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.com>wrote: > 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 >> > > > -- > _____________________________________________________________________ > -- 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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