integracion de elastix y callmanager cisco v7

Discussion in 'Elastix 2.x' started by netillo123X, May 18, 2010.

  1. netillo123X

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    hola qu etal tengo un call manager de cisco el cual maneja extensiones con SCCP

    la ip es 192.168.211.2

    del otro lado tengo un pbx elastix version 1.6 con la ip 192.168.4.201
    en este pbx tengo ext. sip


    necesito crear una troncal sip entre elastix y callmanager


    mis datos del peer son



    host=192.168.211.2
    secret=digestpbx
    type=friend
    canreivite=yes
    qualify=yes
    disallow=all
    allow=ulaw
    context=from-internal
    nat=yes


    los datos de user details

    type=friend
    canreivite=yes
    qualify=yes
    disallow=all
    allow=ulaw
    context=from-internal
    nat=yes



    en mi pbx elastix hago una ruta saliente 9|. saliendo por mi truk sip

    een el call manager tengo una ext 5001 y en elastix tengo un 7004


    este es el sip set debug

    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/7004-08ebbe80", "SIP/callman01/5001|300|") in new stack
    Audio is at 192.168.40.201 port 10530
    Adding codec 0x4 (ulaw) to SDP
    Reliably Transmitting (NAT) to 192.168.211.2:5060:
    INVITE sip:5001@192.168.211.2 SIP/2.0
    Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
    From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
    To: <sip:5001@192.168.211.2>
    Contact: <sip:7004@192.168.40.201>
    Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Tue, 18 May 2010 19:08:02 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 186

    v=0
    o=root 2782 2782 IN IP4 192.168.40.201
    s=session
    c=IN IP4 192.168.40.201
    t=0 0
    m=audio 10530 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    ---
    -- Called callman01/5001
    elastix*CLI>
    <--- SIP read from 192.168.211.2:5060 --->
    SIP/2.0 100 Trying
    Date: Tue, 18 May 2010 19:27:46 GMT
    From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
    Allow-Events: presence
    Content-Length: 0
    To: <sip:5001@192.168.211.2>
    Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
    Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
    CSeq: 102 INVITE


    <------------->
    --- (9 headers 0 lines) ---
    elastix*CLI>
    <--- SIP read from 192.168.211.2:5060 --->
    SIP/2.0 401 Unauthorized
    Date: Tue, 18 May 2010 19:27:46 GMT
    From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
    Allow-Events: presence
    WWW-Authenticate: Digest realm="StandAloneCluster", nonce="ZdxPuvBXxOpByslCQevOuvo6cLF+lGaO", algorithm=MD5
    Content-Length: 0
    To: <sip:5001@192.168.211.2>;tag=1359530374
    Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
    Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
    CSeq: 102 INVITE


    <------------->
    --- (10 headers 0 lines) ---
    Transmitting (NAT) to 192.168.211.2:5060:
    ACK sip:5001@192.168.211.2 SIP/2.0
    Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
    From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
    To: <sip:5001@192.168.211.2>;tag=1359530374
    Contact: <sip:7004@192.168.40.201>
    Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0


    ---
    -- SIP/callman01-08ebfea0 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/7004-08ebbe80", "s-CONGESTION|1") in new st


    me dice que todas las lineas estan ocupadas

    necesito ayuda


    saludos
     
  2. garcia.ronald.d

    Joined:
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    Hola,

    Ya pudiste realizar la integracion?

    He tratado de todo y nada, llamo desde mi softphone registrado en asterisk y me dice "todas las líneas están ocupadas"

    Cualquier ayuda se agradece
     
  3. garcia.ronald.d

    Joined:
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    Hola,

    Ya pudiste realizar la integracion?

    He tratado de todo y nada, llamo desde mi softphone registrado en asterisk y me dice "todas las líneas están ocupadas"

    Cualquier ayuda se agradece
     

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