[Asterisk-video] [Fwd: [asterisk-dev] Asterisk 1.6.0 branch created]

[Klaus Darilion]

I think we should port h324m and other stuff/patches to 1.6

regards
klaus


-------- Original-Nachricht --------
Betreff: [asterisk-dev] Asterisk 1.6.0 branch created
Datum: Tue, 04 Mar 2008 11:07:00 -0600
Von: Russell Bryant <russell at digium.com>
Antwort an: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
An: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>

Greetings,

I just created a 1.6.0 branch in svn.  What does this mean?

1) Asterisk 1.6.0 is now feature frozen and nearing release candidate 
status.  I
have two issues against Asterisk 1.6 that I would like to resolve first:
    - 12130, G.722 transcoding problems
    - 11972, deadlocks related to SIP TLS

2) Asterisk trunk is now completely open for changes.  There are 
multiple heavy
sets of changes that have been waiting to get merged until 1.6.0 is 
done.  They
can now be merged in to trunk.

3) If you are a committer to the Asterisk repositories, you have 1 more 
place to
merge bug fixes.

    a) Merge fix into 1.4 as usual
    b) Merge up trunk as usual
    c) Merge the bug fix from trunk to 1.6.0 using the "mergetrunk6" wrapper
       that is in repotools.

That's all for now.  Questions or comments are welcome, as always.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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