Asterisk is an open-source framework for building communication-based applications.
\nHistorically, Asterisk is an alternative to most proprietary Private Branch eXchange (PBX) systems, dealing with voice communications, conference calling IVR or voicemails.<\/p>\n
Quite modular, Asterisk is shipped with several audio codecs (g711a, g711u, g722, gsm), handles standard protocols (SIP, IAX), and could be used virtually anywhere from multi-tenants providers, to end-user setups.<\/p>\n
There’s a lot of Asterisk-based distributions, starting with FreePBX, and derivatives such as AsteriskNow, Elastix, or alternatives such as PBXinaflash.
\nThe purpose of these systems is to provide end-users with a clear web interface managing their setup.
\nThis is usually a good way to manage your setup. Although, when dealing with several servers, all with their local dialplans, configuring trunks, routes, user extensions, … and guaranteeing all your users are offered with the very same service, you will spend quite a lot of time doing repetitive checks, and sporadically fixing typos and unexpected configurations.<\/p>\n
Before leaving Smile, I worked on a puppet class that could deploy asterisk and configure everything from hiera arrays. No frontend, except for some nginx distributing phone configurations. Minimalistic setup, based on Elastix\/ASTDB-based generated contexts and embedded applications.
\nI didn’t have the time, nor the guts to finish it. Today, I have a working PoC, involving my Freephonie SIP account, a couples softwares and hardware phones, voicemails, DND, CFW, …
\nAnd last but not least: hardware phones default configuration locks them to a private context.\u00a0Users may dial their extension number\u00a0and\u00a0authenticate themselves using a PIN number to get their phone re-configured with their extension.<\/p>\n