Designing a basic Asterisk VOIP system for SIP clients
Asterisk is becoming an increasingly popular way for organizations to deploy Voice over IP (VoIP) without making a huge investment in proprietary systems. One of the major hurdles to get over when deploying Asterisk is to learn how the different configuration files work together and how to configure the system to answer phones.
In this article, I'm going to explain some of the core concepts that are involved in assembling a Session Initiation Protocol (SIP)-based Asterisk Soft PBX environment. By the end of this article, you will have fully functional Asterisk environment capable of SIP client transactions with voicemail features.
For this article, I'll be using Gentoo Linux. Traditionally, Gentoo has a reputation for being difficult to install, but this is no longer true. A Gentoo Live CD install is available from Gentoo.org , which makes installation a snap. I chose Gentoo instead of a more familiar distribution, such as Ubuntu, because a build environment for Asterisk already exists for Gentoo. Asterisk can run on OSX and virtually any other UNIX-based OS. However, if you choose to use another distribution, the install process may be more complex, and you may have to manually compile Asterisk.
If you're using Ubuntu or a distribution that doesn't have a native Asterisk build ready for it, there is quite a bit of leg work you must do first.
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