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<pre>

context cs switch

  routing-table called-e164 appels_provenance_ISDN
    route [0-9]+ dest-service ASTERISK_SRV
    route default dest-service ASTERISK_SRV

  routing-table called-uri appels_vers_ISDN
    route default dest-service isdnports

  mapping-table called-e164 to called-ip transfo
    map [0-9]+ to 10.20.20.1

  mapping-table called-e164 to called-uri transfo2

  interface isdn IF-PBX
    route call dest-table appels_provenance_ISDN

  interface isdn IF-PBX2
    route call dest-table appels_provenance_ISDN

  interface isdn IF-PBX3
    route call dest-table appels_provenance_ISDN

  interface isdn IF-PBX4
    route call dest-table appels_provenance_ISDN

  interface sip IF-ASTERISK
    bind context sip-gateway ASTERISK
    route call dest-table calls_from_SIP
    remote 10.20.20.1 5060


  service sip-location-service ASTERISK_SRV
    bind location-service ASTERISK_SRV
    mode hunt
    hunt-timeout 20

  service hunt-group isdnports
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF-PBX
    route call 2 dest-interface IF-PBX2
    route call 3 dest-interface IF-PBX3

context cs switch
  no shutdown

authentication-service patton
  realm 1 asterisk
  username patton password patton

location-service ASTERISK_SRV
  domain 1 10.20.20.1 5060
  domain 2 asterisk 5060

  identity-group default
  identity patton
    alias name patton

    authentication outbound
      authenticate 1 authentication-service patton username patton

    registration outbound
      registrar 10.20.20.1 5060
      proxy none
      lifetime 3600
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

    call outbound
      use profile tone-set default
      use profile voip default
      use profile sip default
      preferred-transport-protocol udp
      invite-transaction-timeout 32
      non-invite-transaction-timeout 32

    call inbound
      use profile tone-set default
      use profile voip default
      use profile sip default


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