First post... I sucessfully have 3cx talking to my cisco 2651 w/2 FXO and 2 FXS (only have 1 pots line coming into the house and the cisco is also doing my DSL authentication and network default gateway) but I have both fxs port connected (1 to physical phones, 1 connected to my way2call connected to Homeseer) I have 4 instances of Xlite on different computers. Calling works in all directions, softphone to outside, outside to softphone, fxs to outside, softphone to softphone. I have the calls currently forwarded to one of the softphones
my issue. When I setup 3cx, it asked for extensions, fine I did that but I want the incoming call from the pstn gateway to go to the attendant, so I created an additional extension but the attendant box number is not an option in the forwarding... so my question is how do I get an incoming call from the pstn gateway to go to the attendant?
Below is my 2651 config, I searched and there doesn't seem to be a lot of info on unsupported gateways for 3cx. A couple gottcha's first you have to set the pstn to not register and set your outbound rules to strip the 9. Hope the following config can help someone
C2651#sh run
Building configuration...
Current configuration : 4358 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname C2651
!
boot-start-marker
boot-end-marker
!
no logging console
enable secret 5 $1$frux$hI7F0TDJLVfgdNBlFqexk1
enable password **********
!
no aaa new-model
clock timezone GMT-7 7
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
ip domain name *******.***
ip name-server ***.***.***.***
ip name-server ***.***.***.***
vpdn enable
!
voice service pots
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
sip
!
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8 bytes 40
codec preference 4 g723r63 bytes 96
codec preference 5 g726r16 bytes 80
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8 bytes 40
codec preference 4 g723r63 bytes 96
codec preference 5 g726r16 bytes 80
!
interface ATM0/0
no ip address
no ip mroute-cache
atm restart timer 300
no atm ilmi-keepalive
dsl operating-mode auto
pvc 0/32
encapsulation aal5mux ppp dialer
dialer pool-member 1
!
interface FastEthernet0/0
ip address ***.***.***.1 255.255.255.0
ip nat inside
ip tcp adjust-mss 1452
speed auto
full-duplex
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Dialer1
mtu 1492
ip address ***.***.***.118 255.255.255.248
no ip redirects
no ip proxy-arp
ip nat outside
encapsulation ppp
dialer pool 1
dialer-group 1
no cdp enable
ppp authentication chap callin
ppp chap hostname *****@******.***
ppp chap password 0 **********
ppp pap sent-username *******@******.*** password 0 ***********
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 Dialer1 permanent
!
ip http server
no ip http secure-server
ip nat pool mpool ***.***.***.*** ***.***.***.*** netmask 255.255.255.248
ip nat inside source list 1 pool mpool overload
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static udp ******************* extendable
ip nat inside source static udp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static udp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
!
access-list 1 permit ***.***.***.0 0.0.0.255
snmp-server community public RO
snmp-server community ********** RO
snmp-server community host RO
snmp-server community private RW
snmp-server enable traps tty
!
control-plane
!
voice-port 1/0/0
disconnect-ack
input gain 4
output attenuation 4
no comfort-noise
description to House Phones
threshold noise -90
station-id number 7851
caller-id enable
!
voice-port 1/0/1
description to HS
station-id number 7852
caller-id enable
!
voice-port 1/1/0
output attenuation 0
no comfort-noise
connection plar opx 10003
description 3CX
station-id name pots incoming
station-id number ***********
!
voice-port 1/1/1
!
dial-peer voice 10003 voip
description 10003
preference 1
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:***.***.***.17
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10000 pots
preference 1
destination-pattern .T
port 1/1/0
!
dial-peer voice 7851 pots
destination-pattern 7851
port 1/0/0
!
dial-peer voice 7852 pots
destination-pattern 7852
port 1/0/1
!
sip-ua
authentication username 10003 password 0257540B5B55
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers expires 300000
sip-server ipv4:***.***.***.17:5060
!
line con 0
line aux 0
line vty 0 4
password **********
login
!
!
end
C2651#
my issue. When I setup 3cx, it asked for extensions, fine I did that but I want the incoming call from the pstn gateway to go to the attendant, so I created an additional extension but the attendant box number is not an option in the forwarding... so my question is how do I get an incoming call from the pstn gateway to go to the attendant?
Below is my 2651 config, I searched and there doesn't seem to be a lot of info on unsupported gateways for 3cx. A couple gottcha's first you have to set the pstn to not register and set your outbound rules to strip the 9. Hope the following config can help someone
C2651#sh run
Building configuration...
Current configuration : 4358 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname C2651
!
boot-start-marker
boot-end-marker
!
no logging console
enable secret 5 $1$frux$hI7F0TDJLVfgdNBlFqexk1
enable password **********
!
no aaa new-model
clock timezone GMT-7 7
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
ip domain name *******.***
ip name-server ***.***.***.***
ip name-server ***.***.***.***
vpdn enable
!
voice service pots
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
sip
!
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8 bytes 40
codec preference 4 g723r63 bytes 96
codec preference 5 g726r16 bytes 80
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8 bytes 40
codec preference 4 g723r63 bytes 96
codec preference 5 g726r16 bytes 80
!
interface ATM0/0
no ip address
no ip mroute-cache
atm restart timer 300
no atm ilmi-keepalive
dsl operating-mode auto
pvc 0/32
encapsulation aal5mux ppp dialer
dialer pool-member 1
!
interface FastEthernet0/0
ip address ***.***.***.1 255.255.255.0
ip nat inside
ip tcp adjust-mss 1452
speed auto
full-duplex
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Dialer1
mtu 1492
ip address ***.***.***.118 255.255.255.248
no ip redirects
no ip proxy-arp
ip nat outside
encapsulation ppp
dialer pool 1
dialer-group 1
no cdp enable
ppp authentication chap callin
ppp chap hostname *****@******.***
ppp chap password 0 **********
ppp pap sent-username *******@******.*** password 0 ***********
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 Dialer1 permanent
!
ip http server
no ip http secure-server
ip nat pool mpool ***.***.***.*** ***.***.***.*** netmask 255.255.255.248
ip nat inside source list 1 pool mpool overload
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static udp ******************* extendable
ip nat inside source static udp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static udp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
ip nat inside source static tcp ******************* extendable
!
access-list 1 permit ***.***.***.0 0.0.0.255
snmp-server community public RO
snmp-server community ********** RO
snmp-server community host RO
snmp-server community private RW
snmp-server enable traps tty
!
control-plane
!
voice-port 1/0/0
disconnect-ack
input gain 4
output attenuation 4
no comfort-noise
description to House Phones
threshold noise -90
station-id number 7851
caller-id enable
!
voice-port 1/0/1
description to HS
station-id number 7852
caller-id enable
!
voice-port 1/1/0
output attenuation 0
no comfort-noise
connection plar opx 10003
description 3CX
station-id name pots incoming
station-id number ***********
!
voice-port 1/1/1
!
dial-peer voice 10003 voip
description 10003
preference 1
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:***.***.***.17
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10000 pots
preference 1
destination-pattern .T
port 1/1/0
!
dial-peer voice 7851 pots
destination-pattern 7851
port 1/0/0
!
dial-peer voice 7852 pots
destination-pattern 7852
port 1/0/1
!
sip-ua
authentication username 10003 password 0257540B5B55
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers expires 300000
sip-server ipv4:***.***.***.17:5060
!
line con 0
line aux 0
line vty 0 4
password **********
login
!
!
end
C2651#
