[asterisk-users] Question About Conferencing Capabilities

[DHAVAL INDRODIYA]

Hi Siobhan,

Asterisk is all capacity to work-on but you need to find out some way of
handling conference system through WEB part , also one more thing on last
point for switching between conference
i am not much sure about it but i think it is possible if i will look into
code implementation.

regards
dhaval

On Tue, Jan 4, 2011 at 10:34 AM, Siobhan Hamilton <
siobhan.pluggedin at gmail.com> wrote:

> My company is building a VOIP application, and initially were just using a
> barebones OpenSIPS implementation to host one-on-one calls; however, we want
> to expand the functionality to conferencing (which, of course, OpenSIPS
> doesn't handle) and was looking into Asterisk (the other option being
> Freeswitch).  I've been poring through the docs, and have even set up a test
> server myself, but there are some very specific things we are looking for
> that I can't figure out if Asterisk can do or not.
>
> We want to be able to do the following:
> - Create dynamic, on-the-fly conferences that can remain active even when
> initiating user leaves
> - Within a conference, give users the ability to mute and/or deaf
> individual users
> - Give users the ability to enter a "whisper" mode with another user -
> where they are holding a private conversation that can only be heard by the
> two of them ( It sounds like the Meetme module has a functionality like
> this, but it is a little vague in the documentation....)
> - Allow users to be in two conferences at once; the user would most likely
> have one muted at any given time so as to hear the other one, but we want
> them to be able to switch back and forth easily
>
> Could anyone advise me on whether Asterisk can accomplish these needs, or
> perhaps what it might take to do so?  We are not averse to doing some
> customization if we can find the people who know how to make it happen!
>
> Thanks,
> Siobhan Hamilton
>
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