Enhanced Messaging in Asterisk 13 and 15
George Joseph
May 30, 2018
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
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
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.
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
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!
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
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
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
TLS certificates and their management are something we take for granted every day when we visit a website. If you sit down and try to
A basic concept with chan_pjsip/res_pjsip is the endpoint. When a new SIP request comes in, res_pjsip needs to identify which endpoint the request is for.