voice asterisk cisco 2600 trixbox trunk

[tjbradford]

below is a working config pushing calls from a pstn line to a cisco ip phone which is great, but not exactly what i'm after, i wish the calls to be pushed to a ring group.

the ring group i setup was 600
i changed both the "107" references in the below config to 600 but the phones in that group never ring the tones i hear when ringing that number are just like engaged beeps, but this all works fine with any single handset number i put in.

anyone with ideas on this ?


!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
no network-clock-participate slot 1
no network-clock-participate wic 0
no aaa new-model
ip subnet-zero
ip cef
!
!
!
!
!
voice rtp send-recv
!
voice service voip
!
!
!
!
!
!
!
!
!
!
!
!
!
interface ATM0/0
no ip address
shutdown
no atm ilmi-keepalive
dsl operating-mode auto
!
interface FastEthernet0/0
ip address 192.168.1.253 255.255.255.0
duplex auto
speed auto
!
ip classless
ip http server
!
!
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
cptone GB
connection plar 107
station-id name External call
!
voice-port 1/0/1
!
!
!
!
!
dial-peer voice 100 pots
destination-pattern 1..
no digit-strip
port 1/0/0
!
dial-peer voice 2 voip
description Route calls to the Asterisk
destination-pattern 107
session protocol sipv2
session target ipv4:192.168.1.66:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
clid strip
no vad
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:192.168.1.66
!
!
telephony-service
transfer-pattern ..
transfer-system full-blind
!
!
line con 0
line aux 0
line vty 0 4
login
!
!
end

Router#

 

[whykap]

I would think the issue is on the asterisk end. All you are doing is pointing a call through a sip dial-peer.
Are you also changing your connection plar in the 1/0/0 port to match the dial-peer?