Re: [asterisk-users] Question About Conferencing Capabilities

2011-01-04 Thread 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.plugge...@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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Re: [asterisk-users] Question About Conferencing Capabilities

2011-01-04 Thread Siobhan Hamilton
Anyone else know about the holding concurrent conferences (and switching
back and forth) issue ?  Is it possible?
And can you set up dynamic conferences that continue even when the initiator
leaves?

Thanks!


On Tue, Jan 4, 2011 at 7:11 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:

 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.plugge...@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

 --
 _
 -- 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



 --
 _
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 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:
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[asterisk-users] Question About Conferencing Capabilities

2011-01-03 Thread Siobhan Hamilton
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
--
_
-- 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

[asterisk-users] Question About Conferencing Capabilities

2011-01-03 Thread Siobhan Hamilton
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
--
_
-- 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