Asterisk Blog

Asterisk 20Asterisk 21

Sangoma Trunking Module

Sangoma provides two SIP trunking services which are available to customers, SIPStation which is a great solution for the everyday user and VoIP Innovations which ...
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Asterisk 15PJSIP

Upcoming WebRTC Improvements in Asterisk 15

In my previous post I talked about what WebRTC support is like in Asterisk 14. Since Asterisk 15 is going to be released soon let’s ...
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Asterisk 14WebRTC

WebRTC and Asterisk 14

WebRTC has been in Asterisk since Asterisk 11 and over time has evolved just as the WebRTC specification itself has evolved. The WebRTC implementation we ...
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Asterisk 15WebRTC

Asterisk 15 Beta Released

It is with great pleasure I wish to inform the world of the first beta release of the new Asterisk 15 branch. It’s a very ...
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ArchitectureHow To

Asterisk Containers and Objects (ao2)

For those of you budding Asterisk developers out there, you may find yourself browsing the source code. All over, you’ll find function calls prefixed with ...
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People

Interview with new Asterisk developer, Ben Ford

Hello, my name is Matthew Fredrickson and I’m the project lead for the Asterisk project. We recently hired someone new onto the project to help ...
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ArchitectureTesting

Continuous Integration with Asterisk and Docker

How much testing do we really do? A lot, and let’s face it,  Asterisk isn’t the easiest of software packages to test.  Our continuous integration ...
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Architecture

app_queue And The Case Of The In Call Agent

The app_queue module is one of the oldest modules in Asterisk and implements call queueing functionality. You configure a queue with members and a strategy ...
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PJSIPReleases

Recent Security Issues

Security releases of Asterisk were recently created. In this post, we’d like to go into the depths of two of the security issues and how ...
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Testing

Continuous Integration: Helping to find problems.

Continuous integration isn’t something that many people in the Asterisk project think about but it is a critical part of the development of Asterisk. It ...
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PJSIPSIP

rtcp-mux in WebRTC

Do you use WebRTC with Asterisk? Did you notice calls stop working after updating Google Chrome to version 57? Are you curious why that happened? ...
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DocumentationHow To

Asterisk, Opus, packet loss, and FEC

So you’ve heard there is now an Opus codec for Asterisk that’s been released. However, you are having problems with poor audio quality due to ...
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Documentation

Dialplan handler routines allow customization

There are several handler routines available to allow you to customize behavior for the different states of a call. Handler routines execute outside of the ...
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Continuous Integration Update

Back in December in my The Continuing Saga of Continuous Integration blog post I wrote about how we reduced the Testsuite’s “27” layers of file

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