VoIP Monitor - Asterisk

Digium Announces Ninth Annual AstriCon to be Held October 23-25, 2012

VoIP Monitor — Tue, 17 Apr 2012 21:30:58 GMT

Digium has kicked off planning for AstriCon, the Asterisk Open Source Conference and Exhibition. The event, now in its ninth year, will be held in Atlanta, Georgia from October 23-25, 2012 at the Sheraton Atlanta Hotel. Digium is the creator and corporate sponsor of the Asterisk project, the most widely used open source platform for creating custom communication solutions. Speaker topic submissions are open, and the conference organizers are soliciting talk concepts for 2012. Digium invites those who would like to speak at AstriCon to submit information for consideration by May 1, 2012 at http://www.astricon.net/2012/speaking.aspx.

With nearly two million downloads per year, millions of deployments and a community of more than 65,000 members, the acceptance and growth of Asterisk has spawned an ecosystem spanning more than 170 countries. AstriCon gives all members of the Asterisk community – from telephony enthusiasts to businesses – a forum to learn about the technology. Asterisk integrators and business end-users can expect to hear the latest Asterisk news and project updates, gain access to in-depth technical sessions, participate in networking opportunities, meet potential collaborators and review and discuss detailed case studies of Asterisk projects. The exhibition space will consist of more than 40 exhibitors including Aastra, AudioCodes, Grandstream, Jenne, OpenVox, Orecx, Sangoma, Vitality and XORCOM.

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. Using it, developers and other technical pros craft solutions such as IP PBXs, VoIP gateways, interactive voice response systems, conference bridges, voicemail servers and more. Asterisk also forms the basis for Digium’s award-winning Switchvox Unified Communications solution, which offers the most advanced business phone system features in a cost-effective, easy-to-use solution that scales as companies grow.

“As Asterisk continues its phenomenal growth, Digium and the Community continue working to improve and enhance capabilities, as it did most recently with the release of Asterisk 10,” said Bryan M. Johns, community director at Digium. “Digium also continues to devote substantial resources to the development of both the Asterisk project and hardware that more tightly integrates with Asterisk, such as the new VoIP gateways, R-series failover appliance and Digium Phones.”

Digium is once again pleased to partner with Technology Marketing Corporation to promote the event to a broader audience. TMC has helped support other Digium events, including Asterisk World, with training sessions, video production, attendee registration and exhibit management. Companies interested in sponsoring AstriCon and participating on the EXPO floor should contact Joe Fabiano at TMC: +1 (203) 852-6800, ext. 132.

Registration for AstriCon 2012 is open now on the official event site: http://www.astricon.net. The early bird rate of $495 is available until August 1, 2012.

Digium Simplifies Communications With Advanced Asterisk-based VoIP Gateways

VoIP Monitor — Mon, 26 Mar 2012 21:13:40 GMT

Digium introduces the G100 and G200, the first in a family of cost-effective VoIP gateways that simplify the process of deploying converged media networks. Built on a powerful combination of the Asterisk open source communications engine and a state-of-the-art embedded platform, the new gateways provide the best value for Asterisk communications solutions.

Digium’s gateways are built to support both TDM-to-SIP and SIP-to-TDM applications. In a TDM-to-SIP deployment, the gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.

The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. The gateways feature a power-saving embedded design with a highly efficient digital signal processor handling all media-related operations. The combination of an intuitive user interface, the flexibility of Asterisk and the purpose-built media processing capabilities of the DSP results in a gateway platform that outperforms the dated designs in the market today.

Digium beta testers agree. “Setting up the G200 was extremely easy compared to doing it with other gateways. I'm spoiled now!” said Tim Banks of Project Resource Solutions, an Illinois-based Digium Select partner. “Digium has really set the bar high. Their new gateways make it incredibly easy to connect older TDM phone systems with SIP services.”

Digium’s new gateways represent a solution to one of the challenges associated with running Asterisk applications in virtualized environments. TDM interface cards require a card slot – something distinctly missing from virtual servers. By converting the media and signaling from TDM to SIP on a dedicated external device, Asterisk users can migrate applications to virtualized, hosted or cloud environments.

The G100 includes a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. The G200 doubles the capacity with two T1/E1/PRI interfaces and up to 60 concurrent calls. Both models have integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.

The single-span G100 lists for $1,195 USD while the dual-span G200 model lists for $1,995 USD. The gateways are currently available worldwide through Digium’s network of distribution and integration partners.

Digium Introduces World’s First Phones Designed for Asterisk

VoIP Monitor — Wed, 01 Feb 2012 23:01:33 GMT

Digium introduces a new family of high-definition IP phones. They are the first that are engineered to fully leverage the power of Asterisk, the world’s most widely adopted open source communications software, and Switchvox, Digium’s award-winning unified communications system. With Digium technology on both the server and the phone, users will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system.

Asterisk has always been about flexibility, allowing integrators and developers to create highly customized solutions. Likewise, Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren’t simply XML pages; Digium phone apps can interface directly with core phone features.

Digium has leveraged this unique programming interface of the phones to create a suite of productivity applications that work with both Asterisk and Switchvox. Switchvox includes a unique web interface called Switchboard that gives each system user control of their personal communications environment. Digium has extended the capabilities of the Switchboard to the phone, putting advanced features like presence management, searchable contact directory, queue monitoring, recording and voicemail control, all at the user’s fingertips.

The Digium phones include the following models:

D40—An entry-level HD IP phone with 2-line keys. This is Digium’s best value phone, designed for any employee in the company.
D50—A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field keys with an easy to print paper label strip for the user’s most important contacts. This model is perfect for users who spend a lot of time on the phone.
D70—An executive-level HD IP phone with 6-line keys and 10 rapid dial/BLF keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.

Digium plans to have general availability of these new phones in April 2012. The MSRP for these models is as follows: D70 - $279, D50 - $179 and the D40 - $129.

Digium Releases Octal-Span Digital Card; Connects Traditional Telephony Services with Asterisk Communications Systems

VoIP Monitor — Tue, 15 Nov 2011 22:28:44 GMT

Digium announces the availability of the TE820 Octal-Span digital card. This new high-density solution compliments Digium’s existing broad suite of telephony card offerings designed specifically for Asterisk-based communications systems. The TE820 enables Asterisk integrators and OEMs to build large scale telephony deployments that are both high performance and cost-effective.

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. The combination of Digium hardware and Asterisk software provides a cost-effective platform for building numerous communications solutions, from PBX systems and VoIP gateways to IVR servers, call centers and complete unified communications suites. The TE820 supports up to 192 channels (in T1/J1 mode) or 240 channels (in E1 mode) and is available with or without hardware echo cancellation.

The TE820 card supports industry standard telephony protocols, including multiple variants of Primary Rate ISDN. Each span can be configured as either CPE or network for optimal flexibility. The optional VPMOCT256 hardware echo cancellation module, based on the industry-leading Octasic chipset, offloads the task of echo cancellation from the CPU, increasing overall system performance and call quality.

The Octal-Span digital card will be available on November 18, 2011 from Digium and Digium partners.

Digium and Open Source Community Release Asterisk 10 at AstriCon

VoIP Monitor — Fri, 28 Oct 2011 18:02:19 GMT

Digium releases Asterisk 10. Asterisk is a communications platform that allows developers to create powerful business phone systems and unified communications solutions. Since its introduction 12 years ago, Asterisk has been used, free of charge, in nearly every country of the world to power telephone and other communications systems. It has been downloaded millions of times, including two million last year alone, establishing Asterisk as the most popular open source telephony engine.

The most important new feature in Asterisk 10 is its wide-band media engine. Digium has replaced Asterisk’s telephony-grade media engine with a more advanced one, providing support for studio-quality audio and a nearly unlimited number of codecs. By supporting high and ultra high-definition voice, Asterisk can now be used to power communications applications that would have otherwise required specialized or expensive equipment and service in order to convey nuances in speech or emotion. Digium has also updated Asterisk’s media support for Asterisk 10, with several new codecs, including Skype’s SILK codec, 32kHz Speex support and pass-through support for CELT.

Built with open source community support

Digium is advancing Asterisk with version 10, while simultaneously leading work on the Asterisk Scalable Communications Framework. Asterisk SCF will allow developers to create real-time communications applications that include voice, video and text that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions.

Asterisk 10 makes its debut at AstriCon, the Asterisk User Conference & Expo, in Denver. Hundreds of attendees, including software and PBX developers, enterprise IT pros, systems integrators and call center and CRM developers, welcomed the announcement. In its eighth year, AstriCon is offering conference tracks focusing on technical information, carriers and call centers, cloud computing, commerce, government, enterprise and the Asterisk ecosystem. Developer conferences geared toward contributors to the Asterisk and Asterisk SCF projects are also taking place during this year’s AstriCon.

Asterisk 10 is available for free download and is licensed under the GNU General Public License v2.

New features in Asterisk 10

Asterisk 10 offers developers, integrators, resellers and telephony pros a range of new capabilities. A few include:

New media engine—Asterisk 10 supports more media types and virtually any type of audio. The overhaul to the media engine allows Asterisk to support a nearly unlimited number of codecs.
More codecs—The platform includes new codecs, including the wideband version of Speex, Skype’s super-wideband SILK and pass-through support for several CELT variants.
Additional sampling rates—Asterisk previously operated on 8 and 16 kHz sampled audio, but now supports super- and ultra-wideband sampling rates as file format types for file playback or recording. Asterisk now supports 8, 12, 16, 24, 32, 44.1, 48, 96 and 192 kHz rates for superb audio quality.
New conferencing application—Digium replaced the MeetMe conferencing bridge with an HD-capable intelligent bridge application called ConfBridge. It supports all codecs and conference rates and works on any Asterisk 10 system, regardless of operating system or architecture. Intelligent mixing algorithms provide each participant with the optimal audio quality for their connection. Also, ConfBridge is fully customizable, so systems administrators and integrators can configure call-in menus on a caller-by-caller basis.
Support for videoconferencing—ConfBridge relays video of a designated speaker or the current speaker to other participants in the conference. Video-capable SIP devices that use the same codec are required.
Significant new fax capabilities—Asterisk 10 includes T.38 gateway capabilities that allow outgoing fax calls from analog fax machines to be connected to T.38 fax endpoints over SIP and incoming T.38 fax calls to be delivered directly to fax machines. This allows for more straightforward integration of fax capabilities into an Asterisk system and allows users to get delivery confirmation from other fax machines.
Text message routing—Asterisk has long been able to send and receive text messages, but can now route messages as well. Asterisk 10 supports the SIP MESSAGE and XMPP protocols, allowing it to act as a text messaging server and bridge between different messaging protocols.

Digium Flips the Switch on AstriCon 2011, the Event Devoted to Open Source Asterisk

VoIP Monitor — Mon, 09 May 2011 17:45:41 GMT

Digium is making plans for the eighth annual AstriCon, the Asterisk User Conference and Expo and the event of choice for hundreds of technical pros working with the world’s leading open source communications platform. This year, from October 26-28, AstriCon will come to the Westin Westminster in Denver, Colorado. In addition to opening registration today, Digium is now accepting submissions for speaker opportunities at www.AstriCon.net. Topics are open, but past sessions have focused on basic and advanced Asterisk tutorials, IP telephony security, call center and enterprise case studies, and product training.

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. Using it, developers and other technical pros craft solutions such as IP PBXs, VoIP gateways, interactive voice response systems, conference bridges, voicemail servers and more. Asterisk also forms the basis for Digium’s award-winning Switchvox Unified Communications solution, which offers the most advanced business phone system features in a cost-effective, easy-to-use solution that scales as companies grow. The Asterisk community includes more than 65,000 members worldwide, hundreds of whom have made substantial contributions to the software’s development since its release more than 10 years ago.

In addition to AstriCon, Digium will hold developer conferences on Monday, October 24 and Friday, October 28 for Asterisk and Asterisk SCF, respectively. These two special events are open, at no cost beyond the AstriCon registration fee, to any developer who has contributed code to the Asterisk or Asterisk SCF projects. The conferences give contributors an opportunity to help define, in an interactive group of their peers, the direction of each project.

Digium Introduces First Complete Online Training Course for Asterisk

VoIP Monitor — Wed, 30 Mar 2011 18:44:10 GMT

Digium introduces Asterisk Essentials, an online training curriculum for Asterisk users. This virtual training class includes more than six hours of interactive content focusing on the fundamentals required to install and configure a basic Asterisk system. Step-by-step technical examples, best-practice recommendations and Asterisk tips help application developers and systems administrators gain expertise to build communications systems for their companies or customers.

Asterisk Essentials provides numerous examples and teaches developers how to implement their own Asterisk solutions using the core interfaces-configuration files, dial plan scripts and the command line interface. Students will learn to install Asterisk; configure IP phones and VoIP services; create automated attendants; configure voicemail and company directories; set up call queues; and debug Asterisk systems.

Created by the Digium training department, users can be confident that the information and best-practice examples come from the foremost experts on Asterisk. This course prepares students to take the Digium Certified Asterisk Associate exam and other more advanced Asterisk courses.

Asterisk Essentials is available now from the Digium online store for a discounted introductory price of USD $299.00 through July 1, 2011. The course will regularly list for $349.00. Students can access the online curriculum for up to six months from the date of purchase. For more information on Asterisk Essentials and other Digium courses, visit http://www.digium.com/training.

Digium Expands Listing Opportunities for AsteriskExchange and Waives Listing Fees for a Limited Time

VoIP Monitor — Wed, 02 Mar 2011 15:39:24 GMT

Digium is adding a new listing category to AsteriskExchange dedicated to custom applications for Switchvox, the company’s line of VoIP business phone systems. Switchvox provides an easy way to create custom applications that integrate the communications system with business processes to improve productivity. Individuals and companies who are already developing for Switchvox are encouraged to promote and potentially sell their applications in this new category on AsteriskExchange. To jump-start the development and promotion of the Switchvox Solutions category and the AsteriskExchange, the standard fees are being waived for all Asterisk and Switchvox-related listings.

AsteriskExchange is a dynamic marketplace offering everything needed for success with open source and unified communications. Users can quickly find both free and commercial Asterisk- and Switchvox-related products. User reviews, ratings and popularity rankings, along with an advanced search tool simplify the process of finding the right product. The list of AsteriskExchange categories includes IP phones and gateways, speech recognition and synthesis engines, voice prompts, video integrations, call center solutions, vertical application packages and more.

Other applications listed in the new Switchvox Solutions category include:

Connect CRM: CRM Integration by Dynamic Solutions Group
Cabmate Integration: Call Center System Integration for Yellow Cab by Dynamic Solutions Group
Shadow CMS Enterprise: Call Accounting Software by Resource Software Int., Ltd
StarCONNECT: Auto-Dialer/Appointment Reminder by Starnet Data Design, Inc.

Asterisk and Switchvox developers are invited to list a product, service, or application and take advantage of waived fees for a limited time.

Digium Intensifies Momentum Around Asterisk in 2010, Closes Year with More than Two Million Downloads

VoIP Monitor — Mon, 10 Jan 2011 19:01:49 GMT

Digium caps off a year marked by strong growth in the use of Asterisk and substantial technical advances in the product. Contributions from the open source community matched Digium’s investment in Asterisk over the past year. To date, more than 9,800 people have contributed code to Asterisk, including more than 200 who worked on Asterisk 1.8. After releasing version 1.8 in October, the momentum continued later that month when the company also announced a new open source project, Asterisk SCF. Enthusiasm for Asterisk among users, including developers, resellers, integrators and systems administrators, also increased as they downloaded the software more than two million times in 2010.

Asterisk turns an ordinary computer into a communications server that can power IP PBX systems, VoIP gateways, conference servers and other communication applications. In over 170 countries today, small businesses, large enterprises, call centers, carriers and governments are using Asterisk to create standards-based, feature-rich communications systems at a fraction of the cost of proprietary systems. Digium estimates that more than one million servers around the world are currently running Asterisk to handle billions of minutes of phone calls.

A few highlights from 2010 include:

Asterisk Scalable Communications Framework—The new framework, which is currently under development, will allow 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. Digium designed Asterisk SCF to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance. Asterisk SCF does not replace Asterisk; Digium is aggressively developing both projects in parallel.
Asterisk 1.8—Released two months ago, the significant update includes 200 enhancements, including new security features, integration with IPv6 and extensive additions to ISDN-BRI functionality. Asterisk 1.8 is a Long Term Support release, indicating four years of support from Digium.
Gartner Magic Quadrant for Corporate Telephony report—In late summer, Digium was positioned in the Visionaries quadrant of Gartner’sMagic Quadrant for Corporate Telephony report.
AsteriskExchange—In January, Digium announced the official online marketplace featuring both free and commercial products built or integrated with Asterisk. AsteriskExchange helps ecosystem participants market their products to the Asterisk community.

Digium Introduces New Open Source Project, Asterisk SCF

VoIP Monitor — Thu, 28 Oct 2010 16:31:57 GMT

Digium launches the Asterisk Scalable Communications Framework. 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. The Asterisk SCF platform will support, as a part of its basic architecture, the full range of real-time IP communications, including video, multi-channel wideband and ultra-wideband audio, chat, desktop sharing and other media types that may arise in the future.

Asterisk SCF is not a replacement for Asterisk, the world’s most widely used open source voice communications platform. Digium and the Asterisk community are committed to the continued development and support of Asterisk, the telecommunications software. This commitment is emphasized by the recent announcement of the release of the next long-term support version of Asterisk, Asterisk 1.8.

A version of Asterisk SCF for early adopters is now available at the Asterisk website. The project will evolve aggressively during the next 12 months. Digium invites developers interested in contributing or wanting to learn more about Asterisk SCF to visit http://www.asterisk.org today.

Digium Releases Asterisk 1.8 Open Source Telephony Platform

VoIP Monitor — Thu, 21 Oct 2010 17:25:23 GMT

Digium announces the release of Asterisk 1.8, a significant update to the software that developers, integrators, resellers and others around the world use to create customized, feature-rich and cost-effective business phone systems. Asterisk 1.8 includes more than 200 enhancements, including new security features, integration with IPv6 and extensive additions to ISDN-BRI functionality. Asterisk 1.8 is designated as a Long Term Support release, indicating four years of support from Digium. Hundreds of members of the open source Asterisk community will take an in-depth look at the software's new capabilities at next week's AstriCon Conference & Expo, which runs from October 26-28 near Washington, D.C.

Technologists working with small and large businesses, government agencies and municipalities, call centers and carriers have downloaded Asterisk more than two million times over the past 12 months. It can be used to create nearly any type of phone system or voice application, with some of the most popular uses being IP PBXs, voice gateways, voicemail, interactive voice response, conference bridges and automatic call distributors for call centers.

Multitude of New Features Boost Security, Scalability, ISDN-BRI Support

Asterisk 1.8 includes contributions from hundreds of community developers, as well as the Digium development team. For a comprehensive list of additions and changes, and to download the new version, visit http://www.asterisk.org. A few of the updates include:

Secure real-time transport protocol support-New end-to-end VoIPencryption of signaling and media to compliment the existing encrypted signalingsupport.
Security event framework-Modular capability for collecting and distributing security events within Asterisk.
Extensive additions to ISDN-BRI functionality-Call completion services, connected party identification, ETSI advice of charge, message waiting indicator, call rerouting and call deflection.
Session initiation protocol changes-Substantial increase in the speed of registrations, transport layer security improvements and more flexible network address translation handling.
IPv6 support-Integration with next-generation networks.
Calendar integration-Support for Microsoft Exchange, CalDav and iCalendar.
Channel event logging-Enhanced call tracking and logging for better audit trail and billing purposes.
XMPP distributed messaging-Better scalability for message waiting and device state.
Improved internationalization and localization-Asterisk offers improved handling of concatenated audio playback (dates, numbers).
Google Talk and Google Voice support-Inbound and outbound support for Google Talk and Google Voice calling.
High-resolution timestamps for call data records-Carrier and enterprise users can track call times to the microsecond.
Better support for voice codecs-16 kHz signed linear media streams are now supported. Additional HD voice codecs supported.
PacketCable NCS 1.0 support-Allows cable companies to use Asterisk as an option to create business services.
Default de-noise for conference bridge calls-Conference calls will sound clearer.
ConfBridge application enhancements-DAHDI hardware is no longer required to use this software feature. New call conferencing application that does not require the DAHDI kernel interface to operate.
Pitch shift functions-The pitch of audio, including of callers' voices, can be manipulated.
Multicast RTP paging-Extremely efficient and scalable method for handset paging.
Faster development and more robust unit testing-Digium has implementedAgile development and a new automatedtestingframework. The Agile process streamlines development and gives Asterisk users a better view into development plans.

Asterisk 1.8 is released under the GNU General Public License. It is free of charge and available for download at http://www.asterisk.org.

Dialogic Becomes Digium Interoperability Partner

VoIP Monitor — Tue, 15 Dec 2009 17:58:05 GMT

Dialogic has met the program requirements to become a Digium Interoperability Partner by completing the certification of the Dialogic 1000 Media Gateway Series and the Dialogic 2000 Media Gateway Series for use by the Asterisk community. Digium's Interoperability Partners have products that are complementary to and interface with the open source Asterisk telephony platform. These products interact with Asterisk through a SIP standards-based interface and are now certified by Digium for interoperability with Asterisk Business Edition.

Dialogic Media Gateways, including DMG1000 Gateways and DMG2000 Gateways, are widely used to provide PBX integration between applications deployed on SIP-based media servers and the installed base of TDM and hybrid IP-PBX systems. Open source software such as Asterisk has emerged as a viable SIP service creation platform used to create innovative communication applications that can be integrated with existing PBX infrastructures. In the absence of direct SIP to SIP integrations between an Asterisk-based solution and an existing PBX, Dialogic Media Gateways can provide the signaling and media translation necessary to make the solution work.

TELES Unveils Environmentally Friendly Green PBX

VoIP Monitor — Tue, 24 Nov 2009 18:51:11 GMT

TELES is now offering ASTERISK-based PBX software with its family of VoIPBOX gateways to provide enterprises with a compact, fully featured, energy efficient, and environmentally friendly solution that is easy to install and simple to use.

Initially, the integrated PBX will be available on the VoIPBOX BRI and the VoIPBOX GSM products but will soon be incorporated in most of the VoIPBOX family of gateways offering interfaces to BRI, FXS, FXO, GSM, UMTS, and VoIP telephony. Integration into VoIPBOX GSM will also enable customers to benefit from additional cost reductions by converting expensive landline-to-mobile calls into more cost effective mobile-to-mobile calls.

TELES Green PBX will be released with full service and support available to customers.

Asterisk Turns 10; Celebrates a Decade of Powering Telephony

VoIP Monitor — Thu, 08 Oct 2009 15:02:57 GMT

Digium announces the 10th anniversary of the creation of Asterisk, the world’s leading open source telephony engine and toolkit. Asterisk was created by Mark Spencer who is the founder and CTO of Digium, the company that created, leads and coordinates Asterisk. Digium will be celebrating Asterisk’s 10 year anniversary at the AstriCon conference taking place from October 13-15, 2009, at the Renaissance Glendale Hotel and Spa near Phoenix, Arizona. More about this annual event is available at http://www.astricon.net.

When created by Mr. Spencer in his dorm room in 1999, Asterisk provided an opportunity for open source enthusiasts and developers to create and customize a PBX system, which until then was not possible. Asterisk grew in popularity and is now downloaded more than 1.5 million times per year for use by individuals and organizations interested in an alternative to expensive and cumbersome proprietary phone systems.

Over the years, thousands of individuals and organization have contributed to the development and growth of the Asterisk open source project with new codes (more than 2,000 new code commits in 2009), configurations and applications. Today, Asterisk is downloaded nearly 5,500 times a day and boasts a community of 63,000 active participants on Asterisk forums, covering 28,500 topics with 92,000 forum posts.

“When I put the Asterisk platform out there 10 years ago -- using the Linux operating system and my own PBX code -- I never imagined the profound impact that it would have. I just believed that Asterisk could serve as an affordable and flexible telephony solution,” said Spencer. “The strength of Asterisk is a reflection of the creativity and ingenuity of the community along with the value that Asterisk provides its users. It’s been gratifying to be part of its impressive growth so far and we are excited to help it evolve in the future.”

Found throughout the world and in businesses of all sizes, Asterisk is 40-80 percent less expensive than traditional telephony systems and is more flexible, allowing users to integrate their phone systems with existing business-critical applications or easily write custom programs that extend the value of their phone systems. Asterisk can be found across many industries including retail, financial services, insurance, real estate, government and healthcare. In addition, Asterisk has spawned countless new business models, from service providers (more than 200 worldwide today) and traditional telephony companies, to technology integrators and application developers.

Digium and Incendonet Extend Asterisk with Speech Recognition Solutions

VoIP Monitor — Tue, 22 Sep 2009 17:12:43 GMT

Digium and Incendonet announce that companies using Asterisk for their IP-PBX communications needs can now add speech recognition-based solutions in a plug and play manner to improve customer service, reduce operating costs and increase mobile worker productivity with Incendonet’s SpeechBridge.

Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company’s product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own VoIP systems. Software developers, resellers and telecom professionals choose Digium’s products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.

Incendonet empowers organizations to add advanced speech capabilities as a seamless enhancement to their IP-PBX and other IT network investments at a price point never before possible for enterprise speech recognition technology. SpeechBridge is the perfect complement to core enterprise applications, and with its integrated speech-attendant, email and calendaring applications customers will begin realizing a return on their investment from day one. With an emphasis on ease of deployment, self-configuring technology allows for SpeechBridge to be provisioned and deployed in under an hour. Now all the benefits of automated speech recognition technology are no longer reserved for large enterprises and call centers.

Digium and Vestec Partner to Offer Full-featured, Low-priced Speech Recognition for Asterisk

VoIP Monitor — Wed, 16 Sep 2009 16:37:15 GMT

Digium and Vestec partner to provide a robust, low-cost speech recognition engine for use with the open source Asterisk telephony platform. Vestec’s speech engine significantly lowers the cost barrier for introducing feature-rich speech recognition with Asterisk and provides the Asterisk community with a powerful means to enhance the customer experience as well as to generate new revenue.

Speech recognition technology has historically been affordable to only a small segment of the Asterisk community. Vestec’s speech engine significantly broadens the market for speech recognition applications with Asterisk by offering a powerful speech solution at a cost of $99 per port, without any minimum port license purchase requirement. The speech engine is easy to install, supports industry standard grammar, provides a vocabulary size sufficient for most applications, supports all major Asterisk releases and Linux distributions and includes the first year of maintenance. A low-cost, optional annual maintenance subscription will be available after the first year.

Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company’s product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own VoIP systems. Resellers, telecom professionals and software developers choose Digium’s products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.

The Vestec Speech Engine is available immediately from the Digium web store at a list price of U.S. $99 per port, and will be available through Digium North American channel partners in the near future.

Digium and TeleMatrix Announce Interoperability Partnership

VoIP Monitor — Tue, 21 Jul 2009 16:50:01 GMT

Digium and TeleMatrix announce an interoperability partnership that provides Asterisk dealers and developers with a wide variety of new SIP and DECT SIP telephone solutions for hotel guestroom applications. TeleMatrix is an ISO 9001 and RoHS-certified manufacturer of SIP corded, DECT SIP cordless, analog and digital Centrex telephones for hotel, education, healthcare and general business applications. Through the 2006 merger of TeleMatrix and Scitec, the combined companies have deployed seven million telephones at 30,000 commercial customer locations worldwide. Hotel customers include Accor, Best Western, Choice Hotels International, Hilton, Hyatt, International Hotels Group, Marriott, Starwood, Swissotel and Wyndham.

Digium created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software and the cost-effective alternative to proprietary communication software. Digium offers Asterisk free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company’s product line includes a wide range of hardware and software to enable resellers and customers to implement turnkey VoIP phone systems or to design their own custom telephony solutions.

Digium Releases Hardware-based Voice Compression PCI Express Card for Asterisk Systems

VoIP Monitor — Tue, 30 Jun 2009 13:48:20 GMT

Digium releases the TCE400B PCI Express card for use with voice applications based on the open source Asterisk telephony platform. The new card provides hardware-based voice compression and decompression (codec) capabilities to shift transcoding from software to hardware. Using the TCE400B in place of a software-only solution places fewer demands on servers and frees up Asterisk to more efficiently process calls and to provide functionality for phone systems such as call recording, conference calling and interactive voice response.

Asterisk is the most widely used open source telephony engine and tool kit. By offering flexibility to access and alter the software code, Asterisk empowers developers and systems integrators to create VoIP communication solutions that match the specific needs of a business. Many businesses use Asterisk as a PBX to manage phone calls, but it’s also commonly used as a gateway between IP and PSTN networks, a telephony feature server and as the basis for call center applications.

In addition to its support for G.729a transcoding, the TCE400B gives Asterisk the ability to convert G.723.1 compressed audio into other formats, a capability not otherwise possible with Asterisk or software-only solutions. The card’s capabilities allow transcoding between simple, G.711 u-law and a-law, and complex, G.729a 8.0 kbit/s and G.723.1 5.3/6.3 kbit/s, codecs. When running in G.729a mode, the TCE400B can support 120 simultaneous transformations; in a mixed G.729a and G.723.1 mode, it supports 92 simultaneous transformations.

Xorcom Shipping Asterisk Based Channel Banks

VoIP Monitor — Thu, 07 May 2009 18:37:10 GMT

Xorcom has commenced shipping its Astribank 2 channel banks. The new models feature support for the TwinStarT full redundancy solution for complete Asterisk-based PBX systems, including telephony interfaces, which was awarded the "Best of Show" award in the On-site Product Launch category at the IT Expo held earlier this year. The enhanced system architecture in Astribank 2 now enables field-upgrades via a sophisticated licensing system, allowing activation of additional digital ports in existing modules to provide customers with the highest possible flexibility in scaling up their system.

TwinStar - High Availability Solution for Complete Asterisk-based Systems
TwinStarT is a high availability solution for Asterisk-based PBX systems. TwinStar provides automatic detection of server failure and immediate switching of all telephony functions, including telephony interfaces, to a back-up server. This quick and automatic failover process keeps downtime to an absolute minimum.

The TwinStar solution is comprised of two Asterisk servers with identical configurations, and Xorcom Astribank 2 channel banks featuring dual USB ports. It features:

1. Full dual-server redundancy for complete Asterisk PBX systems, including telephony interfaces
2. Automatic detection of server failure and switching to backup server, within minutes
3. Firmware-based switching mechanism that is not network-dependent

New Licensing Mechanism Supports Remote Activation of Ports and Features
A new software-driven licensing mechanism allows an increased number of possible configurations. From now on, all BRI and PRI (R2) modules will be capable of supporting up to 8 and 4 ports respectively. As a result, any PRI (R2)/BRI unit in which not all ports are active at purchase may be upgraded remotely to activate more ports. In addition, each Astribank unit is "TwinStar-ready", and can be upgraded remotely to enable this feature. Visitors to the Xorcom Web site can use the on-line Product Configurator to get an immediate suggestion of the best configuration to meet their needs.

Activating the New Astribank Units
Both Zaptel and the new DAHDI Asterisk drivers are supported in all Xorcom products. Until the new Astribank drivers reach all the distributions, the latest drivers from Xorcom are available for download from the Xorcom Web site in order to activate the new Astribank units. When appropriate software packages are installed, it is possible to use combinations of the old (Astribank) and the new (Astribank 2) devices connected to the same Asterisk server.

Cost-Effective Telephony Expansion in a Compact Unit
The Astribank channel bank provides businesses a tremendous opportunity for expansion at a fraction of the traditional cost of telephony equipment. The Astribank driver has been included as an official part of Zaptel (now DAHDI) since release 1.2.4, making Astribank interoperable with all current Asterisk PBX. The Astribank channel bank supports up to 32 analog ports, up to 8 BRI ports, and up to 144 PRI + FXS/FXO analog ports on a 19" 1U box (144 = quad E1 + 24 analog ports), high speed USB 2 connectivity, full support for fax machines, hot swappable architecture, and is rack- or wall- mountable.

The Atlanta Asterisk Users Group and Asterisk Conference and Install Fest

VoIP Monitor — Mon, 30 Mar 2009 19:05:03 GMT

When: Saturday, April 18, 2009, 9:00AM to 4:00PM

Where: Georgia Institute of Technology Christopher W. Klaus Advanced Computing Building 266 Ferst Drive, Atlanta, GA 30332

Admission: Free to Everyone. No Charge.

What: This year's program consists of two tracks in two lecture halls. One track focuses on the basic installation and configuration of Asterisk. That's where volunteers install and configure Asterisk on attendee supplied computers for free. The installation and configuration will be shown and explained using the lecture hall projection screens.

The second track in the other hall features more advanced topics for current users and system administrators. We have a full day of valuable presentations from internationally prominent users, developers and vendors.

Please review the full agenda and register now at http://atlaug.com/drupal/fest2009

Who: Enjoy presentations from leadingVoIP organizations including Digium, Sangoma, PBX in a Flash, Snom, Aastra, Cbeyond, Aretta, Pika, IP Comms, Fonica, Centric Voice and VoiceRoute.

Atlanta Asterisk Users Group (ATLAUG) - This group is one of the nation's most respected and active Asterisk Users Groups. It meets regularly for the purpose of exchanging information, providing mutual aid and fostering the growth of Asterisk systems in personal and business applications.

Digium Launches Support Services for Open Source Asterisk

VoIP Monitor — Mon, 30 Mar 2009 18:10:48 GMT

Digium announces the general availability of support subscriptions for open source Asterisk. The software, which Digium’s Founder and Chief Technology Officer, Mark Spencer, created and released under the open source GNU General Public License 10 years ago, is now the world’s most pervasive open source telephony platform. The new Asterisk support services allow organizations of any size to leverage the power of open source Asterisk with the confidence that their system is supported by a world-class support organization. The support subscriptions provide technical support, hardware replacements and substantial discounts on training programs to enable users to take full advantage of the power of the Asterisk platform.

Asterisk is a telephony communications engine and toolkit that developers, VARs and systems integrators use to create custom voice applications and sophisticated VoIP communications solutions tailored to meet specific corporate requirements. Asterisk has long been the software of choice for companies that need highly customized or tightly integrated phone systems as well as those who simply wish to take advantage of the many benefits associated with open source software.

Asterisk support subscriptions are bundles of services sold on an annual basis. They include technical and engineering support, consultative services, advance hardware replacement and discounts on Asterisk training and conference passes.

Asterisk support subscriptions are available immediately from the Digium webstore at http://store.digium.com and will be available through Digium channel partners in Q2. SMB pricing begins at U.S. $595 per year for support during the subscriber’s business hours (8:00 a.m.-5:00 p.m., Monday through Friday); 24x7 support for an SMB begins at U.S. $1,995 per year. Enterprise subscriptions, including 24x7 support, begin at U.S. $3,995 per year. Pricing includes a defined number of servers supported and cases opened per year.

Digium Sees Sharp Rise in Asterisk Downloads in 2008

VoIP Monitor — Wed, 17 Dec 2008 15:55:30 GMT

Digium announces that its Asterisk open source telephony software was downloaded 1.5 million times in 2008, more than any other year in company history and an impressive 50 percent higher than last year. The announcement caps a year of milestones and accolades for Digium and marks a promising start to 2009, when Digium celebrates the 10th anniversary of Asterisk, the open source software that has changed the telecommunications world.

Digium’s strong 2008 year highlights the attractiveness of less expensive, easily customizable open source software in the current recession. Asterisk is the world’s dominant open source telephony software. As the economic crisis worsened, Asterisk downloads rose by 32 percent from September through December, compared to a year ago.

Digium’s top 2008 accomplishments include the launch of the beta version of Skype for Asterisk, which enables customers to make, receive and transfer Skype calls from within their Asterisk phone systems. Digium also completed the integration of Switchvox, provider of the world’s foremost open source-based IP PBX, which Digium acquired in 2007. After successfully integrating Switchvox’s channel partners, the Digium Authorized Reseller Program now includes nearly 400 resellers.

Additional accomplishments of 2008 include:

Digium’s Switchvox AA300 – The award-winning AA300 appliance is a flexible, easy-to-use IP PBX that gives companies with up to 150 users a turnkey business telephone solution based on Asterisk software. Its release completed the initial planned build-out of the Switchvox appliance family, along with the AA60 appliance and the AA350 appliance. After reviewing the Switchvox AA300, Matthew Nickasch, a blogger from Network World, wrote, “In my opinion, if there’s an appliance or SMB IP-PBX to beat, this is the one.”
Digium Exceptional Satisfaction Program – A bold new guarantee program ensuring the quality of Digium’s hardware and software products and underscoring the reliability of open source technology. The Digium Exceptional Satisfaction Program is the most comprehensive product guarantee program in open source telephony today.
Four Millionth Port – Digium announced delivery of the four millionth port for connecting telephony systems based on Asterisk to communication networks. The number demonstrates the rapid rise in popularity of Asterisk for managing voice communications and integrating voice with corporate data.
Sponsored Industry Events – 2008 marked the second Digium|Asterisk World, an event to advance open source IP telephony communications in business environments. Additionally, Digium hosted the fifth-annual AstriCon, the industry’s largest, most informative Asterisk community event, which boasted expanded track sessions and high-caliber industry keynote speakers. The event recognized the winners of the annual Digium “Innovation Awards,” which honor businesses using Asterisk in new and exciting ways.
AsteriskNOW 1.5 – Digium released a new version of the award-winning AsteriskNOW software appliance that includes the popular FreePBX Web-based Asterisk management interface and simplifies the process of installing, operating and managing an Asterisk-based telephony system.
Industry Awards – Digium’s Switchvox appliance family continued to collect honors for product excellence, including Unified Communications’ “Product of the Year” and “Best of Show” and “Best of Open Source” at TMC’s INTERNET TELEPHONY Conference and EXPO West 2008. Asterisk won a “2008 Technology of the Year” award from InfoWorld and a “Product Leadership Award” from SearchNetworking.com. eWEEK named Digium Founder and CTO Mark Spencer to its “100 Most Influential People in IT” list and FierceVoIP named Spencer one of 10 “VoIP Leaders.” As a company, Digium was recognized on Linux Magazine’s “2008 Top 20 Companies to Watch,” by VoIP-News as a “Top 20 VoIP Influencer,” and by FierceVoIP as one of the “Fierce 15” VoIP companies of 2008.

FBI Warns of New Vishing Attacks Targeting PBX Systems

VoIP Monitor — Tue, 09 Dec 2008 17:05:00 GMT

The FBI has identified a new technique used to conduct vishing attacks where hackers exploit a known security vulnerability in Asterisk softwar e. Asterisk is free and widely used software developed to integrate PBX systems with VoIP digital Internet voice calling services; however, early versions of the Asterisk software are known to have a vulnerability. The vulnerability can be exploited by cyber criminals to use the system as an auto dialer, generating thousands of vishing telephone calls to consumers within one hour.

Digium released a Security Advisory, AST-2008-003, in March 2008, which contains the information necessary for users to configure a system, patch the software, or upgrade the software to protect against this vulnerability.

If a consumer falls victim to this exploit, their personally identifiable information will be compromised. To prevent further loss of consumers’ PII and to reduce the spread of this new technique, it is imperative that businesses using Asterisk upgrade their software to a version that has had the vulnerability fixed.

Further, consumers should not release personal information in response to unsolicited telephone calls. Providing your PII will compromise your identity.

“As with all types of scams, whether by computer, phone, or mail, using common sense can protect you,” said Special Agent Richard Kolko, Chief, National Press Office, Washington, D.C.

To receive the latest information about cyber scams, please go to the FBI website and sign up for e-mail alerts by clicking on one of the red envelopes. If you have received a scam e-mail, please notify the IC3 by filing a complaint at www.ic3.gov. For more information on e-scams, please visit the FBI's New E-Scams and Warnings webpage.

Camrivox Joins Digium Partnership Program

VoIP Monitor — Wed, 17 Sep 2008 20:40:24 GMT

Digium announces that Camrivox has joined the Digium Technology Partnership Program. This partnership will deliver a true desktop CTI solution to the world’s leading open source telephony ecosystem.

The Camrivox Flexor CTI Software family of products is designed to offer a simple, cost-effective and scalable solution to small and medium-sized businesses by unifying on-demand CRM (Customer Relationship Management) applications with IP PBX telephony and VoIP handsets. With the forthcoming release of Flexor Connect for Asterisk, Camrivox introduces an extra dimension to the Asterisk community and delivers tangible benefits to businesses intent on maximizing their customer interaction.

The partnership and Flexor Connect for Asterisk enable users of Asterisk installations to benefit from the full family of Flexor CTI Software products, including CTI for Outlook, Salesforce, NetSuite and Microsoft Dynamics CRM. Through integration with Asterisk, Flexor CTI Software provides call logging, call reporting, click-to-dial, contact screen pop-ups and on-screen call control. Furthermore, this PBX-centric approach allows future Flexor Software product releases to support home workers and remote workers, bringing the benefits of CTI to a mobile workforce.

Digium is the creator and driving force behind Asterisk, the open source voice communications software deployed by more than four million servers worldwide to manage VoIP calls for businesses and individuals. More resellers, telecom professionals and software developers choose Digium's products than those of any other open source telephony company because only Digium delivers the technical superiority, security and flexibility associated with Asterisk. Asterisk powers Digium’s family of software and hardware appliances including AsteriskNOW, Asterisk Business Edition and Switchvox.

Camrivox will be demonstrating Flexor CTI Software with Asterisk and snom IP Phones at the upcoming AstriCon Open Source Telephony Conference and Expo, booth #311, held in Glendale, Ariz. from Sept. 23-25.

101 Things You Can Do With Asterisk Contest

VoIP Monitor — Wed, 04 Jun 2008 17:12:39 GMT

Over the last nine years Asterisk has emerged as world’s leading open source telephony engine and tool kit, however most people simply know it as an open source phone system.

In order to promote greater visibility as to the myriad of things that people can (and have) done with Asterisk, VoIP Supply has partnered with Digium to launch the “101 Things You Can Do With Asterisk Contest”.

Here are the pertinent details:

1. One contributor (at random) to the list of 101 things, will win a $1,500 VoIP Supply shopping spree sponsored by Digium and VoIP Supply.

2. To be eligible for the contest, one must simply post a use or application of Asterisk that they themselves (or someone they know has) actually done.

3. The contest is open until 101 unique uses or applications have been documented.

This not just an opportunity to pick-up some free some free hardware, but a great way to promote the open source community and find out all of the creative things people are doing with Asterisk.

Digium and Fanstel Announce Partnership

VoIP Monitor — Mon, 02 Jun 2008 18:00:16 GMT

Digium and Fanstel announce a partnership that will help ensure interoperability between Fanstel telephones and Digium products. Fanstel SIP telephones are now “Digium|Asterisk Certified,” allowing companies that want the flexibility and cost-savings of a VoIP platform based on Asterisk Business Edition to benefit from full-featured Fanstel telephones.

Digium is the creator and driving force behind Asterisk, the open source voice communications software deployed by more than 4 million servers worldwide to manage VoIP calls for businesses and individuals. More resellers, telecom professionals and software developers choose Digium's products than those of any other open source telephony company because only Digium delivers the technical superiority, security and flexibility associated with Asterisk. Asterisk powers Digium’s family of software and hardware appliances, including AsteriskNOW, Asterisk Business Edition and Switchvox.

Fanstel has delivered high-quality telephone products to 55 countries. A vertically integrated telephone manufacturer with engineering and production facilities, Fanstel offers customization and branding services for made-to-order products. In addition to Asterisk-interoperable and ready-to-use SIP phones, Fanstel makes customization and branding services available to Asterisk communities.

Fanstel SIP phones offer many advanced features, such as multiple line appearances, DND, transfer, pick-up, park/unpark, conferencing, intercom, busy lamp field, XML messaging and auto-provisioning, as required in enterprise applications. In addition, up to five parties can be conferenced locally, saving PBX processing power to support more extensions.

Digium and Integrics Partner

VoIP Monitor — Mon, 12 May 2008 20:25:48 GMT

Digium and Integrics have partnered around the sales and marketing of Integrics’ Enswitch to carriers worldwide. The agreement provides a route for closer technical, sales and marketing collaboration between the companies and expands Digium’s involvement in the carrier market.

Digium is the creator and driving force behind Asterisk, the open source telephony software deployed by more than 4 million servers worldwide to manage VoIP calls for businesses and individuals. More resellers, telecom professionals and software developers choose Digium’s products than those of any other open source telephony company because only Digium delivers the technical superiority, security and flexibility associated with Asterisk. Asterisk powers Digium’s family of software and hardware appliances, including Switchvox, Asterisk Business Edition and AsteriskNOW.

Integrics sells Enswitch to telecommunications companies, Internet telephony service providers, VoIP providers and others to allow them to create offerings such as full-featured, telephone management and billing solutions, calling card integration, toll free and number translation services, voicemail, call queuing, automatic call distributor, fax to email and multi-level interactive voice response.

Enswitch is based on Digium’s Asterisk Business Edition or open source Asterisk, based on each customer’s preference, as well as other open source software such as OpenSER, MySQL and Linux. Enswitch runs on a single machine or on an Asterisk/OpenSER cluster and supports high availability and failover, allowing for failure of any single machine with only a few seconds of interruption to service. The clustered architecture, which can be geographically distributed, also allows additional machines to be added as the customer’s network grows.

Digium and Infradapt Announce Enhanced Relationship

VoIP Monitor — Tue, 04 Mar 2008 18:57:05 GMT

Infradapt is proud to announce its enhanced relationship with Digium, the Asterisk company, as a Select Reseller. Asterisk offers a full menu of business phone features like unified messaging, advanced voice mail features, conference calling and other highly desirable productivity features.

Digium is the creator and driving force behind Asterisk, the open source voice communications software deployed by more than 3.5 million servers worldwide to manage VoIP calls for businesses and individuals. More resellers, telecom professionals and software developers choose Digium's products than those of any other open source telephony company because only Digium delivers the technical superiority, security and flexibility associated with Asterisk. Asterisk powers Digium's family of software and hardware appliances including AsteriskNOW, Asterisk Business Edition and Switchvox.

Infradapt has deployed and supported Asterisk-based telephony systems since 2003. The Company's current clients include nationally recognized Fortune-500 and Fortune-1000 companies, healthcare providers, financial services firms, banks, real estate companies, manufacturers, ad agencies, non-profits and research centers.

Digium and Metaphor Solutions Announce Partnership

VoIP Monitor — Tue, 26 Feb 2008 18:08:33 GMT

Digium and Metaphor Solutions announce a partnership offering its On-Demand Plug & Play IVR application suite for the Asterisk market. Plug & Play IVR allows Digium customers to use the web to self-configure, customize, deploy, manage and report on high quality speech recognition-based application packages in minutes. Metaphor’s suite of web-configurable speech recognition applications provides businesses unprecedented access to cost-effective speech IVR solutions. Small and mid-size businesses can easily configure these speech IVR solutions to connect to their on-premise or managed Asterisk IP-PBX platforms.

Speech-recognition IVR applications have been proven to increase customer satisfaction, reduce customer service cost and are preferred by callers over touch-tone applications. However, it takes a large upfront investment, specialized development skills and several months of effort to deploy a high quality speech IVR application. Metaphor Plug & Play environment lets enterprises quickly self-configure, deploy and manage high quality speech IVR customer service solutions without specialized technical expertise. Customers can use any of Metaphor IVR’s 55+ pre-built applications by configuring extensions to connect to the Metaphor Asterisk peer.

Digium is the creator and driving force behind Asterisk, the open source voice communications software deployed on more than 3.5 million servers worldwide to manage VoIP calls for businesses and individuals. Asterisk powers Digium’s family of software and hardware appliances including AsteriskNOW, Asterisk Business Edition, and Switchvox. “Speech recognition applications greatly enhance customer service solutions offered on the Asterisk IP PBX platform,” said Jim Webster, director of technology partnerships for Digium. “Metaphor’s Plug & Play Speech IVR dramatically lowers the time, cost and complexity of implementing speech IVR solutions for our small and mid-size business customers.”

Customers can run free trials and dial into live applications by visiting www.metaphorivr.com. Examples of applications include Inbound and Outbound Customer Surveys, Proactive notifications, Telephone Banking, Bill Collection, Payment by Phone, Insurance Claims Service, Order Status, Appointment Scheduling, Prescription Refill, Location Finder, Inventory, Pricing and Catalog Request.

Digium Launches One-stop Shop for VoIP Deployment Needs

VoIP Monitor — Tue, 08 Jan 2008 18:44:43 GMT

Digium unveils the “Digium|Asterisk Marketplace” to give the Asterisk community of end-users, developers, small businesses and enterprises a one-stop online destination for all their open source VoIP deployment needs. Located at www.digium.com/marketplace, the Digium|Asterisk Marketplace enhances Digium’s Technology Partners web portal to allow companies that are not yet Digium partners to advertise their products to the Asterisk ecosystem.

Visitors to the Digium|Asterisk Marketplace will be able to search the company listings by keyword to easily find Asterisk-compatible products such as phones, network infrastructure devices, and application software, as well as services such as VoIP and PSTN providers. The website opens up a new, direct line of communication between independent vendors of Asterisk-related products and the entire Digium|Asterisk ecosystem.

Digium Partners receive prominent and prioritized Digium|Asterisk Marketplace listings as part of their partnership agreement. For vendors not otherwise affiliated with Digium, marketplace listings will be offered starting at $395 per quarter for a basic listing. During a special promotional period, listings will also be available starting at $795 for a full year. Companies interested in a listing can sign up directly on the Digium website, and upon approval of the application, have their listing posted within a few business days. Digium plans to continue to expand the capabilities and features of the Marketplace in the future.

Digium Caps Off Record Year with One Millionth Asterisk Download

VoIP Monitor — Wed, 19 Dec 2007 17:12:39 GMT

Digium hits the one millionth download of Asterisk in 2007, capping off a record year for the leading open source telephony company. Digium, which will complete its 24th consecutive quarter of growth and profitability this year, created headlines with new executive appointments, industry awards, strategic partnerships and acquisitions aimed at further advancing the company’s presence in the small-to-medium-sized VoIP market.

Open source Asterisk continues to revolutionize the VoIP market by giving companies a cheaper and more flexible alternative to proprietary solutions. The 2007 introduction of Digium’s Asterisk Appliance based on Embedded Asterisk Business Edition successfully combined the cost savings and flexibility of open source with the rapid installation, high reliability and ease of use of a business appliance. PC Magazine called the Asterisk Appliance “astonishingly compact” and packed with “high-end features.”

Additional 2007 highlights include:

Executive Appointments – Danny Windham joined the company as CEO from ADTRAN, a $500M+, publicly-traded networking company. Mark Spencer, founder of Digium and creator of Asterisk, assumed the role of CTO to focus on growing the future of Asterisk and developing new unified communications applications based on open source. Digium also added a VP of Sales and VP of Engineering to round out its management team.

Acquisitions – With its new executive leadership team in place, Digium looked to grow its presence within the Asterisk open source community and SMB VoIP market. In July, Digium acquired Sokol & Associates, the premier provider of training for Asterisk users and producer of the AstriCon open source user conference. In September, Digium completed its acquisition of Switchvox, a leading provider of IP PBX phone systems powered by Asterisk for SMBs.

Partnerships – In October, Digium announced a partnership with 3Com Corporation to offer “3Com Asterisk” to small businesses that need a reliable, easy-to-deploy voice solution based on open standards and open source. Digium also announced a licensing agreement with NTT Software Corp., a subsidiary of Japan’s largest phone company, to sell products based on Asterisk Business Edition in Asia.

Industry Awards – Digium won InfoWorld’s Best of Open Source Software award and was the only open source company recognized in the VoIP category. Digium was named the top emerging VoIP vendor by Computer Reseller News and its newly acquired Switchvox SMB 3.0 Software received the publication’s “Tech Innovator” award. Network World named Digium an “open source company to watch” and CRM Magazine gave Digium its “Rising Star” and “Service Excellence” award. Locally, Digium accepted the Alabama “Governor’s Trade Excellence Award” for its commitment to local and international business.

Reseller Program and the Digium Asterisk Marketplace – In 2007, Digium announced a new Authorized Reseller Program and unveiled the Digium Asterisk Marketplace to give the Asterisk community of end-users, developers, small businesses and enterprises a one-stop online destination for all their open source VoIP deployment needs.

Digium|Asterisk World – This year marked the inaugural debut of Digium|Asterisk World, an event solely dedicated to advancing open source IP telephony communications in business environments. The event included a ceremony announcing the winners of the first annual Digium “Innovation Awards” honoring businesses putting Asterisk to use in new and exciting ways.

OrecX Joins Digium as Software Partner

VoIP Monitor — Tue, 04 Dec 2007 21:55:22 GMT

Digium announces OrecX’s support for Asterisk Business Edition, the professional version of Digium's open source telephony engine. Compatibility with Asterisk Business Edition allows OrecX to provide small and mid-sized businesses running the Asterisk platform a modular, cross-platform recording and retrieval system for audio streams. OrecX has also become a Digium|Asterisk Software Partner to enhance the relationship between OrecX and Digium.

By leveraging the power of open source, the OrecX suite of applications delivers end users a voice recording platform that is easy to install and maintain and that continually evolves to meet the needs of Asterisk users at a significantly lower cost than proprietary voice recorders.

OrecX’s support for a wide range of VoIP protocols, such as SIP, MGCP and H.323, makes it possible for businesses to quickly and easily integrate voice recording into VoIP-based phone systems or migrate existing phone systems to OrecX’s VoIP recording platform, according to Bruce D. Kaskey, founder and vice president of marketing for OrecX, who announced OrecX’s support for the Asterisk platform at Fall VON.

Digium Releases Switchvox Free Edition

VoIP Monitor — Wed, 31 Oct 2007 17:29:15 GMT

Digium announces the immediate availability of Switchvox Free Edition. The new product includes the telephony features of the widely used Switchvox SOHO edition and can be downloaded and installed in minutes on existing hardware. The release underscores Digium’s commitment to delivering full-featured telephony systems at low price points to allow organizations around the world to benefit from the Asterisk open source movement and flexibility of VoIP.

Switchvox products are based on Asterisk, the open source telephony software created and owned by Digium. Switchvox Free Edition includes tools such as an interactive voice response editor that allows system administrators to create auto attendants and menus. It also includes features such as voicemail to email, find me/follow me, unlimited calling queues (automatic call distribution, or ACD) and advanced reports on system use.

Switchvox Free Edition supports Digium’s line of analog cards. Users can install Switchvox Free Edition on existing hardware or use Switchvox-certified hardware. For details on hardware support, see http://www.switchvox.com/certified_hardware.

Switchvox Free Edition is on display this week at Fall VON 2007 and Digium|Asterisk World at the Boston Convention and Exhibition Center. The product may be downloaded from http://www.switchvox.com.

Digium Sets the Stage for AstriCon 2007

VoIP Monitor — Tue, 28 Aug 2007 19:37:59 GMT

Digium is making final preparations for AstriCon 2007, the longest running conference of the most widely used open source telephony software, Asterisk. Digium created, owns and is the innovative force behind Asterisk and uses it as the basis for its own voice over IP (VoIP) solutions. Using a dual-licensing model, Digium licenses Asterisk for use by myriad software and telecommunications companies around the world to enable them to deliver their own differentiated solutions. Asterisk’s widespread use and feature set make it one of the most influential open source projects today.

AstriCon will be held from September 24-28 at the Carefree Resort & Villas near Phoenix, Ariz. Registration is open at www.astricon.net. The conference offers sessions for Asterisk experts as well as those just getting started with the software. Industry focus sessions are geared toward enterprise users, carriers and Internet telephony service providers, and call centers. Topics include:

Deploying open source telecommunications in enterprises.
Building an enterprise-class call center with Asterisk.
VoIP security.
Asterisk high availability and load balancing.
Clustering Asterisk.
Running Asterisk on embedded systems.
Peer-to-peer telephony applications.
Speech recognition and Asterisk.
Designing custom Asterisk applications.

“Asterisk will be downloaded more than 1 million times in 2007 and used to power voice communications in businesses with a few people and organizations with tens of thousands,” said Mark Spencer, creator of Asterisk and chief technology officer of Digium. “The software owes its success to the enthusiasm and involvement of the worldwide Asterisk community. In Phoenix at the sixth annual AstriCon, Digium looks forward to meeting with and hearing from all who have made Asterisk a part of their businesses, including customers, partners and software developers.”

Digium Founder Leads the Open Source VoIP Charge at VoiceCon 2007

VoIP Monitor — Tue, 21 Aug 2007 18:21:29 GMT

Digium the Asterisk Company announces that Asterisk creator and Digium Founder Mark Spencer will join leaders from Cisco, Avaya, IBM, Microsoft, Siemens and Skype to discuss the future of VoIP and unified communications. Digium’s Spencer will represent the open source telephony industry in this high-powered discussion as Digium continues its charge to move the revolutionary open source technology deeper into the enterprise market with emerging and new software architectures for unified communications.

Spencer’s “Software Architecture and Unified Communications” panel will take place on Thursday, August 23 from 10:30 am–11:30 am. Also representing the open source industry at VoiceCon 2007 will be Digium’s VP of Product Management Bill Miller who will present “Open Source’s Role in Converged Networks” with Digium customer Southern Company, one of America’s largest generators of electricity with 4.3 million customers. Miller’s presentation will take place on Thursday, August 23 from 8:00 - 8:45 AM.

“As the sole maintainer of the genuine Asterisk code, Digium is excited to see open source’s growing role at mainstream telephony events like VoiceCon 2007,” said Spencer. “Asterisk has grown from an open source technology with limitless potential to a technology that is changing the IP business landscape. Digium will continue to be the Asterisk community’s ambassador to the business world and offer solutions based upon the absolute latest in open source innovation.”

trixbox Pro Allows Employees to Call Each Other for Free

VoIP Monitor — Mon, 13 Aug 2007 18:58:24 GMT

Companies can now download trixbox Pro, a free business-grade phone system that allows companies with five to 500 employees to make free VoIP calls. In addition to traditional PSTN dialing and PBX features, trixbox Pro also includes conference calling, unified messaging between voicemail and email, employee presence management, embedded corporate chat server, and the ability to integrate with Microsoft Outlook, Salesforce.com and SugarCRM.

trixbox Pro includes trixNet, a free in-network calling service which lets any trixbox Pro user call any other trixbox Pro user, using their regular phone numbers. Calls over trixNet are not subject to any local or long-distance charges. Fonality will soon extend trixNet to the trixbox CE (Community Edition) platform, its popular open source Asterisk-based software which has been downloaded more than 1.4 million times. In early 2008, Fonality will also extend free trixNet calling to include anybody using GoogleTalk, Google's instant communications service.

trixbox Pro is a hybrid-hosted phone system, which means the free software is first downloaded by a business and installed on a local computer and local IP phones. After this step, the local computer connects to the Fonality network where server health, call quality and usage are constantly monitored. The hybrid-hosted nature of trixbox Pro also securely extends the phone system outside the corporate firewall, so an employee’s extension can follow them when they work from home, remotely on a laptop, or even on a mobile phone. Fonality’s hybrid-hosted services are market proven and have processed more than 120 million calls for 2,500 companies with 53,000 individual extensions worldwide.

The base edition of trixbox Pro is free and there are two paid editions tailored specifically for the reseller market that are available for a monthly fee or one-time, per-seat charge. trixbox Enterprise Edition and trixbox Call Center Edition have the enterprise feature sets required for larger businesses and professional call centers, but at a low fixed or monthly fee. Fonality developed trixbox Pro EE and trixbox CCE to give resellers and IT professionals simple, easy to use phone software that saves them time so they can focus on growing their business, increasing revenue and delivering a higher level of customer service. Additionally, all three editions of trixbox Pro can be easily branded by resellers for a custom look-and-feel. Product logos, screen appearance and color schemes can all be tailored to match the brand of the reseller or end-customer. trixbox Pro will be available in seven languages: US English, UK English, French, German, Italian, Portuguese and Spanish.

For a complete list of features and more information please visit www.trixbox.com.

Asterisk-based Telephony Platform Trixbox 2.4 Released

VoIP Monitor — Wed, 08 Aug 2007 19:12:46 GMT

Fonality announces the beta release of trixbox 2.4, the latest version of the Asterisk-based telephony platform that has become synonymous with open source telephony and home to the largest Asterisk-based community. trixbox 2.4 incorporates the newest releases of CentOS 5, Asterisk 1.4 and FreePBX 2.3 for broader compatibility, choice and usability. trixbox 2.4 is beta software and timing of the general release will be gated by stability of Asterisk 1.4, as well as integration of all of the components and testing to ensure that it is ready for deployment in a demanding production environment.

For greater compatibility, this release gives trixboxers the option to deploy trixbox on CentOS 5 compatible hardware and reduce install time with the 100 percent Red Hat Package Manager Installer. For greater choice, trixbox incorporates Asterisk 1.4, the latest release from the Asterisk community. The new version of FreePBX 2.3 included in trixbox 2.4 also improves usability with new features including an updated flash operator panel and an improved FreePBX landing page with more status information.

trixbox 2.4 beta, like previous versions of trixbox, includes a host of stable and reliable packages and applications, including Apache, a modified version of Asterisk, FreePBX, Flash Operator Panel, MySQL, phpMyAdmin and SugarCRM. trixboxers who prefer a hardened and supported version of trixbox can still download and use trixbox 2.2. Other key features included in trixbox 2.4 include an automatic back-up and restore system and DHCP manager.

“Listening to the trixbox community is our first priority. They wanted to see Asterisk 1.4 incorporated into trixbox, so we are delivering just that. This beta release is just the first step,” said Andrew Gillis, trixbox founder at Fonality. “And, by having trixbox 2.4 based on CentOS 5, trixbox is compatible with the newest hardware on the market.”

The release of trixbox 2.4 is just one of the latest improvements in a series of updates for both trixbox itself and the community. In recent months, Fonality has added service and support for trixbox, the trixStore, the trixbox Buyer’s Club, additional Fonality trixbox Open Communication Certifications (FtOCC, pronounced "F-talk") and the popular trixbox Appliance that was awarded four stars by PC Magazine and was named a 2007 TMC Labs Innovation award winner.

FtOCC courses consistently sell out and have proven to be a useful and important way for Asterisk resellers to get practical, hands-on experience with the latest software developments, and also share best practices. There are two upcoming FtOCCs scheduled; FtOCC at ITEXPO will take place in Los Angeles on September 10-12, 2007 and FtOCC London will take place in London from October 9-11, 2007. To register for FtOCC go to http://www.trixbox.org/ftocc

For more information on trixbox or to download trixbox 2.4 for free, go to www.trixbox.org.

Digium Expands Asterisk Support with Subscription-based Services

VoIP Monitor — Tue, 31 Jul 2007 19:08:45 GMT

Digium introduces a new subscription-based service offering designed to give Digium Asterisk customers and partners faster, easier and unlimited access to Digium service and support. Offered in Silver, Gold and Platinum levels, Digium’s subscription-based services will be available for all business-class products beginning with the Digium Asterisk Appliance, now available.

The voice communications market continues to experience rapid growth with more companies evaluating and choosing open source as a cost-effective and flexible alternative to proprietary telephony solutions. The increased demand for open source brings a greater business need for service and support during business hours and around the clock. Digium is the first open source telephony provider to offer businesses a service offering rivaling that of proprietary vendors in terms of price, scope and scale.

Benefits of Digium’s new subscription services include: extended hardware and software warranty; unlimited support by email; premium advance hardware replacement; configuration backup; premium support by phone; ongoing software and security updates; and an easy one-click login at Digium.com for access to these premium services.

“Small and medium-sized businesses today are looking for a VoIP solution at an affordable price and a guarantee that they won’t be left in the cold after the contract is signed,” said Danny Windham, CEO of Digium. “Digium’s Asterisk Appliance gives small to medium-sized business a full-featured VoIP solution at 20 to 50 percent of the cost of proprietary systems with a tiered services offering specifically designed to meet the needs of today’s growing companies.”

Pricing and Availability

The first year of subscription-based services is bundled into the Asterisk Appliance, which is now available. Beginning with the Asterisk Appliance, Digium will provide subscriptions for all of its products based on Asterisk Business Edition. For more information, please visit www.digium.com. The Asterisk Appliance with the first-year Silver Subscription will start at U.S. $1,145 and is available from Digium distributors and resellers in the U.S. and Canada. International versions will be available next quarter.

Digium Acquires Sokol & Associates

VoIP Monitor — Wed, 11 Jul 2007 18:01:14 GMT

Digium announces it has acquired Sokol & Associates. Digium has long enjoyed a close relationship with Steve Sokol and his colleagues, all of whom have joined Digium’s team. By acquiring the company, Digium gains an additional channel through which it can communicate Asterisk enhancements to the IT, telephony, software development, reseller and call center professionals who use the software every day.

Digium created, maintains, owns and is the innovative force behind Asterisk, the open source telephony communications software in use by more than two million servers today. Asterisk powers Digium’s family of software and hardware appliances and enables other companies, which often use older versions of the software in their own VoIP products.

Sokol & Associates has more than 10 years of experience working in the VoIP industry. The firm is staffed with network and systems engineers and software developers whose deep knowledge of Asterisk’s many features allow them to run intensive training sessions across North America and Europe. Steve Sokol, a prominent member of the Asterisk community, will take responsibility for planning future software offerings from Digium, which offers, in addition to Asterisk, AsteriskNOW, Asterisk Business Edition™ and the AsteriskGUI.

Sokol & Associates created and has managed AstriCon, the conference dedicated to expanding the knowledge of Asterisk, since 2004. Digium has been the Diamond sponsor of the conference, which will be held in Phoenix, Ariz., this year from September 24 through 28.

“Asterisk is one of the most commonly used open source projects today, and as the community of users and contributors grows, we want to deepen our relationship with them,” said Mark Spencer, founder and chief technology officer of Digium. “Sokol & Associates has run Asterisk training classes around the world and has long been a central part of this community. By bringing Steve and his group on, Digium gains another conduit to new and advanced Asterisk users around the world.”

Digium Launches a New Line of PCI Express Cards for Use with Asterisk

VoIP Monitor — Tue, 12 Jun 2007 16:54:24 GMT

Digium the Asterisk company announces a new family of telephony interface cards based on the PCI Express (PCIe) format. PCIe is fast becoming the dominant form factor in expansion cards for server, workstation and desktop systems because it offers numerous performance benefits over traditional PCI and PCI Extended interface formats. Digium is rolling out PCIe cards across its full product family and will continue to support interface cards based on the PCI and PCI Extended formats.

Digium is the creator and driving force behind Asterisk, the open source voice communications software deployed by more than two million servers and serving well over 10 million people today. The company’s hardware, including the new PCIe cards, is designed to help customers realize the full power and flexibility of Asterisk by meeting precise requirements for scalability; network connectivity; and support for IP, traditional analog or mixed telephony lines.

“PCI Express is quickly becoming the industry preference for server and workstation interfaces for expansion cards,” said Bill Miller, vice president of product management and marketing at Digium. “Digium’s goal in offering a full line of PCIe cards, and in continuing to support PCI and PCI Extended formats, is to offer our customers and partners the widest array of choices in how they use and deploy Asterisk solutions.”

Digium’s four- and two-port digital T1 and E1 PCIe interface cards, the TE420 and TE220, eliminate voltage and slot considerations associated with standard PCI cards. They are available immediately from the company and its authorized resellers. The TE420 will retail for $1,195 USD and the TE220 for $695 USD. Both are compatible with Digium’s existing VPMOCT series of hardware echo cancellation modules. Digium is scheduled to release analog and other cards for PCI Express in Q2 and Q3 2007.

Fonality Continues Leadership

VoIP Monitor — Tue, 17 Apr 2007 22:18:26 GMT

Fonality announces that more than 2,000 customers with 45,000 phones in 37 countries have adopted its open source business phone system, PBXtra, and used it to place more than 75 million calls. Fonality continues to reliably scale as small and medium-sized businesses rapidly adopt PBXtra, the world’s largest commercial Asterisk deployment. Continued growth is fueled by demand among smaller companies for high-quality, value-priced alternatives to the expensive, rigid offerings sold by traditional communications companies.

Fonality is transforming the telephony market by combining the cost savings of open source software with award-winning ease-of-use, making sophisticated business phone systems a reality for small to medium-sized businesses worldwide. Fonality’s patent-pending, hybrid-hosted technology enables seamless use of PBXtra to support branch offices, as well as mobile and home-based workers.

PBXtra includes a modified version of the open source IP-PBX platform Asterisk that Fonality has hardened for reliability and layered with feature-rich applications. Fonality’s business phone systems deliver more functionality with significantly improved ease-of-use at 40 to 80 percent less than the cost of comparable systems from larger companies. Out of the box, PBXtra Standard, PBXtra Professional and PBXtra Call Center support VoIP calling, advanced auto-attendant features, telecommuters, branch offices, drag-and-drop call control, conferencing and intra-office intercom, as well as a host of other advanced features. Fonality provides 24x7 technical support, and its patent-pending hybrid-hosted architecture ensures reliability by allowing real-time monitoring and remote system management.

PBX Prompts: Voice Prompts for Asterisk

VoIP Monitor — Wed, 04 Apr 2007 20:08:21 GMT

Sayers Media Group announces the launch of PBX Prompts. PBX Prompts offers a variety of standard voice prompt packages for the Asterisk Open Source PBX and custom voice prompts for your IVR, Voicemail, or in addition to our standard prompts packages. Recorded in professional sound studios, PBX Prompts’ standard, advanced, and custom voice prompts give your company a voice that is representative of your business. With no minimums, and a quick 72 hour turnaround, PBX Prompts can meet the demands of any voice prompt project.

“Over the past two years, we have heard over and over again about the difficulties many small medium businesses and value added resellers have had finding high quality professional voice prompts for Asterisk Open Source PBX systems,” said Garrett Smith, Director of Sales and Marketing for Sayers Media Group. “Based on these experiences we have created PBX Prompts with the help of these very same companies in order to deliver on a simple, easy to use, ordering interface and installation process for those who want an alternative to the default Asterisk voice for their phone system.”

PBX Prompts is currently offering voice prompts packages for Asterisk systems featuring male and female voice talents in English, English, and Spanish languages. PBX Prompts plans on launching additional languages, such as French, German, and Japanese, in the coming weeks. Prices for the current voice prompt packages for Asterisk systems range from $49.99 for select standard voice prompt packages for Asterisk systems featuring over 500 voice prompts, to $129.99 for advanced voice prompt sets for Asterisk systems that feature over 600 voice prompts. From now until May 1st, PBX Prompts is also offering FREE voice prompt packages for Asterisk® systems that contain 100 of the most popular voice prompts for Asterisk.

For more information on PBX Prompts’ voice prompts packages for Asterisk, visit http://www.pbxprompts.com.

Thanks for the heads up Garrett.

AsteriskNOW Software Preps for Full Release

VoIP Monitor — Wed, 21 Mar 2007 15:28:51 GMT

Digium announces the final beta of its AsteriskNOW software appliance. New AsteriskNOW beta 5 features, as well as extensive product testing and feedback from the Asterisk community, make this the final beta leading up to Q2 availability of AsteriskNOW 1.0. Customers can deploy AsteriskNOW in minutes to start using Asterisk, the most popular open source telephone system in the world, in their organizations.

Introduced last quarter by Digium, the creator and primary developer of Asterisk, AsteriskNOW includes all the software customers need to install and configure a phone system that offers the advanced features found in expensive proprietary products. AsteriskNOW is free and can run on customers’ existing servers. An easy-to-read AsteriskNOW screen helps customers who do not have an Internet VoIP service provider choose a Digium Asterisk Certified Service Provider or add another that they select.

An AsteriskNOW installation demo will be held in the Digium World Theatre at VON on Wednesday, March 21 at 11:00 a.m. Copies of the new book AsteriskNOW for Dummies are also available at the conference. Those not attending VON can view a YouTube video of Mark Spencer walking users through the easy installation.

AsteriskNOW was awarded “Best of Show” for Innovation at Internet Telephony Expo in Fort Lauderdale, Fla. in January of this year. Attendees at this week’s VON can also see the AsteriskNOW demos in Digium’s booth on a regularly scheduled basis.

Highlights of AsteriskNOW beta 5 include:

Asterisk software, a customized Linux distribution and the AsteriskGUI graphical user interface needed to run AsteriskNOW.
Installation using a setup wizard, making AsteriskNOW easy to configure and deploy.
An expanded group of Digium Asterisk Certified Service Providers—IAXtel, New Global Telecom, Simple Signal, VoicePulse and Voilà IP—that ensure interoperability between their VoIP services and AsteriskNOW. Customers may also select their own VoIP service provider.
The ability to manage multiple dial plans to allow organizations to tailor telephony features for different functional groups.
Direct inward dial features that have been added to the setup wizard to more easily let administrators define rules for handling calls placed to direct lines.
Music on hold per queue options that allow the organization to have different recorded messages or music played for callers waiting for customer service, for example, than for sales.
GUI error handling designed to guide users when they have keyed an extension or option that does not exist.
Support for alphanumeric extensions to give system administrators more flexibility in assigning phone number extensions.
Context-sensitive advanced tooltips that help administrators are now fully functional.
Internet Explorer 6 and 7 browser support, in addition to Firefox.
New drivers to support the latest hardware within the Linux distribution.
Additional updates based on significant product quality testing.

Availability

AsteriskNOW is licensed under the open source general public license and is free to download and use. The software will be available at http://www.asterisknow.org/ beginning on Friday, March 23.

trixbox Appliance Mean and Green

VoIP Monitor — Mon, 19 Mar 2007 20:55:43 GMT

Fonality announce a new enterprise telephony appliance based on trixbox and a range of complementary professional support options. The trixbox Appliance comes pre-installed with the trixbox software platform and is an industrial grade rack-mountable server with dual hard disk and dual power supply redundancy options. It is aimed at businesses with five to 500 employees and can be purchased for use with VoIP, E1/T1 or up to 48 analog lines. Created for trixbox and Asterisk resellers, as well as IT professionals, the trixbox Appliance costs $999 and delivers the industry’s best price performance value for a PBX.

The trixbox Appliance is powered by Intel and comes with pre-configured Sangoma line cards with industry-leading Octasic echo-cancellation hardware inside. Broad support for the appliance has been announced by a growing trixbox ecosystem that includes phone manufacturers Polycom, Aastra, and Grandstream, as well as VoIP service providers VoicePulse and Teliax.

“We are excited about the new Intel-based trixbox Appliance, which combines an innovative SME offering with Polycom’s line of standards-based high quality IP phones,” said Sunil Bhalla, senior vice president and general manager of voice communications at Polycom. “The open source telephony market is proving to be very viable and Fonality’s appliance should be a great addition because of its feature set and flexibility.”

The trixbox Appliance includes trixbox 2.2, a new release of the popular distribution that integrates Asterisk with Apache, MySQL, SugarCRM and PHP. Included in this release is the new open source Asterisk GUI (graphical user interface), a component of the AsteriskNow distribution, which gives customers an additional trixbox GUI option.

In addition to delivering an appliance, Fonality is offering professional support options to ensure that a reliable trixbox engineer is always just a phone call away. A new suite of comprehensive technical support packages can be purchased on an annual basis for a complete deployment, or in hourly increments for ad hoc support.

“The trixbox Appliance, built on Intel-based servers, is another proof point that open source telephony solutions can deliver full-featured yet very affordable solutions to the global mid-market,” said Lisa Lambert, managing director, Software and Solutions Group, Intel Capital. “Fonality's investment in trixbox should provide the opportunity for the company to grow their market position and leadership.”

“While Fonality has long offered its own PBXtra appliance, we felt the trixbox community also needed a reliable hardware appliance to run their trixbox software. Price pressure is forcing do-it-yourselfers to put together systems based on mediocre hardware not meant for the rigors of a busy phone system,” said Chris Lyman, Fonality CEO. “Our goal was to create a box that had enterprise performance, but still came in at under $1,000 bucks.”

For more information about the trixbox Appliance, trixbox 2.2 and trixbox support options, please visit www.trixbox.org.

Digium Expands Presence at Spring 2007 VON

VoIP Monitor — Thu, 15 Feb 2007 19:13:52 GMT

Digium the Asterisk company, has more than doubled its presence at Spring VON, through the addition of new exhibiting partners, and the creation of the Digium Theatre on the VON show floor. As a result, the Asterisk Pavilion now occupies 5,000 sq. feet of exhibit space, representing the largest single entity in the VON Expo.

More than a dozen Asterisk partners are showcasing their wares in the Asterisk Pavilion, including: Aspect Software; Intel Corporation; LumenVox; Switchvox; TransNexus; VoicePulse; Polycom; Simple Signal; and others.

On Wednesday, March 21, at 9:30 AM, Digium President Mark Spencer will open the second day of the Expo by delivering his Industry Perspective session, entitled, "Calling Across the Chasm." Newly appointed company executives will be initiated into the Digium experience at VON, including, Steve Harvey, Bill Miller, and many others, who will discuss the company's next steps in the newly-created Digium Theatre -- which is set up adjacent to the Asterisk Pavilion. This theatre also provides a forum to provide training to third- party developers, and any other VON attendee interested in learning more about Asterisk-based, IP telephony solutions.

"This developing partnership is significant to the VON community because we will continue to reflect the global nature of Digium's technology in all of our VON programs around the globe," said Bill Sell, vice president and general manager of VON Events. "We're looking forward to a tremendous show in San Jose, and the largest Digium pavilion ever will certainly be a centerpiece of the VON Expo."

The VON Expo (http://www.von.com/exhibitors.html) will provide attendees with the opportunity to compare and evaluate solutions from more than 300 exhibitors. VON conference sessions run each day from 9:00 AM - 5:00 PM across multiple tracks covering VoIP, IMS, Fixed Mobile Convergence (FMC), and IPTV. For a complete listing of sessions, please visit: http://www.von.com/schedule.html.

Load Balancer for VoIP Networks Released by Vocalscape

VoIP Monitor — Wed, 07 Feb 2007 18:52:55 GMT

Vocalscape announces that they have released a Load Balancer for Voice over IP (VoIP) systems.

"Vocalscape has developed the Load Balancer to meet our customers' needs," commented Ron McIntyre, President of Vocalscape. "As our customers grow their user base, they will need to add additional servers to handle the higher volume of calls. The Vocalscape Load Balancer will allow them to evenly share the load among multiple servers."

The Vocalscape Load Balancer began as an open source project which was adopted and improved upon by Vocalscape. It was made compliant with Asterisk, a popular open source PBX, and the algorithm was revised to more evenly distribute calls. Previously, the Load Balancer would send calls to a primary server and only when the primary server was overburdened would calls be sent to additional servers. The new algorithm balances the load by evenly distributing the calls between the servers. As an additional benefit, the Load Balancer provides failover capabilities. If a server is not responding, the Load Balancer will route all calls to servers that are functional.

trixbox Open Communications Certification Workshops

VoIP Monitor — Tue, 06 Feb 2007 19:10:17 GMT

Fonality announces the Fonality trixbox Open Communications Certification (FtOCC) workshop, the first in a series of training and certification courses for the trixbox application platform. trixbox, the largest and fastest growing Asterisk-based platform and community, is quickly becoming the de facto standard for open source telephony. The FtOCC program will provide the tools and knowledge necessary to successfully implement trixbox in a business environment.

“Telephony is in the midst of a sea change and sea changes mean huge opportunity. But capitalizing on opportunity requires training and education,” said Chris Lyman, Fonality founder and CEO. “The FtOCC courses will quickly teach attendees how to begin earning real money from trixbox and Asterisk.”

The course will focus on a myriad of subjects including, but not limited to: VoIP, PBX deployment, network assessment, telephony troubleshooting, T1/PRI training, and IP handset education. The goal of the FtOCC is to arm data VARs, system integrators and telephony professionals with the knowledge needed to deploy and manage PBX installations for businesses from 1 to 1,000 employees.

“FtOCC is the next logical step for the trixbox community,” said Andrew Gillis, founder of trixbox and director of community development at Fonality. “We are building on the explosive adoption of the trixbox application and providing formal training and certification so businesses can be built upon its customization and deployment.”

The FtOCC course will be hosted by Andrew Gillis, trixbox project founder and Kerry Garrison, senior product manager for trixbox at Fonality and author of “trixBox Made Easy.” Additional training will be given by the trixbox engineering team. Respected partners in the telephony and VoIP industry will also be participating to provide information and free equipment to attendees.

For more information on registration for the FtOCC or to download trixbox 2.0 for free, visit www.trixbox.org.

Helping Small and Medium-Sized Businesses Switch to VoIP

VoIP Monitor — Wed, 24 Jan 2007 18:01:10 GMT

Digium and Polycom announce a high-quality VoIP solution for small and medium-size businesses that simplifies the technical aspects of purchasing, configuring and deploying a complete VoIP phone system. The integrated solution, featuring the Digium AsteriskNOW software appliance and Polycom SoundPoint IP desktop phones can significantly lower the time and technical expertise needed to deploy a high quality VoIP solution; enabling more small and medium businesses to enjoy the cost savings, enhanced quality and productivity benefits of advanced IP telephony.

The simplified purchasing, configuration and deployment process is enabled by new capabilities in the Digium AsteriskNOW software appliance, including a one-click function called BuyNOW that enables customers to purchase Polycom SoundPoint IP phones, and an intuitive configuration process that automatically provisions the phones for immediate customer use. In addition, Digium's Asterisk software now supports Polycom HD Voice technology that enables calls with twice the clarity and sound quality of traditional analog calls.

"This expanded offering with Polycom makes it even easier for SMBs to deploy and manage an Asterisk-based solution -- from the beginning stages to deploying the phones," said Mark Spencer, president and CEO at Digium. "This development is part of our overall strategy to offer an easy, rapid migration to VoIP in an enterprise environment. Partnering with Polycom further emphasizes our commitment to providing users with a simple Asterisk installation, based on only the best quality products."

"Many small and medium businesses want an advanced, high quality and affordable VoIP solution, but the technical challenges can be a barrier," said Sunil Bhalla, senior vice president and general manager, voice communications at Polycom. "We are working with partners like Digium to deliver unique capabilities like Polycom HD Voice and to simplify the deployment process. As open source continues to play an important role in the evolution to VoIP, we look forward to working with Digium to provide innovative, cutting-edge solutions that help customers make this important transition."

The BuyNOW feature, within AsteriskNOW's Digium-designed GUI, greatly simplifies the phone purchasing process by immediately connecting users to NETXUSA, a recognized leading distributor of Voice over Internet Protocol (VoIP) products and services. With regional distribution offices and in-house certified engineers, NETXUSA assists business customers with the entire Digium-Polycom implementation. Polycom's full line of award winning and industry leading standards-based phones are available through BuyNOW including: the SoundPoint IP 650 with HD Voice, SoundPoint IP 601, SoundPoint IP 501, SoundPoint IP 430, SoundPoint IP 301 and SoundPoint IP Expansion Module. The line also includes the SoundStation® IP 4000, Polycom's market leading SIP-based conference phone.

The AsteriskNOW GUI also includes a simplified process for configuring and provisioning Polycom's full line of SoundPoint IP phones. The interface provides a step by step process that helps the user select from the broad array of features that are available in a combined Asterisk and Polycom solution. Once the selection process is complete, the configuration is automatically downloaded to the phone which immediately becomes active on the system.

In addition to the updates in the AsteriskNOW GUI, the Asterisk open source software (release 1.4.0) now supports Polycom's breakthrough HD Voice technology delivering the ultimate communications experience. Polycom's HD Voice includes wideband audio, enhanced signal processing, Acoustic Clarity Technology (which includes next generation technologies for transparent full duplex, echo cancellation, dynamic noise reduction, automatic gain control and microphone management) and specialized system design to deliver unrivaled clarity and richness. This enables significantly better voice clarity and improved intelligibility of information, which significantly improves comprehension and productivity while reducing listener fatigue. Polycom HD Voice is currently available on the SoundPoint IP 650 telephone.

These new capabilities are the result of collaborative efforts between Digium and Polycom and follow the recent announcement of their partnership to supply Polycom SIP-based phones as the exclusive phone in Digium's currently available Asterisk Appliance Developer Kit. Combined with Digium's Asterisk Business Edition, SMB customers benefit not only from a rich feature base capable of rapid deployments; but also, from a more affordable telephony solution as compared to proprietary systems

Trixbox 2.0 Released a Leading Asterisk-Based Platform

VoIP Monitor — Thu, 04 Jan 2007 17:43:13 GMT

Fonality releases trixbox 2.0, a free, easy to use, open source telephony and application platform. The new version, available for immediate download, can be installed in less than 15 minutes, supports multiple languages and provides increased reliability and stability, flexible user customization, and support for a wide-range of hardware vendors. The software also allows the community to upgrade individual deployment components versus having to reinstall from scratch with each upgrade. trixbox.org will also be hosting its first ever training Webinar entitled “Building An Open Source IP-PBX With trixbox 2.0” on January 30, 2007.

“The trixbox community said they wanted a super-reliable Asterisk deployment that removed all the headaches and that is what we strived to deliver in this release of trixbox 2.0,” said Andrew Gillis, founder of trixbox and director of community development at Fonality. “I always envisioned trixbox as a platform that would be both easy enough for people to quickly deploy and stable enough that they could stake their reputation on it. trixbox 2.0 is the realization of this vision.”

trixbox 2.0 comes with a new point-and-click package manager which lets installers, via simple clicks of their mouse, decide which applications they want to install with trixbox. The advantage of the package manager is two-fold: first, it lets installers choose how lean or rich of a deployment they need. Secondly, it informs the installer, over time, of any new updates to any packages within their trixbox installation as vendors release them.

trixbox 2.0 includes a host of packages, or applications, including Apache, a modified version of Asterisk, FreePBX, Flash Operator Panel, MySQL, phpMyAdmin and SugarCRM. In addition, the new release provides call detail reports, an endpoint manager, VoIP service provider wizards, deeper application integration with SugarCRM, drivers for Sangoma and Rhino voice cards and support for multiple languages including English, German, Portuguese and Spanish with more to come.

On January 30, 2007 Kerry Garrison, senior product manager for trixbox at Fonality and author of “TrixBox Made Easy,” a step-by-step guide to installing and running your home and office VoIP system, will host the first trixbox.org Webinar, designed to deliver important information to IT managers and integrators who want to deploy trixbox 2.0 installations. This 90-minute session will provide an overview of trixbox 2.0 and discuss what is included, system requirements, hardware and ITSP support, skills required for successful deployment, installation and configuration steps and additional resources available from trixbox.org.

“The new features in trixbox 2.0 make it the application standard for telephony-savvy businesses and integrators deploying open source IP telephony systems,” said Garrison. “The upcoming Webinar will provide the most up-to-date and useful information about how to successfully implement trixbox 2.0 installations.”

For more information, or to sign up for the trixbox Webinar or download trixbox 2.0 for free, visit www.trixbox.org.

SIPBox Becomes Polycom Certified Reseller

VoIP Monitor — Mon, 18 Dec 2006 19:05:16 GMT

SIPBox announced it has become a certified Polycom reseller. SIPBox recommends the use of Polycom VoIP phones in all of its telephony deployments, specializing in Asterisk, the industry’s first open source telephony platform.

“We are eager to continue our growth in the open source market through our relationship with SIPBox,” said Jim Kruger, Vice President, Voice Marketing at Polycom. “As a value-added-reseller, SIPBox is making significant progress in the VoIP market and together we hope to bring our customers the total VoIP solution and support needed in a fast growing enterprise.”

With customers ranging from K-12 schools to higher education to municipalities with multiple locations, SIPBox provides enterprises with 200+ users a complete VoIP solution and handles the entire process on site, from design and implementation to ongoing management and support. SIPBox guarantees all of their services and all implementations are enterprise and carrier grade.

“Aligning ourselves with a leading technology vendor in the industry such as Polycom allows us to continue to provide our customers with flexible and cost-effective VoIP solutions,” said Chad Agate, co-founder and CEO of SIPBox. “We believe that Polycom’s line of SIP phones will support the business telephony features of Asterisk and meet the growing needs of the market.”