ok,

now i have got some very valuable information to start off with. thank you all.
i´ll be back to report success or further questions...

just one thing, that i think might be a showstopper that i may have not
explained clear enough...:
muting and unmuting a caller should have the effect, that the caller can talk
to the moderator or not... any caller should NEVER hear what other callers
are talking... may he be muted or not...

yves

Am 16.01.2013 23:01, schrieb Danny Nicholas:

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

Exten => s,1,meetme(5555)

The participants would do

Exten => s,1,meetme(5555,m) -- 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

The moderator would need to be able to enumerate the conference by doing

Asterisk --rx "core show channels verbose"|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

Establish a maximum number of users and set this up for each one

Exten => 99,1,meetmeadmin(5555,M,1) let user 1 talk

Exten => 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 => 901,1,chanspy(SIP/XXX,w)
- the modetator must be able to kick off any caller at any time...

Exten => 299,1,meetmeadmin(5555,k,1) kick out user 1

Exten => 666,1,meetmeadmin(5555,K) shut it down

*From:*[email protected] [mailto:[email protected]] *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

Sounds like a conference with all attendees permanently muted (except the "moderator").

The moderator uses "whisper" to communicate with individuals.

--Don

*From:*[email protected] <mailto:[email protected]> [mailto:[email protected]] <mailto:[mailto:[email protected]]> *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

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?
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." <[email protected]
    <mailto:[email protected]>> 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 byhttp://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

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to