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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:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Only trust the wiki if it explicitly refers to your current version (and then you should still test it).<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><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>Carlos Alvarez<br><b>Sent:</b> Tuesday, February 07, 2012 10:46 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] MixMonitor and ChanSpy<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>It's a good thing I never read that warning, since I've been using those in a call center environment for about seven years and never had that issue. Started with 1.2, went to 1.4 and 1.6 now. So I can't answer your question about when it was "fixed" but I've never had a problem doing it (70 concurrent calls max, all recorded, 5 concurrent channels spied max).<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Feb 7, 2012 at 5:48 AM, Tiago Geada <<a href="mailto:tiago.geada@gmail.com">tiago.geada@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-family:"Tahoma","sans-serif";color:#000066'>that means that from 1.4.18 that issue is no longer present ?</span><o:p></o:p></p><div><div><div><p class=MsoNormal>On 7 February 2012 12:44, Jonas Kellens <<a href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>> wrote:<o:p></o:p></p></div></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><div><div><p class=MsoNormal>On 02/07/2012 01:07 PM, Sammy Govind wrote: <o:p></o:p></p><p class=MsoNormal>Hello, <o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded !<o:p></o:p></p></div><div><p class=MsoNormal>Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ?<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'>Using Asterisk 1.6.2.22.<br><br>The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki :<o:p></o:p></p><h3>Attention<o:p></o:p></h3><ul type=disc><li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1'>Up to and including Asterisk 1.4.17 ChanSpy can cause a <strong>crash/segfault</strong> if used together with <a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor" target="_blank" title="Asterisk cmd Monitor">Monitor</a> or <a href="http://www.voip-info.org/wiki/view/MixMonitor" target="_blank" title=MixMonitor>MixMonitor</a> at the same time. 1.4.18 is supposed to attack this issue by using "audiohooks" that replaces the current ChanSpy approach.<o:p></o:p></li></ul><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div><div><p class=MsoNormal>--<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></blockquote></div><p class=MsoNormal><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>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><p class=MsoNormal><br><br clear=all><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>-- <o:p></o:p></p><div><p class=MsoNormal>Carlos Alvarez<o:p></o:p></p></div><div><p class=MsoNormal>TelEvolve<o:p></o:p></p></div><div><p class=MsoNormal>602-889-3003<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>