Abstract: Aspects of present disclosure relates a streaming signal for transmitting digital contents through a set of secure digital communication devices in a secure digital communication system. The streaming signal includes: an audio handshake signal and a streaming media signal. The streaming signal is transmitted between a first workstation and a second workstation in the secure digital communication system.

Abstract: Systems, methods, and computer-readable media for call classification and for training a model for call classification, an example method comprising: receiving DTMF information from a plurality of calls; determining, for each of the calls, a feature vector including statistics based on DTMF information such as DTMF residual signal comprising channel noise and additive noise; training a model for classification; comparing a new call feature vector to the model; predicting a device type and geographic location based on the comparison of the new call feature vector to the model; classifying the call as spoofed or genuine; and authenticating a call or altering an IVR call flow.

Abstract: A monitoring system and a method for operating the monitoring system in an inmate tracking system in a controlled environment is disclosed. The monitoring system receives video and audio data from devices located within the controlled environment and organizes the video and audio data within profiles that allow for searches of the video and audio to be performed. The monitoring system analyzes the video and audio data and generates the profiles to include identified objects associated with the video and audio data.

Abstract: Embodiments of the present disclosure include a microcontroller with a frequency test circuit, a device-under-test (DUT) input, and a calculation engine circuit. The calculation engine circuit is configured to compare a measured frequency from the frequency test circuit measured from the DUT input to a reference frequency stored in memory, and, based on the comparison, adjust frequency of the DUT generating the DUT input.

Abstract: A method and an apparatus for detecting an audio signal according to frequency domain energy is presented. The method may include receiving an audio signal frame; acquiring frequency domain energy distribution of the audio signal frame; obtaining a maximum value distribution characteristic of a frequency domain energy distribution derivative of the audio signal frame according to the frequency domain energy distribution of the audio signal frame; using the audio signal frame and each frame in a preset neighborhood range of the audio signal frame as a frame set, where the frame set includes a to-be-detected frame; and detecting the to-be-detected frame according to a maximum value distribution characteristic of a frequency domain energy distribution derivative of the frame set. In the various embodiments, detection on an audio signal can be implemented.

Abstract: The surrounding environment where an electronic apparatus is placed is detected according to external sounds, and therefore it is difficult to accurately switch the output of a ring-tone. A control apparatus controls the sound generated by the electronic device. The control apparatus includes a sound producing section that outputs a sound output signal causing a sound output section to generate the sound, a sound acquiring section that acquires a sound input signal generated from external sound, and an adjusting section that adjusts the sound output signal or the output of an output device according to the sound input signal acquired in a state where the sound is being generated.

Abstract: A system and method to compress transient signals are described. The system includes a transmitter disposed in a borehole to change a transmitted current and induce a current in an earth formation, and a receiver disposed in the borehole to receive the transient signals resulting from the transmitted current. The system also includes a processor to compress the transient signals, the processor compressing the transient signals based on a spline approximation method or an Eigen value method for each transient signal based on whether the spline approximation method or the Eigen value method results in a fewer number of parameters representing the transient signal.

Abstract: In a baseband processing circuit unit, a sample memory stores a received data sequence obtained by sampling a signal received by an RF receiving circuit unit at a given sampling time interval. A data sequence estimating unit estimates an estimated data sequence, which is obtained when the received signal is sampled at a sampling time shifted by a given shift time from the sampling time of the received data sequence, on the basis of the received data sequence stored in the sample memory. A correlation operation unit performs a correlation operation on the estimated data sequence estimated by the data sequence estimating unit 22 and a replica code. An integration unit integrates the correlation operation result from the correlation operation unit at a given integration time interval.

Abstract: A tone detector and associated method for use with EVRC-B and GSM vocoders to enable reliable detection of system connect tones over a wireless communication system. The tone detection method examines a number of sequential data frames of the signal received from the vocoder and determines that the tone is present if the spectral energy at frequencies around the tone is much higher than that at neighboring frequencies and if the calculated center frequency of the data frames is at or near the frequency of the tone.

Abstract: A computer system for processing audio on a call having a processor in communication with non-volatile memory having code for converting recorded audio into a vector, comparing the recorded audio vector with a plurality of audio vectors and generating a comparative match number for each comparison, identifying a closest audio vector having a highest comparative match number above a predetermined threshold match number, connecting the call to an agent when the closest audio vector is designated as live-voice.

Abstract: A method separates a multi-component signal by first estimating parameters of the signal. Then, using periodicity-based algebraic separation and energy-based demodulation, the signal is separated into components according to the parameters and constraints. Last, a Teager-Kaiser energy detector is applied to each component to provide a direct current signal for each component, and the constraint for each component used by the separating.

Abstract: Methods and systems for providing data communication in medical systems are disclosed.

Abstract: A call is placed to a communications device. A response to the call is received from a call recipient associated with the communications device. The received response is processed based on a set of parameters. Based on processing the received response, a determination is made whether the call recipient is a human or an answering machine. If the call recipient is determined to be a human, the call is handled in a first manner. If the call recipient is determined to be an answering machine, the call is handled in a second manner that is different from the first manner.

Abstract: While a dual tone multi-frequency (DTMF) signal can be easily processed if cleanly obtained, difficulties can arise if the DTMF signal is not cleanly obtained. In one example, part of the signal can be cleanly obtained, but part of the signal can be obstructed. In order to determine the content of the DTMF signal the parts that are cleanly obtained can be compared with known DTMF signals, such as DTMF signals that correspond with buttons on a telephone keypad. A classification of a closest matching known DTMF signal can be identified and that classification can be assigned to the partially obtained DTMF signal. Thus, without clearly obtaining the whole DTMF signal a content of the DTMF signal can be ascertained.

Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: A system for driving a first signal onto a wireline includes a driver and feedback circuitry. The driver drives a deterministic signal onto the wireline with an unknown impedance load. The feedback circuitry measures a quality of the output signal from the wireline and modifies a signal strength of the first signal based on the quality of the output signal. The driver then drives the first signal onto the wireline with the modified signal strength.

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: In one embodiment, a DSP having four arithmetic logic units (ALUs) and able to have two read/write operations per clock cycle performs silence detection and tone detection for data frames containing samples of an audio signal. The ALUs are used together in parallel to process the samples in the data frames received by the DSP. A received data frame is filtered by the silence detection so that substantially silent frames are dropped and non-silent frames are further processed. In the tone detection, a filtered data frame is processed, four samples at a time, to determine the power of the signal at a given frequency, where the power determination is used to determine whether a given tone (i.e., a signal at a given frequency) is present in the data frame.

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: A communication terminal apparatus is connected to a single telephone line and is configured to be selectively connected either to a telephone set or to a facsimile machine. A pseudo ring back tone generator generates a repetitive pseudo ring back tone transmitted to a calling party in response to an incoming call. A pseudo ring signal generator generates a pseudo ring signal for driving a ringer of the telephone set to ring in response to the incoming call. A tone detector detects a CNG signal indicative that the incoming call is a facsimile. A controller controls at least the pseudo ring back tone generator and the pseudo ring signal generator. A table correlates a plurality of ring back tones with corresponding countries. If a ring back tone for a country meets a predetermined condition, the controller sets a predetermined pseudo ring back tone to the pseudo ring back tone generator.

Abstract: Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality.

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: Techniques for processing an incoming message from premises equipment at a monitoring location. In one embodiment, an event message, including digits with varying amplitudes, is generated at a premises (such as a home, office, factory, etc.) in response to an event at the premises. A communication channel is established between the premises equipment and the monitoring location and the event message is sent from the premises equipment to the monitoring location. The entire event message is evaluated based on the number of digits received compared to expected receiving parameters defined in a communication protocol. A number of possible interpretations of the event message is determined. One or more possible interpretations of the event message are eliminated based on the number of digits in each of the possible interpretations to create a subset of possible interpretations.

Abstract: Methods and apparatus to decode dual-tone signals are disclosed. An example receiver to decode a dual-tone signal includes a tone detector to detect a start of a first tone, a first counter to count first cycles of the first tone, a second counter to count second cycles of a system clock while the first counter is counting and the first count is less than a first threshold, state control logic to start the second counter counting third cycles of the clock when a time period elapses, the third count being substantially equal to the second count, the first counter to count fourth cycles of a second tone while the third cycles are counted, and a decoder to compare the fourth count to a second threshold to identify an event represented by the signal.

Abstract: A tone detector is disclosed that is realizable in digital embodiment on a single integrated circuit die and does not require external components, such as a discrete capacitor. An input connects to a comparator, which in turn connects to one or more edge detectors and a flip flop. The edge detector outputs a pulse responsive to a detected edge. A counter is reset by the pulses from the edge detectors thereby preventing the counter from reaching a maximum value, which would otherwise be output from the counter and provided to a flip flop to clock in the comparator output at the D input to the flip flop. In operation, the comparator generates a rail to rail signal responsive to a received tone, which in turn is clocked through the flip flop as a logic high output indicating presence of a tone.

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: Systems and techniques for communication over telephony wireline. A command interface and control device may include a wireline input configured to connect the device to an operative line of a wireline telephony network (such as the PSTN) and to transmit voice signals to and receive voice signals from the wireline telephony network. The device may also include a cellular module configured to transmit signals to and receive signals from a cellular telephony network and a system controller to control communications on the wireline input and the cellular module. One or more wireline adapter modules may be configured to provide control functionality local to a telecommunications device.

Abstract: A tone signal received in a telecommunications network is bandpass filtered based on an identified type of the received tone signal, providing a bandpass filtered tone signal. The tone signal is also bandstop filtered based on the identified type of the received tone signal, providing a bandstop filtered tone signal. The bandpass filtered tone signal and the bandstop filtered tone signal are analysed and the received tone signal is quantified based on the analysis result.

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: A hook switch turns on a transistor by connecting the positive electrode terminal of a capacitor to a base terminal of a transistor in an off-hook state, causing a speech circuit and telephone lines to be connected. The hook switch turns off the transistor by disconnecting the positive electrode terminal of the capacitor from the base terminal of the transistor in an on-hook state, causing the speech circuit and the telephone lines to be disconnected. A holding circuit holds the voltage at the positive electrode terminal of the capacitor at a substantially fixed value by charging the capacitor with electric power supplied from the telephone lines in an off-hook state.

Abstract: A method of tracking and acting on events related to the delivery of a mail piece is provided. An integrated record associated with an addressee of a mail piece is created. The mail piece is given a unique identifier and the mail piece is associated with the addressee. The mail piece is placed in a mail stream and the date and nature of associated delivery events is recorded. After a predetermined delivery event occurs, a follow-up communication is transmitted to the addressee, and the date of this follow-up communication is recorded. The date of any response by the addressee to the follow-up communication is also recorded. To the integrated record associated with the addressee is added the date and nature of the delivery events associated with the mail piece, the date of the follow-up communication, and the date of any response by the addressee to the follow up communication.

Abstract: As part of a system and method for transmitting a signaling tone over a packet switched data (or other communication) network, an audio encoder may have a speech encoding unit and a tone distorter. The tone distorter may distort one or more components of a signaling tone within an audio signal and the speech encoding unit may encode speech and the distorted signaling tone within the audio signal. A control signal may be sent to a tone generator to reproduce the distorted tone.

Abstract: The apparatus may have: a telecommunication network having a tracking calling terminal and a tracked called terminal; a polled geofencing system associated with at least one predetermined area having a predetermined boundary, the tracked called terminal having a location relative to the predetermined boundary; and at least first and second predetermined ring-back tones having a functional relationship with the at least one predetermined boundary, the first predetermined ring-back tone being utilized when the tracked called terminal is within the predetermined boundary, and the second predetermined ring-back tone being utilized when the tracked called terminal is outside of the predetermined boundary; wherein the at least first and second predetermined ring-back tones alert the tracking calling terminal to a location, relative to the predetermined boundary as determined by the polled geofencing system, of tracked called terminal during a call setup between the designated tracked called terminal and the trackin

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: A sound information providing system, which can eliminate inappropriate sound information, is provided. The sound information providing system includes a receiving unit that receives a phone number for identifying a cellular phone of the destination side and a phone number for identifying a cellular phone of the source side. Both information is transmitted from the cellular phone of the source side, a sound information requesting unit for requesting sound information based on the received phone number. A sound information monitoring unit monitors the sound information are output in response to the request. It eliminates inappropriate sound information, and selects the sound information conforming to a predefined condition. A transmitting unit transmits the selected sound information to the cellular phone.

Abstract: In a call transfer of a call there is an original calling network device, an original recipient of the call, and a transferee of the call. Systems, network devices, and methods are provided for delivering local call transfer functionality. Each network device is capable of functioning in the capacity of any one or more of the above three roles, namely, originator, original recipient, and transferee by providing local call transfer functionality. There is no requirement for central processing equipment used to provide local call transfer functionality for transferring calls.

Abstract: Methods and apparatus to decode dual-tone signals are disclosed. An example receiver to decode a dual-tone signal includes a tone detector to detect a start of a first tone, a first counter to count first cycles of the first tone, a second counter to count second cycles of a system clock while the first counter is counting and the first count is less than a first threshold, state control logic to start the second counter counting third cycles of the clock when a time period elapses, the third count being substantially equal to the second count, the first counter to count fourth cycles of a second tone while the third cycles are counted, and a decoder to compare the fourth count to a second threshold to identify an event represented by the signal.

Abstract: A method and device for converting signals from a Human-Interface Device or similar input device to a stream of digital signals that are useable as inputs to an RS-232 or dual-tone multi-frequency (DTMF) circuit is provided. More specifically, the input from the input device may be received at a universal Serial Bus (USB) host and converted to an RS-232 or DTMF signal for ultimate use in controlling a remote device such as a camera in a video security system.

Abstract: The communication terminal device includes a handset, a central controller, a telephone network adaptor unit for converting analog signals transmitted/received between telecommunication network and the communication terminal device into digital signals, an interface unit physically isolating the central controller from the telephone network adaptor unit, and transmitting/receiving the converted digital signals between the central controller and the telephone network adaptor unit, and a telephone circuit unit which transmits the voice signals generated by processing the digital signals received from the telephone network adaptor unit to a handset through the interface unit and transmits the digital signals generated by processing the voice signals received from the handset to the telephone network adaptor unit through the interface unit.

Abstract: A method and system for outputting a call announcement based on presence information of a callee as a ringback tone is provided. The announcement system is provided with presence information of users. When the telephone system receives an indication of an incoming call from a caller, the telephone system identifies the callee. After the callee is identified, the announcement system retrieves the presence information for the callee and generates an announcement based on that presence information.

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: 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 system and method for processing a plurality of channels, for example audio channels, in parallel is provided. For example, a plurality of telephony channels are processed in order to detect and respond to call progress tones. The channels may be processed according to a common transform algorithm. Advantageously, a massively parallel architecture is employed, in which operations on many channels are synchronized, to achieve a high efficiency parallel processing environment. The parallel processor may be situated on a data bus, separate from a main general purpose processor, or integrated with the processor in a common board or integrated device. All, or a portion of a speech processing algorithm may also be performed in a massively parallel manner.

Abstract: A tone detector able to detect multicomponent tones for various tone formats is disclosed. An input signal is received and frequency and amplitude components, if any, of each sample of the incoming signal are estimated. In estimating the components, a current digit value is used to appropriately filter the input signal and estimate frequency and amplitude components. Using the component estimates, a new digit value is determined using a table in storage corresponding to the current tone format. The new digit value corresponds to the symbol whose components are closest in value to the estimated components corresponding to the current digit value. The new digit value is fed back as the current digit value and used in combination with a next sample of the input signal to determine a new digit value. A decision unit is used to process the digit values to determine when a valid symbol is detected.

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: A telephone may include a transmitter configured to transmit signals over a network a receiver configured to receive signals from the network a DTMF signal generator configured to generate DTMF signals, a DTMF gain adjustment module, and an amplifier module in communication with the DTMF signal generator and the DTMF gain selection module, where the amplifier module may be configured to amplify the DTMF signals. In one embodiment, the amplifier modules causes the DTMF signals to increase power by 10 dB so that DTMF signals being communicated over a network, possibly multiple packet networks, may be amplified if the DTMF signals are attenuated over the network and a remotely accessed telecommunications system, such as a voicemail system, cannot communicate with the telephone due to the attenuated DTMF signals.

Abstract: A communication system having an echo canceller is disclosed. One embodiment of the echo canceller includes an adaptive filter used to provide an estimate of reflected echo which is removed from the send signal. The echo canceller may also include a near-end talker signal detector which may be used to prevent the adaptive filter from adapting when a near-end talker signal is present. The echo canceller may also include a nonlinear processor used to further reduce any residual echo and to preserve background noise. The echo canceller may also include a monitor and control unit which may be used to monitor the filter coefficients and gain of the adaptive filter to maintain stability of the echo canceller, estimate pure delay, detect a tone, and inject a training signal. The echo canceller may also include a nonadaptive filter used to reduce the length of the adaptive filter.

Abstract: A method (300) for providing an audio message (105). The method can include detecting at least one external parameter (125, 130, 135). Detecting the external parameter can include detecting a time, a location or a station identifier. Detecting the location can include receiving a signal from a network node (110) or a beacon (115), or receiving location information from a positioning system (240). In one arrangement, detecting the external parameter can include detecting a plurality of identifiers, such as a time, a location and/or a station identifier. The external parameter can be processed to identify at least one audio attribute (265) associated with a voice persona. An audio message then can be generated based, at least in part, on the audio attribute. Identifying the audio attribute can include identifying at least a second audio attribute associated with a background sound.