Abstract: 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.

Abstract: Implementations of the present disclosure involve an apparatus, system and/or method for a central hub of a telecommunications network to provide varying services for a received communication intended for varying telecommunications networks. In one embodiment, the central service hub receives the communication and provides the varying services for the communication based on information concerning the communication, such as the intended telecommunication network. In another embodiment, the central service hub analyzes the communication and determines to which telecommunications network the communication should be routed to apply the appropriate network services. The central service hub may comprise a softswitch, media gateway controller device or other telecommunications device that includes programming, such as an application programming interface (API), configured to receive and process the incoming communications.

Abstract: The invention comprises a method and apparatus for providing at least one network endpoint of an enterprise customer network access to at least one dialing plan. Specifically, the method comprises receiving at least one dialing plan request from the at least one network endpoint, retrieving at least one address mapping in response to the at least one dialing plan request, and responding to the at least one dialing plan request using the at least one address mapping.

Abstract: A method includes receiving a telephone number mapping query associated with a call. The method may also include generating routing information based on the query and forwarding the routing information to a control device. The routing information may indicate that processing associated with routing the call is to bypass a core portion of an Internet multimedia subsystem (IMS) when a calling party associated with the call does not invoke an IMS service, the called party associated with the call does not invoke an IMS service and the called party is not an IMS subscriber.

Abstract: The present invention provides methods, systems and apparatus for providing service provider information for a user communication device. A targeted communication link associated with a target service provider is determined using a user communication device associated with a user service provider. The determination of the indicator for the target service provider may be with a user service provider Central Office or server. An indicator of the target service provider is received and displayed on the user communication device. Communication services available to the communication device are determined and presented to the communication device by audible or visual signals. Representations of services available or restricted to the user communication device are provided to the user device. Restricted services may be enabled by override options.

Abstract: A method and system for operating a multitenancy telephony system including a call queue that stores call requests received from a plurality of users; an expandable and contractible telephony resource cluster that establishes call sessions for call requests; a analysis system that calculates capacity requirements of the system; a resource allocator that manages the scaling and operation of the telephony resource cluster; and a plurality of telephony network channels that are used as telephony communication channels for call sessions.

Abstract: A system and method for enhancing voice telephone communications from a smart phone. The method is implemented on an app for a smart phone. Steps include capturing digits dialed on the smart phone; establishing a data connection with a database external to the smart phone; transmitting to the database the digits dialed on the smart phone; a telephone number for calling the smart phone; and any multimedia capabilities of the smart phone. Additional steps include receiving at the smart phone: a second telephone number sent from the database and any multimedia information accessible to the smart phone; displaying the multimedia information on the display screen when such multimedia information is displayable; and using the second telephone number received from the database to cause the smart phone to establish a voice connection through the public switched telephone network as if the second telephone number had been dialed by the user.

Abstract: A method may include storing a database including a plurality of subscriber identifiers, wherein each subscriber identifier is associated with a telephone number and a subscriber name and address. Each subscriber identifier may be unique to the corresponding subscriber name and address. The method may include storing a database including a plurality of customer identifiers. Each customer identifier may be associated with one of the subscriber identifier and a customer name and address. The method may further include receiving an update to the subscriber name or address associated with one of the telephone numbers and associating a new subscriber identifier with the updated subscriber name or address. The method may further include associating the new subscriber identifier with one of the customer identifiers.

Abstract: A call setup process (28) in a telecommunication device (10) is controlled, wherein the telecommunication device (10) receives a user command (22) to set up a call, initiates the call setup process (28), determines that the call is likely to reach a voicemail system (14), and, in response to the determination that the call is likely to reach the voicemail system (14), automatically terminates the call setup process (28). A telecommunication device (10) comprises related features. A telecommunication network (12) is adapted to support or implement a determination whether or not an incoming call is forwarded to the voicemail system (14). The invention improves the user experience if a called party does not personally answer a telephone call.

Abstract: Disclosed herein are systems and methods for sending and receiving facsimile transmissions in a voice-over-IP system. In certain embodiments, a facsimile machine may include a network interface and a call set up protocol client configured to interface with a call set up protocol server. The call set up protocol client may communicate with the call set up protocol server using the network interface to establish a communication channel with the public switched telephone network. The call set up protocol client may operate according to the session initiation protocol. The facsimile machine may be configured to send and receive facsimile transmissions according to the T.30 protocol. In alternative embodiments, the facsimile machine may be configured to send and receive facsimile transmissions according to the T.38 protocol.

Abstract: Example embodiments of the present invention may include a method and apparatus configured determine an optimal routing path to route a call through a network. One example method may include retrieving and parsing a routing table and storing the parsed routing information in a routing database. The method may also include determining a plurality of routes that could be used to route the call through the network by applying at least one cost metric to the plurality of determined routes. The method may also include determining an order of the plurality of routes used to route the call, and create a route used to route the call based on the determined order of the plurality of routes.

Abstract: A method, computer program product, and an apparatus for transferring a telephone call and associated data from an IVR system to a live agent or other system are provided. An exemplary embodiment may include a voice portal of the IVR system requesting an access number for a live agent and sending data associated with the telephone call to a CTI system. The CTI system may request and receive a routing label from a router and then send the data associated with the telephone call and the request for an access number to a call center using the routing label provided by the router. The CTI system may receive the access number from the call center and send the access number back through the IVR system to an exchange carrier for transferring the caller to the call center.

Abstract: Embodiments of the present invention relate to the field of communications and provide a session transfer method, an application server, and a communications system, which implement a cross-network session transfer between a CS domain and an IMS domain. The method includes: after receiving a session transfer request sent by user B or user C in the IMS domain, sending a release message to the user B currently in a call to terminate a session between user A in the CS domain and the user B, and negotiating media information of the user A and media information of the user C to establish a session between the user A and the user C. The embodiments of the present invention apply to a cross-network session between the CS domain and the IMS domain.

Abstract: Methods, systems, and products are disclosed for enabling an exchange of electronic healthcare messages. An electronic message is received from a sender's address that is destined for a recipient's address. The electronic message may have an attached object associated with a sender's legacy system. The sender's address and the recipient's address are associated to a list of health care participant addresses. When both the sender's address and the recipient's address are matched to the list of health care participant addresses, then the electronic message is sent to a rule-based engine. When the sender's address and the recipient's address do not match the list of health care participant addresses, then an identifier is received that associates the electronic message with another legacy system and with a non-participant legacy system.

Abstract: A mid-point call management node subject to monitoring through a workstation communicatively coupled thereto, provides call services (e.g., through extensible markup language (XML), and in particular call control extensible markup language (CCXML) and/or voice extensible markup language (VXML), instructions) for an inbound call received from an originating network at an originating point-of-presence (POP) associated with multiple, disparate call centers, the call services being provided in response to call management application instructions issued according to enterprise-specific strategies for optimizing call handling between the originating POP and domestic and/or international ones of the disparate call centers communicatively coupled thereto. Call center information (e.g., call load information) received at the management node from the multiple call centers may be used in connection with providing the call services.

Abstract: In one embodiment, the invention provides a method. The method includes dynamically allocating a telephone number to an advertisement on a just-in-time basis; and if the telephone number is not called for a predefined period of time, then unallocating the telephone number.

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: An application server system is provided having a core engine connecting one or more protocol providers to one or more applications. In a preferred embodiment, the one or more applications run on a respective one or more virtual machines having their own execution threads. A thread pool may be used to spawn instances of applications or scripts. Preferably, the core engine, the protocol providers, and each of the one or more virtual machines run in separate AppDomains. In another embodiment, a visual designer is provided preferably having XML conversion capability and interactive debug capability.

Abstract: An intelligent interactive call handling system is provided that typically includes a central office, a call-handling device, and an internet call routing system. The central office typically triggers a query responsive to receiving a call request. The call-handling device is coupled to the central office, receives the query, and triggers an internet call routing query. The internet call routing system, which is coupled to the call-handling device, typically receives the internet call routing query, determines presence of the called party with respect to at least one registered communication device, sends a prompt to the called party at said at least one registered communication device responsive to the presence determination, receives a reply from said at least one registered communication device, and routes the call responsive to the reply. Methods and other systems are also provided.

Abstract: An exemplary method provides access to call origination features contained in an internet protocol multimedia subsystem (IMS) for a subscriber of the IMS who originates a call request from customer premises equipment (CPE) outside of the IMS. The call request is received at a public service identity (PSI) server in the IMS with a phone number to which the call request is routed. An IMS internal desk number is retrieved based on the number of the outside CPE. A first modified call request is transmitted from the PSI server to a telephony application server (TAS) identifying the IMS internal desk number as a request universal resource identity (RURI). Call origination treatment as specified by the subscriber is performed by the TAS.

Abstract: A peering authority or settlement clearinghouse can be used to control access, collect session accounting information, and provide financial settlement of interconnect or session fees among anonymous Internet Protocol (IP) peers or networks. The addition of peering policy criteria, such as price and quality of service, to peer to peer route discovery mechanisms enable a trusted intermediary, such as the settlement clearinghouse, to authorize acceptable interconnection or peering sessions between anonymous IP peers. Any financial settlement transactions which result from the peering sessions may be subsequently executed by the settlement clearinghouse.

Abstract: Implementations of the present disclosure involve an apparatus, system and/or method for a central hub of a telecommunications network to provide varying services for a received communication intended for varying telecommunications networks. In one embodiment, the central service hub receives the communication and provides the varying services for the communication based on information concerning the communication, such as the intended telecommunication network. In another embodiment, the central service hub analyzes the communication and determines to which telecommunications network the communication should be routed to apply the appropriate network services. The central service hub may comprise a softswitch, media gateway controller device or other telecommunications device that includes programming, such as an application programming interface (API), configured to receive and process the incoming communications.

Abstract: Systems and methods are disclosed for a media terminal adapter (MTA) that contains a routing table for routing incoming telephone calls based on caller identification information. The MTA is coupled to a router/hub, where all digital telephones are then connected to the router/hub. Any analog telephones are connected to the MTA. The routing table defines incoming calls using their caller identification information and associates each defined call with a corresponding telephone(s). In this manner, an incoming call that is intended for one predominant telephone user is routed to the desired telephone using the telephone's IP address or a physical connection interface.

Abstract: A method and system for providing call forwarding in an IP telephone network is disclosed. First, when a telephone number for a first telephone from a second telephone is dialed, the call is routing to a call manager. Call setup procedures with a BRG responsible for the first telephone are then initiated. The BRG then checks stored call forwarding profiles to determine whether there is an active call forwarding profile for the first telephone. The call is connected to the first telephone if an active call forwarding profile is not found. However, if an active call forwarding profile is found, the call forwarding information is sent to the call manager. The call is routed to at least a third telephone based on the call forwarding information.

Abstract: A dual network telephone device selectively performs a telephone communication via a public telephone network and via an IP telephone network. The dual network telephone device generates IP telephone identification information for a non-IP telephone device registers to an IP telephone server an association of an IP address of the dual network telephone device and the IP telephone identification information for the non-IP telephone device, receives a first call request sent from the IP telephone server, sends a call request notice instruction to the non-IP telephone device when the first call request includes the IP telephone identification information for the non-IP telephone device as a call request destination, receives IP telephone identification information input to and sent from the non-IP telephone device, sends a second call request to the IP telephone server.

Abstract: A method for negotiating a security capability when a terminal moves is provided. When a user equipment (UE) moves from a second/third generation (2G/3G) network to a long term evolution (LTE) network, the method includes the following steps. A mobility management entity (MME) acquires a non-access signaling (NAS) security algorithm supported by the UE, and an authentication vector-related key or a root key derived according to the authentication vector-related key, selects an NAS security algorithm, derives an NAS protection key according to the authentication vector-related key or the root key, and sends a message carrying the selected NAS security algorithm to the UE. The UE derives an NAS protection key according to an authentication vector-related key thereof. A system for negotiating a security capability when a terminal moves, a UE, and an MME are further provided.

Abstract: In a contact center, a system for processing communication events has an interaction server for managing events waiting to be routed, a routing server for routing the events, a rules engine, and a gateway server executing rules invocation logic and interacting with the rules engine. The interaction server tracks incoming events, initiates and sends a routing request to the routing server, including a special routing object, for each event that requires conformance to business rules, the routing server, executing the routing object, sends an execution request to the gateway server, which upon receipt of the request prepares a business object model (BOM), being a set of facts upon which the rules engine executes rules appropriate to the facts, resulting in routing direction for the routing server to route the event.

Abstract: Architectures, arrangements, systems and/or methods for an Internet protocol (IP) telephony system for enterprise applications that is efficiently provisioned and maintained are disclosed. In one embodiment, a single platform uses a label allocation and routing scheme to provide overlapping dial plans for multiple customers, E.164 mapping and enhanced regulatory enhancements for both wireline and wireless telephony devices.

Abstract: A method and system for providing reliable path for Direct Inward Dial (DID) calls on networks, e.g., Time Division Multiplexed (TDM) networks, the packet networks Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) networks are disclosed. In one embodiment, the present method enables the customers to request for a reliable path for Direct Inward Dial (DID) calls. For example, the present method enables a service provider to connect a customer's switch, e.g., a PBX, to two or more switches (or softswitches) via two separate DID trunk groups. The method also enables the service provider to create a DID trunk group between the two or more switches (or softswitches) connected to the customer's switch. With this architecture, the service provider is able to continue to forward DID calls towards the customer's switch even if one of the two separate DID trunk groups has failed.

Abstract: A method may include receiving a telephone number mapping query from a querying device, the telephone number mapping query being associated with a call from a calling party to a called party. The method may also include retrieving address information based on the telephone number mapping query, the address information including addresses associated a number of network devices. The method may further include retrieving call type information associated with each of the addresses, and forwarding the address information and the call type information to the querying device.

Abstract: A method and system for registering a client for a call-pickup service and for executing the call-pickup service for a new call placed to a given communication device. The call-pickup service may carried out, in part, by providing the client with notification of when the new call is placed to the given communication device, the client requesting that the new call be re-routed to a location other than the given communication device, and a server re-routing the new call to the location other than the given communication device. The location other than the given communication device may be the client that registers for the call-pickup service or another location.

Abstract: A terminal including: a remote control section for transmitting and receiving data with respect to a main device which performs call control processing with a target device via a telephony server in place of the terminal; and a call communication section for performing audio communication with the target device, wherein the remote control section transmits a command including a calling request for the target device and an IP address of the terminal to the main device, and receives an IP address of the target device from the main device, and the call control section performs audio communication with the target device using the IP address of the terminal and the IP address of the target device.

Abstract: When a communication terminal determines that restriction processing is to be applied based on a sound input through the communication terminal or a user instruction, the communication terminal restricts output of conference information received from a counterpart terminal.

Abstract: Methods, systems, and products are disclosed for routing a communication. An identifier of an origination point of the communication is received. A destination point associated with a destination of the communication is determined. The origination point and the destination point are analyzed to determine whether the communication may be entirely routed within a local service provider's geographic area.

Abstract: A positioning system (e.g., GPS) integrated in, or coupled to, a mobile device determines the position of the mobile device. An availability status manager (ASM) in a vPBX acquires the position of the mobile device. The ASM can contain rules on how to route or otherwise control incoming phone calls based on availability status of one or more mobile devices in the vPBX network. When a call is received by the vPBX, the ASM determines the availability status of the receiving mobile device based on the geographic location of the receiving mobile device. The ASM then routes the call to the receiving mobile device, to another mobile device, or to a landline based on the availability status of the receiving mobile device and the availability status based rules.

Abstract: A system may include a destination device that receives information about a caller via a first outgoing leg and exchanges content with the caller via the first outgoing leg. The system may further include a network device that receives an incoming call from the caller, establishes the first outgoing leg, connects the incoming call to the first outgoing leg so the content can be exchanged, monitors the first outgoing leg, selectively establishes a second outgoing leg in response to the monitoring, and connects the incoming call to the second outgoing leg.

Abstract: A system and method are disclosed for intelligent formatting of VoIP telephone numbers. The intelligent VoIP formatting system includes a presentation and user interface layer, an E.164 formatting engine, a location routine, and a database of telephone number data which are used to deconstruct input telephone numbers and reconstruct them as E.164 compliant telephone numbers.

Abstract: A configurable PSTN call-handling (CPC) service enables monitoring and custom responses to calls received over multiple subscriber lines. A CPC Client Program runs on a computer and presents a graphical user interface (GUI) that allows the user to associate call-handling options with call characteristics. The GUI also allows the user to customize the notification signals provided and to provide custom responses to certain callers, by associating call characteristics with various GUI appearances, sound files, or images. The CPC relieves the communications network of performing certain call-handling functions, such as call forwarding, by moving decision-making processes to the user's PC. The user can remotely monitor any subscriber line associated with the user from any browser, because the communications network sends notifications directly to an IP address specified by the user. The IP address associated with a computer connected to the subscriber line need not be the IP address specified by the user.

Abstract: A method and apparatus for providing a reliable Voice Extensible Markup Language (VXML) over packet networks such as Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) network are disclosed. For example, a service provider may utilize a plurality of content servers that can be accessed by at least one telephony browser. The telephony browser can reach the content browsers directly as well as through a shared server that may load balance among the content servers. When a request for a VXML content, e.g., a VXML application, is received, the telephony browser sends the request to the shared server. If the request fails or a response is not received prior to expiration of a predetermined time interval, then the telephony browser sends a second request directly to one of the content servers that is capable of providing the requested content.

Abstract: One or more aspects of the disclosure relate to various functions and processes related to routing of calls originating from other service providers. For example, a method includes receiving a request to route a call or a call session, such as a session initiation request, from a second service provider and proceeding to route the requested call or session through a call routing facility of the first service provider to a requested destination of the call. Routing the call through the call routing facility may include determining a service provider type and/or an origination number type for the call, determining routing information for the call using the type information, and causing routing of the call to a destination of the call based on the routing information. Further aspects of the disclosure relate to various functions and processes related to accessing routing data and/or information from a routing database.

Abstract: The disclosure herein provides data security on a parallel computer system using virtual private networks connecting the nodes of the system. A mechanism sets up access control data in the nodes that describes a number of security classes. Each security class is associated with a virtual network. Each user on the system is associated with one of the security classes. Each database object to be protected is given an attribute of a security class. Database objects are loaded into the system nodes that match the security class of the database object. When a query executes on the system, the query is sent to a particular class or set of classes such that the query is only seen by those nodes that are authorized by the equivalent security class. In this way, the network is used to isolate data from users that do not have proper authorization to access the data.

Abstract: A telephone handset is provided with a user interface which includes a display unit, audio output unit and various user actuable keys. A phone controller implements and responds to instructions from the user interface. Where the instructions from the user interface request the connection of a call, the phone controller connects the handset to a PSTN via one of a plurality of connection means provided. Similarly, in the event of an incoming call connection request received via any one of the connection means, the phone controller enables the connection of the call. The user may choose to accept or reject the connection using the interface. The handling of an incoming or outgoing call is determined by a connection routing engine. The connection routing engine comprises an identification unit, a rules unit, a comparison unit and an enabling unit.

Abstract: A system for handling an outbound call from a call originator to a call target including a database storing a plurality of outgoing telephone numbers, each outgoing telephone number having one of two or more area codes; and an information processor controlled by the call originator. The information processor configured to process a trigger comprising at least an area code of a telephone number of the call target; select from the database a telephone number from the plurality of outgoing telephone numbers where the selected telephone number has at least an area code the same as the area code of the telephone number of the call target; set caller identification data of the outbound call to the selected telephone number; and transmit the caller identification data to the call target in connection with the outbound call.

Abstract: Systems, apparatuses and methods of determining whether to permit a call to an access terminal are disclosed. A second party is determined to be available for a communication session between a first party and the second party. Location information associated with a call permissions rule set is obtained, where the call permissions rule set includes at least one rule related to whether a communication session between the first and second parties is permissible. Further, at least one rule includes one or more location-based rules. Location information is obtained that is associated with the second party. Then, it is determined whether to permit the communication session based on the obtained location information associated with the call permissions rule set and the obtained location information associated with the second party.

Abstract: Systems and methods for providing multi-homed tandem access in a communication system are disclosed. The disclosure may include a private packet network backbone exchange (PPNBE) in connection with a set of access tandems and with a call destination such as an end-user or a communications service. A set of LRNs may be homed across the set of access tandems, with each of the set of LRNs mapped to one or more TNs corresponding to the call destination. An originating party may use any of the TNs to reach the call destination. A plurality of originating calls each including one of the TNs may be received at the PPNBE from any of the access tandems and routed to the call destination. Thus, the present disclosure provides greater call capacity than available access tandem architectures as well as optimizes a maximum number of call paths to a particular call destination.

Abstract: Efficient telephone number mapping (ENUM) based call routing during area code splits is described. An ENUM domain management component can associate an ENUM domain name with multiple records when a numbering plan area code split links an old area code of a telephone number corresponding to the ENUM domain name with a new area code. An ENUM provisioning component can perform permissive dialing of the old and new area code during ENUM based call routing by utilizing the multiple records.

Abstract: Customer communications can be controlled in accordance with customized rules. An initial menu includes options to modify an existing rule or create a new rule for controlling customer communications. An initial selection criteria menu includes options to indicate whether the new customized rule will be built using a preexisting template or starting from initial blank rule criteria. A list of preexisting templates for creating new customized rules can be provided, and input to populate a selected template can be accepted. Initial blank rule criteria for creating new customized rules can also be provided, including initial criteria and rule conditions for selected initial criteria. Communications are processed in accordance with the stored new customized rule.

Abstract: A system and method for processing toll free telephone calls by a telecommunications service provider may include receiving, by the telecommunications service provider, a telephone call request for a telephone call being placed to a toll free telephone number from a telephone within a service area of the telecommunications service provider. The telecommunications service provider may determine that the toll free telephone number is one operated by the telecommunications service provider. The telephone call may be routed via a network of the telecommunications service provider to a telephone associated with the toll free telephone number to connect the telephone call.

Abstract: A method for location relevant directory assistance through a PSTN is provided. The method can include connecting a calling party with a called party through a switch in a PSTN, receiving a directory assistance request from the calling party over the PSTN, and querying a database disposed within the PSTN to retrieve caller identification for the calling party. The method can further include retrieving from over a data communications network separate from the PSTN a call address in response to the querying of the database for the calling party from the database disposed within the PSTN. Location relevant directory information can be identified from a database that is external to the PSTN based upon the retrieved call address. Subsequently, the directory assistance request can be responded to using the identified location relevant directory information.

Abstract: Method and system of establishing a connection between a calling terminal (1) and a called terminal (2) which allows the called terminal (2) to receive calls trough two different networks (3, 4), for example a circuit switched network and a Internet protocol Multimedia Subsystem, using a single number, thanks to a control element which determines the network to be used to reach the called terminal (2), and its number in said network to be used.