Open Source VOIP Connects to Business

Open Source VOIP Connects to Business

Open Source VoIP is slowly making gains in enterprise adoption.

Asterisk: open source's top choice

Digium's Asterisk is far and away the most mature and popular open source IP PBX currently available. Other open source projects are under development -- many, such as OpenPBX, forking the Asterisk code base; others, such as FreeSwitch, being built from the ground up. But despite increasing competition among open source IP PBXes, Asterisk remains the most compelling enterprise VOIP play.

So much so that Sam Houston State University last year migrated 6,000-plus extensions from Cisco CallManager to Asterisk, eliminating phone licensing costs and increasing customization control and security in the process. And Summer Bay Resorts, a time-share vacation property company, logs more than a million voice minutes per month on its 13-server Asterisk system. But despite such proof that large-scale implementations of Asterisk are viable, Digium remains focused predominantly on the midmarket.

"Anything larger is a great opportunity for us, but that's not our core customer base," says Mark Spencer, founder of the Asterisk IP PBX project and chairman and CTO of Digium, which received US$13.8 million in venture capital last year and recently appointed former Adtran COO Danny J. Windham as its CEO. "Asterisk can scale to those levels, but we're looking more toward the middle of the market."

Digium's tempered stance toward widespread enterprise Asterisk adoption is understandable, given the reservations many enterprises have about open source VOIP.

Chief among purported detractors are a perceived lack of support, questions about the availability of features, and concerns about required skills for implementation and management, as well as reservations about platform compatibilities.

A closer look at Asterisk and its rapidly evolving base of developers suggests that these anxieties are unfounded and that Asterisk is ready for targeted enterprise deployment.

Makeup of an enterprise contender

Created by Spencer in 1999, Asterisk is a complete IP PBX released as open source under the GNU General Public License. It is built to run on commodity hardware, providing considerable cost savings when compared with commercial IP PBXes, and it leverages the open source community for additional testing, bug fixes, and feature development. Asterisk is available both as a business edition purchasable just like any other IP PBX -- with seat licenses, warranties, support contracts, and shiny-binder reference materials -- and as a free download, allowing you to take a test run before signing any checks.

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. 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.

Due to gaps in communication between the PSTN and SIP, however, most Asterisk implementations rely on PCI line cards rather than outboard adapters. For example, it isn't possible to send a SIP equivalent of a hook flash from Asterisk to an ATA, meaning that phone features that require hook flashes to the PSTN -- such as call waiting -- won't work. For most businesses, this isn't a problem. It's more indicative of the occasional compatibility issues that exist between old and new technologies. With PCI interfaces in place, however, these problems dissipate.

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