Abstract: A method for distributing incoming telephone calls among inbound links of a service platform is provided. First information is accumulated regarding routing of the incoming telephone calls during a previous time period to the inbound links of the service platform by a network call processor. Second information including a number of the inbound links of the service platform allocated to the network call processor for a future time period is generated based on the first information. A number of the incoming telephone calls are routed to the inbound links of the service platform during the future time period based on the second information.

Abstract: A method of routing a call includes sending a second query when a telephone number mapping (ENUM) database does not identify an internet protocol (IP) address associated with a called number in response to a first query. The second query is directed to a local number portability (LNP) database to determine whether the called number includes a ported number having an associated local routing number (LRN).

Abstract: A wireless communications device that sends and receives presence status information is configured to detect when it is proximate a predefined zone. A receiver in the wireless communications device receives presence information from a transmitter associated with the predefined zone. The presence information defines a new presence status for the wireless communications device, and may include an indicator associated with the new presence status. Upon receipt of the presence information, a controller in the wireless communications device automatically changes the current presence status of the wireless communications device to the new presence status, and publishes the changed status to a remote entity. If the new presence status is not defined in memory, the controller may add the new presence status to memory.

Abstract: Operations of a communication network may include receiving an initiating communication from an initiating device directed to a network administration application, wherein the initiating communication includes an identification of the initiating device and an identification of the network administration application. A target device for which action is being requested may be identified by the initiating device, and a command communication may be transmitted from the network service application to a subscriber line providing service for the target device. More particularly, the command communication may include the identification of the initiating device, an identification of the target device, and a code identifying the action being requested by the initiating communication. Related systems and computer program products are also discussed.

Abstract: A method and apparatus for unified overlay dialing enables simplified telephone dialing to occur within a standard overlay area and eliminates disruption dealing with area code saturation. Each three-digit area code within an overlay area is associated with a single-digit suffix, with the parent area code obtaining the suffix “0”. Intra-overlay area dialing then proceeds on the basis of a seven digit number plus a one-digit suffix, rather than a full 10 or 1+10-digit number. For dialing to the parent area code from within the overlay area, the suffix may be omitted entirely, and the expiration of a suitable timing delay causes the seven digit number provided to be dialed into the original (parent) area code. This method can be implemented in connection with a telephone company central office switch, a modified custom calling service or be made accessible to individual telephone subscribers using computerized customer premises equipment. It can also be used to simplify dialing in non-overlaid situations.

Abstract: A method and apparatus for connecting a plurality of called party terminals in a bridged a call session with a Voice Over Internet Protocol (VOIP) calling party terminal. The method includes the VOIP terminal sending called party terminal identifiers identifying each called party terminal to a calling party Service Control Point (SCP), and creating a bridged call session at the SCP connecting the called party terminals and the calling party VOIP terminal. The apparatus includes a system for sending called party terminal identifiers identifying each called party terminal to a calling party Service Control Point (SCP), and creating a bridged call session at the SCP connecting the called party terminals and the calling party VOIP terminal.

Abstract: 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.

Abstract: A selective router interface translator for placement between legacy selective routers and their location databases translates location queries from legacy selective router interfaces to newer protocol location databases using newer protocol interfaces, e.g., a NENA V8 protocol query to an Emergency Routing DataBase (ERDB), or a Location to Service Translation (LoST) protocol query to a LoST server. In this way existing selective routers may remain as they are today, yet be flexible enough to allow the databases they query to be developed to new standards. The disclosed selective router interface translator includes a NENA V3 protocol to LoST protocol translator to translate a NENA V3 query for location of an emergency caller into a LoST protocol query. The selective router interface translator also comprises an ALI to ERDB protocol translator to translate an ALI query for location of an emergency caller into an ERDB protocol query.

Abstract: Methods, systems, and products are disclosed for routing communications to contact centers. When an incoming communication is detected, calling party identification information is associated to a communications profile. When a called party has multiple communications devices, then a location is received for each of the called party's multiple communications devices. The incoming communication is then routed based on the location of the called party's multiple communications devices.

Abstract: The present invention proposes a method for transmitting data in a communication network, said communication network comprising at least one call control entity (CSCF) for controlling calls to be established from a subscriber (A, B) to a communication destination (B, A), a service control entity (SCP) for controlling services provisioned in said network for a respective subscriber, and a detection entity (DP) adapted to detect at least one of a plurality of predetermined conditions occurring in said network; said method comprising the steps of: forwarding a signaling message from said subscriber (A) to said at least one call control entity (CSCF) for being further forwarded to said destination (B), said signaling message comprising a message header and a message body, said message header including information descriptive for a content carried in the message body, detecting the occurrence of said at least one predetermined condition, and in case the occurrence of said predetermined condition has been detected,

Abstract: The present invention is directed to a method and system for routing communications. A plurality of communication devices are coupled to a communications network. Each communication device has a device identifier. A user-programmable database associates each of a plurality of user identifiers with a device identifier. A call-routing switch arrangement is coupled to the communication network through a portal communication device having a portal identifier. A user designates a destination user identifier when making a communication. The call-routing switches the call, directing the call to a destination device identifier responsive to the user-programmable database association between the destination user identifier and a device identifier. According to the present invention, communications are directed to users and are routed to associated equipment, a user identifier serving as a virtual extension number.

Abstract: A system, method, apparatus, means, and computer program code is provided for routing a call which includes receiving a call request at a first server, the call request including header data identifying a requesting extension and an invited extension, the invited extension associated with a second server. The header data identifying an invited extension is then replaced with proxy data for the invited extension, and the call request (including header data identifying the requesting extension and the proxy data for the invited extension) is forwarded to a second server.

Abstract: An access point operating in a wireless communication network includes a smart antenna for generating directional antenna beams. A method for operating the access point includes communicating with a client station using a selected directional antenna beam, with the client station initially being in a stationary position. Signal to noise ratios of signals received from the client station within a time interval are measured. At least one variation metric of a mean of the measured signal to noise ratios within the time interval is determined. This at least one determined variation metric is compared to a threshold for determining if the client station is moving.

Abstract: A system, method and software for implementing an automated call routing application in a speech enabled call center environment are provided. In operation, the invention provides for the identification of a call center transaction selection from a natural language user utterance and the invocation of one or more scripts operable to route the user to a call center service agent configured to service the selected transaction. In the event a transaction selection cannot be readily identified or can only be partially identified, the invention provides for the initiation of a dialog module or script directed to eliciting a discernable transaction selection and/or the presentation of one or more menus from which the user may select an available call center transaction.

Abstract: A VoIP terminal, which supports to make a phone call through both PSTN and Internet is disclosed. A local subscriber may choose to have a conversation through PSTN or Internet on his own. Besides, by reference to a switch such as a relay, the VoIP terminal is able to establish a PSTN connection under some special cases, such as the terminal power off or Internet shutdown.

Abstract: There is provided herein a system and method for automatic configuration of data routings for use with electronic data such as phone calls, faxes, etc. In the preferred embodiment, when more than one carrier might potentially terminate the transmission, the carriers are ordered based on some screening criterion (e.g., transmission price). Data transmissions are then assigned to the carriers based on the sorting order, with the second place and lower carriers (e.g., the higher priced carriers) not being selected unless the first carrier cannot complete the transaction. The switch instructions necessary to implement this scheme are generated automatically.

Abstract: A data network comprises proxy-call session control functions (P-CSCFs) serving user equipments. Each P-CSCF can request resource reservation from an associated policy manager. A serving-call session control function receives a first call session setup message and determines a set of terminating user equipments associated with a terminating user identity of the setup message. It then transmits a call session initialization message to each identified terminating user equipment via an associated P-CSCF. This message includes a session identity indication and a forking indication which indicates if the first call session is a forked call session. The P-CSCFs and/or the policy managers then restrict the resource reservation for two or more user equipments having the same session identity and forking indications indicative of a forked call session setup to the resource requirement for only one of the user equipments. This may reduce resource usage for forked call sessions.

Abstract: A method and system for handling identical VPN numbers assigned to multiple companies using a nature of address/number identifier initiated as public. The method includes receiving a query from a switch module to determine routing information for a call initiated by a calling party, the query including dialed digits with a nature of address identifier, and a calling party identifier updating the nature of address identifier associated with dialed digits to a private type code if a VPN coding identifier is determined present, looking up routing information associated with the dialed digits in a look-up table corresponding to the calling party identifier if the nature of address identifier includes the private type code, and transmitting the routing information to the switch module.

Abstract: A system and method for processing a plurality of requests for multi-media services received at a call control element (CCE) defined on the system from a calling communication device. The system includes a Network Routing Element, a Service Broker (SB), at least a primary media severs (MS) and at least a secondary MS, a plurality of application servers (ASs) and a plurality of border elements, all of which are coupled to the CCE. The SB is adapted to receive a plurality of requests including parameters for requesting multi-media services, via the CCE, and to selectively redirect the requests to one or more ASs for providing feature processing for the requests. The ASs can instruct either the primary MS or secondary MS, via the CCE, to form a preliminary interactive communication path with the calling communication device for collecting caller-entered data, which can be validated prior to providing the feature processing.

Abstract: Managing packet data network jitter is disclosed. A first call data associated with a mobile network communication session is received. A second call data that is older than the first call data is dropped from a buffer if required to make room in the buffer for the first call data.

Abstract: A method includes jurisdictionalizing a long distance intra-MTA call as a local call. The method may further include receiving an intra-Major Trading Area (MTA) call, determining whether the intra-MTA call satisfies terms of use of a direct connection to a terminating network, and routing the intra-Major Trading Area (MTA) call directly to the terminating network via the direct connection if the intra-MTA call satisfies the terms of use of the direct connection. A method for routing calls includes determining that a received intrastate call is an intra-MTA call, determining whether the intra-MTA call satisfies terms of use of a direct connection bypassing a Local Exchange Carrier, and routing the intra-MTA call for termination via the direct connection if the intra-MTA call satisfies terms of use of the direct connection. A system includes a switch configured to jurisdictionalize a long-distance intra-MTA call as a local call, and may be further configured to route the call over a local route.

Abstract: Methods of controlling media server resources in a VoIP network are disclosed. In an embodiment, an IP node provides a service request. An application server receives the service request and sends a request for media server resources to a media server resource broker. The media server resource broker determines that the request should be handled by a first media server. The media server resource broker queries the first media server to obtain an IP address and port number for use in establishing a call between the IP node and the first media server. The media server resource broker then provides a signal to the IP Node so that it can establishing the call with the appropriate port on the media server. In an embodiment, the media server resource broker updates a database module that tracks assignment levels of the first media server so as to reflect the most recent request, thus decreasing the number of ports available for the first media server.

Abstract: In a system having a switching device in a first location, and an extending device in a second location, a method of providing a communications link between (i) a first phone associated with the extending device and (ii) a second phone local to the second location and not associated with the extending device includes analyzing parameters associated with the first phone and the second phone so as to determine that the first phone and the second phone are local to the second location. The method further includes separating signaling data associated with the communications link from voice traffic associated with the communications link. The method further includes directing the signaling data through the switching device via an extension link between the switching device and the extending device, and directing the voice traffic between the first phone and the second phone.

Abstract: A system and method of delivering targeted advertising to packet-based network telecommunications devices may include accessing a database configured to store telephone services information of telecommunications devices. The telecommunications services information may indicate whether a telecommunications device has packet-based network access. Telecommunications devices that have telephone number information including NPA-NXX of a certain NPA-NXX code and packet-based network access may be identified. Advertising content may be communicated to the identified telecommunications devices for display on the telecommunications device.

Abstract: Systems and methods to route a call over a voice and data network (VDN) are provided. A particular method includes receiving a call from a calling device at a telecommunications gateway (TCG). Authentication data is received via the call. A determination is made whether a user account associated with the authentication data is authorized to route calls via the TCG to a VDN. When the user account is authorized to route calls via the TCG to the VDN, call data received at the TCG via the call is converted into a format compatible with the VDN. The converted call data is sent via the VDN to a destination device.

Abstract: The present invention relates to a method of decoding tokenized Session Initiated Protocol. A token from a tokenized Session Initiated Protocol packet is received. A decoding table is accessed and the corresponding entry for the token is determined. The token is executed according to the corresponding entry. This may be done for all tokens in a tokenized Session Initiated Packet. Once the tokens have been executed according to their entries in a decoding table, the decoded tokenized Session Initiated Protocol packet is provided.

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

Abstract: A universal linking gateway interfaces between a local network of one or more local phones, such as analog phones or PBX phones, a plurality of cell phones and multiple service lines, including a public switched telephone network. Implementation of industry standard protocols such as hook flash signals enable a simplistic local phone to access enhanced calling features including answering calls on multiple service lines, to select and make calls on any desired service line as well as handle simultaneous calls on either an active service line or multiple service lines. Further, the gateway manages security and data transfer features between the gateway and the telephonic devices connected thereto. The gateway interfaces between a variety of other telephonic devices including PSTN, VoIP, and telephonic intercoms. Wireless communications are controlled by the gateway for proper pairing management.

Abstract: Systems for monitoring, storage, and analysis of information in signaling messages, communicated from originating networks to a destination network through one or more intermediate networks are described. Correlation and comparison between signaling messages of interest sent from an origination network to signaling messages received by the destination network allows for the detection of inconsistent information, e.g., information fields in a communicated message that have been altered, either intentionally or unintentionally, by an intermediate carrier. The introduction of an additional communications path over which notification messages are forwarded from signaling monitoring equipment in the origination network to signaling monitoring equipment in the destination network allows for a significant reduction in the memory storage requirements at the destination network and in the amount of processing required to match, compare, and evaluate signaling messages.

Abstract: According to an aspect of the present invention, there is provided a Session Initiation Protocol (SIP) telephone system comprises a server unit, telephone terminals and a module. The server is connected to Internet Protocol (IP) network. The telephone terminals transmit and receive SIP messages to and from the server unit via the IP network. The module applies SIP messages regarding event notification to data transmission to form interactive communication paths among the server unit and each of the telephone terminals.

Abstract: A communication terminal unit has a system control section including a call destination specification information analysis section that analyzes a destination number which is assigned to a communications terminal unit at the other end and input at the time of origination of a call. On the basis of a result of analysis of the destination number performed by the call destination specification information analysis section, a call origination network determination section automatically determines to originate a call over an IP network or over a public network through use of the destination information that has been subjected to analysis and through use of selected network setting information stored in a selected network setting storage section. Selected network setting information required to automatically determine to originate a call over the IP network or over the public network is stored in the call origination network determination section.

Abstract: A method and apparatus for enabling Application Servers to automatically update the databases used by Call Control Elements as changes occur between customer data, such as customer specific logic, and the Application Servers, such as the IP addresses of the Application Servers are disclosed. Whenever there is a change in the location of customer specific data needed by the CCEs, e.g., switching from one AS to a new AS, the new AS will automatically update the relevant database in the CCEs to indicate such an update has occurred. After the automatic update is performed, the CCEs will be able to communicate with the correct AS to retrieve and process the customer specific service logic.

Abstract: A network appliance (100) is provided having a network controller subsystem (110) for coupling the appliance (100) to a data network for providing and receiving data packets to and from a packet data network. A digital signal processing subsystem (120) is coupled to the network controller subsystem (110). A signal conversion subsystem (130) is coupled to the digital signal processing subsystem (120) and a user interface subsystem (160) is coupled to both the signal conversion subsystem (130) and the digital signal processing subsystem (120). The digital signal processing subsystem (120) operates under the control of a computer program which is capable of detecting incoming calls, initiating call sessions, and preferably, implementing advanced telephony features.

Abstract: The present invention relates in general to video communication networks and more specifically to a method for interworking between a number of video communication terminals, to a video communication system, and to a video interactive gateway unit. The solution according to the present invention makes it possible to interwork between different terminals e.g. between a low bit-rate multimedia terminal and a packet based multimedia terminal.

Abstract: A call between a calling party and a called party, one or both of whom may be subscribers to Internet Telephony (IT) services, commences upon the receipt of a call dialed by the calling party to the Plain Old Telephony Service (POTS) number associated with the calling party. A first hub receives the call and routes it to the called party if that party is not an IT services subscriber that is currently on line. If the called party is an IT services subscriber that is on-line, the call is received at an Internet Services Provider serving the called party. The ISP converts the call to an IT format if the call is not already in that format and thereafter delivers the call to the called party.

Abstract: The present invention provides a method and devices for unified messaging. One method provides for receiving a message having a first identifier associated with a user, translating the first identifier associated with the user to a second identifier comprising a zip code and a street address, the second identifier being associated with a network address, and sending the message to the user at the network address. A line interface device of the present invention is associated with an address that comprises a zip code.

Abstract: In various embodiments described herein a TCP/IP based voice communication system is described. The TCP/IP based voice communication system may be useful in a correctional facility or other environments such as college campus, hospitals or other institutions. In addition to providing voice communication from a source to a destination, the voice communication system can perform additional functions for example, validating destination numbers, maintaining user records and storing call details.

Abstract: The invention helps solve problems with technology that detects international calls. When an international call is detected a database is queried. Dependent on the results of the database query, present conditions based up on criteria, the call is either routed using international routing or the call is routed using an internet protocol (IP) gateway.

Abstract: The invention comprises a method and apparatus for processing a call request from a first enterprise customer network to a second enterprise customer network using at least one dialing plan. Specifically, the method comprises receiving at least one dialing plan request from a first network endpoint in the first enterprise customer network for accessing a second network endpoint in the second enterprise customer network, retrieving at least one dialing plan 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 dialing plan mapping.

Abstract: A method and apparatus for accessing communication features in a communication session between at least two communication devices is provided. A first communication path is established between the at least two communication devices via a first communication protocol, the first communication protocol associated with a first set of communication features. A second communication path is established between the at least two communication devices via a second communication protocol, the second communication protocol associated with a second set of communication features thereby giving at least one of the at least two communication devices access to the second set of communication features.

Abstract: The same number is redundantly set as the extension number associated to a fourth telephone number in different data tables associated to different second telephone numbers. For this reason, it is possible to maintain outstanding scalability of the system in a situation where the total number of settable numbers is limited. Further, even if an outgoing call is made to a caller side terminal from a lost receiver side terminal using the call register, the receiver number is recognized to be either an extension number associated to a fourth telephone number or an extension number associated to a fifth telephone number, and an instruction to disconnect a first call is sent to the call controller. As a result, the above-mentioned outgoing call (first call) from the lost receiver side terminal is disconnected, thereby making it possible to fully ensure security when a terminal is lost.

Abstract: The disclosure is directed to a customer support call system including an interactive voice response system configured to receive a customer support call and configured to determine a caller identification. The interactive voice response system is configured to determine whether the customer support call is associated with a computer system having an installed self-support tool. The customer support call system further includes a call center management system responsive to the interactive voice response system. The call center management system is configured to direct the customer support call in response to determining whether the customer support call is associated with the computer system having the installed self-support tool.

Abstract: Disclosed is a unified messaging system and method for combining a voice-mail system with an e-mail system. The present invention is a hybrid event driven system that polls a universal e-mail box for receipts, with the voice-mail system being fully event driven. The present invention maintains two persistent log-ins to the universal e-mail box, one to deliver messages and one to read message receipts. Thus, the system and method does not need to log-in and -out or read the entire list of messages. The present invention uses a file system instead of a state database. The system and method acts in real time such that messages in the unified messaging systems are synchronized within a pre-determined real-time. The system and method uses a “fail-safing” technique in which, after copies of messages are made and stored by each messaging component, allows one system to work if another is down.

Abstract: Switching points in current voice telephony networks (PSTN/ISDN) deal with the control of links and useful channels. Next generation networks which are designed for voice and data work according to the principle of separation in the control of links and useful channels. Different network elements are used to control links and useful channels. Link control is carried out by soft switches which communicate by means of a suitable protocol with media gateways controlling the useful channel. A plurality of services are implemented in existing networks, said services continuing to be needed. This results in the following problem: the new development or porting of the software controlling said services required for the soft switches is extremely heavy and very complex.

Abstract: A method and system for routing messages through a communications network based on the content of the message. The method of the present invention comprises providing a method for retrieving information through a telecommunications network comprising receiving, in a message data structure, a request message comprising a command; sending a query message to a content provider based on the command; receiving a query response message from the content provider comprising response information; and sending a request response message based on the response information.

Abstract: Service resource items for use in call setup in a telephone system are held on servers that are connected to a computer network which is logically distinct from the telephone system infrastructure; this computer network may, for example, make use of the Internet. Each service item is locatable on the network at a corresponding URI and is associated with a particular telephone number. A mapping is provided between telephone numbers and the URIs of associated service resource items. When it is desired to access a service resource item associated with a particular telephone number, this mapping is used to retrieve the corresponding URI which is then used to access the desired service resource item.

Abstract: A methodology for optimizing the access charges paid by a long distance carrier involves the use of a “sizing” operation to determine the optimal balance between direct trunks and access tandems (on a per LATA basis) so as to provide minimal access charges. Once the sizing has been completed, a mixed integer program is used to find the optimal MUX assignments for each LATA. The sizing operation is then re-visited, based on the MUX assignments, to determine if adjustments are required to reduce costs. Any changes in the sizing are then reviewed in the MUX assignment mixed integer program. The end result is a definition of the quantity of direct trunks, access tandem trunks and overflow minutes-of-use charges for each end office, as well as the number of high capacity facilities required for each MUX.

Abstract: An embodiment of the invention is directed to a method, wherein it is received, at an origin device, input from a first telephonic device via an origin telephonic landline. An initiation request is then output based on the input to a destination device, wherein the destination device is configured to output a call request to a second telephonic device via a destination telephonic landline. Another embodiment of the invention is directed to a method, including receiving, at a server, a request from an origin device, wherein the origin device is configured to receive input from a first telephonic device via an origin telephonic landline. Information is then output, based on the input, regarding a destination device to the origin device, wherein the destination device is configured to output a call request to a second telephonic device via a destination telephonic landline.

Abstract: An overlay network of volatile nodes, without centralized servers, using DHT-like search algorithms, is programmed to perform One-Number or One-ID services. A user's public number is either the PSTN or PLMN line that is connected to an overlay node, or a voice (or video) over IP number, which is serviced through the overlay. An overlay node is connected to the Internet, and may connect to a PSTN or PLMN phone line as well. The phone numbers allowed in the One-Number service is generally defined. The One-ID service architecture is based on user-proxy, and does not require social networking other one-ID protocols such as OpenID and protocols suites drafted by OASIS. The user-ID of such a One-ID service is not required to be globally unique while a mapping is used to transform a user-defined non-globally unique ID into a site-specific and site-unique ID.

Abstract: A menu is provided in a first telephone call. The menu provides a plurality of destination options including a first destination option for a residence of a family, a second destination option for a first member of the family at a first location other than the residence, and a third destination option for a second member of the family at a second location other than the residence. A selection of one of the destination options is received in the first telephone call. A second telephone call is initiated to a telephone number corresponding to the selection. The first telephone call and the second telephone call are connected.