AMDEV Communications Corp. - Voice Mail, Voice Processing, Computer Telephony, CTI, Automated Dialers with Parts, Service and Support
 
Voice Mail, Voice Processing, Computer Telephony, CTI, Automated Dialers with Parts, Service and Support
Dialogic
 
>
Contact
 
 
 
PH: 209.962.5900
FAX: 209.962.5800
Welcome - AMDEV Communications, Your Voice Mail Specialists
  Home > Products > Components > Dialogic > D/12H and D/41H

Dialogic D/21H and D/41H

2- and 4-Port Voice Processing Boards

Features and Benefits

  • Freedom of choice: supports Windows* 95, Windows NT* (including TAPI/WAVE), MS-DOS*, OS/2*, and UNIX*
  • Easily expands markets to satisfy international demands by providing a full suite of international telephone network approvals
  • International Caller ID capable: supports North American Bellcore CLASS, UK CLI, and NTT CLIP
  • Enables advanced call processing features for competitive differentiation by supporting software-based features such as Global DPD; and PBX Expert tone characterization utility
  • Provides reliable DTMF detection during voice playback letting callers "type ahead" through voice menus for quicker completion of call transactions
  • Ensures reliability via call progress analysis which monitors outgoing call status quickly and accurately
  • Offers flexible voice coding at dynamically selectable data rates, 24 to 64 Kb/s, selectable on a channel-by-channel basis for optimal tradeoff in disk storage and voice quality
  • Offers superior voice quality through enhanced telephone circuitry and AGC
  • Lets developers build cost-effective scalable systems from two to 64 ports
  • Compatible with legacy telephone switches in the United Kingdom and Northern Europe that use Earth Break Recall
  • Lets developers build flexible, cost-effective Internet telephony platforms for small-business applications

Applications

  • Voice messaging
  • Automated attendant
  • Interactive voice response
  • Audiotex
  • Inbound and outbound telemarketing
  • Small call centers

The four-line Intel® Dialogic® D/41H board and its two-line version, the Intel® Dialogic® D/21H board, are ideal for applications that need high-performance voice processing but don't require the large-scale system sophistication of products based on SCbus or CT Bus. The D/21H and D/41H boards use the same application programming interface (API) as their predecessors, making it easy to scale existing applications upward to take advantage of the power and features of these boards. The D/21H and D/41H boards also have improved voice quality and automatic gain control (AGC) over the legacy D/21D and D/41D boards. Even the weakest of telephone signals traveling over difficult telephone lines can be recorded and played back with complete clarity.

The D/21H and D/41H boards use digital signal processor (DSP) voice processing technology, making them ideal for small- and medium-sized, server-based computer telephony (CT) systems - particularly under the Windows* operating systems. Windows support includes TAPI and WAVE APIs, which facilitate call control, recording, and playback of voice messages under the Microsoft Windows Open Services Architecture (WOSA). The D/21H and D/41H voice processing boards give Windows 95 and Windows NT* application developers a platform for creating sophisticated interactive voice response (IVR) applications. The "H Series" boards also support use in MS-DOS*, OS/2*, and UNIX* operating system environments.

International Caller ID is supported on the D/21H and D/41H boards, letting an application such as IVR receive calling party information via a telephone trunk line. Caller ID is supported for North America (CLASS protocol), the United Kingdom (CLI protocol), and in Japan (CLIP protocol).

The Global Dial Pulse Detection (Global DPD) algorithm is available for both boards, enabling application development for deployment in countries with limited touch tone telephone service. Global DPD is optimized for several countries and provides superior dial pulse detection wherever it has been optimized.

The D/21H board can also be utilized as a cost-effective platform to develop Internet telephony applications that are ideal for the small-business environment.

With all of these advanced features in a half-size ISA board footprint, the D/21H and D/41H boards are perfect for client or small server system development. Both boards offer enhanced DSP power and memory capacity that not only provides a base level of performance for today's requirements, but also provides the "headroom" for future application expansion through software-based technologies.

Configurations

Use the D/21H and D/41H boards to build sophisticated messaging and IVR CT systems with optional technologies, such as automatic speech recognition (ASR), Global DPD, TTS, and PBX Expert. These boards share a common hardware and firmware architecture with other Intel voice boards for maximum flexibility and scalability. More ports and new features can be added while protecting your original investment in hardware and application code. With only minimum modifications, applications can be easily ported to higher line density platforms.

The D/21H and D/41H boards install in IBM PC XT/AT* (ISA bus) and compatible computers (Intel386™, Intel486™ or Pentium® processor-based PC platforms). Both boards provide everything required for building integrated, non-SCbus voice solutions, scalable from two to 64 ports.

Software Support

The D/21H and D/41H boards are supported by System Software and SDK packages for Windows 95, Windows NT, MS-DOS, OS/2, and UNIX. These SDKs contain all the documentation, demo code, and tools necessary for developing complex multichannel applications.

Functional Description

The D/21H and D/41H voice processing boards build on the patented 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 by using faster processors and considerably more memory. This dual-processor approach offloads many low-level decision-making tasks from the host computer and thus enables easier development of more powerful applications. This architecture handles real-time events, manages data flow to the host PC for faster system response time, reduces host PC processing demands, processes DTMF and telephony signaling, and frees the DSP to perform signal processing on the incoming call.

Each of the two (D/21H) or four (D/41H) analog loop start interfaces receives analog voice and telephony signaling information from the telephone network (see block diagram). Each 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 also incorporates circuitry that protects against high-voltage spikes and adverse network conditions and lets applications go off-hook any time during ring cadence without damaging the board.

Part of the telephone interface for the D/21H and D/41H boards includes an on-hook audio path that detects Caller ID information. Depending on the level of service offered by the local telephone provider, Caller ID can include the date, time, caller's telephone number, and (in some enhanced Caller ID environments) the name of the person calling. The on-hook audio path can also detect touch tones while the line is on-hook. This capability lets the board operate behind PBXs that require on-hook touch tone detection for their signaling.

Inbound telephony signaling (ring detection and loop current detection) are conditioned by the line interface and routed via a control bus to the control processor. The control processor responds to these signals, informs the application of telephony signaling status, and instructs the line interface to transmit outbound signaling (on-hook/off-hook) to the telephone network.

The audio voice signal from the network is bandpass filtered and conditioned by the line interface and then applied to a COder/DECoder (CODEC) circuit. The CODEC filters, samples, and digitizes the inbound analog audio signal and passes this digitized audio signal to a Motorola* DSP.

Based on Spring Ware firmware loaded in DSP RAM, the DSP performs the following signal analysis and operations on this incoming data:

  • uses AGC to compensate for variations in the level of the incoming audio signal. The D/21H and D/41H boards also include special circuitry to detect and amplify extremely weak line signals due to harsh telephone line conditions or back-to-back local loops often found in 800 service (toll-free) scenarios.
  • applies an adaptive differential pulse code modulation (ADPCM) or pulse code modulation (PCM) 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-frequency tone
  • uses 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 or user 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 machine, 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 popular 11 kHz, 8-bit linear multimedia WAVE format is also supported on the D/21H and D/41H voice boards. Outbound processing is the reverse of inbound processing. The DSP processed speech is transmitted by the control microprocessor to the host PC for disk storage. When playing back a stored file, the microprocessor receives the voice information from the host PC and passes it to the DSP, which converts the file into digitized voice. The DSP sends the digitized voice to the CODEC to be converted into analog voice and then to the line interface for transmission to the telephone network.

The on-board microprocessor controls all operations of the D/21H and D/41H boards via a local bus and interprets and executes commands from the host PC. This microprocessor handles real-time events, manages data flow to the host PC to provide faster system response time, reduces PC host processing demands, processes DTMF and telephony signaling before passing them to the application, and frees the DSP to perform signal processing. Communications between this microprocessor 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 XT/AT bus. All operations are interrupt-driven to meet the demands of real-time systems. All D/21H and D/41H boards installed in the PC share the same interrupt line. When the system is initialized, Spring Ware 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.

Technical Specifications

Number of ports2 (D/21H) or 4 (D/41H)
Max. boards/system16
Analog network interfaceOn-board loop start interface circuits
MicroprocessorIntel® 80C188
Digital signal processorMotorola DSP56002
HOST INTERFACE:
Bus compatibilityIBM PC XT/AT (ISA)
ISA bus speed4 MHz to 12 MHz, 70 nsec back-to-back bus cycle
Shared memory8 KB page, switch selectable on 8 KB boundaries
Base addressesD000h (default), A000h or C000h
Interrupt levelIRQ 2, 3, 4, 5, 7, 10, 11, 12, jumper selectable. One IRQ is shared by all boards (D/21H or D/41H).
TELEPHONE INTERFACE:
Trunk typeLoop start (or ground start for answer only)
Impedance600 ohms nominal
Ring detection25 Vrms min., 15.3 Hz to 68 Hz, 150 Vrms max.
Loop current range20 mA to 120 mA, dc (polarity insensitive)
Crosstalk coupling-70 dB at 3 kHz channel-to-channel
Frequency response300 Hz to 3400 Hz ±3 dB (transmit and receive)
ConnectorTwo RJ-11 type
POWER REQUIREMENTS:
+5 VDC500 mA
+12 VDC35 mA
-12 VDC35 mA
Operating temperature0°C to +50°C
Storage temperature-20°C to +70°C
Humidity8% to 80% noncondensing
Form factorPC XT (ISA); 7.9 in. long, 0.75 in. wide, 3.85 in. high (excluding edge connector)
REGULATORY CERTIFICATIONS:
United StatesFCC part 68 ID#: EBZUSA-65588-VM-E
REN: 1.0B
UL: E143032
CanadaIC CS-03, 885 4452 A
Load number: 5
ULC: E143032
WarrantyLifetime

Spring Ware Firmware 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 -30 dBm results in full scale recording, configurable by parameter†.
Silence detection -40 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 ADPCM @ 6 kHz sampling
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)
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 +0 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 -36 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 3500 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 +0 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 CCITT Q.321
Transmit level Complies with Bellcore LSSGR Sec 6, TR-NWT-000506
Signaling mechanism Complies with Bellcore LSSGR Sec 6, TR-NWT-000506
Dynamic range for detection -25 dBm to -1 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 CCITT Rec E., Suppl #2. Default uses both frequency and cadence detection. Application can select frequency only for faster detection in specific environments.
Ringback detection Default setting designed to detect 83 out of 87 unique ringback tones used in 96 countries as specified by CCITT Rec E., Suppl #2. Uses both frequency and cadence detection.
Positive Voice
Detection™ accuracy >98% based on tests 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-000506
MF digits 0 to 9, KP, ST, ST1, ST2, ST3
Frequency variation ±0.5% of nominal frequency
Rate 10 digits/s max., configurable by parameter†
Level -5 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)
Japan NTT CLIP
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 600 ohm
Message formats ASCII or binary SDM, MDM message content
ANALOG DISPLAY SERVICES INTERFACE (ADSI):
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

** Analog levels: 0 dBm0 corresponds to a level of +3 dBm at tip-ring analog point. Values vary depending on country requirements; contact your Sales Engineer.

† Average speech mandates +16 dB peaks above average and preserves -13 dB valleys below average.

Hardware System Requirements

  • 80386, 80486, or Pentium IBM PC AT (ISA) bus or compatible computer. Operating system hardware requirements vary according to the number of channels being used.

*All company names, products, and services mentioned are the trademarks or registered trademarks of their respective owners


 HOME | PRODUCTS | SUPPORT | COMPANY
 Copyright ® 2002 - AMDEV Communications Corporation
AMDEV Communications Corporation - Your Voice Mail Specialists! Home Products Support Company