Asterisk

Asterisk Linux Based IP-PBX

VoIP Fulfillment Launched by VoIP Supply

January 31, 2011

VoIP Supply is planning to announce the launch of "VoIP Fulfillment" at ITEXPO next week, but they gave me a sneak peak at this new offering which technically was a two-year quiet beta they've been working on. Their VoIP Fulfillment offering covers quite a few things, however one of the biggest is they have opened their infrastructure to VoIP service providers, including an API that lets service providers follow the package through the fulfillment process all the way to the customer's door.

VoIP Fulfillment by VoIP Supply is a suite of services specifically designed for VoIP service providers that increase operational efficiency, decreases costs and bolster customer experience. I spoke with VoIP Supply's Garrett Smith - Chief Marketing Officer, Mike Russo - Executive Vice President, and Donald Stefanie - VoIP Service Provider Program Manager to learn more.

Posted in Asterisk, ITEXPO, SIP, TMCnet, Unified Communications, VoIP

Sangoma Announces General Availability for NetBorder Express Gateway 4.0

January 27, 2011

Sangoma Technologies Corporation, a provider of hardware and software components that enable or enhance IP Communications Systems for both voice and data, today announced the general availability of NetBorder Express Gateway 4.0, a major upgrade of its TDM-to-SIP Gateway software. NetBorder Express 4.0 includes significant new functionality which extends the market reach of Sangoma channel partners in Europe, Latin America, Japan, Asia, and North America. Sangoma will be demonstrating the latest release of its NetBorder Express Gateway at ITEXPO East 2011, in Miami, Florida, February 2-4, 2011 in booth #401.

Enhancements that provide Sangoma partners with a greater coverage area for worldwide deployments include support for:

Basic Rate Interface (BRI), an ISDN configuration widely used by SME’s in European countries, particularly in Germany (when paired with the Sangoma A500 or B700 boards)MFC/R2 Signaling, used extensively by enterprises in Latin America, especially Mexico, Brazil and Columbia (when paired with any of Sangoma’s lifetime warranty A-Series of digital boards, A101/A102/A104/A108) INS 1500, a Primary Rate Interface (PRI) ISDN variant used in Japan (also supported by Sangoma’s A-Series of digital boards)T.38 Fax Over IP Support, enables reliable transport of fax traffic over a VoIP network

“NetBorder Express is an easily integrated TDM to SIP gateway component that can run concurrently with other software applications, inside a single server,” said Frederic Dickey, director of product management at Sangoma. “Alternate solutions require external gateways in a separate appliance.

Posted in Asterisk, ITEXPO, SIP, TMCnet, VoIP

Digium's Danny Windham to Keynote ITEXPO East 2011

January 21, 2011

Posted in Asterisk, ITEXPO, SIP, TMCnet, VoIP

ITEXPO Named Top 50 Fastest Growing Tradeshow

January 19, 2011

TMC's ITEXPO was named one of the fastest growing trade shows by Trade Show Executive in their Fastest 50 Awards. I've been with TMC since 1994 - right out of college and I've seen this company grow from 15 employees to over 65 employees. It's been an amazing ride. TMC's ITEXPO has had some stellar companies participate over the years, including Asterisk, Avaya, Cisco, Microsoft, Packet8, ShoreTel, and Skype.

TMC CEO Rich Tehrani said, "Much thanks goes to all of TMC's readers as well as our sponsors, attendees, partners, vendors who supply us decorating services, audio/visual, etc and of course the tireless TMC team who works around the clock to over-deliver for customers.

Posted in Asterisk, Avaya, Cisco, ITEXPO, Microsoft, Packet8, SIP, ShoreTel, Skype, TMCnet, Triple Play, Unified Communications, Video Conferencing, VoIP, Wireless

TELTUB + Google Voice = $5 Unlimited Calling to U.S. & Canada

January 17, 2011

TELTUB is offering unlimited free calling to the U.S. and Canada for $5/month. Their gHome product takes a user's Google Voice account and converts it into a SIP-based VoIP service which can then be used as a cheap home phone-line using a SIP analog telephony adaptor (ATA) and your broadband connection. Users can use their own standard VoIP adaptor or order a pre-configured one from TELTUB.

Posted in Android, Asterisk, Reviews, SIP, Social Networking, TMCnet, Unified Communications, VoIP

FREETALK Connect Review

December 15, 2010

FREETALK Connect might be an easy to install IP-PBX, but it doesn't skimp on features, packing quite a punch in advanced features, including Skype for Asterisk & Skype Connect support, paging, conference bridges, PSTN failover, custom CallerIDs per user, find me/follow me, and more. I took a FREETALK Connect system for a test drive and was pretty impressed with how easy it was to setup and configure, but was also impressed with its tight integration with Skype.

First, I should mention that FREETALK Connect runs with Asterisk in its core. Currently running Asterisk v1.4, the box relies on a Web portal for all administrative (add users, configure auto-attendant, etc.) and user interaction (voicemail, call forward settings, etc.). You don't get console access to the Linux kernel.

Posted in Asterisk, SIP, Skype, TMCnet, VoIP

Digium|Asterisk World Returns to ITEXPO

December 10, 2010

TMC and Digium, the Asterisk Company, today announced that Digium|Asterisk World is returning to ITEXPO, February 2-4, at the Miami Beach Convention Center. Asterisk is the world’s most widely used open source communications software, and the Digium|Asterisk World event helps the entire Asterisk community learn how to create more flexible and powerful telephony solutions.

Open Source Communications solutions currently account for more new phone lines than any single conventional PBX or key system manufacturer. While Open Source communications software can be free of charge, it creates significant revenue opportunities for businesses that install Open Source PBXs, and provide GUIs, PBX appliances, gateways, and other related services.

Posted in Asterisk, ITEXPO, TMCnet, Unified Communications, VoIP

Kevin Fleming Details Asterisk SCF

October 27, 2010

Posted in Asterisk, SIP, TMCnet, Unified Communications, VoIP

Asterisk SCF - Universal Communications API?

October 27, 2010

Digium today announced a new open source product called Asterisk SCF which is designed to simplify the creation of complex communications systems. According to Digium, Asterisk SCF is a framework that allows developers to create real-time communications applications that include voice, video and text and that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions. Asterisk SCF is architected to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance.

Asterisk SCF will be delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, transparently.

Posted in Asterisk, TMCnet, Unified Communications, VoIP

Asterisk 1.8 - Major Release Sports 200 New Features

October 21, 2010

Digium today released Asterisk 1.8, a major new release of the popular open source telephony platform. According to Digium, it includes more than 200 notable new features.

I spoke with Digium's Product Manager, Steve Sokol, to learn more about this important release.

Posted in Asterisk, Google, SIP, Skype, TMCnet, VoIP

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