Inside the Asterisk

Keyword: development

Asterisk-18.0.0-rc1 Released!

Greetings Everyone! Another year has flown by and while we now live in interesting times this has not stopped the development and progress of Asterisk or the Astricon conference[1], so much so that the first release candidate of Asterisk 18 is here[2]. The first step in creating the 18.0.0 release of Asterisk is tagging a

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Local Channel Multistream and Re-Negotiation Support

When stream support was initially added to Asterisk we did it in the most backwards compatible way possible to ensure that we did not have to modify the entirety of Asterisk. This has allowed us to gradually improve parts of Asterisk as we’ve expanded our stream and video support. To that end the next part

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Asterisk Versioning and Releases

Prompted by a post on the Asterisk users mailing list I thought I’d write a bit about Asterisk versioning and releases for this blog post. The Asterisk project at any given time has between 2 and 3 major versions of Asterisk being released. When I say major version I’m referring to an entirely different branch

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A Little Glimpse Into ConfBridge

The ConfBridge dialplan application is used countless times each day by people to do conferencing. Long ago this was done using the MeetMe dialplan application that utilized DAHDI functionality but with the requirement on DAHDI for it to operate an alternative was needed and ConfBridge came into existence. Have you ever wondered how it actually

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Building a Channel Driver – Part 3

Review If you’ve been following this blog post series, then you should have a channel driver that’s ready to be integrated with ARI. If not, then check out Part 1 and Part 2 first. The purpose of this final chapter in the series will be to get your channel driver working with ARI, which is

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Issue Walkthrough: DTLS, and the Case of Bad Audio

A past post discusses some tips and tricks to employ when encountering a problem. Here we’ll walk through an issue applying some of those techniques. Recently an Asterisk issue came up involving occasional static and/or silence for audio. Broadly speaking the problem can be described as Alice calls Bob (using SIP), Bob answers, the call

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Adding and Removing Media Streams

When stream support was added to Asterisk it was initially done with the focus being for SFU with a single video stream from each participant with the call starting out with video. This is a use case which is useful for a lot of people and has worked well. Coming soon, however, is the ability

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PJSIP Body Generator Persistence

When PJSIP publish and subscribe functionality was created we knew we wanted to provide a pluggable mechanism to allow modules to easily extend and add new bodies. The result of this is what is known as body generators. Given a set of data they convert it into a format expected by a device, such as

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Building a Channel Driver – Part 2

Review This blog post is the follow up to part 1, which can be found here. If you haven’t read it yet, that would be a good place to start, especially if you want to build your own channel driver. Here’s a recap of what we’ve done so far. We created chan_groovy.c, res_groovy.c, and res_groovy.h,

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