Abstract: This invention relates to a method of processing sensitive information over VoIP. The method provides a method of processing, by a call processor, a media call comprising the steps of: receiving a first signalling stream from a first entity; creating a second signalling stream to a second entity; forwarding signals received from the first signalling stream to the second signalling stream; receiving a third signalling stream from the second entity; creating a fourth signalling stream to the first entity; and forwarding signals received on the third signalling stream to the fourth signalling stream; the first signalling stream containing instructions to set up a media call between the first entity and the second entity such that media is transmitted over a first media stream from the first entity to the second entity and a media is transmitted over a second media stream from the second entity to the first entity without intervention by said call processor.

Abstract: The invention relates to a method of processing sensitive information over VoIP. The method provides a method of processing, by a call processor, a media call comprising the steps of: receiving a first signalling stream from a first entity; creating a second signalling stream to a second entity; forwarding signals received from the first signalling stream to the second signalling stream; receiving a third signalling stream from the second entity; creating a fourth signalling stream to the first entity; and forwarding signals received on the third signalling stream to the fourth signalling stream; the first signalling stream containing instructions to set up a media call between the first entity and the second entity such that media is transmitted over a first media stream from the first entity to the second entity and a media is transmitted over a second media stream from the second entity to the first entity without intervention by said call processor.

Abstract: A method for tracking valid and invalid sequence numbers in a storage system, performed by a processor, is provided. The method includes establishing a table as a key value store in memory in the storage system. The table has sequence numbers as keys and represents valid sequence numbers and invalidated sequence numbers of an open-ended sequence relating to storage of data or metadata in the storage system. The method includes adding to the table an entry that records a first plurality of consecutive sequence numbers, as a first range-valued key associated with a first value indicating the first plurality of consecutive sequence numbers is valid. The method includes adding to the table an entry that records a deletion of a second plurality of consecutive sequence numbers, as a second range-valued key associated with a second value indicating the second plurality of consecutive sequence numbers is invalid.

Abstract: In an aspect, a method performed at a server includes: receiving a message representing a request for a transfer; performing a series of automated processing operations on the message; providing, to a client device associated with an authenticated entity, a transfer interface, the transfer interface including an interface element for selecting the transfer; receiving an indication of a selection of the transfer; in response to receiving the indication of the selection, retrieving stage state data, the stage state data indicating a current one of the automated processing operations in the series; and sending, to the client device, an indication of the stage state data, wherein the client device is configured to update the transfer interface in response to receiving the indication to display a representation of the series of automated processing operations which identifies one of the processing operations in the series as a current processing operation.

Abstract: A numerical sound synthesis method for representing data as audio for use in data sonification employing a Hilbert Space eigenfunction model of human auditory perception is described. The synthesis method comprises approximating an eigenfunction equation representing a model of human hearing, calculating the approximation to each of a plurality of eigenfunctions from at least one aspect of the eigenfunction equation, and storing the approximation to each of a plurality of eigenfunctions. The approximation to each of a plurality of eigenfunctions represents a perception-oriented basis functions for mathematically representing audio information in a Hilbert-space representation of an audio signal space. The model of human hearing can include a bandpass operation with a bandwidth having the frequency range of human hearing and a time-limiting operation approximating the time duration correlation window of human hearing.

Abstract: In a system for performing dual-tone multi-frequency (DTMF) digit detection, an automatic gain controller receives an incoming frame carrying at least one tone sample and adjusts the energy level of the at least one tone sample responsive to the energy level exceeding a frame energy threshold. A single tone detector determines at least one frequency of the at least one tone sample in the incoming frame. A dual tone detector identifies an energy level of each of the determined frequencies and generates an indication of whether a DTMF digit is present in the incoming frame responsive to a first determined frequency corresponding to a frequency in a first grouping of frequencies and a second determined frequency corresponding to a frequency in a second grouping of frequencies. A DTMF digit determiner for determining a DTMF digit based on the indication.

Abstract: A system may include control circuitry for detecting a plosive event associated with a microphone transducer and in response to the plosive event, causing restoration of acoustic sense operation of the microphone transducer and a processing circuit associated with the microphone transducer. A system for configuring a filter having at least two frequency response configurations to achieve an effective frequency response configuration intermediate to the at least two frequency response configurations may include control circuitry for rapidly switching between the at least two frequency response configurations such that a weighted average frequency response of the filter corresponds to the effective frequency response configuration.

Abstract: A method to determine the magnitude MA of a signal component with frequency ?A from a set of N digital samples of an input signal acquired at a sampling rate R, said input signal having a discrete spectral representation having n bins with frequencies ?1, . . . , ?n and corresponding magnitudes M1, . . . , Mn, the spectral representation being derivable from the input signal using a transform, involving choosing an extraction bin with index k?[1, . . . , n] and frequency ?k??A from the spectral representation; determining a magnitude Mk of this extraction bin; determining an allocation factor indicating a portion Mks of a sinusoidal signal with frequency ?A and unity magnitude that is allocated to the extraction bin when the transform is applied to the sinusoidal signal to generate a spectral representation out of the sinusoidal signal; and determining the magnitude MA of the signal component from the magnitude Mk of the bin in combination with the factor.

Abstract: Provided herein are methodologies, systems, apparatus, and non-transitory computer-readable media for providing an adjustable dual-tone multi-frequency (DTMF) phone system. The DTMF phone system includes a DTMF adjustment module that interfaces with a base DTMF phone system and retrieves call parameter values from at least one properties file. The call parameter value is indicative of an audio file to be played by the DTMF phone system, or a DTMF call flow option file. The DTMF adjustment module provides the call parameter values to the base DTMF phone system. The base DTMF phone system includes a number of variables, and a DTMF call flow is generated by assigning at least one of the call parameter values to one or more of the variables within the base DTMF phone system.

Abstract: Briefly, in accordance with one embodiment, a home country associated with a current mobile telephone number for a mobile subscriber is identified. Other countries, such as the following are also identified: a country in which the mobile subscriber is using the internet; a country in which the mobile subscriber is roaming, is operating a current mobile telephone associated with the mobile subscriber's mobile provider account or is operating a current mobile telephone having a SIM card associated with the mobile subscriber's mobile account; a country in which a merchant for a transaction is located; and/or a country in which goods in the transaction, if any, are being shipped. The countries are compared and scoring is employed to facilitate identification of a suspicious transaction.

Abstract: A telephone subnet crawler is used to access automated telephone response systems and index the information, contents and structure contained therein. A database of the information, contents and structure of a plurality of automated telephone response systems is created by the telephone subnet crawler. A user interface provides callers with direct access to the information, contents and structure of the automated telephone response systems contained in the database. Where an automated telephone response system requires user input, the user interface calls the automated telephone response system and navigates to the node requiring user input, provides the user input and displays the results to the user. Where an automated telephone response system connects to an operator, the user interface calls the automated telephone response system, navigates to the node for an operator, and when an operator is detected, calls the user at a user provided callback number.

Abstract: An arrangement for allowing “hands-free” access to DTMF-controlled systems, such as one's voice mail messaging systems, utilizes a speech-to-DTMF tone application that monitors the communication between the user and the DTMF-controlled system. A speech recognition unit is utilized to retrieve certain voice commands (e.g., “next”, “skip”, “repeat”, “forward”, etc.) when uttered by the user. The application then translates the received commands into the proper DTMF tone sequence used by the DTMF-controlled system and transmits the DTMF tones to the system. The application is particularly useful in the cell phone environment and avoids the necessity of the user to constantly switch between using the keypad and listening to messages/commands from the system.

Abstract: A method is specified for the handling of a telecommunications connection between at least two subscribers, wherein the telecommunications connection has a signalling Channel for transmission of a signalling stream and a payload Channel for the transmission of a payload stream, wherein frequency-coded Signals are transmitted in the payload stream, wherein at least one subscriber among the at least two subscribers is assigned to a network which has a switching device and a signal processing device provided separately from the switching device, and wherein the frequency-coded Signals are recognised in the signal processing device and are transmitted independently of the payload stream to the switching device. The invention also relates to a telecommunications arrangement, a switching device and a network coupling device.

Abstract: When receiving Dual Tone Multi-Frequency (DTMF) information, we provide a way to determine whether to store such, which may include sensitive user input information. More specifically, a media gateway (MG) obtains DTMF digit mask control information, which indicates whether DTMF digit information may include sensitive user input information not suitable for storage. Based thereon, it decides, whether to store the received DTMF digit information. In this way, the DTMF digit mask control information controls whether to store the user input information on the DTMF digit information processing device in the architecture with the service function being separated from the bearing function; thus, avoiding information security risks brought by indiscriminate storage of sensitive contents in the user input information.

Abstract: Systems, methods, and computer-readable media for tracking, reconciling, and reporting on data representing Dual Tone Multi-Frequency (DTMF) tones and voice signals communicated between call center and users during interactions therebetween. The system may include a data store adapted to store the DTMF data and a text corresponding to the voice data; a transcription process receiving data representing the voice signal and converting it into corresponding text; a report generation process obtaining the DTMF data and the text, transmitting a report containing the same to a client, and updating the date store accordingly; and a reconciliation process receiving data enabling reconciliation of data as tracked by the call center and as tracked by an entity other than the call center. The transcription process may operate as a third-party or an in-house entity relative to the at least one call center.

Abstract: A secure three-way bridge in a SIP-based processing switch protects sensitive information provided by a party during a call with an agent in a contact center. During the call, the agent may bridge an interactive voice response system (“IVR”) onto the call using the secure three-way bridge so that the party can enter sensitive information, such as a credit card account, to the IVR. The information is provided by the party using dual-tone multiple frequency (“DTMF”) tones that are presented in an unmodified form to the IVR, but modified and then presented to the agent, so that the agent is unable to discern the sensitive information provided by the party. Other embodiments may accomplish similar capabilities by application of components outside of the bridging structure. Further, recording systems may record the call comprising the modified audio, thus protecting the sensitive information from subsequent detection upon review of the recording.

Abstract: Various embodiments associated with signal classification are described. A repeating signal can be partially corrupted and therefore a receiver can obtain an incomplete signal. In one example, the incomplete signal can have discontinuous unobstructed segments. In order for the receiver to understand the contents of the signal, the receiver can identify repeated information in the discontinuous unobstructed segments. This repeated information can be processed to determine content of the signal. In one embodiment, the signal is a dual-tone multi-frequency (DTMF) signal. The receiver can process the content of the DTMF, such as by employing a set of high and low band pass filters, and can identify a high and low frequency of the DTMF. With the high and low frequencies identified, the receiver can access a look-up table and identify a character indicated by the high and low frequency. The receiver can then use or send out the identified character.

Abstract: Aspects relate to provision of enterprise call capabilities to mobile devices. For example, a mobile device can indicate, over a data channel, that a PBX is to make a call on its behalf to a called party. The PBX can call back the mobile device, call the called party, and bridge those call legs to establish the call. The mobile device can employ mechanisms that a particular incoming call is made by the PBX. These mechanisms can include using ANI information, sending, and receiving audible verification codes over the voice channel established after answering the incoming call. The verification codes can be selected based different behaviors of the mobile devices.

Abstract: In a method of processing an incoming sequence of one or more DTMF tones at a mobile device, the sequence is decoded to thereby identify any encoded message. Where the decoding does not identify a message, at least applying an error correction algorithm to the sequence to identify a message. Where the algorithm fails to correct the sequence, the sequence is compared to messages currently expected by the mobile device. If a message is not identified, a negative acknowledgement message may be sent. At a server, upon receipt of a negative acknowledgement message, it is determined if a message was sent to the mobile device in a previous pre-determined time window. If so, the message is re-sent. The server repeatedly sends certain messages to a mobile device until an acknowledgement message indicating receipt of the message by the mobile device is received.

Abstract: A method and a voice-operated arrangement for interacting with a dual-tone multifrequency (DTMF)-controlled system. A record, including representations of DTMF tones and associated with communicating with a remote DTMF-controlled system, is accessed. A communication link between a user voice-communication device and the remote DTMF-controlled system is established through a speech-to-DTMF tone application. A user voice command is translated, using the record, into at least one DTMF tone for interacting with the remote DTMF-controlled system.

Abstract: An exemplary telephone apparatus which can transfer a first phone call from a Bluetooth enabled cellular phone to a home phone and, making a second cellular phone call by a home phone via a Bluetooth enabled cellular phone, includes a Bluetooth enabled integrated chip (IC). The IC includes a Bluetooth modem configured for receiving a Bluetooth signal from a Bluetooth radio frequency (RF) antenna of the cellular phone, a central processing unit (CPU) built-in the IC, for switching controlling of transferring between the first phone call and the second phone call, and a digital signal processor (DSP) and a CODEC built-in the IC. The telephone apparatus makes use of digital signal processing technology to generate and decode various tones and signals. The telephone also avoids the problem of distortion of DTMF tone during transmission through a Bluetooth RF.

Abstract: An enhanced tone detector including an adaptive multi-bandpass filter, a tone detector and a filter control unit. The adaptive multi-bandpass filter has a signal input receiving an input signal, a signal output providing a filtered signal, and at least one control input for receiving at least one control signal. The tone detector has an input coupled to the signal output of the adaptive multi-bandpass filter and at least one output providing at least one tone detection signal when a tone event is detected. The filter control unit has a first input receiving the input signal, at least one second input receiving the at least one tone detection signal, and at least one first output providing the at least one control signal. The control signals may include one or more frequency component signals and corresponding bandwidth signals in which the frequency component signals correspond to component frequencies of a tone event.

Abstract: Systems, methods, and computer-readable media for tracking, reconciling, and reporting on data representing Dual Tone Multi-Frequency (DTMF) tones and voice signals communicated between call center and users during interactions therebetween. The system may include a data store adapted to store the DTMF data and a text corresponding to the voice data; a transcription process receiving data representing the voice signal and converting it into corresponding text; a report generation process obtaining the DTMF data and the text, transmitting a report containing the same to a client, and updating the date store accordingly; and a reconciliation process receiving data enabling reconciliation of data as tracked by the call center and as tracked by an entity other than the call center. The transcription process may operate as a third-party or an in-house entity relative to the at least one call center.

Abstract: Seamless wideband support is afforded by utilizing split-band data streams. For wideband signals, the 8 kHz bandwidth is divided into a low band, with approximately 0-4 kHz bandwidth, and a high band, with approximately 4-8 kHz bandwidth. Narrowband functions and services operate on the low band, while wideband functions and services operate on both low and high bands.

Abstract: A tone detector for detecting a tone having a plurality of targeted frequencies comprises a first filter bank providing frequency estimates for an initial indication of a presence of each of the targeted frequencies; a second filter bank being controlled by the frequency estimates and comprising a plurality of filters operating in different frequency bands, each frequency band comprising at least one of the targeted frequencies, each of the filters providing a modified signal; a third filter bank extracting a noise signal from the input signal; a plurality of level control units (LCU) providing normalized modified signals having a constant amplitude; a power estimation module providing a power estimation parameter for each of the modified signals and the noise signal; and a discrimination module controlling a plurality of demodulator modules, each of the demodulator units providing a fine estimate of one of the targeted frequencies.

Abstract: In one aspect, a method of processing a voice signal to extract information to facilitate training a speech synthesis model is provided. The method comprises acts of detecting a plurality of candidate features in the voice signal, performing at least one comparison between one or more combinations of the plurality of candidate features and the voice signal, and selecting a set of features from the plurality of candidate features based, at least in part, on the at least one comparison. In another aspect, the method is performed by executing a program encoded on a computer readable medium. In another aspect, a speech synthesis model is provided by, at least in part, performing the method.

Abstract: Aspects relate to provision of enterprise call capabilities to mobile devices. For example, a mobile device can indicate, over a data channel, that a PBX is to make a call on its behalf to a called party. The PBX can call back the mobile device, call the called party, and bridge those call legs to establish the call. The mobile device can employ mechanisms that a particular incoming call is made by the PBX. These mechanisms can include using ANI information, sending, and receiving audible verification codes over the voice channel established after answering the incoming call. The verification codes can be selected based different behaviors of the mobile devices.

Abstract: Control and status information between a mobile device and another device can be signaled over a voice channel using DTMF tones, e.g., a pre-defined sequence of DTMF tones can signal a desired message (more generally, a feature code). Tones sent are separated from each other by a time gap (Tgap timer). If a receiving device does not receive a sequence of DTMF tones that can be matched to stored definitions of feature codes, then the receiving device generates a negative ACK. Also, if the sending device does not receive either negative or positive ACK, then sending device resends all tones of the feature code after a retry delay (Tretry timer). Rather than having the Tretry timer start when tones for a feature code start to be transmitted, the Tretry timer is started after the Tgap timer expires for the last tone of a feature code. Thus, only one timer is running (either Tgap or Tretry), rather than having multiple timers running concurrently. Race conditions between the timers are avoided.

Abstract: Mobile devices are authorized to access PBX-based voice services through presentation of audible tones on a voice channel, which are determined based on a code received over a separate data channel. The device can request a code over a data channel from a server in communication with the PBX. The server provides data representative of the code over the data channel to the device, and arranges for local storage of the code. The device makes a connection with the PBX over a separate voice channel. The mobile device presents the received code as a sequence of audio tones (e.g., DTMF tones). The audio tones are used to generate data that can be compared with stored codes by an authentication module. The authorization module can indicate to the PBX that the code is valid; services can be provided by the PBX in response. The valid codes can be maintained, such as by removing used codes from code storage.

Abstract: A telephone communications system includes a quantizing device quantizing a multifrequency signal at a sampling frequency which is twice or more as larger as a sampling frequency necessary for sampling the multifrequency signal to produce quantized data, a low-pass filter low-pass filtering the quantized data at a cutoff frequency higher than a maximum frequency of the multifrequency signal to produce low-pass filtered data, and a multifrequency signal receiver alternately performing an odd-numbered receiver processing and an even-numbered receiver processing on the low-pass filtered data.

Abstract: A method and system for secure call Dual-Tone Multi-Frequency (DTMF) signaling includes entering (202) dial string of a telephone number of a destination device, assigning (210, 212) the dial string to a predefined string having a total length that is greater than the dial string such that there will be at least one leading hexadecimal bit in the predefined string length that is not used when the entered dial string is converted to a hexadecimal, converting (214) the dial string to hexadecimal, reversing the order of the hexadecimal, and placing the reversed hexadecimal at the beginning of the predefined string, appending (216) “one” bits to the predefined string length indicating how many nibbles of the predefined string length are unused, if any, and appending the remaining intervening unused bits in the predefined string length to “zero” bits, and sending (226) the encoded string length.

Abstract: Aspects relate to provision of enterprise call capabilities to mobile devices. For example, a mobile device can indicate, over a data channel, that a PBX is to make a call on its behalf to a called party. The PBX can call back the mobile device, call the called party, and bridge those call legs to establish the call. The mobile device can employ mechanisms that a particular incoming call is made by the PBX. These mechanisms can include using ANI information, sending, and receiving audible verification codes over the voice channel established after answering the incoming call. The verification codes can be selected based different behaviors of the mobile devices.

Abstract: Upon selection of the frequency of a tone signal, a local oscillator generates a frequency signal having a shift frequency corresponding to the frequency of this tone signal. A frequency shifter generates a frequency-converted signal by shifting a frequency of the tone signal by a shift frequency of the frequency signal generated by the local oscillator. A BPF removes an image signal from the frequency-converted signal which the frequency shifter generated, and a normalizer normalizes the amplitude of the signal output from the second BPF. By this, even if the amplitude of the tone signal is small, the amplitude thereof is unified to a suitable magnitude. For this reason, even if there is a difference in the amplitude of the tone signal, the tone signal can be detected reliably and an false detection can be prevented.

Abstract: In one embodiment, a method for discriminating between a machine generated multi-tone signal and a simulated voice multi-tone signal is provided. The method may comprise comparing output profiles generated from sampled outputs of a plurality of filters. The plurality of filters may have a single input sample stream applied to them and each filter may be preset at a measured multi-tone frequency associated with an identified prospective multi-tone signal. Based on the comparison of the output profiles, an inequality degree between the output profiles is generated and compared to an inequality threshold, thereby to determine whether the input sample stream comprises a machine generated multi-tone signal.

Abstract: Systems, methods, and computer-readable media for tracking, reconciling, and reporting on data representing Dual Tone Multi-Frequency (DTMF) tones and voice signals communicated between call center and users during interactions therebetween. The system may include a data store adapted to store the DTMF data and a text corresponding to the voice data; a transcription process receiving data representing the voice signal and converting it into corresponding text; a report generation process obtaining the DTMF data and the text, transmitting a report containing the same to a client, and updating the date store accordingly; and a reconciliation process receiving data enabling reconciliation of data as tracked by the call center and as tracked by an entity other than the call center. The transcription process may operate as a third-party or an in-house entity relative to the at least one call center.

Abstract: In an embodiment of the method for handling subscriber input during interswitch handover of a call from a first switching station to a second switching station, DTMF digits received at the second switching station are sent in a message to the first switching station. The first switching station then performs an operation on the received digits such as playing the DTMF tones to a network or triggering a service.

Abstract: The most preferred embodiments of the present invention are configured to allow a foreign language interpreter to remotely monitor, control, and interpret various legal proceedings for one or more remote locations, such as courtrooms. The interpreter will use a computer-based system to monitor and control the audio-video functions and communications that take place in one or more remote locations and to pass audio and/or video signals to and from the interpreter and the remote locations, including the people in the remote location that need interpretation services. This system allows the interpreter to monitor facial characteristics of the speakers, route the audio to isolated headsets when interpreting into the foreign language, route the audio to sound reinforcement speakers when interpreting from the foreign language, and interpret a “private” conversation between multiple parties speaking different languages.

Abstract: A DTMF signal processing apparatus of the present invention comprises a data divider unit, a DTMF signal component analyzer unit, a weighting processing unit, a buffer, a DTMF signal erasure determination unit, and a DTMF signal erasure processing unit. The data divider unit divides speech data into a plurality of divided speech data, and the DTMF signal component analyzer unit analyzes whether or not the divided speech data has a DTMF signal component. The weighting processing unit applies a weighting value to divided speech data analyzed at this time and stores the resultant speech data in the buffer, and also applies a weighting value to past divided speech data previously stored in the buffer when the result of the analysis indicates that the analyzed divided speech data has the DTMF signal component. The DTMF signal erasure determination unit determines based on the weighting value whether or not to erase the divided speech data stored in the buffer.

Abstract: In one embodiment, a tone event detector first determines whether the presence of a tone is indicated on the input signal, and then, based on this determination, selectively determines whether a tone has been detected on the input signal. For example, in one embodiment, tone detection is performed only when the presence of a tone is first indicated, such that if the presence of a tone is not indicated, tone detection need not be performed. This may help reduce complexity of a tone event detector since a simplified method may be used to indicate the presence of a tone, and the more complex algorithms for tone detection may be enabled only when needed. Also, in one embodiment, detection of a tone includes generating one or more tone characteristics corresponding to the detected tone which may then be used to determine whether the detected tone corresponds to a valid tone event.

Abstract: The method provides a method and device for detecting in a single or multi frequency signal, one or more frequencies from a predetermined set of frequencies. The Signal is subjected to a complex filter substantially passing all of the frequencies in the predetermined set of frequencies. For each of one or more pairs of members of the complex filtered signal, a complex number Yd is determined having a phase indicative of a phase difference between the two members of the pair. The one or more frequencies are then determined based upon the one or more complex numbers. The method and device of the invention may be used in an apparatus such as a call indicator (CI) detector or a DTMF detector.

Abstract: A system and method for generating Dual Tone Multi-Frequency (DTMF) tones in the media gateway of a Universal Mobile Telecommunications System (UMTS) network. The method includes receiving a DTMF signal in a media gateway controller, where the DTMF signal can be a stop tone request, or a start tone request, sending the DTMF signal from the media gateway controller to the media gateway, monitoring the status of a set of timers, where the set of timers includes a minimum tone duration timer, an inter tone timer, and a maximum tone duration timer, and responding to the received DTMF signal, said status of said timers, or a queued request. The system includes a media gateway, a plurality of timers within the media gateway, a media gateway controller, a finite state machine within the media gateway, where the finite state machine includes the following states: No Tone, Play Tone, Queue Stop Request, Inter Tone Timing, Queue Start Request, and Ready to Stop Tone.

Abstract: A method for decoding a DTMF tone which includes collecting the width of serial signals by measuring the time interval between zero-crossing points within a predetermined time; counting within the predetermined time the number of signal zero-crossing points; performing interpolation on the data produced by the collecting and counting steps with a Fourier transform; identifying a first frequency that has the strongest magnitude from the resulting data; and calculating the lower frequency of the tone being decoded by subtracting the first frequency from the average frequency of the digitized signal.

Abstract: A communication apparatus connected to a PBX and an ATM network is provided with a DTMF signal detecting portion for detecting whether or not a DTMF signal is contained in a voice signal from the PBX, a DTMF signal monitoring portion, a DTMF cell generating portion for coding and sending out a DTMF signal, a DTMF cell decomposing portion for decoding a coded DTMF signal and a DTMF signal generating portion for generating a DTMF signal on the basis of the information content of a decoded DTMF signal and sending out this DTMF signal to the PBX, and the transmission side communication apparatus codes a DTMF signal in a specified form and transmits the DTMF signal and the reception side communication apparatus decodes the coded DTMF signal and, on the basis of this content information, restores the DTMF signal in a DTMF signal generating portion.

Abstract: Automated, computer-based systems, method, and computer-readable media for tracking, reconciling, and reporting on data representing Dual Tone Multi-Frequency (DTMF) tones and voice signals communicated between call center and users during interactions therebetween. The system can include a data store adapted to store the DTMF data and converted text corresponding to the voice signals; a transcription process receiving data representing the voice signal and converting it into corresponding text; a report generation process obtaining the DTMF data and the text, transmitting a report containing the same to a client, and updating the data store accordingly; and a reconciliation process receiving data enabling reconciliation of data as tracked by the call center and as tracked by an entity other than the call center. Methods provided by the invention can be practiced using the systems described herein or with other systems.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: A receiver uses probabilistic techniques to identify the value of a received signal based upon a space of potential signals that may be generated by the transmitter. The receiver uses one or more iterations in which received signal values are compared with potential signal models to enable identification of the incoming signal.

Abstract: According to one embodiment, a communication apparatus includes a receiver which receives a Dual Tone Multi Frequency (DTMF) signal for specifying a control request is transmitted from a management apparatus, the control request including connection destination information for specifying a management apparatus, the management apparatus is connected to the public network and remotely controls the communication process, a calculator which obtains a control request from the DTMF signal, and a controller which creates a session for management using a known protocol passable through the firewall with respect to the management apparatus based on the connection destination information included in the control request, and allows the management apparatus to execute a process based on contents of the control request.

Abstract: The invented method involves one or more gateways sniffing the voice channel during the voice mode. If it is determined that V.8bis signals are being initiated, then the gateway breaks these exchanges by suppressing such signaling, thereby avoiding the above-described detrimental effects. Modem relay communications then are allowed to proceed. Briefly, the method and apparatus involve monitoring a call during a voice mode phase for an initiating signal representative of the defined signaling, the monitoring being performed by a local gateway and, if such an initiating signal is detected during the monitoring, then suppressing such detected signaling in such manner that the signaling does not reach the remote gateway. Preferably, the monitoring is for an initiating signal characterized by a dual tone of defined frequency and duration of approximately 1375 Hz and 2002 Hz for a duration of approximately 400 ms or 285 ms, in agreement with the ITU-T Recommendation V.

Abstract: A simple supplementary service protocol (SSSP) is provided for implementing functionally based supplementary services using the H.323 standard. According to the preferred embodiment, a generic and expandable protocol is provided for passing supplementary services information between H.323 entities for the provision of over twenty classes of functionally based supplementary services. The protocol of the present invention is relatively lightweight (i.e. not code intensive) compared to H.450 since it does not rely on ASN.1 encoding. In addition, the mixture of functional and stimulus protocol concepts in the design of the inventive protocol results in robust functional characteristics and implementation capability using only lightweight Endpoints.

Abstract: When it is determined that a sample queue exceeds a first predefined level, samples being received from a IP switched network are modified such that samples are removed within the voiced region of the samples by removing whole pitch periods of samples. If the sample queue is below a second predefined number, additional samples are placed into the queue by analyzing voiced samples from the IP switched network and generating additional pitch periods of samples.