Gday
I am trying to receive calls on the FXO’s of the Patton 4114 connected to FreePBX.
Using Asterisk 1.8 and FreePBX 2.11
SIP Trunk created
[SmartNode]
type=friend
disallow=all
allow=ulaw&alaw
canreinvite=yes
context=inbound
callerid=asreceived
dtmfmode=rfc2833
host=dynamic
insecure=very
qualify=yes
username=SmartNode
secret=nicooo1234
trunk=yes
CFG file of the Patton 4114
cli version 3.20
administrator administrator password gEilf1b9jtu1CmCoVs5b6A== encrypted
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4
system
ic voice 0
profile acl ACL_WAN_PERMIT_ALL_MGMT
permit 1 ip any any
profile acl ACL_WAN_PERMIT_SEL_MGMT
deny 1 tcp any any eq 23
deny 2 tcp any any eq 80
deny 3 udp any any eq 161
permit 4 ip any any
profile acl ACL_WAN_BLOCK_ALL_MGMT
deny 1 tcp any any eq 23
deny 2 tcp any any eq 80
deny 3 udp any any eq 161
permit 4 ip any any
profile acl ACL_WAN_IN
permit 1 tcp any any eq 123
permit 2 tcp any any eq 123
permit 3 tcp any any gt 1023
permit 4 udp any any gt 1023
permit 5 gre any any
profile service-policy SP_WAN_OUT
rate-limit 100000 header-length 18 voice-margin 0
source traffic-class local-voice
priority
source traffic-class default
priority
profile service-policy SP_WAN_IN
rate-limit 100000 header-length 18 voice-margin 200
source traffic-class local-voice
priority
source traffic-class default
queue-limit 4
profile napt NAPT_WAN
profile ppp default
profile tone-set default
profile tone-set Europe
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 g711ulaw64k rx-length 20 tx-length 20
codec 2 g711alaw64k rx-length 20 tx-length 20
codec 3 g729 rx-length 20 tx-length 20
no dtmf-relay
no dtmf-mute-encoder
dejitter-mode static
dejitter-max-delay 120
fax transmission 1 bypass g711alaw64k
fax transmission 2 relay t38-udp
profile pstn default
profile sip default
profile dhcp-server DHCPS_LAN
network 192.168.1.0 255.255.255.0
include 1 192.168.1.10 192.168.1.99
lease 2 hours
default-router 1 192.168.1.1
domain-name-server 1 192.168.1.1
profile aaa default
method 1 local
method 2 none
context ip router
interface IF_IP_LAN
ipaddress 192.168.105.9 255.255.255.0
icmp router-discovery
icmp redirect accept
context ip router
route 0.0.0.0 0.0.0.0 10.10.0.1 0
subscriber ppp SUB_PPPOE
dial out
no multilink
authentication chap
authentication pap
context cs switch
digit-collection timeout 3
routing-table called-e164 CDPN_2_IP
routing-table called-e164 ASTERISK_OUT
route default dest-service ASTERISK
interface sip IF_ASTERISK
bind gateway SIP_GW
service default
route call dest-table ASTERISK_OUT
early-connect
early-disconnect
use profile voip VOIP
interface fxo IF_FXO_00
route call dest-interface IF_ASTERISK
no disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 2
use profile tone-set Europe
interface fxo IF_FXO_01
route call dest-interface IF_ASTERISK
disconnect-signal loop-break
dial-after timeout 3
use profile tone-set Europe
interface fxo IF_FXO_02
route call dest-interface IF_ASTERISK
disconnect-signal loop-break
dial-after timeout 3
use profile tone-set Europe
interface fxo IF_FXO_03
route call dest-interface IF_ASTERISK
disconnect-signal loop-break
dial-after timeout 3
use profile tone-set Europe
service hunt-group ASTERISK
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_FXO_00
route call 2 dest-interface IF_FXO_01
route call 3 dest-interface IF_FXO_02
route call 4 dest-interface IF_FXO_03
context cs switch
no shutdown
gateway sip SIP_GW
bind interface IF_IP_LAN router
service default
domain 192.168.105.10
defaultserver manual 192.168.105.10 5060 loose-router
registration manual 192.168.105.10 5060 use-default-server
user SmartNode authenticate password Hz5gYqcRrjzsQ8Afp0GNzg== encrypted register
gateway sip SIP_GW
no shutdown
port ethernet 0 0
medium auto
bind interface IF_IP_LAN router
pppoe
session SES_PPPOE
bind subscriber SUB_PPPOE
shutdown
port ethernet 0 0
no shutdown
port fxo 0 0
flash-hook-duration 600
use profile fxo za
encapsulation cc-fxo
bind interface IF_FXO_00 switch
no shutdown
port fxo 0 1
flash-hook-duration 600
use profile fxo za
encapsulation cc-fxo
bind interface IF_FXO_01 switch
no shutdown
port fxo 0 2
flash-hook-duration 600
use profile fxo za
encapsulation cc-fxo
bind interface IF_FXO_02 switch
no shutdown
port fxo 0 3
flash-hook-duration 600
use profile fxo za
encapsulation cc-fxo
bind interface IF_FXO_03 switch
no shutdown
In Asterisk I will just add this below to the sip.conf
[inbound]
exten => s,1,NoOp(CALLING Party : ${CALLERID(num)})
exten => s,n,Dial(SIP/100,30,r)
exten => s,n,PlayBack(vm-goodbye)
exten => s,n,HangUp()
My problem is how do one add this to the sip.conf in FreePBX??