Yeah, was looking at this - my "issue" is that the dialplan is already running (the channel is already bridged to a SIP phone), so how do I tell it *which* channel to pause ? <div><br></div><div>Julian<br><br><div class="gmail_quote">
2010/1/25 Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Check this link</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"><a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor</a></span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Depending on your release, you can “pause”
and “un-pause” monitoring.</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
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<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight:bold">On Behalf Of </span></b>Julian Lyndon-Smith<br>
<b><span style="font-weight:bold">Sent:</span></b> Monday, January 25, 2010
8:22 AM<br>
<b><span style="font-weight:bold">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight:bold">Subject:</span></b> [asterisk-users] Call
recordings and sensitive information</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">During a telephone conversation with a customer, they sometimes give
card details over the phone. under the pci-dss regulations we are not allowed
to record the conversation where the details are being given. Is there a
"mute command" or pause that can be sent to MixMonitor ?</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">How has anyone else solved this issue ?</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">Many thanks</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">Julian</span></font></p>
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