built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D> s,1,meetme(5555,m) =96 muted so they can listen but not talk - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator =20 The moderator would need to be able to enumerate the conference by doing Asterisk =96rx =93core show channels verbose=94|grep meetme This is supposed to be doable from the dialplan but my google-fu failed me on it. - the moderator must be able to mute and unmute any caller at any time =20 Establish a maximum number of users and set this up for each one Exten =3D> 99,1,meetmeadmin(5555,M,1) let user 1 talk Exten =3D> 199,1,meetmeadmin(5555,m,1) turn user 1 back off - the moderator must be able to talk to all callers or to a specific caller. Exten =3D> 901,1,chanspy(SIP/XXX,w) - the modetator must be able to kick off any caller at any time... Exten =3D> 299,1,meetmeadmin(5555,k,1) kick out user 1 Exten =3D> 666,1,meetmeadmin(5555,K) shut it down =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Don Kelly Sent: Wednesday, January 16, 2013 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] special conference room =20 Sounds like a conference with all attendees permanently muted (except the =93moderator=94). =20 The moderator uses =93whisper=94 to communicate with individuals. --Don =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yves A. Sent: Wednesday, January 16, 2013 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] special conference room =20 barat and danny, thank you for your input... I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and can help me a lot... but as you already said... "does _almost_ all features..."... unfortunately I need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the "mute" switch to disable the callers to talk to the conference.. (so to say it mutes the callers microphone, not his earphones.... am I wrong?=20 nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all feature you ask for in it On Jan 16, 2013 5:37 AM, "Yves A." <yves030 at gmx.de> wrote: Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello =20 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users =20 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello =20 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users =20 ------=_NextPart_000_0254_01CDF407.F80A37A0 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; charset=3Diso-8859-1"> <html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-microsoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http://www.w3.org/TR/REC-html40"><head><meta name=3DGenerator content=3D"Microsoft Word 14 (filtered medium)"><style><!-- /* Font Definitions */ @font-face {font-family:Calibri; panose-1:2 15 5 2 2 2 4 3 2 4;} @font-face {font-family:Tahoma; panose-1:2 11 6 4 3 5 4 4 2 4;} @font-face {font-family:Consolas; panose-1:2 11 6 9 2 2 4 3 2 4;} /* Style Definitions */ p.MsoNormal, li.MsoNormal, div.MsoNormal {margin:0in; margin-bottom:.0001pt; font-size:12.0pt; font-family:"Times New Roman","serif"; color:black;} a:link, span.MsoHyperlink {mso-style-priority:99; color:blue; text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed {mso-style-priority:99; color:purple; text-decoration:underline;} p {mso-style-priority:99; mso-margin-top-alt:auto; margin-right:0in; mso-margin-bottom-alt:auto; margin-left:0in; font-size:12.0pt; font-family:"Times New Roman","serif"; color:black;} pre {mso-style-priority:99; mso-style-link:"HTML Preformatted Char"; margin:0in; margin-bottom:.0001pt; font-size:10.0pt; font-family:"Courier New"; color:black;} p.MsoAcetate, li.MsoAcetate, div.MsoAcetate {mso-style-priority:99; mso-style-link:"Balloon Text Char"; margin:0in; margin-bottom:.0001pt; font-size:8.0pt; font-family:"Tahoma","sans-serif"; color:black;} span.HTMLPreformattedChar {mso-style-name:"HTML Preformatted Char"; mso-style-priority:99; mso-style-link:"HTML Preformatted"; font-family:Consolas; color:black;} span.BalloonTextChar {mso-style-name:"Balloon Text Char"; mso-style-priority:99; mso-style-link:"Balloon Text"; font-family:"Tahoma","sans-serif"; color:black;} span.EmailStyle22 {mso-style-type:personal; font-family:"Calibri","sans-serif"; color:#1F497D;} span.EmailStyle23 {mso-style-type:personal; font-family:"Calibri","sans-serif"; color:#1F497D;} span.EmailStyle24 {mso-style-type:personal-reply; font-family:"Calibri","sans-serif"; color:#1F497D;} .MsoChpDefault {mso-style-type:export-only; font-size:10.0pt;} @page WordSection1 {size:8.5in 11.0in; margin:1.0in 1.0in 1.0in 1.0in;} div.WordSection1 {page:WordSection1;} --></style><!--[if gte mso 9]><xml> <o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" /> </xml><![endif]--><!--[if gte mso 9]><xml> <o:shapelayout v:ext=3D"edit"> <o:idmap v:ext=3D"edit" data=3D"1" /> </o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite lang=3DEN-US link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>You could set up the caller meetme where the user presses 1 to <o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Exit the conference<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Whisper to the moderator<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Rejoin the conference<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] <b>On Behalf Of </b>Yves A.<br><b>Sent:</b> Wednesday, January 16, 2013 4:35 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] special conference room<o:p></o:p></span></p></div></div><p class=3DMsoNormal><o:p> </o:p></p><div><p class=3DMsoNormal>ok,<br><br>now i have got some very valuable information to start off with. thank you all.<br>i=B4ll be back to report success or further questions...<br><br>just one thing, that i think might be a showstopper that i may have not<br>explained clear enough...:<br>muting and unmuting a caller should have the effect, that the caller can talk <br>to the moderator or not... any caller should NEVER hear what other callers<br>are talking... may he be muted or not...<br><br>yves<br><br>Am 16.01.2013 23:01, schrieb Danny Nicholas:<o:p></o:p></p></div><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>From what I read, neither confbridge or meetme have the whisper feature built-in; This doesn’t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web interface. Let’s say Yves’ “special conference” is 5555. The moderator would start using this command</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Exten =3D> s,1,meetme(5555)</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The participants would do</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Exten =3D> s,1,meetme(5555,m) – muted so they can listen but not talk</span><o:p></o:p></p><p class=3DMsoNormal>- there is one admin / moderator and several "normal" callers.<br>- the callers must not hear any other caller, only the moderator<o:p></o:p></p><p class=3DMsoNormal> <o:p></o:p></p><p class=3DMsoNormal>The moderator would need to be able to enumerate the conference by doing<o:p></o:p></p><p class=3DMsoNormal>Asterisk –rx “core show channels verbose”|grep meetme<o:p></o:p></p><p class=3DMsoNormal>This is supposed to be doable from the dialplan but my google-fu failed me on it.<br>- the moderator must be able to mute and unmute any caller at any time<o:p></o:p></p><p class=3DMsoNormal> <o:p></o:p></p><p class=3DMsoNormal>Establish a maximum number of users and set this up for each one<o:p></o:p></p><p class=3DMsoNormal>Exten =3D> 99,1,meetmeadmin(5555,M,1) let user 1 talk<o:p></o:p></p><p class=3DMsoNormal>Exten =3D> 199,1,meetmeadmin(5555,m,1) turn user 1 back off<br>- the moderator must be able to talk to all callers or to a specific caller.<o:p></o:p></p><p class=3DMsoNormal>Exten =3D> 901,1,chanspy(SIP/XXX,w)<br>- the modetator must be able to kick off any caller at any time...<o:p></o:p></p><p class=3DMsoNormal>Exten =3D> 299,1,meetmeadmin(5555,k,1) kick out user 1<o:p></o:p></p><p class=3DMsoNormal>Exten =3D> 666,1,meetmeadmin(5555,K) shut it down<br><br><br><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><div><div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> <a href=3D"mailto:asterisk-users-bounces at lists.digium.com">asterisk-users-bounces at lists.digium.com</a> [<a href=3D"mailto:asterisk-users-bounces at lists.digium.com">mailto:asterisk-users-bounces at lists.digium.com</a>] <b>On Behalf Of </b>Don Kelly<br><b>Sent:</b> Wednesday, January 16, 2013 3:34 PM<br><b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br><b>Subject:</b> Re: [asterisk-users] special conference room</span><o:p></o:p></p></div></div><p class=3DMsoNormal> <o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Sounds like a conference with all attendees permanently muted (except the “moderator”).</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The moderator uses “whisper” to communicate with individuals.</span><o:p></o:p></p><div><p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=3DEN style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'>--Don</span><o:p></o:p></p></div><div><div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><span lang=3DEN style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> <a href=3D"mailto:asterisk-users-bounces at lists.digium.com">asterisk-users-bounces at lists.digium.com</a> <a href=3D"mailto:[mailto:asterisk-users-bounces at lists.digium.com]">[mailto:asterisk-users-bounces at lists.digium.com]</a> <b>On Behalf Of </b>Yves A.<br><b>Sent:</b> Wednesday, January 16, 2013 3:11 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] special conference room</span><o:p></o:p></p></div></div><p class=3DMsoNormal> <o:p></o:p></p><div><p class=3DMsoNormal>barat and danny,<br><br>thank you for your input...<br>I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and<br>can help me a lot... but as you already said... "does _almost_ all features..."... unfortunately I<br>need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just<br>from reading the doc I realized, that it wont fit all my needs...<br>btw.: I understood the "mute" switch to disable the callers to talk to the conference.. (so to say<br>it mutes the callers microphone, not his earphones.... am I wrong? <br>nevertheless... any more hints for my original feature-request?<br><br>thank you all,<br>yves<br><br><br>Am 16.01.2013 19:03, schrieb Bharat Lalcheta:<o:p></o:p></p></div><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p>Please study meetme application's options. You will get almost all feature you ask for in it<o:p></o:p></p><div><p class=3DMsoNormal>On Jan 16, 2013 5:37 AM, "Yves A." <<a href=3D"mailto:yves030 at gmx.de">yves030 at gmx.de</a>> wrote:<o:p></o:p></p><p class=3DMsoNormal>Hi list,<br><br>I am in need of a "special" asterisk conference room with the following constraints:<br><br>- there is one admin / moderator and several "normal" callers.<br>- the callers must not hear any other caller, only the moderator<br>- the moderator must be able to mute and unmute any caller at any time<br>- the moderator must be able to talk to all callers or to a specific caller.<br>- the modetator must be able to kick off any caller at any time...<br><br>Any hints on how to realize that are highly appreciated..<br><br>Thanx in advance,<br>yves<br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href=3D"http://www.api-digital.com" target=3D"_blank">http://www.api-digital.com</a> --<br>New to Asterisk? 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Join us for a live introductory webinar every Thurs:<o:p></o:p></pre><pre> <a href=3D"http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><o:p></o:p></pre><pre> <o:p></o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a href=3D"http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre></blockquote><p class=3DMsoNormal> <o:p></o:p></p><p class=3DMsoNormal><br><br><br><o:p></o:p></p><pre>--<o:p></o:p></pre><pre>_____________________________________________________________________<o:p></o:p></pre><pre>-- Bandwidth and Colocation Provided by <a href=3D"http://www.api-digital.com">http://www.api-digital.com</a> --<o:p></o:p></pre><pre>New to Asterisk? 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