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) !
These are the paths to get to each CCX logs through CLI. They may be helpful if you are having issues accessing RTMT or downloading logs through it.
If you want to download them you have to prefix "file get " and you can add one of the options (re...