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cli version 3.20
#change to your timezone
clock local default-offset -04:00
clock local default-offset -05:00
# this assumes your time zone is USA, New York. You can replace your clock offset to reflect your timezone and your DST periods
clock local dst-rule SPRING2011 -04:00 from 02:00 mar 13rd 2011 until 03:00 nov 6th 2011
clock local dst-rule SPRING2012 -04:00 from 02:00 mar 11st 2012 until 03:00 nov 4th 2012
clock local dst-rule SPRING2013 -04:00 from 02:00 mar 10th 2013 until 03:00 nov 3rd 2013
clock local dst-rule SPRING2014 -04:00 from 02:00 mar 9th 2014 until 03:00 nov 2nd 2014
clock local dst-rule SPRING2015 -04:00 from 02:00 mar 8th 2015 until 03:00 nov 1st 2015
clock local dst-rule SPRING2016 -04:00 from 02:00 mar 13rd 2016 until 03:00 nov 6th 2016
#best to use sipx as dns server or whatever dns sipx uses
dns-client server 192.168.54.2
webserver port 80 language en
#use sipx as timeserver or another source allowed on or through your network
sntp-client server 192.5.41.40
# this device hostname
system hostname sip-gw.voice.mydomain.loc

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
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
 no autonomous-transitioning

profile aaa default
 method 1 local
 method 2 none

context ip router

 interface LAN
#the ip and mask of this device
   ipaddress 192.168.54.3 255.255.255.0
   tcp adjust-mss rx mtu
   tcp adjust-mss tx mtu

context ip router
#the router of this network
 route 0.0.0.0 0.0.0.0 192.168.54.1

context cs switch
 digit-collection timeout 3

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

 interface sip IF_SIPX
   bind context sip-gateway GW-SIP
   route call dest-table SIP_TO_ISDN
#sipx sip domain name
   remote pbx.voice.mydomain.loc
#use your sip hostname below and your destination, the system AA at "100" is used for this example
   address-translation outgoing-call to-header user-part fix 100 host-part fix pbx.voice.mydomain.loc

 interface fxo IF_FXO0
   route call dest-interface IF_SIPX
   disconnect-signal loop-break
   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_SIPX
   disconnect-signal loop-break
   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_SIPX
   disconnect-signal loop-break
   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_SIPX
   disconnect-signal loop-break
   disconnect-signal busy-tone
   ring-number on-caller-id
   dial-after timeout 2
   mute-dialing
   use profile tone-set US

 service hunt-group OUTBOUND
   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_FXO3
   route call 2 dest-interface IF_FXO2
   route call 3 dest-interface IF_FXO1
   route call 3 dest-interface IF_FXO0

context cs switch
 no shutdown

location-service SIPX_SERVER
 domain 1 sipx.voice.mydomain.loc

context sip-gateway GW-SIP

 interface IF_SIPX
   bind interface LAN context router port 5060

context sip-gateway GW-SIP
 bind location-service SIPX_SERVER
 no shutdown

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

port ethernet 0 1
 medium 10 half
 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