LiveVoip LLC announced today that it will soon be providing small carriers as well as large VoIP service providers with a new, session border control functionality for building next generation, converged networks. LiveVoip’s unique FastTrack3 functionality allows carriers to place LiveVoip’s session border controllers inside their chosen network design, rather than having to make any critical network changes to incorporate the session border controller.

In a FastTrack3 configuration, where a carrier has converged voice and data, FastTrack3 now supports NAT functionality. FastTrack3 now processes management and data traffic from VoIP endpoints without the need for additional devices.

The FastTrack3 configuration is excellent for carriers and service providers that have an existing network in place with segregated voice traffic, and want to introduce added border control functionality and security for VoIP in a non-intrusive manner. FastTrack3’s ability to handle both configurations gives small and large carrier service provider’s maximum flexibility in their network engineering, less disruption to the existing network, and faster time to market with new services and applications. This gives them an important advantage that can separate them from the competition.

This new functionality makes FastTrack3 the one stop solution for the converged networks typically deployed by larger carriers and large service providers. With the ability to accommodate both FastTrack3 topologies, complete with VoIP and data NAT functionality.

“These capabilities were developed as part of LiveVoip’s own network connecting to other carriers,” says Joop Cousteau, Developer and Softswitch Division Vice President, of LiveVoip LLC. “All carriers want a session border controller to easily and flexibly be added and integrated to their IP network designs, rather than having to rework the network to fit the session border controller.”

The FastTrack3 functionality is another key advantage introduced by LiveVoip to service the needs of all carriers migrating to IP. LiveVoip is also the only vendor to provide carriers with architecture for superior application performance and integrated as well as split signaling and media for options and configuration. LiveVoip’s high-end FastTrack3 MAX is the most powerful and efficient Session Border Controllers available providing over 101,000+ concurrent calls in a compact 2U chassis. FastTrack3 MAX is destined to be the leading solution for hosted NAT traversal, and network protection for hosted IP telephony services like IP Centrex and managed Asterisk style IP PBX systems.

About LiveVoip LLC

LiveVoip LLC a new supplier and innovator of session border control technology that ensures customers, expert delivery of IP communications services. Our FastTrack3 MAX series of session border controllers overcome technical hurdles typically encountered at VoIP network borders. The company is headquartered in Mesa, AZ, with offices throughout the world. Launched in late 2003, LiveVoip, LLC is an Arizona based company privately funded Delaware company. More information is available at http://www.livevoip.com

FastTrack3 and LiveVoip LLC are registered trademarks of LiveVoip LLC. All other trademarks belong to their respective companies.

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Exinda, a global provider of WAN Optimisation solutions incorporating Unified Performance Management, will educate IP Expo 2011 visitors on the network performance challenges associated with running todays mission critical business applications and how to resolve them. As video, VoIP and other collaborative applications become must-haves, so businesses need to be able to run them without suffering any network performance issues. At IP Expo, Exinda will demonstrate to visitors how to do this through the latest innovations in WAN Optimisation techniques via two key seminars.

On Wednesday 19th October between 13:50-14:20 within the Data Centre Optimisation theatre, Exinda will discuss its Next Generation WAN Optimisation solution. Visitors will be able to learn how to improve ROI through overcoming network challenges when accessing cloud applications by implementing a Unified Performance Management solution. The presentation will also address the three requirements of application performance management: visibility, control and optimisation in a single unified platform.

On Thursday 20th October between 10:30-11:00 within the Network Optimisation theatre, Exinda will explain the intricacies of how it can help businesses make the most of the new Web 2.0 and video driven applications required by todays enterprises. With VoIP, SaaS, video delivery and hosted cloud traffic all needed on todays business networks, it is critical that the network continues to perform seamlessly even under the scrutiny of these intensive applications. Visitors will be able to learn how implementing Exindas next generation WAN Optimisation solution with policy-based Layer 7 application, visibility, control and dynamic video delivery can improve network performance, and therefore ROI.

Kevin Suitor, Vice President of Marketing at Exinda, explains, Exindas technology continues to lead the way, thanks to our complete range of WAN Optimisation solutions designed for businesses of tomorrow. The demands on organisations to offer the latest technologies that will give them a competitive advantage, both from employees and customers, is constantly increasing. What we at Exinda are offering is a way to maximise the use of these technologies through network optimisation techniques that will ultimately not break the bank.

Suitor continues, Through the two seminars at IP Expo, we are giving visitors the chance to see what can be achieved with the latest in WAN Optimisation techniques. And if this isnt enough, we will also be available on our stand to discuss visitors specific issues and problems.

Exinda will be available at IP Expo on stand number B28 from 19th-20th October at the Earls Court 2 centre in London.

About Exinda

Exinda is a proven global supplier of WAN Optimisation and Application Acceleration products. The company has helped over 2,000 organizations in over 80 countries worldwide improve the end user experience, manage application performance, manage congestion over the WAN and reduce network operating costs for the IT executive. For more information, please visit http://www.exinda.com.

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IPsando, an information technology company focusing on Internet Protocol (IP) communications consulting and Digium, the creator of open source telephony, today announced that AstriCon Europe 2005, the first of two annual Asterisk user conferences, will be held June 15-17, 2005 at the Auditorium Madrid Hotel in Madrid, Spain. In response to the large number of attendees at AstriCon 2004, IPsando and Digium selected the Auditorium Madrid Hotel, the largest urban hotel in Europe, to ensure that AstriCon Europe 2005 would be able to host the large number of telecommunications executives and open-source users expected to attend the conference.

Boasting more than 200,000 users in over 200 countries, Asterisk is changing the way businesses purchase telecommunications services and how carriers build their voice over IP (VoIP) solutions by providing a complete business- and carrier-class PBX solution based on a Linux computer.

“Asterisk has taken the telecommunications industry by storm – initially winning favor with Linux enthusiasts, but now gaining traction among telecommunications carriers and enterprises large and small,” said Steven Sokol, CEO of IPsando and AstriCon’s organizer. “As a result, we wanted to make sure that we are providing our Asterisk users with regular forums to discuss the future in open source telephony.”

The three-day conference will open with a keynote address by Mark Spencer, president of Digium, and consist of a day of tutorials, including three tracks ranging from beginner to expert , a conference day showcasing innovations, strategies and business cases, followed by a one-day developer’s summit. In addition to the conference, there will also be an exhibition, with vendors and service providers that work with Asterisk.

Topics to be addressed include:

Integrating the PBX with IT Infrastructure: Asterisk for the Enterprise

VoIP Migration In-a-Box: Asterisk for Service Providers

Lower Cost, More Flexibility: Asterisk for Call Centers

Your VoIP Swiss Army Knife: Asterisk for Developers

Managing Your Asterisk PBX: From the CLI to the GUI

Registration is open at https://www.AstriCon.net/europe/. An Early Bird discount of 20% is available through April 25, 2005. AstriCon is also accepting potential sponsorship and speaking applications for all AstriCon shows (including AstriCon Fall in the U.S.) through the AstriCon website http://www.AstriCon.net/europe/speakers/. Please include the proposed presentation title, a description of the presentation, as well as speaker bios and contact information.

About Asterisk

Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. It also supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure. Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using packet voice, it is possible to send data such as URL addresses and images in-line with voice traffic, allowing advanced integration of information.

About Digium

Digium is the creator and primary developer of Asterisk, the industry's first open source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures.

Digium solutions reduce the costs of traditional TDM and VoIP implementations through open source, standards-based software and next-generation gateways, media servers, and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, Loopstart, and GR-303. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX™ (Inter-Asterisk eXchange), SIP, MGCP, Cisco Skinny® (SCCP), and H.323 VoIP protocols.

The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respective owners.

About IPsando

IPsando is an information technology company focusing on Internet Protocol (IP) communications consulting and in particular Asterisk, the industry’s first open source PBX.

Founded by Olle E. Johansson and Steven M. Sokol, both well-known contributors to Asterisk and coordinators of AstriCon, the company runs conferences and training seminars, develops software and provides consultancy services. IPsando’s AstriCon training "Introduction to Asterisk" is a five day boot-camp that delivers the knowledge and insights needed learn, implement and produce Asterisk solutions.

Additional information is available at http://www.ipsando.com or by emailing info@ipsando.com.

Media Contacts:

Olle E. Johansson

IPsando

+1 816-278-1035

+46 70 593 68 51

oej@ipsando.com

Janine Savarese

MRB Public Relations

732-758-1100 Ext. 103

jsavarese@mrb-pr.com

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More Asterisk PBX Press Releases

IPsando, an information technology company focusing on Internet Protocol (IP) communications consulting and Digium, the creator of open source telephony, today announced that AstriCon Europe 2005, the first of two annual Asterisk user conferences, will be held June 15-17, 2005 at the Auditorium Madrid Hotel in Madrid, Spain. In response to the large number of attendees at AstriCon 2004, IPsando and Digium selected the Auditorium Madrid Hotel, the largest urban hotel in Europe, to ensure that AstriCon Europe 2005 would be able to host the large number of telecommunications executives and open-source users expected to attend the conference.

Boasting more than 200,000 users in over 200 countries, Asterisk is changing the way businesses purchase telecommunications services and how carriers build their voice over IP (VoIP) solutions by providing a complete business- and carrier-class PBX solution based on a Linux computer.

“Asterisk has taken the telecommunications industry by storm – initially winning favor with Linux enthusiasts, but now gaining traction among telecommunications carriers and enterprises large and small,” said Steven Sokol, CEO of IPsando and AstriCon’s organizer. “As a result, we wanted to make sure that we are providing our Asterisk users with regular forums to discuss the future in open source telephony.”

The three-day conference will open with a keynote address by Mark Spencer, president of Digium, and consist of a day of tutorials, including three tracks ranging from beginner to expert , a conference day showcasing innovations, strategies and business cases, followed by a one-day developer’s summit. In addition to the conference, there will also be an exhibition, with vendors and service providers that work with Asterisk.

Topics to be addressed include:

Integrating the PBX with IT Infrastructure: Asterisk for the Enterprise

VoIP Migration In-a-Box: Asterisk for Service Providers

Lower Cost, More Flexibility: Asterisk for Call Centers

Your VoIP Swiss Army Knife: Asterisk for Developers

Managing Your Asterisk PBX: From the CLI to the GUI

Registration is open at https://www.AstriCon.net/europe/. An Early Bird discount of 20% is available through April 25, 2005. AstriCon is also accepting potential sponsorship and speaking applications for all AstriCon shows (including AstriCon Fall in the U.S.) through the AstriCon website http://www.AstriCon.net/europe/speakers/. Please include the proposed presentation title, a description of the presentation, as well as speaker bios and contact information.

About Asterisk

Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. It also supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure. Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using packet voice, it is possible to send data such as URL addresses and images in-line with voice traffic, allowing advanced integration of information.

About Digium

Digium is the creator and primary developer of Asterisk, the industry's first open source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures.

Digium solutions reduce the costs of traditional TDM and VoIP implementations through open source, standards-based software and next-generation gateways, media servers, and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, Loopstart, and GR-303. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX™ (Inter-Asterisk eXchange), SIP, MGCP, Cisco Skinny® (SCCP), and H.323 VoIP protocols.

The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respective owners.

About IPsando

IPsando is an information technology company focusing on Internet Protocol (IP) communications consulting and in particular Asterisk, the industry’s first open source PBX.

Founded by Olle E. Johansson and Steven M. Sokol, both well-known contributors to Asterisk and coordinators of AstriCon, the company runs conferences and training seminars, develops software and provides consultancy services. IPsando’s AstriCon training "Introduction to Asterisk" is a five day boot-camp that delivers the knowledge and insights needed learn, implement and produce Asterisk solutions.

Additional information is available at http://www.ipsando.com or by emailing info@ipsando.com.

Media Contacts:

Olle E. Johansson

IPsando

+1 816-278-1035

+46 70 593 68 51

oej@ipsando.com

Janine Savarese

MRB Public Relations

732-758-1100 Ext. 103

jsavarese@mrb-pr.com

# # #

LiveVoip LLC announces its entry into the Global Home VoIP market. LiveVoip LLC will be entering the Home VoiP Markets in the, USA, Canada, and UK starting June 2005. LiveVoip intends to become the preeminent VoIP provider in the Global arena. LiveVoip will start with a suite of home user plans. As these plans progress, business and SOHO plans will be added. LiveVoip is a market driven company focused on the future.

”LiveVoip is thrilled to be playing a key role in the 200 Billon Dollar Global Voice Over IP market," ”said Pamela Patterson, acting President, LiveVoip LLC. “Livevoip LLC is proud to deliver VoIP over a network operating at Gigabit speeds that are unmatched anywhere in world today,”

“LiveVoip customers enjoy almost unlimited network transit capacity so the company can offer not only Voice Over IP but, Video Over IP Services as well,” said Joop Cousteau, VP Global Operations.” “LiveVoip technical experts will have service ready to be deployed to broadband connected customers in June 2005 and expects a very large demand.”

"LiveVoip is an amazing place," acting company President Pamela Patterson said. "I see no sign of things even slowing down."

About LiveVoip LLC

LiveVoip, LLC is currently serving the Global VoIP market. North America accounts for 70 percent of the 200 billion telephony markets. It is estimated that 56 percent of businesses in North America are considering migrating to VoIP in the next two years. LiveVoip is the world’s number one Resellers – Reseller to the Asterisk based PBX market. VoIP based systems are headed to the Internet in a big way and LiveVoip will be ready to service all clients worldwide.

Launched in late 2003, LiveVoip, LLC is an Arizona based privately funded Delaware Company. http://www.livevoip.com

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Related Asterisk PBX Press Releases

LiveVoip LLC, a leading global provider of advanced VoIP communications services, announced today yet another capacity upgrade of its VoIP switching NODES. LiveVoip is growing at a very fast rate and has needed to increase capacity on an almost weekly basis. These upgrades have been due to high customer demand and the success of LiveVoip as America's Number One Resellers - Resellers. The first to benefit will be customers located in USA & Canada who will have a lower latency, higher quality connection as LiveVoip constantly improves its Global VoIP network. LiveVoip is a resellers – reseller. The company provides subscribers with a quality solution to control, manage and reduce global telecommunications costs. LiveVoip intends to open a switching facility in Texas within 10 days completing its redundant Trans America VoIP backbone.

"Our VoIP network is based on LiveVoip’s Global IP technology, delivering highly reliable, integrated IP packetized voice service," said Pamela Patterson, acting President and CEO, LiveVoip LLC. "Within a month most subscribers will be able to terminate VoIP traffic on a regional basis without the need to send calls back to NY or LA locations for competition. Subscribers are able to maximize their return on investment and through lower latency, reduce connection times associated with sending voice traffic over a public switched network."

The upgrade to all LiveVoip switching facilities will also allow the company to expand faster. LiveVoip LLC will start to offer Bulk VoiP services to ISP’s, and other companies interested in delivering an ATA home or business enterprise solutions to their customers starting in June 2005. Many ISP's will be able to make the transition to VoIP with the help of LiveVoip.

"LiveVoip’s High-Speed VoIP NODES will reduce possible congestion and simplify VoIP traffic and thus improve quality," said CJ Benson, CTO. "A choice of monthly per minute or flat rate for calling plans provides customers with a greater selection than is available in the market today."

About LiveVoip LLC

LiveVoip, LLC is currently serving the Global market. North America accounts for 70 percent of the multi-billion-telephony market. It is estimated that 56 percent of businesses in North America are considering migrating to VoIP in the next two years. Livevoip is the world’s number one resellers – reseller to the Asterisk based PBX market. VoIP based systems are going to head to the Internet in a big way and LiveVoip intends to be ready to service any and all clients worldwide.

Launched in late 2003, LiveVoip, LLC is an Arizona based company privately funded Delaware company. http://www.livevoip.com.

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Related Asterisk PBX Press Releases

Pronexus Inc., a leader in computer telephony and speech software solutions since 1993, today announced that it has successfully completed an interoperability study between its flagship development environment, VBVoice, and Asterisk, the open source Private Branch Exchange (PBX).

VBVoice is the only telephony development tool inside Visual Studio .NET that blends a high level graphical user interface for call flow design and sophisticated programming. VBVoice delivers the scalability and reliability for a wide range of telephony applications such as speech-enabled IVRs, auto-attendants, call center applications and more.

Linux-based Asterisk™ is a complete IP-PBX in software. It provides the common PBX features, implements VoIP and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. With support for features such as Voicemail services with Directory, Call Conferencing, IVR, Call Queuing and more, Asterisk provides a compelling low-cost alternative to traditional telephony platforms.

VBVoice enables developers to rapidly create advanced telephony applications that seamlessly work with the Asterisk PBX. The tool provides access to sophisticated technologies such as speech recognition and text to speech, enabling the development of applications such as dial-by-name, speech auto-attendant, speech-enabled self-service and many more, bringing additional value to Asterisk, especially in the enterprise environment.

The Pronexus tool also enables the development of hardware-less VoIP applications using Intel’s Host Media Processing (HMP). This makes it a natural fit for Asterisk, which supports various VoIP protocols such as SIP and H.323 without the need for dedicated telephony hardware.

VBVoice facilitates the development of telephony and speech applications for the Asterisk platform with its easy-to-use visual call control and call flow that can be extended using Visual Basic, VB.NET or any other programming language supported in Visual Studio .NET.

“With this announcement, we are pleased to provide developers with an alternative to the intricacies of low level scripting for the Asterisk platform,” commented Gary Hannah, President & CEO of Pronexus. “The graphical user interface and full programmability of VBVoice makes Asterisk even more attractive for organizations looking to leverage IP telephony and powerful PBX features on this powerful low-cost platform.”

About Pronexus

Founded in 1993, Pronexus Inc. is a leader in Computer Telephony and Speech. It is the developer of the award-winning VBVoice™, and the new VBSALT™, Rapid Application Development tools for business-critical telephony and speech solutions. Pronexus also provides VeoSuite, a range of turnkey speech applications for Microsoft Speech Server and a complement of professional services that voice-enable a variety of industries and applications. Comprehensive support and acclaimed training complete the firm’s offerings.

More information:

Sonia Langenberg (Tercler)

Marketing Communications

Pronexus Inc.

Tel: (613) 271 8989 ext.541

sonia.langenberg@pronexus.com

http://www.pronexus.com

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LiveVoip LLC announced today its new Return-Please Object Tracking ID system and VoIP based call center operation. Set to roll out worldwide the first week in June 2005, the Return-Please product line allows users a way to recover Cell Phones, PDA’s, Laptops and more. Each Return-Please ID tag features a unique bar-coded ID that is recorded in multiple secure database systems. Each Return-Please ID tag has a tamper proof feature that deters any attempt at removal. Our unique barcode and toll free reporting number allow lost items to be recovered, then returned to their owners. The smaller devices become, the more they tend to get lost. Corporate and OEM versions of the Return-Please product line will be available at the same time.

”Return-Please products are designed to allow the return of lost or stolen items that many times the general public or police recover but, have no way to contact their rightful owner,” said Pamela Patterson, President, LiveVoip LLC. “Using the LiveVoip VoiP network and call center operation to support the product just makes good business sense. Customers, Police agencies and the general public can call to report finding a lost or stolen item. LiveVoip LLC will then arrange for the item to be returned to its owner at no charge. There is also a reward that is paid to the finder of any Return-Please item.”

“Return-Please is a step forward in ensuring that customers who lose valuable and many times small items are able to have a better than average chance of their return,” said CJ Benson, CTO. “Our VoIP network has been enhanced to handle an entire range of unique tracking-tagging systems for the protection of everything from Cell Phones, Laptops, iPODs, PDA's and more. LiveVoip has discovered new ways to make Return-Please simple and effective for customers to use.”

About LiveVoip LLC

LiveVoip, LLC is currently serving the Global market. North America accounts for 70 percent of the multi-billion-telephony market. It is estimated that 56 percent of businesses in North America are considering migrating to VoIP in the next two years. LiveVoip is the world’s number one resellers – reseller to the Asterisk based PBX market. VoIP based systems are going to head to the Internet in a big way and LiveVoip intends to be ready to service any and all clients worldwide.

Launched in late 2003, LiveVoip, LLC is an Arizona based privately funded Delaware Company. http://www.returnplease.com

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LiveVoip LLC management have decided that it is not in the best interest of the company or its clients to let itself be acquired by RV Wireless, Inc. (Pink Sheets: RVWS) LiveVoip LLC management feels the current multi-billion dollar VoIP market explosion is the “Next Wave” in the future of the Internet. LiveVoip must consider all offers and options presented in order to gain maximum value for its shareholders and customers. The company is retaining a major International banking firm to evaluate all current offers and options. “The Internet has found its Voice in our company," said Pamela Patterson President, LiveVoip LLC. “It is my number one goal to deliver that voice to as many customers as possible.”

About LiveVoip LLC

LiveVoip, LLC is currently serving the Global market. North America accounts for 70 percent of the multi-billion-telephony market. It is estimated that 56 percent of businesses in North America are considering migrating to VoIP in the next two years. LiveVoip is the world’s number One resellers – reseller to the Asterisk based PBX market. VoIP based systems are going to head to the Internet in a big way and LiveVoip intends to be ready to service any and all clients worldwide.

Launched in late 2003, LiveVoip, LLC is an Arizona based privately funded Delaware Company. http://www.livevoip.com

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More Asterisk PBX Press Releases

Satcom Direct, provider of satellite voice, fax, datalink and Internet communications solutions, has launched mobile application versions of five of their services, AeroV