: "asterisk" <sip:[email protected]>;tag=as030665ba 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
Contact: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
User-Agent: Asterisk PBX 
Max-Forwards: 70 
Date: Wed, 09 Mar 2011 09:35:55 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Content-Length: 0 


--- 
[Mar 9 10:35:55] 
<--- Transmitting (NAT) to 192.168.2.170:5060 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.2.170;branch=z9hG4bKa21197f9dd328e24f;received=192.168.2.170 
From: "100" <sip:[email protected]:5060>;tag=2e731562c3 
To: "100" <sip:[email protected]:5060>;tag=as4f3fd042 
Call-ID: c043f3910f009620 
CSeq: 4629 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 
Supported: replaces 
Expires: 360 
Contact: sip:[email protected]:5060;transport=udp;expires=360 
Date: Wed, 09 Mar 2011 09:35:55 GMT 
Content-Length: 0 


<------------> 
[Mar 9 10:35:55] Scheduling destruction of SIP dialog 'c043f3910f009620' in 32000 ms (Method: REGISTER) 
[Mar 9 10:35:55] 
<--- SIP read from 192.168.2.170:5060 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK6019865a;rport=5060;received=192.168.203.35 
From: "asterisk" <sip:[email protected]>;tag=as030665ba 
To: <sip:[email protected]:5060;transport=udp;cpd=on>;tag=4279812662 
Call-ID: [email protected] 
CSeq: 102 OPTIONS 
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO 
Server: Aastra 55i/3.2.0.70 
Supported: path 
Content-Length: 0 


<-------------> 
[Mar 9 10:35:55] --- (10 headers 0 lines) --- 
[Mar 9 10:35:55] NOTICE[2560]: chan_sip.c:13408 handle_response_peerpoke: Peer '100' is now Reachable. (103ms / 2000ms) 
[Mar 9 10:35:55] Really destroying SIP dialog '[email protected]' Method: OPTIONS 
[Mar 9 10:36:01] Scheduling destruction of SIP dialog '[email protected]' in 6592 ms (Method: NOTIFY) 
[Mar 9 10:36:01] Reliably Transmitting (NAT) to 192.168.2.170:5060: 
NOTIFY sip:[email protected]:5060;transport=udp;cpd=on SIP/2.0 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK5ea494d0;rport 
From: "asterisk" <sip:[email protected]>;tag=as0d5f9404 
To: <sip:[email protected]:5060;transport=udp;cpd=on> 
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: 2/0 (0/0) 

--- 
[Mar 9 10:36:01] 
<--- SIP read from 192.168.2.170:5060 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.203.35:5060;branch=z9hG4bK5ea494d0;rport=5060;received=192.168.203.35 
From: "asterisk" <sip:[email protected]>;tag=as0d5f9404 
To: <sip:[email protected]:5060;transport=udp;cpd=on>;tag=3233288977 
Call-ID: [email protected] 
CSeq: 102 NOTIFY 
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO 
Allow-Events: talk, hold, conference, LocalModeStatus 
Contact: "100" <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D264F2E>" 
Server: Aastra 55i/3.2.0.70 
Supported: path 
Content-Length: 0
 
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