Abstract: One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset.

Abstract: In one embodiment, a low cost, simple circuit for detecting an off-hook condition of a telecommunication line comprising tip and ring signal lines is provided. The circuit comprises a voltage divider for coupling between the tip and ring lines without an intervening transistor and having a node at which is presented a scaled version of a voltage across the voltage divider. The circuit further comprises a transistor having a control terminal coupled to the node, a first current flow terminal coupled to a voltage source, and a second current flow terminal coupled to an output terminal, wherein the output terminal bears a value that is indicative of a voltage across the tip and ring lines and thus whether the telecommunication line is off-hook.

Abstract: A telephone includes a transmit channel and a receive channel, each including a bank of sub-band filters having a VAD coupled one to each sub-band filter. Each VAD measures the spectral energy in a sub-band, compares the spectral energy to a first threshold, and produces an output signal representative of whether or not the first threshold is exceeded. The voice activity detector also includes a threshold circuit for calculating a dynamically adjustable noise threshold based upon averaged measured spectral energy. A wide band or system VAD monitors echo canceling circuitry to detect voice activity and double talk.

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: One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset.

Abstract: A system and method for routing and interacting with a dual mode network telephony environment are disclosed. In one embodiment, the system includes an external analog telephony line, a DSL modem, an analog telephony bridge controller, one or more additional analog telephony lines, and a communication center computer. The system routes analog and Voice-over-Internet Protocol calls between the analog telephony lines and the communication center computer. Furthermore, the communication center computer performs operations on calls inbound to the environment and calls outbound from the environment.

Abstract: In one embodiment, a low cost, simple circuit for detecting an off-hook condition of a telecommunication line comprising tip and ring signal lines is provided. The circuit comprises a voltage divider for coupling between the tip and ring lines without an intervening transistor and having a node at which is presented a scaled version of a voltage across the voltage divider. The circuit further comprises a transistor having a control terminal coupled to the node, a first current flow terminal coupled to a voltage source, and a second current flow terminal coupled to an output terminal, wherein the output terminal bears a value that is indicative of a voltage across the tip and ring lines and thus whether the telecommunication line is off-hook.

Abstract: Transmit equalization over high speed digital communication paths may be compensated in a receiver for a probe on that path. In one example, a probe input provides a signal from an electronic communications path, the signal having been processed by a transmit equalizer. A filter circuit processes the signal to compensate for the transmit equalizer, and a decoder decodes the processed signal and produces an output for use by test equipment.

Abstract: A terminal includes a key input unit having a key section and an output unit outputting information inputted by the key input unit. The terminal includes a touch detection unit which detects a touch of an object to the key section and detects a moving distance that the object moves on a surface of the key section while keeping the touch and a control unit which switches information outputted to the output unit according to the moving distance detected by the touch detection unit.

Abstract: The present disclosure generally pertains to systems and methods for identifying subscriber lines coupled to transceivers of interest. In one exemplary embodiment, an end of a subscriber line is coupled to and terminated by a transceiver of interest. The other end of the subscriber line is coupled to a speaker. The transceiver is then selectively activated and deactivated causing it to transmit its normal frequency spectrum in a predefined pattern. The frequency spectrum includes frequencies audible to humans. Thus, the speaker produces sound capable of being heard by a human, and the sound exhibits periods of noise followed by periods of silence. Indeed, the periods of noise and silence are in a pattern according to the activation/deactivation pattern of the transceiver. Thus, a user can identify the subscriber line that is coupled to the transceiver by identifying which of the lines is coupled to the speaker when the speaker emits noise and silence according to the predefined pattern.

Abstract: One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset.

Abstract: In one embodiment, a subscriber line interface circuit includes circuitry to generate metering pulses for insertion onto a subscriber line. The circuitry includes an audio path and a pulse metering path, where the pulse metering path includes a metering generator to generate a digital sine wave from a plurality of stored values, a first interpolator to interpolate the digital sine wave, a filter to filter the interpolated digital sine wave, and a second interpolator to interpolate the filtered interpolated digital sine wave.

Abstract: Methods and apparatus for controlling the audio characteristics of a networked voice communications device (NVCD) are presented. One method presented includes receiving a settings file, extracting at least one audio control parameter from the settings file, deriving audio processing parameters based upon a value selected from the at least one audio control parameter, and controlling the audio characteristics of the networked voice communications device using the audio processing parameters and the at least one audio control parameter. A method for providing audio parameters to an NVCD is also presented which includes establishing a settings file, which includes at least one audio control parameter, receiving a request to send the settings file, and sending the settings file over a network to the networked voice communications device.

Abstract: A telecommunication terminal receives an analog speech signal from a user of the terminal and converts the analog speech signal into a digital signal. A vocoder applies source coding to the speech signal and extracts reconstruction parameters from the speech signal. The reconstruction parameters are modified so that the transmitted voice associated with the signal is modified.

Abstract: Various embodiments are configured to transform characteristics of a communication signal. One embodiment is a method comprising decreasing amplitude of a first detected portion of the communication signal so that the decreased amplitude is in close proximity to a predefined specification; and increasing amplitude of a second portion of the communication signal so that the increased amplitude is in close proximity to the predefined specification, thereby resulting in a transformed communication signal.

Abstract: A VoIP telephone operates in an IP environment wherein at least a portion of the signaling from a remote gateway is included within a media session real time protocol stream. The VoIP telephone comprises a network communication system for encapsulating data into IP frames for exchange with remote devices over a frame switched network. A system client application is coupled to the network communication system and exchanges call set up messages with a remote VoIP gateway to establish a media channel for the exchange of the real time protocol streams. The system client also provides VoIP status signals to a presentation module. A dialog system is coupled to the network communication system and: i) translates frames of compressed digital audio data originated from a remote device to recreate remote voice band; ii) detects and compresses local voice band for transmission to the VoIP gateway; iii) detects in band signaling within the voice band; and iv) generates in band status signals to the presentation module.

Abstract: A dual mode packet phone is provided, comprising a first connector to connect the phone with a data network, and a second connector to connect the phone with a backup network. The phone automatically switches between the data and backup networks in the event that the data network fails to process the call in a timely manner. In a preferred embodiment, the phone is a Voice-over-IP phone, and comprises a voice processing unit, a control unit, a line interface, a data network interface, and a backup switch.

Abstract: A flexible data outlet includes a housing configured for placement in an operating area of a user, such as on a wall adjacent to a user's work space. User interface circuitry in the housing provides an interface to user equipment located in the user operating area, including a user telephone device and a user data device such as a personal computer. The user interface circuitry includes a link-layer interface to the user data device. Both wired and wireless user connections are supported. Premises interface circuitry in the housing provides an interface to premises equipment located generally outside the user operating area. The premises equipment includes data processing equipment and telephone communications equipment providing access to a public telephone network. Bridge circuitry in the housing provides communications connections between the user interface circuitry and the premises interface circuitry in a flexible manner, enabling the provision of a variety of information services to the user.

Abstract: A method (200) and device (100) for answering an incoming call to an electronic communication device enables the device to answer the incoming call using one of a plurality of answer modes, in response to when an answer call signal is input from a user of the device. The method (200) comprises allowing the device (100) to answer the incoming call, using a first answer mode, in response to an answer call signal input from a user at a user interface of the device before expiration of a first predetermined time period (step 205). The device is then allowed to answer the incoming call, using a second answer mode, in response to an answer call signal input from the user at the user interface of the device after expiration of the first predetermined time period (step 210).

Abstract: A telephone applicable to PSTN and IP network having functions of obtaining, when a call is originated by entering a PSTN telephone number, an IP network telephone number of a destination telephone corresponding to the PSTN telephone number from a telephone number translation server connected to the IP network, obtaining IP address information of the destination telephone corresponding to the IP network telephone number from a call agent connected to the IP network, establishing call connection with the destination telephone via the IP network by using the IP address information, and establishing call connection with the destination telephone via the PSTN if the IP network telephone number of the destination telephone cannot be obtained from the telephone number translation server.

Abstract: Techniques for enhancing the sound quality of computer telephony systems are provided. In general, digital signals including telephony sounds are converted from the time domain to the frequency domain. If noise is detected in the frequency domain conversion of the digital signals, the noise is removed utilizing a filter. The noise detection and filtering can be performed in software so that enhanced audio quality can be achieved at a relatively low cost and with flexibility for very diverse environments.

Abstract: In some embodiments, a method includes detecting a tone in each data frame of a sequence of telephony signal data frames. A first data frame of the sequence of telephony signal data frames may be transmitted immediately after detecting the tone therein. Transmission of a last one or last ones of the sequence of telephony signal data frames may be deferred. It may then be determined whether the tone is present in a next data frame that immediately follows the sequence of telephony signal data frames. If it is determined that the tone is not present in the next data frame, the last one or ones of the sequence of data frames and the next data frame may be transmitted. If it is determined that the tone is present in the next data frame, a respective replacement data frame may be transmitted in place of each one of the last one or last ones of the sequence of data frames and in place of the next data frame.

Abstract: The PBX system includes an analog to digital, and digital to analog, (A/D) conversion where the sample size and the sample rate are selectively adjustable. The A/D structure includes registers which hold values indicating the sample size and the sample rate for the conversion. When the A/D structure is configured, the A/D structure reads the values in the register and configures itself to the sample rate and sample size indicated by the registers. When the sample size and sample rate is to be changed, the values in the registers are changed and the A/D structure is reconfigured.

Abstract: The present invention provides a digital communication system which is free from noise generated in a receiver. A transmitter converts an analog input signal to a digital signal, and then a power level detector 5 compares the digital signal with a predetermined level. In accordance with the result, the transmitter transmits a transmission signal which includes the digital signal or a digital signal amplified from that digital signal, and a control bit indicative of the result of comparison. The receiver, in accordance with the control bit included in the received digital signal, outputs an analog signal converted from the received digital signal or an analog signal converted from the received digital signal after it is digitally amplified.

Abstract: A communication apparatus is used to interface signals with any one of a plurality of telephony networks having different impedances. The apparatus includes memory, a processor, a digital-to-analog (D/A) converter, and an interface port. Different sets of impedance control values are stored in the memory. The processor is configured to select one of the sets of impedance control values and to serially transmit the selected set of impedance control values during a communication session. The D/A converter is configured to receive the selected set of impedance control values from the processor and to transmit analog signals based on the selected set of impedance control values. The interface port is configured to interface signals with a communication connection of a telephony network.

Abstract: A method for optimizing the transmission properties and power loss of a high voltage part integrated in a subscriber line circuit for connecting a subscriber line wherein, during XDSL data transmission in the high voltage part, current sources which are integrated in the high voltage part and which supply current to the units present in the high voltage part for the purpose of amplifying or measuring telephone signals and/or data signals (which are situated above the frequency band provided for speech), are set, no later than when such data signals are received in the high voltage part, to current values which are higher than the current values for exclusive transmission of telephone signals situated within the voice band provided for speech, such that the data signals within the high voltage part are transmitted with a high bandwidth largely without distortions and the power loss of the high voltage part is optimized.

Abstract: A method of determining a codec mode for encoding a frame in a communications system, the method comprising the steps of: receiving a sequence of signal samples arranged in frames; analysing a current frame to select a codec mode appropriate for the current frame; predicting the characteristics of a subsequent frame using lookahead samples from the subsequent frame; and determining a codec mode for the current frame and the subsequent frame which suits the current frame and also suits a subsequent frame based on the predicted characteristics.

Abstract: In the case of the inventive telephone, telephone operating signals (bell signal, charge-unit signals and caller identifiers) and audio signals are processed together by an integrated circuit. This requires signal level matching to be performed for the individual signals. Since the telephone signals are processed together by a chip, preferably by a digital signal processor, the hitherto required complexity for discrete components can be reduced.

Abstract: Resistance in parallel with inductors in a series leg of the low-pass filter facilitates changing input and output resistance of the filter with little or no change in the reactance of the inductors. Furthermore, the reactance of the capacitors in the shunt legs of the filter will be substantially unaffected. This assists the designer in matching the impedance of the filter in the pass-band while still providing substantial impedance mismatching in the stop-band without substantially affecting the characteristics of the filter. Facilitating impedance matching in the pass-band and impedance mismatching in the stop-band is accomplished without the need for more complex active components. POTS splitters making use of such filters can facilitate impedance matching of line and load termination in a telecommunications system in response to signals within the frequencies of typical analog telephony service while providing impedance mismatching in response to signals within the frequencies of typical xDSL service.

Abstract: Cascade low-pass filters are useful in attenuating the xDSL band in a POTS splitter. The design of the low-pass filter is a sixth-order filter having two stages. A first stage includes a fourth-order filter, preferably with a stop-band frequency of approximately 48 kHz. A second stage includes a second-order filter in cascade with the fourth-order filter. For this filter, the stop-band frequency is preferably approximately 29 kHz. The inductance value of the second stage is relatively small in comparison to the inductance values of the first stage. In this manner, improvements in xDSL band attenuation are facilitated with little or no eroding of the voice band performance such as insertion loss, pass-band attenuation and return loss.

Abstract: A cost-effective filter consuming low power and occupying minimal space. The filter may be used in a ADSL modem (or CPE) to filter the signal components other than the ADSL signals. A high pass filter first filters the low frequency components to attenuate (or remove) lower frequency components such as that caused by ADSL transmit echo signals and that used for voice transmission. The high pass filter may be modified by adding a few resistors to limit the voltages of the high frequency signals also. The output of the high pass filter is amplified and passed through a low pass filter to filter the high frequency components (HPNA included). Due to earlier filtering operation of the high pass filter, the signal can be amplified substantially before being sent to the low pass filter. The implementation of the low pass filter is simplified due to the prior amplification.

Abstract: A switch based feature that allows subscribers to adjust their incoming or outgoing audio signal strength. The value of the transmit or receive pad values on the subscriber line card are adjusted to control the strength of the audio signal transmitted to, or received from, the network. The subscriber controls the feature by sending a hook flash to the switch, entering the DTMF keypad feature code and transmit or receive volume parameters, and then returning to the call.

Abstract: An integrated circuit is provided to convert an analog telephone signal into a digital format. The integrated circuit includes an analog-digital converter coupled to an averager. Sampled analog values are averaged at intervals and compared to a threshold level to determine a digital value.

Abstract: A full duplex speaker-phone that performs adaptive filtering to provide increased loop stability between users of the speaker-phone. The invention employs a sliding filter that performs adaptive frequency dependent attenuation of the speech signals within a speaker-phone. By detecting the mode of operation of the speaker-phone in real time, the speaker-phone performs adaptive filtering to ensure a high perceptual quality for the users of the speaker-phone. The sliding filtering is performed using a sliding low pass filter (LPF) in certain embodiments of the invention. Perceptually, the speaker-phone does not surrender the appearance of a full duplex speaker-phone from the perspective of the users. In addition, by performing frequency dependent filtering of the speech signals, the speaker-phone offers improved stability in the speaker-phone loop and reduces residual echoes significantly over conventional technology employed in speaker-phones.

Abstract: A method and apparatus for DTMF detection and voice mixing in the code-excited linear prediction (CELP) parameter space, without fully decoding and reconstructing the speech signal. The apparatus includes a Dual Tone Multiplexed Frequency (DTMF) signal detection module and a multi-input mixing module. The DTMF signal detection module detects DTMF signals by computing characteristic features from the input CELP parameters and comparing them with known features of DTMF signals. The multi-input mixing module mixes multiple sets of input CELP parameters, that represent multiple voice signals, into a single set of CELP parameters. The mixing computation is performed by analyzing each set of input CELP parameters, determining the order of importance of the input sets, selecting a strategy for mixing the CELP parameters, and outputting the mixed CELP parameters.

Abstract: An echo canceling device includes a first variable amplifier for variably amplifying an audio signal input through a microphone according to a control signal; a first analog-to-digital converter for converting an analog signal output from the first variable amplifier to a digital signal; a second digital-to-analog converter for converting a digital signal output from a digital signal processor to an analog signal; a second variable amplifier for variably amplifying a signal output from the second digital-to-analog converter according to a control signal; an input/output connector for connecting a signal output from the second variable amplifier to internal and external processing devices; a sound measurer for measuring a level of surrounding sounds input through the microphone; and the digital signal processor for executing an echo canceling routine, performing general digital signal processing and generating the control signals for controlling the first and second variable amplifiers according to the sound l

Abstract: An integrated telephony controller circuit with the ability to perform auxiliary functions is disclosed. The invention consists of a coder-decoder circuit (CODEC) including at least one analog-to-digital converter (ADC), said CODEC being capable of sending and receiving telephony data signals; a multiplexer coupled to said ADC; and a receiving means for receiving an analog sense signal, said receiving means coupled to said multiplexer; wherein said multiplexer allows said telephony signals and said analog sense signal to share said ADC. This integrated circuit provides a state driven means to share the use of the ADC depending on whether the primary telephony function is active.

Abstract: A telephone apparatus is disclosed which comprises: a receiving portion for receiving a telephone signal and generating a voice signal from the received telephone signal; a frequency conversion portion responsive to a control signal for frequency-converting the voice signal; and a reproducing portion for reproducing the voice signal from the frequency conversion portion. The frequency converting amount may be controlled externally or may be predetermined. The frequency converting amount data may cleared every communication for various circumstances. The circumference noise level and the frequency conversion amount data are stored when the frequency conversion is commended in an automatic mode during a communication. In the later communication, the frequency conversion may be effected automatically when the circumference noise level is larger than the stored circumference noise level.

Abstract: A switchboard device and methods of operation of same are disclosed. Embodiments of the invention may provide a flexible means of interconnecting wideband and narrowband communications interfaces, where wideband communications interfaces may transfer wideband data sampled at a higher sampling rate, and narrowband communication interfaces may transfer narrowband data sampled at a lower sampling rate. Data streams sampled at different sampling rates can be combined and the sampling rate of the result adjusted as needed by the destination interface. Methods of operating embodiments of the present invention are included. An additional aspect of the present invention may include machine-readable storage having stored thereon a computer program having a plurality of code sections executable by a machine for causing the machine to perform the foregoing.

Abstract: An interface apparatus interposed between a cordless telephone base unit and a personal computer sound card. Such interface emulates a Central Office connection with respect to the telephone and a mic and speaker connection with respect to the computer sound card. The apparatus is suitable for all cordless telephones regardless of frequency allocation standards because the interface occurs at the RJ-11 modular jack of the cordless phone base, normally intended for connection to a CO line. Differences in telephone line regulations that exist among countries are accounted for with no burden of complexity to the user of the present invention. The interface comprises active and passive circuits perform two-wire-to-four-wire conversion and signal conditioning. Local loop current may be obtained from batteries or a suitable AC power adapter.

Abstract: An apparatus and method to implement a highly efficient low power line driver. In a first embodiment, the invention provides a method to increase the power efficiency of a line driver. The method includes supplying a digital signal processor output to a first subtractor; supplying the first subtractor output as an input to a modulator of a line driver; subtracting the line driver output from the digital signal processor output at the first subtractor; filtering the line driver output with a low pass filter; routing the line driver output to an impedance match filter; providing a first analog-to-digital converter and a second subtractor to subtract the line impedance match filter output from the low pass filter output; providing a digital filter and a second analog-to-digital converter; and subtracting the digital filter output from the first analog-to-digital converter output at a third subtractor to output a feedback signal to the digital signal processor.

Abstract: A terminal adapter or a private branch exchange modulates a calling party number received from a digital subscriber line and sends a modulated signal to an analog terminal in the OFF period of a ringing signal or before the ringing signal is sent.

Abstract: A telecommunication terminal receives an analog speech signal from a user of the terminal and converts the analog speech signal into a digital signal. A vocoder applies source coding to the speech signal and extracts reconstruction parameters from the speech signal. The reconstruction parameters are modified so that the transmitted voice associated with the signal is modified.

Abstract: Cascade low-pass filters are useful in attenuating the xDSL band in a POTS splitter. The design of the low-pass filter is a sixth-order filter having two stages. A first stage includes a fourth-order filter, preferably with a stop-band frequency of approximately 48 kHz. A second stage includes a second-order filter in cascade with the fourth-order filter. For this filter, the stop-band frequency is preferably approximately 29 kHz. The inductance value of the second stage is relatively small in comparison to the inductance values of the first stage. In this manner, improvements in xDSL band attenuation are facilitated with little or no eroding of the voice band performance such as insertion loss, pass-band attenuation and return loss.

Abstract: A method and apparatus for controlling the DC line current on a telephone line and reducing the amount of error introduced to the system. The error is reduced by compensating for a DC error term introduced by an analog to digital converter having a DC offset. The DC offset is controlled digitally, allowing software to be used to limit the DC error in accordance with predefined parameters. Predefined parameters can be set to accommodate varying country specifications instead of using switches to control resistors and capacitors. In addition, changes in a country's requirements can be accomplished through software, instead of changing components or redesigning a circuit board. The use of software results in increased flexibility by allowing an infinite number of settings via software or software updates, allowing changes to be made quickly and easily throughout the world, and allowing error terms to be accommodated digitally.

Abstract: A control device (4) for connection of a telephone set (2) to at least one speaker (41) for reception of an audio signal and/or of at least one microphone (46) for transmission of an audio signal, comprising a digital signal processor (DSP) (10), an analog-to-digital converter (ADC) (11) to convert the digital signal coming from the telephone set (2) and/or from the microphone (46) to a digital signal that can be processed by the DSP (10), and a digital-to-analog converter (12) to convert the digital signal coming from the DSP (10) to an analog signal to be sent to the telephone set (2) and/or to the speaker (41), the control device (4) performing different functions, such as recognition of the voice conductors, storage and transmission of messages, equalization of the signal received and suppression of ambient noise.

Abstract: Techniques for enhancing the sound quality of computer telephony systems are provided. In general, digital signals including telephony sounds are converted from the time domain to the frequency domain. If noise is detected in the frequency domain conversion of the digital signals, the noise is removed utilizing a filter. The noise detection and filtering can be performed in software so that enhanced audio quality can be achieved at a relatively low cost and with flexibility for very diverse environments.

Abstract: A method and apparatus to detect off-hook and intrusion conditions on a telephone line. A voltage to frequency converter is used to convert the voltage measured between the TIP and RING terminals of a telephone line to a frequency. The voltage is continuously monitored, once during each sample period. The measured frequency is compared to predetermined threshold values to determine the state of the telephone line. The predetermined threshold values are updated in each sample period.

Abstract: An adapter (42) for connecting a telephone (52) to a telephone wire (40) has a subscriber line interface circuit (SLIC 113) with a jack for connecting a telephone to the SLIC, a line interface (110) for coupling to a telephone line, a signaling circuit (111) coupled to the line interface for receiving and providing subscriber line signaling on the telephone line, a codec (112) coupled to the SLIC, performing a voice coding/decoding function; and a timing circuit (202) for time multiplexing and demultiplexing of digitized voice signals between the SLIC and the line interface.

Abstract: A method and apparatus is provided for DC feed control for a line card. The method includes determining if the line card is operating in a current limit region of a DC feed curve and synthesizing a curve in the current limit region. The method further includes determining a loop voltage based on the synthesized curve, and applying the loop voltage to the subscriber line. The apparatus includes logic capable of determining if the line card is operating in a current limit region of a DC feed curve and capable of determining a loop voltage based on a synthesized curve in the current limit region. The apparatus further includes circuitry for applying the loop voltage to the subscriber line.