[asterisk-users] Patton 5.3. How to get incoming calls ?

[Olivier]

Hi,

I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN
and SIP interfaces.
I'm using web interface at the moment.

Setup is:

ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone>

I can call from IP phone but can't receive any incoming call : I can't see
any SIP message coming in when a call comes in.

Previously, with 4.2 firmware, you just have to edit routing table binding
ISDN ports to SIP interface to get calls coming in but now with 5.3,
configuration process changed.
Here is an extract from my running config.
Any idea ?

Regards

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 192.168.100.254

  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 appels_vers_ISDN

  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 Otx2vJCEWP+8Bb6tqoGkwA== encrypted

location-service ASTERISK_SRV
  domain 1 192.168.100.254 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 192.168.100.254 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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