Have you ever wanted presence information across multiple Asterisk servers? Well, making that possible is something that I have worked on a bit here and there. I previously made a post asking for some of that code to be tested. Now, this code has been merged in to Asterisk 1.6. Asterisk 1.6.1 will have the ability to share device state between servers using a new module, res_ais, which uses the SAForum AIS to share events between Asterisk servers.

While Asterisk 1.6.0 is currently being stabilized for release, we are also actively developing new features for Asterisk 1.6.1. One feature that has been merged into Asterisk trunk, which will be released in Asterisk 1.6.1, is support for multiple parking lots. This functionality has been available for quite some time from 3rd party modules. So, it’s about time that we got it implemented natively in Asterisk. The code that has been committed was originally written by oej.

In the past week, I have merged patches into Asterisk trunk that provide new features for chan_sip. These features will be available in Asterisk 1.6. TCP and TLS support The Commit In the past, Asterisk has only had support for UDP as a transport for SIP signaling. Asterisk 1.6 will have support for both TCP and TLS, as well. This work was done by Brett Bryant and James Golovich, who were both funded by Digium for their work.

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.

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.

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.

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.

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.

We got a very interesting patch to improve the reliability of the Inband DTMF detector. Some poor debouncing logic was making it very possible to miss a digit when it didn’t start very clean. Here is the commit and also the related discussion on the asterisk-dev mailing list. Many thanks to Tony Mountifield for the patch!