Dialogic ProLine/2V
Feature-Rich, Two-Port Voice Processing Board
Features and Benefits
- Audio connectors
allow convenient off-line recording and playback of
system voice prompts
- Electret
microphone input jack allows convenient online recording
of system voice prompts
- Windows*
95 and Windows NT* Telephony API (TAPI) support and
.WAV audio capability
- Caller ID
capability for "screen pop" applications (supports
Bellcore CLASS Protocols)
- Optional
Global DPD pulse-to-tone conversion software lets
you use the ProLine/2V board in countries with limited
touch tone telephone service
- Voice coding
at dynamically selectable data rates (24 Kb/s to 88
Kb/s, selectable on a channel-by-channel basis) provide
optimal tradeoff between disk storage requirements
and voice quality
- Enhanced
telephone circuitry and AGC maintains recording quality
over a wide dynamic range
- Downloadable
Spring Ware signal and call processing firmware provides
easy feature enhancement and field-proven performance
- Perfect Digit
DTMF (touch tone) provides reliable detection during
voice playback - lets callers "type-ahead" through
menus
- Patented
outbound call progress Perfect Call analyzes outgoing
call status quickly and accurately
- Configure
multiple boards in a single PC for easy and cost-effective
system expansion. Build scalable systems from two
to 32 ports.
- C language
application program interfaces (APIs) for MS-DOS*,
Windows 95, and Windows NT
- Third-party
application generators available for rapid application
development
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 Intel® Dialogic® ProLine/2V board with
its compact 2/3 length, XT height footprint, provides
two telephone line interface circuits approved for direct
connection to analog loop start lines. A dual-processor
architecture comprising a digital signal processor (DSP)
and a general-purpose microprocessor handles all telephony
signaling and performs DTMF (touch tone) and audio/voice
signal processing tasks. Multiple ProLine/2V boards
can be installed in a single PC chassis for system expansion
up to 32 ports.
Windows* 95 and Windows NT* include TAPI/WAVE support
which facilitates recording and playback of messages
or system prompts via the ProLine/2V board's audio connectors
and provides a base TAPI platform for Windows 95 and
Windows NT application development. WAVE support increases
your choices when recording and playing back audio files.
You can record voice prompts directly through the ProLine/2V
microphone input jack and play them back using the ProLine/2V
board's WAVE capability. You can also convert audio
from compact disc and CD-ROM sources (with the help
of PC-based utilities) for use in your computer telephony
(CT) applications.
Caller ID capability lets you create applications where
the incoming caller's number can be used to search a
database to create a "screen pop" of information about
the caller. Additionally, you can use Caller ID to provide
access to an extra level of services in a voice mail
or interactive voice response (IVR) system.
The Global Dial Pulse Detection (Global DPD) algorithm
is available for the ProLine/2V board and lets you use
the product in countries that have limited touch tone
telephone service. Offered as a ProLine/2V software
option, the Global DPD algorithm can also be optimized
on a country-by-country basis to provide dial pulse
detection wherever it is used.
The on-board DSP executes downloaded Spring Ware firmware
algorithms to provide variable voice coding at 24 and
32 Kb/s ADPCM, and 48 and 64 Kb/s µ-law PCM. Sampling
rates and coding methods are selectable on a channel-by-channel
basis. Applications may dynamically switch the sampling
rate to optimize data storage or voice quality as the
need arises. Spring Ware 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
(AGC) maintain recorded voice quality even at extremely
low signal levels.
Configurations
The ProLine/2V board shares a common hardware and firmware
architecture with other Intel® Dialogic® voice
boards for maximum flexibility and scalability. Add
features or grow the system while protecting your investment
in hardware and application code. With only minimum
modifications, you can easily port applications to higher
line density platforms.
The ProLine/2V board installs in IBM PC XT/AT* and
compatible computers (Intel386™, Intel486™,
or Pentium® processor-based PC platforms). The ProLine/2V
board provides what you need for building integrated
voice solutions scalable from two to 32 ports.
Software Support
The ProLine/2V board is supported by System Software
and Software Development Kits for MS-DOS, Windows 95,
and Windows NT. These packages contain a set of tools
for developing complex multichannel applications.
Functional Description
The ProLine/2V 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 off-loads many low-level
decision-making tasks from the host computer and makes
it easier to develop 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 two analog loop start telephone line interfaces
on the ProLine/2V board 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 Type 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.
The line interface conditions the inbound telephony
signaling (ring detection and loop current detection)
and routes it 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 CODEC (COder/DECoder) circuit. The CODEC
filters, samples, and digitizes the inbound analog audio
signal and passes this digitized audio signal to a Motorola
DSP.
Part of the board's telephone interface 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 you use the
ProLine/2V board behind PBXs that require on-hook touch
tone detection for their signaling.
The ProLine/2V board also receives and transmits audio
directly on one channel via line-level input and output
jacks or directly into an electret microphone jack.
This interface bypasses the telephony interface and
lets you record prompts. Line-level input can be used
to load prerecorded prompts or messages via line-level
audio devices, such as a cassette tape recorder or compact
disc player. You can use the line-level output to monitor
calls or play out files in a development environment.
The Spring Ware firmware loaded into the DSP RAM provides
the following signal analysis and operations on the
incoming data:
- Automatically controls the gain to compensate for
variations in the level of the incoming audio signal
- 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 tone
- Detects silence 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 the following
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
- call answered (differentiates whether answered by
a person, answering machine, fax machine, or modem)
When recording speech, the DSP can use different digitizing
rates from 24 to 88 Kbps 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
by the control microprocessor to the host PC for disk
storage.
Outbound processing is the reverse of inbound processing.
When playing back a stored file, the microprocessor
receives the voice information from the host PC and
passes it to the DSP that decodes the compressed file.
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 ProLine/2V 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
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 ProLine/2V boards installed
in the PC share the same interrupt line. When the system
is initialized, Spring Ware firmware, which controls
all board operations, is downloaded from the host PC
to the on-board code/data RAM and DSP RAM. Spring Ware
gives the board its intelligence and enables easy feature
enhancement and upgrades.
Technical Specifications
| Number of ports | 2 |
| Max. boards/system | 16 |
| Analog network interface | On-board loop start interface circuits |
| Microprocessor | Intel® 80C188 |
| Digital signal processor | Motorola DSP56002* |
| HOST INTERFACE: |
| Bus compatibility | IBM PC XT/AT (ISA) |
| 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, 3, 4, 5, 7, 9, 10, 11, 12, jumper selectable. One IRQ is shared by all ProLine/2V boards. |
| TELEPHONE INTERFACE: |
| Trunk type | Loop start |
| Impedance | 600 ohms nominal |
| Ring detection | 25 Vrms min., 15.3 Hz to 68 Hz, 150 Vrms max. |
| Loop current range | 20 mA to 120 mA, dc (polarity insensitive) |
| Crosstalk coupling | -70 dB at 3 kHz channel to channel |
| Frequency response | 300 Hz to 3400 Hz ±3 dB (transmit and receive) |
| Connector | Two RJ-11 type |
| AUDIO INTERFACE: |
| Line input impedance | 10 Kohms |
| Line input signal range | -32 dBv to -2 dBv, AC coupled mono or stereo |
| Line input connector | 3.5 mm stereo audio jack |
| Line output impedance | 600 ohms |
| Line output signal range | -32 dBv to -2 dBv, mono |
| Line output connector | 3.5 mm stereo audio jack |
| MICROPHONE INTERFACE: |
| Mic input impedance | 10 Kohms |
| Mic input signal range | -55 dBv to -25 dBv, AC coupled mono or stereo, +5 Vdc phantom power for electret microphones only |
| Mic input connector | 3.5 mm microphone jack |
| POWER REQUIREMENTS: |
| +5 VDC | 500 mA |
| +12 VDC | 35 mA |
| -12 VDC | 35 mA |
| Operating temperature | 0°C to +50°C |
| Storage temperature | -20°C to +70°C |
| Humidity | 8% to 80% noncondensing |
| Form factor | PC XT (ISA); 7.9 in. long, 0.75 in. wide, 3.85 in. high (excluding edge connector) |
| REGULATORY CERTIFICATIONS: |
| United States | FCC part 68 ID#: EBZUSA-65588-VM-E
REN: 1.0B
UL: E143032 |
| Canada | IC CS-03, 885 4452 A
Load number: 5
ULC: E143032 |
| WARRANTY: | Lifetime |
Spring Ware Firmware Technical Specifications
| AUDIO SIGNAL: |
| Receive range | -50 dBm to -3 dBm nominal for average speech signals**, configurable by parameter† |
| AGC | 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 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 +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 Hz 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. |
| 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.
| | 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) |
| Modem standard | Bell 202 or V.23, serial 1200 b/s (simplex FSK signaling) |
| Receive sensitivity | -48 dBm (-50 dBv) 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 (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 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.
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