Abstract:
Methods and systems for intelligently terminating calls are provided herein. In some embodiments, a method for intelligently terminating calls may include receiving a call request directed to a communication identifier associated with a first user, determining a call termination action to associate with the call request based on (a) information associated with the call request and (b) previous call termination patterns associated with the first user, and terminating the call to one or more devices associated with the communication identifier based on the determined call termination action.
Abstract:
Bei einem Verfahren zur Integration von Funktionen eines Telekommunikationsnetzes (TN) in ein Datennetz (DN) wird ein Vermittlungssystem (PBX) des Telekommunikationsnetzes mit einem IM-Server (XS) des Datennetzes über eine Einrichtung (CCGW) verbunden wird, mit deren Hilfe der IM-Server in die Lage versetzt wird, Computer-Telefonie-Integrations-Dienste des Vermittlungssystems (PBX) einem Kommunikationsteilnehmer des Datennetzes zur Verfügung zu stellen.
Abstract:
The method of the invention comprises : the step of updating a table (D(IDB)) associated with an identifier of a first terminal, said table including an identifier and a state of at least one second terminal (Tc1, Tc2, Tc3), said update being carried out upon reception of a notification from said second terminal indicating a change in one of the states thereof; the step of establishing a list (LID) based on said table, that includes at least one identifier of a second terminal, upon reception of a message (Rl, R2) including an identifier of said first terminal; the step of sending said list to said first terminal; the step of receiving from said first terminal an identifier of the target terminal (TC1) selected from the list; the step of routing said call towards said target terminal.
Abstract:
A method for call connection is provided. The method includes: controlling the switching device to establish the link with the queuing machine according to the received trigger service request; controlling the queuing machine to inquire the number of destination user through the link; controlling the queuing machine to tear down the link; controlling the switching device to connect to the number of destination user. A system, a service control device, a switching device and a queuing machine for call connection are also provided. The problem of link circuity of the queuing machine in the voice value-added service call is resolved.
Abstract:
There is provided a method and system for making a call from a first communications device at any location to a recipient with a second communications device by inputting a contact number of the recipient into the first communications device, selecting a communications mode on the first device, wherein at least one of the first communications device and the second communications device operate using at least one data channel, and wherein a caller makes the call in a manner similar to "dial-and-connect" phone calls. There is also provided a method and system for sending at least one message from a first communications device at any location to a recipient with a second communication device by inputting a contact number of the recipient into the first communications device when sending the at least one message, transmitting the contact number and the at least one message to a server through at least one data channel: storing the at least one message at the server; the server contacting the second communications device; and either the second communications device drawing the at least one message from the server through at least one data channel, or the server transmitting the at least one message to the second communications device, wherein the messenger sends the at least one message in a manner similar to "compose-and-send" messaging.
Abstract:
An exploitation of the proper classification of conventional call-processing states into categories, each category corresponding to one of the call states of the SIP finite- state machine (FSM), yields simplified design, implementation, and operation of heterogeneous call processing. A separate FSM for the S I P- categorized call states, running in parallel with a conventional port event processing (PEP) FSM, implements a VoIP Local Call Controller by passing events between the SIP-type FSM and the PEP FSM only at the points where a transition between SIP categories occurs. The SIP FSM does not require notification of PEP FSM transitions not affecting the SIP call state. For this reason, none of the code associated with the transitions between PEP FSM states within the same major state category requires any modification. Consequently the integration of VoIP and PEP call processing can be done more simply, and at dramatically-reduced cost in software development, support, and maintenance.
Abstract:
Apparatus and method for detecting periods of voice signal silence and allowing a data station (50) to acquire the voice signal TDM time slot to be used for data transmission. This is enabled since all data stations (50) on the multipoint communication line (47) listen to the line (47) during the first part of a voice signal time slot, all the data stations (50) on the communication line (47) know that the remainder of the voice time slot can be used for data transmission.
Abstract:
Methods and systems for processing a signaling message are disclosed. An exemplary method comprises: determining a first transcoding policy (330A) associated with an originator endpoint (320A) contained in a received first call offer (410); determining a second transcoding policy (33OB) associated with an answerer endpoint (330D) contained in the first call offer (410); applying the first transcoding policy (330A) to a first offer codec set in the first call offer (410) to produce a second call offer (420) containing a second offer codec set; applying the second transcoding policy (330B) to the second codec set to produce a third call offer (430) containing a third offer codec set; comparing a first answerer codec set in a received first answer (440) and a second answerer codec set contained in the second call offer (420); and determining whether or not to perform transcoding based on a result of the comparison.
Abstract:
A method, communication device and system for smart route dialling to a destination identifier using a telephone are described. In one embodiment, there is provided a method for routing a call from a communications device having at least voice capabilities, the communications device being connected to a communications network for transmitting and receiving voice data and other data over the communications network, the method comprising the steps of: selecting a destination identifier for the call, the destination identifier comprising one of: a landline telephone number, a mobile telephone number, an instant messaging (IM) address, and a session initiation protocol uniform resource indicator (SIP URI); connecting the call to a Voice over Internet Protocol (VoIP) gateway; and routing the call from the VoIP gateway to the destination identifier, comprising: if the destination identifier is a landline telephone number, routing the call to a respective landline telephone associated with the destination identifier via a public switched telephone network (PSTN), if the destination identifier is a mobile telephone number, routing the call to a respective mobile telephone associated with the destination identifier via a wireless communications network, if the destination identifier is an IM address, routing the call to a respective IM voice network (VoIM) client via a VoIM network associated with the destination identifier, and if the destination identifier is an SIP URI, routing the call to an SIP-compatible VoIP phone associated with the destination identifier via the Internet.