Chapter 17. PBX Setup

Table of Contents

The all-in-one myth
Choosing between Asterisk and FreeSWITCH
Official packages
Contributing patches
Scalabiltiy and code quality
Using Asterisk with the repro SIP proxy

A soft PBX (such as Asterisk or FreeSWITCH) provides features such as voicemail, menu systems and call queues. Many of the typical features in a soft PBX have a particular focus on voice communications, rather than other types of RTC such as IM or video.

It is perfectly feasible to build an RTC environment without these features. It is recommended that a SIP proxy is used to handle all connectivity with users and any external parties, the reasons for this are explained in the section called “All SIP connectivity through a SIP proxy”. This also means that it is better to install and test the SIP proxy before making decisions about the selection and design of the soft PBX.

The soft PBX can be configured to make a SIP registration in the SIP proxy or routing entries can be created in both the SIP proxy and soft PBX to send calls back and forth between them as required.

The configuration and operation of soft PBXes tends to replicate many concepts from the world of traditional telephony. The fact that many of the features of these products are voice-orientated is an example of this trend. Many guides on soft PBX configuration tend to focus on the use of dial plans and phone numbers as the dominant currency in the world of legacy communications, while modern unified communications strategies involve user@domain addressing similar to other Internet-based services.