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 16Project General

Asterisk-16.0.0-rc1 Released!

Hello Everybody! Can you believe how quickly a year has passed? Time is drawing near to many exciting things – Astricon in October and with ...
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ArchitecturePJSIP

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 ...
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DocumentationHow To

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 ...
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ArchitectureTesting

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 ...
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Uncategorized

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 ...
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Asterisk 15Conference Bridge

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 ...
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Asterisk 15PJSIP

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. ...
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Asterisk 15Realtime

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 ...
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Asterisk 16

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 ...
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Asterisk 14Asterisk 15

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 ...
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Asterisk 15WebRTC

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 ...
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DocumentationPJSIP

Asterisk Call Party, Privacy, and Header Presentation

Asterisk allows users to manipulate call party identification information through mechanisms like configuration options and dialplan functions (for instance CALLERID and CONNECTEDLINE to name a ...
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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

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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

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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

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