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</o:shapelayout></xml><![endif]--></head><body lang=EN-ZA link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:14.0pt;font-family:"Calibri Light",sans-serif;mso-fareast-language:EN-US'>Hi Khalil<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:14.0pt;font-family:"Calibri Light",sans-serif;mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:14.0pt;font-family:"Calibri Light",sans-serif;mso-fareast-language:EN-US'>Thanks for the reply. Yup, I’ve switched out the phone with another new-from-box Yealink, also moved that network endpoint to another RJ45 port on the switch...<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:14.0pt;font-family:"Calibri Light",sans-serif;mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:14.0pt;font-family:"Calibri Light",sans-serif;mso-fareast-language:EN-US'>Regards,<o:p></o:p></span></p><p class=MsoNormal><b><span lang=EN-US>From:</span></b><span lang=EN-US> Khalil Khamlichi [mailto:khamlichi.khalil@gmail.com] <br><b>Sent:</b> Sunday, 11 February 2018 16:48<br><b>To:</b> viljoens@verishare.co.za; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Subject:</b> Re: [asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>maybe extension 3082 has some sort of network issue or maybe hardphone issue. test on another phone / network plug.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Feb 9, 2018 2:48 PM, "Stefan Viljoen" <<a href="mailto:viljoens@verishare.co.za">viljoens@verishare.co.za</a>> wrote:<o:p></o:p></p><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>Hi Guys</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument’s sake.)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>A few seconds after 3082 has started speaking to the outside caller</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>- 3082’s call goes dead in their handset.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>- The outside caller goes back into the queue, hears queue MOH and gets answered by another person in the office as if they are dialing in all over again.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>- 3082’s phone starts ringing again after they hang up in puzzlement and if they then pick up they speak to another person who is trying to make an OUTGOING call in their call center.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>This is for a medium sized call center which (along with 17 other centers in the same country) run the same dialplan on Asterisk 1.8.32.3 - only happens at this location.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>Literally 100 000+ calls are handled across these 18 centers every day, only about 10 or 20 at this one center (with carbon-copy dialplan and SIP phone hardware types - Yealink T-21Ps - as at every other branch) keeps disconnecting people picked up and transferred from the incoming queue from the person transferred to, and then connects them and the transferree to other phones - the caller as if he is phoning in AGAIN into the incoming queue, the transferred-to person to someone else who is trying to dial out once the transferred-to person has hung up after losing the incoming caller.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>Anybody ever encountered something similar? The same dialplan on the same Ast version runs fine in 17 other locations, some with ten or twenty times more traffic and none of these issues.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>No errors or strangeness apparent in the CLI, verbose log, DTMF log...</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:16.0pt'>Thanks!</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:16.0pt'> </span></b><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><img border=0 width=780 height=170 style='width:8.125in;height:1.7708in' id="m_3296000724733433052Picture_x0020_1" src="cid:image001.png@01D3A3DB.B7E65FD0" alt="Description: signature"><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div><p class=MsoNormal><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>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">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" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></blockquote></div></div></div></body></html>