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free Patton pstn gateway configurator for snom ONE


mattlandis
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  • 1 month later...

Thanks for posting this Matt. It's very useful.

 

I've been trying without luck to configure my 4114 to have two lines ring one group of extensions and another single line to ring a separate extension but cannot seem to figure out how to do this. I've tried playing with the ports on both the patton and the trunk setup on pbxnsip but it just doesn't seem to work. Can you offer any advice on how I might be able to do this?

 

Thanks

 

Jag

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my bad. I'm mixing up doing it manual and using the configurator.

 

the configurator sends everything to one ext.

 

I'll post a manual config (you need to use patton fw r4.2 or this wont work)

 

in this config you'll need to change "{Port_0/0_Destination}" to the port number. (replace the string inside the quotes to only the extension numbeR)

 

ps-the configurator has some other limitations we will try to fix with patton. one problem is that if you set it to use USATones it uses something not usa. you need to fix it manually. the ability to designate a different extension for each line would also be nice. We will see what we can do about getting patton to update.

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#

#

# SN4114/JO/EUI #

# R4.2 2008-03-11 H323 SIP FXS FXO #

# 1970-01-03T05:14:22 #

# SN/00A0BA03CA4E #

# Generated configuration file #

#

#

cli version 3.20

webserver port 80

language en

sntp-client

sntp-client server primary 129.132.2.21 port 123 version 4

 

system

 

ic voice 0

low-bitrate-codec g729

profile ppp default

 

profile call-progress-tone US_Dialtone

 

play 1 1000 350 -13 440 -13

profile call-progress-tone US_Alertingtone

 

play 1 2000 440 -19 480 -19

pause 2 4000

profile call-progress-tone US_Busytone

 

play 1 500 480 -24 620 -24

pause 2 500

profile tone-set default

 

map call-progress-tone dial-tone US_Dialtone

map call-progress-tone ringback-tone US_Alertingtone

map call-progress-tone busy-tone US_Busytone

map call-progress-tone release-tone US_Busytone

map call-progress-tone congestion-tone US_Busytone

profile tone-set US

 

map call-progress-tone dial-tone US_Dialtone

map call-progress-tone ringback-tone US_Alertingtone

map call-progress-tone busy-tone US_Busytone

map call-progress-tone release-tone US_Busytone

map call-progress-tone congestion-tone US_Busytone

 

profile voip default

 

codec 1 g711ulaw64k rx-length 20 tx-length 20

codec 2 g711alaw64k rx-length 20 tx-length 20

rtp traffic-class local-default

profile pstn default

 

output-gain 5

profile sip default

 

profile aaa default

 

method 1 local

method 2 none

context ip router

 

interface eth0

ipaddress {patton_gateway_ip_address} 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 {router_ip_address} 0

 

context cs switch

 

digit-collection timeout 2

 

routing-table called-e164 RT_TO_SIP_000

route default dest-interface IF_PBXNSIP MAP_TO_000

 

mapping-table called-e164 to called-e164 MAP_TO_000

map default to {Port_0/0_Destination}

 

routing-table called-e164 RT_TO_SIP_001

route default dest-interface IF_PBXNSIP MAP_TO_001

 

mapping-table called-e164 to called-e164 MAP_TO_001

map default to {Port_0/1_Destination}

 

routing-table called-e164 RT_TO_SIP_002

route default dest-interface IF_PBXNSIP MAP_TO_002

 

mapping-table called-e164 to called-e164 MAP_TO_002

map default to {Port_0/2_Destination}

 

routing-table called-e164 RT_TO_SIP_003

route default dest-interface IF_PBXNSIP MAP_TO_003

 

mapping-table called-e164 to called-e164 MAP_TO_003

map default to {Port_0/3_Destination}

 

interface sip IF_PBXNSIP

bind gateway GW_PBXNSIP

service default

route call dest-service FXO_HUNT

remote-party-id calling-party

 

interface fxo IF_CO1

route call dest-table RT_TO_SIP_000

no disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

mute-dialing

interface fxo IF_CO2

route call dest-table RT_TO_SIP_001

no disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

mute-dialing

interface fxo IF_CO3

route call dest-table RT_TO_SIP_002

no disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

mute-dialing

interface fxo IF_CO4

route call dest-table RT_TO_SIP_003

no disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

mute-dialing

 

service hunt-group FXO_HUNT

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_CO1

route call 2 dest-interface IF_CO2

route call 3 dest-interface IF_CO3

route call 4 dest-interface IF_CO4

context cs switch

 

no shutdown

gateway sip GW_PBXNSIP

 

bind interface eth0 router

service default

domain {pbxnsip_server}

defaultserver manual {pbxnsip_server} loose-router

gateway sip GW_PBXNSIP

 

no shutdown

port ethernet 0 0

 

medium auto

encapsulation ip

bind interface eth0 router

no shutdown

port fxo 0 0

 

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_CO1 switch

no shutdown

port fxo 0 1

 

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_CO2 switch

no shutdown

port fxo 0 2

 

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_CO3 switch

no shutdown

port fxo 0 3

 

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_CO4 switch

no shutdown

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Hi Matt,

 

Thanksvery much for the above. I tried it out - changing the {Port_0/0_Destination} strings to the hunt group 200 and 300 for {Port_0/2_Destination} but had no luck. When I dial into port 0/2 which is a different number it rings only the hunt group 200.

 

I'm using the following trunk configs on the PBXnSIP:

 

Trunk 1 (for first two lines)

account: 1000

pw: 1000

Proxy Address: 172.16.0.4:5060

 

Trunk 2 (for 3rd line)

account: 1002

pw: 1002

Proxy Address: 172.16.0.4:5060

 

I think I may be doing something wrong here.

 

Thanks

 

Jag

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Hi Matt,

 

I also tried changing the {Port_0/0_Destination} to the trunk account numbers on pbxnsip and this didn't work either.

 

I've got a linksys SPA 3102 which i could use for the third line instead of the patton which would eliminate the need for this config. Unfortunately I understand the new firmware on the linksys won't work too well with pbxnsip.

 

Jag

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  • 1 month later...

Hi Matt

 

Just been sent your configurator and I wonder if you could assist.

 

We want to add a Snom One to our existing Patton 4960 gateways. These gateways are already in use and carrying live traffic. Can you confirm what effect your configurator might have on the running config of these gateways.

 

Thanks in advance.

 

Wayne

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