ricobok
31/10/2011, 12h50
Bonjour,
Je suis bloqué sur un transfer() qui ne fonctionne pas.
J'ai Asterisk 1.6.2.9-2ubuntu2.1 derriere un Call Manager 7.
Les deux connectés par un trunk SIP.
J'utilise asterisk afin de jouer des fichiers audios et transferer l'appel.
Mais le transfert ne fonctionne pas.
Ci dessous, la conf et les logs :
sip.conf
[ccm]
type=peer
;context=incoming
context=test ; POUR PASSER DANS LE DIALPLAN DE TEST
host=((IPCCM))
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=no
qualify=no
extensions.conf
[test]
exten => _X.,1,Noop("APPEL ENTRANT")
exten => _X.,n,Answer
...
... (pleins d'action avec des backgrounds et playback)
...
exten => _X.,n,Transfer(SIP/10002@ccm)
Voici la trace SIP :
REFER sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Contact: <sip:11111@X.X.X.201>
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Refer-To: <sip:10002@ccm>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Referred-By: <sip:11111@X.X.X.201>
BYE sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK5e1bb074;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
SIP/2.0 200 OK
Date: Mon, 31 Oct 2011 19:48:14 GMT
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Content-Length: 0
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK5e1bb074;rport
CSeq: 103 BYE
NOTIFY sip:11111@X.X.X.201:5060 SIP/2.0
Date: Mon, 31 Oct 2011 19:48:15 GMT
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Event: refer
Content-Length: 20
User-Agent: Cisco-CUCM7.0
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Contact: <sip:10001@X.X.X.200:5060>
Content-Type: message/sipfrag;version=2.0
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Subscription-State: active;expires=59
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3916922948
CSeq: 104 NOTIFY
Max-Forwards: 70
SIP/2.0 100 Trying
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3916922948;received =X.X.X.200
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 104 NOTIFY
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:11111@X.X.X.201>
Content-Length: 0
NOTIFY sip:11111@X.X.X.201:5060 SIP/2.0
Date: Mon, 31 Oct 2011 19:48:15 GMT
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Event: refer
Content-Length: 31
User-Agent: Cisco-CUCM7.0
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Contact: <sip:10001@X.X.X.200:5060>
Content-Type: message/sipfrag;version=2.0
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Subscription-State: terminated
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3b1d8edba2
CSeq: 105 NOTIFY
Max-Forwards: 70
SIP/2.0 487 Request Cancelled
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3b1d8edba2;received =X.X.X.200
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 105 NOTIFY
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:11111@X.X.X.201>
Content-Length: 0
REFER sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Contact: <sip:11111@X.X.X.201>
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Refer-To: <sip:10002@ccm>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Referred-By: <sip:11111@X.X.X.201>
SIP/2.0 500 Internal Server Error
Reason: Q.850;cause=100
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Content-Length: 0
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
CSeq: 102 REFER
Le REFER me parait etrange :
REFER sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Contact: <sip:11111@X.X.X.201>
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Refer-To: <sip:10002@ccm>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Referred-By: <sip:11111@X.X.X.201>
Existe t-il une autre méthode pour du SIP REFER ?
Merci d'avance.
Je suis bloqué sur un transfer() qui ne fonctionne pas.
J'ai Asterisk 1.6.2.9-2ubuntu2.1 derriere un Call Manager 7.
Les deux connectés par un trunk SIP.
J'utilise asterisk afin de jouer des fichiers audios et transferer l'appel.
Mais le transfert ne fonctionne pas.
Ci dessous, la conf et les logs :
sip.conf
[ccm]
type=peer
;context=incoming
context=test ; POUR PASSER DANS LE DIALPLAN DE TEST
host=((IPCCM))
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=no
qualify=no
extensions.conf
[test]
exten => _X.,1,Noop("APPEL ENTRANT")
exten => _X.,n,Answer
...
... (pleins d'action avec des backgrounds et playback)
...
exten => _X.,n,Transfer(SIP/10002@ccm)
Voici la trace SIP :
REFER sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Contact: <sip:11111@X.X.X.201>
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Refer-To: <sip:10002@ccm>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Referred-By: <sip:11111@X.X.X.201>
BYE sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK5e1bb074;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
SIP/2.0 200 OK
Date: Mon, 31 Oct 2011 19:48:14 GMT
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Content-Length: 0
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK5e1bb074;rport
CSeq: 103 BYE
NOTIFY sip:11111@X.X.X.201:5060 SIP/2.0
Date: Mon, 31 Oct 2011 19:48:15 GMT
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Event: refer
Content-Length: 20
User-Agent: Cisco-CUCM7.0
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Contact: <sip:10001@X.X.X.200:5060>
Content-Type: message/sipfrag;version=2.0
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Subscription-State: active;expires=59
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3916922948
CSeq: 104 NOTIFY
Max-Forwards: 70
SIP/2.0 100 Trying
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3916922948;received =X.X.X.200
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 104 NOTIFY
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:11111@X.X.X.201>
Content-Length: 0
NOTIFY sip:11111@X.X.X.201:5060 SIP/2.0
Date: Mon, 31 Oct 2011 19:48:15 GMT
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Event: refer
Content-Length: 31
User-Agent: Cisco-CUCM7.0
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Contact: <sip:10001@X.X.X.200:5060>
Content-Type: message/sipfrag;version=2.0
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Subscription-State: terminated
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3b1d8edba2
CSeq: 105 NOTIFY
Max-Forwards: 70
SIP/2.0 487 Request Cancelled
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.200:5060;branch=z9hG4bK1b3b1d8edba2;received =X.X.X.200
From: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
To: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 105 NOTIFY
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:11111@X.X.X.201>
Content-Length: 0
REFER sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Contact: <sip:11111@X.X.X.201>
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Refer-To: <sip:10002@ccm>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Referred-By: <sip:11111@X.X.X.201>
SIP/2.0 500 Internal Server Error
Reason: Q.850;cause=100
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
Content-Length: 0
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
CSeq: 102 REFER
Le REFER me parait etrange :
REFER sip:10001@X.X.X.200:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.201:5060;branch=z9hG4bK38f48103;rport
Max-Forwards: 70
From: <sip:11111@X.X.X.201>;tag=as1d13f4a6
To: "CCM_poste_1" <sip:10001@X.X.X.200>;tag=443c4a62-649d-4863-b70a-7ff843f3d620-18454632
Contact: <sip:11111@X.X.X.201>
Call-ID: 43f2d880-eae1fb7e-21e44-c86c107e@X.X.X.200
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Refer-To: <sip:10002@ccm>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Referred-By: <sip:11111@X.X.X.201>
Existe t-il une autre méthode pour du SIP REFER ?
Merci d'avance.