[Asterisk-Users] What and When is the next version of Asterisk?

[BJ Weschke]

On 5/25/06, Pavel Jezek <pavel.jezek at i.cz> wrote:
> I think, that sip/rtp jitterbuffer is one of the most wanted feature,
> but because still not included in trunk too few peoples improving it...
> what to try include this soon to trunk, and only if problems will be not
> solved before 1.4 release candidate, remove out of asterisk 1.4 ...
> also good candidate to 1.4 is new codec negotiation algorithm, seems be
> actively maintained/finalized
> http://bugs.digium.com/view.php?id=4825
> PJ
>
 PJ,

 I understand what you're saying, but playing devil's advocate, if
this goes into /trunk it becomes part of the 1.4 release. The
community is then tasked with supporting this feature, whether or not
it is ready for prime time. If it's not ready, people then complain
that features are in the release that "don't work" and "weren't
tested". It's kind of a no-win situation.

 The best thing that can be done at this point to work towards trying
to get this feature in is to have people test it in /trunk in their
environments and report results back. If they are developers and can
contribute to code improvements around it, that would be very much
welcomed as well.

-- 
Bird's The Word Technologies, Inc.
http://www.btwtech.com/