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>