Abstract: Disclosed is a method and system for determining a telecommunications address of a location. A telephone call from a caller located at the location is received. Location data associated with the received telephone call is then obtained. The telecommunications address can then be determined from the location data.

Abstract: A call center maintains various pools of telephone numbers, where each telephone number pool comprises a plurality of telephone numbers in a geographic area. A call originating to a called number uses a calling telephone number that is selected from one of a plurality of telephone number pools based on the area designated by the called number. An index value is used to select the telephone number from the selected telephone number pool. The selected telephone number is used to as a calling telephone number when originating a call to the called party.

Abstract: An examples of active call filtering, screening and dispatching is described, including receiving data representing a call from a first communication endpoint; presenting at a second communication endpoint, prior to connecting to the call, a subset of prompts that specify dispatching actions for the call, at least one of the dispatching actions including transmitting a response via a communications network, detecting selection of a prompt that specifies a dispatching action for the call, and transmitting an instruction from the second communication endpoint to a call messaging system to perform the dispatching action. In some embodiments, a determination is made as to whether a bypass condition exists, the bypass condition specifying that one or more customized screening rules are to be bypassed. If the bypass condition exists, then a call messaging system can bypass the use of a subset of prompts.

Abstract: In an example embodiment, a method is provided that receives a broadcast of available bandwidth from a first routing device. A congestion of traffic is detected along a downstream path to a second routing device. This second routing device is an immediate downstream neighbor. As such, an alternate path is established to the second routing device by way of the first routing device based on the available bandwidth in the network and a portion of the traffic is transmitted along the alternate path.

Abstract: Methods, systems, and apparatus for populating a least cost routing database. In one aspect, a method includes receiving from a carrier a bid for routing a telephone call to a destination; automatically loading the received bid into a least cost routing database to generate an updated least cost routing database; and making the updated least cost routing database accessible for use in determining one or more least cost routes. Further, the received bid can be analyzed to generate a projected business result and the projected business result can be presented to the carrier before loading the received bid into the least cost routing database. Additionally, input can be received from the carrier specifying an historical period and the projected business result can be generated based on data associated with the specified historical period.

Abstract: A communication routing arrangement includes two or more wireless routers coupled to a communication network, and a user device adapted to couple with the communication network through a communication path including at least one of the two or more wireless routers. The communication path between the user device and the communication network is selected based on available resources of the two or more wireless routers.

Abstract: The invention relates to a data communications network and a method of operation thereof. A data communications network (14) is described which includes a call server (58, 60, 62, 64) for the purpose of signalling, and a routing server (66, 68) for the purpose of routing to permit communication between users of the network. The call server (58, 60, 62, 64) is operable to send a request to the routing server (66, 68) for the purpose of obtaining the identity of another call server (58, 60, 62, 64) to which it is required to communicate. The request is provided with a parameter which relates to the origin of the communication within the data network (14) to enable a desired routing of the communication through the network.

Abstract: Registering with a call manager includes connecting to a first call manager and a second call manager. The endpoints register with the first call manager and send keep-alive signals to the first call manager and the second call manager. If a failure is detected in the first call manager, the system initiates a failover process. The endpoints re-register with the first call manager when the first call manager regains operability.

Abstract: Interpersonal communications initiated from devices such as various phones and voice/chat devices automatically transport via an optimal path by invoking a service selector of communication mediums for selecting at least one of several alternate routing paths for user communications from among available communication mediums common to both an initiator and recipient. A user registration database identifies communication mediums applicable to each user, and identifies or maps an appropriate communication medium between users for completing a particular call. The service selector identifies and intercepts an outgoing communication attempt (call, email, text) from a user, identifies the user and recipient in the database, and selects an optimal communication medium defining an alternate routing path from the initially specified (i.e. default) transport mechanism.

Abstract: A signaling node within a telecommunications network automatically detects that a signaling link is fluctuating in and out of service and provides hysteresis to the fluctuating signaling link. The signaling node includes a link controller that monitors the state(s) of the signaling link over a time period and a link blocking module that blocks the signaling link from carrying SS7 traffic when the signaling link fluctuates between a failed state and a stable state over the time period.

Abstract: A telephone call management system first receives a call over a telephone network from a calling device. The calling device is connected to a first called device by a first connection through the telephone network. A first call management message is received at a call routing engine, the first call management message to cause the engine to initiate establishment of a second connection among the calling device, the first called device, and a second called device. The engine issues, in response to the first call management message, a second call management message specifying a DTMF sequence for provision to the telephone network to cause the telephone network to establish the second connection. The telephone network may be a public switched telephone network.

Abstract: A method and apparatus are provided for transferring contacts among automatic contact distributors of an automatic contact distribution system. The method includes the steps of collecting information from a contact through an interactive voice response unit of a broker automatic call distributor of the automatic contact distribution system, incorporating the collected information into a SIP message, forking the SIP message from the broker automatic contact distributor to a plurality of other automatic contact distributors of the automatic contact distribution system and transferring the contact from the broker automatic contact distributors to one of the plurality of automatic contact distributors.

Abstract: In one embodiment, a network device attaches a header including an identifier for an original call to an outgoing refer message and may also modify a table entry for a consultative call to include the identifier. The network device then monitors incoming invite messages for a header containing the identifier and observes that an incoming invite message containing the header correlates to the original call. When a header is not included, the network device extracts information from the incoming invite messages and compares the extracted information to the modified table entry to correlate one of the incoming invite messages to the original call.

Abstract: A system is provided for use by a caller placing a call with a calling device over a communications network linking a plurality of call destinations. The system includes a call-destination information source containing one or more call-destination characteristics associated one or more call destinations. The system further includes a processing unit communicatively linked to the call-destination information source. The processing unit generates one or more call-destination character indicators based on the one or more call-destination characteristics. A call-destination character indicator is generated in response to the system receiving call-destination identifier that identifies a particular call destination.

Abstract: A call manager includes a first device process that controls a first telephony device and that receives a call request from the first telephony device. The call request includes a telephone number associated with a second telephony device. The call manager also includes a call control module that receives the call request from the first device process, and the call manager further includes a route list control process that is associated with the telephone number and that receives the call request from the call control module. The route list process accesses a route list to determine a port of a gateway device that can transmit the call request to the second telephony device. The route list process communicates the call request to a second call manager that is coupled to the packet-based network and that controls the gateway device included in the route list.

Abstract: A system and method for routing an incoming call to a subscriber-selected destination number in accordance with dynamic data concerning the subscriber provided by an address book, a calendar and a presence server. A routing system routes the incoming call in accordance with a subscriber-defined routing rule associated with the originating number of the incoming call, as determined from the address book. The routing rule may specify that the incoming call be routed to a destination number associated with the current date and time, as indicated by the calendar. The routing rule may specify, alternatively or in addition thereto, that the incoming call be routed to a destination number associated with a presence-enabled service on which the subscriber is currently active. The subscriber or caller may also be alternatively notified of the routing of the incoming call.

Abstract: A method provides call pickup in a communications network. The method includes initiating a call from a first device to a second device. The call is initiated over one or more networks, where at least one of the one or more networks includes a data network. The method further includes storing information relating to the call initiation between the first device and the second device, receiving a message from a third device during the call initiation, where the message includes a call pickup indication, retrieving the information relating to the call initiation between the first device and the second device, and establishing a call between the first device and the third device based on the retrieved information.

Abstract: An apparatus for switching a user between an IP network and a PSTN is disclosed in the present invention and includes a user port, a Subscriber Line Interface Circuit, a detection circuit, a PSTN port, an IP network interface circuit and an action unit. A method for switching a user between the IP network and the PSTN is also disclosed. According to the method, the action unit is controlled by the control signal from a CPU, thereby implementing the user switching between the IP network and the PSTN on the apparatus. The apparatus and the method enable one or more users to switch between the IP network and the PSTN, implements flexible user selection between the two networks and offers user communication backup when failure occurs. Accordingly, the present invention increases the satisfaction of users and the core competence of enterprises.

Abstract: The present disclosure is directed to updating routing patterns in an enterprise network. In some implementations, a method for routing sessions includes identifying information associated with call sessions in an enterprise network. The information includes costs to an enterprise for using carrier services during the call sessions. Routing patterns for call sessions are dynamically updated through at least a portion of the enterprise network based, at least in part, on the costs associated with the call sessions.

Abstract: A method of routing calls in a communication system. A call is received from an originating network in the communication system. The identity of the originating network is determined, and the call is routed in the communication system based on the identity of the originating network.

Abstract: A method of directing a call includes receiving a call connection request, the call connection request including an identifier associated with a called party. The method also includes determining a network category associated with the called party in response to receiving the call connection request and generating an instruction operable to direct the call connection request to a selected one of at least two networks accessible for communication by the called party. The network is selected by the instruction in response to the determined network category associated with the called party.

Abstract: At least one telephone network element receives an attempt of a caller to place a telephone call to a callee. A telephone number for the call is registered in a telephone number mapping (ENUM) database to receive incoming telephone calls via a Voice over Internet Protocol (VoIP) network. The at least one telephone network element receives an indication of a preference of the caller to carry the telephone call via one of a public switched telephone network (PSTN) and a public land mobile network (PLMN), instead of via the VoIP network. Based on the preference, the method inhibits an ENUM query of the ENUM database for the telephone call and forwards the telephone call to a gateway to carry the telephone call to the callee via one of the PSTN and the PLMN instead of via the VoIP network.

Abstract: A technique is disclosed that optimizes the background noise experienced by a party who is calling into a call center. Working as part of an overall call-assignment algorithm, the technique considers the acoustic noise that is present in the vicinities of multiple call-center agents who are otherwise satisfactory candidates to handle a call. The technique then selects an agent to handle the call who is associated with an optimal acoustic noise. Typically, the selected agent is associated with lowest background noise level. The background noise is monitored at each call agent's station by evaluating the signals that are present at the agent's microphone. Usually, this is done when a call agent is between calls and, as a result, is not using her headset at that moment. In other words, the background noise is actually measured, and the measurements are then used to assign a call agent to the incoming call.

Abstract: A communication system is provided which includes a digital communication network having a network endpoint and a signaling server. The network endpoint is coupled with the signaling server via signaling tunnels through the digital communication network. Also included is a telephonic device coupled with the network endpoint and configured to engage in a telephone call via the network endpoint over the digital communication network. The network endpoint is configured to generate signaling related to the telephone call, and to direct the signaling through the digital communication network to the signaling server over a selected one of the signaling tunnels, wherein the selected signaling tunnel is selected based on an identity of the telephonic device.

Abstract: A method and system is provided wherein routing information is obtained from previously cached information or from a gatekeeper. An originating gateway in a network receives a request for a connection and determines whether pertinent routing information is cached. If available, the cached routing is used to attempt to establish a connection. If no previously established routing is available, the originating gateway requests preferred routing from a gatekeeper, including prioritized selections of terminating gateways through which to make a connection to a designated call recipient.

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: A method and apparatus for enabling communication of IP multimedia content to a called user during a circuit-switched call between a calling terminal (A) having called a primary communication unit (B1), when the call has been forwarded to a secondary communication unit (B2). The calling terminal receives a connected number during a call set-up procedure, e.g. by subscribing to a called line presentation (COLP) service. The calling terminal stores the received connected number and determines multimedia capabilities of the secondary communication unit based on the connected number. Available multimedia options can then be determined based on the multimedia capabilities of the secondary communication unit.

Abstract: One preferred embodiment of the present invention provides a system and method for evaluating the performance of a network interface sub-system of an inter-messaging network of voice mail systems. This preferred embodiment includes a network query device that connects to the inter-messaging network and requests a test data file to be retrieved from a particular voice mail network interface sub-system in the inter-messaging network. The requests for the test data file are generated according to user command. Accordingly, the performance of a network interface sub-system in the inter-messaging network, as represented by the result of the request attempt, is evaluated according to a defined level of performance, such as a preferred time limit. Other systems and methods are also provided.

Abstract: For use with a path vector routing protocol, a system and method for increasing alternative network route convergence speed and a border gateway router incorporating the system or the method. In one embodiment, the system includes: (1) a router information base (RIB) data receiver, associated with a network, that receives RIB data including: (1a) an update of an active route to a domain in the network that causes the active route to become a withdrawn route and (1b) indications, based on loopback addresses associated with nodes through which the withdrawn route passed, of a reachability of the each of the nodes and (2) route disqualification logic, associated with the RIB data receiver, that disqualifies alternative routes to the domain based on the indications.

Abstract: A communication network that includes a signaling system seven (SS7) network is operated by associating call release messages, which are indicative of call failures in the communication network, with an originating point code (OPC) and/or a calling number. The rate at which calls originating from the originating point code and/or the calling number are admitted to the network may be adjusted based on the association between the call release messages and the originating point code and/or the calling number.

Abstract: Multiple instances of an incoming call in a unified communication system capable of communicating through multiple networks are provided to devices and/or applications associated with the called party. Upon acceptance or a global rejection of the incoming call through one of the end devices/applications, the remainder are provided a cancellation notice for the forked call request that includes information associated with which end device took the action, what type of action was taken, whether the action was based on automatic policy implementation.

Abstract: A computer for processing a call originated by a call originator to a call target. The computer includes memory having at least one region for storing computer executable program code and processor for executing the program code stored in the memory. The program code includes code for processing a telephone number of the call target, code for accessing a database storing a plurality of outgoing telephone numbers, code for identifying one of the plurality of outgoing telephone numbers based on the telephone number of the call target, and code for originating the outbound call to the call target and including the identified outgoing telephone number into a caller ID field of the outbound call.

Abstract: A signal transfer point (STP) within a first telecommunications network enables recovery from a linkset failure between the first telecommunications network and a second telecommunications network by deactivating one or more affected signaling routes between the first and second telecommunications networks that include the failed linkset. The STP includes a linkset controller for monitoring the current state of the linkset coupled between the STP and an additional STP within the second telecommunications network. The STP further includes an input interface for receiving a deactivate common channel signaling (CCS) route command from a user when the current state of the linkset indicates failure of the linkset, and a signaling route control module that changes the route status for the affected signaling routes to “prohibited” to prevent SS7 traffic associated with the affected signaling routes from being sent over the linkset.

Abstract: A radio network controller and methods for reestablishing sessions in a wireless network are described. At least a portion of session information associated with a first session is saved; and in response to detecting an unexpected degradation of the first session, reestablishment of the first session is triggered using the portion of the session information.

Abstract: In a particular embodiment, a system includes a messaging system configured to provide a messaging service. The messaging system includes a centralized Internet Protocol switching network, an Internet access subsystem, a voice access subsystem, a mail subsystem and a storage subsystem. One or more of the subsystems may be coupled to the centralized Internet Protocol switching network via one or more communication links integrated with and internal to the messaging system.

Abstract: A method and system are disclosed where a call center operator having multiple locations, including a call center is able to utilize existing network infrastructure to route telephone communications to the call center to the location where the lowest toll rate is applicable. The telephone communications are converted into data packets and routed to the call center using VoIP technology. No toll would apply to the transmission of the data to the call center.

Abstract: A call manager uses a call management application in conjunction with a live dial database to control routing of calls for managed devices. To generate the live dial database, the call management application accesses configured route patterns and enters these patterns into the live dial database. Upon identifying an expansion indicator in a configured route pattern, the call management application accesses dial plan data that includes multiple route pattern definitions that each define a pattern using one or more sub-strings and, for each sub-string, an associated tag. The call management application then enters patterns defined by the route pattern definitions into the live dial database based on various other criteria established for the configured route pattern having the expansion indicator.

Abstract: A VoIP adapter for POTS a phone comprises: a POTS phone connector, an IP network interface, two sets of signaling senders, signaling receivers, media senders and media receivers for the POTS phone and the IP network respectively, and a controller for controlling the operations of above components. The VoIP adapter enables the user to carry out VoIP communications using a normal POTS phone and further enables use of advanced VoIP functions via the normal POTS phone, such as Call Hold, Call Transfer, Ad Hoc Conference, etc.

Abstract: An interface for use in a telephone communication system provides for connection of a telephone switch to an IP network, allowing data signals between the telephone switch and a VMS to be transmitted by the IP network. In addition to transmission of the signals over an IP network, the interface can be configured to provide additional functionality including translation of the message protocol; connection of the interface to a network operations center; verification of message accuracy and completeness; acknowledgment of the receipt of a message; activation/deactivation of a message waiting indicator; connection to an SS7 network; back-up transmission between the IP network and the switch; and proxy/redirect of voice signals.

Abstract: An internet telephony system (ITS) comprising a browser display having a hyperlink, communicating with a server, retrieving a user's preferences, and initiating a voice communication with the user, through a communications channel defined by the user preferences. An ITS comprising a client system having an Internet browser, and a server hosting a Web site, wherein a message is transmitted from the server to the client system based on the user's status with respect to Web site, the user seeking to establish a voice communication session. A telephony server, comprising an application program communicating directly with telephony hardware, and an application programming interface, wherein the application program includes a call function to an external program. An ITS, comprising a browser display having a hyperlink, the hyperlink communicating with a server, retrieving a user-related data, and initiating a voice communication with the user, wherein the user is charged for the communication.

Abstract: A call management system enables each of multiple prospective callers to specify how to forward (e.g., direct or redirect) future calls to one or more target phone devices or users. For example, each of multiple persons provide call forwarding configuration information to the call management system. Upon initiation of a call by a user, the call management system intercepts and forwards a user's call in accordance with call forwarding configuration information previously provided by the user.

Abstract: A method includes receiving a forwarded call. The forwarded call corresponds to a call forwarded from a first redirecting telephone device to a second redirecting telephone device. The forwarded call has an associated data message that includes a calling number of an originating telephone device and a first redirecting number of the first redirecting telephone device. The method also includes selecting a destination voice mailbox. The destination voice mailbox is selected from a plurality of voice mailboxes associated with a user. The destination voice mailbox is selected based on an evaluation of the first redirecting number and a second redirecting number of the second redirecting telephone device.

Abstract: Systems and methods are disclosed for implementing a split numbering plan area code in an IMS network. The IMS network comprises a subscriber server adapted to store a new phone number including a new area code for a subscriber assigned as a result of the split numbering plan area code. The IMS network further comprises a call control system coupled to the subscriber server adapted to register user equipment of the subscriber assigned the new phone number with the subscriber server. The call control system is further adapted to associate an old phone number of the subscriber used prior to the split numbering plan area code with the new phone number in an implicit registration set. The call control system is further adapted to receive a call for the old phone number and identify the new phone number associated with the old phone number in the implicit registration set and to deliver the call to the user equipment.

Abstract: The communication terminal device includes a handset, a central controller, a telephone network adaptor unit for converting analog signals transmitted/received between telecommunication network and the communication terminal device into digital signals, an interface unit physically isolating the central controller from the telephone network adaptor unit, and transmitting/receiving the converted digital signals between the central controller and the telephone network adaptor unit, and a telephone circuit unit which transmits the voice signals generated by processing the digital signals received from the telephone network adaptor unit to a handset through the interface unit and transmits the digital signals generated by processing the voice signals received from the handset to the telephone network adaptor unit through the interface unit.

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: The communication terminal device includes a handset, a central controller, a telephone network adaptor unit for converting analog signals transmitted/received between telecommunication network and the communication terminal device into digital signals, an interface unit physically isolating the central controller from the telephone network adaptor unit, and transmitting/receiving the converted digital signals between the central controller and the telephone network adaptor unit, and a telephone circuit unit which transmits the voice signals generated by processing the digital signals received from the telephone network adaptor unit to a handset through the interface unit and transmits the digital signals generated by processing the voice signals received from the handset to the telephone network adaptor unit through the interface unit.

Abstract: A system and method that establishes a phone call over a packet network. The process may include receiving a call request from an originating call device on a packet network to call a terminating call device. A first transmission path may be determined between the originating and terminating call devices. Status of a terminating segment at a first terminating trunk may be determined and, if the status of the terminating segment at the first terminating trunk is within a first range, a call between the originating and terminating call device may be established via the first terminating call trunk. Otherwise, if the status of the terminating segment at the first terminating trunk is within a second range, a second transmission path via a second terminating trunk between the originating and terminating call devices may be made.

Abstract: When a call offer is received by a telecommunications peering network destined for a telephone number maintained within a peering database, the telecommunications peering network attempts to route the call directly to the peering partner telecommunications (e.g., broadband, voice over internet protocol, etc.) equipment. If the route fails with a corresponding destination unreachable failure code, the telecommunications peering network can attempt to determine whether or not the telephone number has been ported to a different provider. If the telecommunications peering network determines that the telephone number has been ported, the telecommunications peering network removes the telephone number from the peering database.

Abstract: Independent telephony systems are integrated by using symmetric forking from both systems. Calls received by either system are forked to the other system. Thus, a call can be accepted by one of the systems, using its external interfaces, and presented to all devices in both systems. Each system is enabled to leverage internal call control advanced features to enhance overall user experience. Internally inserted parameters to call requests are used between the two systems for preventing endless loop of calls between the integrated systems.

Abstract: The present invention is a telecommunications system having separate switch fabric and switch intelligence. The system comprises a switch fabric, a switch intelligence, and a feature processor. The switch intelligence is logically separated from the switch fabric and comprises a switch fabric proxy, a facility service, a connection manager service, and a call segment instance service. The switch fabric proxy is coupled to the switch fabric via a vendor-specific first Application Programming Interface (API). The switch fabric proxy supports a second API, which is common across all vendors, representing functions supported by the switch fabric. A facility instance, which is instantiated by a facility service using a facility model, represents the bearer and signaling facilities of a party to a call, and interacts with the switch fabric proxy via the second API to communicate with the switch fabric.