Dear All,
I am configuring my router with the following configurations. I have a UAN number and it have distributed in ten lines. I have a four ports FXO. I terminate four telephone lines in it. When i made the call it tranfer to one of these ten numbers randomly. If the line is not terminate in router than i only hear the ring and call doesn't connect. If the call is transfered to the terminated line than it connect. And i show off-hook on router. But the call is no tranfering to my IP phones throught call manager. Kindly see the configurations and suggest me the right one.
Configurations on router
voice-port 0/1/0
input gain 10
output attenuation 10
no comfort-noise
cptone PK
!
voice-port 0/1/1
input gain 10
output attenuation 10
no comfort-noise
cptone PK
!
voice-port 0/1/2
no comfort-noise
!
voice-port 0/1/3
no comfort-noise
!
!
!
!
!
dial-peer voice 100 pots
destination-pattern 999262397
port 0/1/0
forward-digits 4
!
dial-peer voice 2 voip
destination-pattern 999262397
session protocol sipv2
session target ipv4:192.168.100.1:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 200 pots
destination-pattern 999262395
port 0/1/1
forward-digits 4
!
dial-peer voice 3 voip
destination-pattern 999262395
session protocol sipv2
session target ipv4:Call Manager IP(x.x.x.x):5060
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:Call Manager IP(x.x.x.x)
!
