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</o:shapelayout></xml><![endif]--></head><body lang=EN-ZA link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal><span style='font-size:16.0pt'>Hi Guys<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>I’ve got a situation where an incoming call originated from a trunk provider will ring in a call center and be answered by an agent.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>Usually within 15 seconds of the incoming call being answered, it will randomly blind-transfer to another extension in the same call center.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>It is as if Asterisk is mis-reading some noise in-band as DTMF and doing the transfer, or the caller is emitting DTMF to transfer.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>An examination of CEL records for the relevant call shows the transfer taking place, but no party (the caller or callee) tried to transfer. The relevant queue application already has only small “t” in the parameter list (e. g. only let the called user - e. g. my agent inside - transfer the incoming call), thereby preventing the caller maybe emitting DTMF down the line and transferring at their behest.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>Yet the random, uncommanded blind transfers still take place, always withing about 15 seconds after call initiation in answering an incoming SIP call.<o:p></o:p></span></p><p class=MsoNormal><b><span style='font-size:16.0pt'><o:p> </o:p></span></b></p><p class=MsoNormal><span style='font-size:16.0pt'>We’re using YeaLink SIP phones - T21P - got 17 branches, this only happens at 3 of them at random times. Same hardware and Asterisk versions (1.8.32.3) at all of them, same dialplans.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>Operating over about several years, about 10 000 000 incoming calls handled so far, and this only started happening now... dialplans are untouched for months.<o:p></o:p></span></p><p class=MsoNormal><b><span style='font-size:16.0pt'><o:p> </o:p></span></b></p><p class=MsoNormal><span style='font-size:16.0pt'>Any ideas, pointers, anybody encountered this before?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt'>Thanks<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-ZA'><img width=780 height=170 style='width:8.125in;height:1.7708in' id="Picture_x0020_1" src="cid:image001.png@01D3A0CD.60753430" alt="Description: signature"><o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>