Nearly three years since Jon “maddog” Hall, the executive director of Linux International, predicted that “VoIP using an open source solution, such as Asterisk, will generate more business than the entire Linux marketplace today,” open source VoIP for the enterprise remains a wild frontier.
SMB uptake has been considerable, as open source VoIP’s promise of control and cost savings make it a natural fit. But when it comes to large-scale implementations, open source voice has yet to get most enterprises to listen.
That’s not to say that the enterprise is deaf to the benefits of open source itself. To be sure, companies are increasingly vetting open source alternatives before considering commercial wares. But despite this interest in open source elsewhere in the enterprise, the phone system has, by and large, remained off-limits to open source experimentation.
When it comes to open-sourcing dial tone, the feeling among most enterprises is that there’s just too much at stake. After all, network troubles translate to help desk calls and lost revenue, but if the phones go down, it could mean life or death.
That said, the notion of an all-out VoIP implementation — ripping and replacing to the core — is fast fading away. Yes, dial tone has crossed the network boundary, but not pervasively. Moreover, many traditional PBX vendors are backing into the VoIP market, allowing telephony admins the comfort of tried-and-true PBXes with some of the benefits of VoIP. Introducing VoIP modules that permit VoIP trunks between locations is becoming common, yielding long-distance cost reductions without disrupting the status quo for local voice.
Largely, VoIP is becoming a PBX replacement on an as-needed basis. And such targeted installations could prove a sweet spot for open source VoIP. As the technology gathers steam, convincing enterprises of its efficacy in increments, it will most certainly join the pack of large-scale go-to VoIP candidates down the line. After all, the considerable cost savings and flexibility of open source VoIP are just too great to ignore.
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 customisation 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 (see page 21: Summer Bay resorts proves Asterisk worthy). 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 (NZ$19.7 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.
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 analogue phones to a central switch, but scalability is an issue. It can interface with analogue handsets through use of FXS (foreign exchange station) line cards; IP-to-analogue converters, such as Digium’s IAXy ATA (analogue 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 channelised T1s to deliver dial tone from the PSTN.
Individual analogue 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.
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 operating system 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 customised Linux distributions based around Asterisk.
Lending an ear to open-source VoIP
Whereas commercial VoIP vendors typically supply their own phones, tying them to their IP PBXes for solid integration and providing phone setup, configuration, and maintenance as part of their packages, when it comes to deploying an open source VoIP solution such as Asterisk, it’s strictly a BYOP (bring your own phone) affair.
That said, a massive number of Asterisk-compatible phones are already available, from around US$50 (NZ$70) for a basic handset to US$800 per device. Several 802.11 wireless phones currently on the market include support for multiple VoIP protocols, including SIP, MGCP (Media Gateway Control Protocol), and IAX (Inter-Asterisk eXchange).
Among the major IP phone vendors, Polycom has been working closely with Digium to ensure that its phones are fully Asterisk-compatible, according to Mark Spencer, founder of the Asterisk IP PBX project and chairman and CTO of Digium. Phones from Aastra, Grandstream Networks, Linksys, and Snom all ship with SIP support and can easily be integrated into an Asterisk deployment.
As for those phone manufacturers that sell their own IP PBXes, such as Cisco, Spencer’s comments are salient. “This illustrates the classic problem,” he says. “Asterisk is both competitive and complementary to their products.”
Whereas Cisco would prefer customers buy its complete VoIP package, its phones now support SIP so that they can also be sold a la carte. Despite some versions of the SIP firmware on its 7000-series phones having difficulty with various Asterisk functions, Cisco’s move toward SIP compatibility has gone reasonably well.
The downside, of course, is that most phones won’t have the server-side support found in commercial products. As such, setup and configuration may be more difficult, although not terribly so. Many models do, however, come equipped with plenty of enterprise-ready features, such as large LCDs and built-in 802.1q-compliant switches to reduce cabling and configuration. Polycom also sells a version of its conference-room speakerphone with full SIP compliance.
Summer Bay resorts proves Asterisk worthy
Despite Digium's current positioning of Asterisk for the midmarket, plenty of large-scale implementations speak to the scalability and versatility of the open source IP PBX. One such rollout — that of Summer Bay Resorts, a time-share vacation property company — provides ample evidence that if the phone is the lifeblood of your business, Asterisk is up to the task.
Nearly all of Summer Bay’s business relies on its phones, and nearly all of its phones run on Asterisk. With around 60,000 customers, three main resorts, and several call centres throughout the United States, Summer Bay’s phone and data infrastructure is necessarily large. There are nearly 600 seats in Summer Bay’s call centres, and they log more than a million voice minutes each month and 7.1 million calls each year. It’s safe to say that Summer Bay’s 13-server phone system gets a workout.
Just a few years ago, the company’s sole, 40-seat call centre relied on a legacy PBX. Rapid growth, including the addition of several call-centre sites, pushed this system to its limit, and the decision was made to move everything over to Asterisk. The company has since deployed Polycom SIP phones on every desk and at least one HP DL360 running CentOS and Asterisk at each site, tying the whole system together at the company’s headquarters.
Beyond basic dial tone, Summer Bay has implemented SIP trunks to international resorts for simple call transfers and has wrapped Asterisk around legacy phone systems where necessary. The company is in the process of writing a predictive dialer application for Asterisk. For reporting, it has developed in-house tools and is writing a .Net app to handle all call monitoring and reporting throughout the organisation.
According to David Kurtz, Summer Bay’s director of IT, flexibility was a key factor in the decision to move to Asterisk rather than a commercial solution.
“We have to turn on a dime. Adding 40 inbound toll-free numbers and handling call routing for those numbers can happen at a moment’s notice,” Kurtz says. “We can do that very easily with Asterisk. There’s no way we could have been this flexible, this mobile, and this quick on rollouts with any other platform.”
In addition to the PBX side, Summer Bay also moved all inbound and outbound calls from PRI circuits to SIP channels provided by Global Crossing, a connection made by its voice and data rep at CDW. In this way, Summer Bay’s long-distance and toll-free costs have dropped right along with its data costs because everything is data. But the cost savings didn’t stop there.
“Our implementation costs for the entire voice operation were around US$90,000 (NZ$128,300), including the phones,” Kurtz says. “Quotes from commercial vendors were around US$1,000 per seat. There’s no way we could have done this with any other phone system.”
Surprisingly, Summer Bay isn’t a Linux shop. “We’re predominately a Microsoft infrastructure,” says Jason Brown, Summer Bay’s network manager. “But I can’t see us running a PBX on Windows.”
When asked about reliability, Brown’s comments show his Novell roots. “The Linux servers have been 100% rock-solid,” he says. “The Asterisk process has abended once or twice over the years, but it’s never been a problem.”
“Our ROI has been unreal,” Kurtz adds. “It’s not only saved us hard dollars right away, but our productivity has gone up in terms of what we can do with the phones and how quickly we can deploy solutions.”