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Discussion: Probleme de configuration SIP

  1. #1
    Membre Junior
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    Probleme de configuration SIP

    Bonjour,

    Je rencontre un problème avec un did que j'ai possède lorsque je compose le numero ça coupe automatiquement.

    Voici les logs asterisk 1.8 qui pourrait m'aider à les traduires je n'arrive pas à comprendre d'ou vien le problème.

    [/CODE]


    <--- SIP read from UDP:46.4.71.238:5060 --->
    INVITE sip:13363105403@sip.norlecom.com SIP/2.0
    Via: SIP/2.0/UDP 46.4.71.238:5060;branch=z9hG4bK28526f39
    Max-Forwards: 70
    From: "13364052775" <sip:13364052775@46.4.71.238>;tag=as57bdb5d2
    To: <sip:13363105403@sip.norlecom.com>
    Contact: <sip:13364052775@46.4.71.238>
    Call-ID: 52b3b9b1379a11612271bd8832fb9b1f@46.4.71.238
    CSeq: 102 INVITE
    User-Agent: Asterisk
    Remote-Party-ID: "13364052775" <sip:13364052775@46.4.71.238>;privacy=off;screen=n o
    Date: Sat, 18 Feb 2012 12:06:22 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 308

    v=0
    o=root 1570846146 1570846146 IN IP4 46.4.71.238
    s=Asterisk PBX 1.6.2.17.2
    c=IN IP4 46.4.71.238
    t=0 0
    m=audio 13470 RTP/AVP 8 0 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    <------------->
    --- (15 headers 14 lines) ---
    Sending to 46.4.71.238:5060 (no NAT)
    Using INVITE request as basis request - 52b3b9b1379a11612271bd8832fb9b1f@46.4.71.238
    Found peer '13364052775' for '13364052775' from 46.4.71.238:5060

    <--- Reliably Transmitting (no NAT) to 46.4.71.238:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 46.4.71.238:5060;branch=z9hG4bK28526f39;received=4 6.4.71.238
    From: "13364052775" <sip:13364052775@46.4.71.238>;tag=as57bdb5d2
    To: <sip:13363105403@sip.norlecom.com>;tag=as5141fb3 d
    Call-ID: 52b3b9b1379a11612271bd8832fb9b1f@46.4.71.238
    CSeq: 102 INVITE
    Server: FPBX-2.10.0rc1(1.8.9.2)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="093544b1"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '52b3b9b1379a11612271bd8832fb9b1f@46.4.71.238' in 32000 ms (Method: INVITE)

    <--- SIP read from UDP:46.4.71.238:5060 --->
    ACK sip:13363105403@sip.norlecom.com SIP/2.0
    Via: SIP/2.0/UDP 46.4.71.238:5060;branch=z9hG4bK28526f39
    Max-Forwards: 70
    From: "13364052775" <sip:13364052775@46.4.71.238>;tag=as57bdb5d2
    To: <sip:13363105403@sip.norlecom.com>;tag=as5141fb3 d
    Contact: <sip:13364052775@46.4.71.238>
    Call-ID: 52b3b9b1379a11612271bd8832fb9b1f@46.4.71.238
    CSeq: 102 ACK
    User-Agent: Asterisk
    Remote-Party-ID: "13364052775" <sip:13364052775@46.4.71.238>;privacy=off;screen=n o
    Content-Length: 0

    <------------->
    --- (11 headers 0 lines) ---

    <--- SIP read from UDP:46.4.71.238:5060 --->
    INVITE sip:13363105403@sip.norlecom.com SIP/2.0
    Via: SIP/2.0/UDP 46.4.71.238:5060;branch=z9hG4bK3a399d8b
    Max-Forwards: 70
    From: "13364052775" <sip:13364052775@46.4.71.238>;tag=as2f59cbc4
    To: <sip:13363105403@sip.norlecom.com>
    Contact: <sip:13364052775@46.4.71.238>
    Call-ID: 7b512c206652dc4a636b649a682d56da@46.4.71.238
    CSeq: 102 INVITE
    User-Agent: Asterisk
    Remote-Party-ID: "13364052775" <sip:13364052775@46.4.71.238>;privacy=off;screen=n o
    Date: Sat, 18 Feb 2012 12:06:23 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 304

    v=0
    o=root 43122963 43122963 IN IP4 46.4.71.238
    s=Asterisk PBX 1.6.2.17.2
    c=IN IP4 46.4.71.238
    t=0 0
    m=audio 18188 RTP/AVP 8 0 18 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    <------------->
    --- (15 headers 14 lines) ---
    Sending to 46.4.71.238:5060 (no NAT)
    Using INVITE request as basis request - 7b512c206652dc4a636b649a682d56da@46.4.71.238
    Found peer '13364052775' for '13364052775' from 46.4.71.238:5060

    <--- Reliably Transmitting (no NAT) to 46.4.71.238:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 46.4.71.238:5060;branch=z9hG4bK3a399d8b;received=4 6.4.71.238
    From: "13364052775" <sip:13364052775@46.4.71.238>;tag=as2f59cbc4
    To: <sip:13363105403@sip.norlecom.com>;tag=as44cf8a5 9
    Call-ID: 7b512c206652dc4a636b649a682d56da@46.4.71.238
    CSeq: 102 INVITE
    Server: FPBX-2.10.0rc1(1.8.9.2)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46230d1a"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '7b512c206652dc4a636b649a682d56da@46.4.71.238' in 32000 ms (Method: INVITE)

    <--- SIP read from UDP:46.4.71.238:5060 --->
    ACK sip:13363105403@sip.norlecom.com SIP/2.0
    Via: SIP/2.0/UDP 46.4.71.238:5060;branch=z9hG4bK3a399d8b
    Max-Forwards: 70
    From: "13364052775" <sip:13364052775@46.4.71.238>;tag=as2f59cbc4
    To: <sip:13363105403@sip.norlecom.com>;tag=as44cf8a5 9
    Contact: <sip:13364052775@46.4.71.238>
    Call-ID: 7b512c206652dc4a636b649a682d56da@46.4.71.238
    CSeq: 102 ACK
    User-Agent: Asterisk
    Remote-Party-ID: "13364052775" <sip:13364052775@46.4.71.238>;privacy=off;screen=n o
    Content-Length: 0

    <------------->
    --- (11 headers 0 lines) ---
    Really destroying SIP dialog '52b3b9b1379a11612271bd8832fb9b1f@46.4.71.238' Method: ACK
    Really destroying SIP dialog '7b512c206652dc4a636b649a682d56da@46.4.71.238' Method: ACK

    [/QUOTE]

  2. #2
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    bonjour,
    Je ne suis pas un expert mais :

    SIP/2.0 401 Unauthorized
    Le problème part de la .

  3. #3
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    Il faut ajouter insecure=port,invite dans le definition de peer entrant.

  4. #4
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    Oui le problème c'etait bien ça mais peux tu m'expliquer.

  5. #5
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    Bonjour, en deux mots vous avez autorisé les appels entrants sur votre peer sans authentification.

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