Digium Receives $13.2MM in VC Investment

August 9, 2006 by Garrett Smith

By BRUCE MEYERSON AP Business Writer

NEW YORK Aug 9, 2006 (AP)— Matrix Partners is investing $13.2 million in Digium Inc., the creator and coordinator of Asterisk, a popular open-source PBX software platform for business phone systems that’s based on Linux and Internet Protocol.

The funding announced Wednesday marks the first round of venture capital for Digium, which oversees development of the free Asterisk platform, but is also one of a growing number of companies selling a customized phone system derived from the software.

The investment comes amid modest, but growing interest in a new generation of technologies that seek to replace the traditional office switchboard and phone system often referred to as a PBX, or private branch exchange. Earlier this year, for example, Azure Capital Partners invested $5 million in Fonality, which makes an Asterisk-based phone system.

At last count, there were 130 vendors of Asterisk-based business voip systems, and more than a thousand software developers are downloading the source code each day, according to Digium. However, none of that activity generates any revenue. Digium won’t disclose any specific data on its financial performance, but says it has been profitable since 2002, generating 100 percent growth in revenues each year since.

Where the conventional products sold by major vendors can cost tens or hundreds of thousands of dollars for a business to buy, the new systems based on Asterisk and other Internet technologies can be purchased by a small company for as little as $1,000.

The IP-based systems also inject new programming flexibility and features for users to customize to the needs of their companies. Some of these products are delivered remotely over the Internet from a vendor’s servers rather than being installed on a computer at a company’s facility.

Matrix manages $2.5 billion in assets from offices in Boston, Silicon Valley and India.

“We have maintained steady growth and have been consistently profitable. We felt seeking outside funding was unnecessary, but Matrix Partners’ success and vision in the open source industry was too compelling to ignore,” said Mark Spencer, president of Digium and creator of Asterisk.

More from: Asterisk Garrett Smith

Popular Hardware for Asterisk Open Source PBX

August 4, 2006 by Garrett Smith

Thinking of implementing Asterisk, Asterisk Business Edition or TrixBox (Formerly Asterisk @ Home)? Asterisk is growing in popularity as a viable alternative to often expensive, proprietary IP PBX solutions from tier 1 vendors. Asterisk is open source software which is maintained by Digium as well as a host of volunteer coders around the globe.

If you are new to open source telephony, below are a few good places to begin learning about these technologies:

Asterisk Open Source PBX

Asterisk WIKI at VoIP-Info.org

TrixBox – Formerly known as Asterisk @ Home

YATE – Yet Another Telephony Engine

With the advent of various GUIs (Graphical User Interfaces) for Asterisk, the barriers to entry for businesses and other users lacking Uber-Geeks in the backoffice are slowly being removed. Asterisk supports both SIP and IAX protocols for VoIP calling, with SIP being the most commonly used.

In addition to the Asterisk software and Linux variant operating system, you’re going to need some basic hardware elements to get started. Asterisk is commonly hosted on a server, which can be a desktop type PC, or a more industrial, rackmount type server.

Both Intel and AMD based machines will work. Configured properly, Asterisk is not an extremely “resource intensive” application, and can run on a fairly lean hardware configuration. A good base system would require:

  • Intel or AMD 2.0 GHz CPU
  • 512MB RAM
  • 20GB or Larger SATA or EIDE Hard Drive
  • CD-Rom, Video Card, Sound Card, 10/100 Ethernet Card
  • At least one free PCI or PCI-X Expansion Slot

Asterisk is commonly deployed in a “hybrid” scenario, with calls being placed via VoIP, as well as over the PSTN. Digium, Sangoma and Rhino produce PCI interface cards allowing for PBX connectivity to the legacy PSTN, as well as digital T1/PRI, E1 and ISDN/BRI. Analog cards are available with FXS Ports, FXO ports, or a combination of both. Additional upgrade modules providing echo cancellation are also available. Some Digium PCI cards require a specific voltage PCI slot, either 3.3V or 5V. Be sure to adequately research your motherboard prior to purchasing a Digium PCI board.

Asterisk supports the use of both IP enabled telephones, as well as traditional analog telephones used in conjunction with an ATA (Analog Telephone Adapter), Gateway or Channel Bank. IP Enabled telephones can be connected directly to the network, whereas analog telephones are going to require a bit of extra hardware to connect to your Asterisk server.

Popular SIP phones for use with Asterisk include:

In future posts, we’ll explore setting up Asterisk, and performing basic configuration on the various hardware elements involved.

More from: Asterisk Garrett Smith

VoIP Week in Review for August 4th

This is the first installment of the VoIP Week in Review, a segment that will become a weekly feature on the VoIP Supply blog. Here are some of the biggest stories in the industry this week:

Check back next Friday for the next installment of the VoIP Week in Review.

Grandstream BudgeTone 200 – Hot New Product!

July 31, 2006 by Garrett Smith

Our good friends over at Grandstream Networks have done it again with another hot new IP phone for under $100 that is about to hit the market! The BudgeTone 200 is a SIP based IP Phone featuring Dual 10/100 BaseT Ports for WAN/LAN switching or routing and a Full Duplex Speakerphone.

This afforable IP phone sits between Grandstream’s extremely popular GXP-2000 and their low cost BudgeTone 101 and 102 phones. The GS-200 has a list price of $89.95, and is set to arrive on the dock August 4th.

Calling All Service Providers – The Motorola VT1005 is Coming Soon

July 28, 2006 by Garrett Smith

Yes, Yes, we all know that the Motorola VT1005 has been used by Vonage for quite sometime. But until recently, the only way to get a hold of them was to A) be Vonage or B) be as large as Vonage. VoIP Supply is proud to announce a partnership with Motorola that allows VoIP Supply to sell the un-locked, non-vonage version of their VT1005 SIP Telephone Adaptor.

The adaptor features a compact, low profile design with easy web based configuration. Add two 10/100 Base-T Ethernet ports for Wan/Lan connectivity and two RJ-11 telephony ports and you have the ideal solutions for your residential VoIP service customers. The VT1005 also allows for Voice over data prioritization, PPPoE support, and VPN-pass through. with a suggested retail price of $79.99, and an estimated street price of $65.00 in single quantities, this telephone adaptor will be a great alternative to the Linksys 2100 and Grandstream 486.

Linksys Releases Two New 8 Port PoE Switches for the SMB Marketplace

July 26, 2006 by Garrett Smith

Linksys, a division of Cisco Systems, has recently released 2 low density power over ethernet switches aimed at SMB’s. The Linksys SRW208P and SRW208MP, which retail at VoIPSupply for $274.99 and $324.99/ea respectively.

Both are managed, 8 port switches with a variety of Security and QoS features and both confirm to the IEEE 802.3af Power Over Ethernet Standard. Throw in dual Gig-E Uplinks for increased bandwidth and redundancy, port authentication and MAC filtering, VLAN tagging and a whole lot more, and you’ve got value that’s difficult to beat.

The major difference between the two models is that the SRW208MP, or “Maximum PoE” model, provides a full 15.4W of power per port, whereas the SRW208P version supports 15.4W of power up to 4 ports, and then drops down to 7.5W of power per port on up to 8 ports.

Both models are affordable and feature-rich options for small businesses deploying VoIP or premise based IP PBX solutions.

New Digium T1 Cards with Octasic Echo Cancellation Now Shipping!

June 29, 2006 by Garrett Smith

Digium has recently released (4) new products featuring onboard DSP echo cancellation from Octasic.

The TE412P Quad T1 for 3.3V PCI, the TE407P Quad T1 for 5.0V PCI, the TE212P Dual T1 for 3.3V PCI and the TE207 Dual T1 for 5.0V PCI.

These new models offer an on-board DSP-based echo cancelation module. They support E1, T1, and J1 environments and are selectable on a per-card or per-port basis.

Echo cancellation is provided by Digium’s new VPM450M Octasic DSP-based echo cancelation module. The VPM450M provides a certified carrier-grade algorithm that has been labeled a benchmark for echo cancelation. This new module improves upon our older VPM400M and TE411P/TE406P products. With the VPM400M, 16ms or 128taps of echo cancelation were possible across 128 channels. The new VPM450M enables users to eliminate echo tails up to 128ms or 1024 taps across all 128 channels in E1 mode or 96 channels in T1/J1 modes. Further, this module takes advantage of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls.

Digium has designed these products to be fully compatible with existing software applications and they are fully integrated with the Asterisk Open Source PBX/IVR platform. Also, the open source driver supports an API interface for custom application development. With the combination of Digium hardware and Asterisk software, numerous combinations of telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications.

These products support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features for business VoIP.

More from: Asterisk Garrett Smith

Polycom IP430’s IN Stock at VoIPSupply.com

Polycom IP430 IN Stock at VoIPSupply.com

The Polycom IP430, Polycom’s latest addition to their business class handset line-up is now in stock at VoIPSupply.com. The IP430 is a two line desktop IP phone that featues Polycom’s Acoustic Clarity Technology, delivering outstanding voice quality and smooth two way conversations. In addition to the 2-line appearances, the IP430 features Polycom’s award winning full duplex speakerphone.

Additional Features Include:

  • 132 x 46 pixel graphical LCD with line LED indicators
  • Intuitive user interface
  • Robust feature set and security
  • Dual switched 10/100 Mbps Ethernet ports
  • Built-in auto-sensing IEEE 802.3af PoE support
  • SIP

An enterprise-grade IP phone, The SoundPoint® IP 430 is designed to meet the telephony needs of general business users – cubicle workers that conduct a low-to-medium volume of calls – by delivering a robust feature set encompassing traditional telephony features such as call, park, pick-up, hold and transfer, as well as more advanced capabilities such as shared call / bridged line appearance, multiple call appearances, and presence.


Polycom C100S Communicator Now in Stock at VoIPSupply.com

June 27, 2006 by Garrett Smith

The highly anticipated Polycom C100S Communicator is finally in stock at VoIPSupply.com.

The Polycom Communicator gives you the ultimate hands-free Skype experience. Based on the same technology used in Polycom’s legendary line of triangular SoundStation conference phones, the Skype-certified Polycom Communicator enables crystal-clear, natural conversations when using Skype. Enjoy the freedom of not wearing your headset for hands-free Skype calls, or plug into the built-in stereo headphone port for private conversations.

The Polycom Communicator is on sale for $129.99 and can be purchased by Clicking Here!

Download Polycom Communicator C100S Data Sheet
Download Polycom Communicator C100S Frequently Asked Questions
Features and Benefits
Download Polycom Communicator C100S User Guide

VoIP Supply, llc Increases Headcount 49% in May

Headcount grows from 35 to 52; Company Unveils Interactive Training Center.

Cheektowaga, NY June 20, 2006 — VoIP Supply, llc, a leading provider of Voice over IP (VoIP) hardware, software and services, today announced a 49% increase in headcount during the month of May. The expansion focused on bolstering the customer service, engineering, and sales staff to support the tremendous growth the company has experienced this year.

“The company’s tremendous headcount growth in May is a direct result of the continued success of our business and the emerging Voice over IP (VoIP) industry,” stated Benjamin P. Sayers, President and CEO of VoIP Supply.com. “The launch of our interactive training center reinforces our commitment to our customers by ensuring our staff is the most knowledgeable in the industry.”

The interactive training center features 12 individual learning centers featuring the latest in PC, VoIP, and presentation technology. The training center will be used to training new hires, conduct vendor lead seminars, and to foster continued staff development. The training center is located at the company’s 254 Sonwil Drive location in Buffalo, NY.

For more information about VoIP Supply, llc visit www.voipsupply.com.

About VoIP Supply, llc
VoIPSupply.com, llc (www.voipsupply.com), is a leading supplier of Voice over IP Hardware, Software, and Services. In addition to a comprehensive selection of IP phones, networking equipment and software platforms, the company offers provisioning and fulfillment, custom configuration, technical support, extended warranties and logistical services for end-to-end customer solutions. VoIP Supply is a certified partner with these industry leading manufacturers, including vegastream, zoom, polycom, digium, cisco, linksys, grandstream, sangoma, uniden and snom. For additional information on VoIP Supply, llc please contact Garrett Smith at 716.250.3408.