Jitter buffer functionality has been in Asterisk for quite some time now. However, knowing what jitter is in a voice over IP (VoIP) application and
The Digium Phone Module for Asterisk and the g729a, silk, siren7 and siren14 codec binary modules hosted at downloads.digium.com can now be automatically downloaded and installed
Overview When the development goals were laid out at AstriDevCon for Asterisk 12 way back in 2012, we had two primary missions: Build a new
Asterisk’s REST Interface (ARI) in both Asterisk 12 and 13 has the ability to originate (create) outgoing channels. The functionality in ARI mirrors that of
