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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:10.5pt;font-family:Consolas;color:#1F497D'>Sounds like you are running FreePBX.<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 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>Mitul Limbani<br><b>Sent:</b> Thursday, September 04, 2014 6:34 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Special functionality for Secretary/Boss<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><p>Kevin,<o:p></o:p></p><p>With your dialplan with g option on external trunk, if the call finishes the boss's leg of call also gets disconnected. So the next instruction would make a call to secratary, however with no one on other end.<o:p></o:p></p><p>Mitul <o:p></o:p></p><div><p class=MsoNormal>On 04-Sep-2014 11:44 PM, "Kevin Larsen" <<a href="mailto:kevin.larsen@pioneerballoon.com">kevin.larsen@pioneerballoon.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal><tt><span style='font-size:10.0pt'><a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> wrote on 09/04/2014 11:57:40 AM:</span></tt><span style='font-size:10.0pt;font-family:"Courier New"'><br><tt>> We are currently migrating from a Nortel pbx to Asterisk and we </tt><br><tt>> have been able to convert most of the functions that people are used to </tt><br><tt>> but there is one I have no clear idea how to do. The scenario is:</tt><br><tt>> </tt><br><tt>> Boss calls secretary from outside the office to get connected to </tt><br><tt>> another outside destination. The secretary dials the destination and </tt><br><tt>> then trasfers call to the boss. When boss finishes with that person </tt><br><tt>> they want to send the call back to the secretary in order to make </tt><br><tt>> another connection or simply to talk to the secretary.</tt><br><tt>> </tt><br><tt>> The first part is not a problem, but after the boss finishes his </tt><br><tt>> call how can we send the call back to the secretary? I was thinking of </tt><br><tt>> using a conference room but how would the secretary know when the boss </tt><br><tt>> has finished? Anyone know how to handle this scenario?</tt><br></span><br><tt><span style='font-size:10.0pt'>I haven't tested this, but my initial thought would be to create a special context or extension that the secretary could route through when doing the call transfer. The Dial application could be called with the 'g' option to continue the dialplan at the next priority when the call hangs up. Something like a normal call transfer would just dial the number as normal, but for the special transfer, you could prepend the dialed number with a #.</span></tt> <br><br><tt><span style='font-size:10.0pt'>For example (using a local US dialstring, change to fit your needs):</span></tt> <br><br><tt><span style='font-size:10.0pt'>; This is a normal external call.</span></tt> <br><tt><span style='font-size:10.0pt'>exten => _NXXNXXX,1,Dial(SIP/your_external_trunk/${EXTEN})</span></tt> <br><tt><span style='font-size:10.0pt'> same => n,Hangup()</span></tt> <br><br><tt><span style='font-size:10.0pt'>; This is a call that should be transfered back to the secretary's extension when external call is finished</span></tt> <br><tt><span style='font-size:10.0pt'>exten => _#NXXNXXX,1,NoOp(Special Dial for Boss/Secretary Transfer)</span></tt> <br><tt><span style='font-size:10.0pt'> same => n,Dial(SIP/your_external_trunk/${EXTEN:1},,g)</span></tt> <br><tt><span style='font-size:10.0pt'>; First call has ended, now we go back to the secretary)</span></tt> <br><tt><span style='font-size:10.0pt'> same => n,Dial(SIP/1234)</span></tt> <br><tt><span style='font-size:10.0pt'> same => n,Hangup()</span></tt> <br><br><tt><span style='font-size:10.0pt'>That's at least where I would start with my testing and then develop the solution from there.</span></tt><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></div></body></html>