December 31, 2007

New Channel Driver: chan_console

I just merged a new channel driver into Asterisk trunk which will be in Asterisk 1.6. The module is called chan_console. It is a new console channel driver which uses portaudio as a cross platform audio interface instead of using something like ALSA or OSS directly. I wrote it to give myself a console channel driver that I could use on my Mac. However, portaudio supports a number of other audio interfaces, as well.

December 31, 2007

team/russell/chan_refcount: Improving Asterisk Performance

I have a branch that is ready for testing that makes some significant changes to the Asterisk channel handling core. The changes improve the data structure management for Asterisk channels. This will provide a large performance benefit. See the following post to the asterisk-dev mailing list for more information, as well as where to get the code for testing. Asterisk-dev - Request for Testing: team/russell/chan_refcount

December 18, 2007

Asterisk as a video soft phone

Asterisk trunk recently got a pretty cool new feature. You can now use Asterisk as a highly configurable video soft phone. The commit to trunk is here. The way it works is pretty neat. Asterisk already had a couple of console channel drivers: chan_oss and chan_alsa. These channel drivers allow you to use a local OSS or ALSA sound device as an endpoint for a call. These interfaces are commonly used to interface with overhead paging systems.

November 11, 2007

chan_unistim: Nortel IP Phones with Asterisk

A new channel driver has been committed to Asterisk trunk which allows you to use Nortel IP phones with Asterisk. It was submitted by Cedric Hans. See the commit and the original mantis issue. This module will be available in Asterisk 1.6. There is some basic documentation and a sample configuration file available. Nortel phones which have been verified to work with this module are the Nortel i2002, i2004 and i2050.

November 11, 2007

pbx_lua: Asterisk Dialplan in Lua

Recently, a new module for writing Asterisk dialplan in the Lua programming language was merged into Asterisk trunk. It was developed by Matt Nicholson of Digium, Inc. See the commit and mantis issue. It will be available in Asterisk 1.6. From lua.org: Lua is a powerful, fast, light-weight, embeddable scripting language. Lua combines simple procedural syntax with powerful data description constructs based on associative arrays and extensible semantics. Lua is dynamically typed, runs by interpreting bytecode for a register-based virtual machine, and has automatic memory management with incremental garbage collection, making it ideal for configuration, scripting, and rapid prototyping.

October 18, 2007

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.

October 10, 2007

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.

October 9, 2007

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.

September 27, 2007

Digium Acquires Switchvox, Open Source Community to Benefit

I am extremely proud to be able to point out that Digium has acquired Switchvox. Switchvox is an amazing packaged PBX system that is based on Asterisk. A team from Digium did an extensive analysis of the products available in this area, and concluded that the Switchvox package was superior in terms of features as well as ease of use. Check out the Switchvox features and some screenshots. Part of the plan is to work with the Switchvox team to take parts of the technology they have developed, and contribute it back to the open source community.

September 20, 2007

Thoughts on Asterisk release management

The process of developing, releasing, and maintaining Asterisk 1.4 has certainly been a learning experience. I have been putting a lot of thought into the things that we have been dealing with and would like to propose some changes to the way that we manage releases. Over the past few years we have gone from not having managed releases, to Asterisk 1.0, 1.2, and now 1.4. Over this time period we have transitioned from everyone using the development code directly to now nobody using the development code for any real purpose.