308results about "Interconnection arrangements" patented technology

The method and systems for automatically identified currently viewing website domain name or a uniform resource locator (URL), receiving, by the server, said automatically identified website domain name or the uniform resource locator (URL), a request or an invitation for initiating a communication including live chat, puts the request for initiating a communication including live chat from the first user and the second user in a communication including live chat queue of the website or the uniform resource locator (URL) associated account, for available or identified or relevant agents to pick up, routing request or invitation for initiating a communication including live chat from the first user and the second user according to a pre-set rules and start a communication including live chat by available agent by selecting particular request for initiating a communication including live chat from the queued communication including live chats.

Systems and methods detect and prevent vishing attacks through an integrated framework combining SIP header customization, STIR / SHAKEN frameworks, AI / ML analysis, and real-time speech analysis using the Viterbi algorithm. The system begins with call initiation, embedding authentication information in the SIP header. The SIP data is transmitted and verified using STIR / SHAKEN frameworks, ensuring the authenticity of the caller's identity. Verified data is cross-referenced with third-party databases and analyzed by an AI / ML engine to detect anomalies. If potential fraud is detected, the call is blocked, and the customer is notified. Calls that pass initial checks are further analyzed using the Viterbi algorithm, which converts speech to text and identifies suspicious patterns. An anomaly pattern detector processes the converted text to detect vishing indicators, terminating the call if a match is found. This multi-layered approach ensures robust protection against vishing, enhancing the security and reliability of voice communications while safeguarding users from fraud.

Disclosed are example methods, systems, and devices for allowing caller computing devices to authenticate calls via a service provider computing system. Users may opt to have entities register to contact the user with a positive ID, icon, or other notification on the user's computing device transmitted by the service provider computing system. A caller computing device may use a unique security token of the user to activate the notification on the user's device. The user device may be used to exert control over the security token via a service provider client application running on the user device. The caller computing device may initiate authentication via an API call to the service provider computing system. The caller computing device is able to have items (text, images, documents, etc.) delivered to the user computing device if authenticated.

Systems and processes are disclosed for real-time fraud detection and prevention in in-person transactions. The invention utilizes an AI / ML engine to analyze customer application data for inconsistencies and unusual requests indicative of potential fraud. Concurrently, a real-time conversation analysis engine with speech recognition algorithms monitors interactions between bank associates and customers, identifying suspicious speech patterns, hesitations, and keywords associated with scams. By combining insights from application data and conversational analysis, the system generates a comprehensive risk assessment. When a high probability of fraud is detected, an alert notifies the bank associate, security personnel, and other relevant individuals. This proactive approach enables immediate action to prevent fraudulent transactions, reducing manipulation risks and minimizing financial losses. The system continuously learns from new data, adapting to evolving fraud tactics, thus providing robust, long-term protection for financial institutions and their customers.

The invention relates to an MCX group call service penetration packet implementation method. The MCX server comprises a call control module, a speaking right control module and a media distribution module; a plurality of MCX clients, wherein each MCX client comprises a call processing module, a speaking right processing module and a media processing module; the speaking right control module and the media distribution module are configured to receive and process a UDP penetration packet and update address mapping; the speaking right processing module and the media processing module are configured to send UDP penetration packets and process speaking right and voice streams. According to the invention, the MCX client is enabled to acquire the extranet address and the port of the MCX server for processing the speaking right and the voice stream by using the call signaling established by MCX group call, the speaking right SSRC and the voice SSRC are respectively used for identifying the speaking right of the MCX client and the voice stream of the MCX client according to the MCX protocol, the MCX client is enabled to send the fixed format UDP penetration packet with the SSRC identification to the MCX server, and the UDP penetration packet is enabled to be transmitted to the MCX server. And an intranet penetration function of speaking right and voice between the MCX client and the MCX server is realized.

The present application discloses an active echo canceller for a hearing device, where the hearing device comprises an output interface configured to output a first sound based on a far-end signal, an input interface configured to provide a plurality of input audio signals indicative of the first sound and a near-end sound, and one or more processors, a filter configured to: filter a first filter input signal based on a far-end signal using a first filter to provide a first filtered signal, the first filter configured to estimate an echo signal originating from a first sound; combining the first filtered output signal with a first input signal to provide a first output signal; filtering a second filter input signal based on the plurality of input audio signals using a second filter to provide a second filtered signal, the second filter configured to attenuate the near-end sound; combining the second filtered output signal with the first filtered output signal to provide an update signal; one or more first filter coefficients of the first filter are updated based on the update signal.

Embodiments of the present disclosure disclose apparatuses, methods, and computer programs for acoustic echo cancellation (AEC) based on one or more diagonal regularized correlation matrices. At least some example embodiments may allow for an intelligent and efficient solution for multi-channel AEC and / or stereo AEC, providing improved performance and simpler implementation.

Methods and systems related to supporting real time communications are disclosed. In one type of a real time communication session, prior to an offer and acceptance to perform real time communications, a communication channel may be established between a caller device and a callee device. Information related to addresses of the caller and callee devices can be contained in, or pointed to, by a Universal Resource Locators (URLs) for the caller and callee devices. A network computing device may communicate messages relating to establishing and maintaining the communication channel between the caller and callee devices. In a scenario where the network computing device becomes unavailable, another network computing device can use the URLs and other channel information that may be stored in the database, to handle messages.

Model-based updating of triage data includes receiving problem analysis data associated with a computing system and determining one or more reference codes in the problem analysis data that are associated with one or more defects. A portion of the problem analysis data to be used as training data for a model is determined based on data usage associated with the one or more defects. An indication of each of the one or more reference codes within the portion of the problem analysis data is input to the model. The model is configured to output one or more data confidence scores based on the one or more reference codes. An association of the one or more reference codes to the one or more defects is updated based on the one or more data confidence scores.

This invention provides a nurse call system, a nurse call method, and a nurse call program that can reduce patient waiting times. [Solution] The nurse call system according to this disclosure includes: a display means that, when a nurse call button located in a patient's room is pressed, displays the name of the patient who pressed the nurse call button and the elapsed time on a nurse's terminal and a display terminal; and a notification means that, when the nurse call button located in the patient's room is pressed, first sounds a notification tone on a nurse's terminal held by the nurse in charge of the patient; if the receiver button on the nurse's terminal held by the nurse in charge is not pressed within a predetermined time, sounds a notification tone on a display terminal located in the nurse's station; and if the receiver button on the display terminal is not pressed within a predetermined time, sounds a notification tone on a nurse's terminal held by each nurse in the ward.

To provide an entrance system capable of remotely controlling the operation of equipment connected to the entrance system by using a voice messaging circuit.SOLUTION: An entrance system according to the present invention includes a communication circuit and a control circuit. The communication circuit includes a SIM (SubscriberIdentityModule) and obtains out-of-band DTMF (DualToneMulti-Frequency) signals from the communication terminal and converts the out-of-band DTMF signals into control signals having a format suitable for operation of the control circuit. The control circuit operates the entrance system or operates a device connected to the entrance system based on the control signal.SELECTED DRAWING: Figure 1

Disclosed herein are embodiments of a hearing device. The hearing device can filter a first filter input signal to provide a first filtered signal, where filtering is performed to estimate an echo in a first input signal. The hearing device can combine a first filtered output signal with the first input signal to provide a first output signal, the first filtered output signal is based on the first filtered signal. The hearing device can filter a second filter input signal to provide a second filtered signal, where the filtering is performed to attenuate a near-end sound. The hearing device can combine the second filtered output signal with the first filtered output signal to provide an update signal, where the second filtered output signal is based on the second filtered signal. The hearing device can update one or more filter coefficients based on the update signal.

The invention discloses a method (500) for a cellular communications system (100), in which traffic is sent in frames (200), each frame comprising a first number of subframes (201-210), with a second number of said subframes being available for at least either uplink or downlink traffic. At least one of said second number of subframes is made to comprise at least three parts (515), as follows: • One part (520) which is utilized for uplink traffic, • One part (525) which is utilized for downlink traffic, • One part (530) which is utilized as a guard period, with said guard period part (525) being scheduled between the uplink and the downlink parts. The duration of at least two of said three parts (520, 525,530) may be varied to fit the current system need.

A user of a user device initiates a contact center engagement with a first contact center. The first contact center processes a prompt received, by a first contact center software service associated with a first entity. Determine, by the first contact center software service and based on stored data associated with the prompt or an account of the contact center user device, that the prompt relates to a second contact center software service associated with a second entity. Train, using online learning techniques, an artificial intelligence agent of the first contact center software service based on prior responses received from the second contact center software service during a merged contact session. The artificial intelligence agent is capable of providing a resolution to a user of the contact center user device without contacting the second contact center software service after a response is provided to questions in the merged contact session.

To provide an inexpensive telephone facility that can perform advanced call control including calling a designated telephone handset by utilizing the transmission function of the transmission line of an R-type fire alarm facility for call control. [Solution] A fire detector with a unique address is connected to a transmission line drawn from the receiver to the security zone, and a telephone line drawn from the receiver connects a telephone base unit installed in the receiver to a telephone handset installed in the security zone. The receiver is equipped with a telephone base unit control unit that controls the call connection of the telephone base unit, and a telephone repeater with a unique address that is connected to the transmission line and the telephone line and controls the call connection of the telephone handset. The telephone handset is remotely connected to the telephone repeater via a voice line, and when the telephone repeater receives a predetermined call control signal via the transmission line from the telephone base unit control unit, it connects the telephone line and the voice line to control the call connection between the telephone base unit and the telephone handset.

The invention provides a translation box and a real-time conversation translation method based on the translation box, which are applied to a fixed-line telephone, and comprise a telephone host port used for being connected with a telephone host; the telephone receiver port is used for being connected with a telephone receiver; the audio acquisition module is used for acquiring an analog voice signal; the analog-to-digital conversion module is used for converting the analog voice signal into a digital voice signal; the digital-to-analog conversion module is used for converting the digital translation signal into an analog translation signal; the communication module is used for accessing the Internet and transmitting digital signals with a cloud server; the audio output module is used for sending an analog voice signal to the telephone host or the telephone receiver; and the main control module is used for setting a translation process and controlling the audio acquisition module, the analog-to-digital conversion module, the digital-to-analog conversion module, the communication module and the audio output module to execute the translation process. The translation box can realize a real-time translation function with zero transformation access, and cross-language real-time dialogue can be realized by installing the translation box on an original fixed-line telephone by a single party of a talker.

A virtual telephony platform may establish at least one communication connection with one or more multiple modem array devices that are deployed in one or more geographical regions. The platform may be provided with access to multiple Subscriber Identity Modules (SIMs). The platform may provision a unique device profile to a particular modem in a specific modem array device deployed in a particular geographical region through a communication connection. The platform may further provide the particular modem with a specific SIM profile of a particular SIM of the multiple SIMs through the corresponding communication connection, such that the particular modem is configured to use the specific SIM profile to access a base station of a wireless carrier in the particular region. The platform may further route telecommunication data between a client device and the wireless carrier network in the particular region through the particular modem.

This application provides a call process implementation method and an apparatus, applied to the field of communication technologies, to enrich visual and auditory experience of a user by adding an electronic gift service or a virtual item service in a calling phase or a ringing and waiting phase of the user, so as to improve user experience, and meet diversified requirements of network users. The method includes: A terminal receives a media data stream corresponding to a virtual item, where the terminal is a recipient of the virtual item; and the terminal plays media information corresponding to the virtual item in a call process, where the terminal is a terminal of a caller user of a call service or a terminal of a callee user of the call service.

To provide an intercom device that can effectively prevent theft when it is installed on a wall surface, and an installation structure for the intercom device.SOLUTION: A magnet holder 17 that holds a magnet 16 is attached to a fixed frame 13, and a magnetic sensor 11 capable of detecting the magnet 16 is built in a main body case 2 of a collective entrance device 1 at a position facing the magnet 16. Then, a control unit 7 that controls the main operation of the collective entrance device 1 constantly monitors the magnetism of the magnet 16 using the magnetic sensor 11, and when the positional relationship between the magnetic sensor 11 and the magnet 16 changes in such a case, for example, that the collective entrance device 1 is about to be removed from the fixed frame 13, activates a speaker 3 to emit a warning sound.SELECTED DRAWING: Figure 2

We provide a resident monitoring system that allows for the monitoring of residents while respecting their privacy. [Solution] The resident monitoring system of this disclosure is provided in an intercom system 1 for an apartment building. The intercom system 1 comprises a main entrance unit 4, a plurality of entrance sub-units 2, a plurality of room master units 3, a management room master unit 5, and a control unit 6 that controls the intercom system 1. The room master unit 3 has a camera that takes pictures inside the room R1 and a transmitting unit that transmits the camera's image to the management room master unit 5. The camera generates an image consisting of point cloud data based on changes in brightness in the room.

To stably establish a communication with a terminal device.SOLUTION: A communication control device used in a medical treatment facility is connected with a terminal device so as to enable a communication via a first line and a second line. The communication control device sends a first calling signal via the first line to the terminal device upon detection of a calling from a calling terminal installed in the medical treatment facility, and sends a second calling signal via the second line to the terminal device after the sending or at the same time of the sending.SELECTED DRAWING: Figure 3

The application discloses a trusted business card calling system and method, comprising: an electronic authentication module, which is used for issuing a digital certificate for an enterprise or an institution with a first user; a business card issuing module, which is used for editing and exporting a vCard format file of the first user's business card information, and signing the business card of the first user's vCard format file by using a private key of the digital certificate to obtain a trusted business card, and publishing the trusted business card to a directory server for a calling terminal to download; the calling terminal and the called terminal; the directory server, which is used for authenticating other system structures, and providing specified access rights for the calling terminal and the called terminal, and is used for the calling terminal to download the trusted business card and the called terminal to verify the trusted business card; the calling terminal downloads the trusted business card of the first user from the directory server, and transmits the trusted business card to the called terminal for display. The application has the advantages of independent issuing by the enterprise or the institution, guaranteeing the authenticity of work role information, relatively low implementation cost, better compatibility with existing systems and applications, and the like.

The present invention relates to a method of augmenting a telecommunications call, the method comprising detecting a telecommunications call event at a user telecommunications device 10; selecting an item of media content 101, the item of media content 101 being associated with a uniform resource identifier or URI; augmenting the call with the item of media content 101; detecting a user interaction with the item of media content 101; and on termination of the call, performing an action in dependence on the resource identified by the URI. The invention extends to a corresponding system and apparatus.

The invention provides a device of an SIP telephone terminal integrated with an SIP server, and the device comprises an SIP client module which comprises audio input and output equipment and is used for being responsible for a basic voice call function; the SIP server module is used for managing all registered SIP clients, processing routing, transferring, group calling and conference conversation of calls, and providing a voice mailbox and call records; the SIP relay module is used for converting or forwarding internal and external SIP signaling and media data; the network interface module is used for providing wired or wireless connection options to be connected to a local area network and the Internet and is responsible for data transmission; and the user management and configuration interface module is used for providing user-friendly interface configuration and various functions of management equipment. In this way, the SIP server function and the SIP telephone function are integrated into the single device, multiple functions are integrated into the single device, the hardware cost and the installation cost are reduced, and the stability of the communication system is enhanced.

A call connection method and apparatus are provided, and are applied to a process of establishing a call between a calling side and a called side in a scenario in which a called terminal subscribes to a caller ring back tone business. The method includes: In a process of signaling interaction between a calling terminal and the called terminal, after an action of tapping answering by a user of the called terminal, if the called terminal has not received first ACK signaling replied by a network side (for example, an S-CSCF), but receives second INVITE signaling, in this case, the called terminal assumes that the network side has sent the first ACK signaling, and replies to the network side with second 200 OK (INVITE) signaling after receiving the second INVITE signaling. In comparison with a manner of replying 491 signaling in this case, the method in embodiments of this application reduces a probability of no audio in a call, can increase a probability of successful calls, and helps improve user call experience.

The systems and methods disclosed herein provide automated consent management for ensuring compliance with user-consent laws, rules, and / or regulations integrated with the automatic generation and transmission of autoreply messages. In accordance with the disclosed embodiments, a computer system may be configured to communicate with a user over both first and second communication channels, where the system requires the user's consent before it can communicate with the user over the second communication channel. The system may receive an incoming communication, such as a telephone call, from the user over the first communication channel. The computer system may automatically generate an autoreply message, such as a text message, to send to the user over the second communication channel in response to the incoming communication if the computer system automatically determines that the user has provided all necessary; user consents for sending the autoreply message over the second channel.

A monitoring system for communicating between a customer device operating in a tone-based reporting format and a central monitoring station (CMS) operating in an IP-based reporting format may include a message tone detector, an IP message generator, an IP acknowledgement detector, a handshake and acknowledgement tone generator, and a processor communicatively linked with each. The message tone detector may be configured to intercept one or more messages sent from the customer device towards the CMS. The IP message generator may be configured to generate and transmit a converted message to the CMS. The IP acknowledgement detector may be configured to receive one or more acknowledgements sent from the CMS. The handshake and acknowledgement tone generator may be configured to generate and transmit one or more acknowledgements to the customer device based on a received acknowledgement from the CMS.

A service registration method, a terminal, and a network side device are provided. The terminal includes a first subscriber identity card module associated with a first subscriber identity card and a second subscriber identity card module associated with a second subscriber identity card. The method includes: accessing a first mobile network based on the first subscriber identity card; and transmitting a voice over wireless fidelity (VoWiFi) registration message of the second subscriber identity card based on the first mobile network.

A computerized system is configured to automatically manage an incoming phone call from a caller to a cellular phone or an internet phone. The computerized system includes a first user device (e.g., a cell phone, internet phone, etc.) assigned to a first user. The system further includes an automated assistant module in communication with the first user device, a synthesized speech generator in communication with the automated assistant module, and a phone number directory in communication with the automated assistant module. The automated assistant module is configured to, when the incoming call is received by the first user device, determine (a) an importance of the incoming call based on (i) a location or number from which the incoming call originated, (ii) words used by the caller, (iii) a schedule of the first user, and (b) based on the importance of the incoming call, an appropriate response.