359results about "Intelligent networks" patented technology

A system and method include implementing and accessing a subscriber's telecommunications services, using a graphical user interface (GUI) via the Internet, and an interactive voice response (IVR) system via the public switched telecommunications network (PSTN). The system and method enable a subscriber to review, schedule and modify call forwarding information stored and implemented by a service control-point (SCP) through a common server from any dual tone multi-frequency (DTMF) telephone and IVR system via the PSTN or any GUI and Web client via the Internet. The subscriber is able to build and edit call forwarding data, such as a scheduler and priority and rejection screening lists, by the GUI via the Internet for subsequent implementation.

AIN based methods and apparatus for transitioning telephone numbers and customers from the PSTN to a VOIP network are described. AIN line number portability features are used to allow a few gateway switches that interconnect the PSTN and VOIP networks to service customers whose telephone numbers were originally serviced by several remote PSTN switches. AIN LNP triggers are used to forward PSTN calls, directed to the PSTN switch previously used to service a telephone number, to the gateway switch assigned to route such calls to the IP network. AIN triggers set at the gateway switch insure that the subscriber with the ported telephone number continues to receive AIN services provided before the telephone number was ported to the IP network. Calls from ported telephone numbers to telephones in the PSTN are billed from PSTN switches through the use of AIN functionality and triggers set at the gateway switch.

System and method for deriving call routing information utilizing a network control data base system and voice recognition for matching spoken word sound patterns to routing numbers needed to set up calls. Public access is provided to a common database via a common access number or code. Terminating parties sponsor the call and originating parties need not be pre-subscribed to use the service. The common access number is used to initiate or trigger the service. The system advantageously operates under the direction of a service control point, which combines technologies for switching, interactive voice response, and voice recognition with the data base to automate the processes of assisting callers in making calls for which they do not know the phone number. Usage information is gathered on completed calls to each terminating party for billing. Three alternative deployments in the U.S. telephone network are described, and vary based on the location of the service control points or intelligent processors and the degree of intelligence within the network.

A system for granting access to agents at a communication center in response to requests for connection from network-level entities starts a fairness timer having a fairness time period when a first request is received for and agent, monitors any other requests for the same agent during the fairness time, and executes an algorithm at the end of the fairness time to select the network-level entity to which the request should be granted. In a preferred embodiment the fairness time is set to be equal to or greater than twice the difference between network round-trip latency for the longest latency and shortest latency routers requesting service from the communication center. In some embodiments an agent reservation timer is set at the same point as the fairness timer to prevent calls to the same agent, and has a period longer than the fairness timer by a time sufficient for a connection to be made to the agent station once access is granted, and for notification of the connection to be made to network-level entities.

A system and method for providing Value-Added Services (VAS) in an integrated telecommunications network having a packet-switched network portion (PSN) operable with Session Initiation Protocol (SIP). The integrated telecommunications network includes a SIPext SSP server, a trigger server, and a service node having a Service Logic Program (SLP) that is operable with Intelligent Network Application Protocol (INAP). The SIPext SSP and service nodes are provided with the capability to communicate using SIP-compliant messaging. New header fields are provided that specify operations to be performed by the service node with respect to a service. INAP service parametric data is also provided in the header fields in a sequential form. When a call is received in the SIPext SSP server for a user having a subscription for a VAS, it queries the user profile stored in the trigger server. If the user is subscribed for a service, a SIP request message is formulated based on the user profile, wherein appropriate headers are populated with relevant parametric information and call context data. The service node launches the SLP based on the information provided in the request message and sends a SIP response message to the SIPext SSP server with an instruction concerning the provisioning of the VAS. The SIPext SSP server, thereafter, takes an appropriate action based on the response message and any parametric information contained therein.

A telecommunications network that handles call center calls by providing advanced interactive voice response (IVR) services using a next generation service node (NGSN). The NGSN nodes combine with several other interacting network components to provide a scaleable architecture which allows several different, customized and flexible call center services to be performed for telecommunications service provider customers. In addition the network includes virtual call routers, parking managers, host connect gateways, report servers, advanced network databases, data access points, and data gateways to deliver calls to customer call centers with various features enabled.

An enhanced caller-id service providing calling name and calling number information to subscribers via a server system. The server system can combine the caller-id information with other data to provide more detailed reports for the subscriber. The server can also be used to maintain long-term calling records for subscribers. The caller-id information can be accessed remotely by the subscriber and can be downloaded from the server to another computer for use with other applications.

A telecommunications network is enhanced with a presence component. A session initiator requests a session with a session terminator by contacting a presence server. The presence server receives the request for presence information and processes the request by comparing the session initiator's identity to preferences of the session terminator to identify a preferred treatment. The presence server returns the preferred treatment to the session initiator. The session is then initiated or not initiated based upon the preferred treatment.

A system and method of for exchanging information between a wireless network and an external system is disclosed. The wireless network detects trigger events related to communications with wireless subscribers. Certain event triggers cause the wireless system to create a message to send to the external system. In response the external system may send information to the wireless system. The wireless system may create messages upon detection of handoff, origination, status, registration or other activity by the wireless subscriber. The invention may also detect trigger events related one mass media broadcasts. An interface translates between the different protocols of the wireless and the external or wireline networks to allow for automatic call redirection. The application can be utilized between many networks and facilitates the utilization of the architecture of a wirelined network and the home location register of the wireless network for various applications on the wireless network.

A system and method of sending and receiving voice calls over the Internet is described. The system includes a PSTN-derived intelligent call routing engine that receives call control messages from the Voice over IP (VoIP) network and routes calls via previously stored, customized routing procedures. Special dialing plan, 800 numbers, minimized cost routing, time and day-based least cost routing, transferring, conferencing, and additional messaging capabilities are provided. The system operates on standard server computers with interface cards connected to the packetized networks via TCP / IP and, indirectly, to VoIP gateways to accessing PSTN networks and to virtual and / or private networks (PBXs).

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

An interactive assistant for managing telephone communications and services is disclosed. In one of many possible method embodiments, a chat interface is provided to a device associated with an intended recipient of an incoming voice call. A chat message from the intended recipient is received through the chat interface. The chat message is presented to a calling party associated with the incoming call prior to a disposition of the call being determined. In other embodiments, an interface provides controls for individually managing calls on a conference call, without affecting other calls on the conference call.

An enhanced caller-id service providing multi-media based caller information to subscribers via a server system. The server system can combine user-provided profile information with other data to provide more detailed reports for the subscriber. The server can also be used to maintain long-term calling records for subscribers. The caller information can be accessed remotely by the subscriber and can be downloaded from the server to another computer for use with other applications.

A telecommunications service controls and monitors telephone calls by storing at a service control point (SCP) restricted telephone numbers and priority telephone numbers. Upon receiving call data relating to a call between a subscriber telephone number and an outside telephone number, the SCP determines whether the outside telephone number matches the restricted or priority telephone numbers. When the outside telephone number matches a restricted telephone number, the call is terminated and the SCP stores the call data. When the outside telephone number does not match a restricted telephone number, or matches a priority telephone number, the call is connected. The SCP transmits stored call data to a service management system, which is accessible by the subscriber via the Internet or an interactive voice response, for generating reports. The service further includes time schedules for implementing call restrictions and personal identification number for bypassing call restrictions.

A service includes verbally editing at least one parameter of a subscriber's advanced intelligent network (AIN) service using speech recognition functionality of an intelligent peripheral. A call is received from the subscriber at a switch, including a dialed number for editing the AIN service. A connection is established with the intelligent peripheral in response to the dialed number. Call service data, including the parameter, is retrieved from a database. The intelligent peripheral plays a voice announcement identifying the parameter to the subscriber, receives a voice instruction from the subscriber in response to the voice announcement and translates the voice instruction into digital command data relating to a change to the parameter. The digital command data and the change to the parameter are stored in the database for near real-time implementation. The intelligent peripheral and the database are likewise accessible by the subscriber through a packet switched data network.

A system and method of service provisioning in an integrated telecommunications network having a packet-switched network portion (PSN) and a circuit-switched network (CSN) portion with a gateway disposed therebetween. A service node comprising a Service Control Point (SCP), a Service Data Point (SDP), or both, is provided with an interface operable with the PSN portion. A first entity (e.g., served entity) is capable of accessing the service node for the purpose of invoking a service. A return result from the service node is provided, based on the decision of what action is to be taken with respect to the service request, to the first entity which then determines how the action is to be effectuated. An indication is thereafter provided by the first entity to a second entity (e.g., re-routing entity) preferably using H.450.X messages. In response, the second entity (i.e., re-routing or switching entity) takes an appropriate service action based on the return result obtained from the service node. In a presently preferred exemplary embodiment, legacy service nodes built on the basis of WIN / IN service architecture are utilized for the provisioning of Value-Added Services (VAS) or Supplementary Services (SS) in VoIP networks operable with the H.450.X Recommendations.

A subscriber associates a ring-tone media selection based on a media genre, author, or title to a calling number. When an incoming call is detected, a particular ring-tone media selection, based on the subscriber's association, may be streamed from a central server to the subscriber's customer premises equipment and then played. Alternatively, when an incoming call is detected, a particular ring-tone selection indicator, based on the subscriber's association, may be sent from a central server to the subscriber's customer premises equipment which then plays a pre-stored ring-tone selection media associated with the indicator.

A resource management system for an intelligent communications network having one or more distributed service nodes, each service node for providing services relating to an event received at a network resource associated with a service node. The system comprising a first processing tier comprising one or more local execution environments located at a node, each execution environment including a mechanism for instantiating one or more service objects capable of performing event services at a first local execution environment, and, for generating status information relating to executing service objects; and, a second processing tier associated with a service node and including a system processor for tracking status and availability of service objects and local execution environments. Upon receipt of service requests, the system processor communicates with the first processing tier for receiving the status information and initiating service object instantiation in the one or more local execution environments in the first processing tier at the node based upon the status and availability information of the requested service object.

A method for verifying access authorization for voice telephony in a fixed network line or mobile telephone line, as well as a communications network having such access authorization verification are described. The access authorization is verified by analysis of a voice signal which was entered by the subscriber placing the call, before or during a call in progress. In one variant, the voice signal is entered as a password before the connection is established; in another variant, voice signals are analyzed for voice recognition and subscriber identification, the same voice signals also being transmitted to the person being called, making concealed access verification possible which does not hamper the normal flow of conversation.

AIN based methods and apparatus for transitioning telephone numbers and customers from the PSTN to a VOIP network are described. AIN line number portability features are used to allow a few gateway switches that interconnect the PSTN and VOIP networks to service customers whose telephone numbers were originally serviced by several remote PSTN switches. AIN LNP triggers are used to forward PSTN calls, directed to the PSTN switch previously used to service a telephone number, to the gateway switch assigned to route such calls to the IP network. AIN triggers set at the gateway switch insure that the subscriber with the ported telephone number continues to receive AIN services provided before the telephone number was ported to the IP network. Calls from ported telephone numbers to telephones in the PSTN are billed from PSTN switches through the use of AIN functionality and triggers set at the gateway switch.

Embodiments of the invention provide novel solutions, including systems, methods and / or software, for providing handoffs between cellular providers and VoIP providers. In some cases, for example, upon the initiation of a call from a dual-mode cellular phone, the cellular network (and / or a component thereof) may be configured to store the dialed number and / or substitute a predetermined number for the dialed number. The predetermined number may be associated with a VoIP provider's system. Hence, in a particular embodiment, the call may be routed (e.g., via the PSTN) to the VoIP provider's system, which may be configured to obtain (perhaps from an application server) the original dialed number and / or to route the call (e.g., via the PSTN) to the original dialed number. If the dual-mode phone subsequently obtains IP access, a VoIP connection may be established between the VoIP system and the phone and / or a handoff may be performed (e.g., in the VoIP system) to transfer the call from the cellular connection to the VoIP connection. Optionally, the cellular connection may be terminated.

Methods and systems are disclosed for audibly announcing information associated with a calling party in an advanced intelligent network comprising a service switching point, a service control point and a services node. The service switching point is adapted to suspend calls directed to an audio caller identification service and to forward information concerning the calling and called parties to the service control point. The service control point is adapted to direct the call to a selected services node. The services node sends a query to retrieve information concerning the calling party from the service control point. If necessary, other service control points may be queried for the information. While the aforementioned processing is occurring, the calling party continues to hear a ringing signal. When the aforementioned information is retrieved, the service directs a second call to the called party. When an operator from the called party station answers the phone, the service audibly announces information concerning the calling party and the called party is provided an opportunity to accept or reject the call.

A system for estimating call waiting time for a call in a queue takes into account multiple queues wherein agents are shared between queues, abandoned call history, and virtual and priority queues. The system in a preferred embodiment is a computer-telephony integration (CTI) software application adapted to execute on a CTI processor, which may be coupled to switching equipment at network level in a connection-oriented, switched telephony (COST) network or to a switch at call-center level, or both.

A system and a method store, display and report outgoing call log data associated with telephone calls from a subscriber's telephone. The system includes a switch in one network that receives a telephone call from the subscriber's telephone and launches a query in response to an AIN trigger to a service control point. In response, the service control point forwards associated call data to an interface server in another network via a generic data interface. The interface server obtains additional information from a directory database and sends the call data and the additional information to an outgoing call log database, which stores the call data and the additional information as the outgoing call log data. A web server retrieves the outgoing call log data and displays it to the subscriber at a web client.

A method and system are disclosed for facilitating location-based services. In the arrangement described, when a mobile station is engaged in a data session with a server, the mobile station may be made to initiate a voice call to a designated number. As a result, a switch or other such entity may send a signaling message (such as an Origination Request message) to a central controller, thereby providing the central controller with an indication of the mobile station's current location. The central controller may then make that location information available to the server. In turn, the server may use the location to perform a location based service, such as providing location-based information to the mobile station.

Methods and systems for conference call buffering are disclosed. Methods and systems consistent with the present invention record a conference call between a plurality of users. A service center in conjunction with a conference bridge and conference server establishes a conference call between a plurality of users. The service center may also ascertain the identities of a plurality of destination devices for an audio stream corresponding to the conference call, where the destination devices corresponding to the plurality of users. An audio streaming server provides the audio stream to the plurality of destination devices. Memory located centrally and / or local to the destination devices stores the audio stream. A user of one of the destination devices may choose to replay a selected portion of the audio stream.

AIN based call routing, transfer and conferencing methods and apparatus are disclosed. In various embodiments initial call routing is based on the availability of a party to service a call as determined from a computer system associated with the party selected to service the call. The party's computer system supports a telephone programming application interface (TAPI) which allows a telephone network server to determine the availability of the party selected to service the call from, in part, TAPI obtained telephone line status information. The network server supplies call related data to the computer system of the party assigned to service the call. Call transfer and conferencing operations along with the transfer of call related data are also supported. AIN mid-call triggers are used in some embodiments to support call transfer and conferencing operations.

A switched telephone network is arranged in a manner to enable packet voice communication between telephone terminals via multiple redundant packet switched networks. The packet switched networks may utilize different protocols, be operated by different entities, and have primary functions other than voice communication. One example of such a network may be internetworked networks, such as the Internet. One example of an alternate packet switched network may be a network whose primary function is control of a circuit switched telephone network. The common channel interoffice switching system (CCIS) of a public switched telephone network (PSTN) is one such example.