[OpenSIPS-Users] Call from Asterisk to Opensips

[Duong Manh Truong]

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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