Method and Apparatus for Automatically Building Conversational Systems

A system and method provides a natural language interface to world-wide web content. Either in advance or dynamically, webpage content is parsed using a parsing algorithm. A person using a telephone interface can provide speech information, which is converted to text and used to automatically fill in input fields on a webpage form. The form is then submitted to a database search and a response is generated. Information contained on the responsive webpage is extracted and converted to speech via a text-to-speech engine and communicated to the person.

Tandem Access Controller Within The Public Switched Telephone Network

A system that includes a tandem access controller (TAC) coupled to the PSTN, where the TAC allows a subscriber to set-up and change the configuration of the phone line or other communications device, including selective call forwarding, using the web. The TAC is coupled internally to the PSTN in a local service area and is outside the central office of the subscriber. A calling party makes a first call to the subscriber using the subscriber's public telephone number. The TAC receives the first call prior to the call reaching the subscriber's terminating central office, which in some cases avoids a toll. The TAC then carries out the subscriber's instructions for the first call, such as making one or more second calls using telephone numbers different from the subscriber's public telephone number. When the second call is answered, the answering phone is connected by the TAC to the caller.

Voice instant messaging

Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

COMMUNICATIONS SWITCHING ARCHITECTURE

Presently described is a system and method for switching multimedia data communications, including but not limited to Voice over IP (VoIP) telephony, cable TV, digital audio and video. The system utilizes a single, integrated device to provide all PacketCable-compliant functionality, including enhanced user privacy, compliance with CALEA, E911 and other mandated services not available in conventional distributed PacketCable systems. High speed and efficient, low cost operation are provided by means of an optimized data unit encapsulation scheme for internal switching and routing. A proprietary fiber optic backplane and removable optical connectors are used to enable lightspeed internal communications hot-swapping of components. Furthermore, the present system is extensible to all forms of digital data switching and is secure, resistant to Denial of Service attacks, and fault-resilient. 116.-. (canceled)17. A packet-based telephony architecture comprising:a first addressable communication terminal configured for real-time packet-based communication;a second addressable communication terminal configured for real-time packet-based communication, the first and second packet telephone terminals communicatively coupled with a packet network;a switch in the packet network, wherein said packet-based telephony architecture is configured such that when a real-time communication session is established between said addressable communication terminals, each said addressable communication terminal is not provided the address of the other said addressable communication terminal, and the communication from each said addressable communication terminal during said session is instead directed to an address of said switch.18. The architecture of in which the first packet telephone terminal comprises a first Multimedia Terminal Adapter (MTA) claim 17 , and in which the second packet telephone terminal comprises a second MTA.19. The architecture of in which the switch provides its own ...

A telephone outlet for implementing a local area network over telephone lines and a local area network using such outlets

An outlet for coupling at least one data unit to digital data carried over wiring that simultaneously carry a packet-based serial digital data signal and a power signal over the same conductors. The outlet includes: a wiring connector for connecting to the wiring; a transceiver coupled to the wiring connector for transmitting and receiving packet-based serial digital data over the wiring; a LAN connector coupled to the transceiver for bi-directional packet-based data communication with at least one data unit; a bridge or a router coupled between the transceiver and the LAN connector for passing data bi-directionally between the at least one data unit and the wiring; and a single enclosure housing the above-mentioned components. The enclosure is mountable into a standard wall outlet receptacle or wall outlet opening, and the transceiver and the bridge or router are coupled to the wiring connector to be powered from the power signal.

Method and Apparatus for Automatically Building Conversational Systems

A system and method provides a natural language interface to world-wide web content. Either in advance or dynamically, webpage content is parsed using a parsing algorithm. A person using a telephone interface can provide speech information, which is converted to text and used to automatically fill in input fields on a webpage form. The form is then submitted to a database search and a response is generated. Information contained on the responsive webpage is extracted and converted to speech via a text-to-speech engine and communicated to the person.

Systems and methods for thermal mitigation with multiple processors

A wireless communication device for thermal mitigation with multiple processors is described. The wireless communication device includes a first communications processor that processes a data call. The wireless communication device also includes a second communications processor coupled to the first communications processor. The first communications processor performs a thermal mitigation operation by sending instructions to the second communications processor when at least one thermal threshold is reached. The second communications processor receives and executes the instructions.

NETWORK NODE, METHOD, AND MOBILE TERMINAL FOR PROVIDING VOICE CALLS TO A MOBILE TERMINAL IN A PACKET-SWITCHED-ONLY NETWORK

A network node, method, and mobile terminal for providing circuit-switched (CS) voice calls to a mobile terminal operating in a packet-switched (PS)-only network. A control node such as an EPC node in an SAE/LTE PS-only network is modified to mimic the messaging functionality of a Serving GPRS Service Node (SGSN). The EPC node is connected to an MSC/VLR in a CS network through a Gs interface. The EPC node registers the terminal with the MSC/VLR and provides a VLR TMSI to the terminal. When the MSC/VLR receives an incoming CS voice call for the terminal, the MSC/VLR notifies the EPC node, which sends a paging message to the terminal. To set up the CS voice call, the terminal sends a paging response directly to the MSC/VLR. 129-. (canceled)30. A method of providing a circuit-switched (CS) voice call to a mobile terminal operating in a packet-switched-only wireless telecommunication network , said method comprising:in a control node in the packet-switched-only network, emulating the messaging functionality of a Serving GPRS Service Node (SGSN);connecting the control node in the packet-switched-only network to a mobile switching center/visitor location register (MSC/VLR) in a CS network through an interface;receiving by the control node, a Paging Request from the MSC/VLR utilizing the interface; andsending an equivalent of a CS paging request message from the control node to the mobile terminal in the packet-switched-only network, the equivalent of the CS paging request message indicating the MSC/VLR has received an incoming CS voice call for the mobile terminal, wherein the equivalent of the CS paging request message causes the mobile terminal to send a CS paging response directly to the MSC/VLR.31. The method according to claim 30 , further comprising claim 30 , after the connecting step claim 30 , the steps of:registering the mobile terminal with the MSC/VLR by sending a first message from the control node in the packet-switched-only network to the MSC/VLR, said ...

Systems, Methods, and Computer Programs for Transitioning from a Phone-Only Mode to a Web Conference Mode

Various embodiments of systems, methods, and computer programs are disclosed for transitioning from a phone-only mode to a web conference mode. One embodiment is a method for participating in an audio conference comprising: a conferencing system establishing an audio conference between a plurality of participants via a communication network, at least one of the participants participating in the audio conference in a phone-only mode; and a server in communication with the conferencing system presenting to the phone-only participant, via a user interface on a computing device, a phone-only control menu for enabling the phone-only participant to selectively transition from the phone-only mode to a web conferencing mode. 1. A method for participating in an audio conference , the method comprising:a conferencing system establishing an audio conference between a plurality of participants via a communication network, at least one of the participants participating in the audio conference in a phone-only mode as a phone-only participant; anda server in communication with the conferencing system presenting a phone-only user interface to a computing device associated with the phone-only participant, the phone-only user interface comprising a phone-only control menu for transitioning from the phone-only mode to a web conferencing mode by replacing the phone-only user interface presented to the computing device with a web conference user interface presented to the computing device.2. The method of claim 1 , wherein the phone-only control menu comprises a list identifying the participants claim 1 , a mute/unmute control claim 1 , a disconnect control claim 1 , and a switch mode control for requesting the transition from the phone-only mode to the web conferencing mode.3. The method of claim 1 , further comprising:receiving a transition request via the phone-only control menu;in response to transition request, the server presenting the web conference user interface to the ...

Systems and Methods for Making Low-Cost International Phone Calls

Various embodiments are described herein that relate to computer hardware and programs for facilitating international calls between individuals. More specifically, an international call system can route voice traffic between multiple telephone network endpoints. For example, a client application (also referred to as a “client”) may be accessed by a traveler on a mobile phone. The client can be configured to connect to a network-accessible application server, which can communicate with a call processing system that routes voice traffic from the traveler's mobile phone to the mobile phone of a desired contact (and, in some embodiments, vice versa). The international call system described herein makes use of the local providers in each country traversed by the traveler. Consequently, low-cost calling can be enabled in both directions, while predictable reachability is guaranteed as the traveler moves between different local providers. 1. A computer-implemented method for routing calls between users of an international call system , the method comprising:receiving, at a client residing on a first computing device associated with a user located in a first country, a first user input indicative of a request to place a call to a second computing device associated with a contact located in a second country; 'wherein the call is routed to a first gateway over a Public Switched Telephone Network (PSTN);', 'sending, by the client to a phone application residing on the first computing device, an inter-process message that causes the phone application to place a call to an access number associated with the contact,'}causing the first gateway to send a first message to a call processing system across an Internet Protocol (IP) network; retrieving a destination number for the contact from an application server;', 'sending the first message to a second gateway over the IP network; and', 'causing the second gateway to place a call to the destination number that is received by a phone ...

Method for Automatically Transmitting an Imminent Event Via an Interface to a Terminal Point Associated with a User, and a Conversion Device Designed Therefor

We provide a computer-implemented method for automatic transfer of an upcoming event, which is stored in an event-processing entity () of a client () assigned to a user, through an interface to an endpoint () assigned to the user, comprising the following steps: 110-. (canceled)11. A computer-implemented method for automatic transfer of an upcoming event , which is stored in an event-processing entity of a client assigned to a user , through an interface to an endpoint assigned to the user , comprising the following steps:sending, from the client, first additional information concerning the upcoming event to a conversion unit,generating, from the conversion unit, an appointment reminder for the upcoming event and sending the appointment reminder to a telecommunication platform,processing, by the telecommunication platform, the appointment reminder, generating a signal, and sending the signal to the endpoint, wherein the signal is adapted to the endpoint, andprocessing, by the endpoint, the signal and the first additional information, accepting or rejecting the first appointment reminder as a result, and sending that result back to the client.12. The method of claim 11 , wherein the signals and information are sent by a CSTA Appointer as the conversion unit in an environment based on the Session Initiation Protocol and the Computer Supported Telecommunications Applications protocol claim 11 , to an endpoint based on analog claim 11 , Time Division Multiplexing claim 11 , or IP.13. The method of claim 11 , wherein the signals and information are sent by a connector based on a Universal Telephony Connector or Advanced Telephony Connector claim 11 , to a WebRTC-based client.14. The method of claim 11 , wherein the signals and information are sent by a cloud-based event-processing signal and information entity and a WebRTC application server claim 11 , and wherein event-related information is locally distributed by the client through a connector based on a Universal ...

Service System Supporting Voice Call Using Digital Assistant Device, Method Thereof, And Non-Transitory Computer Readable Medium Having Computer Program Recorded Thereon

The present invention relates to a service system supporting a voice call using a digital assistant device, a method thereof, and a non-transitory computer readable medium having a computer program recorded thereon, and more particularly, to a service system which performs a call between a digital assistant device and a terminal corresponding to a call opponent through a voice command of a user and supports call conversion between the digital assistant device and a user terminal through the voice command of the user, a method thereof, and a non-transitory computer readable medium having a computer program recorded thereon. 1. A service method supporting a voice call using a digital assistant device , the service method comprising:a checking step of checking, by a service server, a digital assistant device associated with a receiving number and receiver information included in a call processing request signal transmitted from a communication company server;a call connecting step of attempting, by the service server, a call connection to the digital assistant device checked based on the call processing request signal while attempting a call connection to a receiving terminal corresponding the receiving number checked based on the call processing request signal through the communication company server;a response step of requesting, by the service server, a call connection between a transmitting terminal and the digital assistant device to the communication company server based on a response signal received from the digital assistant device which recognizes a voice command for a call connection request of the receiver in response to the call connection attempt; anda call connection step of maintaining, by the service server, a call channel between the digital assistant device and the transmitting terminal formed by the communication company server in response to the requested call connection.2. The service method of claim 1 , further comprising:before the checking step, ...

PRIORITY CALLING USING CLOUD-BASED SIGNALING FOR VoLTE AND NON-CELLULAR CALLS

This Application describes mechanisms to provide priority levels for call connection requests and/or messages from a sending device to a receiving device. The receiving device is configured to allow prioritized notification alerts for incoming call connection requests and/or for incoming messages from a sending device to override a silent or do not disturb mode for one or more contacts based on i) a priority level provided with the call connection request or message and ii) a priority setting for the sending device or a set of devices associated with a user of the sending device.

Computer, Internet and Telecommunications Based Network

A method and apparatus for a computer and telecommunication network which can receive, send and manage information from or to a subscriber of the network, based on the subscriber's configuration. The network is made up of at least one cluster containing voice servers which allow for telephony, speech recognition, text-to-speech and conferencing functions, and is accessible by the subscriber through standard telephone connections or through internet connections. The network also utilizes a database and file server allowing the subscriber to maintain and manage certain contact lists and administrative information. A web server is also connected to the cluster thereby allowing access to all functions through internet connections.

Facsimile apparatus, communication apparatus, communication method and storage medium

Provided are a facsimile apparatus, a communication apparatus, and a communication method that perform communication using a session established according to a session initiation protocol (SIP). If an INVITE in which a media type is designated as “audio” is received, an INVITE in which a media type is designated as “image” is transmitted. If a success response is made to the transmitted INVITE, an image session is established and facsimile communication or image communication is executed. If an error response is made to the transmitted INVITE, calling processing for calling an operator is started.

TELEPHONY TERMINAL

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format. 1. An apparatus , comprising:a phone port configured to connect to a telephone;a network port configured to connect to a network interface that provides access to a data network; anda controller configured to provide a network voice service that converts information to and from a voice format of the data network and exchanges the information between the phone port and the network port.2. The apparatus of claim 1 , wherein the network port is configured to connect to the network interface via a cable.3. The apparatus of claim 1 , wherein the controller is further configured to provide a public switched telephone network service that processes telephone information from the phone port for transmission via a network device.4. The apparatus of claim 1 , wherein the phone port comprises a register jack 11 (RJ-11) port.5. The apparatus of claim 1 , wherein the phone port comprises a universal serial bus port.6. The apparatus of claim 1 , wherein the controller is further configured to provide a broadband service that processes data for exchange with a broadband network interface device.7. The apparatus of claim 1 , wherein the controller is further configured to process service-related data received via at least one of the phone port or the network port according to a service claim 1 , of a plurality of services supported by the ...

IMAGE PROCESSING APPARATUS, CONTROL METHOD THEREOF, AND STORAGE MEDIUM

An image processing apparatus registers voice call destination information and data transmission destination information for a destination in association with each other. The image processing apparatus executes a voice call with the destination using the voice call destination information, and executes, after the voice call, data transmission to the destination using the data transmission destination information in accordance with a data transmission instruction input by the user. 1. An image processing apparatus capable of executing a voice call and a data transmission , comprising:a registration unit configured to register voice call destination information used to execute the voice call in association with data transmission destination information used to execute the data transmission for a destination; anda control unit configured to execute a voice call with the destination using the voice call destination information, and to execute, after the voice call, data transmission to the destination using the data transmission destination information in accordance with a data transmission instruction input by a user.2. The apparatus according to claim 1 , further comprising a display control unit is configured to identifiably display a function which is executable for the destination in accordance with a content registered by said registration unit.3. The apparatus according to claim 2 , wherein said registration unit is configured to register a plurality of items of voice call destination information in association with a plurality of items of data transmission destination information for the one destination claim 2 , andsaid display control unit is configured to display any of the plurality of items of voice call destination information and the plurality of items of data transmission destination information to be selectable.4. The apparatus according to claim 2 , wherein said display control unit is configured to gray out a button used to instruct a function which ...

Method and apparatus for establishing communications with a remote node on a switched network based on hypertext calling received from a packet network

Disclosed is a system and a methodology enabling a first provider party to provide to a second recipient party software and services to enable the recipient party to embed in an Internet e-mail of the recipient party directed to an addressee, an HTML hyperlink file which includes a telecommunication initiating trigger, sending the e-mail to the addressee via the Internet, the addressee upon receiving the e-mail actuating the hyperlink via its trigger and, responsive to the trigger actuation, connecting the addressee via the Internet to a web page located on an Internet server, wherein the web page includes a request for a telephone number of the addressee. The addressee entering its telephone number information at the server, which information is communicated to a database associated with the server, the server communicating with the database and with a telephony switch, including sending to the switch instructions which result in the switch originating and managing a public switched telephone network (PSTN) communication between the recipient party and the addressee.

Communication System

When a first user intends a call with a second user via a packet switched communication service (e.g. VoIP), but the second user terminal or second user is currently unavailable to answer a call through the packet-switched service, then a call intent notification is sent to the second user terminal via an out-of-band communication channel, the call-intent notification notifying the second user about the intended call. Includes in the notification is a gateway PSTN number enabling the second user terminal to establish a hybrid PSTN call with the first user terminal via a connection between the second user terminal and a gateway over a PSTN network, and via a connection between the gateway and the first user terminal over the packet-switched network using the packet-switched communication service. 1. Equipment for use in relation to a communication service for establishing a packet-switched call between a first user terminal and a second user terminal over a packet-switched network , the first user terminal being used by a first user and the second user terminal being used by a second user; the equipment comprising:an interface for sending a notification to the second user terminal via an out-of-band communication channel not involving said communication service;memory comprising one or more memory units, the memory storing code; andprocessing apparatus comprising one or more processing units, the processing apparatus being arranged to run the code from said memory, wherein the code is configured so as when run on the processing apparatus to perform operations of:receiving an indication that the first user intends a call with the second user via said communication service;in response to the receipt of said indication, determining whether the second user terminal is currently unavailable to answer a call through said communication service, and/or determining whether the second user is currently unavailable to answer a call through said communication service;in response ...

METHOD AND APPARATUS FOR COST-SAVING TELEPHONE SERVICE

The present invention provides a method for a cost-saving telephone service, comprising: receiving a request for telephoning a third party client device from a communication resources using device, wherein the communication resources using device and the third party client device are connected to a first and a second telephone service networks respectively; selecting a communication resources providing device from a plurality of client devices registered to the cost-saving telephone service and connected to the second telephone service network; and having the communication resources providing device telephone the third party client device through the second telephone service network such that the communication resources using device communicates with the third party client device via the communication resources providing device. 1. A method for a cost-saving telephone service , which is performed by a server comprising:receiving a request for telephoning a third party client device from a communication resource using device, wherein the communication resource using device is a client device registered to the cost-saving telephone service, and is connected to the server through a data network, and the communication resource using device and the third party client device are respectively connected to a first telephone service network and a second telephone service network;selecting a communication resource providing device from a plurality of client devices registered to the cost-saving telephone service and connected to the second telephone service network, wherein the communication resource providing device is connected to the server through the data network; andhaving the communication resource providing device telephone the third party client device through the second telephone service network such that the communication resource using device communicates with the third party client device via the communication resource providing device.2. The method for the cost- ...

Method for faster connection of users using voice over ip communication

A method for establishing communication between a first communication device ( 110 ) of a first communication network ( 180 ) and a second communication device ( 120 a, 120 b ) of a second communication network ( 160 ) is disclosed. In one embodiment, a service provider ( 130 ) receives a message indicative of an intention of the first device to call the second device. A communication provider ( 140 ) in communication with the service provider establishes a communication channel with the second device via the second network based on the message received at the service provider and connects the communication channel established between the communication provider and the second device with another communication channel established in response to an incoming call from the first device to the communication provider via the first network. The establishment of the communication channel between the communication provider and the second device is initiated before establishment of the other communication channel is completed.

CALL CONTINUITY USING SHORT-RANGE COMMUNICATIONS HANDOFF

A call is made from a user equipment (UE) device to a first other UE device that is is paired to a short range communications device. The first other UE device monitors the respective connections between short range communications device and the first other UE device and a second other UE device and automatically hands over the telephone call from the first other UE device to the second other UE device when the connection between the short range communications device and the second other UE device is stronger than the connection between the short-range communications device and the other first UE device. The cordless headset 172 is concurrently handed off from the first other UE device to the second other UE device. 1. A device comprising:a call interface that receives call data from and provides call data to a first other device;a network interface through which a communication channel is established with a second other device;a short-range communication interface through which data is communicated between the device and a short-range communication device; and process the call data received through the call interface to establish a first communication channel between the device and the first other device;', 'control the network interface to establish a second communication channel between the device and the second other device and to provide the call data to the second other device via the second communication channel;', 'determine at least one of relative strength, latency or quality of service (QoS) of respective connections between the short range communication device and the device and between the short range communication device and the second other device; and', 'when the short range communication device is determined to have a better connection with the second other device than the short range communication device has with the device, transfer the call data to the second other device and concurrently hand off the short range communication device to the ...

BANDWIDTH UTILIZATION DURING SILENCE FRAMES

Briefly, in accordance with one or more embodiments, an apparatus of a user equipment (UE) comprises one or more baseband processors to generate a voice call invite message for a remote UE, and a memory to store the voice call invite message. The voice call invite message includes a header p-use-silence-period to indicate that the UE is configured to transmit or receive non-voice data during a silence period of the voice call. 130-. (canceled)31. An apparatus of a user equipment (UE) , comprising:one or more baseband processors to generate a voice call invite message for a remote UE; anda memory to store the voice call invite message;wherein the voice call invite message includes a header p-use-silence-period to indicate that the UE is configured to transmit or receive non-voice data during a silence period of the voice call.32. The apparatus of claim 31 , further comprising a radio-frequency (RF) transceiver to transmit the voice call invite message to the remote UE via a core network.33. The apparatus of claim 31 , wherein the header indicates a type of non-voice data to be transmitted or received during a silence period.34. The apparatus of claim 31 , wherein the one or more baseband processor are to generate one or more non-voice data packets to be sent to the remote UE during a silence period of the UE.35. The apparatus of claim 31 , wherein the one or more baseband processors are to decode one or more non-voice data packets received from the remote UE during a silence period of the remote UE.36. The apparatus of claim 31 , wherein the one or more baseband processors are to generate a Silence Indicator Description (SID) frame to be sent from the UE to the remote UE prior to sending one or more non-voice data packets to the remote UE during a silence period of the UE.37. The apparatus of claim 31 , wherein the voice call comprises a Voice over Long Term Evolution (VoLTE) call.38. The apparatus of claim 31 , wherein the non-voice data comprises message data claim ...

USING VIRTUAL SUBSCRIBER IDENTIFIERS IN COMMUNICATION SYSTEMS

Systems and methods for using virtual subscriber identifiers in communication systems. An example method may comprise: receiving, by a processing device, a first subscriber identifier and a second subscriber identifier, wherein the first subscriber identifier identifies a subscriber of a first communication system; producing a third subscriber identifier based on the first subscriber identifier and the second subscriber identifier, wherein the third subscriber identifier identifies a subscriber of a second communication system; and causing a communication session to be established between the subscriber of the first communication system and the subscriber of the second communication system. 1. A method comprising:receiving, by a processing device, a first subscriber identifier and a second subscriber identifier, wherein the first subscriber identifier identifies a subscriber of a first communication system;producing a third subscriber identifier based on the first subscriber identifier and the second subscriber identifier, wherein the third subscriber identifier identifies a subscriber of a second communication system; andcausing a communication session to be established between the subscriber of the first communication system and the subscriber of the second communication system.2. The method of claim 1 , wherein the second subscriber identifier is provided by a virtual identifier.3. The method of claim 1 , wherein producing the third identifier comprises retrieving the third subscriber identifier from a memory data structure associating the third subscriber identifier with a combination of the first subscriber identifier and the second subscriber identifier.4. The method of claim 1 , wherein at least one of: the first subscriber identifier claim 1 , the second subscriber identifier claim 1 , or the third subscriber identifier comprises a public switched telephone network (PSTN) phone number.5. The method of claim 1 , wherein at least one of: the first subscriber ...

METHOD AND SYSTEM FOR LEAST COST ROUTING (LCR) OF INTERNATIONAL MOBILE TELEPHONE CALLS WITH AN INTEGRATED MONEY TRANSFER FACILITY

A method and system for least cost routing for mobile telephone calls between the United States and Mexico in such a way so that the call is separated into several call legs and the international portion or call leg is routed via a low-cost or no-cost route independently of the local call legs, essentially achieving an international call with local call rates. This uses a SIM card that automatically identifies any incoming or outgoing call and provides the ability to convert the call from an expensive international and/or roaming call to a local (free) call. Several slight variations are disclosed, and all embodiments have an integrated money transfer service that can be used with a very simple low-cost basic mobile phone using IVR, SMS and voice recognition, or can be accessed via the web or via a smartphone app. 1. A method for transferring money between a country and a foreign country , comprising the steps of:pre-establishing a target area code in foreign country; and recognizing that a call to a destination number is being initiated to said target area code,', 'completing the call over a data connection via Session Initiation Protocol (SIP);', 'communicating SIP call control information to a local control center including called number, calling number, HLR and VLR information, said local control center carrying out the steps of;', 'recognizing and authenticating said calling number and terminating and parking said call,', 'formulating a call setup request to the called number in said foreign country,', 'automatically determining a plurality of lowest cost routing options,', 'transmitting said routing options and their costs to the SIM card application,', 'initiating a new call to the destination number via a selected routing option,', 'terminating on the destination number,', 'merging the parked call and with the call terminating on the destination number, and', 'transferring funds on said the call terminating on the destination number.', 'to any one of an ...

TELEPHONY TERMINAL

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format. 1. An apparatus , comprising:a network port configured to connect to a data network; a public switched telephone network service configured to process analog telephone data received via the phone port,', 'a network voice service configured to convert information to or from a network voice format of the data network and exchange the information with the network port, and', 'an electronic mail service; and, 'a controller configured to, in response to receipt of header data from a phone port of the apparatus, initiate a service specified by the header data from a group of services supported by the apparatus, wherein the group of services comprises at least'}a user interface configured to display at least a portion of the information.2. The apparatus of claim 1 , wherein the analog telephone data comprises voice information claim 1 , and the group of services further comprises a voice over internet protocol service configured to convert the voice information to digital data and store the digital data in an internet protocol packet.3. The apparatus of claim 1 , wherein the public switched telephone network service is configured to process the analog telephone data for transmission via a network device of a public switched telephone network.4. The apparatus of claim 1 , wherein the group of services further comprises a browser ...

WORKLOAD BALANCING TECHNIQUE FOR A TELEPHONE COMMUNICATION SYSTEM

A workload balancing technique enhances balancing of workloads processed by components of a telephone communication system configured to render telephony services over a computer network. The telephony services illustratively include private branch exchange (PBX) services suitable for customers with geographically dispersed telephone service demands. The workload balancing technique is invoked by the communication system to reduce contention among the components of the system, such as telephone access servers (TASes) coupled to a proxy router, when servicing telephony requests issued by the customers, thus enabling rendering of the telephony services in an efficient and cost-effective manner. As described herein, an available TAS is randomly selected to service a telephony request received by the proxy router from a network coupled to the communication system. 1. A method comprising:configuring a plurality of components of a telephone communication system to render telephony services, wherein the components include a plurality of telephone access servers (TASes) coupled to a proxy router via a plurality of proxy servers;in response to receiving a telephony request, transmitting a first capability message from the proxy router to the proxy servers;in response to a first responding proxy server that responds to the first capability message, selecting the first responding proxy server to receive a first invitation to service the telephony request;forwarding the telephony request to the selected proxy server upon receiving a response to the first invitation;randomly selecting a subset of the plurality of the TASes to receive a second capability message transmitted by the selected proxy server;in response to a first responding TAS that responds to the second capability message, selecting the first responding TAS to receive a second invitation to service the telephony request; andforwarding the telephony request to the selected TAS.2. The method of further comprising ...

TELEPHONY TERMINAL

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format. 1. An apparatus , comprising:a phone connection configured to connect to an analog telephone;a network connection configured to connect to a network that provides access to a public switched telephone network and to a data network; anda controller connected to the phone connection and the network connection, a voice over internet protocol service that converts a voice signal received over the phone connection to an internet protocol packet containing voice data and sends the internet protocol packet to the network connection,', 'a public switched telephone network service, and', 'a broadband service., 'wherein the controller is configured to provide'}2. The apparatus of claim 1 , wherein the controller is further configured to claim 1 , in response to receipt of header data over the phone connection claim 1 , initiate a service specified by the header data from a group of services comprising the voice over internet protocol service claim 1 , the public switched telephone network service claim 1 , and the broadband service.3. The apparatus of claim 1 , wherein the public switched telephone network service processes analog telephone data received over the phone connection for routing to the public switched telephone network via the network connection.4. The apparatus of claim 1 , wherein the broadband service processes data ...

METHOD AND APPARATUS FOR ESTABLISHING INTERNETWORK COMMUNICATION BETWEEN TELECOMMUNICATION DEVICES

Systems and methods for connecting long distance and international calls between two telecommunications devices, without incurring tariff charges, are disclosed. 1. A telecommunication method comprising:retrieving, using a telecommunications system, an electronically stored first identifier associated with a first device;retrieving, using the telecommunications system, an electronically stored second identifier associated with a second device;temporarily assigning, using the telecommunications system, first local access information to the first device;associating, using the telecommunications system, the first local access information with the first identifier;temporarily assigning, using the telecommunications system, second local access information to the second device;associating, using the telecommunications system, the second local access information with the second identifier; andestablishing a bridged communication, under the control of the telecommunications system, between the first device and the second device using the first local access information and the second local access information.2. The method of claim 1 , wherein the first identifier and second identifier each comprise one or more of a phone number claim 1 , SIM card number claim 1 , ESN claim 1 , IMEI number claim 1 , IMSI number claim 1 , proxy ID claim 1 , personal user ID claim 1 , mobile device user ID claim 1 , application user ID claim 1 , MAC address claim 1 , and biometric information.3. The method of claim 1 , wherein the first identifier is selected from one or more items maintained in a database accessible from the first device.4. The method of claim 1 , wherein the local access information is a local access telephone number.5. The method of claim 1 , wherein the local access information comprises multiple local access telephone numbers claim 1 , each associated with a different network.6. The method of further comprising wherein communication between the first device and the second ...

WORKLOAD BALANCING TECHNIQUE FOR A TELEPHONE COMMUNICATION SYSTEM

A workload balancing technique enhances balancing of workloads processed by components of a telephone communication system configured to render telephony services over a computer network. The telephony services illustratively include private branch exchange (PBX) services suitable for customers with geographically dispersed telephone service demands. The workload balancing technique is invoked by the communication system to reduce contention among the components of the system, such as telephone access servers (TASes) coupled to a proxy router, when servicing telephony requests issued by the customers, thus enabling rendering of the telephony services in an efficient and cost-effective manner. As described herein, an available TAS is randomly selected to service a telephony request received by the proxy router from a network coupled to the communication system. 1. A method comprising:configuring a plurality of components of a telephone communication system to render telephony services over a computer network, wherein the components include a plurality of telephone access servers (TASes) coupled to a proxy router; andin response to receiving a telephony request at the communication system, routing the telephony request on a random basis to an available TAS to service the telephony request.2. The method of wherein routing the telephony request comprises:determining whether there are TASes available to service the telephony request and, if there are no available TASes, dropping the telephony request; andif there are available TASes, randomly selecting the available TAS to service the telephony request.3. A method comprising:configuring a plurality of components of a telephone communication system to render telephony services over a computer network, wherein the components include a plurality of telephone access servers (TASes) coupled to a proxy router via a plurality of proxy servers;in response to receiving a telephony request at the proxy router, forwarding the ...

Video Calling and Conferencing Addressing

Novel tools and techniques are provided for implementing video calling and conferencing addressing. A computer might receive a call request from a caller. The call request might include a callee address in a first protocol. In some cases, the callee address might include a uniform resource locator (“URL”) associated with the callee, and the first protocol might be hypertext transfer protocol (“HTTP”). The computer might determine a callee at a calling destination based at least in part on the URL associated with the callee or callee's device(s), and might establish a voice or video call between the caller and the callee based at least in part on the URL associated with the callee (or callee's device(s)). In some cases, the computer might map the callee address in a second protocol (e.g., SIP, XMPP, PSTN protocol, etc.) different from the first protocol, and might call the callee using the second protocol. 1. A method , comprising:receiving, at a computer and from a caller, a call request including a callee address in a first protocol, wherein the first protocol is hypertext transfer protocol (“HTTP”), and the callee address includes a uniform resource locator (“URL”) associated with the callee;determining, with the computer, a callee at a calling destination based at least in part on the URL associated with the callee; andestablishing, with the computer, a call between the caller and the callee based at least in part on the URL associated with the callee.2. The method of claim 1 , wherein the computer is a server located over a network.3. The method of claim 1 , wherein the URL associated with the callee is a URL associated with at least one device associated with the callee.4. The method of claim 1 , wherein establishing the call between the caller and the callee based at least in part on the URL associated with the callee comprises:mapping, with the computer, a callee address in a second protocol different from the first protocol;logging into a calling server that ...

SYSTEMS AND METHODS OF DISTRIBUTED SILO SIGNALING

The embodiments described herein recite a telephone communication system used for handling information such as messages, typically voice mail messages, and, more particularly, is directed to a system that provides distributed session initiation protocol (SIP) silos. Distributed SIP silos (DSS) is a Communications Application Platform (CAP) feature that maintains the site's call capacity even when a signaling server fails. DSS uses multiple non-redundant signaling servers to provide SIP signaling for the same set of media ports. Because there are multiple signaling servers providing signaling for the same set of ports, the failure of one signaling server only terminates the calls it was actively processing and once those calls have been cleaned up, all the available (non-suspended) ports in the configuration are available to the remaining signaling servers. 1. A computer-implemented method comprising:establishing, by a signaling server, a real-time transport protocol (RTP) session stream for a call between a media server and a remote endpoint using a port;receiving, by the signaling server from a host server, a command to end a call;transmitting, by the signaling server to the media server, a command to release the port of the call;receiving, by the signaling server from the media server, an acknowledgement of the command to release the port of the call;transmitting, by the signaling server to the remote endpoint, a command indicating the call should end; andreceiving, by the signaling server from the remote endpoint, an ok message indicating the command indicating the call should end was received.2. The method of claim 1 , wherein the command to end the call contains the network interface unit identification (NIU id) claim 1 , primitive identification (Prim id) claim 1 , channel claim 1 , call tag claim 1 , and the port to terminate.3. The method of claim 1 , wherein the command to release the port of the call claim 1 , releases the port and returns the port to the ...

Telephony terminal

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format.

Systems and methods of distributed silo signaling

The embodiments described herein recite a telephone communication system used for handling information such as messages, typically voice mail messages, and, more particularly, is directed to a system that provides distributed session initiation protocol (SIP) silos. Distributed SIP silos (DSS) is a Communications Application Platform (CAP) feature that maintains the site's call capacity even when a signaling server fails. DSS uses multiple non-redundant signaling servers to provide SIP signaling for the same set of media ports. Because there are multiple signaling servers providing signaling for the same set of ports, the failure of one signaling server only terminates the calls it was actively processing and once those calls have been cleaned up, all the available (non-suspended) ports in the configuration are available to the remaining signaling servers.

CONTROLLER FOR THE INTELLIGENT INTERCONNECTION OF TWO COMMUNICATION NETWORKS, AND METHOD OF USE FOR SAME

A caller ID based call routing feature. A processing system in the public switched telephone network (PSTN) receives first identifying information for identifying the source of a telephone call and associates additional information stored in a memory with the first identifying information. The additional information may then be transmitted to the subscriber via the Internet for display. Another feature is a branch calling feature where the subscriber may program a processing system within the PSTN to forward an incoming call to two or more end units (e.g., telephones) simultaneously. If the call at an end unit is answered, answer supervision signaling is transmitted back to the processing system which then terminates all other calls. The processing system then connects the calling party to the subscriber. The branch calling may be made for any combination of local, long distance, and cellular telephone numbers. 1. A method of routing calls from a first network to a second network , comprising:receiving at a controlling device connected to a tandem switch, a first information relating to a call from a first user connected to the first network, the call being directed to a second user connected to the second network, wherein said receiving takes place prior to the second user answering the call;generating by the controlling device a second information, wherein the second information is a function of at least one control criteria available to the controlling device; andusing the second information to process the call from the controlling device through the second network.2. A method according to claim 1 , wherein the controlling device enables a communication path between the first user on the first network and the second user on the second network.3. A method according to claim 2 , wherein the first network and the second network operate on disparate communications protocols.4. A method according to claim 2 , wherein the at least one control criteria has been ...

METHODS FOR HANDING OVER A CIRCUIT SWITCHED CALL TO AN INTERNET PROTOCOL CALL AND RELATED ELECTRONIC DEVICES AND COMPUTER PROGRAM PRODUCTS

A method includes establishing a circuit switched call with an automated answering system at a destination, the destination having a communication server associated therewith, transmitting a request to switch to Internet Protocol (IP) communication to the automated answering system, receiving an IP request acknowledgement message from the communication server responsive to transmitting the request to switch to IP communication, and establishing an IP communication session with the communication server responsive to receiving the IP request acknowledgement. 1. A method , comprising:establishing a circuit switched call with an automated answering system at a destination, the destination having a communication server associated therewith;transmitting a request to switch to Internet Protocol (IP) communication to the automated answering system;receiving an IP request acknowledgement message from the communication server responsive to transmitting the request to switch to IP communication; andestablishing an IP communication session with the communication server responsive to receiving the IP request acknowledgement.2. The method of claim 1 , further comprising:terminating the circuit switched call with the automated answering system responsive to establishing the IP communication session.3. The method of claim 2 , wherein terminating the circuit switched call comprises:receiving a notification from the communication server that the circuit switched call is associated with the IP communication session; andterminating the circuit switched call responsive to receiving the notification.4. The method of claim 1 , wherein transmitting the request to switch to IP communication comprises:transmitting the request to switch to IP communication to the automated answering system responsive to receiving a communication from the automated answering system presenting an option to switch to IP communication.5. The method of claim 1 , further comprising:downloading an IP communication ...

Call Intent Notification for Establishing a Call

Techniques for call intent notification for establishing a call are described. According to one or more implementations, a user of a first user terminal attempts to establish a call with a second user terminal. Based on determining that the second user terminal and/or a user of the second user terminal is not available to answer the call, a call intent notification is sent to the second user terminal. According to one or more implementations, the call intent notification includes a number that enables the second user terminal to establish a call with the first user terminal. 1. A system comprising:a processing apparatus; and receiving an indication that a first user performs an action on a first user terminal to initiate a call with a second user of a second user terminal and via a communication service;', 'determining one or more of that the second user terminal is currently unavailable to answer a call through said communication service, or that the second user is currently unavailable to answer a call through said communication service;', 'sending, automatically and responsive to said determining, a call intent notification to the second user terminal via an out-of-band communication channel that is separate from the communication service; and', 'including in the notification a Public Switched Telephone Network (PSTN) number enabling the second user terminal to initiate a PSTN call with the first user terminal using the PSTN number and via a connection between the second user terminal over a PSTN network., 'memory storing code that is executable by the processing apparatus to perform operations including2. A system as recited in claim 1 , wherein said sending comprises sending said notification to enable the second user terminal to provide a prompt with a control that is actuatable to establish the PSTN call to the first user terminal based on the PSTN number without requiring manual dialing of the PSTN number at the second user terminal.3. A system as recited in ...

TELEPHONY TERMINAL

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format. 1a phone port configured to connect to a phone interface that provides access to a publicly switched telephone service network;a network port configured to connect to a network interface that provides access to a data network; anda controller configured to, in response to receipt of header information from the phone port that identifies a service of a group of services comprising at least a public switched telephone network service, an electronic mail service, and a network voice service, initiate the service,wherein the network voice service facilitates conversion of information to or from a voice format of the data network and exchange of the information between the phone port and the network port.. An apparatus, comprising: This application claims the benefit of U.S. Provisional Application No. 60/414,242, filed Sep. 27, 2002, and of U.S. Provisional Application No. 60/454,464, filed Mar. 13, 2003, the disclosures of which are incorporated herein by reference.A typical analog telephone system sends and receives voice information to and from the PSTN (public switched telephone network) using a POTS (plain old telephone service) connection. An alternative approach for sending voice information has been developed using IP networks, sometimes referred to as VoIP (voice over IP). One common way to use VoIP is to provide ...

Information communication system, information communication apparatus, information communication method, and recording medium

정보 통신 시스템은, 제1 이용자가 조작하는 제1 전화와 제1 기기가 접근한 것에 의해 작동하여 양자를 접속하는 제1 근접 통신 수단과, 제2 이용자가 조작하는 제2 전화와 제2 기기가 접근한 것에 의해 작동하여 양자를 접속하는 제2 근접 통신 수단을 포함한다. 제1 전화가 제1 근접 통신 수단을 통하여 제1 기기로부터 그 IP 어드레스를 판독하고, 전화망을 통하여 제2 전화로 송신하고, 또한 제2 근접 통신 수단을 통하여 제2 기기에 통지한다. 전화기와 그 근방의 전자 기기와의 제휴 동작에 의해 전화로 통화 중인 상대와의 사이에서 정보의 공유나 정보의 교환을 행할 수 있다. The information communication system includes a first proximity communication means for operating a first telephone operated by a first user and a first device accessed to connect them, and a second telephone and a second apparatus operated by a second user. Second proximity communication means for operating by accessing and connecting both. The first telephone reads its IP address from the first apparatus via the first proximity communication means, transmits to the second telephone via the telephone network, and also notifies the second apparatus via the second proximity communication means. By the cooperative operation of the telephone and the electronic device in the vicinity thereof, it is possible to share information or exchange information between the parties who are talking on the telephone.

IP Phone having ringback tone generating apparatus and Method for transmitting ringback tone thereof

본 발명은 인터넷 폰에 관한 것으로, 인터넷 폰(IP Phone)에 링백 톤 발생 장치를 구비시켜 공중교환전화망(Public Switched Telephone Network) 폰(phone) 등으로부터의 호 착신시 자체적으로 링백 톤을 생성하여 발신측으로 전송할 수 있는 링백 톤 발생 장치를 구비한 인터넷 폰 및 그의 링백 톤 전송 방법에 관한 것이다. 이와 같은 본 발명 인터넷 폰의 링백 톤 전송 방법은, 인터넷 네트워크와 공중교환전화망을 연결하는 네트워크 포인트 그리고 브릿지 수단 및 제어 수단을 포함하는 망을 이용한 통화에 있어서, 전송된 호 설정 메시지가 공중교환전화망 게이트웨이에서 발생된 메시지이면 착신측 단말에서 링백 톤을 생성하여 이를 실시간 전송 프로토콜 패킷에 포함하여 상기 공중교환전화망 게이트웨이로 전송하는 단계, 상기 실시간 전송 프로토콜 패킷을 링백 톤으로 변환하여 상기 공중교환전화망의 발신측 단말로 전송하는 단계를 포함하여 이루어진다. BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an internet phone, and includes a ringback tone generating device in an IP phone to generate and send a ringback tone by itself when a call is received from a public switched telephone network phone. The present invention relates to an internet phone having a ringback tone generating device capable of transmitting to a side, and a ringback tone transmitting method thereof. The ring back tone transmission method of the Internet phone of the present invention includes a network point connecting an internet network and a public switched telephone network, and in a call using a network including a bridge means and a control means, the transmitted call establishment message is a public switched telephone network gateway. If the message is generated by the receiving terminal to generate a ringback tone, including the real-time transport protocol packet and transmitting to the public switched telephone network gateway, converting the real-time transport protocol packet to a ringback tone to the calling party of the public switched telephone network It comprises the step of transmitting to the terminal.

Bidirectional data communication system for communication between bidirectional data communication device such as portable telephone set or the like and computer, bidirectional communication device used therefor and its usage method

(57)【要約】 【課題】 携帯電話機、双方向ページャー等の軽量の 移動式双方向データ通信デバイスをクライアントとして 使用可能な双方向データ通信ネットワークシステムを提 供する。 【解決手段】 クライアントモジュールを備えた双方 向データ通信デバイスが、例えば、ＷＡＮ、ＬＡＮ、イ ンターネット等を含む双方向データ通信ネットワークを 介して、ネットワークに接続されたサーバコンピュータ と通信を行う。双方向データ通信デバイスは、クライア ントモジュールを用いてユーザが選択したリソース選択 情報を含むメッセージをコンピュータネットワーク上の サーバコンピュータへ伝送し、サーバはこのメッセージ を処理して応答をクライアントモジュールに返送し、ク ライアントモジュールはこの応答を翻訳し、ユーザイン タフェースを介してユーザに対して表示する。サーバが 応答を別の位置または別のユーザに伝送するリクエスト をユーザがサーバに送ることもできる。

Branch calling and caller ID based call routing telephone features

A caller ID based call routing feature is described for blocked and non-blocked caller ID's. A processing system in the public switched telephone network (PSTN) receives first identifying information for identify the source of a telephone call and associates additional information stored in a memory with the first identifying information. The additional information may be information about the calling party initially downloaded to the memory by a subscriber. Once retrieved from the memory by the processing system, the additional information may then be transmitted to the subscriber via the Internet for display on a monitor or to the subscriber's telephone for display on a telephone display. Another feature described is a branch calling feature where the subscriber may program a processing system within the PSTN to forward an incoming call to two or more end units (e.g., telephones) simultaneously. If the call at an end unit is answered, answer supervision signaling is transmitted back to the processing system which then terminates all other calls. The processing system then connects the calling party to the subscriber. The branch calling may be made for any combination of local, long distance, and cellular telephone numbers.

Service providing system and method therefor

提供一种可基于进行多个通信的观点实现一维状况管理的服务提供系统及其提供方法。通信系统10把Web浏览器发出的请求作为呼叫尝试来执行，通过HTTP协议针对该请求起动呼叫尝试管理部100，通过联合管理部96和58作成SIP协议的请求，在呼叫尝试管理部56中对该请求进行处理，经相反的通信路径把该处理结果的应答送给Web浏览器50来显示，顺序操作服务代理UA管理部98、联合管理部58乃至服务代理UA管理部54，提供服务代理UA注册，按照相反的顺序把注册结果返回到Web系统，之后，从Web系统执行事项，确立呼叫。

Interactive device control system for integrating the internet with the public switched telephone network

An Interactive Device Control System has a method and means for providing device control, call control, and communications system information for telecommunications and Internet voice and multimedia communications. A communications system, service, or device offers the user of a communications device having a display screen, such as a screen phone, the ability to control or influence the functions of the communications system. The PSTN establishes and controls Internet communications between communications systems and user devices. Messages comprising information to be presented on the user's display are sent from the communications system, service, or device, and interactive responses from the user indicate the selected control function. Call control messages from the communications system to the PSTN execute the desired control function, such as transferring the call. The Interactive Device Control System operates across the Public Switched Telephone Network (PSTN) and the Internet, further enhancing the benefits of integrating the Internet with the PSTN.

System and method for generating audio calling signal

FIELD: communication line terminals; generation and transmission of audio calling signals. SUBSTANCE: proposed method for transmitting audio calling signal to communication line terminals depending for their operation on different communication protocols includes following procedures: first terminal depending for its operation on Internet protocol where audio calling signal is generated identifies type of network to which second terminal requesting call initialization is assigned; audio calling signal data to be transferred to second terminal are generated in compliance with type of network; audio calling signal data are inserted in message responding to call initializing message; then response message incorporating mentioned inserted data is transferred to second terminal. EFFECT: provision for using different communication protocols for signal shaping and transmission. 20 cl, 4 dwg ÐÎÑÑÈÉÑÊÀß ÔÅÄÅÐÀÖÈß (19) RU (51) ÌÏÊ 7 (11) 2 252 488 (13) C1 H 04 L 12/66, H 04 M 3/42, H 04 J 3/12 ÔÅÄÅÐÀËÜÍÀß ÑËÓÆÁÀ ÏÎ ÈÍÒÅËËÅÊÒÓÀËÜÍÎÉ ÑÎÁÑÒÂÅÍÍÎÑÒÈ, ÏÀÒÅÍÒÀÌ È ÒÎÂÀÐÍÛÌ ÇÍÀÊÀÌ (12) ÎÏÈÑÀÍÈÅ ÈÇÎÁÐÅÒÅÍÈß Ê ÏÀÒÅÍÒÓ (21), (22) Çà âêà: 2003127439/09, 10.09.2003 (72) Àâòîð(û): ËÈ Äîíã Êüþ (KR) (24) Äàòà íà÷àëà äåéñòâè ïàòåíòà: 10.09.2003 (73) Ïàòåíòîîáëàäàòåëü(ëè): Ýë Äæè Ýëåêòðîíèêñ Èíê. (KR) (30) Ïðèîðèòåò: 07.10.2002 KR 2002-061051 2 2 5 2 4 8 8 R U (54) ÑÈÑÒÅÌÀ È ÑÏÎÑÎÁ ÔÎÐÌÈÐÎÂÀÍÈß ÒÎÍÀËÜÍÎÃÎ ÑÈÃÍÀËÀ ÂÛÇÎÂÀ (57) Ðåôåðàò: ðàáîòàþùåì íà îñíîâå ïðîòîêîëà Internet, Èçîáðåòåíèå îòíîñèòñ ê òåðìèíàëàì ëèíèé çàêëþ÷àåòñ â èäåíòèôèêàöèè òèïà ñåòè, ê êîòîðîé ñâ çè è, â ÷àñòíîñòè, ê ñèñòåìàì è ñïîñîáàì ïðèíàäëåæèò âòîðîé òåðìèíàë, çàïðàøèâàþùèé ôîðìèðîâàíè è ïåðåäà÷è òîíàëüíîãî ñèãíàëà èíèöèàëèçàöèþ âûçîâà; ôîðìèðîâàíèè äàííûõ âûçîâà. Òåõíè÷åñêèì ðåçóëüòàòîì âë åòñ òîíàëüíîãî ñèãíàëà âûçîâà, ïåðåäàâàåìûõ íà ñîçäàíèå ñïîñîáà è óñòðîéñòâà äë ôîðìèðîâàíè âòîðîé òåðìèíàë â ñîîòâåòñòâèè ñ òèïîì ñåòè; è ïåðåäà÷è òîíàëüíîãî ñèãíàëà âûçîâà òåðìèíàëàì âêëþ÷åíèè äàííûõ òîíàëüíîãî ñèãíàëà âûçîâà â ñâ ...

system and method for unified messaging system service using voice over Internet protocol

본 발명에 따른 VoIP(voice over Internet protocol)를 이용한 UMS 서비스 제공시스템에서는 VoIP 인터페이스를 갖는 VoIP-UMS에서 VoIP 게이트웨이로 음성 데이타를 보낼 때, 미리 녹음된 음성 데이터를 보내주는 UMS의 특성을 이용하여 음성 데이타를 버스트하게 보내고, VoIP 게이트웨이에서는 사용자에게 들려줄 음성 데이터를 수신하여 버퍼에 미리 저장해두었다가 일정한 간격으로 재생하여 들려줄 수 있게 되어 지연과 지터에 의한 잡음을 제거할 수 있다. In the UMS service providing system using the voice over Internet protocol (VoIP) according to the present invention, when the voice data is sent from the VoIP-UMS having the VoIP interface to the VoIP gateway, the voice is transmitted by using the characteristics of the UMS which transmits the pre-recorded voice data. The data can be sent in bursts, and the VoIP gateway can receive the voice data to be heard by the user, store it in the buffer in advance, and play it back at regular intervals to remove noise caused by delay and jitter. UMS, VoIP 게이트웨이, 버퍼, 버스트, 실시간, PCM, PSTN UMS, VoIP Gateway, Buffer, Burst, Real Time, PCM, PSTN

Method and architecture for an interactive two-way communication network

URL notification device for mobile phones

Bandwidth management system and method for quality of service in voice over internet protocal network

본 발명은 호를 설정하거나 해제하는 경우 사용되는 사용 대역폭 정보를 제공하여 사용 대역폭 정보가 제공될 때 대역폭을 재설정하여 변경되는 가용 대역폭 정보를 통보하는 브이오아이피 네트워크에서 큐오에스 보장을 위한 대역폭 관리 시스템 및 그 방법에 관한 것이다. The present invention provides a bandwidth management system for guaranteeing QOS in a VIP network that provides information on the available bandwidth by resetting the bandwidth when the usage bandwidth information is provided by providing usage bandwidth information used when establishing or releasing a call. And to a method thereof.

For transmitting imminent event automatically by interface to the method for the endpoint for distributing to user and the conversion equipment constructed thus

本发明涉及一种计算机实现的、用于通过接口自动传输即将发生的事件到分配给使用者的端点（60，80）的方法，所述事件存储在分配给使用者的客户端（30）的事件处理机构（20）中，所述方法包括以下步骤：‑所述客户端（30）发送关于即将发生的事件的第一附加信息到转换单元（40）上，‑所述转换单元（40）产生关于所述即将发生的事件的预约提醒并且发送所述预约提醒到电信平台（50）上，‑所述电信平台（50）处理所述预约提醒，产生信号并且发送所述信号到所述端点（60，80）上，其中，所述信号匹配于所述端点（60，80）并且优选地是声学的和/或光学的信号，以及‑所述端点（60，80）处理所述信号和所述第一附加信息，以便作为结果接受或拒绝所述第一预约提醒，并且发送所述结果到所述客户端（30）上。所述方法此外涉及一种为此构造的转换装置以及具有在其上存储的计算机程序产品的机器可读的数据载体（90）。

Dynamically providing terminal connected to public communication network with services offered by private telecommunication network, is capable of simultaneously exchanging signalling data on first channel and voice data on second

The server (2) comprises a control unit (6) for transmitting to a terminal (4) connected to the first network (PLMN) over the first channel and in accordance with the selected criteria, configuration data designed to enable a connection with the server to be set up on the first channel by the terminal, during a voice link on the second channel, so as to provide the terminal during the voice link, with services offered by the second network.

System and method for providing unified messaging system service using voice over internet protocol

一种使用VoIP(基于互联网协议的语音)来提供UMS服务的系统和方法，通过发送语音数据到VoIP网关，带有VoIP接口的VoIP－UMS利用传输已记录的语音数据的UMS的特点，以突发模式发送所述语音数据，VoIP网关接收用户要接听的该语音数据，在缓冲器中提前存储该语音数据，然后以恒定时间间隔再现该语音数据以便用户接听到该语音，由此消除了由于延迟和跳动产生的噪音。

Communication system

A method of transmitting a communication event from a calling entity to a called entity located in a communication network comprising a first network and a second network, said method comprising the steps of: determining a first identity associated with the called entity, wherein the first identity is recognisable in the first network; allocating a calling identity to the called entity, wherein the allocated identity is recognisable in the second network; associating at a first node located in the first network the first identity with the -calling identity; providing the calling identity to the calling entity; establishing a connection with a second network node located in the second network; wherein the connection with the second network node is established using the calling identity, querying the first node with the calling identity associated with the called entity to determine the first identity associated with the called entity; transmitting the communication event to the called entity using the first identity associated with the called entity; and disassociating the calling identity from the called entity after step of querying the first node to determine the first identity associated with the called entity.

Service providing system cooperative with VoIP and web environments and a method therefor

In a communication system, a request is generated from a Web browser as a call attempt. A call attempt manager is booted in response to the request under the HTTP protocol. The request is turned by cooperative managers into a request under the SIP protocol, which will be processed by another call attempt manager. A response to the result of processing by the other manager is sent to a Web browser in a reverse direction on the transmission path of the request for display on the Web browser. Service proxy UA managers and the cooperative managers on both systems are brought into play in order so that the response is registered in a service proxy user agent. The result of the registration is returned to the Web system in the direction reverse to the processing result. A task is subsequently executed from the Web system to establish a call.

System and method for access to multimedia structures

The present invention relates to a system (10) in which a telephone set (16) is capable of using access devices or apparatuses (12) and a network such as the Internet (80) to access multimedia structures consisting for example of automatic answering messages or of messages that can be interpreted by a speech synthesis device, said structures being stored in a storage device (15). The invention also relates to the method for implementing the system (10) and to the generation equipment (14) capable of producing the multimedia structures to be instantiated by means of the telephone set (16) and the access device (12). In particular, the system (10) and method make it possible to implement universal automatic answering machines, i.e., answering machines that are capable of managing a plurality of answer message structures stored in a single device (15) and accessible by means of access devices (12) distributed on the network (80) or in the telephone sets (16).

A kind of converged communication system based on micro-base station

本发明公开了一种基于微基站的融合通信系统，该系统包括：微基站子系统、固话网子系统、办公网子系统以及通信交换子系统，所述通信交换子系统接收所述微基站子系统、所述固话网子系统发送的通话请求，并在通话请求中的被呼叫用户是移动通信内网用户时，将通话请求向所述微基站子系统发送，在通话请求中的被呼叫用户是固话网用户时，将通话请求向所述固话网子系统发送，接收所述微基站子系统发送的业务请求，并将所述业务请求发送给所述办公网子系统。本发明实施例可以实现企业内的移动通信网、固话网、办公网的三网融合通信，提供企业的办公效率，并解决了现有的通信系统具有很高的复杂性，部署和使用成本高等问题。

Voice call transfer method and electronic equipment

本申请涉及一种语音通话转移方法及电子设备。该方法包括：在目标设备作为被叫方收到来电语音呼叫时，在通话应用界面上显示能够进行语音通话转移的第一设备的设备标识、接通来电语音呼叫的第一按钮和拒绝来电语音呼叫的第二按钮；接收对设备标识的选择操作，触发目标设备进入语音通话转移模式；在确定目标设备与主叫方设备进行语音通话且处于语音通话转移模式时，将来自主叫方设备的第一语音数据发送至第一设备，以使第一设备播放第一语音数据。本申请实施例所提供的方法及电子设备，将目标设备与主叫方设备的语音通话转移至第一设备，用户所需执行的操作少、便捷性强，且语音通话过程中所传输的语音数据的品质高、传输速度快，适用范围广。

IP telephone exchange equipment

Communication system

A method of transmitting messages from a network node in a communication network to a first user device can be provided. The network node transmits to a second user device a message-waiting notification. The first user device is identified, and a node holding the message is identified. In response to that notification, the second user device transmits a connection request to the first user device. The connection request is acted on by the first user device to establish a connection. With the identified node, the message can be optionally transmitted to the first user device.

Call centre

Communicaton system

A data store associates a first identity recognizable in a first network with a second identity recognizable in a second network for both a user and a calling entity. Both identities for the calling entity are provided to the receiving device via the second network. A node receives the communication event with the second identity for the calling entity, the communication event being addressed with the second identity of the user. The data store is queried with the second identities of the calling entity and user to determine the first identities of the calling entity and user. The receiving device receives the communication event with the first identity for the calling entity via the first network, the communication event being addressed with the first identity of the user. The receiving device determines the second identity of the calling entity using the first identity of the calling entity.

Communications switching architecture

Presently described is a system (200) and method for switching multimedia data communications, including but not limited to Voice over IP (VoIP) telephony, cable TV, digital audio and video. The system utilizes a single, integrated device (210) to provide all Packet Cable-compliant functionality, including enhanced user privacy, compliance with CALEA, E911 and other mandated services not available in conventional distributed Packet Cable systems. High speed and efficient, low cost operation are provided by means of an optimized data unit encapsulation scheme for internal switching and routing. A proprietary fiber optic backplane and removable optical connectors are used to enable lightspeed internal communications hot-swapping of components. Furthermore, the present system (200) is extensible to all forms of digital data switching and is secure, resistant to Denial of Service attacks, and fault-resilient.

Url notification device for portable telephone

System and method for switching telecommunications service in a multiple dwelling unit/multiple tenant unit environment

A system and method for managing, switching, and reconfiguring physical connections of telecommunications services that provides improved cost and timeliness. The method comprises the steps of: providing telecommunications service for telecommunications equipment of a first subscriber via a cross-connect switch connected to a telecommunications network, the telecommunications equipment of the first subscriber connected to a first telecommunications service, the cross-connect switch implemented between a central office and a subscriber location; receiving, at a network operations center connected to the cross connect switch, an indication that the first subscriber has terminated or changed service; in response to receiving, at the network operations center, an indication that the first subscriber has changed service, transmitting a command to the cross connect switch to switch the connection of the telecommunications equipment of first subscriber from the first service to a second service; and in response to receiving the command at the cross-connect switch, switching the connection of the telecommunications equipment of first subscriber from the first telecommunications service to a second telecommunications service.

System and method for switching telecommunications service in a multiple dwelling unit/multiple tenant unit environment

A system and method for managing, switching, and reconfiguring physical connections of telecommunications services that provides improved cost and timeliness. The method comprises the steps of: providing telecommunications service for telecommunications equipment of a first subscriber via a cross-connect switch connected to a telecommunications network, the telecommunications equipment of the first subscriber connected to a first telecommunications service; receiving, at a network operations center connected to the cross connect switch, an indication that the first subscriber has terminated or changed service; in response to receiving, at the network operations center, an indication that the first subscriber has changed service, transmitting a command to the cross connect switch to switch the connection of the telecommunications equipment of first subscriber from the first service to a second service; and in response to receiving the command at the cross-connect switch, switching the connection of the telecommunications equipment of first subscriber from the first telecommunications service to a second telecommunications service.

Unified access platform distributing atm cells and telephony signals to subscriber premises over twisted wire pairs

A Unified Access Platform capable of providing telephone and high speed data services in a number of different local loop configurations. In a first, embodiment, a broadband digital terminal (BDT) receives high speed data and telephony signals, and combines them into a cell based signal which is transported to an access multiplexor. At the access multiplexor, a first linecard generates an analog telephone signal, and a second linecard generate; a high-speed data signal. Analog telephone service is provided over a first twisted wire pair drop cable, while the high speed data service is provided over a second twisted wire pair drop cable. In an alternate embodiment, the BDT's cell based signal is transported to two separate terminals. Analog telephone service is provided to a subscriber location from the first terminal, while high speed data service is provided to a second subscriber location from the second terminal. Yet another embodiment provides analog telephone and high speed data services from a single linecard located in an access multiplexor. The analog telephone and high speed data signals are generated on the linecard and combined using a diplexor. At the residence, a receiving diplexor is used to separate the combine signal. Still another embodiment generates a high speed data signal at an access multiplexor which contains a digital representation of the analog telephones signal. The high speed data signal is sent to the residence, where a receiving device generates an analog telephony signal and transmits the high speed data signal to appropriate terminal equipment.

Method and device for displaying supplemental information while rendering a playlist

A media player device and method for receiving and playing a playlist is provided. The device has a transceiver to wirelessly send and receive communication, a memory, a control system connected to the memory and the transceiver, and at least one speaker to play the plurality of songs. The control system and method receive a command to log into an account before receiving a playlist, receive and display the playlist identifying a plurality of songs in a menu with context sensitive icons representing different functions, receive and display information associated with an artist from a currently playing song of the plurality of songs in the playlist, receive and display information on a related song that is from a genre similar to that of one of the currently playing song, and receive and display advertising. The method provides for the speaker to play the plurality of songs.

Telephone outlet for implementing a local area network over telephone lines and a local area network using such outlets

A network for transporting power and multiplexed data and digital telephone signals. The network includes at least three nodes and first and second wiring segments in a building for carrying the multiplexed data and digital telephone signals, and at least one of the segments is configured to additionally carry a power signal. A power consuming component is connected to the at least one wiring segment and is powered by the power signal carried by that segment. Each wiring segment connects a different pair of the nodes together to form, with nodes nodes, a packet based bi-directional communication link. One of the nodes contains communication link composed of a repeater, a bridge, or a router connectable to a data unit. At least one of the nodes is connected to a remote data unit external to the building for coupling the remote data unit to at least one of said communication links.

Multimedia network transposition

Methods and systems for routing multimedia traffic are described. A method may include receiving a communication originating from a user, transposing an alias transport network over one or more existing networks to route the communication, and routing the communication based on mapping rules.

Method for collect call service based on voip technology and system thereof

The present invention relates to a collect call method and system thereof, more particularly, in order to charge the called party with an uniform toll for collect call, which is determined by only the type and location of called party terminal, the collect call method, system and a counsel service providing method use a free VoIP network as a part of voice call link and a charge PSTN network as the rest of voice call link. According to the preferred embodiment of the present invention, if the first link corresponding to the collect call request is established, the collect call switch calls the called party terminal to establish the second link, and billing on the second link is initiated.

Telephone outlet and system for a local area network over telephone lines

A local area network (70, 80, 90, 110, 112) using the telephone wiring within a residence or other building simultaneously with telephony signals, with network outlets (31a, 31b, 31c, 31d) having telephone connectors (14a, 14b, 14c) and data connectors (113a, 113d) allowing connection of Data Terminal Equipment to the network. The network outlets use high pass filters (22a, 22b, 22c) to access the high-frequency band across the media, whereas the standard telephone service uses low pass filters to access the low-frequency voice / analog telephony band across the same media. The high pass filters of the network outlets at each segment end are connected to modems (23a, 23b, 103a, 103b) or other Data Communication Equipment, thus supporting data communication networks of various topologies, including point-to-point topologies. The invention also contemplates a network outlet (31a, 31b, 31c, 31d) having a data interface connector (101) for allowing connection of Data Terminal Equipment thereto and having a telephone connector for connecting a standard telephone thereto. A splitter (50a, 50b, 50c, 50d) is adapted to be connected to a telephone line for separating data and telephone signals transported on the telephone line, whilst a coupler (50a, 50b) is adapted to receive telephone and data signals and feeds a composite signal at an output thereof to the telephone line.

Telecommunications network

Information communication system, information communication device, information communication method, and computer program

An information communication system includes a first telephone operated by a first user, first proximity communication means operated when a first device comes into its vicinity so as to connect the both, a second telephone operated by a second user, and second proximity communication means operated when a second device comes into its vicinity so as to connect the both. The first telephone reads an IP address of the first device via the first proximity communication means, transmits it to the second telephone via a telephone network, and notifies it to the second device via the second proximity communication means. By link operation between a telephone and an electronic device in its vicinity, it is possible to share information and exchange information with a conversation partner via the telephone.

Network-enabled audio device

A network-enabled audio device that provides a display device that allows the user to select playlists of music much like a jukebox is disclosed. The user can compose playlists from disk files, CD's, Internet streaming audio broadcasts, online music sites, and other audio sources. The user can also select a desired Web broadcast from a list of available Web broadcasts. In addition, the user can play standard audio CD's and MP3 encoded CD's and have access to local AM/FM stations. Further, the software, the user controls, and the display in the network-enabled audio device are operably configured and connected such that the user can listen to playlists that include CD's and other audio sources just as the user would choose a playlist in a jukebox. The user accesses a server site via a PC and the Internet. From the server site, the user obtains a list of the devices in his or her Internet Personal Audio Network (IPAN) and what songs are on those devices. The IPAN includes an IPAN server, an IPAN client, and IPAN software stored on the network-enabled audio device. Thus, the network-enabled audio device provides people who are or are not comfortable with computers a way of taking music from various sources and putting it into one place for listening pleasure. In one embodiment, the Personal Computer (PC) is used to compose the playlists, but the user is able to listen to playlists and other audio sources without using the PC.

Communication system

A method of transmitting a communication event from a calling entity to a called entity located in a communication system comprising a first network and a second network, said method comprising determining a first identity associated with the calling entity and a first and second identity associated with the called entity, wherein the first identities are recognizable in the first network and the second identity is recognizable in the second network; associating at a first node located in the second network the first identities with the second identity; establishing a connection between the calling entity and a second node; wherein the connection with second node is established using the first identity associated with the called entity; querying the first node with the first identity associated with the called entity and the first identity associated with the calling entity to determine the second identity associated with the called entity; and transmitting the communication event to the called entity using the second identity associated with the called entity.

An internet time multiplexed circuit connection

Here is presented a method whereby an Internet Time multiplexed Circuit Connection is established enabling data communications at both ends in exactly the same way as in the case where the Internet Connection so established is a PSTN switched circuit connection, at wire speed with same transmissions quality. At predetermined periods (the single individual periods of which may be different in time durations at each nodes) all selected nodes between source and destination are arranged to automatically switch incoming signals to next node at wire speed without buffering delay/route computation delay, thus establishes a Time Multiplexed Circuit Connection for the durations of the predetermined periods, as in the case where a simplex PSTN dedicated circuit connection has been established.

Call completion via instant communications client

Determining presence for interaction routing

An Internet Protocol Network Telephony call center having a plurality of agents for serving clients also processes e-mails addresses to the call center, but not to specific agents. An e-mail server receives and routes the e-mail, and includes a router and a database storing skill set information regarding agents. The router extracts information from the e-mails, matches key words in the extracted information with key words from the skill sets, and routes the e-mails to appropriate agents having the necessary skills to respond to the service requirements in the e-mails.

System and method for generating ringbace tone

公开一种产生和发送回铃音的方法及其系统。在基于因特网协议的第一终端产生回铃音的方法包括：识别请求呼叫建立的第二终端所属网络的类型，根据网络类型产生被发送到第二终端的回铃音数据，将回铃音数据插入到呼叫建立的响应消息，以及将插入的响应消息发送到第二终端。

Method of operating a contact center

Some known Contact Centers rely on a Network Level Router to determine which of a number of available Contact Centers should deal with a given contact. This decision is made by the NLR based on real time information provided by the Contact Centers. However, the format of such information is proprietary which means that the NLR must translate and collate this information. This invention seeks to use session initiation protocol (SIP) Presence for such information transfer which enables improved decision making and further enables additional functionality to be added to the Contact Center system. The invention provides a SIP enabled Contact Center (180, 190, 200) comprising a Contact Center server arranged to send SIP messages to one or more other SIP enabled nodes in a SIP communications network, said SIP messages comprising SIP presence information about the current state of the Contact Center.

Method and apparatus for automatically building conversational systems

A system and method provides a natural language interface to world-wide web content. Either in advance or dynamically, webpage content is parsed using a parsing algorithm. A person using a telephone interface can provide speech information, which is converted to text and used to automatically fill in input fields on a webpage form. The form is then submitted to a database search and a response is generated. Information contained on the responsive webpage is extracted and converted to speech via a text-to-speech engine and communicated to the person.

Apparatus and method for providing enhanced telecommunications services

An apparatus and method is disclosed for providing transparent enhanced telecommunications services to subscribers by implementing an enhanced services platform on a local network exchange within the public telephone network. More specifically, the apparatus and method of the present invention is capable of providing a plurality of enhanced telecommunications services to a subscriber by using a voice-activated interface to enable the subscriber to access at least one of the plurality of enhanced telecommunications services. In addition, the apparatus and method of the present invention may use a web-activated interface to enable the subscriber to access at least one of the plurality of enhanced telecommunications services.

Url notification device for portable telephone

A URL notification device for a portable telephone comprises telephone network control means for responding to a received call directed to a telephone number to be contacted for service, electronic mail network connection means for exporting an optional electronic mail, a URL storage unit, telephone number/electronic mail address conversion means for acquiring the telephone number B of a caller when receiving the call and converting the number B to the mail address D allocated to the telephone of the number B, and electronic mail sending means for reading at least one URL from the URL storage unit in response to the received call from the caller telephone number B, creating an electronic mail message in which the URL is described in a text of the message in hypertext format and sending the mail to the address D by the network connection means.

Internet protocol telephony for a mobile station on a packet data channel

Method and device for obtaining playlist content over a network

A network-enabled audio device that provides a display device that allows the user to select playlists of music much like a jukebox is disclosed. The user can compose playlists from disk files, CD's, Internet streaming audio broadcasts, online music sites, and other audio sources. The user can also select a desired Web broadcast from a list of available Web broadcasts. In addition, the user can play standard audio CD's and MP3 encoded CD's and have access to local AM/FM stations. Further, the software, the user controls, and the display in the network-enabled audio device are operably configured and connected such that the user can listen to playlists that include CD's and other audio sources just as the user would choose a playlist in a jukebox. The user accesses a server site via a PC and the Internet. From the server site, the user obtains a list of the devices in his or her Internet Personal Audio Network (IPAN) and what songs are on those devices. The IPAN includes an IPAN server, an IPAN client, and IPAN software stored on the network-enabled audio device. Thus, the network-enabled audio device provides people who are or are not comfortable with computers a way of taking music from various sources and putting it into one place for listening pleasure. In one embodiment, the Personal Computer (PC) is used to compose the playlists, but the user is able to listen to playlists and other audio sources without using the PC.

Voice instant messaging

Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender (602a) an at least one recipient (602b) through an instant messaging host (604). In addition, voice communication is enabled between the sender (602a) and the recipient (602b) through the instant messaging host (604).

Communication system

一种将通信事件从位于包括第一网络和第二网络的通信网络中的呼叫实体传输到被叫实体的方法，所述方法包括以下步骤：确定与被叫实体相关的第一标识，其中，第一标识在第一网络中是可识别的；将呼叫标识分配给被叫实体，其中，所分配的标识在第二网络中是可识别的；使得所述第一标识在位于第一网络中的第一节点处与呼叫实体相关；将呼叫标识提供给呼叫实体；与位于第二网络中的第二网络节点建立连接；其中，与第二网络节点的连接利用呼叫标识来建立，用与被叫实体相关的呼叫标识来询问第一节点，以确定与被叫实体相关的第一标识；利用与被叫实体相关的第一标识将通信事件传输到被叫实体；以及在询问第一节点以确定与被叫实体相关的第一标识的步骤之后使呼叫标识与被叫实体分离。

Telephone outlets and systems for local area networks on the telephone line

System and method for initiating a conference call

The present invention is a system and method for initiating conference calls via an instant messaging system to reduce the effort required to initiate and manage the call. The system uses an IM connection between a requesting party and a conference call server to inform the conference call server of the desire to initiate the conference call. The conference call server may initiate the conference call by having involved parties called by a conference bridge, thus reducing the effort required by the parties to join the call.

Apparatus and methods in improving and enhancing telephony telecommunications

A plurality of aspects (113, 114, 120, 124) and embodiment of unique telephony apparatus and methods are disclosed, including but limited to video routing and conferencing, coordinating telephone calls with data pertaining to the calls, methods for implementing and operating call centers (110), routing calls by statistical modeling, using multiple object states in telephony software systems, and routing of electronic documents.

Home agent call center system and method

A call center system supports remote agent stations (126) through a network (125) by establishing a data link (119, 129) between a computer platform (131) at the remote agent station (126) and a computer-telephony integration (CTI) processor (111) connected to a telephony switch (109) at a call center (101) and switching incoming calls from the call center (101) to a telephone (133) at the remote agent station (126) while sending data pertaining to the calls via the data link (119, 129) to the computer platform (131) at the remote agent station (126) to be displayed.

Home Agent Call Center System and Method

Communication system

一种控制在通信网络的用户和通信系统之间的通信会话的方法，其中通信网络包括第一网络和第二网络，并且其中通信系统在第一网络上操作所述方法，该方法包括：将会话请求包从与用户相关的设备传输到位于第一网络内的第一节点；响应于接收到所述会话请求而在第一节点处为用户启动与通信系统的会话，其中在会话期间该方法包括以下步骤：在用户和实体之间经由通信系统的通信事件期间，从第一节点向设备提供设备使用的呼叫标识，其中所述呼叫标识用于建立设备和位于第二网络中的第二节点之间的连接；判定所述实体的地址；以及使呼叫标识与实体的地址相关联；其中当终止会话时使呼叫标识与实体的地址解除关联。

Computer Telephony for Cellular Phones

The present invention relates to a computer telephony terminal provided in a personal computer ( 10 ) and includes a computer telephony handling unit, a first cellular phone ( 12 ), and a control unit. The control unit is arranged to order the setting up of a voice-over-IP connection over a computer network ( 18 ) to a second terminal ( 14 ) based on phone call information received by the first cellular phone from a second known cellular phone ( 16 ) via a cellular network ( 20 ) or to order the setting up of a cellular voice connection from the first cellular phone to the second known cellular phone based on a received voice-over-IP call from the second terminal received by the computer telephony handling unit via a voice-over-IP connection.

Method and apparatus for accessing communication data relevant to a target entity identified by a number string

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 ).

System and method for initiating a conference call

The present invention is a system and method for initiating conference calls via an instant messaging system to reduce the effort required to initiate and manage the call. The system uses an IM connection between a requesting party and a conference call server to inform the conference call server of the desire to initiate the conference call. The conference call server may initiate the conference call by having involved parties called by a conference bridge, thus reducing the effort required by the parties to join the call.

Ringback tone transmission method, terminal, ringback tone generation method, and system for generating ringback tone

Computer, internet and telecommunications based network

A method and apparatus for a computer and telecommunication network which can receive, send and manage information from or to a subscriber of the network, based on the subscriber's configuration. The network is made up of at least one cluster containing voice servers which allow for telephony, speech recognition, text-to-speech and conferencing functions, and is accessible by the subscriber through standard telephone connections or through internet connections. The network also utilizes a database and file server allowing the subscriber to maintain and manage certain contact lists and administrative information. A web server is also connected to the cluster thereby allowing access to all functions through internet connections.

Method and apparatus for call distribution and override with priority

A communication router receives incoming communications and assigns a priority to each, which are then placed in descending order of priority from the head of the queue, to be routed from the head of the queue to selected destinations (31, 33). The invention is particularly applicable to routing telephone calls in a call center (21). Destinations (31, 33), in the call center agent stations, are selected on a basis of agent availability, and an agent status may be available if the agent is involved in a call of lower priority than a call to be routed. In some embodiments displaced calls may be placed back in the routing queue in order of priority. The invention is applicable to routing a wide variety of communications, such as telephone calls (13), e-mails, video calls, and Internet Protocol (11) calls.

Apparatus and method for coordinating telephone and data communications

(57)【要約】 第１のサイトと第２のサイトとの間の電話通信およびデータ通信を調整するシステムが開示されている。第１のサイトのコンピュータ（１１４）がサーバ（１３２）から生成されたデジタル情報を表示する。第２のサイト（１０２）は複数の代行者を有している。第２のサイトはまた入ってくる電話通話をこれらの代行者に向ける電話スイッチ（２０２）を備えている。第１のサイトのコンピュータは、デジタル通信ネットワーク（１０６）を通して第２のサイトを呼び出すリクエストを第２のサイトに送る。第２のサイトは代行者を選択して、この電話通話用の電話スイッチのルーティングポイントを確保する。この電話番号を持つ入ってくる通話をスイッチが受信すると、この入ってきた通話は選択された代行者にルーティングされる。サーバから生成されたデジタル情報も選択された代行者に関係するコンピュータ（１４６）に送られる。結果として、代行者は同じデジタル情報を即座に表示し、第１のサイトとの電話通信を維持することができる。