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Asterisk como gateway

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  • Asterisk como gateway

    Hola buena noches
    Quiero integrar a asterisk (elastix) como gateway y que este equipo contenga una tarjeta E1

    y atras del PBX se encuentre un NBX

    Se puede realizar esto

    Tratare de realizar esto esperando suayuda en caso de complicarse

    Saluds

  • #2
    Si es posible. Solo asegúrate de que el viejo pbx igual use una E1 y listo.

    Comentario


    • #3
      Hola buen dia ya crealize el trunk con el conmutador
      y estos son algunos datos

      El esquema es

      E1 --- ASTERISK PBX ------ ==========3COM- --- PHONES 3COM
      TRUNK SIP


      sip_additional.conf

      [pbx]
      disallow=all
      allow=g729
      dtmfmode=rfc2833
      context=incoming-pbx
      host=10.10.200.253
      type=peer

      y mi diaplan es el siguiente

      ivr

      [incoming-pbx]

      exten => 3000,1,Wait(1) ;espera un segunfo
      exten => 3000,2,Set(TIMEOUT(digit)=7) ;
      exten => 3000,3,Background(bienvenida)
      exten => 3000,n,Background(ext-or-zero)

      exten => 3000,n,WaitExten()
      exten => 0,1,Dial(SIP/3061@10.10.200.253:5060,30)
      exten => _3XXX,1,Dial(SIP/${EXTEN}@10.10.200.253:5060,30)
      exten => _3XXX,2,Hangup



      Salida


      [sip-pbx]
      exten => _9XXXXXXX,1,Dial(DAHDI/g0/${EXTEN:1},90,Tt)
      exten => _9XXXXXXX,n,hangup()




      El detalle ahora es que puedo tano recibir y realizar llamadas desde las extensiones conectadas al conmutador 3com

      1.- Pero cuando se trasnfiere una llamada no se escuhcha mada del MOH musiconhold
      2.- las llamadas que se reciben
      si los telefonos 3com cuelgan en el tel del llamnte no se escucha que xcuelguen

      es decir no hay hangup
      pero si cuelga el llamnet no hay problema

      Y si se realiza una llamada de los tels 3com hacia afuera no hay problema si hay hangu




      este es un degug de una netrada de una llamada
      entrante

      ---
      [Feb 22 10:13:17] VERBOSE[30996] app_dial.c: -- Called SIP/3166@10.10.200.253:5060
      [Feb 22 10:13:17] VERBOSE[25194] chan_sip.c:
      <--- SIP read from UDP:10.10.200.253:5060 --->
      SIP/2.0 100 Trying
      From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
      To: <sip:3166@10.10.200.253:5060>
      Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
      Cseq: 102 INVITE
      Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK33c3146d;rport=50 60
      Content-Length: 0

      <------------->
      [Feb 22 10:13:17] VERBOSE[25194] chan_sip.c: --- (7 headers 0 lines) ---
      [Feb 22 10:13:17] VERBOSE[25194] chan_sip.c:
      <--- SIP read from UDP:10.10.200.253:5060 --->
      SIP/2.0 180 Ringing

      From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
      To: <sip:3166@10.10.200.253:5060>
      Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
      Cseq: 102 INVITE
      Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK33c3146d;rport=50 60
      Date: Fri, 22 Feb 2013 16:13:11 GMT
      Contact: sip:10.10.200.253
      User-Agent: 3Com-SIP-Phone/v2.6.0 (sipXphone) (VxWorks)
      Accept-Language: en
      Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY
      Supported: replaces
      Content-Length: 0
      <------------->
      [Feb 22 10:13:17] VERBOSE[25194] chan_sip.c: --- (13 headers 0 lines) ---
      [Feb 22 10:13:17] VERBOSE[25194] chan_sip.c: list_route: hop: <sip:10.10.200.253>
      [Feb 22 10:13:17] VERBOSE[30996] app_dial.c: -- SIP/10.10.200.253:5060-0000020f is ringing
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c:
      <--- SIP read from UDP:10.10.200.253:5060 --->
      SIP/2.0 200 OK
      From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
      To: <sip:3166@10.10.200.253:5060>;tag=1361549593
      Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
      Cseq: 102 INVITE
      Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK33c3146d;rport=50 60
      Content-Type: application/sdp
      Content-Length: 225
      P-Asserted-Identity: "SITE Central"<sip:3166@10.10.200.253>
      Contact: "3ComCallProcessor"<sip:3ComCallProcessor@10.10.20 0.253:5060;transport=UDP>
      Date: Fri, 22 Feb 2013 16:13:13 GMT
      Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY
      User-Agent: 3Com-SIP-Phone/v2.6.0 (sipXphone) (VxWorks)
      Accept-Language: en
      Supported: replaces

      v=0
      o=- 0 0 IN IP4 10.10.200.138
      s=NIP Session
      c=IN IP4 10.10.200.138
      t=0 0
      m=audio 8048 RTP/AVP 18 101
      a=SilenceSuppff
      a=rtpmap:18 G729/8000
      a=fmtp:18 annexb=no
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-16
      <------------->
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: --- (15 headers 11 lines) ---
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found RTP audio format 18
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found RTP audio format 101
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found audio description format G729 for ID 18
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found audio description format telephone-event for ID 101
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)


      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Peer audio RTP is at port 10.10.200.138:8048
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: list_route: hop: <sip:3ComCallProcessor@10.10.200.253:5060;transpor t=UDP>
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: set_destination: Parsing <sip:3ComCallProcessor@10.10.200.253:5060;transpor t=UDP> for address/port to send to
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: set_destination: set destination to 10.10.200.253:5060
      [Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Transmitting (NAT) to 10.10.200.253:5060:
      ACK sip:3ComCallProcessor@10.10.200.253:5060;transport =UDP SIP/2.0^M
      Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK4ff04ab2;rport^M
      Max-Forwards: 70^M
      From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b^M
      To: <sip:3166@10.10.200.253:5060>;tag=1361549593^M
      Contact: <sip:PBX AST@10.10.200.210:5060>^M
      Call-ID: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0^M
      CSeq: 102 ACK^M
      User-Agent: Asterisk PBX 1.8.11.0^M
      Content-Length: 0^M
      [Feb 22 10:13:19] VERBOSE[30996] app_dial.c: -- SIP/10.10.200.253:5060-0000020f answered DAHDI/1-1
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: bits changed in chan 1
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - Bits changed from 0x00 to 0x08
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - CAS Rx << [CLEAR FORWARD] 0x08
      [Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: Chan 1 - Far end disconnected. Reason: Normal Clearing
      [Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: MFC/R2 call disconnected on channel 1
      [Feb 22 10:13:36] VERBOSE[30996] pbx.c: -- Executing [h@incoming-pbx:1] Hangup("DAHDI/1-1", "") in new stack
      [Feb 22 10:13:36] VERBOSE[30996] features.c: == Spawn extension (incoming-pbx, h, 1) exited non-zero on 'DAHDI/1-1'
      [Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: Scheduling destruction of SIP dialog '1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:50 60' in 32000 ms (Method: INVITE)
      [Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: set_destination: Parsing <sip:3ComCallProcessor@10.10.200.253:5060;transpor t=UDP> for address/port to send to
      [Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: set_destination: set destination to 10.10.200.253:5060
      [Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: Reliably Transmitting (NAT) to 10.10.200.253:5060:
      BYE sip:3ComCallProcessor@10.10.200.253:5060;transport =UDP SIP/2.0^M
      Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK45c1d39f;rport^M
      Max-Forwards: 70^M
      From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b^M
      To: <sip:3166@10.10.200.253:5060>;tag=1361549593^M
      Call-ID: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0^M
      CSeq: 103 BYE^M
      User-Agent: Asterisk PBX 1.8.11.0^M


      ---
      [Feb 22 10:13:36] VERBOSE[30996] pbx.c: == Spawn extension (incoming-pbx, 3166, 1) exited non-zero on 'DAHDI/1-1'
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: disconnecting MFC/R2 call on chan 1
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: ast cause 16 resulted in openr2 cause 6/Normal Clearing
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - Call ended
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - CAS Tx >> [IDLE] 0x08
      [Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - CAS Raw Tx >> 0x09
      [Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: MFC/R2 call end on channel 1
      [Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: -- Hungup 'DAHDI/1-1'
      [Feb 22 10:13:36] VERBOSE[25194] chan_sip.c:
      <--- SIP read from UDP:10.10.200.253:5060 --->
      SIP/2.0 100 Trying
      From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
      To: <sip:3166@10.10.200.253:5060>;tag=1361549593
      Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
      Cseq: 103 BYE
      Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK45c1d39f;rport=50 60
      Content-Length: 0

      <------------->


      Espero me puedan ayudara resolver el problema

      Saludos

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