Open Source Options For VoIP

Open source PBX means the breadth of VoIP products for SMBs is bigger than ever

CRN logo By Fahmida Y. Rashid, ChannelWeb

12:00 AM EDT Mon. Mar. 17, 2008
From the March 17, 2008 issue of CRN
Page 1 of 2
For many solution providers, it's time to start asking, "Open source or proprietary PBX?" They already have a veritable banquet of choices for SMB VoIP -- throw in open source systems to the mix and it becomes a feast.

The dizzying array of SMB VoIP products and unified communications available on the market means solution providers can pick and choose which features and capabilities to push to customers. That is even more so the case with open source, since there is flexibility with pricing, hardware add-ons and bundled services. There's a lot of money up for grabs, too; the Dell'Oro Group, Redwood City, Calif., has projected total PBX revenues to exceed $7.5 billion in 2011, driven by strong IP PBX sales. Over the same period, the installation of IP lines into the SMB is expected to grow by 30 percent every year, making up 60 percent of line shipments into the SMB by 2011.

Most open source IP PBX solutions are based on one of two software packages: Asterisk, a complete feature-rich PBX solution, or SIPfoundry, which includes SIP-based projects such as sipX and reSIProcate. Asterisk offers traditional PBX functionality, which means nearly all the features that a buyer would expect to find in an enterprise-class PBX such as directory-based voicemail, conference calling, integrated messaging, interactive voice response (IVR) systems, three-way-calling, caller ID and call queues. While several vendors offer Asterisk-based systems, solution providers comfortable with Linux and software development can download the source code, build the solution on their own hardware and sell their own, branded PBX.

Being Linux-based, Asterisk does not require any elaborate or expensive hardware, freeing up solution providers to source the hardware at a lower cost. If the system needs to work with analog lines or FXO/FXS adapters, special network interface cards can easily be added to the machine. If compiling the source code sounds too intimidating or time-consuming, Asterisk-based vendors, such as Digium Inc., Huntsville, Ala.—the primary developer and sponsor—and Fonality, Los Angeles, are just some companies that offer pre-packaged solutions.

Why Open Source PBX?
Open source PBX has many selling points including customizability, stability, performance, flexibility and deep functionality. The legacy PBX market is expensive, complicated and often not very easy to customize. Customers also like the price tag; bids for open source PBX deployments often are 45 percent to 60 percent less than deployments using proprietary systems. Cheap doesn't have to dent the solution provider's bottom line, either. Open source telecom products like Digium's can offer as much as a 50 percent product margin, which is far higher than the 10 percent range commonly seen in proprietary offerings. Digium offers Switchvox, which is essentially the Asterisk IP PBX on a pre-configured server with some other enhancements. The software version, Switchvox Free Edition, can be installed on any hardware, supports 15 virtual, IP or analog phone extensions and can handle up to eight concurrent phone calls at once. Digium also has Switchvox as a hosted PBX solution. The Switchvox Hosted Edition runs many copies of Switchvox IP PBX software in a virtual environment, allowing several customers to be supported on a single server.

There are many ways to deploy an Asterisk solution and to add-on various services. For example, solution providers can install the FXS analog adapter to support analog phone lines instead of getting it pre-installed by the vendor. An entire IP PBX solution can be rolled out without using a single phone; customers can have a solution based entirely on softphones and conference-calling applications installed directly on their desktop machines. Services revenue can often be double that generated from traditional deployments, especially if the customer pays for support via a subscription.

Switchvox supports mashups, where the phone system can be integrated with other packages to create applications that are accessible from the user's dashboard. These applications can be as simple as displaying a Google Maps page based on the caller ID location, or looking up the caller ID information in a Sugar CRM application. Other applications can show what other users on the system are doing. While integration with Sugar CRM is built-in, creating applications to show images from Flickr or to return search engine results based on the caller's name is also straightforward.

Firewall configuration is another common task. If the firewall is Linux-based, configuring the firewall to prioritize all UDP packages (SIP relies on UDP) and setting some traffic rules can improve voice quality and reliability. If the firewall is not flexible, the VoIP solution can be integrated with a gateway/firewall product that can handle these configurations. Creating rules on an Asterisk server can allow callers to call in from outside the network with a cellphone and then use the service to dial out anywhere.

Another benefit of open source PBX: most VoIP phones should work with the system, so existing phones can be used. If a specific phone type is preferred, Polycom Inc., Pleasanton, Calif., has a multiyear agreement to integrate Asterisk telephony features into its SIP-based desktop and phone products for the SMB. Since any VoIP phone will work, it's actually more cost-efficient to migrate from an existing VoIP implementation to this one, than to roll out a brand new one.

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