SIP/BRI gateway - Patton SN4554 in the UK

[ncg]

Hi folks,

I've just picked up a Patton SN4554 cheap on eBay and will use it to connect PIAF and my BT ISDN2 (aka British Telecom BRI) lines. I have ISDN2 to carry 4 simultaneous voice calls.

Has anyone done this in the UK who can let me have pointers and/or a secure configuration file for the Patton?

I'll post a how-to once the system is running.

Thanks,

Nigel

Edit: There are some interesting pointers at http://www.trixbox.org/forums/trixbox-forums/trunks/patton-4638-sip-trunk - especially if you speak French!

 

[claudiopizzillo]

Start from this: Italian ISDN

clock local offset +01:00
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 193.204.114.105 port 123 version 4 
sntp-client server secondary 193.204.114.233 port 123 version 4
sntp-client poll-interval 36 
sntp-client local-clock-offset 
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 call-progress-tone IT_Dialtone

  play 200 425 -12
  no play 200
  play 600 425 -12
  no play 1000

profile call-progress-tone IT_Alertingtone

  play 1000 425 -12
  no play 4000

profile call-progress-tone IT_Busytone

  play 500 425 -12
  no play 500

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 congestion-tone IT_Congestion
  map call-progress-tone release-tone IT_Busytone

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


profile voip VOIP
  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 no-silence-suppression

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface WAN
    ipaddress 192.168.1.250 255.255.255.0
    no napt-inside
  interface LAN
    ipaddress 192.168.2.1 255.255.255.0
    no napt-inside

context cs switch

  digit-collection timeout 4
  no digit-collection terminating-char
  address-completion timeout 5
  national-prefix 0
  international-prefix 00

  routing-table called-e164 FROM_SIP
    route .%T dest-service TELECOM_GROUP MAPPING
    use profile tone-set IT

  routing-table called-e164 ROUTING
    route .%T dest-interface IF_ASTERISK MAPPING

  mapping-table itc to itc MAPPING
    map default to speech

  interface isdn IF_1
    route call dest-table ROUTING
    use profile tone-set IT

  interface isdn IF_2
    route call dest-table ROUTING
    use profile tone-set IT
    isdn-date-time

  interface isdn IF_DEV0

  interface sip IF_ASTERISK
    bind gateway GW_ASTERISK
    service default
    route call dest-table FROM_SIP
    early-disconnect
    remote-party-id called-party
    remote-party-id calling-party

  service hunt-group TELECOM_GROUP
    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
    drop-cause destination-out-of-order
    route call 1 dest-interface IF_1
    route call 2 dest-interface IF_1
    route call 3 dest-interface IF_2
    route call 4 dest-interface IF_2

context cs switch
  no shutdown

gateway sip GW_ASTERISK
  bind interface WAN router

  service default
    domain 192.168.1.240
    realm 192.168.1.240
    authentication 1001 password 1234
    default-server 192.168.1.240 loose-router
    registrar 192.168.1.240 5060
    user 1001
    session-timer 1800

gateway sip GW_ASTERISK
  no shutdown

port ethernet 0 0
  medium 10 half
  bind interface WAN router
  no shutdown

port ethernet 0 1
  medium 10 half
  bind interface LAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_1 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_2 switch

port bri 0 1
  no shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side net
      bchan-number-order ascending

port bri 0 2
  shutdown

 

[ncg]

Hi Claudio,

Thank you for this: it is really helpful for setting up the Patton.

How do you set up the connection between the Patton and PIAF in FreePBX? Do you "Add Sip Trunk", and what are the Outgoing and Incoming Settings, and the User Context?

Is the Patton a peer or a friend?

Also, which version of SmartWare firmware is your Patton running. The syntax seems a little different on mine (v5.3)

Thank you again for your help.

Nigel

 

[ncg]

We found an extremely helpful config at http://fonality.com/trixbox/forums/...bound-cid-not-being-set-patton-smartnode-4554 which pretty much rounded off this issue.

Two problems remain:

- DTMF: it seems to be sent multiple times, and there is an audible a stutter in the audio stream that follows DTMF tones being sent, that lasts about 5 seconds for each DTMF tone pressed. (ie a 4 digit PIN gives 20 seconds of stutter...) And the multiple sends of each DTMF tone mean callers are routed incorrectly. Does anyone have any thoughts?

- a SIP OPTIONS packet being sent from the Asterisk to the Patton every minute, receiving an immediate 405 response METHOD NOT ALLOWED. Suggestions would be appreciated!

Thank you!

 

[ncg]

For the record the DTMF issue now appears to be solved. The phone handsets are Snom 190's and I have switched their extensions in FreePBX to DTMF: inband. RFC2833 had previously been working when the server had PCI BRI cards. Perhaps both RFC2833 and inband were getting through and causing duplicates?

 

[a1itsols]

Hi,
Any chance you can post your working config, I've got an install to do the end of this week with exactly the same setup (UK) and it'd be a lot easier to look at a known good config as time will be an issue and I haven't got anywhere I can test this prior as BT are migrating from analogue to ISDN on the day.
Thanks in advance,
Julian

 

[ncg]

Sure.

In PBXinaFlash set up the SIP trunk.

In the UI set the maximum channels to 4.

In the PEER box:

host=192.168.1.XXX
username=SIPUSER
secret=SIPPASSWORD
disallow=all
allow=ulaw
qualify=yes
type=friend
context=from-trunk
canreinvite=yes

The Patton startup.cfg is attached. I've removed the Patton and SIP usernames and passwords.

Please can you post any comments or improvements. There is one recurrent log error, but it works perfectly.

Good luck!