Abstract: A method of detecting narrow-band signals in an echo canceller to control the coefficient update process, echo suppression process, and the disablement of the echo canceller is provided. The method includes the steps of searching for the presence of a generic tone, searching for DTMF tones, searching for a 2100 Hz tone and, if a generic tone is found, searching for a dial tone. If a generic tone is detected on the far end signal, a tone flag is set which triggers the disablement of the echo canceller coefficient update process and the enablement of the echo suppressor process. If a 2100 Hz tone is detected, the echo canceller is disabled. If a DTMF signal is detected, DTMF flags are set which are used by the echo suppressor for suppressing the near end echo. Finally, if a dial tone is detected, dial tone flags are set which cause the disablement of the echo suppressor.

Abstract: A communication system in which a subscriber terminal moves between networks having different number plans. The moved subscriber terminal can receive the same service as that of the network from which the subscriber terminal has moved (movement originator network) without being conscious of the movement. In the system, when the subscriber terminal moves between the networks having the different number plans, the movement of the subscriber terminal is registered in the network to which the subscriber terminal has moved (movement destination network), data is exchanged with the movement originator network, an allocated number correlative table is prepared and stored in the movement destination network; while, a movement destination number correlative table is prepared and stored in the movement originator network.

Abstract: A caller ID system is disclosed in which a prescaler is coupled between a source of a caller ID signal and a caller ID receiver, and a processor is coupled to the caller ID receiver. The prescaler is responsive to the processor for scaling the caller ID signal.

Abstract: A telephone number redial function is provided wherein only digits corresponding to a completed telephone number are stored within a redial buffer. Any digits entered by a user after completion of a telephone number, such as digits entered in response to queries by a voice mail system, are not stored within the redial buffer. As such, digits entered by a user after completion of a telephone number do not affect the telephone number of the redial buffer allowing the telephone number to be properly redialed. In one arrangement, the redial buffer is incorporated within a cellular telephone. Digits entered by a user are first entered within a new number buffer until a send command is received. Thereafter, the digits stored within the new number buffer are transferred to the redial buffer and also transmitted from the cellular phone. Any additional digits entered by the user are transmitted, but not stored within the redial buffer.

Abstract: A new speaker verification method, termed Mixture Decomposition Discrimination (MDD), and a new apparatus for using MDD are presented. MDD takes mixture component score information from a speaker independent recognizer and transmits this information while it is still decomposed as a mixture of component scores that indicate the response of the states of a Hidden Markov Model (HMM) before this information is combined into a single speaker independent recognizer parameter. MDD can be very effective in improving the performance of existing verification methods based on speaker dependent HMMs with cohort normalization because the errors of the two speaker verification methods are very uncorrelated statistically. Experimental results have shown that when MDD is incorporated into a system that also uses speaker dependent HMMs, the resulting hybrid system has its average equal error rate reduced by 46% compared to cohort normalized speaker independent HMMs.

Abstract: A tone blocking system and method for use preferably in conferencing systems in order to prevent control tones from being transmitted to other conferees is disclosed. The buffer length used to process the signal and detect tones is varied, being increased when a tone is suspected to allow for sophisticated tone detection algorithms, and being decreased when silence is present by trimming the silence away. The technique minimizes delay, and its degrading effect on echo, but nonetheless provides for a lengthy buffer required to do reliable tone detection. The invention is applicable to any signal other than tone as well.

Abstract: A telephone ringing system in which one or more user-selected sound segments are received using the microphone portion of a telecommunications instrument, stored in digital or analog form, and audibly reproduced using the loudspeaker portion of the telecommunications instrument when a ring signal is detected to provide a distinctive, personalized indication of an incoming call.

Abstract: An amplified telephone handset includes a current sensing circuit that senses the polarity of DC voltage supplied from a telephone base, irrespective of whether the DC voltage is of standard or reverse polarity. This feature allows the amplified telephone handset to operate with telephone bases that supply voltages through a telephone cord of either standard or reverse polarity. The amplified telephone handset also includes a voltage detection circuit that detects "off-hook" or "on-hook" states by detecting the absence or presence of the voltage. If a DC voltage is detected, then the amplified telephone handset is coupled to the telephone base and the voltage detection circuit disconnects the battery from receiver and transmitter circuits in the amplified telephone handset, placing the amplified telephone handset in a "on-hook" state.

Abstract: A method for controlling echo canceling in an echo cancellation system using a state machine controller. A far-end speech signal transmitted through a communications channel may be passed through an echo channel to produce an echo signal. The echo signal is summed with a near-end speech signal and transmitted back to the far-end. An echo canceller rejects the echo signal by forming an estimate of the echo signal and subtracting the estimate from the sum of the near-end speech signal and the echo signal. The echo canceller includes a state machine which is configured into a predetermined state of a plurality of states depending on the presence near-end speech signal, far-end speech signal, or both near-end and far-end speech signals. Based on the predetermined state of the state machine, the controller in the state machine controls the update of coefficients of a plurality of adaptive filters.

Abstract: An add-on telephony server platform provides inter alia high-speed call switching. The outbound automatic call distributor establishes one "nailed" voice connection between the telephony served platform and trunks connected to the automatic call distributor. The outbound call distributor establishes another "nailed" voice connection between the telephony server platform and the voice sets at the agent stations assigned to outbound operations.

Abstract: Telephone numbers along with alphanumerical identifications representing the same are stored in the memory of a mobile phone. The number is selected with quick dialing so that the user first scrolls the identification corresponding to the required number on the display (11). In the method of this invention the display is a touch-sensitive display (21). For each identification several alternative telephone numbers are stored in the quick dialing memory, for instance to work (W), home (H), or mobile phone (M). The desired number is selected and a call initiated by writing a gesture (23) corresponding to the required number on the display (21) or in a specified field of the display (24).

Abstract: This invention permits transferring calls along with data that identifies, for example, the caller's name, pertinent account information, any transaction information acquired prior to the need to transfer the call and why the caller is being transferred between a plurality of call centers. The method and system of the invention, therefore, coordinate a voice component and a data component of a call to transfer the call from a first call center across a communication path such as a switching network to a second call center. The first call center includes the necessary network control circuitry to direct an associated switching network to transfer to the second call center the voice component of the call together with an identifier for uniquely identifying the call. The identifier may be generated using DNIS and/or ISDN protocol principles. The method and system also generate and direct a transfer screen from the first call center to the second call center.

Abstract: An improved method and apparatus (12) for handling calls in a call center predicts (206, 312, 410) a caller's level of motivation for making the call and uses (208-212, 314-330, 412-416) the prediction as a basis for determining how to handle the call. The method and apparatus respond to a call coming in to the call center by determining (206, 306, 404) a number of call attempts--either all call attempts whether answered or unanswered by an agent, or only unanswered call attempts--made by the caller to the call center within a predetermined elapsed period of time, and possibly also by determining (308-310, 406) other predictors of the level of motivation of the caller in making the call, prior to the call being answered.

Abstract: A speech recognition system is disclosed which comprises a core speech recognition program and a plurality of utility program modules for adjusting various recognition parameters such as gain, sensitivity and acceptance threshold and for improving the training of word models. The system further provides a decision tree and utility controlling program module which can be invoked by a user confronting problems during the running of the core program. The system utilizes user input to traverse the decision tree and to initiate appropriate ones of the utility program modules thereby to alter the on-going behavior of the core recognition program.

Abstract: A modular switch box which is responsive to a control signal, is diagnosable, and is not dependent upon the type of protocol being switched. There may be a plurality of switch boxes wherein each switch box includes a plurality of inputs and an output. There is a selector for selecting which input is connected to the output and a controller for controlling the selector. There is also a control means for controlling the switch boxes. The switch boxes are connected in a chain so that the input of one switch box is connected to the output of the next, etc., until the output of the last switch box is connected back to the controller. The controller sends a command message to the first switch box which then executes the command and forwards the message on to the next switch box, etc., until the command returns to the controller. In this manner, the controller knows that all switch boxes have received the command.

Abstract: To improve the signal to noise ratio of the receiver of a transmitting modem, the transmitting modem is equipped with the ability to cancel the far listener echo. Advantageously, doing so allows higher data rates or reduced error rates. Cancellation of the far listener echo is achieved by coupling the signal containing the in-phase, quadrature (also known as X and Y, or real and imaginary) components of the modem transmit signal to a far echo canceller that is included in the modem in addition to the far echo canceller conventionally found in modems. However, the bulk delay employed by the additional far echo canceller is arranged to be a delay of two round trips between the transmitting modem and the receiving modem rather than the conventional single round trip delay of the conventional far echo canceller.

Abstract: An echo canceling loudspeaker telephone includes a loudspeaker which produces a sound pressure wave in response to an input signal which is applied to an audio input thereof. This sound pressure wave includes a desired linear component which is a linear function of the input signal, and an undesired non-linear component which is a non-linear function of the input signal, and the sound pressure wave is transmitted along an acoustic path. A microphone is positioned in the acoustic path and converts the sound pressure wave into an output signal. An echo filter is responsive to the input signal and generates an estimated echo signal. This echo filter includes a loudspeaker model which generates an estimate of the sound pressure wave including an estimate of the linear component and an estimate of the non-linear component. This echo filter also includes an acoustic path model which generates an estimate of the acoustic path from the loudspeaker to the microphone.

Abstract: In one aspect, the invention provides word recognition systems that operate to recognize an unrecognized or ambiguous word that occurs within a passage of words. The system can offer several words as choice words for inserting into the passage to replace the unrecognized word. The system can select the best choice word by using the choice word to extract from a reference source, sample passages of text that relate to the choice word. For example, the system can select the dictionary passage that defines the choice word. The system then compares the selected passage to the current passage, and generates a score that indicates the likelihood that the choice word would occur within that passage of text. The system can select the choice word with the best score to substitute into the passage. The passage of words being analyzed can be any word sequence including an utterance, a portion of handwritten text, a portion of typewritten text or other such sequence of words, numbers and characters.

Abstract: The present invention provides a novel method of analyzing speech signals in order to reduce the computational power required to perform both speech compression and voice recognition operations. Digital speech signals are provided to a speech analyzer which generates a linear predictive coded (LPC) speech analysis signal that is compatible for use in both the voice recognition circuit and the speech compression circuit. The speech analysis signal is then provided to the compression circuit, which further processes the signal into a form used by an encoder and then the encoder encodes the processed signal. The same speech analysis signal is also provided to a voice recognition circuit, which further processes the signal into a form used by a recognizer and then the recognizer performs recognition on the processed signal.

Abstract: A voice system and method in which messages are stored as data files. One or more tags are associated with each file, which identify the mailboxes intended to receive the message contained in the file. An owner of a mailbox can transfer a message to another mailbox by changing the tag to designate a transferee-mailbox. This tag manipulation saves memory space because the message file is not copied into the transferee-mailbox, only a tag is changed. Time is also saved because no delay is imposed by a copying process.