Enterprise Telephony News
3Com and IBM to Deliver Integrated Collaboration Solution for System i
 3Com Corporation and IBM announce the availability of the System i Integrated Collaboration solution, adding a software developer kit, contact center, presence and unified messaging solutions for the System i IP Telephony solution first introduced in November 2006. System i is a complete IP telephony and business computing solution that combines 3Com's VCX IP Telephony software on the IBM System i platform (VCX was previously available on the IBM X Series server). The new solution targets mid-market businesses (100 to 2,000 users) that want a single system for all business applications - telephony applications and business computing. System i customers can add the 3Com telephony solution to their existing data system, saving money by eliminating the need for additional servers to get IP telephony functionality and benefits.
The IBM System i with built-in database software, storage, Internet connectivity and security can run several operating systems simultaneously - i5/OS, Windows, Linux and AIX 5L. Now, System i customers can also run 3Com's open-standards IP Telephony solution. 3Com software available for IBM System i since November 2006 includes IP Telephony, IP Messaging, IP Conferencing and Presence, as well as 3Com hardware: 3101, 3102, 3103 and 3105 IP telephones, the Convergence Client softphone and various analog and digital gateways. With the newly-announced System i Integrated Collaboration solution due out in second quarter 2007, customers will be able to take advantage of four new capabilities: (1) 3Com Software Developer Kit (SDK) and new Application Programming Interfaces (APIs) for adding third party applications, (2) Contact Center with intelligent call routing, screen pops and more, (3) Lotus Sametime 7.5 for click-to-call, click-to-conference and presence information, and (4) Unified Messaging for managing voicemail, fax and e-mail in a single inbox (Lotus Notes and Microsoft Outlook e-mail clients).
System i IP Telephony is currently being deployed by a number of businesses. 3Com's VCX (VCX V7000™ Voice Core eXchange) SIP-based platform for Linux servers targets medium to large enterprises that want an end-to-end SIP solution and total system survivability, including customers as small as 100 users and as large as 5,000. Since the user profile for System i is 100 to 2,000 users, VCX software is a perfect fit for the target audience, according to 3Com.
System i IP Telephony is available in pre-configured “Express” bundles of 100, 250, 500 and 1,000 users, including the software elements for IP telephony and IP Messaging along with the System i server. The appropriate hardware (phones and gateways) is added according to capacity needs. Express Editions come in a single configuration (starts at $37,900) or double configuration (starts at $51,900) – the double configuration for “high availability” includes a primary system and a backup system. Larger customers will need a customized quote from an IBM business partner. System i IP Telephony is only sold through authorized IBM System i Business Partners. Domino and Sametime integration starts at $500 per server for each application. 3Com will disclose pricing for the Contact Center offering at a later date.
Cisco Introduces Smart Business Communications System for Small Businesses
Cisco announces the Cisco Smart Business Communications System for small business aimed at simplifying set up and configuration as requested by Cisco’s customers and partners. The new all-in-one solution is “simple, complete and highly secure” and provides telephony, messaging, and mobility through a single, compact and easily-administered device. Functionality includes IP voice, data, voice messaging and auto attendant, video support, integrated VPN, Wireless LAN access and embedded security. Optional desktop applications enable collaboration such as instant messaging, single number reach and desktop control of communications.
The Cisco Unified Communications 500 Series platform (pictured) combines Unified Communications Manager Express (formerly CallManager Express) and Unity Express (voice and desktop messaging) and Cisco IP Phones and comes preloaded with all software and licenses. Cisco Configuration Assistant (free) is a Graphical User Interface that simplifies system administration and management.
The standard unit (rack- or wall-mount) houses eight Power over Ethernet ports for IP phones, four FXO or two BRI ports for connection to the PSTN, four FXS ports for analog devices, one Ethernet LAN port, an expansion port for an additional Ethernet switch (Cisco recommends the Catalyst Express 520 switch with embedded security which is preconfigured to operate with this new solution), an optional Wireless LAN Access Point, an audio jack for music on hold, an auxiliary port for emergency access and an expansion port for additional PSTN or analog interfaces.
Third party partner applications are being developed for Smart Business Communications, including IPcelerate’s IPsmartsuite productivity applications focused on healthcare, legal, retail and manufacturing environments.
Options are priced as follows: Cisco Catalyst Express 520 ($1,395), Cisco 521 Wireless Express Access point ($499) and Cisco 526 Wireless Express Mobility Controller ($1,799) and Cisco Monitor Director and Monitor Manager startup package ($2,900).
In other news, Cisco announces Select Certification, an entry-level certification for channel partners who focus on SMBs.
Digium Delivers Asterisk Appliance Telephony Solution
 Digium, Inc. announces the forthcoming Asterisk Appliance, a telephony solution for small businesses and branch offices with up to 50 users. The Asterisk Appliance is Digium’s “official entry” into the Customer Premises Equipment (CPE) market and was available last fall as a “developer kit” with the components needed to build a turn-key telephony platform. In early May 2007, the company will offer the new Asterisk Appliance, a full telephony solution that is easy to configure and which includes Asterisk Business Edition software, Digium hardware, the Digium developed Asterisk GUI and documentation. With a range of features and simplified management, the new solution makes “open source a meaningful option to small businesses,” says Mark Spencer, Founder and CTO of Digium.
The Asterisk Appliance includes a complete Asterisk server, support for VoIP and analog phones, full featured PBX functionality with interactive voice response (IVR), voicemail, conferencing and automatic call distribution (ACD) with hold time announcements, music on hold and ring strategies. The unit houses eight analog ports (FXO and FXS), a built-in router, five Ethernet ports (one WAN and four LAN), a compact flash card or MMC for voicemail and wireless and a craft port for debugging.
Pricing starts at $995 for the Appliance with SIP/IAX trunking (all IP) or approximately $1,600 with analog ports. Volume pricing will also be offered.
The company has traditionally offered two end user options: the Asterisk Business Edition (a commercially licensed version of Asterisk and enterprise business class offering based on the Open Source Edition, but fully tested and supported by Digium) and the Asterisk Open Source PBX software, as well as an OEM license that allows other telephony manufacturers to tailor the Asterisk telecommunications Open Source software and develop their own PBX solution (e.g. Switchvox SMB and SOHO).
In other news, Digium reports strong global demand for its Asterisk open source telephony solution, now deployed in 140 countries across North America, South America, Europe, Asia and Africa. Digium also announces the general availability in May 2007 of AsteriskNOW, the company’s first software appliance. AsteriskNOW is the open source distribution of Asterisk with a Digium-designed graphical user interface (GUI) and new setup wizard for easy installation and configuration in less than 30 minutes. The software will be a free download from www.asterisknow.org.
ESI Introduces New Server Family
Estech Systems Incorporated (ESI) announces a new family of communications servers, expanding the range of platforms and capabilities available for customers from small, single-site businesses to larger, multi-site enterprises. The ESI Communication Servers offer varying levels of capacity, as well as flexibility in the mix of digital and IP technologies to best meet a customer’s specific business needs.
 Four server platforms are now available: the compact ESI-100 Communications Server (up to 84 digital or IP stations), the ESI-200 which targets small or mid-sized business (190 digital or IP stations), the ESI-600 (pictured) introduced in 2006 for larger offices or multi-site customers (408 digital or IP stations) and ESI’s largest platform to date, the ESI-1000 with data redundancy to satisfy larger enterprise needs (816 digital or IP stations and currently in beta testing). All ESI Communications Servers include built-in voice mail, auto attendant and ACD with options for advanced applications such as unified messaging and VoIP (requires port card and licenses). A Mirrored Memory Module is an option for ESI-200 (24-port/600-hour voice mail) and ESI-600, but comes standard with the ESI-1000 server. (The ESI-200 with 16-port/140-hour voice mail uses solid state memory (Compact Flash) and does not need the mirrored drive.)
ESI has reduced the number and types of VoIP cards available and re-bundled the IP licenses to make it more cost-effective to purchase the amount the customer requires. Additionally, on the ESI-200, ESI-600, and ESI-1000, the new port cards do not require a separate card carrier, which also eliminates additional cost. Three new “Intelligent” VoIP cards enable up to 24 local or remote IP phones, or 24 Esi-Link channels or a combination of 12 IP phones and 12 Esi-Link channels on one card. Esi-Link technology uses the WAN or the Internet to network up to 100 ESI phone systems into one IP-based system. A new programming tool, ESI System Programmer, uses object-oriented technology and industry protocols (XML as a standard interface and Microsoft .Net framework) to provide installers and administrators with common Windows features, backup and restore functionality and applications for programming and debugging.
 ESI also announces additional flexibility for its Visually Integrated Phone (VIP) software suite, adding customer-recommended enhancements such as an improved GUI and simplified installation. VIP combines ESI phone system functionality and Microsoft Outlook for call control, voice mail and contact management via a PC. Two versions are available: basic VIP and VIP Professional for managing telephone calls and voice mail, organizing contacts, call logging, telephone set programming (ESI Feature Phone). The VIP Professional version adds secure instant messaging, call recording and compatibility with other ESI solutions such as VIP PC Attendant Console, VIP ACD and VIP Softphone.
ESI-100, ESI-200 and ESI-600 Communications Servers are currently available in North America through ESI Certified Resellers; ESI-1000 is undergoing beta testing. ESI will continue to offer its earlier line: IVX C-Class, S-Class, E-Class and X-Class systems.
Iwatsu Releases Enterprise-CS 3.0
Iwatsu Voice Networks announces new software for its Enterprise-CS communications system, a converged TDM/IP communication system for small and medium businesses that scales to 1,024 IP ports in a single gateway cabinet. Version 3.0 software improves the performance of the Platinum Series IP and digital telephones, adds new soft key features, enables new features for SIP stations and supports SIP integration to the Enterprise TOL messaging solution.
SIP integration to Esnatech’s Telephony Office-LinX 7.0 unified communications platform reduces hardware costs, while adding high availability and resiliency and flexible remote site deployments. Expensive voice cards and PBX interface cards are not needed, and with SIP, the Telephony Office-LinX solution can be deployed anywhere on the local or wide area network and communicate with both the voice and data network, saving costs in multiple servers and interface hardware. In addition, TOL 7.0 adds a number of improvements - mobile workers will benefit from the ability to manage their schedule through Microsoft Outlook, to filter calls (by Caller ID or contact information in a Microsoft Outlook address book) and to use speech access to corporate directories and contact information. Mobile LINK software enables presence management from a PDA, while the new mobile gateway lets users login from a WAP-compliant device to manage messages and settings or send text messages.
Iwatsu’s Enterprise Platinum Series terminals, available in IP or digital versions, include the 18-key and 12-key models, optional 100-button DSS unit and the Platinum series Softphone. New soft key features for these phones let users set call forwarding from another extension, use a single key to turn on/off call forwarding, access speed dial from the display, redial and more. SIP telephones are also compatible, including the Polycom SoundPoint 601 which has been tested and approved. New SIP station features add hold, call hold pickup, unattended/attended call transfer, message waiting lamp and hook flash to access features using SLT feature access codes.
The modular Enterprise CS supports multiple voice, data and video protocols and advanced applications such as unified communication, text-to-speech, voice-activated auto attendant, SNMP for network administration and in-building wireless. QuadFusion technology embedded in the system consolidates four protocols - Time Division Multiplexed (TDM), Voice over Internet Protocol (VoIP), Session Initiation Protocol (SIP) and H.323 data traffic - running these four protocols simultaneously on one platform without the need for external servers.
Iwatsu’s Enterprise-CS has the same 500+ telephony features as Iwatsu's traditional TDM-based ADIX telephone system. All earlier ADIX phones and all ADIX features are supported so that the end-user has minimal obsolescence when migrating from an ADIX to the Enterprise-CS. Further, all ADIX systems installed since 1989 can be upgraded to Enterprise-CS with no loss in hardware or application support.
Panasonic Announces KX-TDA Version 3.2
Panasonic Communication Systems announces new functionality for the KX-TDA Hybrid IP PBX system, adding new hardware, new features and enhancing the existing features available for the KX-TDA100/200 systems. Version 3.2 brings the system “up to par” with the larger KX-TDA600 Hybrid IP PBX introduced by Panasonic in 2006 (this software is planned for the smaller KX-TDA50 in May 2007).
 A new 16-port single line extension card transmits Caller ID information to all 16 ports and includes four power failure circuits, a benefit to hotel environments that often use single line phones in guest rooms. A new Automatic Call Waiting feature, assigned per extension, adds the option to automatically send a call waiting tone for specific call types such as CO line calls, door phone calls or intercom calls. If the option is programmed, the caller will hear a ring back tone rather than a busy tone.
Also new are Primary and Secondary Directory Numbers (PDN/SDN) that benefit a “boss-secretary” environment in which a call arriving to the boss’s extension will also ring and flash at the secretary’s extension, along with display of Caller ID information. Call handling is flexible – the secretary can answer, hold, transfer or make calls using the SDN button (the system supports several SDN buttons per extension (one secretary and several bosses) and up to eight PDN buttons per extension). With Version 3.2, Panasonic also enhances several existing features, including Caller ID Voice Mail Transfer, Call Log to Busy Extension, 5-digit Extension Numbers (up from four digits), SMDR Memory (expanded from 200 to 4,000 calls), Incoming Call Log for an ICD Group, Toll Restriction improvements and more.
Version 3.2 software is a free software upgrade that can be downloaded from the Panasonic dealer website.
| Telephones, Gateways, Messaging |
Aastra Telecom Launches DECT Mobility in North America
Aastra Telecom announces the launch of a SIP-based DECT (Digital Enhanced Cordless Telecommunications) wireless solution in the North American market. Targeting small and mid-sized businesses, enterprises with multiple locations or large campus environments, the Aastra SIP-DECT solution keeps mobile employees in touch within a building or campus with full roaming capabilities, seamless hand-over and high quality voice communication.
 The Aastra SIP-DECT solution includes DECT Handsets (DECT 142 Handset pictured) and IP DECT Access Points (up to 256) and operates in the 1920-1930 MHz frequency range released from the FCC in the North American market in 2005. DECT provides a higher quality of voice transmission and better area coverage compared to the earlier 900 MHz and the widespread 802.11 standard which is intended for data transmission. OpenMobility Manager software, to administer handsets, roaming and handover configurations, is installed on the IP DECT Access Point, eliminating the need for an additional server. Future releases will allow the software to run on any Linux-based server which can be a PBX or standalone server.
The Aastra SIP-DECT solution is expected to become available on June 1, 2007 through a “select group” of Aastra authorized distributors in North America. List price (handsets and access points) is $500-$800 U.S. per user, depending on configuration. Initially, one DECT handset model (DECT 142) will be available.
Aastra's DECT technology (variants of the access points and handset) has been deployed in Europe for at least two years, according to the company. The new SIP-DECT solution is based on proven DECT technology and Aastra VoIP and SIP capabilities. The SIP-DECT handset joins a family of Aastra IP Telephones, including the latest 5i Series. Aastra SIP phones are standards-based and interoperate with VoIP platforms (hosted or premise-based) from a number of vendors, including Asterisk/Digium, Fonality, Epygi, Vertical Communications, Nortel, Sphere and Teltronics and hosted systems from Sylantro, MetaSwitch and BroadSoft.
Active Voice Announces New Releases for Repartee and Kinesis Unified Messaging
Active Voice, LLC announces a new version of Repartee for Windows, a unified messaging solution for a broad set of customers in the small to medium-sized business market that need a voice mail or an integrated messaging solution. Repartee for Windows enables workers to manage voice, fax, and e-mail messages from a PC, mobile device or over the Internet. The platform scales from four to 96 ports and supports up to 65,000 mailboxes. Optional software packages can be activated via an upgrade code purchased from Active Voice. These include ActiveFax, multilingual prompt sets, and the Visual Messaging suite. Repartee for Windows integrates with over 200 PBX, Centrex and hybrid key systems from leading vendors including Avaya, Mitel, NEC and Nortel, as well as IP systems from Mitel.
Repartee for Windows Version 2.5 adds new features for the hospitality industry and improves migration from earlier Repartee versions. The Hospitality Package is enhanced with several new features, including All Hotel Guests Message Group administration that allows hotel staff to record and send broadcast messages, Custom Hotel Guest Distribution Lists (closed messaging group), scheduling of wake-up call report printouts and easy access to the Follow Me Here option (now on the main menu). Version 2.5 also adds a data migration utility that streamlines upgrades from earlier Repartee OS/2-based systems by backing up the original database, detecting database corruption and attempting to repair the database prior to the upgrade. The tool will report errors on-screen, save error information to a text file and abort the upgrade if a problem occurs.
 For enterprise sites that have globally dispersed workforces or need a scalable, redundant and clustered system capability, Active Voice offers Kinesis, the company’s Next Generation Unified Messaging platform. Kinesis is a Windows-based messaging server that can serve as an affordable voice mail only solution or a full unified messaging solution with which subscribers manage voice, fax and e-mail messages from a Microsoft Outlook inbox; all message types can be managed from a PC, telephone or the Internet. Users can also access calendars and listen to e-mail over a phone using optional text-to-speech technology (English) or take advantage of optional Speech Recognition (English) functionality. Kinesis integrates with a variety of telephone systems, including traditional (150+) and IP or combined IP/PBX environments such as those from Avaya, Cisco, Mitel, NEC, Nortel and Panasonic.
Clustering of five Kinesis servers increases capacity to 37,500 mailboxes (7,500 per server) and improves reliability since the servers share a SQL database and Exchange message store. Servers can be co-located or geographically dispersed. Version 2.9 adds a SQL Server-based “Active Voice (AV) Message Store,” a new, cost-effective message store option for voice mail only configurations which eliminates the cost of the Exchange server and Outlook clients. Kinesis (with AV Message Store) supports up to 16,000 mailbox users on a single server with no clustering of systems.
Esnatech and Iwatsu Deliver Enterprise Communication Suite
Esna Technologies Inc. (Esnatech) and Iwatsu Voice Networks announce the new Enterprise Communication Suite for small and medium size businesses, packaging Esna’s Telephony Office-LinX unified communications solution with Iwatsu’s Enterprise-CS IP telephony platform. The Enterprise Communication Suite is a new configuration that packages the Telephony OfficeLinX mobility, messaging and presence services as standard capabilities for the switch solution. And, Esnatech has made several customizations that extend beyond standard SIP signaling, enabling complete presence status and native support for fax and message light support on both digital and IP telephone sets, according to Esnatech.
SIP integration between Telephony Office-LinX and the Enterprise-CS IP PBX reduces hardware costs, while adding high availability and resiliency and flexible remote site deployments. Expensive voice cards and PBX interface cards are not needed, and with SIP, the Telephony Office-LinX solution can be deployed anywhere on the local or wide area network and communicate with both the voice and data network, saving costs in multiple servers and interface hardware.
In addition, TOL 7.0 adds a number of improvements - mobile workers will benefit from the ability to manage their schedule through Microsoft Outlook, to filter calls (by Caller ID or contact information in a Microsoft Outlook address book) and to use speech access to corporate directories and contact information. Mobile LINK software enables presence management from a PDA, while the new mobile gateway lets users login from a WAP-compliant device to manage messages and settings or send text messages.
Iwatsu Enterprise-CS currently supports two Esnatech Telephony OfficeLinX platforms: Telephony OfficeLinX Small Business Edition and Telephony OfficeLinX ELITE for larger configurations.
Polycom Adds New VoIP Desktop Phones
Polycom, Inc. introduces three new VoIP desktop phones, adding two new entry-level models and a high-performance phone with Polycom’s High Definition (HD) Voice technology that incorporates wideband audio, enhanced signal processing, Acoustic Clarity Technology and a specialized system design. The new SoundPoint phones expand the company’s standards-based telephone family to a full range of telephone models for the employee, the attendant or the conference room. The new phones join Polycom’s current family of SIP-based phones: the SoundPoint IP 650 with HD Voice, SoundPoint IP 601, SoundPoint IP 501, SoundPoint IP 430, SoundPoint IP 301 and Sound Station IP 4000 conference phone.
 Available in April worldwide (China, Korea, Brazil in third quarter 2007), the new SoundPoint IP 330 and 320 cost-effective, entry-level models (US$179 and US$139, respectively) feature a full duplex speakerphone, graphical LCD and Power over Ethernet support. SoundPoint IP 330 includes a dual-port 10/100 Ethernet switch for LAN and PC connection (beneficial to contact center agents), while the SoundPoint IP 320 has a single Ethernet port (suitable for common areas).
Pictured is SoundPoint IP 550 (US$369) with “cutting edge SIP feature set,” an enterprise-class Ethernet IP phone that features high-fidelity voice quality and enhanced clarity via Polycom HD Voice technology. Beneficial to users who require advanced features (bridged line appearance, busy lamp field and presence), the Polycom SoundPoint IP 550 includes a micro-browser that lets users access XHTML-based applications such as stock quotes or weather forecasts. All three new models have SIP 2.0 compatibility which enables support for Microsoft Live Communications Server and Microsoft Office Communicator instant messenger client.
The enhanced clarity of Polycom HD Voice is part of the company’s “UltimateHD” strategy which includes HD Voice, HD Video, HD Infrastructure, and HD Global Services.
All Polycom SIP-based desktop phones are designed to work with a variety of IP PBX platforms and softswitches. Recently, Polycom has added eight new certified members to its VoIP Interoperability Partner (VIP) Program, adding 3Com, Aptela, CommuniGate, Fonality, Inter-Tel, NEC Philips Unified Solutions, pbxnsip and TeleWare. The certified partners, now 19 members in total, offer interoperability with Polycom’s VoIP desktop and conference phones. Existing VIP members include Digium, Comverse/Netcentrex Converged IP Communications, Objectworld, Pingtel, Whaleback Systems, BroadSoft, Sphere, Sylantro, ADTRAN, Interactive Intelligence and Nortel.
Sphere and Quintum Certify Survivable Remote Office Solution
Sphere Communications and Quintum Technologies announce the certification and interoperability of Sphere’s Sphericall IP PBX and Quintum’s VoIP gateways to enable the transmission of analog and digital voice and fax traffic over a corporate intranet or IP network. Sphere’s IP PBX uses the Quintum Tenor VoIP gateway as a remote office solution to integrate legacy equipment such as fax machines or postal meters, for example, as well as analog and digital trunking and stations. Since survivability is a key issue for remote offices, the Quintum gateways provide local PSTN access and E911 emergency services, as well as continued voice service for SIP endpoints with an embedded SIP proxy agent in the remote-office switch. Essential calling functions can continue even if the connection to a central or hosted IP PBX is lost. This proxy agent provides sufficient routing intelligence to provide users with dial-tone and basic calling capabilities.
 Quintum VoIP technology addresses the important issues in a VoIP deployment such as PSTN connectivity, ease of management and survivability should an IP failure occur. Quintum’s SelectNet QoS technology ensures quality calls with backup routing to the PSTN, while PacketSaver technology improves bandwidth utilization. Other notable features include automatic call type detection (voice, modem or fax), public and private dial plan support, Caller ID delivery and MultiPath Call Routing for intelligently routing calls between the PBX, the PSTN, and the IP network. For management, Quintum offers the Tenor Configuration Manager (GUI) for configuration of remote individual Tenors, the Tenor Monitor, a VoIP network management tool that monitors alarms, call events and Call Detail Records and the Tenor Remote Management Session Server (even behind NAT firewalls).
Support for industry standard interfaces (H.323 and SIP) makes Tenor VoIP gateways compatible with most vendors' Key and PBX systems. The gateways have been certified by Avaya, Nortel, Inter-Tel, Vertical, Dialexia, Intel, 3Com, Sphere and others to assure VoIP interoperability and additional functionality, such as survivability. Tenor gateways are available in analog (AS, AF or AX) or digital (DX - pictured) versions that suit varying business sizes from SOHO and branch office environments to larger businesses and service providers’ smaller locations. For enterprises using ISDN Basic Rate Interface (BRI) lines, Tenor “BX” digital gateways are available with two, four or eight BRI S/T interfaces.
LG-Nortel Launches ipLDK60 in CALA
 LG-Nortel announces the availability of the ipLDK60 telephony system in Brazil in April 2007. Part of LG-Nortel’s global strategy is a focus on enterprise markets, and Brazil is an important market within CALA, according to the company. The ipLDK60 is also currently available in India and Turkey and will be available in EMEA in 2007 through Nortel’s sales channels.
Designed with a simplified, plug-and-play architecture, the ipLKD60 scales to 48 extensions (up to 16 IP stations (including WiFi handsets) and 48 analog or digital stations). The basic unit supports popular telephony features such as Short Message Service, Caller ID and conferencing (15 participants in nine virtual conference rooms), while offering options for voice mail, interactive voice response (IVR) and IP connectivity. Additional options are activated by a keycode, including TAPI, the EzPhone TAPI soft phone, a Windows-based VoIP soft phone called Phontage, IP networking (up to 72 nodes) and the ezAttendant PC attendant console.
The ipLDK60 joins a family of ipLDK telephony systems offered in various regions of the world and sold by Vertical Communications (formerly Vodavi) in North America.
In November 2005, LG Electronics and Nortel finalized the formation of the joint venture and new company called LG-Nortel Co. Ltd. that specializes in 'leading-edge communications and networking solutions in the wireline, optical, wireless and enterprise markets in South Korea and the rest of the world.' LG-Nortel is headquartered in South Korea.
NEC Philips Adds Support for Polycom SoundPoint IP Phones
NEC Philips announces SIP@Net software that supports a SIP extension interface to connect SIP devices (phones, video phones, Voice over WLAN phones, conference bridges and IP DECT phones) to the company’s’ SOPHO iS3000 Platform. Certified SIP Phones include Polycom SoundPoint IP 430 (pictured), IP 501 and IP 601 and Polycom’s SoundStation 4000 conferencing unit (new Polycom SIP telephone models will also be compatible). Now, all NEC Philips platforms (IPC, 2000 IPS, SV7000 and iS3000) support SIP and the Polycom SIP telephones and conference units. CounterPath’s eyebeam telephony client is also available.
In other news, NEC Philips announces a new version (Release 21) of the UNIVERGE SV7000 IP communications system for enterprises such as large international businesses and government organizations, adding advanced integration with Microsoft Office Communication Server. In addition, NEC Philips plans to introduce NEC’s UNIVERGE Assured Mobility in Europe by May 2007. UNIVERGE Assured Mobility is a new mobility solution that uses NEC’s wireless LAN architecture (Wireless Optimized Architecture) with WLAN controller(s) and access points to enable a voice over wireless LAN solution for any size organization – a small or mid-sized businesses or larger enterprise.
Also new from NEC Philips is the Business ConneCT CTI application for call control and group information, voicemail, directory services, operator and call routing functionality. This client-server application with .NET technology provides PC-based call handling, contact management and presence management for employees, call handling for operators, or enables contact center agents to view caller information and take advantage of routing, queuing and IVR functions. Business ConneCT is available on iS3000, 2000 IPS and SV7000 telephony platforms sold throughout the EMEA region. There is strong interest from other NEC divisions to adopt this application in other regions, but at present, Business ConneCT is not available outside EMEA, according to NEC Philips.
Swyx To Offer Fixed Mobile Convergence Solution
Swyx, a Germany-based IP telephony solution provider, announces the forthcoming availability of SwyxMobile, a Fixed Mobile Convergence (FMC) and Mobile Extension (Mobex) solution. Planned to be available in late second quarter 2007, SwyxMobile will enable employees to use a Nokia E-Series dual mode handset to access Swyx IP PBX applications and features within a wireless local area network (WLAN) and also when the worker leaves the campus and requires a cellular network. This hand-off between a local WLAN and a cellular network means that users need only one telephone number and one voice mailbox. Key features include one number access, user-defined call routing, call twinning (rings desk phone and mobile phone), ad-hoc conferencing, voicemail and fax delivery to the wireless handset.
In addition to the Nokia E-Series handset, SwyxMobile will run on other Symbian-based mobile phones, and Swyx also offers a variant for Windows-based Mobile Phones.
In other news, Sywx is offering its IP PBX Swyxware software as a hosted IP voice application available via Service Providers for a per-user, per-month license fee. Two solutions are offered, both with the same functionality: Hosted Sywxware Product for Service Providers with data centers, or Hosted Swyxware Server, an outsourced hosted facility provided by Swyx. In additional to the Swyx functionality and range of telephone equipment, Service Providers can offer their own additional services. While Swyxware has always been sold on a license basis, it has not previously been offered by Swyx as a hosted product or service (though some channel partners have developed and offered their own hosted version based on the standard CPE Swyx product.)
For questions, feedback and product information,
contact sgustavsen@accessintel.com. |
Polycom is “future-focused,” introducing a number of new solutions in recent months such as the Polycom RMX 2000 real-time media conferencing platform, an on-demand platform that enables ad-hoc, multi-point video conferencing in conference rooms, on desktops and via mobile devices. Polycom sees a transformation in the way businesses use video conferencing, noting a trend away from scheduled video conferences (that require technical personnel for assistance) toward unattended, on-the-fly conferencing, combining both audio and video participants. Also, new in conferencing is the Polycom RealPresence Experience High Definition (RPX HD), a series of telepresence solutions for an interactive “face-to-face” meeting experience. 
