Error 401 Unauthorized en Elastix 2.0.0-14

Discussion in 'Elastix 2.x' started by andrecho239, Mar 10, 2010.

  1. andrecho239

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    Hola que tal!!!

    He descargado la version beta de Elastix 2.0 32 bits en una maquina virtual para efecto de pruebas y para ver como funciona con Asterisk 1.6. Tan pronto se instaló realizé un yum -y update y se actializó elastix a la version 2.0.0-14 y asterisk se actualizó a la version 1.6.2.1, en el proceso de actualización todo va bien, luego empiezo a configurar el PBX creando una extension sip y luego la registro en un softphone y se registra sin problema pero cuando hago una llamada hacia la extension 102 me devuelve error 401 Unauthorized, lo mismo sucede cuando me registro desde un telefono IP de Linksys pero si lo hago con una version distinta de elastix no tengo problema. el sip debug me muestra lo siguiente:

    <--- SIP read from UDP:192.168.1.112:64588 --->
    INVITE sip:102@192.168.1.114 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.112:64588;branch=z9hG4bK-d8754z-e03c684b71244160-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.112:64588>
    To: "102"<sip:102@192.168.1.114>
    From: "ext101"<sip:101@192.168.1.114>;tag=f20c6d6f
    Call-ID: MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: eyeBeam release 1013t stamp 43070
    Content-Length: 342

    v=0
    o=- 6 2 IN IP4 192.168.1.112
    s=CounterPath eyeBeam 1.5
    c=IN IP4 192.168.1.112
    t=0 0
    m=audio 19256 RTP/AVP 0 8 101
    a=alt:1 2 : 04/AvuPV ZcO7gHUi 192.168.56.1 19256
    a=alt:2 1 : NYSmC7r+ ADjk9hVu 192.168.1.112 19256
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    a=x-rtp-session-id:67E888E7DDB840968CF94D567B5D02FA

    <------------->
    --- (12 headers 12 lines) ---
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    Sending to 192.168.1.112 : 64588 (no NAT)
    Using INVITE request as basis request - MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    Found peer '101' for '101' from 192.168.1.112:64588

    <--- Reliably Transmitting (NAT) to 192.168.1.112:64588 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.112:64588;branch=z9hG4bK-d8754z-e03c684b71244160-1---d8754z-;received=192.168.1.112;rport=64588
    From: "Ext101"<sip:101@192.168.1.114>;tag=f20c6d6f
    To: "102"<sip:102@192.168.1.114>;tag=as4f93cb9d
    Call-ID: MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    CSeq: 1 INVITE
    Server: Asterisk PBX 1.6.2.1
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6216ed5d"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.' in 9024 ms (Method: INVITE)
    elastix*CLI>
    <--- SIP read from UDP:192.168.1.112:64588 --->
    ACK sip:102@192.168.1.114 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.112:64588;branch=z9hG4bK-d8754z-e03c684b71244160-1---d8754z-;rport
    To: "102"<sip:102@192.168.1.114>;tag=as4f93cb9d
    From: "ext101"<sip:101@192.168.1.114>;tag=f20c6d6f
    Call-ID: MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    CSeq: 1 ACK
    Content-Length: 0


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

    <--- SIP read from UDP:192.168.1.112:64588 --->
    INVITE sip:102@192.168.1.114 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.112:64588;branch=z9hG4bK-d8754z-f73fac3f1540734c-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.112:64588>
    To: "102"<sip:102@192.168.1.114>
    From: "ext101"<sip:101@192.168.1.114>;tag=f20c6d6f
    Call-ID: MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: eyeBeam release 1013t stamp 43070
    Authorization: Digest username="101",realm="asterisk",nonce="6216ed5d",uri="sip:102@192.168.1.114",response="167350dc0d7091547eeec24281ae1c44",algorithm=MD5
    Content-Length: 342

    v=0
    o=- 6 2 IN IP4 192.168.1.112
    s=CounterPath eyeBeam 1.5
    c=IN IP4 192.168.1.112
    t=0 0
    m=audio 19256 RTP/AVP 0 8 101
    a=alt:1 2 : 04/AvuPV ZcO7gHUi 192.168.56.1 19256
    a=alt:2 1 : NYSmC7r+ ADjk9hVu 192.168.1.112 19256
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    a=x-rtp-session-id:67E888E7DDB840968CF94D567B5D02FA

    <------------->
    --- (13 headers 12 lines) ---
    Sending to 192.168.1.112 : 64588 (NAT)
    Using INVITE request as basis request - MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    Found peer '101' for '101' from 192.168.1.112:64588
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 192.168.1.112:19256
    Looking for 102 in from-internal (domain 192.168.1.114)

    <--- Reliably Transmitting (NAT) to 192.168.1.112:64588 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 192.168.1.112:64588;branch=z9hG4bK-d8754z-f73fac3f1540734c-1---d8754z-;received=192.168.1.112;rport=64588
    From: "ext101"<sip:101@192.168.1.114>;tag=f20c6d6f
    To: "102"<sip:102@192.168.1.114>;tag=as4f93cb9d
    Call-ID: MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    CSeq: 2 INVITE
    Server: Asterisk PBX 1.6.2.1
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.' in 9024 ms (Method: INVITE)
    elastix*CLI>
    <--- SIP read from UDP:192.168.1.112:64588 --->
    ACK sip:102@192.168.1.114 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.112:64588;branch=z9hG4bK-d8754z-f73fac3f1540734c-1---d8754z-;rport
    To: "102"<sip:102@192.168.1.114>;tag=as4f93cb9d
    From: "ext101"<sip:101@192.168.1.114>;tag=f20c6d6f
    Call-ID: MjhjZTVkOTRlNjI5NDU2NTFhNzJlZWEwMGJmMGRhNTc.
    CSeq: 2 ACK
    Content-Length: 0


    Agradezco a quien me pueda ayudar con este problema.

    Gracias.
     
  2. jcastellanos

    Joined:
    Feb 10, 2009
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    lo intentaste antes de actualizar' que extraño tengo el mismo beta en produccion y no tengo ese problema
     

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