[Asterisk-Dev] Re: MeetMe max users and cutoff time

Eric Wieling eric at fnords.org
Thu Feb 26 15:39:38 MST 2004


Notice the option to dynamically create a conference.  I do, however,
know if MeetMeCount works on dynamic conferences.

fs-1*CLI> show application meetme
fs-1*CLI>
  -= Info about application 'MeetMe' =-
                                                                                                                              
[Synopsis]:
Simple MeetMe conference bridge
                                                                                                                              
[Description]:
  MeetMe(confno[|options]): Enters the user into a specified MeetMe conference.
If the conference number is omitted, the user will be prompted to enter
one.  This application always returns -1.  A ZAPTEL INTERFACE MUST BE INSTALLED
FOR CONFERENCING TO WORK!
                                                                                                                              
The option string may contain zero or more of the following characters:
      'm' -- set monitor only mode (Listen only, no talking
      't' -- set talk only mode. (Talk only, no listening)
      'p' -- allow user to exit the conference by pressing '#'
      'd' -- dynamically add conference
      'v' -- video mode
      'q' -- quiet mode (don't play enter/leave sounds)
      'M' -- enable music on hold when the conference has a single caller
      'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
              Default: conf-background.agi
             (Note: This does not work with non-Zap channels in the same conference)
      Not implemented yet:
      's' -- send user to admin/user menu if '*' is received
      'a' -- set admin mode
                                                                                                                              
fs-1*CLI>



On Thu, 2004-02-26 at 15:35, Tony Mountifield wrote:
> In article <1077815104.5211.13.camel at critch>,
> Steven Critchfield <asterisk-dev at lists.digium.com> wrote:
> > On Thu, 2004-02-26 at 10:34, Tony Mountifield wrote:
> > > 
> > > I'm working on a conference bridging application, which will update
> > > meetme.conf as conferences are created and deleted. I am planning to
> > > enhance the meetme.conf format to enable rooms to specify a maximum
> > > participant count (with a suitable message to excess participants).
> > > 
> > > I'm also looking to specify an expiry time for a room, as either a
> > > "soft" expiry (no new participants can join after the expiry, but
> > > the existing members can continue till they leave), or a "hard"
> > > expiry (at the deadline, all participants receive a message and
> > > a hangup).
> > > 
> > > I don't anticipate a problem implementing either of these, but I would
> > > be interested in people's thoughts as to the best approach.
> > 
> > You can use Meetmecount to get number of users in a conference and are
> > able to use a goto to either enter a conference or not. Since you cane
> > do conditional gotos based on time already in the dial plan, that is
> > already covered.  Hard expiry could be dealt with by computing time and
> > setting a timeout or developing a side app that does a transfer on the
> > members via the manager interface at specified times.
> 
> For fixed conference numbers that is true, but MeetMeCount needs a room
> number as an argument, within extensions.conf (AFAIK). I want one access
> extension for all conferences, and I call MeetMe without a room number,
> so that the caller is prompted for it interactively.
> 
> The fact that meetme.conf is read afresh each time someone joins a
> conference is great - I can have a process that just rewrites the file
> each time a room needs to be created or deleted.
> 
> But I don't see any way to use MeetMeCount in this context, which is
> why I'm thinking of enhancing the code. Similarly for conditional gotos
> based on time: I would need them to be room specific on a common extension.
> 
> Cheers,
> Tony
-- 
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
"Unofficial Links" section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

BTEL Consulting




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