Dial Pattern

Discussion in 'Elastix 2.x' started by wolverin0, Feb 5, 2011.

  1. wolverin0

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    Buenas... migré de Elastix 1.6 a 2.0, y la verdad que ya no se que hacer para poder marcar desde mi interno
    Mi proveedor VoIP maneja el dial plan, o sea que, por ejemplo, si llamo a un número de capital, y mi prefijo asignado es 5411, marco XXXX XXXX como número local, y x ej: 02477XXXXXX para una ciudad del interior.

    Antes, en Elastix dejaba el dial pattern default, si no mal recuerdo era 9|. y marcaba normalmente, y funcionaba la salida (marcaba 9005411XXXXXXXX y salia a 005411...)

    La recepción de llamadas es perfecta, entran por DID, incluso por internos, pero al querer llamar... nada... solo a internos con 9|INTERNO

    Que puede ser? Que cambió o me olvidé de configurar?
     
  2. wolverin0

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    Update: borré el Dial Plan que me predefine la plataforma del proveedor IP
    Ahora, en mi interno Elastix marco 95411XXXXXXXX y llama perfectamente
    Pero si marco 954+cualquier otro prefijo que no sea 11 (954341XXXXXXXX por ejemplo) me dice "your call cannot be completed as dialed.
    El dial pattern es 9|.
     
  3. fmvillares

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    claramente tu proveeedor no esta tomando los digitos o vos no se los pasas correctamente...
    alguna traza??? hablaste con el proveedor a ver que reciebn ellos??? porque la bola magica no esta por estos lados si no pasas info de logs etc real
     
  4. wolverin0

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    Acá pego el debug de la llamada.
    Sigue pasando lo mismo, llamo al 9100005 (el interno de mi proveedor IP) y funciona OK
    Llamo al interno 1002, de mi plataforma, y funciona OK
    Llamo a un número de PSTN con el dial completo (00542477NUMEROMOVIL) saliendo con el 9 (dial pattern 9|. en outbound routes) y nada, me da el error que figura en el debug.

    Espero respuesta, gracias desde ya

    Meto el debug en el spoiler:

    Code:
    m=audio 5062 RTP/AVP 8 0 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendrecv
    
    <------------->
    --- (14 headers 12 lines) ---
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
    Sending to 192.168.1.101 : 5060 (no NAT)
    Using INVITE request as basis request - 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    Found peer '1001' for '1001' from 192.168.1.101:5060
    
    <--- Reliably Transmitting (NAT) to 192.168.1.101:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23e20cf3070ff46;received=192.168.1.101;rport=5060
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>;tag=as0242d79d
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 120 INVITE
    Server: FPBX-2.7.0(1.6.2.13)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f2923c6"
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog '80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101' in 6400 ms (Method: INVITE)
    
    <--- SIP read from UDP:192.168.1.101:5060 --->
    ACK sip:090054247715660092@192.168.1.150 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23e20cf3070ff46;rport
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>;tag=as0242d79d
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 120 ACK
    Content-Length: 0
    
    
    <------------->
    --- (7 headers 0 lines) ---
    
    <--- SIP read from UDP:192.168.1.101:5060 --->
    INVITE sip:090054247715660092@192.168.1.150 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;rport
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 INVITE
    Contact: <sip:1001@192.168.1.101:5060>
    Authorization: Digest username="1001", realm="asterisk", nonce="6f2923c6", uri="sip:090054247715660092@192.168.1.150", response="d9c2238df805f8d9e55d5e23fd2073d3", algorithm=MD5
    Content-Type: application/sdp
    Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
    Max-Forwards: 70
    Supported: 100rel, replaces
    User-Agent: SIPPER for phoner
    P-Preferred-Identity: <sip:1001|1001@192.168.1.150>
    Content-Length:   257
    
    v=0
    o=- 4177321316 0 IN IP4 192.168.1.101
    s=SIPPER for phoner
    c=IN IP4 192.168.1.101
    t=0 0
    m=audio 5062 RTP/AVP 8 0 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendrecv
    
    <------------->
    --- (15 headers 12 lines) ---
    Sending to 192.168.1.101 : 5060 (NAT)
    Using INVITE request as basis request - 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    Found peer '1001' for '1001' from 192.168.1.101:5060
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 101
    Found audio description format PCMA for ID 8
    Found audio description format PCMU for ID 0
    Found audio description format GSM for ID 3
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 192.168.1.101:5062
    Looking for 090054247715660092 in from-internal (domain 192.168.1.150)
    list_route: hop: <sip:1001@192.168.1.101:5060>
    
    <--- Transmitting (NAT) to 192.168.1.101:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;received=192.168.1.101;rport=5060
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 INVITE
    Server: FPBX-2.7.0(1.6.2.13)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Contact: <sip:090054247715660092@192.168.1.150>
    Content-Length: 0
    
    
    <------------>
        -- Executing [090054247715660092@from-internal:1] ResetCDR("SIP/1001-00000018", "") in new stack
        -- Executing [090054247715660092@from-internal:2] NoCDR("SIP/1001-00000018", "") in new stack
        -- Executing [090054247715660092@from-internal:3] Progress("SIP/1001-00000018", "") in new stack
    Audio is at 192.168.1.150 port 17636
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    
    <--- Transmitting (NAT) to 192.168.1.101:5060 --->
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;received=192.168.1.101;rport=5060
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>;tag=as647736d1
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 INVITE
    Server: FPBX-2.7.0(1.6.2.13)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Contact: <sip:090054247715660092@192.168.1.150>
    Content-Type: application/sdp
    Content-Length: 286
    
    v=0
    o=root 1544643527 1544643527 IN IP4 192.168.1.150
    s=Asterisk PBX 1.6.2.13
    c=IN IP4 192.168.1.150
    t=0 0
    m=audio 17636 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    <------------>
        -- Executing [090054247715660092@from-internal:4] Wait("SIP/1001-00000018", "1") in new stack
    
    <--- SIP read from UDP:200.80.226.247:5060 --->
    NOTIFY sip:190.246.232.94:60130 SIP/2.0
    Via: SIP/2.0/UDP 200.80.226.247:5060;branch=0
    From: sip:keepalive@200.80.226.247;tag=59d9e51a
    To: sip:190.246.232.94:60130
    Call-ID: 744de4af-207e1437-145b62@200.80.226.247
    CSeq: 1 NOTIFY
    Event: keep-alive
    Content-Length: 0
    
    
    <------------->
    --- (8 headers 0 lines) ---
    
    <--- Transmitting (no NAT) to 200.80.226.247:5060 --->
    SIP/2.0 489 Bad event
    Via: SIP/2.0/UDP 200.80.226.247:5060;branch=0;received=200.80.226.247
    From: sip:keepalive@200.80.226.247;tag=59d9e51a
    To: sip:190.246.232.94:60130;tag=as1e83de74
    Call-ID: 744de4af-207e1437-145b62@200.80.226.247
    CSeq: 1 NOTIFY
    Server: FPBX-2.7.0(1.6.2.13)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0
    
    
    <------------>
        -- Executing [090054247715660092@from-internal:5] Playback("SIP/1001-00000018", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
        -- <SIP/1001-00000018> Playing 'silence/1.gsm' (language 'en')
        -- <SIP/1001-00000018> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
    
    <--- SIP read from UDP:192.168.1.101:5060 --->
    CANCEL sip:090054247715660092@192.168.1.150 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;rport
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 CANCEL
    Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
    Max-Forwards: 70
    User-Agent: SIPPER for phoner
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
    Sending to 192.168.1.101 : 5060 (NAT)
    
    <--- Reliably Transmitting (NAT) to 192.168.1.101:5060 --->
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;received=192.168.1.101;rport=5060
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>;tag=as647736d1
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 INVITE
    Server: FPBX-2.7.0(1.6.2.13)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    upported: replaces, timer
    Content-Length: 0
    
    
    <------------>
    
    <--- Transmitting (NAT) to 192.168.1.101:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;received=192.168.1.101;rport=5060
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>;tag=as647736d1
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 CANCEL
    Server: FPBX-2.7.0(1.6.2.13)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0
    
    
    <------------>
      == Spawn extension (from-internal, 090054247715660092, 5) exited non-zero on 'SIP/1001-00000018'
        -- Executing [h@from-internal:1] Macro("SIP/1001-00000018", "hangupcall") in new stack
        -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000018", "1?noautomon") in new stack
        -- Goto (macro-hangupcall,s,3)
        -- Executing [s@macro-hangupcall:3] NoOp("SIP/1001-00000018", "TOUCH_MONITOR_OUTPUT=") in new stack
        -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1001-00000018", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,7)
        -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1001-00000018", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,10)
        -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1001-00000018", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,12)
        -- Executing [s@macro-hangupcall:12] Hangup("SIP/1001-00000018", "") in new stack
      == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1001-00000018' in macro 'hangupcall'
      == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-00000018'
    
    <--- SIP read from UDP:192.168.1.101:5060 --->
    ACK sip:090054247715660092@192.168.1.150 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK80ef1f387e32e011b23f20cf3070ff46;rport
    From: "Phoner" <sip:1001@192.168.1.150>;tag=4051173607
    To: <sip:090054247715660092@192.168.1.150>;tag=as647736d1
    Call-ID: 80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101
    CSeq: 121 ACK
    Content-Length: 0
    
    
    <------------->
    --- (7 headers 0 lines) ---
    Really destroying SIP dialog '80EF1F38-7E32-E011-B23D-20CF3070FF46@192.168.1.101' Method: ACK
    
    <--- SIP read from UDP:192.168.1.101:30488 --->
    
     
  5. fmvillares

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    no podes marcar asi un celular....desde el extranjero los celulares a argentina se marcan con 9 despues del 54 es decir en tu acso y con la regla generica 9 54 9 341 num celular sin 15
     
  6. wolverin0

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    a ver...
    una cosa es el dial plan de mi proveedor y otro es el mio
    para marcar desde un softphone, mi proveedor toma este tipo de marcado (0054247715660066) por ejemplo, para un celular de PERGAMINO
    si fuera uno de rosario en el softphone marcas: 00543411566006600

    en la plataforma, si salgo con 900543411566006600 me tira ese error del debug.
    se entiende mas o menos?

    edit: no me preguntes porque, pero marco 90247715660092 y anda... en ningún lugar está definido el 0054, pero así anda........
    y si marco afuera, tengo que marcar 959XXXXXXX (para Mexico por ejemplo)

    sigo sin entender quien o que define el 54 como default.....
     
  7. fmvillares

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    tu proveedor no te esta dando las reglas exactas de como marcar por eso el local de pergamino sale bien no tenes que poner el 54...pasame el dato de esa empresa asi la voy tachando de la lista
     
  8. wolverin0

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    no hay que marcar 54 para ningún numero de argentina
    solo marcar 9 (outbound route) y el prefijo del pais que no sea Argentina
    parece que 54 lo marca por defecto.
     

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