AVST Adds New SIP Integrations to CallXpress

Applied Voice & Speech Technologies (AVST) announced it has added new SIP integrations to its flagship CallXpress unified communications platform. As part of its ongoing goal to support interoperability and offer the most integrations in the market, AVST's CallXpress will support SIP integration to the Nortel (News - Alert) Communication Server 1000 and Avaya (News - Alert) Communications Manager PBX (News - Alert) products. AVST plans plans to provide additional SIP integrations with leading communication products, including the Microsoft (News - Alert) Live Communications Server.

"Because SIP is a popular protocol for IP telephony, video, instant messaging, and presence engines, it is the ideal protocol to be used by unified communications solutions like CallXpress," said AVST's Vice President of Product Management, Tom Minifie. "For companies that are interested in migrating to VoIP, either now or in the future, CallXpress can be easily integrated via SIP to IP telephony systems, and at the same time, continue to support legacy PBX systems." In addition, AVST now provides a CallXpress Turnkey option for all of its servers, which enables customers to purchase the T2000, I4000, I6000 and the new V2000 models preloaded with CallXpress software. This option was developed to make deployment easy for system integrators and enable resellers to quickly fulfill orders by delivering the server with all the software pre-installed.

"This is just another example of AVST's commitment to providing best-of-breed technology built to ensure ease of use, simple installation and administration, and low cost of ownership," added Minifie. "Using our CallXpress Turnkey option, final onsite installation is reduced to site specific configuration tasks. System integrators save valuable time by avoiding hardware integration, operating system configuration, and basic software installation tasks."

www.avst.com

Telrex Extends SIP IP PBX Support

Telrex (News - Alert), developer of VoIP call recording and monitoring software for SMBs using IP PBXs or hosted PBX services, announced support for IP PBXs from Pingtel, Fonality (News - Alert), TalkSwitch (News - Alert), and Switchvox. Pingtel, Fonality, TalkSwitch, and Switchvox are all innovative providers of SIP-based IP PBXs. Support for these IP PBXs strengthens CallRex's leadership position in the VoIP call recording market for small and medium businesses.

With hundreds of deployments on SIP-based IP PBXs and softswitches from multiple manufacturers, CallRex supports more SIP-based IP PBXs than any other call recording solution. In addition to Pingtel, Fonality, TalkSwitch, and Switchvox, CallRex supports SIP-based IP PBXs from Cisco, Asterisk, Avaya, Nortel, Mitel (News - Alert), ShoreTel (News - Alert), 3Com, NEC (News - Alert), Siemens (News - Alert), Mitel, Zultys, and Vertical, and SIP-based softswitches from BroadSoft (News - Alert), Sylantro, and Tekelec (News - Alert).

SIP is growing rapidly as an IP PBX standard and makes it easy for IP PBX resellers to quickly sell and deploy VoIP in businesses of all sizes, enabling small and medium businesses in particular to access a wide variety of previously unaffordable applications such as call recording.

"Telrex is pleased to bring affordable call recording to those businesses that have invested in SIP IP PBXs, including those from Pingtel, Fonality, TalkSwitch, and Switchvox," said Robert Kapela, president of Telrex. "For many of these businesses, it is the promise of applications like call recording that makes standards-based IP PBXs so attractive."

www.telrex.com

TelePlus Group to Deploy SIP Platform
By Cindy Waxer

Thanks to a two-year strategic partnership with Digitrad France, TelePlus Group is gearing up to offer VoIP services using a turnkey SIP network - the PsipTN (Public SIP Telephony Network). The deployment of PsipTN, set to go live later this year, will allow TelePlus to offer customers a telecommunications and billing system, using the Vocalyz product for on-the-road travel and at home using the VoIP enabled 1TC product.

"By allowing customers to continue making and receiving long distance calls and to access the TelePlus proprietary 'inlanguage' interpreter/concierge services beyond the travel experience, we are not only creating an efficient form of seamless communications, but also a continuous revenue stream by unifying access components (GSM/VoIP/WiFi (News - Alert)) and consolidated billing applications," said Jim Gibson, vice president of business development for TelePlus Group.

PsipTN is developed by TelTel (News - Alert), a developer of VoIP SVNO platforms. Together with PsipTN comes a full-featured VoIP softphone client, which allows the customer to seamlessly continue using TelePlus communication services well beyond the travel experience.

The process is universal and allows users to make low cost calls, continue receiving calls from their local number, manage personal buddy lists and access the TelePlus international services and more from anywhere around the world. PsipTN is integrated with the existing Digitrad GSM billing system, Stand4U, which is a Web-based IVR platform that allows for the efficient immediacy and accuracy of account provisioning, data access and billing transfers.

www.teleplusgroup.com
www.digitrad.com
www.teltel.com

JPS Rolls Out Analog Radio Adapter for SIP Networks
By Laura Stotler

Raytheon JPS Communications has announced a new Analog Radio Adapter, the ARA-1. The new solution is comparable to an analog telephone adapter and enables a standard radio to operate on a SIP network.

The ARA-1 extends the coverage and capabilities of existing SIP PBXs by enabling the interface of land mobile radios to the system. Radios are connected through the ARA-1 and are assigned unique extensions, so they can easily be dialed via any IP phone, softphone, or voice device associated with the SIP PBX. A number of services are also available, including call logging, call recording, and call forwarding. The SIP PBX can also enable video conferencing, document sharing, and text messaging between compatible devices.

The new solution brings radio networks into the SIP arena and also brings SIP-based communications to areas not serviced or reachable using a SIP network. For instance, an ARA-1 could be used to extend SIP communications into tunnels, across bodies of water or through rugged terrain.

"The ARA-1 is a perfect marriage between land mobile radios and IP-based networks," said Mike Cox (News - Alert), vice president of engineering for JPS Communications. "It combines a supremely capable radio interface to the standards-based open SIP protocol that is rapidly becoming the acknowledged pathway to the convergence of voice, data, and video."

www.jps.com

Verizon Business Unveils Advanced IP-Based Services

Verizon Business (News - Alert) introduced two new Internet protocol-based capabilities for its Contact Center Services and VoIP portfolio to help businesses enhance customer-service operations and leverage the benefits of VoIP.

The new offerings are: IP Tollfree Service; IP IVR, an interactive voice-response system for contact center services; and new IP Trunking options, all featuring interoperability with Avaya enterprise communications software.

"We continue to advance and extend our VoIP offerings to meet customer needs where it matters most - at the heart of their business operations," said Tom Roche, vice president for network voice and data services for Verizon Business. "IP Tollfree and IP IVR will help redefine how companies implement contact centers in the future. As businesses increasingly make the move to IP, Verizon Business continues to outpace the industry, delivering one of the most complete suites of VoIP services available today."

Verizon IP Tollfree routes incoming toll-free calls over IP to enable greater efficiency and support multiple-contact media, such as phone calls, e-mail, or IM from around the globe. The service enables contact center agents to transfer calls using capabilities inherent to SIP, which enables real-time communication on the Internet. Since IP Tollfree is a networkbased service, companies benefit from lower total cost of ownership because they do not have to own and operate costly gateway equipment.

Verizon IP IVR provides call processing in a pure IP environment over a carrier-grade, global network infrastructure, enabling customers to benefit from network efficiencies such as voice compression and dynamic bandwidth allocation. The service offers administrators an extensive selection of call-routing and processing features and terminates incoming calls to both TDM and IP endpoints, allowing customers to adopt IP technology at their own pace.

Verizon Business has designed and certified IP Tollfree and IP IVR to be compatible with Avaya Communication Manager with SIP enablement services (SES) 3.1 software and other leading SIP-enabled endpoints.

"Our research shows that there is a huge pent up demand for SIP-based call delivery, and Verizon Business is meeting that demand," said Robin Goad, lead analyst, contact centers, Datamonitor. "Businesses can improve customer service by working with suppliers that offer robust but simple-to-use-service offerings and whose portfolios reflect a clear IP strategy."

www.verizonbusiness.com

BandTel Partners with Quintum
By Cindy Waxer

BandTel and Quintum Technologies (News - Alert) have formed a relationship that will allow customers to acquire a Tenor VoIP switch or gateway with the BandTel Services. BandTel is a provider of next-generation VoIP termination to the PSTN for high-volume telecom users such as call centers, enterprise users, teleconferencing companies, and IVR users. The company's VoIP network solutions are now fully interoperable with Tenor VoIP MultiPath switches and gateways.

Enterprise users will be able to access the combination of BandTel's VoIP network to deliver call center and enterprise users a cost-effective, faulttolerant voice architecture and Quintum's Tenor VoIP MultiPath solutions.

According to Chris Dunk, president and CEO of BandTel, "Quintum's Tenor VoIP MultiPath solutions are the perfect complement to BandTel's network because both integrate so easily into an existing network, even if legacy devices are present and both guarantee dependable switching and routing. In addition, by mutually supporting the SIP standard, we are working together to advance the future of VoIP technology."

Because Quintum's Tenor switches support industry standards like SIP, they are wholly interoperable with BandTel's SIP Softswitch technology. Customers deploying Quintum''s MultiPath VoIP systems with BandTel's VoIP services will have the unique ability to connect call systems via a trunk-to-trunk transfer and cost-effectively terminate calls to and from any location in the world. This allows users to instantly unite their global businesses with multiple users in various locations.

In addition to Quintum, BandTel has partnered with several other names as well, including Sphere, Digium, AudioCodes (News - Alert), and VegaStream.

www.bandtel.com
www.quintum.com

Sipera IPCS 310 Supports More VoIP Environments
By Johanne Torres

VoIP security technology provider Sipera Systems of Richardson, Texas, announced expansion of its IP communications security solutions to more environments. The company's IPCS 310 will now offer protection of VoIP and IM to more environments, allowing enterprises, especially in the financial services and healthcare industries, to secure legacy systems and additional IP communications applications against attacks, misuse, and service abuse.

The IPCS 310 system monitors VoIP traffic, detects anomalies in call traffic patterns, and identifies threats to protect end-user devices and network infrastructures for enterprises with up to 1000 users.

The new upgrades features include multiple protocol support to protect VoIP assets in mixed protocol environments; border control to enable the deployment of SIP trunks and voice extranets, as well as extend VoIP infrastructure to mobile users; expanded IM protection for SPIM and IM spoofing to protect internal and external IM communications; IM logging functionality; and additional media protection to prevent threats that may be launched through the media channel after a VoIP connection has been established.

"Giving customers choices and flexibility is particularly important as new technology is integrated into existing enterprises," said Sipera's president and CEO, Seshu Madhavapeddy. "For enterprises of all sizes and configurations, comprehensive network-level and end-user security is a prerequisite to deploying today's growing suite of collaborative VoIP, IM, and other IP communications applications."

Madhavapeddy added: "With these new features, the Sipera IPCS 310 allows enterprises to efficiently take advantage of today's real-time, IP communication applications without compromising the network's security requirements."

www.sipera.com

CommuniGate Systems Rolls Out Upgrades for VoIP and SIP
By Laura Stotler

CommuniGate Systems has announced its "Trade In and Trade Up" program to enable carriers to upgrade to new value-added services including VoIP, SIP/XMPP secure IM and hosted PBX. The program will enable carriers to trade in legacy e-mail software, saving on support and maintenance services. They may then "trade up" to replace antiquated messaging systems and offer a full suite of IP communications solutions.

CommuniGate's initiative will enables service providers to smoothly migrate their communications services to an IMS-ready platform. It is targeted toward providers that have customers unable to access SIP-based IP communications networks because their email solutions use legacy technology.

The providers will be offered a full CommuniGate Pro license, replacing their existing ongoing support costs. A simple migration will then enable providers to offer a number of enhanced IP communications services. Carriers may choose which features of the CommuniGate Pro suite they wish to deploy. The modules offered include SIP Proxy, Session Border Controller (SBC), IP PBX, clustered voice through a SIP Farm, audio conferencing, XMPP support and voice mail.

"The feedback we are receiving is tremendous as the world is now adopting SIP based Communications. Mobility and productivity will rise and traditional closed and location locked access will die," said Jon Doyle, vice president of business development for CommuniGate Systems.

Doyle continued: "We will witness a fundamental change in the communications landscape over the next five years just as we saw in the early 90s with email becoming the communication standard medium for business," added Doyle. "Holding users to a location with a phone number, or charging them for roaming to other locations will soon be replaced with the mobility and portability of VoIP - where one address finds a person anywhere, anytime."

www.communigate.com

Pretty Good Security for SIP Communications
By Erik Linask

BorderWare Technologies and Pretty Good Privacy (PGP) founder Phil Zimmermann announced that BorderWare is poised to become the first commercial licensee of Zfone, a secure VoIP media encryption software solution also created by Zimmermann. By integrating integrates Zfone with BorderWare's SIPassure VoIP Security Gateway (News - Alert), this agreement brings a new level of security and ease of use to VoIP systems.

According to BorderWare, SIPassure is the industry's first VoIP security gateway that takes VoIP application security to a new level by combining the best features of an enterprise firewall, application layer gateway, and a Session Border Controller.

SIPassure is designed to secure all SIP-based applications, including VoIP services, video conferencing, and other messaging applications. SIPassure ensures that organizations using SIP-based communications remain harbored from impairment and service disruption from internal and external attacks, interference spam, and other threats.

The integration of Zfone media encryption with SIPassure ensures that BorderWare's customers have the security to ward off Spam threats, DoS attacks, eavesdropping, spying, and wiretapping,--all without limiting the performance and convenience customers have come to expect from their e-mail.

www.borderware.com

Patton's SmartNode VoIP CPE Certified by Thomson
By Erik Linask

Patton-Inalp Networks has announced that its SmartNode brand of VoIP routers has been certified by Thomson for interoperability with Thomson's Cirpack brand softswitches. The certification assures carrier customers of a smooth migration as they move from multiple legacy infrastructures to fully converged IP-based voice and data services.

Patton has actively sought softswitch interoperability for both its SmartNode and SmartLink families of VoIP and triple play platforms via its own Interoperability Program, which acknowledges third-party softswitches that passed tests proving they work with Patton's equipment - primarily in the area of live carrier deployments.

For use with Thompson's equipment, SmartNode gateways connect business and residential equipments to the nextgen VoIP features and services enabled by Cirpack using SIP. As such, SmartNode products are designed with the future of the communications industry in mind, capable of supporting new advanced products as they emerge. In addition to being SIP-compliant, SmartNode supports H.323, T.38 fax relay, fax bypass, modem bypass, Voice over VPN, AES/DES-IPSec voice encryption, and DownStream QoS.

"Patton brings a lot to the table," said Fabien Maisl, Marketing Director of Thomson's Network Intelligence Solutions Cirpack. "Their extensive portfolio is a great fit with our worldwide customer base. Carriers always prefer a single-source supplier, and SmartNode pretty well covers it. We are very pleased to welcome Patton as a certified technology partner.

www.patton.com
www.cirpack.com

Interactive Intelligence Releases Upgraded Unified Communications Software

Interactive Intelligence (News - Alert), a global developer of business communications software, has released an upgraded version of its unified communications software, Communite. Communite is a voice mail replacement system offering standards-based unified messaging, interactive voice response, and real-time communications services for all types of organizations, including large, distributed enterprises and higher education institutions.

The latest Communite release, version 2.4, was designed to further increase organizational productivity while minimizing infrastructure requirements with enhanced functionality for speech-enabled mobile applications, simplified creation of customized voice mail menus, flexible personalization options for presence management, and an alternate message store for unified messaging applications.

"These latest enhancements to Communite make it an even more compelling unified communications system for organizations with mobile employees, those looking to replace aging voice mail systems, and companies that must comply with increasingly stringent archive messaging requirements," said Yankee Group senior analyst, Ken Landoline. "With IP telephony driving faster adoption of unified communications solutions, Communite's built-in SIP support gives it an added competitive advantage."

New in Communite 2.4 is an enhanced Interaction Mobile Office(TM) application, which builds on its existing speech-enabled auto-attendant by adding a speech-enabled menu for message retrieval, status changes, and company directory access. A new telephone user interface enables organizations to use XML for easy emulation of existing voice mail menus, thus reducing user training requirements when replacing voice mail systems.

Other enhanced features such as new customizable status codes that can be applied to different messaging rules - for instance, "if out of office, forward calls to mobile phone" - give organizations unsurpassed flexibility.

www.inin.com

Huawei  Deploys Ubiquity Software  at Telefonica for IP Conferencing

Ubiquity Software (News - Alert) has announced a joint win with Huawei Technologies (News - Alert), whereby Ubiquity and Huawei have deployed Ubiquity's IP conferencing solution at Telefónica Argentina, the first service provider to launch IP conferencing services using Speak from the Huawei/ Ubiquity relationship.

Ubiquity's Speak IP conferencing solution is a scalable, carrier-class, IP conferencing application that enables Conferencing Service Providers to offer hosted audio and Web conferencing services. Speak leverages SIP-based VoIP and Ubiquity's carrier-grade SIP Application Server (SIP A/S) architecture. This easy-to-use, browser-based solution offers a complete conferencing application feature set as well as a web portal for scheduling, initiating, managing and terminating multi-party conferences.

"With years of experience deploying next-generation network solutions in over 20 countries, Huawei understands carriers' needs, and is experiencing a strong demand for SIP-based applications," said Mr. James Yuan, Vice President, Strategy & Marketing for Application and Software Product Line for Huawei Technologies. "Ubiquity's Speak IP conferencing solution allows us to market a world-class solution to address our customers' needs."

www.ubiquity.com
www.huawei.com