SIP Trunk Error!

Hello Everybody here,

I have one trunk and outbound route to my VSP and previously i can call the phone numbers that my VSP support but since last 2 or 3 weeks ago I can not. Whenever i call it fails and says like that “all circuit are busy now” and in the asterisk cli it shows me like that

" GotoIf(“SIP/22012-0972b108”, “0?disabletrunk|1”) in new stack "

" Everyone is busy/congested at this time (1:0/1/0)"

““TRUNK Dial failed due to CONGESTION - failing through to other trunks”) in new stack”

And i check my sip regsitry request in CLI , it shows me like that
"Host Username Refresh State Reg. Time"
x.x.x.x:5060 123456 120 Request Sent

Thats why i complaint my VSP but they replied me that they didnt change anyconfiguration about my trunk and their server is running up and to check my tb again.ALthough i checked my tb but i didnt find anything.
So pls anyone help or point me out how to solve this problem. Does SIP registration need to be registered?.

Any help will be greatly appreciated.