[OpenSIPS-Users] Call from Asterisk to Opensips

[Max Mühlbronner]

Hi,

i would suggest doing sip-traces on asterisk (sip debug) and opensips 
(ngrep) while watching the corresponding log messages of both servers 
(asterisk/opensips). Most of the time it´s difficult to find a problem 
by looking at it from just one side.

BR

Max M.

Am 06.05.2011 05:53, schrieb 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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