[asterisk-dev] Asterisk 16 is now a Thing!

George Joseph gjoseph at digium.com
Fri Jul 20 07:47:59 CDT 2018


Yesterday we cut the 16 branch of Asterisk from master so it's now
available in the gerrit repo at https://gerrit.asterisk.org/asterisk

We also cut the 16.0 branch which will be the branch for the 16.0.0-rc1
release due in the next 2 weeks.

If you have patches currently up for review, you should cherry-pick them to
the 16 branch (not 16.0) and we'll work with you to get them in an RC and
the final 16.0.0 release.  Any new patches should also be cherry-picked to
16 but unless they fix regressions or some other critical issue, they will
not be included in 16.0.0.

If you are a developer with custom code or modules, now is the time to make
sure your stuff works with 16.  There HAVE been some changes, especially in
module loading, that may affect you so review the CHANGES and UPGRADE files
and test.

If you're a packager, you could wait for the rc1 tarball but we'd suggest
starting early to make sure that at least you can compile,

The external codecs (g729a, opus, silk, siren7, siren14) are now available
on the downloads site (http://downloads.digium.com/pub/telephony/) for
Asterisk 16.  res_digium_phone (DPMA) is also available but it may have
issues when used with an Asterisk installation configured with
--enable-dev-mode.  We're working on that now.

As always, feedback of any kind is appreciated.

-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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