Abstract: According to another example, a wearable medical device controller is provided. The device controller includes a memory and a processor coupled to the memory. The processor is configured to determine a correlation between a phenomenon identifiable by the wearable medical device controller and at least one response pattern associated with a patient and store, responsive to detecting the correlation, an adaptation path to address the at least one response pattern, the adaptation path specifying an adaptation of at least one characteristic of an alarm. The at least one response pattern may include a plurality of response patterns and the adaptation path may reflect adaptations made to address at least some of the plurality of response patterns.

Abstract: A consumer electronics device is arranged for outputting a hearing loss compensated signal. The consumer electronics device comprises an operating system whereon at least one application can be run that yields a sound output signal. Wherein the consumer electronics device further comprises: a first software module adapted for rerouting the sound output signal, and a second software module adapted for receiving from the first software module the rerouted sound output signal, for performing hearing loss compensation on the rerouted sound output signal and for outputting the hearing loss compensated signal.

Abstract: Provided is an amplification system for a public safety service. The amplification system may be installed to cover a radio-shadow zone, and stably maintain the service despite a distortion in a wireless environment and an excessive input of a spurious interference signal.

Abstract: A hearing device includes: a processing unit configured to compensate for hearing loss of a user of the hearing device; and a memory unit; wherein the processing unit is configured to: obtain an access right certificate, the access right certificate comprising an access right identifier, verify the access right certificate, and if the access right certificate is verified, provide an access right according to the access right identifier.

Abstract: In a multiband compressor 100, a level calculation unit 121 calculates a signal level inputted for each of bands, a gain calculation unit 122 calculates a gain value from the calculated signal level, and a gain limitation unit 130 limits a gain value by comparison with a gain value of the other band in a compressor for each band. With this configuration, provided is a multiband compressor capable of achieving a balance between the quality of sound and the effect of enhancing the sound level at a high level.

Abstract: A network interface device may include an I/O unit configured to receive data packets from a packet network. A processing unit may be in communication with the I/O unit and be configured to identify data packets including data representative of DTMF signals, convert the data in the data packets into analog DTMF tone signals, and determine if power of the analog DTMF tone signals is below a threshold value. A digital to analog converter may be in communication with the processing unit and be configured to generate the analog DTMF tone signals based on the data in the data packets. A DTMF amplifier may be in communication with the processing unit and the digital to analog converter, where the processing unit may cause the DTMF amplifier to amplify the power of the analog DTMF tone signals in response to determining that the power of the analog DTMF tone signals are below the threshold value.

Abstract: Distortions of both amplitude and phase along a transmission line are compensated for by a trace canceller inserted between a transmitter and a receiver. The trace canceller has an equalizer that compensates for a trace length between the transmitter and the trace canceller. A variable gain amplifier between the equalizer and an output buffer has its gain controlled by an automatic gain control circuit that compares low-frequency swings of the input and output of the trace canceller. The gain of the variable gain amplifier is reduced to prevent the output buffer from saturating and clipping peak voltages on its output. Thus both the variable gain amplifier and the output buffer remain in the linear region. Training pulses from the transmitter are passed through the trace canceller without clipping of peak voltages, allowing the transmitter and receiver to adjust transmission parameters to best match the transmission line.

Abstract: A device for continuous time quantization of an input signal, in order to supply a continuous time output signal that is quantized as two bits, the device including: an electronic circuit, designed to supply a first bit of the output signal called the sign bit which at any time takes a first value when the input signal is positive and a second value when the input signal is negative, and an envelope analysis circuit designed to supply a second bit of the output signal called the envelope variation bit which at any time takes a first value, called high value, when an envelope signal of the input signal is increasing, and a second value, called low value, when the envelope signal is decreasing.

Abstract: The invention includes a method and apparatus for transmitting packets in a network comprising a plurality of nodes. A method includes encapsulating a packet intended for at least one of the nodes where the packet is encapsulated using a bit set, comparing the bit set of the packet and a bit mask of an interface, and forwarding the packet over the interface in response to a determination that the comparison of the bit set and the bit mask indicates that one of the nodes for which the packet is intended is the same as one of the nodes available from the interface. The bit set and bit mask each include bit positions associated with the respective plurality of nodes. The bit positions of the bit set and bit mask identify each of the at least one of the nodes for which the packet is intended and each of the nodes available from the interface, respectively.

Abstract: A DSL line conditioner achieves high performance sufficient to simultaneously support video, voice and data signals on standard telephone twisted pair lines over substantially greater distances than are currently available. The line conditioner automatically adjusts and sets the upstream and downstream preamplifier gains and attenuations according to the actual degradation imposed upon the upstream and downstream signals by the twisted pair line in which the line conditioner is used. The line conditioner achieves high performance by optimizing the signal-to-noise ratio and signal quality of DSL signals, and has a low power budget that enables it to operate by deriving power from the twisted pair lines over which it is used.

Abstract: In accordance with the present invention, a loop extender for improving the transmission of digital subscriber line (DSL) signals over a local loop is disclosed. The loop extender includes selectable line termination and equalization (SLTE) DSL amplification circuitry capacitively coupled to the local loop via bypass relay switches, a plain old telephone service (POTS) loading coil adapted to be coupled to the local loop for improving transmission of POTS band signals over the local loop, and a diagnostic/control unit (DCU) coupled to the local loop for receiving and processing control signals from a central office, coupled to the bypass relay switches via a bypass relay for controlling the bypass relay switches, and coupled to the SLTE DSL amplification circuitry via a plurality of switch control lines for controlling the SLTE DSL amplification circuitry.

Abstract: The communications system has a transmission network for transmitting useful data and control data. Call numbers are input at a subscriber terminal. A subscriber line unit with an adjustable transmission characteristic is connected between the subscriber terminal and the transmission network. A recognition unit connected to the transmission network recognizes particular call number patterns and outputs particular control signals corresponding to call number patterns. A control unit connected between the recognition unit and the subscriber line unit sets the transmission characteristic of the subscriber line unit depending on the control signals provided by the recognition unit.

Abstract: There are provided a voice switching system and a voice switching method which are capable of carrying out highly accurate voice detection and performing a switching action properly even in the presence of an unstationary noise or under circumstances where the S/N ratio is low.

Abstract: An inter-network line level adjustment system includes a switch (SW1) and a processing unit (24) for identifying calls between a remote telephone network and a local telephone network. Echo cancellers (20) and/or speech coders (22) collect line level data associated with the identified calls. The processing unit (24) determines statistical properties of the collected line level data. Finally, the processing unit (24) instructs the echo cancellers (20) and/or speech coders (22) to adjust, based on the determined statistical properties, line levels of future calls between the remote telephone network and the local telephone network to a desired line level.

Abstract: For signal level adjustment of a multitone signal received over a transmission line (1) during a training sequence in a DSL system, a programmable high-pass filter (2) is connected to the transmission line (1). A programmable gain control unit (4) is connected between the filter (2) and an A/D converter (5). A digital signal processor (6) is connected with its inputs to the A/D converter (5), with control outputs (7, 9) to control inputs (8, 10) of the filter (2), and with control outputs (11, 13) to control inputs (12, 14) of the gain control unit (4). The processor (6) detects signal level differences between the highest and the lowest tone of the multitone signal and programs the filter (2) to make the signal levels of the highest and the lowest tone equal.

Abstract: A method and apparatus for customizing a device based on a frequency response for a hearing-impaired user are disclosed. The device receives an identifier associated with a hearing-impaired user and retrieves a frequency filter associated with the hearing-impaired user from a database by providing the identifier. The device applies the frequency filter to audio signals in a call session conducted over a network for presentation to the hearing-impaired user at the device.

Abstract: A voice signal processing system with multiple parallel control paths, each of which address different problems, such as the high peak-to-RMS signal ratios characteristic of speech, wide variations in RMS speech levels, and high background noise levels. Different families of input-output control curves are used simultaneously to achieve efficient peak limiting and dynamic range compression as well as low-level dynamic expansion to prevent excessive amplification of background noise. In addition, a delay in the audio path relative to the control path makes it possible to employ an effective look-ahead in the control path, with FIR filtering smoothing-matched to the look-ahead. Digital domain peak interpolators are used for estimating the peaks of the input signal in the continuous time domain.

Abstract: A system and method for automatic gain and slope equalization for provisioning telephone circuits is provided. A first unit is installed at the company end of a copper pair transmission line. A second unit is installed between customer premises equipment (CPE) and the copper pair transmission line. The second unit is programmed to disconnect the CPE, and to transmit a test signal, preferably comprising a square wave with a known amplitude and fundamental frequency of 1 kHz. The signal is received at the first unit, and the fundamental frequency and odd harmonic frequencies are analyzed. Based on the analysis, the first unit thereafter compensates for impairments in the transmission line. The test procedure can be initiated from either the first unit or the second unit. The first unit is programmed to produce a command sequence, preferably comprising a series of DC voltages applied to the tip and ring leads, and received at the second unit.

Abstract: Systems and methods are described for asymmetric digital subscriber loops.

Abstract: A method and apparatus is provided for improving at least one gain bandwidth path. At least one signal being transmitted is monitored. A gain/bandwidth control process is performed upon the monitoring of the signal. The apparatus taught by the present invention comprises: a first circuit portion capable of driving a signal onto a subscriber line; and a second circuit portion electrically coupled with the first circuit portion, wherein the second circuit portion is capable of separating a plurality of signal paths based upon at least one characteristic of the signal path for applying an appropriate gain factor upon the signal path.

Abstract: In a data communications system, a first node transmits an output digital signal to a second node through a transmission line and receives an input digital signal from the second node through a reception line, a signal processing amplification block of the data communications system compensates an attenuation in the input digital signal and prevents a crosstalk between the transmission line and the reception line, and a regulating block for preventing a crosstalk receives a branch signal and generates a control signal based on a capacity of the branch signal.

Abstract: An improved, more intelligent network interface unit capable of processing signals received from a high speed transmission media and outputting the received signals without substantial amplification or reduction. The network interface unit comprises a regenerator circuit which detects T1 signals from said high-speed transmission media and outputs a digital stream to a serial data processor for processing via a micro-controller. A wave shaper circuit receives the output from the serial data processor and, in conjunction with the micro-controller and a level detector circuit, regenerates signals having substantially the same wave shape and amplitude as the original T1 signals received from said transmission media.

Abstract: A method and system for automatic gain control is provided. The automatic gain control may be used, for example, with a microphone and voice codec of a speakerphone telephone. In an exemplary embodiment, an amplification gain lookup table stores the gain values. The power of the input signal is estimated and the appropriate gain value is selected as a function of the estimated power of the input signal. The gain value is applied to the input signal to provide an automatic gain controlled output signal. In another embodiment, the gain lookup table may be adapted to compensate for the non-linearity of the microphone and voice codec of the speakerphone. In this alternate embodiment, the output power of the automatic gain controlled signal is estimated and compared to a reference signal to generate an error signal. The error signal may be scaled and used to update and dynamically adapt the gain values in the gain lookup table.

Abstract: There are provided a voice switching system and a voice switching method which are capable of carrying out highly accurate voice detection and performing a switching action properly even in the presence of an unstationary noise or under circumstances where the S/N ratio is low.

Abstract: A user-programmable bass/treble control circuit is embedded within a conventional remotely located codec. A subscriber station set is capable of transmitting out-of-voiceband “control” tones to increase/decrease bass and treble. A tone detector within the codec recognizes and removes these control tones, forwarding them to a programmable equalization filter inserted in the transmit signal path in the codec. The equalization filter will adjust the bass and treble of the to-be-transmitted signal in accordance with these received control tones. Advantageously, the “listener” at the other end will not hear the tones if the tone detector is inserted before the receiver. The user may adjust the tones during the entire conversation and the filter may be “reset” at call completion.

Abstract: An improved, more intelligent network interface unit capable of processing signals received from a high speed transmission media and outputting the received signals without substantial amplification or reduction. The network interface unit comprises a regenerator circuit which detects T1 signals from said high-speed transmission media and outputs a digital stream to a serial data processor for processing via a micro-controller. A wave shaper circuit receives the output from the serial data processor and, in conjunction with the micro-controller and a level detector circuit, regenerates signals having substantially the same wave shape and amplitude as the original T1 signals received from said transmission media.

Abstract: The present invention customizes a network connection to customer premises equipment using a calibration signal. A network signal processor emits a tag signal indicating that a signal processing platform is in use. Customer premises equipment (CPE) issues a calibration signal to the network signal processor based on detection of the tag signal. The customer premises equipment outputs the calibration signal to the network signal processor which measures the signal loss, adjusts the gain mapping and issues a calibration signal to the customer premises equipment. The customer premises equipment then estimates spectral shaping based on the calibration signal from the network signal processor and equalizes the receive path so that the signal processing platform is detected and implemented. Embodiments of the present invention are illustrative of a more general concept of the use of CPE as an agent in telephone network maintenance.

Abstract: Different hearing impairment compensation parameters are stored in a searchable attribute database indexed by the telephone numbers of hearing impaired users. Calls made to or from the stored telephone numbers have incoming voice samples redirected to signal processing resources that perform customized hearing compensation. The compensation parameters are downloaded from the attribute database to the chosen signal processing resource at call setup time. The signal processing resource then compensates the audio signals of the telephone call for specific user hearing impairment using the downloaded compensation parameters.

Abstract: A user-authentication system includes an authorized user device (AUD) for transmission of dtmf tones to a user-verification system (UVS) over a telephone network, where user-verification is performed and authorized users are granted access to a desired service. In a memory module of the AUD, a plurality of passphrases and identifying data are stored for subsequent transmission to the UVS. The compensating system comprises a pair of tone devices that generate low frequency tones and high frequency tones, respective that are amplified to different amplification levels. The amplification levels compensate for the low and high frequency tone transmission characteristics associated with different microphones in use in a public telephone system. The systems further performs channel normalization with the user-activated AUD and the UVS. Tone signals representative of the access telephone number transmitted by the AUD to reach the UVS, have a reference amplitude of zero.

Abstract: The present invention provides a two-wire line conditioning apparatus having an improved calibration and cancellation circuit and provides an improved calibration method using a multitone calibration signal. The output is monitored by an AGC circuit which controls the output, and prevents hard clipping. This is particularly important for DTMF receivers. Additionally, the apparatus automatically adjusts an electronic hybrid as part of a calibration procedure so as to eliminate undesirable echo effects when calls are placed along the line. The electronic hybrid balancing process and echo cancellation circuitry may also be utilized in a circuit for converting four-wire communications systems to two-wire communications systems. The line conditioning device of the present invention utilizes a multitone signal to simulate the voice frequency spectrum between approximately 500 Hz and 3500 Hz to probe the line side with a plurality of tones (e.g., 500, 1500 and 3500 Hz; or 500, 1000, 1500, 2000, 2500, 3000, 3500 Hz).

Abstract: A telecommunications network (10) includes an Originating Switching System (12.sub.1) and a Terminating Switching System (12.sub.2) linked by a plurality of trunks (20.sub.1 -20.sub.4). For each incoming call it receives, the Originating Switching System establishes a Circuit Selection Capability Indicator (CSCI) value that determines what enhancement, if any, the Originating Switching System will provide to the call. The Originating Switching System also utilizes the CSCI value to select an appropriate trunk that is compatible with a type of call enhancement, if any, to route the call to the Terminating Switching System which also enhances the call in accordance with the CSCI value. Determining the call enhancement in accordance with the CSCI value allows different types of calls which are routed on the same trunk to get different call quality enhancements, depending on the call type.

Abstract: A system (130) for providing enhancement to a voice-band signal in a telecommunications network (138) is provided. The present system (130) includes a power averager (18) for determining the average power of the voice-band signal. The present system (130) also includes an equalizer (132) for attenuating a predetermined portion of the voice-band signal and an output scaler (134) for scaling the equalized voice-band signal with a scaling factor. The system (130) of the present invention also includes an automatic gain enhancer (136) for applying an automatic gain factor to the scaled voice-band signal and wherein the automatic gain factor is dependent upon the average power of the input voice-band signal.

Abstract: In a communications system containing signal enhancement circuitry capable of selectively enhancing portions of a plurality of signals carried on a corresponding plurality of channels through the signal enhancement circuitry, a system and method for controlling enhancement on a per call basis. The system includes: (1) command generation circuitry for generating an enhancement command for transmission to the signal enhancement circuitry and (2) command reception circuitry, associated with the signal enhancement circuitry, for receiving the enhancement command from the command generation circuitry and selectively enhancing the portion of the signals on the per call basis according to the enhancement command.

Abstract: A "loudness balance" procedure is used to determine the amount of dynamic range for syllabic compression loss in an individual's cochlea. In particular, an individual is asked to adjust a set of tone volumes, which are played to them, until the tone volumes sound equally loud. Based on the adjustments performed by the individual, a compression ratio is determined for that individual over at least one frequency band. The determined compression ratio is then subsequently used by a telecommunications system to provide an individual-specific sound enhancement in a telephone call. As a result, the individual-specific sound enhancement particularly compensates for an amount of hearing loss unique to that individual.

Abstract: An automatic line equalizer for use with a T1 or an E1 repeater in a telecommunications system is disclosed. The equalizer includes a first equalizing means for equalizing an input line having a line length falling within a first range and a second equalizing means for equalizing an input line having a line length falling within a second range. The equalizer includes a line condition/length detector. The equalizer includes switch means for selectively coupling one of the first and second equalizers to the selected input line as a function of a line condition indication. The switching means prevents hunting between said first and said second equalizers.

Abstract: A telecommunications network (120) is provided with adaptive gain control (AGC) of the voice signals in network (120). Network (120)includes an input (12) for receiving a voice signal, an output (14) for receiving the voice signal, and a coupling between input (12) and output (14) including at least one switch (124) or (126). Network (120) also includes voice enhancer (10) including power averager (18) for measuring and determining the average power of an input signal. Voice enhancer (10) also includes bass band equalizer (16) to attenuate a predetermined portion of the input signal to provide an equalized input signal. From the average power of the input signal is determined a scaling factor from a gain/attenuation look-up table (28). Voice enhancer (10) also includes output scaler (30) coupled to output (14), output scaler (30) scales the equalized input signal with the scaling factor and provides the scaled signal to output (14).

Abstract: A level of transmit attenuation and a level of receive attenuation are generated using a discrete state machine of a controller. Local audio signals generated with a local microphone are attenuated with a transmit attenuator at the level of transmit attenuation for transmission to a remote node. Remote audio signals received from the remote node are attenuated with a receive attenuator at the level of receive attenuation for playback at a local speaker.

Abstract: A channel interface unit for interconnecting a two-wire transmission path which transmits analog signals and a digital signal transmission path. The channel interface unit includes a digital interface unit coupled with the digital signal transmission path and producing digital signals based on digital signals received from the digital signal transmission path, a converter for converting the digital signals to analog signals, a filter for removing noise, produced by the converter, from the analog signals, a driver, coupled with the two-wire transmission path, for amplifying the filtered analog signals, and a power supply for providing regulated voltage to the driver.

Abstract: A channel interface unit for interconnecting a two-wire transmission path which transmits analog signals and a digital signal transmission path. The channel interface unit includes a digital interface unit coupled with the digital signal transmission path and producing digital signals based on digital signals received from the digital signal transmission path, a converter for converting the digital signals to analog signals, a filter for removing noise, produced by the converter, from the analog signals, a driver, coupled with the two-wire transmission path, for amplifying the filtered analog signals, and a power supply for providing regulated voltage to the driver The channel interface unit also includes a device for decreasing effective capacitance by modifying the output impedance of the driver.

Abstract: An audio teleconferencing apparatus which permits two-way communications between more than two parties without the necessity of complete switching and does not allow a loss to increase in a communication channel of a particular speaker. The audio teleconferencing apparatus further comprises more than two line interface circuits, each provided with a voice activity detector for detecting voice activity in a received speech signal, a received-speech variable loss for inserting a loss into a speech receiving path when the received speech signal is voice non-active, and a sending-speech variable loss for inserting a loss in a speech signal transmission or sending path when the speech to be sent is voice-active, whereby a certain loss is inserted in any four-wire multiplex loop in the line interface circuit.

Abstract: Methods and apparatus are disclosed for improving the clarity and intelligibility of incoming speech in a telephone connection. Improved sound quality is provided by automatically selecting, or enabling the selection of, an appropriate sound enhancement for a particular individual at a given telephone number. Sound enhancements applied to calls originating from or terminating at a given telephone station are determined on a call-by-call basis. The sound enhancement set at the beginning of a telephone call can be changed mid-call to a different sound enhancement. Thus, different occupants of a household may select their own sound enhancement, or may select no enhancement at all.

Abstract: A system and method to provide voice enhancement circuits on a voice path on an individual basis, without having in-line circuits inserted into the lines or trunks. The enhancement circuit is inserted within one or more of the switches. This system comprises a switching system having a switching fabric and a central control, wherein the switching fabric includes several extended time slot interchange (XTSI) units connected to a time multiplex switch. The extended time slot interchange unit has approximately twice the capacity of a standard time slot interchange unit. Such XTSI is connected to voice enhancing circuits, which the TSI may then switch enhancement circuits into a call path. A method of setting up a call according to this invention includes routing the call to a switch, routing the call in the switch into an XTSI, which routes a call through the enhancement circuit, and then back through the XTSI.

Abstract: A method and apparatus for utilizing DTMF tones and supervisory tones in a call processing system to adjust parameters for future tones and/or speech. The tones initially received are analyzed to determine characteristics such as signal-to-noise ratio, twist, signal level, etc. Processing parameters for future tones and speech are adjusted based upon the characteristics of a first tone.

Abstract: An enhanced amplifier for interconnecting a two-wire, bidirectional cable side transmission path and a two-wire bidirectional equipment side transmission path in a communication system includes a first splitter and a second splitter. The first splitter is formed of a cable-to-equipment voltage sense amplifier, a shunt current driver, and a series current compensator for generating a first simulated terminating impedance. The second splitter is formed of an equipment-to-cable voltage sense amplifier, the shunt current driver and the series current compensator for generating a second simulated terminating impedance. Other features disclosed include manual balancing circuitry for manually adjusting the cable side balance networks. Further, a loopback circuit is provided form facilitating remote testing and alignment of the amplifier from the cable side.

Abstract: A telecommunications network (120) is provided with adaptive gain control (AGC) of the voice signals in network (120). Network (120) includes an input (12) for receiving a voice signal, an output (14) for receiving the voice signal, and a coupling between input (12) and output (14) including at least one switch (124) or (126). Network (120) also includes voice enhancer (10) including power averager (18) for measuring and determining the average power of an input signal. Voice enhancer (10) also includes bass band equalizer (16) to attenuate a predetermined portion of the input signal to provide an equalized input signal. From the average power of the input signal is determined a scaling factor from a gain/attenuation look-up table (28). Voice enhancer (10) also includes output scaler (30) coupled to output (14), output scaler (30) scales the equalized input signal with the scaling factor and provides the scaled signal to output (14).

Abstract: A telephone line repeater (28) includes a pattern decoder/recognizer (100) which recognizes, inter alia, an input interrogating signal transmitted on a telephone network (20). The interrogating signal includes an address field (302) having at least one subfield (304) therein. The pattern decoder (100) determines that its repeater (28) is to be tested, if a predetermined subfield (304) of the interrogating signal is configured according to a predetermined select convention. When the repeater (28) recognizes that it is being addressed, a controller (110) causes the addressed repeater to "loop back", returning, in alternating sequence, test-indicative no-operation signals and diagnostic information ("DSM") signals generated by the repeater. Methods for testing a repeater and methods for testing telephone networks are also disclosed.

Abstract: An integrated control circuit for generating a regulated control voltage for voltage-controlled attenuators used in speakerphone circuits. The circuit provides separate transmit and receive voltage regulation to provide accurate gain settings for the voltage-controlled attenuators. The circuit eliminates the need for absolute value external components or external signal feedback between the voltage-controlled attenuators and the control circuit.

Abstract: In a telephone system 1 connectable to both integrated service digital network (ISDN) lines 3a and 3b and a public subscriber telephone network (PSTN) line 7 and having a plurality of extension telephone sets 11a to 11d, the speech sound volume through the ISDN lines 3a and 3b is so adjusted as to become equal to that through the PSTN line 7, by changing the loss rate (i.e. gain) of digital pads 55 and 47 provided in ISDN interface units 5a and 5b. That is, a level confirmation tone is generated in the system 1; this tone is transmitted to a first extension telephone set 11a via only the ISDN on one hand and to a second extension telephone set 11b via both the ISDN and PSTN in series on the other hand; and the loss rate (i.e. gain) of the digital pads 55 and 47 are changed or adjusted so that the sound of the level confirmation tone can be heard at the same volume level by both the telephone sets 11a and 11b.

Abstract: A Data Communication Device permits extended range RS-232 communications between two devices by utilizing a current loop between the remote device and the Data Communication Device. The Data Communication Device may be used for any mix of signal or control lines specified for RS-232 and will support multiple remote devices in a broadcast mode or half duplex polled mode. Only a single Data Communication Device per remote communications circuit is required.

Abstract: A method for controlling a voice-controlled voice direction recognition unit for a digital telephone station, which can be connected to termination units having different hybrid attenuations. The method prevents self-interruption in the transmission side due to an excessively low hybrid attenuation. This is achieved by providing a counter unit that identifies the switching occurrences of the voice direction recognition per time unit and, when a predetermined number is exceeded, that causes the voice direction recognition unit to connect an auxiliary attenuation element into every voice channel. The method can be utilized in digital telephone stations whose termination units are located in an appertaining switching system.