search for: appkonference

Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul...

I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonference That said Asterisk 13 doesn?t get that much attention because I use Asterisk 1.4 + some hacks. Here?s a link to my Asterisk 1.4 github repository: https://github.com/pjalbrech...

Hi, I have released AppKonference 2.4 today. This release includes a frame cache for asterisk frames which should be more efficient than the asterisk frame cache is for this type of application. It's ulaw/alaw only for now. This release also added a speaker list which the conference thread uses to mix speaking members. This...

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going pa...

....com] On Behalf Of Murthy Gandikota Sent: Monday, December 08, 2014 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Playing audio to bridged channels using ControlPlayBack There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going...

Hi masters, I?m not an expert on this my friends, but I?m trying to understand which the expected behaviour is from Asterisk side when you deal with the following scenario: Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?> extension 1000 GSM gateway with call waiting activated on SIM A Queue with ?skip busy agent? disabled and ringall strategy. SIP extension 1000 with call

Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -------------- next part -------------- An HTML

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin