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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
Busy Asterisk systems can be affected by the SIP timers T1 and B timeout values configured. Consideration of their values impacts how quickly a transaction
A new feature that was initially implemented during a recent visit to SIPit has now been merged into the 13, 14, and master Asterisk branches.
Slight interlude from your regularly scheduled programming. For any interested, Matthew Fredrickson, manager of the Asterisk project, will be giving a webinar today about Asterisk
When the PJSIP work for Asterisk began one of the primary concerns kept in mind was that it be extensible. One of the APIs derived
For those of you who were unable to attend, AstriDevCon this year at Phoenix continued to impress. There were 39 people signed up, and at
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
I’ve been a part of the Asterisk project for many years now, over 10 in fact. I’ve seen it evolve and grow and one of
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
