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: There is provided a method of forwarding a traffic flow in a communications network having at least one network node for which network traffic is selectively prevented from transiting the network node. The method includes the step of selecting a specified category of network traffic that is to be prevented from transiting the network node. The method also includes the step of determining a path within the network for forwarding the traffic flow, whereby the network node is excluded for transit in establishing the path where the traffic flow is identified with the specified category of network traffic.

Abstract: An automated method for determining user satisfaction with a call comprising (1) maintaining a count of the number of times a user was asked for input during the call (the “actual count”), (2) maintaining a count of the number of times a user should have been asked for input based on the dialog path the user traversed (the “ideal count”), and (3) associating no difference between the actual count and the ideal count with maximum user satisfaction and associating increasing differences between the counts with decreasing user satisfaction. In one embodiment, if the difference between the actual count and ideal count reaches a certain threshold, the call system changes the interaction with the user.

Abstract: A communication system that is configured to communicate with more than one call manager systems, the communication system comprising a routing system configured to receive signaling for a call wherein the signaling indicates a called number including an international country code, and process the international country code to transfer a first query including the international country code, and the communication system further comprising a call processing system configured to receive the first query, process the international country code to select one of the call manager systems from the more than one call manager systems, transfer a second query to the one call manager system, and process a response from the one call manager system to transfer a routing instruction for the call to the routing system.

Abstract: A method and system for establishing a call between two endpoints residing on different media gateways in a decomposed voice over packet architecture. Cluster attributes identifying alternate bearer possibilities are exchanged between media gateways as part of capabilities negotiation which occurs during call setup conducted by an external call control element over a core network. The media gateways analyze the cluster attributes to determine whether bearer possibilities exist apart from the core network. If an alternate bearer possibility exists, the alternate is employed and the call is established via the alternate bearer directly between the media gateways, thus bypassing the core network.

Abstract: A method of signalling in an AIN, in which in an SSP, the initial message of a new transaction is addressed to a Service Global Title (SGT). The SSP translates that SGT to a first address of the first choice SCP. The SCP responds with an acknowledgement message which has a second address of the SCP as its source address, and includes the Nodal Global Title (NGT) of that SCP. The SSP sends subsequent messages for that transaction addressed to that NGT, and the SSP translates that NGT to the second address of the SCP. When the SCP is taken out of service, the first address is made unavailable, so the SSP sends new initial messages to the second choice SCP, but subsequent messages for existing transactions may still be sent to the second address of the first choice SCP. When all existing transactions have ceased, the SCP is taken out of service.

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: 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 method and system is presented to route calls to a voicemail system. The method includes receiving a call from an originating device at a first redirecting device and forwarding the call from the first redirecting device to a second redirecting device. The forwarded call has an associated data message that includes a calling number of the originating device and the first redirecting number of the first redirecting device. The method also includes forwarding the call from the second redirecting device to a destination voice mailbox. The destination voice mailbox is selected based on an evaluation of the first redirecting number and the second redirecting number.

Abstract: A forwarding telephone number in a programmable telephone number list associated with a called number is dialed in response to a busy or no answer signal at the called number. The programmable telephone number table may include preferred caller telephone numbers. The system and method may be may be implemented on an AIN based wire line system.

Abstract: A method of routing a telephone call includes receiving an indication, at a handset device 112, to initiate a telephone call. It is then determined, preferably by the handset device 112, whether the telephone call is an emergency services telephone call. If the telephone call is not an emergency services telephone call, it is routed to a proxy server 134. On the other hand, if the telephone call is an emergency services telephone call, it is routed to a local gateway 116 without first accessing the proxy server 134.

Abstract: A method and apparatus for ensuring that every path in an IP network that interconnects a pair of VoIP network elements comprises diverse network routes and is engineered with adequate bandwidth to support VoIP phone calls are disclosed. In the event that a single point of failure or inadequate bandwidth is detected between a pair of VoIP network elements, an alarm is issued and network engineers are notified to reconfigure the routes between the VoIP network element pair.

Abstract: At least one telephone network element receives an attempt by a caller to place a telephone call to a telephone number of a callee. The telephone number 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 determines a preference to carry the telephone call via a public switched telephone network (PSTN) or a public land mobile network (PLMN) instead of the VoIP network. Based on the preference, the telephone call is forwarded to a gateway to carry the telephone call to the callee via the PSTN or the PLMN instead of the VoIP network.

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 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: A method is presented for routing a call from a PBX to a voice mailbox on a communication platform wherein a call is received from an originating device at the PBX and the PBX is associated with a voice mail box on a communication platform. A redirecting number associated with the PBX is assigned to an ISDN message set-up string. The call is forwarded to the communication platform along with the ISDN set-up string information. After verification against a subscriber database, the forwarded call, including related ISDN information is delivered to the communication platform voice mail box server for storage and further processing. The redirecting number may represent the voice mail box identifier, and for increased security and privacy, the voice mail box may be different from the number for the PBX extension from which the voice and data messages are received.

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: A duplicate point code routing node is capable of routing messages to nodes having the same point code but being located in different national networks. The duplicate point code routing node associates a group code with received messages and uses the group code to route the messages to the appropriate national network. The group code may be determined based on the link or linkset from which a message is received or from the message itself.

Abstract: A method of detecting misrouted inter-office transport facility routes in a telecommunications system is disclosed. The method includes receiving a plurality of provisioning orders corresponding to a plurality of newly requested inter-office facility routes to be added to the telecommunications system, and comparing the plurality of newly requested inter-office facility routes to a plurality of preferred routes that have been stored in a preferred routes database to identify a first set of routes that match the preferred routes, a second set of misrouted routes where a preferred route has been defined and previously stored, but where the newly requested route does not match the preferred route, and a third set of unmatched routes where a preferred route has not been previously stored in the preferred routes database.

Abstract: A method, system, and medium for determining a least-cost, international-routing path to communicate data across a telecommunications network are provided. The path implementation variables can be presented on a user interface or used to automatically generate a switch update transaction that implements the path. The method includes identifying variables that affect the determination of a least-cost path, receiving data values that correspond to the variables, and presenting the data values so that the path can be determined in response to observing the variables. Confidence and route testing can be conducted using the present invention.

Abstract: A method and apparatus for enabling a network provider to download routing profiles into an intelligent Customer Premise Equipment (CPE) device to enable the CPE to determine the best provider to route a particular call to are disclosed. The CPE determines the best provider based on criteria specified by the subscriber and the network provider and then sends the call to a specific IP address, that has been previously been downloaded and configured in the CPE, to reach a selected network provider.

Abstract: A communication system includes a communication network, and an originating telephony device, a target telephony device, and a call manager coupled to the communication network. The call manager signals the target telephony device to indicate an incoming call from the originating telephony device. The call manager includes an alternate number list that identifies at least one alternate telephony device to be signaled in response to the incoming call directed to the target telephony device. The call manager additionally includes one or more mapping tables associating a packet network address with each identified alternate telephony device.

Abstract: A dial plan for a gateway in a packet based voice system provides call re-direct functionality to the system. The dial plan allows a gateway to process all incoming and outgoing calls for an enterprise, thereby minimizing required interface with an existing PBX. The dial plan identifies a given call as an incoming call from a data network, an incoming call from the PSTN, or an outgoing call from the PBX at an enterprise. The dial plan further identifies a desired route for transmitting the call to a desired destination. If the desired route is over a data network, the dial plan tests the route to determine its availability, and if it is not available determines an alternate re-direct route over the PSTN.

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: In an example embodiment, a system that appropriately routes calls to an agent at a contact center based on the agent's voice and/or hearing characteristics. The agent is selected by matching speech and hearing characteristics of a caller with the speech and hearing characteristics of an agent. In order to find the best match for the caller, the contact center determines if the caller is hearing impaired, and if so determines a suitable frequency range for the caller. If a match cannot be found, the agent's and/or caller's voice may be shifted in real time and adjusted to a frequency range that is best suited for the caller.

Abstract: A request to delete a dialing plan is received. A list is generated of switching modules that are connected to a central processor of a telephone switch. Each switching module is queried to determine if the switching module includes a copy of the dialing plan. It is determined whether the dialing plan is in use by querying switching modules that include a copy of the dialing plan. If the dialing plan is not in use, then the dialing plan is deleted from at least one switching module identified as including a copy of the dialing plan.

Abstract: A telephony interface system has a controller and multiple communication interfaces to provide voice-over-packet (VoP) telephone service with PSTN telephone service as backup. The telephony interface system has a communication interface for at least one telephony device, a communication interface for the PSTN, and a communication interface for a packet-switched network. To originate a call, the telephony interface system connects the telephony device to the PSTN if the dialed digits for the call correspond to a predetermined telephone number or numbers, such as 911. Otherwise, the telephone interface system connects the telephony device to the packet-switched network.

Abstract: A system, wherein a mobile station can switch to an IP access protocol, whereupon data transmitted between the mobile station and a mobile services switching centre change over from a base station system, reached using radio communication, to being transmitted through an interworking unit (40) coupled in connection with an IP connection and the mobile services switching centre. When the mobile station switches to the access network, it transfers the dynamic data relating to the mobile station's state and the calls in progress (state machine) to the interworking unit.

Abstract: The present invention provides a way to associate multimedia clients with telephony devices and create multimedia sessions related to a voice connection between the telephony devices.

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 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: The present invention is directed to a method and apparatus for accessing a packet node or device through a circuit-switched network. Circuit-switched technology, such as local number portability techniques, and packet-switching technology, such as soft switches, are integrated in a network. As such, calls may originate from a circuit-switched network and terminate in a packet-switched network.

Abstract: The present invention provides a method and system for maintaining and updating records of the routing of switches in a telecommunication network. Routing information is retrieved from a routing guide, and corresponding routing information is retrieved from switches in the telecommunication network. Routing information is retrieved from switches by using the routing information from the routing guide to iteratively retrieve routing information from switches by NPA code, switch, rate center, and NXX codes. Routing information from the routing guide is compared to routing information retrieved from the switches and discrepancies in the routing information is identified. Identified discrepancies are corrected in either the routing guide or the routing of the switch.

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: The system automatically generates electronic commands necessary to modify or provision different types of program control switches as may be included in a public switch telephone network. A data schema and organization of tables and data structures provide for command generation for each of the different switch types. Additionally, a self-modifying dynamic data structure is used as a common area to communicate information between various programs and formulate the commands due to a limited number of parameters used to communicate information between programs that generate the command for the switches. A technique for verifying switch commands automatically generated by a second new provisioning system is described. A technique is described for automatically extracting switch specific commands and options as generated by a first system for use in populating data stores used in a second system for command generation.

Abstract: Emergency calls placed from telephone devices connected to a data network are completed to the appropriate public safety answering point (PSAP). In one embodiment of the present invention, a method for completing an emergency call comprises receiving a call from a device on a data network, the call destined for an emergency service, the device having a physical location, obtaining a telephone number of an emergency call center that handles emergency calls for the physical location of the device, and routing the call to the emergency call center using the telephone number.

Abstract: A method and system enable optimization of trunk group design in a public switched telephone network (PSTN). A community of interest, including two end offices connected by a direct trunk and at least one tandem switch, is identified using out-of band signaling data collected from the PSTN. An application server interfaced to the PSTN determines whether traffic in the community of interest passes through the tandem switch during a predetermined time period. When traffic passes through the tandem switch, the application server further determines whether the direct trunk between the end offices experienced an overflow condition during the same predetermined time period. When the direct trunk did not experience an overflow condition, an exchange code associated with the traffic passing through the tandem switch is designated as a misrouted code. The misrouted code is flagged so that associated traffic is redirected through the direct trunk.

Abstract: A system and method is provided for routing telephone calls on a standard telephone network to one of a PSTN network and a VoIP network. The system comprises at least one telephone capable of making local and long distance calls on the network. A Central Office is coupled to the telephone and comprises a trigger for identifying calls to be routed to either a VoIP platform or the PSTN network. Calls to be converted to IP protocol are routed to a VoIP platform. An IP network is coupled to the VoIP platform and routes VoIP calls to one or more devices in communication with the telecommunications system.

Abstract: A telecommunications system is disclosed that enables a telecommunications switch to be “user” programmable in the “field”—in contrast to only programmable by the manufacturer or third-party software company—with audio samples. Each audio sample is exemplar of a sound that might be encountered during or after the establishment of a telecommunications call that is handled by the telecommunications switch. In accordance with the illustrative embodiment, the audio samples are received by the telecommunications switch through one or more of the same ports that the telecommunications switch uses to route calls, in contrast to programs and program data that are received by the telecommunications switch through a programming port.

Abstract: A method for routing an international call includes receiving a call and identifying a first international carrier based on an Internet protocol address associated with the call or a tunnel through which the call was received. The method also includes identifying a second international carrier based on the identified first international carrier. The second international carrier may represent a terminating carrier associated with forwarding the call to a called party. The method may further include forwarding the call to the identified second international carrier.

Abstract: A method of routing calls through a network (5) includes: receiving an incoming call at a signaling transfer point (STP); transferring information related to the incoming call to a routing engine by decoding ISUP signaling information related to the incoming call; determining an optimized route based at least in part on the transferred information; and signaling the STP to a destination number based on the optimized route.

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: 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 delivering digital subscriber line (DSL) service to a subscriber. In a telecommunications network, the signal passing through a protector field is diverted to a cross connect block that can be selectively configured to establish a communication route between the cross connect block and a splitter selectively receiving a DSL signal. The combined signal is then placed back upon the network by the cross connect block, which directs the combined signal back to the protector field.

Abstract: In a telephony call management system, a method of distributing calls among a plurality of packet-based load balancing resources can include receiving over a telephony channel a call request specifying call information, comparing the call information with selection criteria to select one of the plurality of packet-based load balancing resources, and querying the selected packet-based load balancing resource to resolve the call request. A response from the selected packet-based load balancing resource can be received which indicates whether the call request can be resolved.

Abstract: A method and system for providing computer-network related information about a calling party. First, the called party receives a telephone number of a calling party. The called party's customer premises equipment (CPE) or elements in a telephone network then use the telephone number to index a database, which contains combinations of telephone numbers and computer-network addresses. Once the calling party's computer-network address is retrieved, calling-party-customized information present at the computer-network location specified by the computer-network address can be sent to and displayed on the called party's CPE. Also, the calling party's telephone number can be sent to an application in the computer network, causing the calling-party-customized information to be automatically displayed on the called party's CPE.

Abstract: A method of operating a telecommunications network includes, in one embodiment, detecting a call at an originating switch of the telecommunications network and, when a destination for the call matches a predetermined destination, routing the call on designated data trunks. When traffic on the designated data trunks exceeds a capacity threshold, the method further includes routing the call to a terminating switch using interoffice trunks and completing the call to the destination from the terminating switch.

Abstract: A method and system for routing service calls made from lines resold to a competitive service provider includes advanced intelligent network and standard switching functionality. The call may be made into an AIN service switching point (60) or a non-AIN switch (62-64). All calls are routed to an AIN hub (190), such as a service switching point, via a trunk group. The AIN or non-AIN switch may access a line class code table (160-164) to determine the proper routing information. Calls made from resold lines are routed to the AIN hub (190). Once the call is routed to the AIN hub, a trigger (187) provisioned on the AIN hub and assigned to the incoming trunk group causes the hub to suspend the call and launch a query to a service control point (200). The query includes the directory number of the resold calling line and the originally dialed number. The SCP (670) accesses a data-base (675) containing directory numbers for resold lines and identifiers for competitive service providers.

Abstract: The present invention provides methods, apparatuses, and computer media for re-routing outgoing calls from a communications terminal. A call routing module selects a set of re-routing plug-ins. One of plug-ins determines re-routing information for an outgoing call based on call information in accordance with a priority list. A priority order for offering a call event to a re-routing plug-in is determined from at least preference list. A preference list may be configured by a subscriber through a user interface. Also, preference lists and re-routing plug-ins may be downloaded over a communications channels. Another re-routing plug-in is offered a call event if a first re-routing plug-in is not able to re-route an outgoing call. One of a plurality of communications systems is selected for re-routing an outgoing call. A communications systems supporting a close-proximity technology may be selected when a communications terminal is sufficiently near the communications system.

Abstract: A system and method is provided for call routing from a wired telephone switch. A wireless device is registered with a database of the wired telephone switch. Incoming wireless calls destined for the wireless device are routed to a standard telephone. The wireless device is unregistered to disable the call routing to the standard telephone.