FreePBX (13.0.188.8) & Patton 4120 No inbound

Hello to everyone,
I have a new installation of FreePBX 13.0.188.8 connected with a Patton 4120 2xBRI (SN4120/2BIS4V/EUI). I am mostly familiar with pci cards and not gateways, but this is a virtual enviroment and i couldn’t use a pci card. I searched on the internet for something that might be usefull with a little luck.
Anyway I achieved to do some configuration at Patton and the outbound calls are working fine. But, incoming calls aren’t . This is not a freepbx issue cause the inbound route routes to hold and when i make an inbound call it doesn’t route me to hold, it keeps ringing somewhere.

Any help or reference is appreciated.

This is my current Patton configuration

#----------------------------------------------------------------#

SN4120/2BIS4V

R6.9 2016-09-01 H323 SIP

2016-10-13T10:43:06

Generated configuration file

#---------------------------------------------

cli version 3.20
clock local default-offset +03:00
dns-client server 192.168.1.1
webserver port 80 language en
sntp-client
sntp-client server primary 2.gr.pool.ntp.org port 123 version 4
sntp-client server secondary 3.gr.pool.ntp.org port 123 version 4
sntp-client poll-interval 600
system hostname patton

system

ic voice 0
low-bitrate-codec g729

system
clock-source 1 bri 0 0
clock-source 2 bri 0 1

profile ppp default

profile tone-set default

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface eth0
ipaddress 192.168.1.149 255.255.255.0
icmp router-discovery
icmp redirect accept
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context cs switch

routing-table called-e164 RT_ISDN_TO_SIP
route .T dest-interface IF_SIP

interface isdn IF_ISDN_00
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit

interface isdn IF_ISDN_01
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit

interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-service SRV_HG
remote 192.168.1.150
address-translation outgoing-call from-header user-part fix Patton4120 host-part call

service hunt-group SRV_HG
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_ISDN_00
route call 2 dest-interface IF_ISDN_01

context cs switch
no shutdown

authentication-service AUTH_SRV
username Patton4120 password encrypted

location-service SER_LOC
domain 1 192.168.1.150
match-any-domain

identity Patton4120

authentication outbound
  authenticate 1 authentication-service AUTH_SRV username Patton4120

registration outbound
  registrar 192.168.1.150
  lifetime 3600
  register auto
  retry-timeout on-system-error 10
  retry-timeout on-client-error 10
  retry-timeout on-server-error 10

context sip-gateway GW_SIP

interface IF_GWSIP
bind interface eth0 context router port 5060

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

port ethernet 0 0
medium auto
encapsulation ip
bind interface eth0 router
no shutdown

port bri 0 0
clock auto
encapsulation q921

q921
permanent-layer2
uni-side auto
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_ISDN_00 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
permanent-layer2
uni-side auto
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_ISDN_01 switch

port bri 0 1
no shutdown

Hi, this is my working configuration for patton 4120 (but i have only 1 bri port)

cli version 3.20
clock local default-offset +02:00
dns-client server 172.16.150.10
webserver port 80 language en
sntp-client server primary 172.16.150.10 port 123 version 4
system hostname pattonbri-ISDN

system

  ic voice 0

system
  clock-source 1 bri 0 0

profile napt NAPT_WAN

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

profile call-progress-tone IT_Alertingtone
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_Busytone
  play 1 500 425 -12
  pause 2 500

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
  response-preferred-codec g711alaw64k

profile pstn default

profile sip default
  no autonomous-transitioning

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface LAN
    ipaddress 192.168.70.31 255.255.255.0
    icmp redirect accept
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context ip router
  route 0.0.0.0 0.0.0.0 192.168.70.19 0

context cs switch
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_OUT
    route .% dest-service SER_HG_ISDN speech

  routing-table called-e164 RT_IN
    route .%T3 dest-interface IF_PBX

  mapping-table itc to itc speech
    map default to speech

  interface isdn IF_ISDN0
    route call dest-table RT_IN
    use profile tone-set IT

  

  interface sip IF_PBX
    bind context sip-gateway GW_PBX
    route call dest-table RT_OUT
    remote 192.168.70.30 5060
    early-disconnect

  service hunt-group SER_HG_ISDN
    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_ISDN0

context cs switch
  no shutdown

authentication-service AS_PBX
  realm 1 asterisk
  username pattonbri password dZ8edXkjFnM= encrypted

location-service LS_PBX
  domain 1 192.168.70.30 5060

  identity pattonbri

    authentication outbound
      authenticate 1 authentication-service AS_PBX username pattonbri

    registration outbound
      registrar 192.168.70.30 5060
      register auto

context sip-gateway GW_PBX

  interface IF_PBX
    bind interface LAN context router port 5060

context sip-gateway GW_PBX
  bind location-service LS_PBX
  no shutdown

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

port bri 0 0
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_ISDN0 switch

port bri 0 0
  no shutdown

And this is the trunk configuration in freepbx

username=pattonbri
type=friend
secret=_**password**_
qualify=yes
insecure=port,invite
host=dynamic
dtmfmode=rfc2833
defaultip=192.168.70.31

192,168.70.31 is the patton, 192.168.70.30 is freepbx.

Update: The configuration seems to be right. There is a compatibility issue with the NT device that provided by the carrier and the Patton 4120. I tried an other NT device and is working fine.

@perrcla Does the caller id display at you ip phones for inbound calls?

oh yes !

UPDATE: The issue was that my ISDN provider doesn’t send the DID. For this to work on Patton needed a different configuration as mentioned here http://support.patton.com/index.php?/Knowledgebase/Article/View/43/60/how-to-configure-a-smartnode-if-calls-does-not-contain-a-called-party-number

Ciao sto avendo il tuo stesso problema ma il link ke hai indicato non è più raggiungibile puoi postare qui il template e la configurazione per favore ? Grazie

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