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 Freeswitch (the other option being Asterisk). 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 Freeswitch 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 (which I know can already be done with the "relate" command) - 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 - 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 Freeswitch 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