<p dir="ltr">You can follow AMI events with linkedid. you have a newchannel event for that.</p>
<p dir="ltr">Ludovic Gasc (GMLudo)<br>
<a href="http://www.gmludo.eu/">http://www.gmludo.eu/</a></p>
<div class="gmail_quote">On 18 May 2015 08:09, "Muteesa Fred" <<a href="mailto:fred.muteesa@infocom.ug">fred.muteesa@infocom.ug</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-GB" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks Ludovic,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I have tried the AMI,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">But I only see the variables I need after the call has ended. Look below. I want to capture AnswerTime, and Starttime as soon as the call has been answered.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Event: Cdr<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Privilege: cdr,all<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">AccountCode:<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Source: asterisk<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Destination: 123456<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">DestinationContext: from_talklite<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">CallerID: "asterisk" <asterisk><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Channel: SIP/sip_virtual-00000001<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">DestinationChannel:<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">LastApplication: Playback<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">LastData: custom/0414-holdmusic6<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">StartTime: 2015-05-18 08:48:59<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">AnswerTime: 2015-05-18 08:49:04<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">EndTime: 2015-05-18 08:49:25<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Duration: 26<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">BillableSeconds: 21<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Disposition: ANSWERED<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">AMAFlags: DOCUMENTATION<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">UniqueID: 1431928139.2<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">UserField:<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks and regards,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Fred<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif""> <a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Ludovic Gasc<br><b>Sent:</b> Friday, May 15, 2015 1:37 PM<br><b>To:</b> Asterisk Developers Mailing List<br><b>Subject:</b> Re: [asterisk-dev] capture the time before Answer of call in dialplan<u></u><u></u></span></p><p class="MsoNormal"><u></u> <u></u></p><p>Hi,<u></u><u></u></p><p>I've no idea to do that with pure dialplan, however you can do that via a daemon that talks AMI protocol, I already do that. Via AGI/FastAGI it may be possible, I'm not really sure. ARI should also usable for this use case, never tested yet.<u></u><u></u></p><p>For the AMI daemon, you can use the programming language you want, Personally, I use Python for that.<u></u><u></u></p><p>BTW, I recommend you to use an external daemon instead of to write a C module because if your business logic crashes, it's less grave outside of Asterisk instead of inside, it should crash all of your telephony.<u></u><u></u></p><p>However, apparently it's possible via dialplan, it's better to use that with an external daemon.<u></u><u></u></p><p>Regards. <u></u><u></u></p><p>Ludovic Gasc (GMLudo)<br><a href="http://www.gmludo.eu/" target="_blank">http://www.gmludo.eu/</a><u></u><u></u></p><div style="border:none;border-left:solid #cccccc 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt"><p class="MsoNormal">Hello Everyone,<br><br>I apologise this might look like a user mailing list question but it's more<br>of a developers question. I think this is the only forum that can help me.<br><br>I have spent a week trying to achieve this. I realize I might have to write<br>a C or PERL extension to achieve it but I don't know where to start. Or<br>might need to rebuild the dial funtion.<br>Objective:<br>I want to dial a number in asterisk via a sip-trunk, and when the other<br>party answers the call, I want to test if the call was answered before 5<br>seconds, or after 5 seconds.<br>If it was answered in less than 5 seconds I want to cancel the call,<br>otherwise the call should continue.<br><br>Your guidance will highly be appreciated.<br><br>Regards,<br>Fred Muteesa<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><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br>   <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><u></u><u></u></p></div></div></div><br>--<br>
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