how can i do things with a user after he entered a conference? in the dialplan? via script? ... like unmuting him.
you can do most of the stuff from event_socket, monitor the specific conference event and control it from there.
a more general question is how to play no-sound for certain events. if there is something like: <param name="kicked-sound" value="conference/conf-kicked.wav"/> can i set it to something like: <param name="kicked-sound" value="nosound"/> ?? %<
I think empty out the value parm should do it. so, put this in your conference profile <param name="kicked-sound" value=""/> On Fri, Aug 8, 2008 at 3:18 PM, Michael Collins <mcollins@fcnetwork.com>wrote:
-----Original Message----- From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev- bounces@lists.freeswitch.org] On Behalf Of t Sent: Friday, August 08, 2008 3:24 AM To: freeswitch-dev@lists.freeswitch.org Subject: [Freeswitch-dev] conference with background sound + some details ont
i already asked this on the fs-users list. so i like to ask the developers if they can give me some hints.
%< i like to setup conferences with some ambient music in the background.
is there a background sound parameter?
when i use the perpetual sound with local_stream its fine. but all the users have unmute themselve first.
how can i do things with a user after he entered a conference? in the dialplan? via script? ... like unmuting him.
a more general question is how to play no-sound for certain events. if there is something like: <param name="kicked-sound" value="conference/conf-kicked.wav"/> can i set it to something like: <param name="kicked-sound" value="nosound"/> ?? %<
... some more background about me ...
i thought it is a really common setup to have conferences with backgroundchannels.
by thinking deeper about the idea with queuebased conferencing and room-oriented ivr-system (voice-muds) i started the project blabla (but its still a on a very beginning) [python + yate] -- https://blabla.bountysource.com/
i also took part on the idea of VARE (virtual a rendering engines) which is hyperlinked conferencing. or a base for some "audio-web". http://blog.justdigits.de/?cat=11 <-- sorry german but basically a audiointerpretation of documents like http://justdigits.de/vare/Erdgeschoss.vare#S by some 16 channel mixer ontop of asterisk and xmpp.
since 1 year i work on something called dialstation. which is php+rubyonrails+freeswitch. more about this can you see on my talk on asterisktag. [where i also announced the release of FreeSWITCH 1.0 ;-) ] http://www.asterisk-tag.org/wiki/Programm_27.05.2008_(Mixed_Track)<http://www.asterisk-tag.org/wiki/Programm_27.05.2008_%28Mixed_Track%29>
thats for now. lets see who i meet here on this list.
t. t,
I think the devs might have some questions for you about what you are trying to implement. I think your best bet is to get on the IRC channel next week when everyone is back from ClueCon. (All of the devs and a good chunk of the community were in Chicago for most of this week and the Internet connection was terrible. Things should get back to normal next week.)
-MC
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