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Discussion: Echec d'enregistrement telephone SIP

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  1. 18/11/2010, 15h54 #1
    chab
    chab est déconnecté
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    etant débutant dans le domaine d'asterisk, je n'ai rien modifié de spécial...

    rien de particulier non plus dans le /etc/hosts.

    Pour info voici les logs lorsque je connecte ekiga avec le même compte (siemens) et depuis la même connection internet. On y voit clairement que asterisk repond gentillement a ekiga alors qu avec le telephone asterisk ne repond rien....


    Code:
    <--- SIP read from UDP:89.4.2xx.1xx:61772 --->
REGISTER sip:server.domain.com SIP/2.0
CSeq: 5 REGISTER
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKd71ac623-c502-1910-9d9a-000f66cf7cc5;rport
User-Agent: Ekiga/3.2.7
From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5
Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi
To: <sip:siemens@server.domain.com>
Contact: <sip:siemens@89.4.2xx.1xx:61772>;q=1, <sip:siemens@89.4.2xx.1xx>;q=0.667, <sip:siemens@192.168.0.10>;q=0.334
llow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Content-Length: 0
Max-Forwards: 70


<------------->
--- (12 headers 0 lines) ---
Sending to 89.4.2xx.1xx : 61772 (no NAT)

<--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKd71ac623-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772
From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5
To: <sip:siemens@server.domain.com>;tag=as3384ec04
Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi
CSeq: 5 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78df32a9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:89.4.2xx.1xx:61772 --->
REGISTER sip:server.domain.com SIP/2.0
CSeq: 6 REGISTER
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKdb2ec823-c502-1910-9d9a-000f66cf7cc5;rport
User-Agent: Ekiga/3.2.7
Authorization: Digest username="siemens", realm="asterisk", nonce="78df32a9", uri="sip:server.domain.com", algorithm=MD5, response="fec67185c669f9a966e67d5194284a93"
From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5
Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi
To: <sip:siemens@server.domain.com>
Contact: <sip:siemens@89.4.2xx.1xx:61772>;q=1, <sip:siemens@89.4.2xx.1xx>;q=0.667, <sip:siemens@192.168.0.10>;q=0.334
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Content-Length: 0
Max-Forwards: 70


<------------->
--- (13 headers 0 lines) ---
Sending to 89.4.2xx.1xx : 61772 (no NAT)
    -- Registered SIP 'siemens' at 89.4.2xx.1xx port 61772
       > Saved useragent "Ekiga/3.2.7" for peer siemens

<--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKdb2ec823-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772
From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5
To: <sip:siemens@server.domain.com>;tag=as3384ec04
Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi
CSeq: 6 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 3600
Contact: <sip:siemens@89.4.2xx.1xx:61772>;expires=3600
Date: Thu, 18 Nov 2010 13:44:25 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:89.4.2xx.1xx:61772 --->
SUBSCRIBE sip:siemens@server.domain.com SIP/2.0
CSeq: 2 SUBSCRIBE
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK4ca0c823-c502-1910-9d9a-000f66cf7cc5;rport
User-Agent: Ekiga/3.2.7
From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5
Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi
To: <sip:siemens@server.domain.com>
Contact: <sip:siemens@89.4.2xx.1xx:61772>
Accept: application/simple-message-summary
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Event: message-summary
Content-Length: 0
Max-Forwards: 70


<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 89.4.2xx.1xx : 61772 (no NAT)
list_route: hop: <sip:siemens@89.4.2xx.1xx:61772>
Found peer 'siemens' for 'siemens' from 89.4.2xx.1xx:61772

<--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK4ca0c823-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772
From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5
To: <sip:siemens@server.domain.com>;tag=as0f9d2526
Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33349611"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi' in 32000 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:89.4.2xx.1xx:61772 --->
SUBSCRIBE sip:siemens@server.domain.com SIP/2.0
CSeq: 3 SUBSCRIBE
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKbe11c923-c502-1910-9d9a-000f66cf7cc5;rport
User-Agent: Ekiga/3.2.7
Authorization: Digest username="siemens", realm="asterisk", nonce="33349611", uri="sip:siemens@server.domain.com", algorithm=MD5, response="4aa386369a65250b3c10a494a5b42da9"
From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5
Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi
To: <sip:siemens@server.domain.com>
Contact: <sip:siemens@89.4.2xx.1xx:61772>
Accept: application/simple-message-summary
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Event: message-summary
Content-Length: 0
Max-Forwards: 70


<------------->
--- (15 headers 0 lines) ---
Creating new subscription
Sending to 89.4.2xx.1xx : 61772 (no NAT)
Found peer 'siemens' for 'siemens' from 89.4.2xx.1xx:61772

<--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 --->
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKbe11c923-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772
From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5
To: <sip:siemens@server.domain.com>;tag=as0f9d2526
Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Nov 18 14:44:26] NOTICE[17316]: chan_sip.c:21518 handle_request_subscribe: Received SIP subscribe for peer without mailbox: siemens
Really destroying SIP dialog '7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi' Method: SUBSCRIBE

<--- SIP read from UDP:89.4.2xx.1xx:61772 --->
PUBLISH sip:siemens@server.domain.com SIP/2.0
CSeq: 7 PUBLISH
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK81c3c823-c502-1910-9d9a-000f66cf7cc5;rport
User-Agent: Ekiga/3.2.7
From: <sip:siemens@server.domain.com>;tag=afbbc823-c502-1910-9d9a-000f66cf7cc5
Call-ID: dcb3c823-c502-1910-9d9a-000f66cf7cc5@cdi
To: <sip:siemens@server.domain.com>
Contact: <sip:siemens@89.4.2xx.1xx:61772>
Expires: 500
Event: presence
Content-Type: application/pidf+xml
Content-Length: 346
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:siemens@server.domain.com">
<tuple id="sip:siemens@server.domain.com_on_cdi">
<note>online - I'm online using Ekiga</note>
<status>
<basic>open</basic>
</status>
<contact priority="1">siemens@server.domain.com</contact>
</tuple>
</presence>

<------------->
--- (13 headers 10 lines) ---

<--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK81c3c823-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772
From: <sip:siemens@server.domain.com>;tag=afbbc823-c502-1910-9d9a-000f66cf7cc5
To: <sip:siemens@server.domain.com>;tag=as62cfd18d
Call-ID: dcb3c823-c502-1910-9d9a-000f66cf7cc5@cdi
CSeq: 7 PUBLISH
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:89.4.2xx.1xx:61772 --->


<------------->
Debian-50-lenny-32-minimal*CLI>
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  2. 19/11/2010, 15h30 #2
    chab
    chab est déconnecté
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    Ok en fait c'etait juste un probleme NAT... il suffisait donc de mettre un serveur stun dans la config du tel.

    Comme on peut le voir dans les logs ci dessus, l'adresse de la requete envoyée par le telephone etait l'adresse locale (192.168.0.50) d'ou le probleme.
    Ekiga quant à lui donne bien l'adresse publique dans l'invite...

    Ce site explique tres bien tout cela avec des petits schemas et tout:
    http://www.ordinoscope.net/index.php...tes/SIP_et_Nat
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