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</o:shapelayout></xml><![endif]--></head><body 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'>Not the greatest solution, but since you are most likely using a script for the AMI process, you could do an <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Asterisk –rx “core show channels verbose”|grep SIP/testmachine-0000000d <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>And get the dialed number from that.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Actually you could issue the AMI command core show channels verbose.<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><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Tiago Geada<br><b>Sent:</b> Thursday, January 24, 2013 11:24 AM<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><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><span style='font-family:"Tahoma","sans-serif";color:#000066'>Have you tried and looked up all events generated when you place the call?</span><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><span style='font-family:"Tahoma","sans-serif";color:#000066'>some of them are bound to have the variable callerid set</span><o:p></o:p></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On 24 January 2013 16:46, Jerry Geis <<a href="mailto:geisj@pagestation.com" target="_blank">geisj@pagestation.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>When I am monitoring the AMI I see the following event<br>for a call I just made over a SIP trunk.<br><br>Event: Newchannel<br>Privilege: call,all<br>Channel: SIP/testmachine-0000000d<br>ChannelState: 0<br>ChannelStateDesc: Down<br>CallerIDNum:<br>CallerIDName:<br>AccountCode:<br>Exten:<br>Context: testmachine<br>Uniqueid: 1359035395.20<br><br>In this event or any event following I do not see<br>the phone number that I dialled. How do I "correlate"<br>the "SIP/testmachine-0000000d" to the number I just dialed????<br>(purpose is to hangup the call later if I need to interrupt it)<br><br>Now if I am using a machine with actual hardware cards, the phone<br>number is included as part of the Channel so I can look that up.<br>but for a SIP trunk the phone number dialled does not come over the AMI.<br><br>How do I match up the call I just started (using AMI over SIP trunk) to the number I called?<br><br>Thanks,<br><br>jerry<br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>