<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">Hi Truong<br><br>first thing you should try to read the asterisk SIP TRUNK and here is the basic example and i think the problem is asterisk not opensips<br>http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/<br><br>and make sure to check the debug from asterisk and opensips, i think you will get the clues :D<br><br><br>Ha`<br><br>--- On <b>Thu, 5/5/11, Duong Manh Truong <i><ngoahotanglongbk@gmail.com></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><br>From: Duong Manh Truong <ngoahotanglongbk@gmail.com><br>Subject: [OpenSIPS-Users] Call from Asterisk to Opensips<br>To: "OpenSIPS users mailling list" <users@lists.opensips.org><br>Date: Thursday, May 5, 2011, 9:53 PM<br><br><div id="yiv413426246">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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