<div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Jun 12, 2013 at 3:21 AM, Avi Marcus <span dir="ltr"><<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>But the media is handled by FreeSWITCH, so it doesn't really matter if sip supports it... It's something theoretically FreeSWITCH could handle for you...</p>
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<p>-Avi</p></font></span></blockquote><div>This is a classic case of a feature that could/should be handled by the phone itself but since no two manufacturers would ever do it the exact same way it falls to the server to do the dirty work. <br>
<br>I suppose the most elegant solution would be a mod_dptools app that knew exactly what to do. However, that would require a bit of programming skill and some time and patience.<br><br>An alternative would be silently dropping all 3 parties into a conference and using the "relate" feature where you can define who hears whom. However, I don't know if the transition into the conference room would cause a blip in the audio or not, which is why an explicit "whisper_page" dp app would be best.<br>
<br></div><div>-MC<br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On Jun 12, 2013 12:17 PM, "Jeff Bernhardt" <<a href="mailto:jeff@askcornerstone.net" target="_blank">jeff@askcornerstone.net</a>> wrote:<br type="attribution"><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"><span style="font-size:10.0pt">Yes, we have a customer who used to work in a place with an old Panasonic PBX and Panasonic phones that had this feature and he loved it. His secretary could whisper in his handset, he’d hit a key and be able
to talk to the secretary really quick without the other party even knowing he’d done anything. We were told by a guy who used to consult for us on Asterisk stuff that this isn’t possible due to the nature of how SIP works, but we’ve always kind of wondered.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Jeff Bernhardt<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Systems Administrator<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Cornerstone Consulting<u></u><u></u></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><u></u> </span></p></div></div></blockquote></div></div></div></blockquote><br clear="all"></div><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br>
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