Configurazione Patton Smartnode 4112JO 2 FXO (unico trunk sip)

Questo documento spiega come configurare il Patton Smartnode 4112JO 2 FXO e farlo funzionare con FreePBX / Asterisk.

PREMESSA:
Abbiamo realizzato e messo online già da novembre 2014 il sito http://www.patton-smartnode-configuration.com/ dal quale si possono configure quasi tutti i modelli di smartnode con software 6.X (FXO, BRI, PRI e FXS).

Questo un esempio di file di configurazione per il Patton Smartnode 4112JO 2 FXO:
DATI DI PARTENZA:
- IP PBX: 192.168.1.100
- IP PATTON: 192.168.1.101/255.255.255.0
- GATEWAY: 192.168.1.1

NECESSITA':
- configurazione con 1 trunk sip

#----------------------------------------------------------------
#
# http://www.patton-smartnode-configuration.com
# Asterweb Srl - Milan - Italy
# info@asterweb.org
#
# Firmware 6.x
# Generated configuration file 2015-07-23 07:03:36
#
#----------------------------------------------------------------

cli version 3.20
clock local default-offset +01:00
dns-client server 8.8.8.8
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.1.100 port 123 version 4
system hostname SN4112

system
ic voice 0

profile ppp default

profile call-progress-tone IT_Dialtone
play 1 200 425 -12
pause 2 200
play 3 600 425 -12
pause 4 1000
play 5 200 425 -12
pause 6 200
play 7 600 425 -12
pause 8 1000
play 9 200 425 -12
pause 10 200

profile call-progress-tone IT_Alertingtone
play 1 1000 425 -12
pause 2 4000
play 3 1000 425 -12
pause 4 4000
play 5 1000 425 -12
pause 6 4000

profile call-progress-tone IT_Busytone
play 1 500 425 -12
pause 2 500
play 3 500 425 -12
pause 4 500
play 5 500 425 -12
pause 6 500

profile tone-set default

profile tone-set IT
map call-progress-tone dial-tone IT_Dialtone
map call-progress-tone ringback-tone IT_Alertingtone
map call-progress-tone busy-tone IT_Busytone
map call-progress-tone release-tone IT_Busytone
map call-progress-tone congestion-tone IT_Busytone

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
codec 3 g729 rx-length 20 tx-length 20
fax transmission 1 relay t38-udp

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router
interface LAN
ipaddress 192.168.1.101 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
route 0.0.0.0 0.0.0.0 192.168.1.1 0

context cs switch
digit-collection timeout 3
no digit-collection terminating-char
national-prefix 0
international-prefix 00

routing-table called-e164 INBOUND_0
route default dest-interface IF_SIP_SERVICE_0 MAP_DID_0

routing-table called-e164 INBOUND_1
route default dest-interface IF_SIP_SERVICE_1 MAP_DID_1

routing-table called-e164 OUTBOUND
route default dest-service BOUND_PSTN

mapping-table called-e164 to called-e164 MAP_DID_0
map default to 021234567

mapping-table called-e164 to called-e164 MAP_DID_1
map default to 021234568

interface isdn IF_PSTN_0
route call dest-table INBOUND_0
loop-break-duration min 60 max 5000
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing
use profile tone-set IT

interface isdn IF_PSTN_1
route call dest-table INBOUND_1

loop-break-duration min 60 max 5000
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing
use profile tone-set IT

interface sip IF_SIP_SERVICE
bind context sip-gateway GW_SIP
route call dest-table OUTBOUND
remote 192.168.1.100
early-disconnect
privacy

service hunt-group BOUND_PSTN
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
drop-cause user-busy
route call 1 dest-interface IF_PSTN_0
route call 2 dest-interface IF_PSTN_1

context cs switch
no shutdown

authentication-service AUTH_SVC
username PattonUser1 password kivqPdVFgwhvMgy

location-service LOCATION_SVC
domain 1 192.168.1.100

identity PattonUser1
authentication outbound
authenticate 1 authentication-service AUTH_SVC username PattonUser1

registration outbound
registrar 192.168.1.100
lifetime 3600
register auto


context sip-gateway GW_SIP
interface IF_GW_SIP
bind interface LAN context router port 5060

context sip-gateway GW_SIP
bind location-service LOCATION_SVC
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface LAN router
no shutdown

port fxo 0 0
encapsulation cc-fxo
bind interface IF_PSTN_0 switch
no shutdown

port fxo 0 1
encapsulation cc-fxo
bind interface IF_PSTN_1 switch
no shutdown



Lato FreePBX

Creare ora il FASCIO SIP su FreePBX.
Selezionare "Connectivity" => "Fasci" e quindi cliccare (a destra) su "Aggiungi Fascio SIP".

FASCIO:
  • Identificativo Chiamante in uscita: [il numero di telefono della prima FXO o il numero che dovrà comparire in uscita]
  • Maximum Channels: 1 (il canale della FXO)
  • Impostazioni in uscita
    • Nome fascio: PattonUser1
    • Dati PEER:
      username=PattonUser1
      secret=kivqPdVFgwhvMgy
      type=friend
      host=dynamic
      port=5060
      nat=no
      directmedia=no
      defaultip=192.168.1.101
      context=from-pstn
      insecure=port,invite
      dtmfmode=rfc2833
      qualify=yes
      disallow=all
      allow=alaw&ulaw
  • Impostazioni in entrata (tutto vuoto)


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