Share this story • • • With the advent of voice-over-IP (VoIP) technology, there has been a dramatic movement toward IP-only telecommunications, leaving the twisted pair of yesteryear in the dust. Lower costs, automated directories, centralized monitoring, and ease of call routing are just a few of the advantages that a good VoIP implementation can bring to a workplace. But many businesses have been held back from jumping on the VoIP bandwagon because it can seem daunting or expensive to set up. ![]() The reality of VoIP is that some very modest hardware and a suite of free software tools can make for an enterprise-class VoIP system that can serve up to 1,000 office users in a single building. In this article, we'll walk through the basics of doing VoIP with Asterisk, an open-source, software private branch exchange (PBX). Note that this isn't a detailed how-to—it's more of a overview of the basics of building a VoIP system, with some notes on best practices. After reading this article, you should have a sense of what's involved in a moderately sized VoIP setup, and of what such a setup can do for your business. Note: This article revisits the same topic as our similarly titled by Kurt Hutchinson. But given the progress since then, we thought it was time for an update/expansion. ABCs of VoIP Most people don't know what goes on behind the scenes when they place a call on their telephone at home or at work. Historically, the phone line consisted of a single circuit from your desk to a closet on your floor or section of your building (whether your office building, apartment building, or even your house). Setting up a small office or home office VOIP system with Asterisk PBX – Part 2. A computer to act as the PBX Server. Apart from leading Cypress North, Matt. Asterisk is a PBX software which can interoperate with multiple low-cost VoIP providers, thus giving the users many more choices. There are several pre-built software that bundles Asterisk, e.g. PBX in a Flash, FreePBX Distro and AsteriskNOW. Calls were then punched over to what are called 'house pairs,' or the internal wiring of the building that's permanent and isn't moved or modified. In most multi-story office buildings, these house pairs were arranged in blocks, and they went down into a master closet where the phone service provider had brought in either circuits or trunks for each of the individual phones. In recent years, calls have been multiplexed into some sort of digital solution such as ISDN or PRIs on a T1, DS3, or a similar digital connection. When you dial a call, you're dialing on a switch in the nearest calling office, which then routes the calls through the various telecommunications switches between you and your calling destination. Sometimes the call is routed over the Internet,but on private circuits. Your call is 'attended' during the entire duration, which means that, while you're on the phone, the switch through which it's routed has a busy circuit and one less resource that can be used, because it's maintaining a single complete circuit from handset to handset. Credit: Kurt Hutchinson This approach makes perfect sense if you're in an office in Boston and you're calling an office in Miami, but what if you're calling downstairs? When it comes to inter-office calls, you're inefficiently using up two switch ports in your private branch exchange (PBX) for that phone call, when those ports could instead be used to switch inbound or outbound calls. VoIP, in contrast, uses an all-digital network between both call endpoints (using a VoIP phone or analog telephone adapter). This makes VoIP very dynamic, because it can be set up and moved without causing any disturbance to the current building infrastructure. VoIP uses the same logical structure as an analog PBX, but does it in a much more efficient and far more cost-effective manner, by implementing that structure with standard data networking equipment. In a VoIP scenario, if you're in Boston and you call Miami, you pick up the phone and dial the number. Mp3 songs download naa songs. Your VoIP phone contacts the call manager or proxy that it is registered with and requests that number. The call manager comes back and essentially says, 'Yes, I have a path. Hold on, let me set it up for you.' The call manager then handles the protocol and codec negotiation for the call and makes the call to the remote gateway, whether to the telco's gateway on its analog network or to the destination's VoIP gateway. Most call managers then attend the call, meaning your audio traffic will pass through and even be processed by the call manager for the entire duration of the call, especially if both sides can't use the same audio codec for some reason, which means that the call will need to be transcoded in real-time. (More on that later.).
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