Inside the Asterisk

Improved PJSIP Qualify Support Performance

One of the most difficult things in PJSIP is ensuring that the experience is the best it can be for not just people who configure their Asterisk from normal configuration files but also from a database. This presents quite a challenge and one of the areas that has been problematic has been qualify support. Qualify

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Using DEBUG_THREADS to find deadlocks.

Asterisk’s DEBUG_THREADS is a compile time tool that helps find deadlocks involving Asterisk locks. You enable DEBUG_THREADS in menuselect’s “Compiler Flags” menu along with other useful compile time options like DONT_OPTIMIZE and BETTER_BACKTRACES. It is strongly recommended that you enable BETTER_BACKTRACES for the output of the Command Line Interface (CLI) “core show locks” command to

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Asterisk Test Suite: Building Better Tests

Let’s talk some more about testing, understanding the test framework for Asterisk, and building better tests. An exciting topic I know! In a previous post, we discussed how unit testing Asterisk worked. Here, though we’ll be talking a bit about the Asterisk Test Suite. The Asterisk Test Suite is a way to write automated, functional, “black-box”

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Meet Chris Savinovich!

Earlier this year, the Asterisk team at Digium got a little bit bigger!  For today’s blog post, I’m going to interview Chris Savinovich, the most recent addition to Digium’s Asterisk development team with the intent that all of you may get to know him a little better. [Matt] Hey Chris, first off, we’re super excited

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Enhanced Messaging in Asterisk 13 and 15

The next releases of Asterisk 13 and 15 extend MESSAGE support in chan_pjsip and add it to conference bridges.  While Asterisk has supported the SIP MESSAGE method in both chan_sip and chan_pjsip for some time, with this enhancement, if a conference bridge participant (connected via chan_pjsip) sends an in-dialog MESSAGE to a conference bridge, the

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Receiver Estimated Maximum Bitrate Support

For the last few months I, along with Ben Ford, have been working on improving the user experience side of the WebRTC support in Asterisk. When one thinks of user experience the first thing that comes to mind is usually a user interface but in this context I’m referring to underlying technology. Ben has been

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RTP: retransmission for video to combat packet loss

Introduction Packet loss can be an annoying problem when dealing with real time communication, especially when dealing with video. It’s very noticeable when the screen freezes for multiple seconds, then the footage resumes with everything in a completely different position than it was originally. We’ve all seen this before. Packet loss is inevitable, but it

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It’s getting to be that time of year!

With the changing weather, some of us are thinking about summer with excitement, others about winter with trepidation.  Regardless of which hemisphere you live in though, it’s time to start thinking about Asterisk 16!

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A Call to Action!

Hey Everybody! For those of you that don’t know me, my name is Matthew Fredrickson and I’m the Asterisk Open Source Project Lead. Does that mean I’m the best software developer or most prolific committer to the project? No. Does it mean I can answer every single question about using Asterisk? Not necessarily. Oftentimes it

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WebRTC and Asterisk: When It Goes Wrong

Like many things WebRTC is a complex stack of technology within Asterisk and also within the browser. The result of this is that to the best of our ability it doesn’t always work. The browser can change things, the network can stop things from working, the Javascript client may have an issue. This blog post

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