With the open access framework, the SHT series is a powerful and versatile analog telephony interface platform with on-board DSP resources to process fax, conferencing and voice-processing. It is designed to provide developers of highly reliable telecommunications systems with an optimized, cost-effective solution that saves cPCI/PCI/USB slots over traditional designs by combining 2~16 analog Loopstart/Subscriberloop connections with DSP-based enhancements such as fax, conference and voice processing capabilities.
Rooted in all-in-one architecture, hardware design of SHT series adopts modular structure to deliver custom products and high application flexibility for developers and integrators. SHT series boards can be installed with specific modules for station, trunk and recording functions and have high analog interface capability to fit to diverse applications needs. The "USB Box" in the SHT Series is handy for setting up 2~4 port systems for application developers and household users.
The SHT series is also supported by in-house unified API that is operating systemindependent, which makes it highly versatile. With a single, protocol-independent API, programmers can easily and quickly develop applications that run on a variety of telecommunication networks simply by executing protocols on the SHT DSP resources. The in-house unified API unifies application development across all CTI and call recording products.
KEY FEATURES AND BENEFITS
2 ~ 16 Analog Ports
Configurable 2 ~ 16 analog Loopstart/Subscriber interfaces with call control and switching in cPCI/PCI/USB slot.
Independent conference resource in each port, support conference recording/monitoring;
DTMF filtering to eliminate DTMF signal in confere nce; configured as interactive teleconferencing without limit on the number of conferees and conferences of high-density
Universal port capability supports conference, fax, voice-processing and built-in SHT Series functions to ensure scalability, compatibility, and high performance to leverage developer time and application investment.
In-House Unified API
Protocol-independent API minimizes system development and deployment efforts and feature-rich software development kits support for Windows® and Linux®.
Handle call connection with special algorithm for dialing handing and recognizing caller ID in FSK and DTMF modes.
Real-time monitoring by audio jack for any of the channels in teleconferencing without need of an additional channel.
Built-in Voice-Processing Resource
Support: A-law, µ-law, record/playback of Windows WAV file, programmable tone generation/ reception/ detection for DTMF, FSK receiver/transmitter, adaptive echo cancellation and barge-in functionality.
Cover all IVR voice functionality, 4, 8 or 12 ports of DSP based Group 3 fax resources available; transmission speed up to 14400bps; support .TIF file, fax transmission of all kinds of printable files, easily viewed and printed; faxing capacity is software configurable.
Rich DSP Resources
Used to implement call control, switching, and enhancements such as voice playback/record, DTMF/MF, call progress functions , echo cancellation, fax, conferencing; DSP resources are efficiently managed to minimize host overhead and maximize host processing time.
Full-speed H.100/H.110 bus with 4,096 timeslots to support interoperability with other boards in openarchitecture, high-capacity systems.
|PRODUCT SPECIFICATIONSPRODUCT MODELS
SHT-8B/PCI (PCI interface, 2 to 8 ports configurable)
SHT-16B-CT/PCI (PCI interface, 2 to 16 ports configur able,H.100)
SHT-16B-CT/cPCI (CompactPCI interface, 2 to 16 ports configurable, H.110)
SHT-4B/USB (USB interface, 2 to 4 ports configurable)
SHT-2B/USB (USB interface, 2 ports)
Ring detection: 30Vrms (min), 16 to 68Hz
Ringer equivalence number: < 0.2/programmable
Voltage detection: -62V to +62V Step=0.5V
Caller ID: FSK/DTMF line connector: RJ11(4-8port) /RJ45(16port)
Feed voltage: -48V Max current: 35 mA per channel
Tone generator: programmable calling tone generation
Ring generator: 55Vrms 50/60Hz Max user line length: 5.5 km
|HARDWARE SYSTEM REQUIREMENTS
Pentium 3 or equivalent
1 GHz or better PCI Bus, Rev. 2.2,
33Mhz, 32Bit/cPCI/USB 2.0
Operating temperature: 0? to +55?
Storage temperature: -20? to +85?
Humidity: 8% to 90% non-condensing
Storage humidity: 8% to 90% non-condensing
Max boards per system: any combination up to 16
Max ports per system: any combination up to 256
Resource sharing BUS: H.100/H.110( for SHT-16B boards)
16Bit PCM 128Kbps
8Bit PCM 64Kbps
Audio connector: f3.5mm headphone jack
Output impedance: 32Ohms
Output power: 50mW
Play volume: programmable
SAFETY AND CERTIFICATIONS
|SAFETY AND CERTIFICATIONS
Emission: FCC Part 15 Class A & Class B
Emission: AS/NZS CISPR 22:2004 Class B
Immunity: AS/NZS CISPR 24:2002
Estimated MTBF(PCI bus interface): 158,000 hours per Bellcore Method I
Estimated MTBF(cPCI bus interface): 156,800 hours per Bellcore Method I
Receive range: -68 dBm to + 3 dBm
Input gain control: +20 to -20 dB
Silence detection: programmable from API
Automatic Gain Control (AGC): programmable from API
Activity detection: programmable from API
Frequency response: 300-3400Hz(+/- 3dB)
Signal/Noise ratio: >=42dB
Idle channel noise: <=60dB
Crosstalk coupling: <=-70 dB
|DTMF TONE DETECTION
DTMF digits: 0 - 9, *, #, A, B, C, D
Dynamic range: -38 dBm to 0 dBm
Minimum tone detection: 40 ms
Fax tone detection: 1100/2200Hz
Calling process tone detection: programmable*4
DTMF digits 0 - 9, *, #, A, B, C, D
Frequency variation: less than 1 Hz
Rate: API programmable
Duration: API programmable
ON-BOARD FUNCTIONAL MODULES AVAILABLE
offers seven types of modules for SHT series board, including MT, MU, MTU, MR, MR+, MTR+ and MTR. A single module interfaces to an on-board socket and corresponds to two channels. For instance, SHT-16B-CT/PCI is designed with 8 on-board sockets (corresponding to 16 ports), so customers can freely configure 8 modules on the board from the following module types:
MT (Trunk module)
MT is the module that connects to the telephone network (analog loop start lines). It can carry out a series of work such as tone detection, playback/record, caller-ID/called-ID retrieval, polarity reversal detection and DTMF acquisition. Moreover, MT can check the status of telephone wires for its line voltage detection ability. With MT on the board, inbound and outbound calling, teleconferencing, voicemail, IVR and the like solutions can be implemented.
MU (Station module)
MU internally interfaces to the analog telephone set to perform a series of work such as ringing signal, caller-ID (DTMF/FSK), playback/record and DTMF detection. MU works well in solutions such as operator service.
MTU (Trunk/station combined module)
An MTU also corresponds to 2 ports, with one channel performing MT module's function and the other MU's. Besides all the functions inherited from MT and MU, this module offers an added advantage, i.e. phone calls will not be interrupted even when power supply is out thanks to the internal physical connection between the MT and MU.
MR (High-impedance record module)
MR is used for recording inbound and outbound calls and can carry out a series of work such as DTMF detection, tone detection, recording voice and signaling data (caller-ID, called-ID), etc. With MR on the board, a perfect recording solution can be implemented.
MR+ (Microphone record module)
MR+ is the module that connects to a microphone for recording voices and signals through the microphone. MR+ enhances and expands the diversity and flexibility in recording solutions.
MTR+ (Trunk/Microphone record combined module)
An MTU module corresponds to 2 ports, with one performing MT's function and the other MR+'s.
MTR (Trunk/record combined module)
An MTR module corresponds to 2 ports, with one performing MT's function and the other MR's