VFX/40ESC plus  International SCSA 4-PORT voice/ FAX BOARD With caller ID

FEATURES & BENEFITS

• Next generation of the VFX/40ESC with additional DSP RAM that provides expanded voice processing capability:
• International caller ID 
• Pulse-to-tone conversion software-Global Dial Pulse Detection (DPD) 
• Supports software-based speech technologies, included TextTalk™ TTS software 
• WAVE file support at 11 kHz linear PCM 
• PBXpert™, a free utility that simplifies switch integration
• Combines baseline VFX/40ESC fax performance with enhanced fax features including transmission and receive MH, MR and MMR coding
• 14,400 b/s (v.17) with ITU-T Group 3 and ETSI NET/30 compliance ensures operation with fax machines worldwide, provides seamless voice + fax integration at hardware and software levels, and supports full compliance of ITU-T (T.4, T.30) specifications
• 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 for undistorted images while sending and receiving 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 artwork can be combined on same page permitting mixing of coding methods
• Support of TIFF/F, MH data formats provides compression of fax data
• Multiple polling modes can automatically collect documents from remote devices and 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 interfaces (APIs) for MS-DOS®, UNIX®, Windows NT®, Windows® 95 shorten your development cycle so you can get your applications to market faster
• Windows TAPI/WAVE support (including 11 kHz, 8-bit file format)
• Four-channel design supports low- to medium-density voice/fax systems scalable from 4 ports to 24 ports and beyond
• Modified Modified READ (MMR) reduces transmission and receive time when connected to fax units with MMR capability and allows more efficient use of disk storage space and network bandwidth
• Error Correction Mode (ECM) ensures error-free data reception and transmission when connected to fax units with ECM capability

• Unified messaging
• Fax mail
• Integrated voice mail and fax mail
• Integrated voice/fax response
• Single call fax-on-demand
• Remote database transactions
• Fax confirmation
• Fax store-and-forward
• Fax broadcast
• Fax notification

• International caller ID (CLASS, CLIP, ACLIP, and Japan CID) 
• Global Dial Pulse Detection (DPD) 
• Speech recognition 
• Text-to-speech 
• WAVE file support at 11 kHz linear PCM

 The VFX/40ESC plus 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 shared DSPs provides efficient on-demand voice and fax image processing. Consistent voice and fax commands ensure transportability between operating systems.

 The VFX/40ESC plus board incorporates an SCbus or PEB interface for compatibility with a wide range of call processing products.

 The VFX/40ESC plus is built on the 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, incorporating 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

 The VFX/40ESC plus board incorporates an SCbus (or PEB) connector for compatibility with a wide range of call processing products.

 The VFX/40ESC plus board installs in IBM® PC AT (ISA bus) and compatible computers (80386, 80486, or Pentium™-based PC platforms). The board occupies a single expansion slot and up to 16 boards can be configured in a system with each board sharing the same interrupt level. 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.
 
 

SOFTWARE SUPPORT

 The VFX/40ESC plus is supported by Dialogic System Software and Software Development Kits for MS-DOS, UNIX, Windows 95 and Windows NT. These packages contain a set of tools for developing complex multi-channel applications.

 The Dialogic Fax Development Package supports all Dialogic 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 voice 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/40ESC plus 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/40ESC plus 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/40ESC plus provides call processing for either analog loop start telephony signals input from the telephone network or resource sharing for digital signaling and digital voice information input via the SCbus or PEB. 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 the analog audio bus to a fax channel duplexer. The duplexer enables the fax modem to alternately transmit and receive analog audio (modem) signals over the single channel analog audio bus.

 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

• 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

• 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.

 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/40ESC plus via the SCbus interface. These signals are managed by an SC2000 chip that 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 or PEB. 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 or 24 for PEB in T-1 mode, 32 in E-1). 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. In PEB configuration, the local and external SCbus operates at 1.544 or 2.048 Mb/s.

 The on-board control microprocessor controls all operations of the VFX/40ESC plus board via a local control 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, filters 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 dual-port 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 ISA 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 high-speed SRAM to control all board operations. This downloadable firmware gives the board all of its intelligence and enables easy feature enhancement and upgrades.

 The Board Locator Technology™ circuit operates in conjunction with a rotary switch to determine and set non-conflicting slot and IRQ interrupt-level parameters. This feature eliminates the need to set confusing jumpers or DIP switches.

TECHNICAL SPECIFICATIONS*
 
 

Number of ports	4 channels of call processing and facsimile
Max. boards/system	16
Analog network interface	On-board analog loop start/ground start (answer only) interface
Resource sharing bus	SCbus or PEB
Fax modem	Rockwell R144EFXL
Fax control microprocessor	Intel 80C186 @ 16 MHz
D/41ESC control microprocessor	Intel 80C186 @ 16 MHz
Call processing
digital signal processor	Motorola DSP56001 @ 33 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 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), MMR (Modified Modified READ), with or without ECM
Receive Data Modes	MH (Modified Huffman), MR (Modified READ), MMR (Modified Modified READ), 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 & 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	IEEE P996 ISA compatible (IBM PC XT/AT)
Bus speed	12.5 MHz maximum
Bus mode	Automatically configures to 8- or 16-bit transfer mode
Shared memory	8 Kbytes page
Base addresses	8000h to E800h, on 32 K boundaries. All VFX/40ESC plus boards share the same base address. Shared memory is page mapped in/out dynamically as needed.
Interrupt level	IRQ 2/9, 3, 4, 5, 6, 7, 10, 11, 12, 14, 15, software selectable. One IRQ line must be shared by all VFX/40ESC plus boards.
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	2.5 A max.
+12 VDC	175 mA max.
-12 VDC	125 mA max.
Operating temperature	0°C to +50°C
Storage temperature	-20°C to +70°C
Humidity	8% to 80% non-condensing
Form factor	PC AT, 13.34 in. long. 0.79 in. wide, 4.8 in. high
SAFETY & EMI CERTIFICATIONS:
United States	FCC part 68 ID #: EBZUSA-75385-VM-T; FCC Part 15, Class A
	UL: E96804
Canada	DOC: 885-5542A
	CSA: LR-84340
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	ADPCM @ 6 kHz sampling
32 Kb/s	ADPCM @ 8 kHz sampling
48 Kb/s	æ-law PCM @ 6 kHz sampling
64 Kb/s	æ-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 IBM PC AT (ISA) bus or compatible computer. Operating system hardware requirements vary according to the number of channels being used.