The present invention suppresses the number of connections from another network, and reduces the load on call processing apparatuses in a local network. An ENUM server 30 includes: a congestion detection unit 322 that collects traffic information regarding calls that include call destination numbers from call processing apparatuses in the local network, calculates a call amount per unit time for each call destination number, and stores the call amounts per unit time as calculated call amount information 331, and also extracts a call destination number included in an ENUM query received from the other network, references the calculated call amount information 331, and determines that calls to the call destination number are in a congested state if the call amount regarding the call destination number exceeds a predetermined threshold value; a limit processing unit 323 that, if it was determined that the call destination number included in the ENUM query is in the congested state, determine whether or not the ENUM query is to be a limit target based on predetermined limit processing; and a signal response unit 325 that, if it was determined that the ENUM query is a limit target, generates an ENUM answer that indicates connection unavailability, and transmits the ENUM answer to the other network.

Systems and methods for preventing a robocall from leaving a voicemail are described. When an incoming call without a history information header (HIH) destined for a subscriber is detected, the incoming call is prevented from leaving a voicemail if the subscriber does not answer the incoming call. If the incoming call has the HIH, additional determination process based on a visited network address of the subscriber is performed to determine whether the incoming call is to be prevented from leaving a voicemail.

A system and method for integrating an existing telephony system of a business with various other existing communication equipment in a unique sequence and using call servers to facilitate voice calls to be made over data networks between businesses and customers for the purpose of low cost customer service provision. The system and method comprises a business connecting voice calls over data networks to customers registered in a dedicated directory using a digital to analog converter to convert the existing telephony system format of the business into digital format to transmit digital packets to the call server which will function as a telephone switch and connect the call to the customer and customers making voice calls over data networks to the business using the directory. Customers use a computer or mobile application which transmits digital packets to the call server which will connect the call through the converter, converting the format from digital to analog, to existing telephony system.

A system and method for integrating an existing telephony system of a business with various other existing communication equipment in a unique sequence and using call servers to facilitate voice calls to be made over data networks between businesses and customers for the purpose of low cost customer service provision. The system and method comprises a business connecting voice calls over data networks to customers registered in a dedicated directory using a digital to analog converter to convert the existing telephony system format of the business into digital format to transmit digital packets to the call server which will function as a telephone switch and connect the call to the customer and customers making voice calls over data networks to the business using the directory. Customers use a computer or mobile application which transmits digital packets to the call server which will connect the call through the converter, converting the format from digital to analog, to existing telephony system.

This document discloses techniques, apparatuses, and systems for inter-service communication management to manage inter-service communication between a server-side service and a client-side service. The inter-service management system may comprise memory and a controller configured to perform the inter-service communication management. In aspects, a configuration file may be used to obtain information specifying a first inter-service communication mechanism for communicating one or more messages between the client-side and server-side services. The inter-service communication management system may instantiate a stub associated with the inter-service communication mechanism information, which can be used to communicate a message from the server-side service to a proxy that interacts with the client-side service. The message may then be forwarded from the proxy to the client-side service to enable inter-service communication between the server-side service and the client-side service.

This document discloses techniques, apparatuses, and systems for inter-service communication management to manage inter-service communication between a server-side service and a client-side service. The inter-service management system may comprise memory and a controller configured to perform the inter-service communication management. In aspects, a configuration file may be used to obtain information specifying a first inter-service communication mechanism for communicating one or more messages between the client-side and server-side services. The inter-service communication management system may instantiate a stub associated with the inter-service communication mechanism information, which can be used to communicate a message from the server-side service to a proxy that interacts with the client-side service. The message may then be forwarded from the proxy to the client-side service to enable inter-service communication between the server-side service and the client-side service.

Verification of the identities of parties participating in network-based communication, such as telephone communication, including SMS/text communication, email communication and the like is provided. Communication identifiers (IDs) (e.g., telephone numbers, email addresses or the like) are verified as being associated with one or more communication parties and, in response, a verified communication (ID) database is established. The verified communication ID database is relied upon when a user/communication is selecting a communication ID as an address for an impending communication or receiving a communication for determining whether the communication ID is verifiably associated with a known communication party. If the communication ID is determined to be verifiably associated with a known communication party, a visual or audible output may be provided on the user's device or within the communication that indicates that the communication ID is verifiably associated with the known communication party.

A failure monitoring device and a failure monitoring method that reduce maintenance operations at the time of a failure of an SIP server are provided. A failure monitoring device (10) includes: a monitoring unit (11) configured to transmit a monitoring signal from the outside of an SIP server (2) to the SIP server (2) and detect a failure of the SIP server (2) in accordance with presence/absence of a response from the SIP server (2) or details of a response from the SIP server; and a control unit (12) configured to inhibit, in a case in which the failure has been detected by the monitoring unit (11), a connection request toward a failed SIP server (2_1) in which a failure has been detected by the monitoring unit (11) in cooperation with a DNS server (1).

A system comprising: (a) a compliance server adapted to receive an audio feed of a telephone conversation; (b) an identification module adapted to ascertain IP addresses from which said audio feed originates and identify participants in said telephone conversation using said IP addresses; and (c) an analysis module adapted to ascertain a subject of said telephone conversation.

A system comprising: (a) a compliance server adapted to receive an audio feed of a telephone conversation; (b) an identification module adapted to ascertain IP addresses from which said audio feed originates and identify participants in said telephone conversation using said IP addresses; and (c) an analysis module adapted to ascertain a subject of said telephone conversation.