• An MC-3 multichassis interconnect board. This board provides for the connection of 4,000+ time slots between either local or off-site locations. It also provides enhanced conferencing so developers can utilize a single board slot for both multichassis connectivity and conferencing.
• An MVIP 24-port, high-density, loop-start board. The board includes DTMF detection and generation, energy detection for each port, local tone resources for each port, AGC and gain control for each port, and all standard call progress tones. This board is also available as a 12-port board, and may be combined with the high-density station board for a combination of 12 loop start and 12 station ports all in a single card slot. A local power supply located on each analog line interface board provides the required negative voltage. Thus, port capacity is not limited, and developers can build their applications to the size needed.
• A series of high density "smart" analog port boards. The first in the series is a 24-port station board. The XDS High Density Station Port Board is designed for call center applications where a high volume of operators and telephone stations are necessary. The board includes DTMF detection and generation, energy detection and all standard call progress tones.
• An SCSA/MVIP bridge board. This board is a single chassis solution for combining MVIP and SCSA technologies. This board allows SCSA boards and MVIP boards to be connected in the same chassis without the need for special drivers or complex software. System clocks for this board may be provided from the MVIP bus, the SCbus, or from the bridge itself. Clock fall back is implemented according to MVIP or SCbus standards, depending on the clock source.
• A SCO UNIX MVIP driver with MVIP call functions. This driver is designed for use with Amtelco XDS (eXternally Defined Switching) boards. Other drivers include DOS, OS/2, UNIXWARE, and a new Windows NT driver. Also available is a QUNIX driver for MVIP from XDS developers.
• An industrial quality 20-slot rack mount chassis. Features include vertical board placement for easy access, five half-height bays for disk and other auxiliary equipment placement, lockable bays, enhanced multilevel cooling, and dual hot swap power supplies.

Active Voice's ViewMail
Active Voice announced ViewMail for Microsoft Exchange, the newest addition to Active Voice's TeLANophy family of applications. ViewMail for Exchange offers unified messaging by providing access to voice, fax, and e-mail messages from the Microsoft Exchange inbox. Users can browse messages in any order and process them with a few mouse clicks, handling them much more quickly than they could using the telephone. Users can also send broadcast voice mail and faxes, reply to voice mail and redirect voice or fax messages to other Exchange subscribers.

ViewMail for Exchange users can also check voice, fax, and e-mail from any touch-tone telephone, and listen to e-mail via text-to-speech conversion. They can reply to e-mail messages with voice mail, and they can redirect faxes and e-mail to any fax machine in the world. For more information, please contact Diane M. Williams at 206-441-4700.

Lucent Brings Speech Processing Solutions to New Markets
Lucent Technologies announced it will offer its speech processing capabilities to a new set of customers. Lucent Speech Processing Solutions, currently offered to telecommunications companies around the world, are now being introduced to customers who integrate and develop their own applications. Now systems integrators, value-added resellers of computer-telephony solutions, and network service providers such as telephone companies will have access to advanced Bell Labs technology to design network-based, speech-driven applications.

Richard DiPietro, Lucent's speech technology director, noted that "Lucent is the first to bring to the speech processing market a general-purpose, RISC-based technology. Our RISC-based design supports more applications at greater channel capacities than are possible with DSP-based solutions. And using general purpose processors allows us to create a real cost-savings for our customers."

The Lucent Speech Processing Solutions make speech processing more like human conversations with a function called key-word spotting, which allows callers to get responses to open-ended questions, rather than merely responding to system prompts. For example, a movie service could provide answers to a caller's question, "Where can I see a good comedy near Springfield Friday night?" by picking up on key words, "comedy... Springfield...Friday," and offering a list of choices. The technology eliminates nonessential words and pauses.

According to Lucent, the accuracy of this technology allows today's most advanced applications to perform well even in noisy backgrounds such as cars, airports, or train stations. It is designed to accommodate wide variations in accents, pitch variations in both male and female as well as older and younger voices, and supports large vocabularies.

Lucent Speech Processing Solutions include the Lucent Speech Processing Board, T-1 Interface Card, and Echo Canceller Board. The boards are designed to fit into a rack-mounted personal computer engineered for network-based applications and will be available to customers in the first quarter of 1997. For more information, call Lucent at 800-772-5785.

"For far too long, consumers have been charged far too much for international long-distance. AlphaNet Mondial will change international long-distance forever, and for the better," says Andre LeBel, AlphaNet's President and CEO. "Subscribers will have access to a fully digital international telephone service offering them substantial discounts. And they won't have to leave their regular domestic carrier."

AlphaNet says its service will differ from Voice-over-Internet (VOI) products that provide computer-to- computer communication aimed at the hobbyist. Instead, the Mondial voice service will be available to millions of global users of ordinary telephones, while fax service will operate via standard fax machines.

AlphaNet Mondial will take advantage of Internet Protocol (IP) standards for the transmission of voice and fax over data networks. AlphaNet anticipates that the low cost of these digital networks, combined with AlphaNet's proprietary compression technology, will allow the company to transport voice and fax calls at a considerable price advantage over traditional voice networks.

AlphaNet Mondial users make a phone call by dialing the telephone number of the person to whom they wish to speak, causing the target telephone to ring in the ordinary way. The person called will be unaware the call is being carried over an IP network. AlphaNet Mondial's digital technology offers exceptional call clarity, so calls sound virtually identical to those made over standard long-distance networks.

AlphaNet has signed Memoranda of Understanding with major partners in Australia, New Zealand, Singapore, Hong Kong, Taiwan, Korea, and other countries. Arrangements with partners to distribute AlphaNet Mondial in additional countries worldwide are under negotiation. Marketing in North America will be through a variety of channels, and conducted in concert with one or more marketing partners. AlphaNet will also be working closely with domestic telephone companies and long-distance providers in terminating their international traffic. For more information, contact Michael Reichmann at 416-923- 2222.

The new versions of Montana and Mariner include new cellular performances and connectivity features for use with all Motorola data-ready cellular phones, including popular 3-Watt phones and the new StarTac model, and are shipped with Netscape Navigator. Montana and Mariner have received MC2 Certification from Motorola's Cellular Subscriber Group. MC2 certification includes features such as auto-dialing, auto-answering, and a cable detection feature that automatically reconfigures the modem for cellular operation. For more information, contact Jessica Chipkin at 847-538-3415.

At the heart of the NetXchange approach are two key protocols. The first, the Simple Fax Transport Protocol (SFTP), is designed to establish a distributed communications infrastructure between all Internet Xchange servers on the network (or on interconnected networks) to share status, load levels, and least- cost routing information. Thus, each server routes traffic independently, eliminating a single point of failure. The system automatically performs load balancing and ensures reliability of the network, re- routing traffic around a failed server and backup.

The second protocol, the Real-Time Messaging Protocol (RTMP), is designed to guarantee reliability of transmission despite the volatility of Internet bandwidth. This is accomplished by establishing a peer-to- peer connection between the servers, then using signaling, buffering, and compression to maintain the connection, avoid bottlenecks, and avoid re-transmission.

NetXchange technology also addresses enhanced network management. The XchangeView module uses the industry-standard Simple Network Management Protocol (SNMP) running on HP's OpenView and is augmented with tracking, logging, and billing for each message transaction. The message event information is stored on an SQL database and can be exported to any third-party cost accounting or billing application.

NetXchange also announced Internet Xchange for Fax, its first application built around the switching architecture. This application is designed to reduce or eliminate fax transmission charges for corporations using their Intranets. It also enables service providers to offer cost-effective, value-added messaging services to customers. Forthcoming NetXchange messaging software offerings will encompass voice, audio, and video applications. For more information, contact NetXchange at 415-346-4131.

After initial development by Sun Microsystems and Lucent, the proposed Java Telephony API was shared with Intel, Nortel, and Novell, all of whom made contributions and devoted resources to refining the API. Sun has placed the Java Telephony API proposed specification on its JavaSoft Web site, and is submitting the API to the Enterprise Computer Telephony Forum (ECTF).

The Java Telephony API specifies the standard telephony API for computer-telephony applications under Java. It provides the definition for a reusable set of telephone call control objects which enable application portability across computer platforms. The API is designed to simplify application development for a wide range of CTI applications including customer service/management, Internet telephony, automated voice response, and directory services. The scaleability of the API will enable it to be implemented on devices ranging from hand-held phones to desktop computers to large servers.

The Java Telephony API is designed to allow enterprises to blend together Internet and telephony technology components within a single application environment as they design and deploy new business strategies for improving customer service levels, and for launching their Web presence -- creating new distribution channels and communications with their customers.

The Java Telephony API is a member of the Java Media API set being developed for Sun's Java platform. The telephony API is a standard extension to the Java platform licensed broadly across the computer and software industry. For more information, contact Ted DuPont (Burson-Marsteller for Sun) at 212-614- 4562.

The InLine series supports up to 4 ports of fax per card (with a maximum of 16 cards per chassis). The Daytona series supports up to 12 ports of fax per card. For the highest density, the Premiere series can be configured to handle up to 48 ports of send/receive fax on a single Premiere card.

Developers receive CAS 2.0 drivers and documentation in Release 4.5 of PIKA's software development toolkits for DOS and Windows NT. Firmware upgrades are available from PIKA at a cost of $150 per port. DSP module upgrades on the Daytona cards will be released in the near future. For more information, contact Rahul Virmani at 613-591-1555.

Both TA120 models feature two analog (POTS) ports for connecting analog telephones, fax and answering machines, and modems to the ISDN line. This feature provides conventional analog devices direct access to BRI making ISDN more affordable for work, home, and remote office applications.

Each model also provides one (DB25) data port for clear channel data rates up to 64 Kbps, or asynchronous data rates up to 230.4 Kbps. The TA120 automatically adapts to B channel protocol and utilizes dynamic bandwidth allocation with Multilink PPP. Async to sync PPP or Multilink MP is supported. The TA120 features remote on-line configuration to minimize field support service. For more information, contact Nancy Keen at 205-881-3106.

PhoNet Announces EtherPhone
PhoNet Communications announced its EtherPhone telephony system, which is designed to add sophisticated PBX capabilities to a standard LAN. In addition, PhoneNet has introduced Internet connectivity. With EtherPhone, PhoNet not only offers heightened telephony functionality, but also simplifies the organizational infrastructure.

EtherPhone integrates telephony and computer data networks by employing existing Ethernet LAN structures, existing Windows 95 PCs, and standard phone sets as building blocks for the system. According to PhoNet, the EtherPhone solution is cost-effective because it eliminates dedicated wiring, specialized PBX hardware, and custom telephone sets.

PhoNet's technology converts telephone signals to data packets, which are then transmitted over the data network in real-time. In addition to existing PCs and LAN infrastructure, specific EtherPhone components include a card in each desktop PC, which serves as an interface between the telephone instrument, the PC, and a telephony server. The server is a specially configured Windows 95 PC that provides the interface to the PSTN. PhoNet asserts that EtherPhone's PSTN interface works equally well with digital or analog trunks.

The server houses the software modules that support advanced telephony applications, such as voice mail, ACD, and billing information, and control both the external PSTN interfaces and the telephony system. Server hardware and software can be installed on an existing PC, or a pre-configured server may be purchased from PhoNet.

EtherPhone is intended as a versatile, low-cost solution from companies upgrading to a PBX system for the first time, perhaps from a smaller key system. The EtherPhone solution is designed to be inherently modular and easily extended as needs increase. It also eliminates the need for dedicated phone lines to each extension. EtherPhone users can implement a range of features, including dialing, call transfer, "follow-me," call pick-up, call waiting, and IVR. Third-party voice mail and billing systems are supported. PhoNet is providing an open API, so other developers can easily provide custom or specialized applications for the EtherPhone.

PhoNet also has an integrated Internet link capability for the EtherPhone. The link enables two EtherPhone-equipped sites to have an Internet-based voice link. Just as users now dial "9" to reach an "outside" line, they'll dial "8" for an Internet connection. The call will then be routed as though the user were calling from a station on the remote PBX. The same user interface - telephone set or GUI - is used in either case. For more information, contact PhoNet at 212-642-6563.

APC Enhances Matrix-UPS
In order to increase power management for data centers, American Power Conversion (APC) has enhanced its Matrix-UPS product series of high-performance uninterruptible power supplies. According to APC, system administrators will now be able to simplify their disaster recovery plans by managing their power the same way they manage the rest of the systems and data. The Matrix-UPS now includes APC's SmartSlot, an internal accessory option slot and a bundle of monitoring requirements.

APC's SmartSlot is designed to enable the user to customize and enhance the management, control, and performance of the Matrix-UPS unit with accessories such as the PowerNet SNMP Adapter card for SNMP management or the UPS Interface Expander card for multiple platform, unattended, graceful shutdown. The SmartSlot accessories are installed directly into the Matrix-UPS without the need for additional wiring or cables. PowerChute Plus, APC's power management software, is also available with the Matrix-UPS. For more information, contact APC at 800-877-4080.

RackMaster Ships New FRS Chassis
RackMaster Systems announced it has started shipping the first units of its newly designed FRS (Fault Resilient Series) Chassis. The FRS Chassis is an enclosure that can be configured as either a desktop, tower, or rackmount enclosure by adding the appropriate configuration kit.

The new FRS Chassis is available with 8, 14, or 20 expansion slot capability. The FRS Chassis features dual 300 W AC "hot swap" power supplies as a standard, with options available for 400 W AC or 250 W - 48 V DC input. The power supplies are removed and replaced through the front of the enclosure. Only one power supply may be removed at a time because of the lockout mechanism that is located behind a locking aluminum door on the faceplate. LEDs located on the front of the power supply shuttles give a visual status of each power supply. Power failure LEDs are also located on the control panel. The user can, at-a-glance, determine which power supply to swap out in the event of a failure.

Available as an enhancement to the FRS Chassis are additional monitoring and alarm capabilities of critical system functions. Enclosure temperature is sensed at up to three locations, and all power supply output voltages (+/- 5 V and +/-12 V) and fan rotations are continuously monitored. The status of all of these functions is displayed on an LCD that is located on the locking door. The display cycles through all of the functions and is updated in real-time with the status of each function. For more information, contact Charles Koch at 800-480-4384.

AnswerSoft Announces TAPI 2.0 Initiatives
AnswerSoft announced availability of its Sixth Sense and SoftPhone Agent applications running in a standard NT/TAPI 2.0 environment utilizing full LAN-to-PBX connectivity. According to AnswerSoft, call centers will be able to utilize full-scale client applications that directly communicate with NT servers operating against the TAPI 2.0 standard. Northern Telecom's Meridian 1 system will be the first switch supported in this environment.

Tom Hite, AnswerSoft's vice president, Software Development, said, "Initial releases of TAPI supported a direct connection between the PC and the desktop telephone. While TAPI 2.0 provides most of the direct PBX client/server capabilities required, a connectivity component needed between the client application and the server will be unavailable for several months. In order to meet the immediate needs of our customers, AnswerSoft has developed the client software component -- a TAPI 2.0 re-director -- needed to provide NT users access to client applications in this environment.

The company will immediately begin bundling its TAPI 2.0 re-director with its flagship application Sixth Sense and SoftPhone Agent for shipment. SixthSense automates agent workflow throughout the life span of a call. As a call arrives on an agent's desktop, Sixth Sense uses rules-based intelligence to automatically launch and navigate desktop applications, without human intervention. Events trigger automatic gathering of relevant data about the caller from the user's client/server and host repositories and delivers the data together with the call to the agent's desktop. SoftPhone Agent provides call center agents a specialized interface for call management and control.

AnswerSoft has also announced the availability of a Universal TAPI 2.0 Service Provider (UTSP) for PBX manufacturers seeking rapid development of direct communication links between PBX systems and Windows NT Server 4.0-based machines. Hardware manufacturers now have the ability to significantly reduce the development effort required to provide Telephony Service Providers or drivers necessary to fully integrate client/server applications with PBX switches in a Windows NT 4.0/TAPI 2.0 environment. Rapid development of these drivers will enable call centers to utilize information obtained from a telephone switch to drive the communications process across the network.

Specifically, the UTSP provides the communication layer between the TAPI 2.0 server and the switch, including the capability of queuing numerous telephony events. Additionally, the UTSP provides the mapping layers required between telephony events, and the devices affected by these events. Using the UTSP, the programmer simply reads or writes to the queue, translates the event, and feeds them to the UTSP to be correctly bundled to send to the server. This process eliminates the requirement to write mapping and transition tables as well as programming switch-to-server communications links - the most time consuming and complex layers within the Telephony Service Provider. For more information, contact Lisa Chiranky at 214-612-5114.

Konexx Announces New Product Line
Konexx (Unlimited Systems Corporation) announced a new line of interface products designed to provide analog RJ-11 connections to digital, PBX, and multiline phone systems. The new line is intended to address the need for cost-effective Internet access at the desktop. According to Konexx, the increasing use of modems and other devices requiring analog telephone connections is causing a shortage of available analog lines in the workplace.

New products include Office Konnector, which connects desktop computer fax modems to office phone systems; DataTalk I, which connects VoiceView (AVD) modems to office phone systems, allowing alternating voice and data through the same connection, without the need for additional headsets of handsets; DirectTalk II, which connects Digital Simultaneous Voice and Data (DVSD) modems to office phone systems, allowing simultaneous voice and data through the same connection, without the need for additional headsets or handsets. Konexx asserts that it maintains its commitment to research and development by providing solutions for the latest modem technology including VoiceView/DSVD modems and by supporting V.34 applications that transmit data up to speeds of 33.6 K bps. For more information, contact KONEXX at 800-275-6345 or 619-622-1400.

LiveTouch allows the Web site visitor who needs additional information or wants a secure way to finalize a transaction to request telephone contact from a call center representative by pressing an on-screen button. In the call center, calls are delivered to the representatives best able to handle them. The appropriate application and data arrive at the representative's desktop with the phone call. This data may include information on the Web page the call requester was visiting, together with a customer record from the corporate database or information the visitor entered on the Web page. The representative is ready to assist the web customer immediately.

LiveTouch also provides facilities for linking data on Web use to corporate databases. This data may include the number of visitors to a certain page, time of the day visits are occurring, and an array of data captured in Web site guest logs or survey forms.

LiveTouch also provides facilities for compiling names and phone numbers from guest logs or survey forms into telemarketing lists. Telemarketers can perform speedy, single-key outdialing from these lists, and receive the appropriate application and data at their workstations with the calls. For more information, contact Aristacom at 510-748-1500.

Personal Caller ID will work with any Windows Telephony-compliant desktop hardware that captures caller ID and can dial computer-linked phones, modems, ISDN modems, dialers, and proprietary desktop interfaces to PBXs and key systems. For Windows 95 environments, a 486 or higher processor with 8 MB of RAM is required. For Windows 3.1 environments, a 486 or higher processor with 4 MB of RAM is required. For more information, contact Michael Stanford at 202-332-2110.

Combining hardware and software in a single plug-in board, Antares is part of Dialogic's SCSA product family for providing voice, fax, ISDN, wireless, and Internet solutions. "The combination of Antares and products from our ASR and TTS partners will enable developers to respond more quickly to new business opportunities and to deliver solutions more easily and efficiently," said Jeff Hill, Dialogic product manager for Antares. "With a single investment, they can maximize their options without having to invest in multiple, proprietary and incompatible development solutions."

"Continued growth in the wireless and wireline markets is driving tremendous demand for ASR and TTS applications, particularly among businesses that are focused on providing improved customer services," said Howard G. Bubb, Dialogic's president and chief executive officer. "For example, mobile business people want the safety and convenience of voice-activated dialing from car phones. Likewise, information providers are looking to make their services easier to use for customers."

The Antares hardware, available now from Dialogic, sells for approximately $3,500 in small quantities, while software packages from Antares Technology Developers range in price from $250 to over $1,000 per channel, based on the software's sophistication and capability. Software is available either from Dialogic or through the Antares ASR/TTS partner directly. For more information, contact Dialogic at 800- 755-4444 or 201-993-3030.

The computers are intended for server, processing, embedded systems, mutimedia, and other throughput- intensive applications. Sharing the same backplane technology as ICS's Intel-based platforms, the SB21064 Series is designed to bring workstation performance to standard PCs and is backwardly compatible to existing ISA passive backplane systems as well as future PICMG-compliant, ISA/PCI designs.

Announced prices are $2749 (275 MHz; NT) to $3599 (300 MHz; Digital Unix). Off-the-shelf delivery is available.

ICS also announced the publication and immediate release of its new 1997 Master Source-Book Edition 1 for data acquisition/control and telephony applications. New offerings include Pentium ATX motherboards and chassis with up to 9 PCI expansion slots; 8300 Series flat-panel displays with up to 4 x 3.5" drive bays; benchtop and PC monitor enclosures for harsh environment applications; expanded line of hubs, switches, concentrators, and other networking products; more instrumentation cards, peripherals, and accessories. For more information, call ICS at 800-677-7329.

Based on the ISA bus, the TAP-802 accommodates up to 12 TMS320C32 processors providing 720 MFLOPS of power. This performance enables the TAP-802 to run multiple vocoders for real-time voice over the Internet applications and modem algorithms for remote data access. The TAP 802 is available with 2 Mbytes of shared global DRAM and 256 Kbytes of local, zero wait-state SRAM per processor. In addition, the TAP-802 open DSP platform supports both MVIP and SCSA standard interfaces, 128 local TDM channels, and is available with complete UNIX- and Windows NT-compatible software. For more information, contact Terry Shugart at 508-977-3000, ext. 3521.

Alliances/Acquisitions

Guest Messaging Systems Supplied By Applied Voice Technology And MCORP
Applied Voice Technology announced that Signature Inns has standardized on Emissary guest messaging systems developed by AVT and marketed by MCORP and has installed Emissary in its 23 inns across the United States. Emissary Guest Voice Mail Plus provides Signature Inns nationwide with voice mail and wake-up call management for its guests.

Emissary, which is based on AVT's CallXpress3 Automated Agent IVR software, incorporates customized features for hotels that cater to the business professional. Emissary can also be easily configured to meet the requirements of other hotel and hospitality providers.

With the installation of Emissary, the 23-property lodging chain provides guests the ability to receive detailed messages without having to involve the front desk. Emissary also provides an efficient means of disseminating information among hotel personnel. For example, Housekeeping can update the status of a room and submit maintenance requests with the use of a touch-tone phone. Front-desk staff can then access the information on their computers to find out if a room is ready for the next guest. For more information, contact AVT at 206-820-6000 or MCORP at 908-321-4144.

ATCOM/INFO Teams With GTE For Internet Access At Airport
ATCOM/INFO announced that 10 of its kiosks will be installed by GTE at Dallas-Fort Worth International Airport. Designed for business travelers in high-traffic, high-wait areas, the kiosks provide users with access to e-mail, the World Wide Web, multimedia presentations, Internet newsgroups, ticketing services, information about local attractions and events, and targeted advertising.

Each four-sided, pyramid-shaped kiosk is seven feet tall and has one sit-down and one stand-up Internet access terminal, plus two stations for telephone service. The Pentium-based kiosks feature large, easy-to- read screens with easy-to-follow instructions, ergonomically designed keyboards, a centrally located touch-pad mouse, and an electronic reader slot that will accept credit card payments for usage.

Internet access will be provided by GTE via an ISDN connection that provides speeds 10 times faster than today's most commonly used 28.8 Kbps computer modems. And access speed is further increased by ATCOM/INFO's implementation of two innovative concepts: stored custom configurations and automated remote administration. Stored custom configurations allow rapid Internet access via any kiosks in the world without having to provide user configuration information each time. For more information, contact Peter Van Horne at 619-685-3168.

Voysys Partners With Siemens For Voice Messaging Solution
Voysys, in partnership with Siemens Business Communication Systems, will design and manufacture the voice messaging system for Siemens' OfficePoint ISDN system. The intent is to provide OfficePoint customers with complete message handling and retrieval for remote or small offices.

Message Valet, Voysys' voice message system, will be offered with the purchase of OfficePoint for professionals seeking advanced messaging capabilities with their high-speed ISDN phone system. Features and functions include off-site message notification to a pager, personal mailbox greetings, password privacy, and broadcast messages, with an interface designed for the non-telephony expert. For more information, contact Ashlie Bryant (Voysys) at 510-252-1100 or Anita Giani (Siemens) at 408-777- 4546.

WilTel To Provide Web/IVR Integration With InterVoice's VisualConnect
Williams Telecommunications systems and InterVoice announced an expansion to their business relationship whereby WilTel will provide business users Internet and voice processing integration through VisualConnect, InterVoice's Internet interactive information exchange system. VisualConnect will enable WilTel customers to access IVR, transaction, workflow-based, and telephony integrated applications via the Internet. For more information, contact Dawn Brimberry (WilTel) at 713-307-4872 or Gordon Givens (InterVoice) at 214-454-8834.

Dialogic And Syntellect Announce License Agreement
Dialogic, a manufacturer of computer-telephony components, and Syntellect, a provider of interactive communications solutions, announced a licensing and distribution agreement in which Dialogic will offer its customers a portfolio of CT patents from the Syntellect subsidiary, Syntellect Technology Corporation (STC). The agreement gives CT developers another option for obtaining these patent rights when they purchase specified Dialogic CT products and provides Syntellect with an efficient channel to spread distribution of their core patent rights.

Under the terms of the agreement, application developers have the option to purchase a pass-through license for STC patents directly from Dialogic. Dialogic customers can now choose to take advantage of these patent rights on a per-port basis. Customers will also benefit from being able to acquire both CT components and patent rights from a single source. Additionally, as part of the agreement, STC will absolve Dialogic customers from patent infringement claims, provided they participate in this program.

The STC patent portfolio consists of 17 U.S. patents and several foreign counterparts covering such CT features as automated attendant and specific voice mail functions, announcing wait times in call queuing systems, and use of the calling party's telephone number in call processing systems. STC has previously licensed 16 companies under one or more of the patents covered by this agreement, including voice and call processing companies and PBX manufacturers. STC offers licensees either a royalty-based or fully paid license for individual or groups of their patents. For more information, contact Dialogic at 201-993- 3000 or Syntellect at 770-587-0700.

Ingram Micro will provide Mitel's existing dealer network with access to its training and certification programs to further their data expertise. In addition, Ingram Micro will offer engineering support services such as configuration centers, network consulting and implementation, as well as dealer financing, inventory floor planning, and leasing services. For more information, contact Kirsten Frosh (Ingram Micro) at 714-566-1000, ext. 2727.

The solution, which consists of GeoTel's Intelligent CallRouter and InterVoice's OneVoice platform, is designed to enable companies to leverage their network infrastructure with intelligent call routing to the most appropriate call center location, automated call processing system, or customer service representative. Incoming calls can be screened at the network level for available agents, routed to any available InterVoice system for queuing and further prompting, or delivered to agents. For more information, contact Nancy Pieretti (GeoTel) at 508-486-1149 or Cathy Kingeter (InterVoice) at 214-480- 8934.

"NetSpeak's partnering with the 'technology leader of their market' will result in the rapid introduction of a number of exciting new Internet telephony-based products for worldwide distribution in the retail market," said Robert Kennedy, president and Chief Operating Officer of NetSpeak Corporation.

Creative is bundling and distributing Creative WebPhone with SoundBlaster and with its communications products Phone Blaster 33.6 PnP and Internet Blaster 33.6 PnP, bringing NetSpeak's technology into the consumer retail market.

Creative WebPhone is the first in a line of Internet-based telephony products to be introduced over the next year. It is designed to offer low-cost, long-distance telephone calls over the Internet. TrueSpeech audio, caller ID, directory assistance, conferencing, and chat functionality within a familiar user interface are included. Creative intends to build upon the technology of its multimedia products to create a suite of products suited for home and business use. For more information, contact Creative Technology at 408- 424-5700 or NetSpeak at 561-997-4001.