Patton 4114 (4 FXO)
From SIPfoundry sipXecs IP PBX, The Open Source SIP PBX for Linux - Calivia
The following is a configuration for a Patton 4114 4 port FXO gateway.
This is a simple file assuming you are using three digit extensions and using all 4 ports for FXO use and a device with a sigle ethernet port. it starts dialing out port three and hunts to port 1 for outbound calls. make sure you bind the ethernet port of your sip gateway to the proper ethernet port as shown below.
gateway sip GW-SIP bind interface eth0 router
After importing the config file, remeber to save and copy to running-config.
Remove all of the above lines when ready to save the config and import it.
#----------------------------------------------------------------#
# #
# SN4114/JO/EUI #
# R3.20 2006-07-27 H323 SIP FXS FXO #
# 1970-01-01T00:15:40 #
# Generated configuration file #
# #
#----------------------------------------------------------------#
cli version 3.20
# set your password in the system first, then do an EXPORT config, REMOVE THIS LINE
# replace the line below with your username and encrypted password REMOVE THIS LINE
administrator admin password xxxxxxxx encrypted
# replace these nameservers with your own REMOVE THIS LINE
dns-client server 198.6.1.5
dns-client server 198.6.1.2
dns-relay
webserver port 80 language en
# good idea to change the snmp read string from default of "public" REMOVE THIS LINE
snmp community public ro
sntp-client
# put in your own ntp server(s) REMOVE THIS LINE
sntp-client server primary 192.5.41.41 port 123 version 4
sntp-client server secondary 192.5.41.40 port 123 version 4
sntp-client poll-interval 36000
sntp-client local-clock-offset
# put in your own timezone REMOVE THIS LINE
sntp-client gmt-offset - 05:00:00
# change the hostname to your own for this device REMOVE THIS LINE
system hostname patton.mydomain.com
system
ic voice 0
low-bitrate-codec g729
profile napt NAPT
profile ppp default
profile call-progress-tone US_Dialtone
play 1 0 350 -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
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 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
profile pstn default
profile sip default
profile aaa default
method 1 local
method 2 none
context ip router
# ip address and mask of this device REMOVE THIS LINE
interface eth0
ipaddress 192.168.1.11 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
# default route of this device REMOVE THIS LINE
route 0.0.0.0 0.0.0.0 192.168.1.1 0
context cs switch
digit-collection timeout 3
routing-table called-e164 TAB-OUT
route .%T dest-interface IF-SIP
routing-table called-e164 TAB-IN
route .%T dest-service FXOHUNT
interface sip IF-SIP
bind gateway GW-SIP
service default
route call dest-table TAB-IN
# your sipxpbx SIP Domain Name (hostname if not using SRV records) REMOVE THIS LINE
remote mydomain.com
interface sip IF-SIP1
bind gateway GW-SIP
service default
route call dest-table TAB-IN
# your sipxpbx SIP Domain Name (hostname if not using SRV records) REMOVE THIS LINE
remote mydomain.com
# change the '100' to the number for your auto attendant REMOVE THIS LINE
address-translation outgoing-call to-header user-part fix 100 host-part interface
interface fxo PSTN
ring-number on-caller-id
use profile tone-set US
interface fxo IF_FXO0
route call dest-interface IF-SIP1
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 2
mute-dialing
use profile tone-set US
interface fxo IF_FXO1
route call dest-interface IF-SIP1
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 2
mute-dialing
use profile tone-set US
interface fxo IF_FXO2
route call dest-interface IF-SIP1
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 2
mute-dialing
use profile tone-set US
interface fxo IF_FXO3
route call dest-interface IF-SIP1
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 2
mute-dialing
use profile tone-set US
service hunt-group FXOHUNT
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_FXO3
route call 2 dest-interface IF_FXO2
route call 3 dest-interface IF_FXO1
route call 4 dest-interface IF_FXO0
context cs switch
no shutdown
gateway sip GW-SIP
bind interface eth0 router
service default
# your domain, dns_srv records (host name if not using SRV) may be different REMOVE THIS LINE
domain mydomain.com
# your sipxpbx sip domain (host name if not using SRV) REMOVE THIS LINE
default-server mydomain.com loose-router
gateway sip GW-SIP
no shutdown
port ethernet 0 0
medium auto
encapsulation ip
bind interface eth0 router
no shutdown
port fxo 0 0
flash-hook-duration 50
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO0 switch
no shutdown
port fxo 0 1
flash-hook-duration 50
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO1 switch
no shutdown
port fxo 0 2
flash-hook-duration 50
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO2 switch
no shutdown
port fxo 0 3
flash-hook-duration 50
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO3 switch
no shutdown
