Abstract: A network-enabled user interface device, for example a VoIP telephony device, includes a display, a user input interface, an interface controller, and an application controller. The interface controller is configured for generating display elements for the display, controlling the user input interface based on received commands, and outputting responses to the application controller. The application controller is configured for outputting the commands to the interface controller based on application operations executed locally (i.e., within the user interface device), or remotely (e.g., by a server in communication with the user interface device). Remote application operations may include communications between the application controller and the remote server. The interface controller also may generate the display elements based on messages received from the network.

Abstract: An interface is provided between a conventional telephone and a conventional computer configured for network voice applications including a sound card and modem. The interface allows for use of the telephone during network voice applications such as IP telephony and voice chat. When the telephone is not used, the interface allows for sound to be input and output via the computer sound card and conventional computer microphone and speakers. When the telephone is picked up off-hook, the interface senses the off-hook state, causes the computer sound card to be connected to the telephone instead of the conventional microphone and speakers, and allows for network voice applications to be conducted using the telephone.

Abstract: A system and method for providing telephony services, such as caller ID, voice mail, selective call forwarding, and the like, to a remote subscriber, typically located outside the providers regional territory. The system can include customer premise equipment (CPE) selectively connectable to a remote central office (CO), a packet-switched network gateway, such as an Internet telephony gateway (ITG), for remotely communicating service requests to a local service node. The local service node can be located within the service provider's territory and returns telephony service responses via the packet-switched network gateway to the remote CO, and ultimately the remote CPE.

Abstract: The invention relates to a method for transmitting information such as fax, voice or data to a destination information device via a dynamic transmission network. The dynamic transmission network includes one or more transmission control servers (TCSs) and one or more transmission client nodes (TCNs). The TCS is always connected to the computer network, while the TCN is dynamically connected with the computer network. When connected to the computer network, the TCN checks its system status and then registers to the TCS. After getting the information from the TCN, the TCS stores said information into a dynamic transmission route table, wherein the information includes the country code, area code, Client ID, IP address, etc. of the TCN. Once any user requests the TCS to send a information to a destination site, the TCS will locate an available TCN physically nearest to the same country code and area code as those of the destination site, and then transmits the information to the TCN through the computer network.

Abstract: Service resource items (49) for use in call setup in a telephone system are held on servers (58) that are connected to a computer network which is logically distinct from the telephone system infrastructure; this computer network may, for example, make use of the Internet. Each service resource item (49) is locatable on the network at a corresponding URI and is associated with a particular telephone number. A mapping is provided between telephone numbers and the URIs of associated service resource items. When it is desired to access a service resource item (49) associated with a particular telephone number, this mapping is used to retrieve the corresponding URI (step 55) which is then used to access the desired service resource item (56, 57, 59).

Abstract: The present invention relates to method and system for providing online information in a networked user environment in which an end-user runs an application program and transmits data to an online server while running the application program. It is proposed to provide a request-button at the end-user application program dedicated to requesting information, and in particular help-information. When a help request is received at the communication server, a communication channel is promptly established between end-user and an agent. Information about the user activities sent in one or more transaction parts of an end-user intended business process and performed in the current application program session is read from the storage in the application server and is provided to the terminal of said agent in the help center. Advantageously, the same communication channel as used for performing the transactions is used for voice transmission for providing help or other information to the end-user.

Abstract: Phone systems for an integrated chat client service are provided. A representative phone system includes a call processing system operable to check a chat server for accessibility of a calling party chat client associated with a calling party and coupled to the internet, the call processing system being further operable to prompt a called party to select to talk or chat. Methods to connect telephone calls and answer telephone calls are also provided.

Abstract: An in-line power tap device that can be deployed in a data communication network in which two ports of the in-line power tap device are interchangeably coupled between any two communicating nodes to pass in-line power and data signals. The in-line power tap device also has a pair of tap ports that are coupled to external analyzers to monitor the quality of data signals without interrupting the on-going communication between the two communicating nodes. The apparatus allows one to tap into an Ethernet unshielded twisted pair data transmission line, or other source of data transmission, without disturbing the in-line power transmission and to maintain the integrity of data transmitted.

Abstract: A method is provided in an application server configured for responding to hypertext transport protocol (HTTP) requests. The method includes storing, in response to a first HTTP request, an XML document that specifies for a user, a call number of a second party. The stored XML document is retrieved based on a second HTTP request by the user. A first HTML document is generated based on the retrieved XML document. The first HTML document has instructions including the call number for accessing the second party. A second HTML document is generated based on a prescribed input received from the second party. The second HTML document has instructions for connecting the second party with the user. Hence, a user may speak with a called party without ever having to remain on hold.

Abstract: An Electronic Document Answering Machine is adapted to connect to remote information sources and to download digital documents including e-mail and WEB pages, and to store the documents for later review by a user. The Answering machine has an alert device for alerting a user that new documents are available for review, and an initializing device for a user to cause documents to be communicated for review one-at-a-time. In a preferred embodiment documents are communicated over a speaker via voice synthesis. In an alternative preferred embodiment a small display is provided wherein documents may be displayed in alphanumeric characters. The answering machine can be implemented as a stand-alone device or can be connected via a serial port to a host PC for transferring documents to the PC for processing and review. In other embodiments the Answering Machine is implemented as a system in a PC, with alert and selective input added as an interface.

Abstract: Phone systems for an integrated chat client service are provided. A representative phone system includes call processing system operable to check whether a called party has a called party chat client accessible through a chat server coupled to the internet. The call processing system is further operable to prompt a calling party to select to talk or chat with the called party when the called party chat client is accessible. Methods to connect and place telephone calls are also provided.

Abstract: If there is an incoming voice call to a telecommunication terminal equipment, a response cannot be instantly made to the call and a caller's telephone number and an electronic mail address associated with the caller's telephone number are registered with a telephone directory, then a document prepared in advance and a document automatically formed out of latest schedule data and time are automatically transmitted to the electronic mail address over electronic mail. Therefore, the latest schedule data of a called person is transmitted as a part of a content of a transmitted voice message or a content of document information and a caller can be notified of information.

Abstract: An apparatus and method for recording/reproducing voice messages using an internet gateway without separately procuring a VMS (Voice Mailing System). Upon receiving the voice message from a subscriber, the voice message is compressed in a data compressor of a digital signal processor (DSP) and stored in a flash memory. Then, the voice message is stored in a buffer of the DSP at a predetermined time interval, and subsequently read and decompressed by a data decompressor, and thereafter reproduced to a calling subscriber.

Abstract: A method for providing telecommunications services in a telecommunications network. A telecommunications service provider monitors data relating to communications initiation or termination criteria for at least two communicants in the telecommunications network and initiates or terminates communications between the at least two communicants if, by processing the data, it is determined that the criteria have been met.

Abstract: A method for high quality voice communication over an IP network. The method is implemented using an IP network device. Voice communication quality of service is initiated through the transmission of a path message for the voice communication. The path message is configured for establishing a communications path through the nodes of the IP network. A reservation message is received in response to the path message. The reservation message is configured for specifying a range of voice streams for a bandwidth reservation, allowing a single reservation message to specify bandwidth for quality of service for multiple voice calls. The bandwidth reservation for the range of voice streams is implemented in accordance with the reservation message. The path message can be transmitted from an originating IP network device, such as an originating VoIP gateway, and can be generated by a first voice application executing thereon.

Abstract: The present invention is drawn to a call trace system and method that is simple to establish in a short amount of time. The call trace is implemented through a Web browser page which remotely programs central offices to trace calls to particular number. When a trace is established a message is sent back to the Web browser which displays the calling number, the called number and the central office detecting the call. Also, a page is sent to a pager which displays the traced information.

Abstract: One or more elements in a telecommunications network (10, 10? and 10?) receive a call hold-initiating message when a first party places a second party to the call on hold, and in doing so, triggers a call hold event in the network. In response to the call hold-initiating message, the network element may take one or more actions, including releasing one or more network resources and/or providing a service to the second party during the period the call remains on hold. Upon resumption of the call, the network will re-acquire the resources that had been previously released.

Abstract: An enhanced IP telephone taking advantage of the many call-related functions provided by existing PBX systems. The enhanced IP telephone includes a thin computer telephony integration (CTI) client transmitting to a CTI server a command associated with a particular call-related function provided by a PBX unit. The CTI server translates the command to a PBX command, and transmits the PBX command to the PBX unit for providing the particular call-related function to the enhanced IP telephone. Instead of the thin CTI client, the enhanced IP telephone may include an enhanced IP signaling protocol stack for accessing the PBX functionality using its signaling protocol. A CTI translator translates the enhanced signaling command to a CTI command for transmitting to the CTI server.

Abstract: The invention discloses an Internet phone transmitting signals through a USB interface, which comprises a thin client and a digital phone. The USB interface connects the thin client and the digital phone. The phone communication is achieved through the online function of the thin client. The voice message sending procedure transmits outgoing message signals to the thin client through the USB interface. The thin client converts the outgoing message signals into an outgoing message VOIP package. The outgoing message VOIP package is then transmitted through a network to a target. The voice message receiving procedure transmits an incoming message VOIP package from the target back to the thin client through the network and converts the incoming message VOIP package into incoming message signals. Further through the USB interface, the incoming message signals are transmitted to the digital phone.

Abstract: A method and apparatus for effecting low cost audible communication between a local machine at a local site and a remote modem, over a wide area network. A remote modem, e.g. in a modem pool is configured to include a converter added to a telephone interconnection of a modem in the modem pool. As the remote modem receives communication(s) from the local machine the signal generated by the modem onto a local phone line for communication with the remote computer, is split and the converter receives the signal (heading to or from a WAN) from the telephone side of the modem. The converter provides an audio output that is input to sound processing hardware on an interface PC that runs an audio streaming program which packetizes and puts the audio signal back onto the WAN for transmission to the local computer.

Abstract: A virtual private network includes an internet protocol (IP) network and a public switched telephone network (PSTN). An enterprise gateway is operably connected to the IP network. The enterprise gateway is operably connected to a switch of the PSTN through a direct access line (DAL). The set-up signaling for virtual private network calls and the calls themselves are transported across the internet protocol network and the public switched telephone network through the direct access line.

Abstract: An intelligent network service control point, which is connected with a plurality of switching systems in a transport layer network over a common channel signaling network and with an Internet protocol network through a service control gateway, stores information for indicating that a first terminal is being connected to the Internet in a user information management table when the service control point receives a call waiting service request from the first terminal connected with the transport layer network. When one of said switching systems notifies that the first terminal is called from a second terminal, the service control point sends an incoming notification message to the first terminal through the gateway by referring the user information management table.

Abstract: A voice processing system is disclosed having a first interface for receiving telephone calls from the Public Switched Telephone Network (PSTN) and application software for processing received PSTN telephone calls. The voice processing system supports an API to allow the first interface to inform the application software of the presence of the call. The voice processing system further has a second interface for receiving telephone calls from an internet protocol (IP) network, and a signaling subsystem for routing an IP call from the second interface to the application software. This signaling subsystem uses the same API as is used to inform the application software about PSTN calls. Therefore, it is transparent to the application software whether an incoming call has been received from the IP network rather than the PSTN.

Abstract: In a communications network, a method and apparatus are disclosed for using a graphical user interface for telephony call control. A method is also disclosed for sending a datagram from a called party to a calling party. Call control of a call is diverted to a voice-mail system. The IP address of the caller is determined. The IP address of the caller is determined by either a search on a look-up table or from a voice over Internet call set up message. Call control information, or a datagram is sent to the caller. In the case of call control information, the caller inputs call control commands which are received by the voice mail system. The invention provides an easier method and apparatus to navigate voice-mail menus and reduces the problem of “voice-mail jail”.

Abstract: A backplane (11) of a telecommunications chassis includes a pair of controller slots that can receive respective controller modules. The backplane (11) also has a plurality of module slots that can receive respective telephony modules. Each of the pair of controller slots is connected by a plurality of buses to each of the plurality of module slots. The backplane (11) provides the ability to place any module type into any of the plurality of module slots.

Abstract: An efficient interface for service exchange is provided between a circuit-switched communications network and a domain naming system (DNS) in a packet-switched communications network. A node in the circuit-switched communications network receives from a first entity a request for a first service involving a second entity associated with a second entity identifier. A message is sent from the circuit-switched network node to a DNS server in the packet-switched network requesting a second service related to the second entity identifier. The result of the second service from the DNS server is used in providing the first service.

Abstract: A line card for a data packet router interfaces to a high-speed standard data link, and has a first portion interfacing to the router and having a plurality of slower ports, and a second portion having a framer compatible with and coupled to the data link. The framer is coupled through an ingress and an egress data path between the framer and the slower ports, each with separate ingress buffers and egress buffers for each port. An interface control circuit controls data packet transfers between the slower ports and the framer in both directions. In a preferred embodiment a function is used by the control circuit to map packets from the link to the ports, using keys extracted from the incoming packets. For an IP packet the key is the source address, destination address (SA/DA)pair, which constrains packets for same IP conversations to be routed by the same path.

Abstract: A method and system for dynamically selecting a destination gateway to complete a call over a path supported at least in part by an IP telephony network and a public switched telephone network. The method and system further provide for dynamically detecting available gateways, dynamically removing failed and/or unavailable gateways, and automatically recovering failed and/or unavailable gateways after a predetermined period of time. A method is also provided for detecting available destination gateways using a ping method, where a message is transmitted to a plurality of destination gateways on a one-by-one basis to ascertain the availability status of each destination gateway.

Abstract: A computer program, which is used to voice talking to cause an information terminal to execute voice talking, managing an ID of the information terminal and a first address over a first network, stores instructions for execution on a computer system enabling the computer system to perform determining that the information terminal moves over a second network which is different from the first network, acquiring a second address over the second network, and transmitting a request for re-registering a combination of the second address and the ID instead of the first address into another device having a combination of the first address and the ID registered therein.

Abstract: One embodiment of the present invention provides a system that facilitates developing applications with telephony functionality. The system includes a session initiation protocol (SIP) gateway configured to interface with a public switched telephone network (PSTN). The SIP gateway translates telephone calls from telephones coupled to the PSTN to SIP protocol messages. A SIP server coupled to the SIP gateway accepts these SIP protocol messages and an application server accesses a voice extensible markup language (VXML) page on behalf of the SIP server. A VXML gateway provides access to the VXML page by a users of the telephones.

Abstract: A teleconferencing system for voice and data provides interconnections among user sites via a central station. User stations at user sites each alternate operation between a data mode connecting a user computer and modem to a user telephone communication path and a voice mode connecting a telephony circuit to the communication path. The teleconferencing system is adapted for conducting a voice conference over standard telephone lines while allowing simultaneous viewing of data objects such as slides, graphs, or text. A host computer connected to the central station serves as a central repository for storage and retrieval of data objects for use in teleconferences.

Abstract: Perception of video clip is enabled in an instant messaging communications session by storing on a host system one or more video clips, receiving the request at the host system for a selected video clip to be delivered to an instant messaging participant system, accessing the selected video clip stored at the host system, and communicating the selected video clip from the host system to the instant messaging participant system for rendering in an instant messaging application running on the instant messaging participant system.

Abstract: A network processing system is described that is able to monitor IP network traffic, including the ability to perform trap and trace on IP communications flowing over the IP network. The network processing system is able to scan the entire contents of data packets passing through it, and to associate related data packets into discrete sessions, or flows, which allows the network processing system to search for predetermined search criteria contained within those flows. If a flow is found to contain a predetermined search criteria, the network processing system is able to maintain a record of the flow or to replicate the flow and save it or send it to another IP address for monitoring. The monitoring of a flow can include the entire contents of the flow, or any subset of information in the flow such as call identifying information.

Abstract: A system and method for implementing large and/or multiple telecommunication protocols utilizing memory-limited processors. Telecommunication protocols are expressed as virtual machine instructions defining a finite state machine. A telecommunication protocol engine implements the telecommunication protocols using a virtual machine operating under the control of a CPU. A telephony gateway comprising a telecommunication protocol engine interfaces with a communication device and a packet switched network. A network of telephony gateways communicating with a provider gateway facilitates communication over a packet switched network. A call placed by a calling telephony gateway to a receiving device on the same network or another network is routed according to a prefix in a telephone code sent by the calling telephony gateway.

Abstract: A system and method for integrating a legacy IP-PBX to a session initiation protocol (SIP) network supporting one or more SIP sets. A SIP-PBX proxy server coupled to the legacy IP-PBX and the SIP network maintains a logical IP set for each SIP set on the SIP network. The logical IP set is configured to maintain IP-based signaling and media connectivity with the legacy IP-PBX. The SIP-PBX proxy server receives media directed to a logical IP set associated with a particular SIP set, and redirects the media to the particular SIP set.

Abstract: When an operator of an Internet telephone inputs a telephone number, a server provides an IP address corresponding to the telephone number. Upon receiving the IP address and confirming a predetermined symbol in a host address portion of the IP address, the predetermined symbol is converted into a numeric value. Based on the converted IP address, a call destination is accessed. When the call destination is unavailable, the converted numeric value is incremented in order to generate another IP address. Then, the newly generated IP address is used to access another destination.

Abstract: A telephone line status notification system including at least one telephone line having a status, a communications network, at least one communications terminal which is connectable to the communications network and which is employable by a seeking user to communicate via the communications network a status request concerning the status of the at least one telephone line, apparatus for processing the status request the apparatus for processing is connectable to the communications network for receiving the status request from the seeking user therethrough and communicating the request, and apparatus for acquiring the status of the at least one telephone line, the apparatus for acquiring is in communication with the apparatus for processing for receiving the status request therefrom, and the apparatus for acquiring is connectable to the communications network for communicating the status via the communications network.

Abstract: The multimedia call distribution system maintains call connection devices, a multimedia call distribution device, and terminal devices. When a call connection request is received, each of the call connection devices issues an inquiry to the multimedia call distribution device about a terminal device to which a call is to be connected. The call connection device connects the call to the terminal device as a response to the inquiry by the multimedia call distribution device The multimedia call distribution device monitors communication conditions of each terminal device about respective types of media, and selects one terminal device that can preform real-time communication and informs the call connection device that has issued the inquiry of the selected device as a response, when the inquiry is received from the call connection device, and when a type of media of the call connected by the call connection device that has issued the inquiry is a type of media that preform real-time communication.

Abstract: A system and method for remote activation of customized telecommunication services are provided for data network telephony subscribers in a data network. In response to a call originator placing a call to a terminating telephony subscriber in the data network, a telephony server routes the call according to custom telecommunication service instructions provided in advance by the telephony subscriber and stored remotely from the telephony subscriber's point of presence on the data network. The call originator may be a data network telephony subscriber or a subscriber in the Public Switched Telephone Network (PSTN). The telephony server preferably performs the call routing based custom telecommunication service instructions (along with the subscriber's network address) retrieved from a subscriber profile server that maintains a subscriber profile database.

Abstract: This invention is a gateway apparatus arranged between a circuit switched network and the Internet. The gateway apparatus comprises a prior connection section setting a channel on the Internet according to the call setting request received from the circuit switched network before the call setting request is received from the circuit switched network, and a call setting section setting a connection corresponding to the call setting request on the Internet by using the channel set by the prior connection section when the call setting request is received from the circuit switched network.

Abstract: A data terminal, a data distributing system, and an Internet telephoning system are provided where desired data such as advertisements can selectively be determined and used for enjoying its relevant services. The data terminal receives a summary of service data from a data provider apparatus over the Internet, and if desired, demands the data provider apparatus to provide the service data over the Internet. The service data from the data provider apparatus may selectively be received in at least one of an audio form, a text form, and a video form at the data terminal. The service data may be advertisement data, commercial product data, and entertainment data such as music, movies, and games. The entertainment data may selectively be presented as at least one of the title, the artist name, the cost, and a trial sample piece of the content.

Abstract: A web based agent backed system provides sales and service and sales support for a company. The system includes a call center computer (112) coupled to the Internet (or another packet switched network) (118) that receives a call request from a customer computer (104), which may have been made using a link contained in a company web page maintained on a web server (108). Upon receiving the call request, the call center computer (112) determines customer information and call information and connects an agent through agent station computer (114) for call servicing. The call center computer (114) completes an IP telephony call made by the customer computer (104) to an agent or assigns an agent to call the customer by placing an IP telephony call to the customer computer (114) or by placing a PSTN call to the customer's phone (106).

Abstract: A switching system that enables the provision of a hold tone in a hold operation to be performed efficiently, comprising a first communication terminal (100) capable of packet communications, a second communication terminal (110) that performs packet communications with the first communication terminal (100), and a network (130) that connects the first communication terminal (100) and second communication terminal (110). When the first communication terminal (100) puts the second communication terminal (110) on hold, the first communication terminal (100) converts hold data (103) that it possesses to audio that can be interpreted by the second communication terminal (110) by means of a hold data conversion section (104), and transmits the converted audio as a hold tone.

Abstract: A directory service, containing unique identifiers (e.g., telephone numbers) and related Internet address(es), accessible through the Internet. An Internet-enabled communication system, such as IP-PBX or voice mail system, accepts a telephone number as a destination address. It then contacts the directory service and requests the Internet address related to that telephone number. The communication system uses the Internet address to set up a delivery path via the Internet to the destination. Thus, all time-dependent charges normally associated with message delivery by the telephone network can be avoided.

Abstract: A system and method for communicating screen display images on a computer to another computer using a telephony network. A screen share button on a data network telephone initiates a screen shot request to a first computer associated with the data network telephone. The first computer uses a screen shot application to retrieve the image on the computer display and send it to the data network telephone. The data network telephone receives the screen shot image and sends the image to another data network telephone which is a party to a telephone conversation with the first data network telephone. The first data network telephone preferably uses the SIP invite with a session description protocol that permits the transfer of image data on SIP messages. The second data network telephone implements the image data session description protocol and communicates the screen shot image to the second computer.

Abstract: A dial plan for a data network telephony system is provided. A variable-length-code scheme is used to assign communication numeral strings to user identifiers and/or device identifiers, such as SIP URIs (Universal Resource Identifier). The dial plan enables a user to enter a user identifier (or device identifier) into a conventional keypad of a data network telephone, such as an Internet telephony phone. Additionally, embodiments of the disclosed dial plan include techniques for dialing from one data network telephone to another data network telephone utilizing the same service provider, dialing from one data network telephone utilizing a first service provider to a second data network telephone utilizing a second service provider, and dialing from a data network telephone to a PSTN phone and vice versa.

Abstract: A telecommunications network environment (10) includes a signaling server (20) accessible by maintenance interfaces (12) and network interfaces (14). The signaling server (20) includes a signaling link controller (83) that receives and processes signaling information received from network elements and other signaling servers. The signaling link controller (83) extracts a data portion of the signaling information for message transfer part level two processing. For additional processing, the data portion of the signaling information may be routed over an Ethernet switch (76) to another signaling link controller (83) or to a signaling services controller (99) for signaling connection control part processing. The signaling services control element (99) may perform local number portability lookup and global title translation on the received data portion. After processing, the signaling link controller (83) generates a return message in response to the signaling information.

Abstract: In an automatic telephone call transfer apparatus in an Internet connection mode, an initial value setting portion sets a telephone number of an exchange of a telephone office which will perform a call transfer, a telephone number of an ISP to be subject to the call transfer, and an automatic/manual mode. An automatic call transfer setting/canceling portion transmits data corresponding to the telephone number of an exchange of a telephone office set at the initial value setting portion, call transfer request data, and the telephone number of the ISP to be subject to the call transfer at the connection of an Internet service, and at the termination of the Internet service, transmits the number of the exchange of the telephone number set at the initial value setting portion and data for canceling the call transfer. A modem is connected to a communication line.

Abstract: The present invention is drawn to a call trace system and method that is simple to establish in a short amount of time. The call trace is implemented through a Web browser page which remotely programs central offices to trace calls to particular number. When a trace is established a message is sent back to the Web browser which displays the calling number, the called number and the central office detecting the call. Also, a page is sent to a pager which displays the traced information.

Abstract: An Internet communication control apparatus and its transmission control method is provided, to which communication terminals such as facsimile apparatuses and/or telephones are connected, that is connected to the Internet to perform transmission/reception with other communication terminals. When an off-hook state of a connected communication terminal is detected upon transmitting, a setting for dial tone signal reception is prepared, and the received signals are converted to a destination number. By checking the corresponding table, the destination number is converted into a destination address, and according to the content of the table, whether the communication is for facsimile transmission or telephone call is determined. According to the result of the determination, a facsimile process or telephone process is selectively performed so that the transmission operation does not seem abnormal and complicated.