Abstract: The present invention discloses a Softswitch and system for implementing the intelligent network in the next generation networks and a method for interconnecting the intelligent networks to the PSTN. The Softswitch comprises: a network adaptive device for implementing the communication between the Softswitch and other devices in said network, as well as receiving the call request; a call server for determining whether the call received by said network adaptive device is a common call or the call of the intelligent network and processing the common call; and an INAP adapter for responding the call of the intelligent network and encoding/decoding the INAP message.

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 and apparatus for enabling multiple services to be invoked in a specific precedence per call are disclosed. For example, the present method allows a subscriber to interact with a plurality of services provided by a service provider on a single call.

Abstract: A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service.

Abstract: A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service.

Abstract: A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service.

Abstract: A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service.

Abstract: A network management system communicates with a plurality of intelligent distributed network nodes. The network management system dynamically distributes managed objects across the plurality of intelligent distributed network nodes. Examples of managed objects distributed by the network management system include a service control managed object, a call control managed object, a bearer control managed object, and a resource proxy managed object.

Abstract: A method and apparatus for enabling a packet network provider, e.g., a VoIP network provider, to terminate long distance calls using both CLEC PRI access links and regulated LEC access links depending on whether or not the wholesale customer wants to treat outgoing calls in a regulated manner similar to traditional Public Switched Access Network (PSTN) calls or apply FCC information service exemptions to outgoing calls are disclosed. Based on the calling party number of a call, the network can determine if the call should be terminated as an information service call or a regulated PSTN call. Once the call termination method is determined, the appropriate access link can be chosen for call completion.

Abstract: Example methods, apparatus and articles of manufacture to determine authorization levels in a virtual private network are disclosed. A disclosed example method includes determining whether a call originated from an originating station associated with a virtual private network, the call originating via a private branch exchange associated with the virtual private network by a first user of a plurality of users associated with the virtual private network, querying a private-number table based on at least one of a switch identifier representing a service switching point, a trunk group identifier representing the private branch exchange, or an originating station identifier representing the originating station to obtain a private number assigned to the first user, and providing an authorization level associated with the private number to the service switching point, the service switching point to process the call according to the authorization level.

Abstract: Aspects of the disclosure relate to routing of an emergency communication under fault conditions. Routing can be implemented in a packet-switching (PS) network that provides voice service. For a network node of the PS network, availability to route the emergency communication through a specific emergency service routing number (ESRN) can be determined and, in response to the network node being unavailable, administrative data associated with the ESRN can be accessed. In addition, an identifier associated with the administrative data can be updated (e.g., created or modified) with data indicative of the device that originated the emergency communication. The administrative data and the resulting identifier can be delivered to a second network node that can route the emergency communication to an emergency service network.

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: Method and communication system for providing a value added service, such as an intelligent network (IN) service, which is available in a first network (5), to a subscriber (3) in a second network (7). The first network (5) comprises a first network node (11) for executing the value added service. The method comprises detecting in a terminating call to the subscriber (3) that the subscriber (3) desires to use the value added service, forwarding control of the call towards the first network node (11) associated with a forwarding number in the first network (5), executing the value added service by the first network node (11), and, when necessary, further directing the call towards the subscriber (3) associated with the terminating call. The present method can also be applied to originating calls from a subscriber. In this case, the call is further directed towards a destination associated with the call.

Abstract: Methods and devices for controlling a set-up of a call from a calling entity (CE201) to a called entity in a telecommunications network are described. The telecommunications network comprises a service control entity (SCE205) and a service switching entity (SSE204). The service control entity (SCE205) is adapted to perform the steps of receiving from the service switching entity (SSE204) an indication that the first call attempt failed, receiving from the service switching entity (SSE204) an indication of the further call attempt, obtaining call information related to the first call attempt, determining a call establishment procedure for the further call attempt based on the call information related to the first call attempt, sending an instruction to the service switching entity (SSE204) to continue the set-up of the further call attempt according to the call establishment procedure.

Abstract: The present invention discloses method, system and network server for recording use of network service capability by applications (APPs), and the key points of the methods comprise: a network server acquires an identity of the APP and type information of the network service capability providing services for the APP, conducts service logic processing, and writes a message of use of network service capability, which comprises at least the identity of the APP and the type information of the network service capability providing services for the APP. With the methods of this invention, the purpose of making effective records of use of network service capability by APPs provided by service providers is implemented. Furthermore, management of APPs, such as charging, may be conducted by utilizing the records.

Abstract: A communications manager in a telecommunications transaction server communicates service request messages and service response messages between one or more remotely executing service request processes and one or more service applications executing on the transaction server. A main thread in the communications manager monitors connections requests and initializes other threads. A unique logical communications connection is established between one of the service request processes and the communications manager. The communications manager creates both a read thread and a write thread for each logical communication connection that is validated and active. The read thread communicates service request messages from a service request process to an appropriate service application. The write thread communications service response messages from the service application to the service request process that issues the service request message.

Abstract: A method and apparatus for enabling a network provider to instantly assign a virtual number to a subscriber to use until their primary number is successfully ported to the VoIP network is disclosed. For example, a virtual number can be a phone number for callers outside a subscriber's local calling area to call the subscriber for the price of a local call if the subscriber chooses a virtual phone number in their local calling area. After the porting of the primary number is successful, the user can be reached by either having callers directly dial the primary number or the previously assigned virtual number. The subscriber can continue to use both phone numbers or choose to disconnect the virtual number and only use the successfully ported primary number.

Abstract: The invention provides location information indicating the geographical location of a mobile station via a core net-work node serving the mobile station The method includes the steps of initiating a communication between the mobile station and the network by calculating the geographical location, supplying corresponding location information indicating the geographical location to the core network node, and subsequently routing the communication between the mobile station and the network.

Abstract: A system and method for processing calls in a call center are described. A call session from a caller via a session manager and including incoming text messages of a verbal speech stream is assigned. The incoming text messages are progressively visually presented throughout the call session to a live agent on an agent console operatively coupled to the session manager. The incoming text messages are progressively processed through a customer support scenario interactively monitored and controlled by the live agent via the agent console. The incoming text messages are processed through automated script execution in concert with the live agent. Outgoing text messages are converted into a synthesized speech stream. The synthesized speech stream is sent via the agent console to the caller.

Abstract: A user equipment is provided. The user equipment comprises a processor that, when the UE goes out of service area of a first cell, is configured to search for a suitable cell before the UE loses a connection to a network by checking for a suitable cell among cells proximate to the first cell, based on a timer before the UE loses the connection to the network.

Abstract: Embodiments of the technology disclosed herein provide an overall, or holistic view, of an enterprise's interaction with its customers. These embodiments can also provide a holistic view of other types of interactions. In one particular embodiment, a system collects and aggregates information related to user inquiries and/or response generated by different types of response systems. Such an embodiment could collect data about phone response system activities and aggregate that information with data about automated response system activities. Other embodiments collect and aggregate information related to customer information, contact resolutions and other information. Other embodiments of the disclosed technology generate reports based on aggregated information and/or generate recommendations to address problems with the individual response systems or the overall strategy for responding to customer inquiries.

Abstract: The present invention describes how a trusted network routing authority, such as a VoIP inter-exchange carrier or clearinghouse can provide routing and secure access control across multiple network domains with a single routing and admission request. This technology can improve network efficiency and quality of service when an Internet Protocol (IP) communication transaction, such as a Voice over IP (VoIP), must be routed across multiple devices or administrative domains. This technology defines the technique of performing multiple route look-ups at the source of the call path to determine all possible routes across intermediate domains to the final destination. The VoIP inter-exchange carrier or clearinghouse then provides routing and access permission tokens for the entire call path to the call source.

Abstract: Methods and apparatus to provide a network-based caller identification (CID) service in a voice over Internet protocol (VoIP) network are disclosed. An example method comprises receiving a session initiation protocol (SIP) communication session initiation request message comprising an early-media indicator, disabling a ringtone generator, sending a SIP ringing message containing a session descriptor in response to the early-media indicator, receiving a ringtone based on the session descriptor, and presenting the ringtone to alert a user of a requested communication session.

Abstract: A system for causing the simultaneous, or near simultaneous, receipt by multiple pre-identified communication termination endpoints of electronic messages sent at times respectively delayed relative to one another and inviting each of the communication termination endpoints to join a communication session. The system receives identification of associated communication termination endpoints that are to receive, at substantially the same time, electronic messages to join a communication session. The system determines and stores send time delays for each such communication termination endpoint that are calculated by a respectively appropriate method.

Abstract: Example methods, apparatus and articles of manufacture to determine authorization levels in a virtual private network are disclosed. A disclosed example method includes determining whether a call originated from an originating station associated with a virtual private network, the call originating via a private branch exchange associated with the virtual private network by a first user of a plurality of users associated with the virtual private network, querying a private-number table based on at least one of a switch identifier representing a service switching point, a trunk group identifier representing the private branch exchange, or an originating station identifier representing the originating station to obtain a private number assigned to the first user, and providing an authorization level associated with the private number to the service switching point, the service switching point to process the call according to the authorization level.

Abstract: A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service.

Abstract: A wireless telecommunication device and method of generating a domain ID from the wireless telecommunications network identifier of other communicating devices, such as a telephone number or mobile identification number (MIN), and creating a SIP URL to direct communications across a TCP/IP capable network. The wireless telecommunication device can generate more than one domain ID for same network identifier, and can include geographic information in the domain name to provide additional information about the other communicating device within the SIP URL address.

Abstract: A first call is received at a reserved address and is forwarded based on the reserved address. A second call is placed to a designated address, and a message is played to the designated address.

Abstract: Architecture for enabling user identities of callers to be collected from data sources and aggregated into respective meta-identities for each caller. Alternate user identities are searched, collected and associated with the meta-identity that can be a user name. A routing rule applied to the meta-identity is then applied across the alternate identities. The user identities can include a name of the caller, a phone number of the caller, or caller information collected from an external source. The phone numbers can include a partial phone number that is normalized into a full phone number format. The user identities can be mapped to the meta-identity and stored according to a hierarchy of confidence ratings. The user identities can be tagged with corresponding data source identifiers that designate respective data sources of the user identities. Conflict resolution is provided for selecting a suitable call routing path between callers having similar meta-identities.

Abstract: An application server that provides RBT treatment to a caller and simultaneously notifies the called party, as subscriber of the service facilitated by the present invention, of an incoming call, whereby the called party is able to select a treatment for the call to be presented to the caller based on input collected from the called party on a real-time basis.

Abstract: Communication networks and associated methods are disclosed that provide third party charging for SIP sessions. A communication network of the invention comprises a session control function (e.g., an IMS CSCF), a network node (e.g., an IMS application server), and a charging system. The session control function serves a SIP session initiated between a first and second party. The network node identifies a third party to charge for the session. The network node includes a third party identifier for the third party in a SIP message for the session, and transmits the SIP message to the session control function. The session control function generates a charging message, maps the third party identifier to the charging message, and transmits the charging message that includes the third party identifier to the charging system. The charging system then initiates charging of the session to the third party based on the third party identifier.

Abstract: Architecture and methods to provide an intermediary model for inter-carrier Multi-Media Messaging (MMS) services so that numerous wireless operators can transmit MMS messages between them. The architecture also provides for delivery, at least of notification messages, to those wireless operators who are not part of the overall inter-carrier MMS infrastructure.

Abstract: Methods, systems, and computer program products for number portability with local directory number support are disclosed. According to one aspect, the subject matter herein includes a system for providing number translation with local directory number support. The system includes a communications node for receiving, from a message source, a first message that includes information identifying a subscriber. The system also includes a number translation module, operatively associated with the communications node, for determining a network address and a local directory number associated with the identified a subscriber, and for at least one of: including the network address and local directory number in the first message and forwarding the first message to a destination of the first message; and including the network address and local directory number in a second message and sending the second message to the source of the first message.

Abstract: The present invention relates to the field of telecommunications. More particularly, to a method and an apparatus that would allow callers to time-share telephone, tele-communication or communication lines in order to truthfully vary the telephone line from which they are calling parties through a communication, telecommunication or Public Switch Telephone Network (PSTN). The system would allow customers to utilize methods that would manage their calls in ways that would maximize the probability that a called party would answer.

Abstract: The present invention relates to a method in a communication network for connecting at least one application server (AS) (213) of an internet protocol multimedia subsystem (IMS) (200) to an intelligent network (IN) (100) through an interface unit (301). The IMS (200) comprises a call server control function (CSCF) unit (205) connected to the at least one AS (213) arranged for processing call signaling. In the method the interface unit (301) directs the call signaling to the at least one AS (213) through the CSCF unit (205) and the at least one AS (213) processes the call signaling and sends the processed call signaling through the CSCF unit (205) back to the interface unit (301). Based on the information received in the call signaling processed by the at least one AS (213), the interface unit (301) then directs the call to a destination.

Abstract: A method, apparatus and software is disclosed in which a gateway server provides communications between an Internet Protocol (IP) network and Public Switched Telephone Network (PSTN) to enable messaging between an IN server device in the PSTN and a Session Initiation Protocol (SIP) Application Server (AS) on the IP network.

Abstract: To use the supplementary services of a conventional mobile communication service, such as Short Message Service (SMS) and Voice Mail System (VMS) service, there are problems in that an access method to each service must be known and several steps must be undergone. The present invention relates to a method and apparatus for sending voice messages in a mobile communication network and, more particularly, to a method and apparatus that gain access to the service providing page of the wireless Internet in such a way as to input the telephone number of a counterpart and a wireless Internet access key, and send a voice message to the counterpart.

Abstract: A method for providing voice telephony services can include the step of receiving a call via a telephone gateway. The telephone gateway can convey call identifying data to a resource connector. A media port can be responsively established within a media converter that is communicatively linked to the telephone gateway through a port associated with the call. A call description object can be constructed that includes the call identifying data and an identifier for the media port. The call description object can be conveyed to a telephony application server that provides at least one speech service for the call. The telephony application server can initiate at least one programmatic action of a communicatively linked speech engine. The speech engine can convey results of the programmatic action to the media converter through the media port. The media converter can stream speech signals for the call based upon the results.

Abstract: Embodiments of systems for selecting a communication path related to a networked contact center comprise a communication module configurable to receive data indicating that a customer of a tenant is seeking to initiate a communication with the tenant, and a path selection module configurable to register a first location, the first location being a geographical location of an agent associated with the tenant, determine a second location, the second location being a geographical location of the customer of the tenant, and based on the first location and the second location, establish the communication between the agent and the customer using a communication path having low communication latency.

Abstract: The present invention discloses a corporation switchboard platform and a call processing method. The corporation switchboard platform comprises a corporation switchboard service control point, a corporation switchboard service switching point and an intelligent peripheral, wherein the corporation switchboard service control point comprises: a basic call processing module for determining whether the capability of an existing network switching device can support a call triggered by a service switching point of a switching network and sending the call to an enhanced service capability module when determining that the capability of the existing network switching device cannot support the call; and an enhanced service capability module for controlling the intelligent peripheral and the corporation switchboard service switching point to complete a service related to the call.

Abstract: In accordance with an improved call routing methodology, a call is routed within a communication network by reference to one or more databases. In response to receiving a local number portability request at a service control point (SCP), the SCP accesses one or more databases to determine whether or not to apply custom call routing to a call directed to a called telecommunications number. In response to determining to apply custom call routing to the call, the SCP performs at least a second access to the one or more databases to obtain a local routing number (LRN) associated within the one or more databases with a calling telecommunications number associated with the call. The SCP thereafter provides the LRN to a service switch point in the communication network so that the call is routed to a called station associated with the called telecommunications number.

Abstract: A method, software and system of developing personalities for interactive and/or automated call center applications are provided. According to teachings of the present invention, sample population questionnaires and interviews may be used to identify key personality traits. The impact of the identified key personality traits are then empirically determined. Based on the empirical determination of the key personality traits' impact on customer satisfaction, a plurality of personality profiles may be generated for evaluation. A plurality of application types and voice talents may then be employed to evaluate the impact of each personality profile on customer satisfaction, for different user populations and different types of automated systems. The personality traits for each automated system are preferably rated and reviewed to ensure a system accurately represents the identified key personality traits.

Abstract: Systems and methods are disclosed for providing a usage monitoring telephony service. A service-providing network, which has the capability of monitoring attributes, including duration, for calls that traverse it, provides usage-monitoring services for calls directed to a telephony device on a native transport network. The native transport network is a network that naturally carries a subscriber's calls but does not have the capabilities required to provide usage monitoring. The system uses a virtual number to map a service in the service-provider network with a device in the native transport network. The usage monitoring services capture usage-monitoring data for multiple telephone numbers in multiple native transport networks.

Abstract: A method (200) for determining capability of a serving system for anchoring a call using an intelligent network protocol is provided. A communication network (100) includes a plurality of serving systems. Each serving system of the plurality of serving systems is associated with a unique identifier (ID). The method at a communication device (124) includes detecting (204) a movement of the communication device from a first serving system (102) to a second serving system (104) based on the unique ID of the first serving system and the unique ID of the second serving system. Further, the method includes accessing (206) a list that includes a mapping of the unique ID of each serving system with the intelligent network protocol capability of the serving system. Furthermore, the method includes determining (208), based on the list, whether the second serving system supports the intelligent network protocol for anchoring a call.

Abstract: In one aspect, a network-centric processing (NCP) system includes sensors configured to monitor activities associated with testing of an asset, a first system configured to provide the testing on the asset, a second system configured to provide training of personnel using the asset, a third system configured to record problems associated with the testing and the training of the asset as input for an acquisition process and a network connected to the first system, the second system, the third system and the sensors.

Abstract: A method of operating a communication network comprises receiving a call setup request from a calling party in a calling network to a called party in the communication network wherein the call requires an early media service provided by a service network to the called party, forwarding the call setup request to the service network wherein the service network returns a first call setup response in response to the call setup request, and generating a second call setup response and transmitting the second call setup response to the calling network wherein the calling network generates and transmits a call setup instruction to the calling party in response to the second call setup request.

Abstract: Included are embodiments for detecting presence in a call control protocol environment. At least one embodiment of a method includes subscribing to events associated with a communication system and receiving a notification for an event, the event being associated with a communication between a first communications device and a second communications device.

Abstract: Apparatus and methods for creating a composite data source having a common data representation from disparate sources of voice data. Data transmission links are established to heterogeneous messaging data sources, requests for voice data is sent using data access protocols, the voice data is received, and a set of voice data transformation rules are selectively applied to the voice data to transform the data into a common data representation. The common data representation can also be used as a source for reporting and graphical displays to monitor the operational aspects of the sources of voice data.

Abstract: A method and apparatus for enabling multiple services to be invoked in a specific precedence per call are disclosed. For example, the present method allows a subscriber to interact with a plurality of services provided by a service provider on a single call.

Abstract: Systems and methods for providing a call to a destination address are disclosed. Generally a first network element is queried regarding treatment of a call to a destination address. The call is then routed to the destination address when the first network element determines, based on an originating destination address associated with the call, that the call has been previously treated at a second network element.