Yeah!! Been a while but I will make some time to look at any needed doxygen updates or phoneprov stuffs.
On Fri, Jul 20, 2018 at 11:34 AM George Joseph <[email protected]> wrote: > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev -- - Andrew "lathama" Latham -
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