DIALOG/4 Half-Size, Four-Port Voice Processing Board

Features and Benefits

• Four independent voice processing ports in a single, half-size PC ISA slot supporting low- to medium-density voice systems

 
• Dialogic downloadable signal and call processing firmware, SpringWare(tm), facilitates feature enhancement and provides field-proven performance based on over three million installed ports

 
• C language application program interfaces (APIs) for MS-DOS(r), Windows(r) 95, Windows NT(r), OS/2(r), and UNIX(r) shorten your development cycle so you can get your applications to market faster

 
• Application generators available from third-party providers

 
• Configure multiple DIALOG/4(tm) boards in a single PC for easy and cost effective system expansion, and to build scalable systems from 4 to 64 ports

 
• 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

 
• Enhanced telephone circuitry and automatic gain control maintains recording quality over a wide dynamic range

 
• PerfectDigit(tm) DTMF (touchtone) provides reliable detection during voice playback - allows callers to "type-ahead" through menus

 
• Patented outbound call progress analyzes outgoing call status quickly and accurately

 
• Supports PBXpert(tm) and PBXpert/32(tm), free utilities that simplify switch integration

 
• Lifetime warranty

• Voice mail/voice messaging
• Interactive voice response
• Audiotex
• Inbound and outbound telemarketing
• Operator services
• Dictation
• Auto dialers
• Telecomputing servers
• Notification systems
• On-line data entry/query

 

 Dialogic voice products offer a rich set of advanced features, including state-of-the-art DSP technology and signal processing algorithms, for building the core of any computer telephony system. With industry-standard ISA bus expansion boards and a variety of channel densities to choose from, you can integrate Dialogic voice products easily into exactly the type of system you require at a price and performance level unmatched in the computer telephony industry.

 Downloaded firmware algorithms, SpringWare(tm), executed by the on-board DSP, provide voice coding at 24 and 32 Kb/s ADPCM, and 48 and 64 Kb/s PCM. Sampling rates and coding methods are selectable on a channel-by-channel basis. Applications may dynamically switch sampling rate and coding method to optimize data storage or voice quality as the need arises. SpringWare also provides reliable DTMF detection, DTMF cut-through, and talk off/play off suppression over a wide variety of telephone line conditions. Enhanced telephone circuit design and automatic gain control maintains recorded voice quality even at extremely low signal levels.

 The DIALOG/4 voice board

• connects directly to the telephone line
• automatically answers calls
• detects touchtones
• plays voice messages to a caller
• digitizes, compresses, and records voice signals
• places outbound calls and automatically reports the results all in real time on four independent channels.

 The DIALOG/4 board shares a common hardware and firmware architecture with other Dialogic voice boards for maximum flexibility and scalability. You can easily add new features and/or expand the size of the system while protecting your original investment in hardware and application code. Applications can be ported to lower or higher line-density platforms with minimal modifications.

 The DIALOG/4 board installs in IBM(r) PC XT(r)/AT(r) (ISA bus) and compatible computers (80386, 80486, or Pentium(tm)-based PC platforms).The DIALOG/4 board provides everything required for building integrated voice solutions scalable from 4 ports to 64 ports.
 
 

Software Support

 The DIALOG/4 is supported by Dialogic System Software and SDK for MS-DOS(r), Windows NT(r), Windows(r) 95, OS/2(r), and UNIX(r). These packages contain a set of tools for developing complex multichannel applications.

 Functional Description

 The DIALOG/4 board 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 80C188 control microprocessor. This dual processor approach offloads many low-level decision-making tasks from the host computer enabling 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 four loop start telephone line interfaces on the DIALOG/4 board receives analog voice and telephony signaling information from the telephone network (see block diagram). Each 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 allowing applications to go off-hook any time during ring cadence without damaging the board.
 
 

Inbound telephony signaling (ring 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 sent through a bandpass filter, conditioned by the line interface, and then applied to a CODEC (COder/DECoder) circuit. The CODEC filters, samples, and digitizes the inbound analog audio signal and passes the digitized signal to a Motorola DSP.

 Based on SpringWare firmware loaded in DSP RAM, the DSP performs the following signal analysis and operations on this 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-frequency tone
• silence detection to determine whether the line is quiet and the caller is not responding

• 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

• 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.

 The on-board microprocessor controls all operations of the DIALOG/4 board 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 signals 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 increasing 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 DIALOG/4 boards installed in the PC share the same interrupt line. When the system is initialized, SpringWare firmware to control all board operations is downloaded from the host PC to the on-board code/data RAM and DSP RAM. This downloadable firmware gives the board all of its intelligence and enables easy feature enhancement and upgrades.

 Technical Specifications*
 
 

Number of ports	4
Max. boards/system	16
Analog network interface	On-board loop start interface circuits
Microprocessor	80C188
Digital signal processor	Motorola DSP56001
HOST INTERFACE:
Bus compatibility	IBM PC XT/AT (ISA)
Bus speed	4 to 12 MHz, 70 nsec back-to-back bus cycle
Shared memory	8 KB page, switch selectable on 8 KB boundaries
Base addresses	D000h (default), A000h or C000h
Interrupt level	IRQ 2 to IRQ 7 jumper selectable; one IRQ is shared by all DIALOG/4 boards
TELEPHONE INTERFACE‡:
Trunk type	Loop start (or ground start for answer only)
Impedance	600 ohms nominal
Ring detection	40 Vrms min; 15.3 to 68 Hz, 130 Vrms max.
Loop current range	20 to 120 mA, dc (polarity insensitive)
Receive signal/noise ratio	70 dB, referenced to -15 dBm
Crosstalk coupling	-70 dB at 1 kHz channel to channel
Frequency response	300 Hz to 3400 Hz ±3 dB (transmit and receive)
Connector	Two RJ-14 type
POWER REQUIREMENTS:
+5 VDC	.75 A
+12 VDC	40 mA
-12 VDC	40 mA
Operating temperature	0°C to +50°C
Storage temperature	-20°C to +70°C
Humidity	8% to 80% noncondensing
Form factor	PC (ISA) half size: 7 in. long, 0.652 in. wide, 4.5 in. high (excluding edge connector)
REGULATORY CERTIFICATIONS:
United States	FCC part 68 ID#: EBUSA-65588-VM-E UL: 143032
Canada	DOC: 885-4452A ULC: 143032
Warranty	Lifetime

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 and legal limit cap
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
DTMF TONE DETECTION:
DTMF digits	0 to 9, *, #, A, B, C, D per Bellcore LSSGR Sec 6
Dynamic range	Default set to -36 dBm to -3 dBm per tone, configurable by parameter‡
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 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(tm):
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 125 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(tm):
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-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 CCITT Rec. E., Suppl. #2; default uses both frequency and cadence detection; application can select frequency only for faster detection in specific environments
Ring backdetection	Default setting designed to detect 83 out of 87 unique ring back tones used in 96 countries as specified by CCITT Rec. E., Suppl. #2; uses both frequency and cadence detection
Positive Voice
Detection(tm) 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(tm) 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
Dialtone 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
Pulse rate	10 pulses/s, nominal, configurable by parameter‡
Break ratio	60% nominal, configurable by parameter‡
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.
** Average speech mandates +16 dB peaks above average and preserves -13 dB valleys below average. 
‡ 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 IBM PC AT (ISA) bus or compatible computer. Operating system hardware requirements vary according to the number of channels being used.