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Discussion: SIP REFER et fonction Transfer [RESOLU]

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  1. #1
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    SIP REFER et fonction Transfer [RESOLU]

    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
    Code:
    [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
    Code:
    [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 :

    Code:
    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 :
    Code:
    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.
    Dernière modification par ricobok ; 31/10/2011 à 17h14.

  2. #2
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    [Synopsis]
    Transfer caller to remote extension

    [Description]
    Transfer([Tech/]dest[,options]): Requests the remote caller be transferred
    to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
    an incoming call with the same channel technology will be transfered.
    Note that for SIP, if you transfer before call is setup, a 302 redirect
    SIP message will be returned to the caller.

    The result of the application will be reported in the TRANSFERSTATUS
    channel variable:
    SUCCESS Transfer succeeded
    FAILURE Transfer failed
    UNSUPPORTED Transfer unsupported by channel driver

    Je ne vois pas de @

    Si tu remplace:

    exten => _X.,n,Transfer(SIP/10002@ccm)

    Par:

    exten => _X.,n,Transfer(SIP/ccm)

    Ça donne quoi ?

    Et pourquoi transfer et pas dial ?

  3. #3
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    Question: Where - except for a SIP REFER - is this application of use, why not simply use Dial(), maybe together with a Local channel?

    Answer: For example, a SIP provider offering DIDs, you want calls to go to a colo box and then be transfered to other locations rather then waiting for the DID provider to update routes, by using this you save headaches with 1 way audio due to tracking firewalls not being able to track asyncronious SIP and RTP paths, and call delay if you and the colo are in different countries. Downside to doing something like this is you incur a slightly increase call setup delay while the call is bounced between servers, but no one will ever notice since most people take a few rings to actually answer.
    C'est ton cas ?

  4. #4
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    De mémoire, il y avait un bug avec le transfer via REFER dans les versions précédant la 1.6.2.16.

  5. #5
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    En fait, Le Call Manager doit conserver la gestion de l'appel.
    Asterisk reçoit l'appel du call manager, joue les messages qui vont bien après être passé dans un AGI.
    Et le but est de relacher l'appel de l'Asterisk.

    Le transfert me permet de libérer l'appel et le call manager gère l'aboutement.

    exten => _X.,n,Transfer(SIP/ccm)
    => cela ne va pas convenir car on ne sais pas vers quel DN réaliser le transfert

    J'ai testé ça :
    exten => _X.,n,Transfer(SIP/10002)

    Cette configuration passe ( mais 1 fois sur 10 environ)
    Des fois => Musique attente CCM mais le poste 10002 ne sonne pas
    Des fois => Musique attente CCM et poste 100002 sonne ===> OK, on voudrais ça à tous les coups...
    Des fois => Congestion

    Peux être plus la configuration du Call Manager ?

  6. #6
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    Dans tous les cas, en CLI j'ai :
    Code:
    CLI>
      == Using SIP RTP CoS mark 5
        -- Executing [11111@test:1] NoOp("SIP/ccm-00000119", ""APPEL ENTRANT"") in new stack
        -- Executing [11111@test:2] Answer("SIP/ccm-00000119", "") in new stack
        -- Executing [11111@test:3] Transfer("SIP/ccm-00000119", "SIP/100002") in new stack
        -- Auto fallthrough, channel 'SIP/ccm-00000119' status is 'UNKNOWN'
        -- Incoming call: Got SIP response 500 "Internal Server Error" back from X.X.X.200

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