Hi all, I've created sip trunk on Asterisk and defined asterisk server ip on address table of opensips
Then, from extension of Opensips , i can dial out to pstn through Asterisk Now, i want to route PSTN call to the extension but when Asterisk receive the call from PSTN and dial Opensips through the Sip Trunk i always got the message in the asterisk's console: *Called to-opensips/1001* * -- SIP/to-opensips-00000745 is circuit-busy* * == Everyone is busy/congested at this time (1:0/1/0)* (1001 is the extension of Opensips) Then the call hangs up. Anyone got this problem ? please help me the way to deal with! Thanks so much!
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