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