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
"GosubIf("SIP/22012-0972b108", "0?sub-pincheck|s|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.
SOUL00
Member Since:
2009-07-29