Inside the Asterisk

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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Happy New Year – A Community Update

Hello everyone…for those of you who don’t know me, I’m Jared Smith, Sangoma’s new Vice President for Open Source Community Development. Ever since I first started using open source telephony software back in 2002, I’ve tried to give back to the community. I’ve done Asterisk consulting, I’ve written Asterisk documentation, I’ve taught Asterisk training classes,

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SIP and RTP Routing

One of the most common issues I see when people deploy SIP is calls hanging up after approximately 30 seconds or traffic not going to where it should. This can be hard for users to grasp and is primarily due to the fact that SIP embeds routing information (IP addresses and ports) within the signaling

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Year End Message to our Open Source Community

Hello Everyone, As a public company CEO, I write a “Letter to Shareholders” once per year, leading up to our Annual General Meeting. But shareholders are one of the multiple ‘key constituencies’ at Sangoma (others would include customers, employees, and yes, our valued open source community). So, as 2019 draws to a close, I thought

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AstriDevCon 2019 – A Recap

Greetings All, Time certainly flies and it seems like AstriDevCon and AstriCon 2019 were a century ago. It’s hard to imagine that in fact they were only a month and a half ago! Since everyone has had some time to recharge I thought now was a great time to review and talk about what was

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Building a Channel Driver – Part 1

Overview Today we’re going to be talking about channel drivers and how to get started on creating your own. This topic is going to be covered in three separate blog posts, so keep an eye out for the next two! In the first one, we will cover the following: some basic tips, some template code,

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Announcing a New Compile Option for app_voicemail Storage.

Asterisk offers its users great flexibility in most of its features. One of them is the choice between three different modules for different ways to store voicemail. These include: To a file  (app_voicemail – default) To an ODBC database (app_voicemail_odbc) To IMAP (app_voicemail_imap) The good news for Asterisk administrators is starting with version 17, we

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