ProductsDialogic
VFX/PCI
International SCSA/H.100 4-Port PCI Voice and Fax Board with Caller ID


Features and Benefits

  • Four independent voice and fax processing ports in a single PCI slot
  • Supports low- to medium-density enterprise Unified Messaging, integrated voice and fax mail, and single call fax-on-demand applications
  • SCSA™ SCbus™ connectivity enables applications requiring switching and allows access to additional hardware resources
  • H.100 connector allows for easy future expansion to the new industry standard CT Bus™
  • Plug and Play ready. Hardware installation is simplified by eliminating DIP switches and jumper settings and enabling software-controlled configuration.
  • Configure multiple boards in a single chassis, PCI bus or mixed PCI/ISA bus, for easy and cost-effective system expansion of up to 64 analog ports
  • Dialogic SpringWare™, downloadable signal and call processing firmware, provides easy feature enhancement and field proven performance based on over three million installed ports
  • PerfectDigit™ DTMF (touchtone) provides reliable detection during voice playback—allows callers to "type-ahead" through menus
  • A-law or µ-law voice coding at dynamically selectable data rates, 24 Kb/s to 64 Kb/s, selectable on a channel-by-channel basis for optimal tradeoff in disk storage and voice quality
  • International Caller ID capability via on-hook audio path. Supports Bellcore CLASS™, UK CLI, and other international protocols.
  • Patented outbound call progress analysis monitors outgoing call status quickly and accurately
  • 14,400 b/s (v.17) fax with ITU-T Group 3 and ETSI NET/30 compliance ensure operation with fax machines worldwide. Provides seamless voice + fax integration at hardware and software levels. Supports full compliance of ITU-T (T.4, T.6, T.30) recommendations.
  • ASCII-to-fax conversion on the fly allows direct transmission of text files and frees host computer for other processing
  • Page header information is generated automatically
  • Scan line error correction detects, reports, and repairs faulty scan lines for improved image quality
  • Dynamic page-width conversion preserves aspect ratio resulting in undistorted images. Sends and receives varying page sizes according to capabilities of remote machine.
  • Phase B, Phase D return allows real-time application control at beginning and end of fax process
  • Operator intervention provides switching between voice and fax during same call enabling integrated voice/fax mailbox applications and a proprietary communication channel to allow confidential transmissions
  • Page concatenation on the fly allows sending of multiple images in one fax
  • Text and graphics can be combined on same page permitting mixing of coding methods
  • Support for TIFF/F, MH, and MMR data formats compresses fax data
  • Multiple polling modes can automatically collect documents from remote devices. After transmission of a document, turnaround polling allows the calling unit to poll (to receive a fax) the remote fax unit during the same call.
  • Fine resolution supported for improved image quality
  • Fill minimization results in faster transmission and more compact file storage; no unnecessary fill bits are sent or stored
  • Fully integrated voice/fax API enables easy incorporation of fax capability into existing voice applications
  • C-language application program interface (API) for Windows NT® shortens your development cycle so you can get your applications to market faster
  • Supports PBXpert32™, a free utility that simplifies switch integration
  • Optional, on-board Global Dial Pulse Detection™ (DPD™) feature enables callers without touchtone phones to access applications. No additional pulse-to-tone conversion equipment needed.
  • Windows TAPI/WAVE support (including 11 kHz, 8-bit file format)
Applications
  • Unified messaging
  • Fax mail
  • Integrated voice mail and fax mail
  • Integrated voice/fax response
  • Single call fax-on-demand
  • Remote database transactions
The VFX/PCI™ supports four simultaneous voice or V.17 fax instances in a single PCI slot. The board incorporates powerful new features found on the latest Dialogic voice processing boards, including support for:
  • International caller ID (CLASS™, CLIP, ACLIP)
  • Global Dial Pulse Detection (DPD)
  • Speech recognition
  • Text-to-speech 
  • WAVE file support at 11 kHz linear PCM
You can build unified messaging systems with the VFX/PCI board that give customers unrestricted use of call and fax processing in the same call, or build voice response systems equipped with remote hard copy capability via fax. The VFX/PCI on-board analog interface is approvable in over 30 countries, so you can sell your VFX/PCI-based applications worldwide. The fax channels on the VFX/PCI can be used as a dedicated resource or they can be shared by multiple call processing resources through an SCbus™ interface. The VFX/PCI board provides everything required for building integrated call processing and fax services scalable from 4 ports to 24 ports and beyond.

 The VFX/PCI combines the best voice processing and fax technology in the call processing industry: the reliable, field-proven SpringWare™ voice processing capability and fax modems from Rockwell International.

 Fax functions are seamlessly incorporated into Dialogic standard voice drivers. Complete voice and fax integration at the hardware level via a shared DSP provides efficient on-demand voice and fax image processing. Consistent voice and fax commands ensure portability between operating systems.

 The VFX/PCI board incorporates an H.100 interface for compatibility with next generation CT Bus™ products, and at the same time is backwardly compatible with existing SCSA products through the use of an optional CT Bus/SCbus adapter.

 The VFX/PCI is built on the proven dual-processor architecture pioneered by Dialogic. This configuration combines a Motorola digital signal processor (DSP) with an Intel-compatible control processor to provide balanced processing capacity and a robust software architecture to enable advanced signal computing. The architecture allows the product to run Dialogic downloadable signal and call processing firmware, SpringWare™, which provides an advanced set of call processing features, including selectable rate, high-quality voice coding with speed control, outstanding DTMF detection with cut through, and advanced outbound call progress analysis.
 


Configurations

 Use the VFX/PCI board to build sophisticated, computer telephony systems to which capabilities such as speech recognition, facsimile, and text-to-speech can be added. The VFX/PCI shares a common hardware and firmware architecture with the other Dialogic SCbus boards for maximum flexibility and scalability. Features can be added and systems can grow while protecting investment in hardware and application code. With only minimum modifications, applications can be easily ported to lower or higher line-density platforms.
 


The VFX/PCI installs in any PCI-based personal computer or server (PCI bus or mixed PCI/ISA) and compatible computers (Intel 80386, 80486, or Pentium™-based PC platforms). The VFX/PCI provides everything required for building integrated voice and fax solutions scalable from 4 ports to 64 ports. The maximum number of lines that can be supported is dependent on the application, the amount of disk I/O required, and the host computer CPU and power supply.

 Applications developed to run on the Dialogic VFX/40ESC plus™ product will run on a similar VFX/PCI configuration.
 


The VFX/PCI board can operate within a mixed chassis containing Dialogic PCI and ISA products. The VFX/PCI's forward-looking design incorporates the new H.100 connector to simplify connection to next generation CT Bus products. The VFX/PCI can also connect to existing SCbus products through the use of an optional CT Bus/SCbus adapter. The adapter provides both SCbus and H.100 physical connectors required to link the VFX/PCI to current SCbus products.

 Software Support

 The Dialogic System Software and Software Development Kit for Windows NT® (Native) currently supports the VFX/PCI. This package contains a set of tools for developing complex multi-channel applications. In addition, the package also contains a Dialogic Fax Development Package that supports all VFX series fax products. Applications developed on one platform can be easily ported to the other. In addition, the fax interface is designed to work consistently with Dialogic standard voice and call processing interfaces. A C language-based library/ application programming interface (API) lets you develop a wide variety of applications. You can use the same commands to dial and receive voice and fax calls. The application can then play or record voices files, or send or receive fax files in the same session, providing the utmost application flexibility. An API with logical addressing resource management eliminates the need to keep track of physical resource addresses such as time slots and board numbers.

 Functional Description

 The VFX/PCI board combines four-line voice processing with 14,400 b/s (v.17) facsimile that can perform all standard functions of a fax machine and more, including sending and receiving multiple documents, polling, broadcasting, and turnaround polling. VFX/PCI uses a unique dual-processor architecture that combines the signal processing capabilities of a DSP with the decision-making and data movement functionality of a general-purpose control microprocessor. The VFX/PCI provides call processing for either analog loop start telephony signals from the telephone network or resource sharing for digital signaling and digital voice information input via the SCbus. This architecture handles real-time events, manages data flow to the host PC for faster system response time, reduces host PC processing demands, filters DTMF and telephony signaling, and frees the DSP to perform signal processing on the incoming call.

 A telephone line interface on the baseboard receives analog voice and telephony signaling information from the telephone network (see block diagram). This telephone line interface uses reliable, solid state hook switches (no mechanical contacts) and FCC part 68 class B ring detection circuitry. This FCC-approved ring detector is less susceptible to spurious rings created by random voltage fluctuations on the network. Each interface incorporates circuitry that protects against high-voltage spikes and adverse network conditions and allows applications to go off hook any time during ring cadence without damaging the board. Ring and on-hook/off-hook information and voice signals are conditioned by the telephone line interface and applied to a signal processor called a CODEC.

 The CODEC samples and digitizes the audio signal and sends a digitized audio signal via a digital audio bus (TDM) to a Motorola DSP for processing. When the Motorola DSP identifies the call as a fax call, this information is communicated to the control processor which switches the voice channel into fax mode and activates the corresponding fax channel.

 When a fax call is received, it is connected via an internal analog audio bus to a fax channel duplexer. The duplexer enables the fax modem to alternately transmit and receive analog audio (modem) signals via the internal analog audio bus to the telephone line interface or SCbus.

 For call processing, the DSP processes the digitized voice data based on SpringWare firmware loaded in code/data RAM. The DSP performs the following signal analysis and operations on the incoming data:
 


  • automatic gain control to compensate for variations in the level of the incoming audio signal
  • applies an ADPCM (Adaptive Differential Pulse Code Modulation) or PCM (Pulse Code Modulation) algorithm to compress the digitized voice and save disk storage space
  • detects the presence of tones—DTMF, MF, or an application-defined single or dual tone
  • silence detection to determine whether the line is quiet and the caller is not responding For outbound data, the DSP performs the following operations:
  • expands stored, compressed audio data for playback
  • adjusts the volume and rate of speed of playback upon application request
  • generates tones—DTMF, MF, or any application-defined general-purpose tone The dual processor combination also performs outbound dialing and call progress monitoring
  • transmits an off-hook signal to the telephone network
  • dials out (makes an outbound call)
  • monitors and reports results—line busy or congested; operator intercept; ring, no answer; or if the call is answered, whether answered by a person, an answering machine, a facsimile, or a modem.
When recording speech, the DSP can use different digitizing rates from 24 to 64 Kb/s as selected by the application for the best speech quality and most efficient storage. The digitizing rate is selected on a channel-by-channel basis and can be changed each time a record or play function is initiated. The DSP processed speech is transmitted via the control processor to the host PC for disk storage. When replaying a stored file, the processor retrieves the voice information from the host PC and passes it to the DSP which converts the file into digitized voice. The DSP sends digitized voice and appropriate signaling responses to the CODEC to be converted into analog voice and ring and on-hook/off-hook signals for transmission to the telephone network. Outbound processing is the reverse of inbound processing.

 Ring and on-hook/off-hook signaling data is passed to the on-board control microprocessor and transmitted to the application via a dual-port shared RAM and the host PC ISA bus.

 When the SCbus is selected, digital voice and signaling information from a network board or other resource enter the VFX/PCI via the SCbus interface. The SC2000 chip manages these signals and acts as the traffic coordinator to store (buffer) the high-speed digital data from the bus until the data for each channel can be transmitted to the DSP.

 The SC2000 chip transmits several lower-speed data streams over a single high-speed channel in the SCbus. This chip also incorporates matrix switching capabilities. Under control of the on-board control processor, the SC2000 chip can connect any call being processed or any fax channel to any of the four analog lines or to any time slot (1024 for SCbus). This enables the application to switch calls to or from other resources (such as call processing, speech recognition or TTS) as they are needed, or to reroute calls.

 The SC2000 chip can bundle time slots to carry high-bandwidth data and can broadcast to multiple resources over the SCbus.

 In SCSA configuration, the external SCbus can be programmed to operate at 2.048 or 4.096 Mb/s while the local bus operates at 2.048 Mb/s.

 The on-board control processors control all operations of the VFX/PCI board via a local bus and interpret and execute commands from the host PC. These processors handle real-time events, manage data flow to the host PC to provide faster system response time, reduce PC host processing demands, process DTMF and telephony signaling before passing them to the application, and free the DSP to perform signal processing.

 Communications between a processor and the host PC is via the Shared RAM that acts as an input/output buffer and thus increases the efficiency of disk file transfers. This RAM interfaces to the host PC via the PCI bus. All operations are interrupt-driven to meet the demands of real-time systems. When the system is initialized, SpringWare firmware is downloaded from the host PC to the on-board code/data RAM and DSP RAM to control all board operations. This downloadable firmware gives the board all of its intelligence and enables easy feature enhancement and upgrades.

 With the rotary switch on the VFX/PCI set to 0, the board is hardware Plug and Play enabled. Configuration is handled exclusively by software. Alternatively, you can set the rotary switch to another value to manually control board location for ease of cabling or backwards compatibility with Dialogic Board Locator Technology™ (BLT) installation.

 Technical Specifications*
 
 
Number of ports 4 channels of call processing and facsimile
Max. boards/system 16
Analog network interface On-board analog loop start
Resource sharing bus SCbus or CT Bus
Fax modem Rockwell R144EFXL
Fax control microprocessor Intel 80C186 @ 16 MHz
D/41EPCI control microprocessor Intel 80C186 @ 16 MHz
Call processing  
digital signal processor Motorola DSP56002 @ 49 MHz, with 128 K word private, 0 wait state SRAM
FACSIMILE SPECIFICATIONS:
Fax compatibility ITU-T G3 compliant (T.4, T.30), ETSI NET/30 and T.6 compliant
Data rate 14,400 b/s (v.17) (max.)
  Variable speed selection
  Automatic step-down to 12,000 b/s, 9600 b/s, 7200 b/s, 4800 b/s, and lower
Transmit Data Modes
MH (Modified Huffman), MR (Modified Read), and MMR (Modified Modified Read) with or without ECM
Receive Data Modes MH (Modified Huffman), MR (Modified Read), MMR (Modified Modified Read), all with or without ECM
File data formats TIFF/F (Tagged Image File Format) for transmit/receive
  MH, ASCII and Katakana text, transmit
ASCII-to-fax conversion On-board conversion
  Direct transmission of text files
  Multiple fonts supported
  Embedded formatting commands supported
  Page headers generated automatically
Error correction Detection, reporting, and correction of faulty scan lines
Image widths 215 mm (8.5 in), 255 mm (10.0 in), and 303 mm (11.9 in)
Image scaling Automatic horizontal and vertical scaling between page sizes
Polling modes Normal and turnaround
Image resolution Normal (203 pels/in x 98 lines/in)
  Fine (203 pels/in x 196 lines/in)
Fill minimization Automatic fill bit insertion and stripping
HOST INTERFACE:
Bus compatibility PCI. Complies with PCISIG Bus Specification, Rev. 2.1.
Bus speed 33 MHz max
Bus mode 32 bit to 16 bit conversion in target mode
Shared memory 64 Kbyte page
I/O ports None
TELEPHONE INTERFACE:
Trunk type Loop start
Impedance Configurable by parameter†
Loop current range 8 to 100 mA
Ring detection 15 Vrms min, 13 to 68 Hz†
Echo return loss Configurable by parameter†
Cross talk coupling Less than –70 dB at 1 kHz channel to channel
Signal/noise ratio 70 dB (referenced to –15 dBm)
Freq. response 300 Hz to 3400 Hz ±3 dB (transmit and receive)
Connector Four RJ-11 type
POWER REQUIREMENTS:
+5 VDC 2.62 A max.
+12 VDC 200 mA max.
–12 VDC 150 mA max.
Operating temperature 0°C to +50°C
Storage temperature –20°C to +70°C
Humidity 8% to 80% non-condensing
Form factor PCI long card, 12.3 in. long. (without edge retainer) or 13.3 in. long (with edge retainer), 0.79 in. wide (total envelope), 3.87 in. high (excluding edge connector)
SAFETY & EMI CERTIFICATIONS:
United States FCC part 68 ID #: EBZUSA-75385-VM-T;
  FCC Part 15, Class A
  UL: E96804 UL1950
Canada DOC: 885–5542A
Warranty 3 years standard

SpringWare Technical Specifications*
 
 
AUDIO SIGNAL:
Receive range –50 to –13 dBm (nominal), for average speech signals.** Configurable by parameter.‡
Automatic Gain Control Application can enable/disable. Above –18 dBm results in full scale recording, configurable by parameter.†
Silence detection –38 dBm nominal, software adjustable†
Transmit level (weighted average) –9 dBm nominal, configurable by parameter†
Transmit volume control 40 dB adjustment range, with application definable increments
Frequency response  
24 Kb/s 300 Hz to 2600 Hz ±3 dB
32 Kb/s 300 Hz to 3400 Hz ±3 dB
48 Kb/s 300 Hz to 2600 Hz ±3 dB
64 Kb/s 300 Hz to 3400 Hz ±3 dB
AUDIO DIGITIZING:
24 Kb/s
32 Kb/s ADPCM @ 8 kHz sampling
48 Kb/s µ-law PCM @ 6 kHz sampling
64 Kb/s µ-law PCM @ 8 kHz sampling
Digitization selection Selectable by application on function call by call basis
Playback speed control Pitch controlled; available for 24 and 32 Kb/s data rates.
  Adjustment range: ±50%; Adjustable through application or programmable DTMF control.
WAVE AUDIO: Supports 11 kHz linear PCM, 8-bit mono mode (available only when running Windows® 95 and Windows NT)
DTMF TONE DETECTION:
DTMF digits 0 to 9, *, #, A, B, C, D per Bellcore LSSGR Sec 6
Dynamic range Programmable, default set at –36 dBm to +3 dBm per tone
Minimum tone duration 40 ms, can be increased with software configuration
Interdigit timing Detects like digits with a 40 ms interdigit delay
  Detects different digits with a 0 ms interdigit delay
Twist and frequency variation Meets Bellcore LSSGR Sec 6 and EIA 464 requirements
Acceptable twist 10 dB
Signal/noise ratio 10 dB (referenced to lowest amplitude tone)
Noise tolerance Meets Bellcore LSSGR Sec 6 and EIA 464 requirements for Gaussian, impulse and power line noise tolerance
Cut through Detects down to –30 dBm per tone into 600 ohm load impedance
Talk off Detects less than 20 digits while monitoring Bellcore TR-TSY-000763 standard speech tapes (LSSGR requirements specify detecting no more than 470 total digits). Detects 0 digits while monitoring MITEL speech tape #CM 7291.
GLOBAL TONE DETECTION™:
Tone type Programmable for single or dual
Max. number of tones Application dependent
Frequency range Programmable within 300 to 4000 Hz
Max. frequency deviation Programmable in 5 Hz increments.
Frequency resolution Less than 5 Hz. — Note: certain limitations exist for dual tones closer than 60 Hz apart.
Timing Programmable cadence qualifier, in 10 ms increments
Dynamic range Programmable, default set at –36 dBm to +3 dBm per tone
GLOBAL TONE GENERATION™:
Tone type Generate single or dual tones
Frequency range Programmable within 200 to 4000 Hz
Frequency resolution 1 Hz
Duration 10 msec increments
Amplitude –43 dBm to –3 dBm per tone, programmable
MF SIGNALING:
MF digits 0 to 9, KP, ST, ST1, ST2, ST3 per Bellcore LSSGR Sec 6, TR-NWT-000506 and ITU-T Q.321
Transmit level Complies with Bellcore LSSGR Sec 6, TR-NWT-506
Signaling mechanism Complies with Bellcore LSSGR Sec 6, TR-NWT-506
Dynamic range for detection –25 dBm to +3 dBm per tone
Acceptable twist 6 dB
Acceptable freq. variation Less than ±1 Hz
CALL PROGRESS ANALYSIS:
Busy tone detection Default setting designed to detect 74 out of 76 unique busy/congestion tones used in 97 countries as specified by ITU-T Rec. E., Suppl. #2. Default uses both frequency and cadence detection.
  Application can select frequency only for faster detection in specific environments.
Ring back detection Default setting designed to detect 83 out of 87 unique ring back tones used in 96 countries as specified by ITU–T Rec. E., Suppl. #2. Uses both frequency and cadence detection.
Positive Voice  
Detection™ accuracy >98% based on a database of real world calls
Positive Voice Detection speed Detects voice in as little as 1/10th of a second
Positive Answering  
Machine Detection™ accuracy 80 to 90% based on application and environment
Fax/modem detection Preprogrammed
Intercept detection Detects entire sequence of the North American tri-tone.
  Other SIT sequences can be programmed.
Dial tone detection  
before dialing Application enable/disable; supports up to three different user definable dial tones; programmable dial tone drop out debouncing.
TONE DIALING:
DTMF digits 0 to 9, *, #, A, B, C, D; 16 digits per Bellcore LSSGR Sec 6, TR-NWT-506
MF digits 0 to 9, KP, ST, ST1, ST2, ST3
Frequency variation Less than ±1 Hz
Rate 10 digits/s max., configurable by parameter†
Level –4.0 dBm per tone, nominal, configurable by parameter†
PULSE DIALING:
10 digits 0 to 9
Pulsing rate 10 pulses/s, nominal, configurable by parameter†
Break ratio 60% nominal, configurable by parameter†
ANALOG CALLER IDENTIFICATION:
Applicable Standards Bellcore TR-TSY-000030
  Bellcore TR-TSY-000031
  TAS T5 PSTN1 ACLIP: 1994 (Singapore)
  British Telecom SIN 242 (Issue 01)
  British Telecom SIN 227 (Issue 01)
Modem Standard Bell 202 or V.23, serial 1200 bits/sec (simplex FSK signaling)
Receive sensitivity –48 dBm to –1 dBm
Noise tolerance Minimum 18 dB SNR over 0 to –48 dBm dynamic range for error free performance
Data formats Single Data Message (SDM) and Multiple Data Message (MDM) formats via API calls and commands
Line impedance AC coupled 600 Ohm (@1.8 kHz) termination during caller ID on-hook detection interval
Message formats ASCII or binary SDM, MDM message content
ANALOG DISPLAY SERVICES INTERFACE:
  FSK generation per Bellcore TR-NWT-000030. CAS tone generation and DTMF detection per Bellcore TR-NWT-001273.
* All specifications are subject to change without notice.
** Average speech mandates +16 dB peaks above average and preserves –13 dB valleys below average.
† Configurable to meet country specific PTT requirements. Actual specification may vary from country to county for approved products.
‡ Analog levels: 0 dBm0 corresponds to a level of +3 dBm at tip-ring analog point. Values vary depending on country requirements. Contact your Dialogic Sales Engineer.

 Hardware System Requirements

  • 80386, 80486, or Pentium PCI bus or mixed PCI/ISA bus PC or compatible computer
  • Operating system hardware requirements vary according to the number of channels being used
  • System must comply with PCISIG Bus Specification Rev. 2.1 or later.
Additional Components
  • Optional multi-drop CT Bus cable
  • Optional CT Bus/SCbus Adapter

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E-mail : dialogic@foremost-systems.com