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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal style='text-align:justify'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> I remember doing the testing and two calls going out at the same time don’t actually have to go out at the *exact* same time. The remote end will pick up one of the two calls, but there is no guarantee which one it will be. Also, if you let the first call ring too long, yes, the second call will go to voicemail, but the first call will start ringing, which is something we wanted to avoid.<o:p></o:p></span></p><p class=MsoNormal style='text-align:justify'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>John<o:p></o:p></span></p><p class=MsoNormal style='text-align:justify'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><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"'>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>Matt Riddell (lists)<br><b>Sent:</b> Monday, February 06, 2017 12:32 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Call List Campaign to an IVR<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. <o:p></o:p></p></div><div id=AppleMailSignature><p class=MsoNormal><o:p> </o:p></p></div><div id=AppleMailSignature><p class=MsoNormal>If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. <br><br><span style='font-family:"UICTFontTextStyleBody","serif"'>Kind regards,</span><o:p></o:p></p><div><p class=MsoNormal><span style='font-family:"UICTFontTextStyleBody","serif"'><br><br></span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-family:"UICTFontTextStyleBody","serif"'>Matt</span><o:p></o:p></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>On Feb 6, 2017, at 12:29 PM, Tech Support <<a href="mailto:asterisk@voipbusiness.us">asterisk@voipbusiness.us</a>> wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=MsoNormal>That's the basics, but you have to nail the timing just right. The timing is<br>really important to do it the right way.<br><br><br>-----Original Message-----<br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>[<a href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Steve Edwards<br>Sent: Monday, February 06, 2017 12:25 PM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: Re: [asterisk-users] Call List Campaign to an IVR<br><br><br><br><o:p></o:p></p><p class=MsoNormal>On Mon, 6 Feb 2017, Tech Support wrote:<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'><p class=MsoNormal> We were able to develop a feature to send the call to voicemail<o:p></o:p></p></blockquote><p class=MsoNormal>about 90% of the time. That way, an end user could (1) not be bothered by<br>having to answer the call, (2)<br><br><o:p></o:p></p><p class=MsoNormal> delete the message without listening to it, or (3) listen to the<o:p></o:p></p><p class=MsoNormal>message when it was most convenient for them. That way, they were in control<br>and things were done on<br><br><o:p></o:p></p><p class=MsoNormal> their terms.<o:p></o:p></p><p class=MsoNormal><br><br><o:p></o:p></p><p class=MsoNormal>On 6/02/2017, at 11:34 AM, Steve Edwards <<a href="mailto:asterisk.org@sedwards.com">asterisk.org@sedwards.com</a>><o:p></o:p></p><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>wrote:<o:p></o:p></p></blockquote><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'><p class=MsoNormal>Love the idea. How?<o:p></o:p></p></blockquote><p class=MsoNormal><br>On Mon, 6 Feb 2017, Matt Riddell wrote:<br><br><br><o:p></o:p></p><p class=MsoNormal>exten => <o:p></o:p></p><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>_X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1${EXTEN}@myprovider,3)<o:p></o:p></p></blockquote><p class=MsoNormal><br>Amazing. Who knew?<br><br>So how/why does this work?<br><br>I see 2 calls going out to my cell. Does the first 'busy out' my number at<br>my cell provider so the second goes straight to VM? What part does the<br>'0111' play?<br><br>--<br>Thanks in advance,<br>-------------------------------------------------------------------------<br>Steve Edwards <a href="mailto:sedwards@sedwards.com">sedwards@sedwards.com</a> Voice: +1-760-468-3867 PST<br> <a href="https://www.linkedin.com/in/steve-edwards-4244281">https://www.linkedin.com/in/steve-edwards-4244281</a><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>Check out the new Asterisk community forum at:<br><a href="https://community.asterisk.org/">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div></blockquote></div></body></html>