Regarding all ciscuits are busy now regarding

Hi all,
I’m new to asterisk world. And trying to configure a asterisk server to terminate voip traffic to pstn network.
I have installed asterisk now and installed TDM800P 8port fxo module on it.
Then i configured simple outbound route and pstn trunk on it.
But when im trying to take a call to a mobile number from sip phone thru fxo card im getting all circuits are busy now recording. Below are the debug info. Please advice & this will be a grate help for me.

[[email protected] ~]# lsdahdi

Span 1: WCTDM/0 “Wildcard TDM800P Board 1” (MASTER)

1 FXS FXSKS (EC: MG2) RED
2 FXS FXSKS (EC: MG2) RED
3 FXS FXSKS (EC: MG2) RED
4 FXS FXSKS (EC: MG2) RED
5 FXS FXSKS (EC: MG2) RED
6 FXS FXSKS (EC: MG2) RED
7 FXS FXSKS (EC: MG2) RED
8 FXS FXSKS (EC: MG2) RED

Get rid of the Dial Rules in the trunk setting. The Trunk Dial Rules are for adding country codes and area codes to outgoing numbers. So your users don’t have to do 10 digit dialing or adding the country codes to certain routes.

The way you have it now, you may have to dial 20 digits to use your Zap g0 trunk.