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Hi,<br>
<br>
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>
<br>
BR<br>
<br>
Max M.<br>
<br>
Am 06.05.2011 05:53, schrieb Duong Manh Truong:
<blockquote
cite="mid:BANLkTim73q7yj3F6dnbMTrSi2ZDHQJB38g@mail.gmail.com"
type="cite">Hi all,
<div>I've created sip trunk on Asterisk and defined asterisk
server ip on address table of opensips</div>
<div><br>
</div>
<div>Then, from extension of Opensips , i can dial out to pstn
through Asterisk</div>
<div>
<br>
</div>
<div>Now, i want to route PSTN call to the extension </div>
<div>but when Asterisk receive the call from PSTN and dial
Opensips through the Sip Trunk</div>
<div>i always got the message in the asterisk's console: </div>
<div>
<div> <i>Called to-opensips/1001</i></div>
<div><i> -- SIP/to-opensips-00000745 is circuit-busy</i></div>
<div><i> == Everyone is busy/congested at this time (1:0/1/0)</i></div>
</div>
<div><br>
</div>
<div>(1001 is the extension of Opensips) </div>
<div>Then the call hangs up. </div>
<div><br>
</div>
<div>Anyone got this problem ? please help me the way to deal
with!</div>
<div><br>
</div>
<div>Thanks so much! </div>
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</pre>
</blockquote>
<br>
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