As an update on the Git migration, here is the current state of the world:
1. The SVN repos have been marked read-only. While you will still be able to checkout from SVN, you shouldn't commit to any of the branches. Note that even if you do, those commits won't make it into the Git repos - so it's not really a good idea to try :-)
2. The project has been moved over to a Git repo under Gerrit (https://gerrit.asterisk.org). You can clone the project using the instructions on the 'asterisk' repo project page:
3. Mirrors for the project have been set up on git.asterisk.org and Github. These mirrors are particularly useful for providing source browsing of the repo.
4. A patch has been put up against 'master' to rework the source file version macros. By rework, I mostly mean "remove", although the macro itself could not be fully removed due to other features relying on the file name being registered. See https://gerrit.asterisk.org/#/c/54/ for more details.
So what are some immediate next steps?
1. We need to determine the best way to handle maintaining the long running branches. While rebasing is appropriate for topic branches (team branches) that closely track a mainline branch, the mainline branches are a bit different. They not only don't have ever commit merged into them (either going 'up' from 11 => 13 => master or 'down' from master => 13 => 11), but patches are highly likely to merge cleanly. ABI issues in 11/13 are a bigger concern than those in master; APIs will have changed; etc.
As a result, my initial plan was to have developers cherry-pick to the mainline branches as appropriate, with the initial commit being done to 'master'. There are some downsides to this approach:
a) Each cherry-pick has to be reviewed. This can make it difficult to track the reviews, as each review is completely separate from the others.
b) Cherry-picks through the Gerrit UI will not always work. Folks will need to be careful when cherry-picking back to a branch that requires fixing merge conflicts, as getting the Change ID correct in such a case is important to keep all of the Gerrit reviews tied together.
c) We'll be changing our process from merging 'up' branches to 'down' branches. Change is scary.
I think we're going to need some time to work through the implications of how we handle the mainline branches. I suggest hanging out in #asterisk-dev over the next few days as we work through the details. In the meantime, I've restored the TEST-Asterisk repo so folks can play around with the cherry-picking, and/or other models of branch maintenance. I certainly welcome any suggestions on the best way to make the process work.
2. Russell noted in George's .gitreview/.gitignore review (https://gerrit.asterisk.org/#/c/42/) that we may want to include the fullname of contributors/users along with their e-mail address. I think that's a good idea, but that would mean updating our commit message template, script, and our process. Comments/suggestions welcome here on that proposal.
3. The 'make update' Makefile target needs to be updated. If you have some Makefile-foo and git-foo and would like to look at that, that'd be awesome.
Over the next few days, I'll be updating the Asterisk wiki pages with more information. I'll reply to this thread as that happens. If you have any questions, please don't hesitate to reply here or jump in #asterisk-dev.
Digium, Inc. | Director of Technology
Its been a few months since our update on Asterisk 12, and during that time period, the Asterisk Community has been hard at work enhancing and testing Asterisk both for existing users of Asterisk 12 as well as in preparation for the next Long Term Support release, Asterisk 13. These new features focus heavily on improving the user experience in the new PJSIP stack and enhancing the existing APIs as well as the new Asterisk REST Interface (ARI). Several of these new features have been released recently in versions 12.2.0 and 12.3.0
The A2Billing Team has posted a note asking people to upgrade their systems urgently:
It has been widely reported that a recently discovered bug in OpenSSL compromises the security of communications data. Any A2Billing administrator who has installed an SSL certificate to provide HTTPS encryption on their website should upgrade OpenSSL.
We've also had a 3rd party security expert audit A2Billing in the interests of increasing security. They have found a major security issue. We've no evidence that this exploit is in the wild, but we advise in the strongest possible terms that you upgrade to A2Billing version 2.0.8 as a matter of urgency.
If you need assistance with upgrading your operating system or A2Billing or an installation with transfer of A2Billing data, then please contact us at firstname.lastname@example.org
39 Free Softphones August 14, 2009 Average Vote: 17
I decided to do another round up article, this time focusing on the 39 best free softphones.
Asterisk 11.0.0-beta1 Now Available August 11, 2012 Average Vote: 10
The Asterisk Development Team is pleased to announce the first beta release of Asterisk 11.0.0 (the first Long Term Support release since 1.8).
Asterisk 10.0.0-rc1 Now Available November 11, 2011 Average Vote: 10
The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0.
oFono 1.0 has been released November 10, 2011 Average Vote: 10
Steve Totaro has forwarded details of the latest release of a project called oFono.
Discount for Astricon 2012 August 31, 2012 Average Vote: 10
Astricon 2012 is rapidly approaching and will be held in Atlanta between October 23 and 25. The Daily Asterisk News has secured a 20% discount on all tickets if you use the discount code.
Asterisk and Google August 26, 2011 Average Vote: 10
Malcolm Davenport has posted a request on behalf of Digium for someone to help maintain the Google channel driver when Google makes changes.
espeak module for Asterisk August 22, 2011 Average Vote: 10
Lefteris Zafiris has posted details of a new version of the app_espeak application for Asterisk - another speech synthesizer.
New SIP channel driver project page November 23, 2012 Average Vote: 10
Mark Michelson has posted details of a wiki page that has been created to guide the development of a new SIP channel driver for Asterisk.