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

[Patrick]

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