Configurar o encaminhamento de chamadas de saída através de gateways Cisco

Este artigo explica como deve configurar o seu gateway Cisco CME (ou CUBE) para enviar chamadas para o nosso sistema

Guia passo a passo

Primeiramente, obtenha seu Nome de usuário SIP e Senha com a MultiTEL.

    Pode criar um par de nome de utilizador/palavra-passe SIP na nossa secção Ligações SIP (Trunks):
    https://multitel.net/trunks/accounts


    Também pode adicionar aqui o seu endereço IP:
    https://multitel.net/trunks/ips

    1. Inicie sessão no seu gateway de voz Cisco e use uma configuração semelhante à mostrada abaixo (modifique onde necessário)
      Quando terminar, teste suas chamadas a partir do console Cisco CLI usando o seguinte comando:

      csim start <enter_here_your_phone_number>

      Por exemplo, se seu número de telefone fosse 18886868581, este comando CLI apareceria assim

      csim start 18886868581

      Este comando originaria uma chamada telefônica do seu gateway de voz Cisco através do serviço da MultiTEL de entrega de chamadas de saída ao destino e seria usado para testar sua configuração.

      Além disso, você sempre pode usar os dois comandos a seguir:

      show call active voice brief
      show call history voice brief

      Eles indicariam quais dial-peers foram tentados (procure pela palavra-chave pid:) e por que as chamadas falharam.
    #This section will vary depending on your configuration.
    #We do suggest setting the variable min-se which controls session timeout to 900 seconds.

    voice rtp send-recv
    voice service voip
    !
      allow-connections sip to sip
    sip
        min-se 900
    !
    voice class codec 711
     codec preference 1 g711ulaw
     codec preference 2 g711alaw
    !
    voice class codec 729
    codec preference 1 g729r8 bytes 40
    codec preference 2 g729br8 bytes 40
    !
    voice class codec 700
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    codec preference 3 g729r8 bytes 40
    codec preference 4 g729br8 bytes 40
    !
    ! We support E164 format only. US/CA numbers have to be dialed in full 11 digit format.
    ! Use this template if you wish to convert from 7 or 10 to 11 digits.
    ! Replace NPA with your local area code when using 7 digit numbers
    !
    ! translation rule definition below
    !
    voice translation-rule 1
      rule 1 /^9\(1[2-9]..[2-9]......\)$/ /\1/     ! send out if using 9 dial-out prefix and 11 digit format ; strip 9 
      rule 2 /^9\([2-9]......\)$/ /1NPA\1/         ! send out if using 9 dial-out prefix and 10 digit format ; strip 9 and 1 country code
      rule 3 /^\(.......\)$/ /1NPA\1/              ! 7 digit format, add 1 and the "NPA" and send out (don't forget to replace NPA with your local prefix)
    rule 4 /^011\(.*)$/ /1\1/ ! strip 011, send call out
    rule 5 /^00\(.*)$/ /1\1/ ! strip 00 , send call out

    !
    ! translation profile using the translation rules above

    !

    voice translation-profile multitel
    translate called 1
    !
    ! Outbound dial-peer definition
    !

    dial-peer voice 1 voip
    permission term
    description Termination towards MultiTEL translation-profile outgoing multitel destination-pattern .T ! or replace with 1[2-9]..[2-9]......T if only sending US calls session protocol sipv2 session target dns:sbc-us.multitel.net ! -- replace here with the SBC which is closest to you
     session transport udp  ! or "tcp", depending how your account is configured on MultiTEL
    voice-class codec 711 dtmf-relay rtp-nte no vad
    !
    ! Inbound dial-peer definition
    !
    dial-peer voice 2 voip
    permission orig
    description Origination from MultiTEL
    huntstop
    session transport udp ! or "tcp", depending how your account is configured on MultiTEL
    incoming called-number .T
    dtmf-relay rtp-nte
    voice-class codec 711
    fax-relay ecm disable
    fax-rate 9600
    fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through g711ulaw
    no vad
    !

    !
    ! IMPORTANT: please make sure you DO HAVE the below config for security purposes
    ! this will allow calls to your gateway only from authorized IP addresses
    voice source-group myawesomecompany
    access-list 99
    disconnect-cause call-reject
    ! add here inbound translation profile if you need to
    !
    ! access-list definition - list here your IP addresses
    ! default action is deny
    !
    access-list 99 permit 192.168.1.0 0.0.0.255
    ! You may have to add MultiTEL's origination IP addresses here to allow inbound DID/origination calls from MultiTEL
    ! Please consult the article below for complete (and usually up to date) list of IP addresses we use to send DID/origination calls
    ! https://multitel.net/docs/view/161/knowledgebase/
    ! Sample entries below:
    access-list 99 permit X.Y.Z.0 0.0.0.255
    access-list 99 permit A.B.C.0 0.0.0.255

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