In terms of replacing your traditional PBX, Asterisk can tie analog phones to a central switch, but scalability is an issue. It can interface with analog handsets through use of FXS (foreign exchange station) line cards; IP-to-analog converters, such as Digium’s IAXy ATA (analog telephony adapter); or competing products from Grandstream Networks and Linksys, among others. That said, Asterisk is built primarily for IP phones based either on its native IAX (Inter-Asterisk eXchange) VoIP protocol or standard SIP (see “Lending ear to open source VoIP”). Asterisk modules that can talk SCCP (Skinny Client Control Protocol) to Cisco phones are generally less reliable, given the protocol’s proprietary nature.
Despite Asterisk’s IP phone bias, outbound trunks do not have to be IP. Not only can Asterisk link with commercial VoIP providers such as BroadVoice and VoicePulse, but with the right hardware in place, it can also handle TDM circuits such as channelized T1s to deliver dial tone from the PSTN. Individual analog PSTN lines can also be brought into play with PCI line cards within the Asterisk server or via outboard FXO (foreign exchange office) ATAs such as the Grandstream GXW-4108, which can handle eight POTS lines, each addressable as a unique SIP trunk within Asterisk.
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Light on Linux requirements
Perhaps the most fundamental misconception about Asterisk is that it requires you to be a Linux shop. Not true. The open source
PBX runs as a service on many platforms, including Windows, as there are projects available to enable Asterisk to run on 32-bit
Windows.
Constructed much like your traditional PBX, Asterisk is based on a Unix-like OS hidden by a CLI or GUI management layer. You can deploy a standard Linux server and install the Asterisk package to create your own PBX or go with one of the several customized Linux distributions based around Asterisk.
Paul Venezia is senior contributing editor of the InfoWorld Test Center.
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