Source: snapvoip.blogspot.com

The Digium Innovation Award is designed to recognize developers, customers and partners for innovative new and ground breaking thoughts, methods that are implemented to keep the bottom line of a company individual or an institution at an elevated state.

"The 2007 Digium Innovation Award winners are truly redefining the VoIP industry through revolutionary and inspiring open source deployments," said Mark Spencer, CTO of Digium and creator of Asterisk. "We continue to be impressed with the ways in which developers, partners and customers are using Asterisk to tackle business and social problems that would otherwise be too cost prohibitive or time consuming to address."

Digium Innovation Award will be presented during a ceremony at Digium|Asterisk World in Boston, MA. October 30 and 31

The results are out and I am happy to have recognized some of them. The Winners are;

Aheeva (Montreal, Canada),leveraged input from the global Asterisk community and, in the true spirit of open source, gave a portion of the resulting work back to the community. For more information about Aheeva, www.aheeva.com. Selected for its Aheeva Contact Center Suite (CCS).

OneBizTone, www.onebiztone.com. For aiding UPenn (I wrote about the UPenn implementing Asterisk at the campus.) and selected for working with the University of Pennsylvania to migrate more than 10,000 voicemail users off a discontinued and unsupported legacy voicemail system.

Shelton|Johns (Marietta, Georgia), selected for using Asterisk to design and deploy a turn-key, campus-wide VoIP network for Life University. www.sheltonjohns.com.

iPLATEu, selected for inventing the first-ever social platform for motorists using anonymous, mobile-to-mobile voice messaging through license plate numbers. www.iplateu.com.

Super Technologies (Pensacola, FL), selected for providing Asterisk open source VoIP services to wholesale telephony service providers in 98 countries.www.didx.net.

Source: snapvoip.blogspot.com

The Philadelphia-based Ivy League university, University of Pennsylvania, currently has over 1,250 Session Initiation Protocol (SIP) IP phones on desktops, tied to a back end based on SIP Express Router — an open source VoIP call-control and routing stack, and Asterisk for voice mail messaging. Don’t go away! This is just the start. The University has extending the VoIP network in to a 15000 seats, in it’s plans!

Deke Kassabian, the university’s senior technology director for information systems and computing, plans to grow that installed base by a factor of more than 10 over the next five years. Driving the project is the desire to get off costly Centrex monthly fees and infrastructure, and the promise of an open source, standards-based VOIP infrastructure that provides superior integration and control.

"If we can run one modern IP network for voice, video and data …. there’s a clear win," Kassabian says. "If we provide business telephony internally, less money leaves the university."

The Linux-based SER call control and Asterisk messaging servers were a better fit with UPenn’s standard back ends for authentication (Kerberos and RADIUS), its OpenLDAP directory structure, and e mail. While commercial IP PBXs are adaptable to these platforms, "they don’t work that way out of the box" typically, he adds.

With open source running extensively throughout the university — from directories, to e-mail, DHCP and DNS — the level of expertise in open source troubleshooting and development was there to support the Asterisk plans, Kassabian says.

"For years UPenn has had a strong open source talent pool. As a result, we have the staff and expertise to develop and roll out open source VOIP."

UPenn’s work with the Asterisk community is also paying off by improving the product itself. University programmers have already contributed to two additions to the code base, which is now supported in the main release. One change integrates IMAP-based voice mail and messaging stores, and another involves improvements in SMDI signaling between IP phones and voice-mail system back end.

"Instead of going off and making changes ourselves," Kassabian says, "we get our changes built into [the code base] and don’t have to maintain them ourselves. They’re part of the next distribution."
The infrastructure Kassabian and his team built is designed for high-availability VOIP, with redundant connections to IP call and feature servers, PSTN and IP telephony service provider (ITSP) point-of-presence links. Two data centers on campus host redundant clusters of Asterisk boxes, SIP proxy servers, and media/messaging feature servers. Phones on the network can register to and access any set of servers. "In this way, there’s no single failure, and no single site failure that would take out the servers," Kassabian says.
For outbound calling, UPenn is using a mix of VoIP and PSTN services. For long distance service and other calls, UPenn is plugging its campus VoIP network directly into a SIP trunking service from Level 3. A pair of dedicated Cisco 3600 routers also support PSTN links for local calls, and as a back for ITSP service.

Kassabian and his staff, mostly with IP networking backgrounds, also had to get up to speed with voice system jargon and terminology before being able to understand user needs. "We had to learn about how people use their phones," Kassabian says. "I had to learn what a bridge line appearance actually was."

Like many large organizations converging voice and data networks, consolidating the school’s telecom and data network teams helped tremendously.

"Having people from our traditional telecom organization learning IP technology has been great," he says. "And our networking staff has been learning telephony. That’s all been part of it. As time goes on, more of us are more well cross-trained and the two technologies come together very well."

More information at UPenn voice