Asterisk 1.6 Release Management Proposal

I have published a document describing the details of the release management for Asterisk 1.6. See the full post to the mailing list here.

A few weeks ago, I proposed to this list that we create a new release series that is managed with a short release cycle to introduce smaller sets of new features. I also wanted to increase the emphasis that we put on testing new sets of functionality for potential regressions.

The feedback on this list was positive, as was all of the feedback I have received directly. I spoke to people about this a lot at Astricon, and received no negative feedback.

So, I would like to move ahead with formalizing this new release series, Asterisk 1.6. I have documented the new release policy that will apply to this release series, as well as some of the history that inspired these changes to release management.

I have included the document and would appreciate any feedback from the development community.

Asterisk 1.4.13 Released

The Asterisk Development Team has released version 1.4.13.

This release fixes a couple of security issues in the implementation of IMAP storage for voicemail. One of the issues is remotely exploitable. Any systems that do not use IMAP storage for voicemail are not affected by these issues. For more details on this issue, see the Asterisk security advisory here:

http://downloads.digium.com/pub/asa/AST-2007-022.pdf

This release also contains some other bug fixes that have been merged in the past week or so. The other fixes include resolutions for a few different deadlocks, a couple of problems in res_jabber, chan_sip and RTP fixes, and a few more minor issues. See the ChangeLog for a full listing of the changes:

http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.4.13

Thank you very much for your support!

Asterisk Jitterbuffer support for Applications

In a post on asterisk.org, I described how the generic jitterbuffer works in Asterisk 1.4. Because of the way it was designed, it does not work when a call is connected to an Asterisk application such as Voicemail or MeetMe. It will only work when two channels are bridged together. So, it has been extremely useful for people who are terminating SIP calls to the PSTN, for example.

Someone at Astricon asked me about this, and it made me think of a way to enable the use of the jitterbuffer for calls connected to Asterisk applications. It was quite simple to implement, too. I added support for the generic jitterbuffer to the Local channel driver, which provides a channel interface back into the local Asterisk dialplan.

Since this is a new feature, it was only added to trunk, our development tree, which will soon become Asterisk 1.6. However, I have posted an unofficial backport of the patch to Asterisk 1.4.

To use the feature, you simply use the new ‘j’ option in the Dial() string, in addition to the ‘n’ option that already existed for chan_local. For example:


exten => 5551212,1,Dial(Local/1234@somecontext/nj)


[somecontext]
exten => 1234,1,MeetMe(1234)


See the commit of the code here.

Download instructions for Asterisk 1.4:

$ svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4 1.4-backports

There is a file in there called chan_local_jitterbuffer.patch.txt. To apply the patch:

$ cd asterisk-1.4
$ patch -p0 < /path/to/chan_local_jitterbuffer.patch.txt.
$ make
$ sudo make install


Enjoy!