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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>This might have changed but IIRC /etc/asterisk/manager.conf controls what events you have access to.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Jerry Geis<br><b>Sent:</b> Thursday, January 24, 2013 1:14 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] question on SIP trunk and AMI to place call<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal><o:p> </o:p></p></blockquote><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><pre style='orphans: 2;text-align:start;widows: 2;-webkit-text-size-adjust: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>You probably want the Dial event. It is raised both at the beginning of<o:p></o:p></pre><pre>the Dial, as well as when the Dial completes.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial">https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Note that the Channel: field will contain the name initiating the Dial,<o:p></o:p></pre><pre>the Destination: field will contain the channel name being dialled, and<o:p></o:p></pre><pre>the Dialstring: field will contain the non-technology specific portion<o:p></o:p></pre><pre>of the thing being dialled.<o:p></o:p></pre></blockquote><p class=MsoNormal>I get that even on the system with the PRI card and using DAHDI<br>however I am not getting that event on the system with the SIP trunk .<br><br>Is there something to enable to get that???<br>Both systems are running Asterisk 11.0.2.<br><br>Thanks,<br><br>Jerry<o:p></o:p></p></div></body></html>