Abstract: The present invention is directed to a system and method for implementing an intelligent redial system. A method for storing redial data for a connection between a user phone system and a remote phone system may include establishing a connection with the remote phone system by the user phone system. A timer is started in response to the established connection. At least one digit is dialed by a user and corresponding time data of the timer with the dialed digit is monitored. The monitored digit and corresponding time data is stored. The stored monitored digit and corresponding time data are suitable for performing a redial.

Abstract: In an enhanced 911 (E911) response system, a callback monitor including a microphone and a speaker is connected to a telephone. When a caller dials 911 from the telephone, upon the telephone going on-hook, the callback monitor is activated to allow a 911 operator who calls back to detect sounds from the location of the caller's telephone without that phone ringing or being taken off hook. The callback monitor includes a timer; when the timer expires, the monitoring ends. In alternative embodiments, the timer can be extended.

Abstract: An H.323 client terminal (102) according to an embodiment of the invention employs primary and secondary H.323 control units or state machines (110a, 110b). The primary control unit (110a) sends signaling messages to a primary gatekeeper (108a) and triggers the secondary control unit (110b) to send a message with an appropriate correlation identifier to a secondary gatekeeper (108b). The primary control unit (110a) establishes a call per standard H.323 protocols. The secondary control unit (110b) also establishes a call via a secondary gatekeeper (108b). However, no media channels are established for this back up call. As the secondary control unit (110b) receives signaling from the secondary gatekeeper (108b), it checks the status of the call with the primary control unit (110a). If the call signaling on the primary control unit (110a) is proceeding normally, no further action is taken. If the call signaling with the primary gatekeeper (108a) fails, the secondary control unit (110b) takes over communication.

Abstract: Multiplex transmission apparatus 10 on the transmitting side multiplexes received signals from non-voice signal input output apparatus 14 and voice signal input output apparatus 16, and also transmits the multiplexed signal from communications network 30 by digital portable telephone 12 on the transmitting side. The digital portable telephone transmits a signal per 20 msec, so that multiplex transmission apparatus 10 alternately transmits the voice signal and the non-voice signal at an interval of 20 msec, for example. Multiplex transmission apparatus 20 sorts out the voice signal and the non-voice signal from the supplied signal by checking the error detecting code of the supplied signal. Multiplex transmission apparatus 20 can judge the type of the signal according to the type of the error detecting code, because a different error detecting code is added to each signal.

Abstract: Systems and methods for interfacing a service component written in any one of a variety of programming languages to a native operating system application program interface (API) are described. For example, in one embodiment, a generic interface between the Win32 API (application program interface) and Windows NT service components written in C, C++, and JAVA is provided. In one aspect, an interface module is configured to load a service component written in a non-native programming language. In another aspect, an interface module is configured to retrieve service component information from a configuration database.

Abstract: A TAPI method according to an implementation of the present invention includes installing a TAPI service provider and associating a first PID with the TAPI service provider; re-installing the TAPI service provider and associating a second PID with the TAPI service provider; and automatically associating TAPI devices that had been associated with the first PID with the second PID.

Abstract: A web-based call center and a method for its operation include multiple agent and customer terminals connected to a web server. The agent terminals and the customer terminals include a web browser for accessing web documents from the web server and annotation plug-in software for recording and replaying static and dynamic annotations on web documents. An HTML document registration tool formats web documents prior to storage on the web server, so that the web documents are formatted for annotation. An annotation server includes a database for storing annotated documents received from agent and customer terminals. Indexing software associated with the annotation server extracts key information from annotations received from customer terminals as part of callback requests. The extracted key information is indicative of a skills set which is required by an agent in order to handle the callback request. An ACD server utilizes the key information to route callback requests to qualified call center agents.

Abstract: A multipoint control unit coordinator (MCUC). The MCUC tracks all conferences in a telecommunications system and determines how they can be best configured and modified over time. The MCUC instructs multipoint control units (MCUs) to break down and reconfigure calls, if necessary. A MCUC according to an embodiment of the invention maintains a database of all the MCUs in the system, a measure of processing coding resources, and a geographical location. When two parties seek to add a third in a conference call, the MCUC determines coding resources, geographical locations, and determines the most appropriate media stream mixing location based on preferences, such as network cost or endpoint coding resources or quality. The MCUC then instructs the MCUs to handle the conference accordingly.

Abstract: The present method for selective web caching comprises the steps of generating a table, the table comprising a URL, a time of last access and a time stamp of a web page, the table further comprising a URL, time of last access and time stamp of elements found on the web page; when a request for a web page is made, checking the requested web page's URL and time stamp and URL and time stamp of the requested web page's corresponding elements with the URL and time stamp listed in the table to determine whether any modification has been made; and selectively downloading in a cache only those elements which are deemed to have been modified.

Abstract: A jitter buffer controller allows the depth of the jitter buffer to be adjusted dynamically according to the varying jitter of the current sequence. The contents of the jitter buffer are examined during a transmission. If the delay or average delay within the buffer drops to a predetermined threshold, then the size or depth of the jitter buffer is increased.

Abstract: A system integrates a hearing aid with devices such as wireless telephones, advantageously avoiding radio frequency (RF) interference. In one embodiment, a processor transforms an electrical signal to compensate for a hearing impairment. The function used for signal transformation can be accessed via a processor memory enhancement such as a smart card. A digital-to-analog converter (DAC) converts the transformed signal to an analog signal, which then goes to an amplifier and speaker. In other embodiments, an analog amplifier transforms the electrical signal.

Abstract: A system and method for filtering a jittered clock is disclosed. The filter of the present invention may check if a reference clock is received at a point in time within a pre-defined window of time. If the clock is received at a point in time within the window, the reference clock may be utilized as a reference for a system application such as a reference for a phase-locked loop. If the reference clock is not received at a point in time within the window, an interpolated clock representing an ideal received clock may be utilized. This may minimize the disturbance of system applications by blocking extreme deviations of the reference clock.

Abstract: Systems for dialing an emergency telephone number from a teleworking client according to the invention include apparatus that implement the steps of detecting at a teleworking client when an emergency number is dialed, disconnecting the teleworking client from the PBX/MLTS, connecting the teleworking client to the PSTN, and dialing an associated stored number. Though the invention is described with reference to a teleworking client, it may also be applied to any other dialup network connection.

Abstract: An IP Telephony Emergency Connections (ITEC) system and method that determines the precise origin of an emergency call and routes the call to the proper Public Service Answering Point (PSAP). A source-based routing mechanism is provided in an IP telephony type network, such as a VoIP or IP over LAN/ATM. Emergency calls are routed to the correct PSAP jurisdiction. Each server/switch may include the mechanism such that the IP Telephony network can identify an E911 connection and egress to a public network at a point closest to the emergency call point of origin. Whenever an emergency number call is made, the call's origin is determined during call setup establishment. Every port or end user jack in the network is assigned a Source Group Index (SGI), which is a number or index representing each PSAP jurisdiction in the network. All ports/jacks within the same PSAP jurisdiction are assigned the same SGI. Users may be in different areas of multiple PSAP jurisdictions.

Abstract: A system and method is presented for providing a ringback tone at the initiating end of a call while substantially reducing bandwidth usage. The initiating end initiates a call through a network. The receiving end of the call responds by transmitting an alerting message containing information on the characterization of a ringback tone to the initiating end of the call. At the initiating end of the call, the proper ringback tone is determined based on this information, and a ringback tone is generated. When the call is picked up at the receiving end, a transmission to the initiating end alerts the initiating end to cease the ringback tone. Silence suppression may be used at the receiving end to prevent the transmission of an audio stream before the terminal (e.g., a phone) is picked at the receiving end.

Abstract: A short cell multiplexing is provided for chiefly transmitting data shorter than the payload of a standard ATM cell (basically the data of less than 48 bytes, but the data more than 48 bytes can be allowed). A standard ATM cell assembler 1 converts various forms of input information into short cells, places the short cells in standard ATM cells efficiently considering their information length, and output them to a B/ISDN network 7. The standard ATM cell disassembler 2 disassembles the standard ATM cells, which are assembled by the standard ATM cell assembler 1 and is input through the B-ISDN network 7, into short cells, converts the short cells into those with the original input information forms, and outputs them to channels. The configuration makes it possible for the short cell ATM cell multiplexing to improve channel efficiency with small delay, and matching to the standard ATM system.

Abstract: A telecommunications system includes a network (101); a plurality of endpoints (102a, 102b) operably coupled to the network; a gatekeeper and a database engine, typically located in a gatekeeper server (108), for downloading database information such as a corporate directory, to the endpoints. The gatekeeper server and the endpoints further include load monitors (2002) for monitoring device performance. If the load is higher than a threshold, the gatekeeper will not provide a requested database information. The endpoint then requests the information from another endpoint.

Abstract: An apparatus and method are provided for a user to create a temporary e-mail address to which the user's mail can be addressed. The mail is then forwarded to the user's permanent e-mail address. The forwarding can be customized. In one embodiment, the temporary address expires after a predetermined amount of time has elapsed. In an alternative, after time-out, the forwarding expires.

Abstract: In an isochronous telecommunications system sending packets over two routes in a network, a server of the network compares overall transmission delay between the routes. If the new route is faster, a server buffers cells on the new route until the last cell on the old route is delivered. Then the first cell from the new route is delivered at the old delay and subsequent new route cells are delivered with the same delay. If the old route is faster, a server identifies the type of transmission; if it is voice, a signal is inserted to alert the receiver of delay, and if the transmission is video, the system resends the last frame from the old route until frames from the new route arrive.

Abstract: A telecommunications system according to an embodiment of the present invention includes an instant messaging (IM) server (106) and a VoIP server (110); and a plurality of system clients (112, 118, 120, 122, 124) having both VoIP and IM sub-clients. The system clients can log on to their respective servers in parallel. The IM server and the VoIP server can communicate lists of common participants and allow for IM conferences among at least subsets of the system clients while an audio or video teleconference is ongoing.