Abstract: Described are computer-based methods and apparatuses, including computer program products, for sharing control of a voice over internet protocol (VoIP) call. In some examples, the method includes establishing a direct connection between a VoIP endpoint device and a computing device associated with the VoIP endpoint device. The method further includes communicating, via the direct connection, VoIP control signaling between the VoIP endpoint device and the computing device. The method further includes modifying a VoIP call based on the VoIP control signaling.

Abstract: A switching system that receives communications over a trunk incoming from a communications network and routes them to its subtending communications terminals responds to a stimulus, such as actuation of an actuator, by either (a) returning all of the communications that it receives on the incoming trunk at least subsequently to the stimulus to the network over a trunk outgoing to the network with instructions directing the network to redirect the returned communications to a second switching system for routing to communications terminals served by the second system, or (b) routing all of the communications that it receives on the incoming trunk at least subsequently to the stimulus to the second switching system over a trunk outgoing to the second switching system for routing to the communications terminals served by the second switching system.

Abstract: A technique is disclosed that enables a contact center to provide enhanced services to its agents. An agent server is interconnected with the automatic call distributor (ACD) and agent telephones of the contact center. The server intercepts messages sent by the ACD, which contain caller-provided data obtained by an interactive voice response system. When it is determined that the particular agent has access to both a telephone and a personal computer, the agent server draws a distinction in the information content of the received message. In doing so, the agent server provides the call initialization information in the message to the agent's telephone, and appears as if the information were coming from the ACD; additionally, the agent server provides the caller-provided data to the agent's personal computer. The personal computer is then able to execute a predetermined application, such as a display-oriented presentation application, based on the caller-provided data.

Abstract: A system for and method of providing a gateway transfer mechanism is presented. In one exemplary embodiment, the system and method of providing a gateway transfer mechanism may be realized as a method, comprising identifying, at a gateway, a call to be routed to a first location, determining whether to transfer the call to a second location, in the event the call is to be transferred to the second location, transferring the call, wherein transferring the call comprises ending a process to route the call to the first location, generating transfer information for the second location, and transferring the call to the second location using the generated transfer information.

Abstract: A temporary ENUM architecture assigns a temporary ENUM at call time. For an originated communication requiring routing to a party which needs a particular callback format not supported, the TempENUM provides consistency with the specific format required by the receiving party. The TempENUM is assigned for the duration of the call or communication, or for a given amount of time thereafter. The TempENUM service gateway includes a TempENUM database, a module that assigns a suitable TempENUM at the time of call routing, and a module that handles a lookup in the TempENUM database, and maps the assigned TempENUM to a calling party identifier (CgPID). The TempENUM database stores a TempENUM from a coordinator system and relates the actual calling party identifier (CgPID) for use in callback attempts. Either during the initial call or communication, or shortly thereafter, the receiving party uses the TempENUM number to return communication the initial caller.

Abstract: The present invention provides for routing calls between disparate domains, such as a circuit-switched subsystem and a multimedia subsystem. When a user element is homed in a first domain and roaming in a second domain, an incoming call will arrive at a gateway node in the first domain. As a result, a message identifying the user element and indicating that an incoming call has been received at first gateway node for the first domain is sent to a continuity control function (CCF) residing in the multimedia subsystem. The CCF will create and effect delivery of an inter-domain routing number to the first gateway node. The inter-domain routing number is used by the first gateway node to route the call to the second domain. The inter-domain routing number may be associated with a second gateway node of the second domain.

Abstract: A telephone system comprising switching circuitry configured for coupling a call to a telephone extension coupled to the system, voice processing circuitry configured for automatically interacting with the call, a microprocessor, a first data bus connected between the microprocessor and the switching circuitry, and a second data bus connected between the microprocessor and the voice processing circuitry.

Abstract: Systems and methods for customizing messages are described. The apparatus may include an interaction engine to enable interactions between a customer and an agent, a user interface module to display a listing of life events and a life event engine to retrieve the listing from a data store. The method may include receiving a life event indication and playing one of a plurality of life event messages using at least the life event indication.

Abstract: A busy lamp field (BLF) system for remote telephones and a method for the same are disclosed. The busy lamp field system for remote telephones mainly has a plurality of switchboard systems and a XML server interconnected via the internet. Each switchboard system has a PBX and at least a telephone. Each telephone is connected to the XML server via internet and periodically uploads local phone status. The BLF key of each telephone is set to associate with a corresponding specific remote telephone by a subscriber of each telephone. Each telephone periodically retrieves the phone status of the corresponding remote telephone from the XML server displayed on the BLF key of the local telephone. Thus, a subscriber monitors status of a remote telephone and dials out via each PBX in addition to extension telephone of the local switchboard system.

Abstract: A system includes a network entry point configured to receive and process internet protocol (IP) based connection requests originating from outside the communications network, and a telephone number mapping (ENUM) database configured to store a uniform resource identifier (URI) of the network entry point and to respond to a request for routing information for a subscriber terminal within the communications network received from a remote entity outside the communications network by providing a naming authority pointer resource (NAPTR) record including the URI of the network entry point to the remote entity in response to the request for routing information.

Abstract: A telephone system comprising switching circuitry configured for coupling a call to a telephone extension coupled to the system, voice processing circuitry configured for automatically interacting with the call, a microprocessor, a first data bus connected between the microprocessor and the switching circuitry, and a second data bus connected between the microprocessor and the voice processing circuitry.

Abstract: A call services manager and methods for improving the quality and efficiency of voice of Internet Protocol (“VOIP”) calls are provided. In an embodiment a method to improve communication link utilization on a call requiring the use of a central service platform is provided. An additional method is provided to validate caller identification information exchanged between originating and terminating communication devices in an IP voice network. In a further aspect of the invention, a method using a call services manager is provided to dynamically switch from one call route to another based on quality of service considerations. A call services manager is also provided. The call services manager is a software proxy that can reside at any type of network element. The call services manager includes software modules to implement the above methods for improving VOIP quality, and can include other call service and quality improvement modules.

Abstract: Embodiments of the invention include a method for handling communication sessions by a central gateway, such as session border controller. The method may include receiving an indication that the interaction between a customer and an agent has ended and automatically routing by the session border controller the customer to an automated feedback application.

Abstract: Methods and systems are disclosed for enabling a dynamic computer telephony integration campus call center that leverages the assets of a school communications system including internal telecommunications networks, information systems, data networks, and applications, of public telecommunications networks, of public data networks, and/or of various communications devices to facilitate improved access, sharing, notification, and/or management of communications (e.g., external and internal communications) and associated data to enhance school safety services.

Abstract: A communication network and method are disclosed that block calls from a calling party based on the network access type of a called party. The communication network of the invention includes a permission system, an access system, and a call control system. Based on a call from a calling party to a called party, the permission system identifies permissions associated with the calling party. The access system identifies a network access type for the called party, such as wireless, circuit-based, etc. The call control element determines if the network access type for the called party is permitted by the calling party based on the permissions associated with the calling party. If the network access type for the called party is not permitted by the calling party, then the call control system releases the call.

Abstract: A method for setting up a communication connection between a calling communication terminal and another communication terminal belonging to a group in a communication network selected by the calling party dialing an allocated group telephone number, the other communication terminal being selected by an exchange unit of the selected communication network by means of availability information, and the connection initiated through the exchange of a start message and a number of connection confirmation messages between the calling communication terminal and the selected other communication terminal. A private branch exchange with a server which comprises an exchange unit and to which a plurality of communication terminals combined in groups is connected, carries out the method.

Abstract: Provided is an apparatus and method for providing a voice service without discontinuance in a mobile communication system. The apparatus includes a controller for receiving a system information block of a first network system or a system information block of a second network system from an interface module, and transmitting a network selection change request to the interface module to change a network system mode; and a network selector for, when the system information block of the first network system does not comprise voice switched domain information for location area update, driving a timer according to a selection of a user, changing the network system mode into the second network system, and executing location area update to thereby provide the voice service.

Abstract: A method, call setup device (10) and computer program product for controlling telephone call setup for an associated user and having access to different communication networks are provided. The call setup device (10) comprising a data receiving unit (52) receiving a first data message (DM1) from the user comprising callee identifying information and user identifying information and a control unit (48) having access to call placing units (12, 14, 58), each connected to corresponding network (24,32) and providing telephone call set up via these networks, wherein the control unit analyses the callee and user identifying information, orders a call placing unit to set up a first telephone call to a user terminal, orders a call placing unit to set up a second telephone call to a callee terminal and orders an interconnection of the first and the second call.

Abstract: When an outgoing call is to be requested on a terminal, the terminal transmits a first outgoing call request containing the priority of outgoing call operation. When the terminal transmits the first outgoing call request, and then receives a second outgoing call request addressed to the terminal before it receives an incoming call response with respect to the first outgoing call request from a destination terminal, the terminal compares the priority contained in the second outgoing call request with the priority contained in the first outgoing call request. The terminal executes an incoming call response procedure corresponding to the second outgoing call request when the priority in the second outgoing call request is higher as the result of comparison between the priorities of outgoing call operation, and executes a outgoing call control following the first outgoing call request when the priority in the first outgoing call request is higher.

Abstract: The invention provides a method and apparatus for enabling DTMF signal processing in the core VoIP network. More specifically, the present invention enables a VoIP network to recognize and respond to special DTMF signals entered by a user and initiate the appropriate service logic response to satisfy the user's service request.

Abstract: A temporary ENUM architecture assigns a temporary ENUM at call time. For an originated communication requiring routing to a party which needs a particular callback format not supported, the TempENUM provides consistency with the specific format required by the receiving party. The TempENUM is assigned for the duration of the call or communication, or for a given amount of time thereafter. The TempENUM service gateway includes a TempENUM database, a module that assigns a suitable TempENUM at the time of call routing, and a module that handles a lookup in the TempENUM database, and maps the assigned TempENUM to a calling party identifier (CgPID). The TempENUM database stores a TempENUM from a coordinator system and relates the actual calling party identifier (CgPID) for use in callback attempts. Either during the initial call or communication, or shortly thereafter, the receiving party uses the TempENUM number to return communication the initial caller.

Abstract: A telecommunication call center handles a call from a telecommunication network. The call center comprises a call center processor and a call center destination system. The call center processor receives and processes a message from the telecommunication network for the call to select a route code and to select a key, wherein the route code and the key are separate from one another. The call center processor transfers the route code and the key to the telecommunication network. The call center processor transfers the key and call-handling information for the call to the destination system. The destination system receives the key and the call from the telecommunication network, and associates the call with the call-handling information based on the key.

Abstract: A system includes a network entry point configured to receive and process internet protocol (IP) based connection requests originating from outside the communications network, and a telephone number mapping (ENUM) database configured to store a uniform resource identifier (URI) of the network entry point and to respond to a request for routing information for a subscriber terminal within the communications network received from a remote entity outside the communications network by providing a naming authority pointer resource (NAPTR) record including the URI of the network entry point to the remote entity in response to the request for routing information.

Abstract: A method includes receiving a media signal at an interface device in communication with and disposed between an Internet Protocol private branch exchange (IP-PBX) and a first endpoint. The media signal is associated with the first endpoint and is associated with a session established between the first endpoint and a second endpoint within a session over internet protocol network. The session is established at least in part by the IP-PBX based on a session control signal received at the interface device from the first endpoint and sent from the interface device to the IP-PBX. The method also includes modifying the media signal at the interface device based on a per-device service associated with at least one of the first endpoint or the second endpoint.

Abstract: Multiple access internet portal billing systems are provided. A representative system, among others, includes a communication facility, a wireless internet server, and a transaction billing system. The communication facility includes a billing platform, and is operable to connect to a plurality of wireless device platforms through a mobile network, and to connect to the wireless internet server. The wireless internet server provides internet access to the wireless devices and communicate at least one billing information record including a usage time to a transaction billing system. The transaction billing system receives the billing information record from the wireless internet server, formats the billing information record, and communicates the formatted record to the communication facility billing platform. Methods and other systems for multiple access internet portals are also provided.

Abstract: A method for setting up a communication connection between a calling communication terminal and another communication terminal belonging to a group in a communication network selected by the calling party dialing an allocated group telephone number, the other communication terminal being selected by an exchange unit of the selected communication network by means of availability information, and the connection initiated through the exchange of a start message and a number of connection confirmation messages between the calling communication terminal and the selected other communication terminal. A private branch exchange with a server which comprises an exchange unit and to which a plurality of communication terminals combined in groups is connected, carries out the method.

Abstract: Methods, systems, and apparatus, including computer program products, in which a first telephone number is associated with a second telephone number, and a web-based advertisement is served for an advertiser that includes the first telephone number. A call placed to the first telephone number is received, and the call is routed to the second telephone number based on the association of the second telephone number with the first telephone number. A determination as to whether a call-through event in response to receiving the call placed to the first telephone number has occurred, and an advertiser associated with the second telephone number is charged a cost-per-call amount if the call-through event is determined to have occurred.

Abstract: According to one aspect of the present invention, there is provided a system for establishing a communication between a first party and a second party, comprising: a data receiving subsystem for receiving from a third party details of the first and second party; and control logic for establishing a communication between the first and second party using the details supplied by the third party.

Abstract: A user may utilize a communication device, e.g., a wireless phone, mobile device or personal digital assistant (PDA), to contact an information/call center where an operator provides personalized information and communications services to the user in accordance with the invention. Such services include, e.g., providing listing information, contacts information, appointments information, etc. To provide effective connectivity, for example, when a user locates a phone number in his/her private directory using a web browser, the user may utilize a direct connect feature of the invention to initiate a phone call to the desired number over the Internet. In addition, the invention allows for delivery of the appointments information to the user not only via voice connection, but also SMS, email, facsimile, PDA, etc. It also allows for rendering a calendaring service in combination with such services as a TeleConcierge® service which helps the user to plan a future event, e.g., making a restaurant reservation.

Abstract: The invention relates to a method, call setup device (10) and computer program product for controlling telephone call setup for an associated user and having access to different communication networks. The call setup device (10) comprises a data receiving unit (52) receiving a first data message (DM1) from the user comprising callee identifying information and user identifying information, and a control unit (48) having access to call placing units (12, 14, 58), each connected to corresponding networks (24/32) and providing telephone call setup via these networks. The control unit analyses the callee and user identifying information, determines a preferred call setup scheme based on said analysis, orders a call placing unit to set up a first telephone call to a user terminal, orders a call placing unit to set up a second telephone call to a callee terminal, and orders an interconnection of the first and the second call.

Abstract: An short message service switching private branch exchange system and a method can reduce costs. The method can include certifying whether a usable DSP exists or not when an office line and a speech path are connected to each other, and transmitting an short message service signal transmitted from the office line to the DSP through a PCM channel if the usable DSP exists. A main processor of the private branch exchange can certify an extension line terminal that will receive short message service, generate an short message service signal having a format corresponding to a kind of the extension line terminal by using the DSP, and transmit the formatted SMS signal to an extension line interface unit of a corresponding extension line terminal. The system switches an SMS signal between an office line and an extension line by sharing the DSP, for example, by time division.

Abstract: A method of negotiating a call capability between signalling points in a telecommunications system. The method comprises sending a capability preference or prioritised list of preferences from an originating signalling point to a terminating signalling point or signalling transfer point, at the Call Control level. A capability acceptance is returned from the terminating signalling point or signalling transfer point to the originating signalling point at the Call Control level, if the terminating signalling point or signalling transfer point accepts a preference sent by the originating signalling point.

Abstract: A method and system are provided for processing an incoming call and providing an interactive voice response unit with application flow control and call arrival and departure control. The system includes a call router including rules for handling call arrival and call departure and an interactive voice response unit. The interactive voice response unit includes a state and event table for operation as a state machine for intermediate processing between call arrival and call departure. An additional external resource is provided for updating the state and event table in order to control interactive voice response application flow. The method includes processing call arrival at the call router, performing intermediate processing with the state machine at the interactive voice response unit, and handling call departure procedures at the call router. The method additionally includes updating the state and event tables and providing application flow control using an external resource.

Abstract: An apparatus and corresponding method for managing busy lamp field (hereinafter, referred to as BLF) information in an inter-networking system are provided which are capable of collecting and managing BLF information of every PBX in a BLF information repeater added to the inter-networking system. The BLF information repeater reduces the load of the inter-networking system by transmitting the BLF information to a PBX that requests the BLF information.

Abstract: An IPNT call center is provided wherein agent's computers may be locally-connected to a managing computer on a local area network, or remote agents may act over the Internet with the managing computer. The managing computer establishes an IPNT call with each remote agent on duty, and then routes incoming IPNT calls to the remote agents by substituting an incoming call for an existing call without closing the existing call, thereby avoiding the necessity of establishing a new call with the remote computer for each call routed to the remote agent. In some embodiments the IPNT system is a part of a multimedia call center, wherein incoming calls may be either IPNT or plain old telephony service (POTS), and calls are routed to agents according to a set of business rules without regard to type.

Abstract: A personal directory system 10 has a translation server 40 for verifying the identity of a subscriber and recognizing the name of one or more subscribers that are stored on a personal directory of another subscriber. A caller is connected by a public telephone switch to the server 40 that accesses a personal directory computer 60 located on the World Wide Web 50. The requested number and/or speed dial number are converted from text to speech and the public telephone switch automatically dials the requested number.

Abstract: A personal directory system 10 has a translation server 40 for verifying the identity of a subscriber and recognizing the name of one or more subscribers that are stored on a personal directory of another subscriber. A caller is connected by a public telephone switch to the server 40 that accesses a personal directory computer 60 located on the World Wide Web 50. The requested number and/or speed dial number are converted from text to speech and the public telephone switch automatically dials the requested number.

Abstract: A system and method to connect a caller to a callee through the services of an operator, the method comprising: connecting the caller to the operator through an origination switch; connecting the operator to the callee through a destination switch; sending a release message to the destination switch with a request for a callee address; receiving the address at the operator; sending a release message to the origination switch with the callee address; and connecting the callee with the caller by utilizing the callee address. The method can also include putting the callee on hold at the destination switch and sending the callee address from the destination switch. The method can also include putting the caller on hold at the origination switch while receiving the callee address from the operator. The operator can be an automated operator, a manual operator, or a private branch exchange, for example. The network can be an asynchronous transfer mode network and, or an Internet Protocol network, for example.

Abstract: An operator call-fielding scenario system which displays a response scenario to an operator who is responding to an incoming call includes a memory unit which stores a plurality of scenario parts making up a plurality of response scenarios; a scenario setting unit which selects predetermined scenario parts from the plurality of scenarios parts in the memory unit, in accordance with one of a plurality of conditions, and sets a response scenario with the predetermined scenario parts according to the one condition; and a display unit which displays the set response scenario. Using this configuration, response scenarios matched in the optimum way with various circumstances can be set.

Abstract: An apparatus and corresponding method for managing busy lamp field (hereinafter, referred to as BLF) information in an inter-networking system are provided which are capable of collecting and managing BLF information of every PBX in a BLF information repeater added to the inter-networking system. The BLF information repeater reduces the load of the inter-networking system by transmitting the BLF information to a PBX that requests the BLF information.

Abstract: The present invention is an apparatus, system, and method for converting point codes in a signal transfer point in a telecommunications signaling system. The STP converts point codes which designate the origination and destination signaling points for the message. The conversion is based on information defined by the messages, such as origination or destination information. The present invention creates a virtual signaling system which can be reconfigured at the STP by converting point codes, and thus, altering the identities of the signaling points. The present invention is also operable to convert circuit identification codes and transfer integrated services user part messages to a user part.

Abstract: An ISDN primary rate interface circuit having a plurality of communication channels, includes a channel division controller. According to predetermined conditions, the channel division controller selects a communication channel out of the plurality of communication channels when generating or receiving a call. The channel division controller provides private branch exchange by enabling allocation of the selected communication channel to an extension group according to a predetermined condition when generating or receiving the call.

Abstract: A system and method for directing an incoming telephone call. The system comprises a control unit that receives images associated with two or more regions of a local environment. The two or more regions are each serviced by a respective telephone extension. The control unit processes the images to identify, from a group of known persons associated with the local environment, any one or more known persons located in the respective regions. For each known person so identified, an indicium is generated that associates the known person with the respective region in which the known person is located.

Abstract: A call transfer may be prompted by a first endpoint, from the active status—open bearer channels between the first endpoint and a second endpoint—in a communications network based on ITU-T Standard H.323. The call transfer is effected by a series of steps, namely a facility message for call transfer, including an address for the third endpoint, is transmitted to a proxy function; a connection setup message is transmitted from the proxy function to the third endpoint and the third endpoint responds by transmitting a call connect message to the proxy function; the connection between the first endpoint and the second endpoint is released; and the proxy function initiates a connection to be set up between the second endpoint and the third endpoint.

Abstract: A system for service control and operations for a telecommunications network. In particular, an architecture and method for a service control and operations element system. The system communicates with a plurality of interconnected telecommunications network elements via a switching and signaling subsystem. The system provides and controls the various functions of the telecommunications network, such as call processing and routing, automatic fault detection and correction, providing services to customers in an interactive manner, fraud detection and control, identification of patterns of abuse of the network, collecting data regarding call activity at each network element, and producing a record of each call placed within the network.

Abstract: A method and server for providing intermediate telephone call notification services while increasing the reliability of other call handling services is disclosed. Calls including called party identifier are received at a switch (16,28) forming part of a network (26). The calls are forwarded to a first available notification server (12, 18, 34) in communication with the switch (12, 18, 34), preferably using pre-configured route lists. Each server (12, 18, 34) is capable of providing call notification service for at least some subscribers (10, 22, 30). A server (12, 18, 34) receives a call, including a called subscriber identifier and determine whether the call should be processed as a call for a primary or a secondary subscriber. For calls for a primary subscriber, the server (12, 18, 34) dispatches an indicator of the call. For a call to a secondary subscriber the server forwards the call to a call handling system, such as a voice mail server (36, 38, 40), interconnected with the network.

Abstract: A portable keyboard enables touch typing with both hands without external support. The keyboard includes a first array of keys and a second array of keys on opposite sides of the keyboard. The keyboard is held between the hands and is supported against the palm of one hand for touch typing with both hands.

Abstract: In an environment of competitive local and interexchange carriers, offering number portability between local carriers serving a common region and between switches serving that region, each local carrier accesses a regional database to determine the identity of the carrier and switch serving a local customer. In addition, interexchange carriers access a national database to determine the identity of the carrier and switch serving the customer specified by the number dialed by an originating customer. For customers requiring high reliability service, alternate carriers can be used to serve such customers in case the primary carrier is unavailable; the databases identify these alternate carriers. Advantageously, this arrangement allows a high degree of freedom of movement of customers between carriers and geographic relocation without requiring a number change.

Abstract: A service node is used in an intelligent communications network for providing services for customers.

Abstract: A method and system for enabling dynamic location of a voice and video mixing site in a telephony-over-LAN (TOL) automatic call distributor (ACD) system includes establishing a first call between an ACD agent terminal and a customer terminal, wherein the first call includes first voice data transmitted from the agent terminal and second voice data transmitted from the customer terminal. Upon receiving a monitoring session request from an ACD supervisor terminal, agent processor monitoring circuitry determines whether sufficient agent processor resources are available to mix voice, and if applicable, video data samples. If sufficient agent processor resources are available, an agent terminal call control cooperates with a supervisor terminal call control to establish a first monitoring call. The agent call control monitors the first call for simultaneously transmitted agent and customer voice and video data.