[Asterisk-Users] RE: [Asterisk-biz] Asterisk as a Voice ConferenceServer

[BJ Weschke]

 We've merged in 3599 and 4252 against a version of HEAD from around
the April timeframe of this year.

On 11/3/05, Patrick <asterisk at puzzled.xs4all.nl> wrote:
> On Thu, 2005-11-03 at 09:31 -0500, BJ Weschke wrote:
> >  We're using SIP exclusively. We do use the meetme features that have
> > enter/leave sounds and name announcement and we've taken alot of the
> > patches (putting the playback of conference-wide announcements) and
> > integrated them in even though those patches were not merged with the
> > CVS-HEAD tree from the bug tracker.
>
> Which patches from bugs.digium.com or elsewhere did you apply? I have
> found the following bug reports:
>
> Audio delay in MeetMe using SIP when not 'q' mode
> http://bugs.digium.com/view.php?id=3599
>
> Increasing delay over time on non-Zap channels in MeetMe
> http://bugs.digium.com/view.php?id=4252
>
> Asynchronous generation of outgoing frames when timing device available
> http://bugs.digium.com/view.php?id=5374
>
> MeetMe doesn't recreate pseudo when Local channel masquerades back to
> non-Zap channel (already fixed in cvs around 9/25)
> http://bugs.digium.com/view.php?id=5274
>
>
> >  Clients have been happy with the results thus far.
>
> Good for you :)
>
> Regards,
> Patrick
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