Dialogic 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
- Downloadable
signal and call processing firmware facilitates feature
enhancement and provides field-proven performance
based on over 3 million installed ports
- C language
application program interfaces (APIs) for MS-DOS*,
Windows 95*, Windows NT*, 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 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
- Perfect Digit
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
PBX Expert and PBX Expert/32, utilities that simplify
switch integration
- Lifetime
warranty
Applications
- 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
The DIALOG/4 board, with its half-size footprint, is
an ideal solution for computer telephony installations
that cannot take full-size voice boards. It provides
four telephone line interface circuits that are approved
for direct connection to analog loop start lines. A
dual-processor architecture, consisting of a DSP (Digital
Signal Processor) and a general-purpose microprocessor,
handles all telephony signaling and performs DTMF (touch
tone) and audio/voice signal processing tasks. Multiple
DIALOG/4 boards can be installed in a single PC chassis
enabling system expansion up to 64 ports.
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. Downloaded firmware algorithms
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. Downloaded firmware algorithms
also provide 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 touch tones
- 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.
Configurations
The DIALOG/4 board shares a common hardware and firmware
architecture with other 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 PC XT/AT* (ISA bus)
and compatible computers (Intel386, Intel486,
or Pentium® processor-based PC platforms). The DIALOG/4
board provides everything required for building integrated
voice solutions scalable from four ports to 64 ports.

Software Support
The DIALOG/4 is supported by System Software and SDK
for MS-DOS*, Windows NT*, Windows 95*, OS/2*, and UNIX*.
These packages contain a set of tools for developing
complex multichannel applications.
Functional Description
The DIALOG/4 board uses a 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 Spring Ware 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
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 can be selected on a channel-by-channel
basis and can be changed each time a record or play
function is initiated. 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 replaying 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 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 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, 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 |
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 -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: |
| 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: |
| 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 back
detection |
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
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 |
| 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 Intel Sales Engineer.
Hardware System Requirements
Intel386, Intel486®, 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.
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