----- Original Message ----- > From: "Richard Kenner" <[email protected]> > To: [email protected] > Sent: Monday, April 25, 2011 8:42:07 AM > Subject: Re: [asterisk-users] The new ConfBridge application is now in > Asterisk Trunk! > > To help get you started, Malcolm Davenport has written some > > fantastic > > documentation on the asterisk.org wiki. It can be found below. > > I've looked at this documentation, but can't find the documentation > the > realtime interface, which is needed in order to schedule conferences > in > the future. Is the documentation missing or is this not part of the > application? We depend on it HEAVILY. >
No, conference scheduling is not a feature that we have built directly into ConfBridge, and I'm debating on what it would look like. This is primarily a result of taking a different approach to the way the conferencing application is configured. Conferences in confbridge are only created dynamically. The scheduling would probably have to live on the user or bridge profile. So a dynamic profile would be created and that profile would have a start and end time. I'll have to think about this more. While it isn't simple, there are existing tools that can be leveraged to achieve similar functionality. For example, func_curl can be used to pull this conference information from a database directly in the dialplan, or an AGI script can be used to determine when a conference is available or not. The hard part would be warning the conference users the conference is about to end. This would require some dialplan cleverness. I suppose joining a local channel into the conference using the Wait and Playback application would work. The local channel could be the marked user, when it enters the conference begins, and when it leaves everyone is kicked. The local channel would just wait for the duration of the conference, and can provide playback warnings towards the end. I hope this helps, If you have any feedback on how you would like this feature to look in with the new ConfBridge configuration system please let us know. If the feature is well understood enough, perhaps someone will come around and develop it before Asterisk 1.10 is released. We are marking this as a feature request on the wiki, Thanks! -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _____________________________________________________________________ -- 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
