Patent classifications
H04M7/009
CALL FLOW SYSTEM AND METHOD FOR USE IN A VOIP TELECOMMUNICATION SYSTEM
A method of establishing a communication link between a mobile terminal of a wireless network and a subscriber of a network, such as an enterprise network, and/or a residential network.
System for cloud-enabling a premise PBX
In an example embodiment, a solution that allows a PBX to utilize bridged mobile or desktop applications for collaboration and softphone use is provided. The solution works in conjunction with existing hardwired PBX devices without requiring additional hardware such as an edge router to be integrated with the PBX hardware. Incoming calls are able to ring a user's hardwired device, such as a desk phone at an office, while simultaneously ring mobile device or desktop applications (apps). App users can place outbound PSTN calls and dial PBX extensions just as if they were using their hardwired device. Additionally, PSTN services are utilized from the existing PBX rather than bypassing the existing PBX, which makes tracking and billing more straightforward. A Session Initiation Protocol (SIP) tie trunk is used between the existing PBX and a software bridge controller.
Hybrid Cloud PBX
Disclosed is a system for telephones by providing an improved and streamlined user experience and enhanced fail over mechanisms. A decentralized system managed through a web site which allows for continued operation even when the primary systems fail includes a mechanism for restoring the primary systems automatically when they become available again. Phones connect to two PBX systems at the same time, one local and one at a remote location. The two PBX systems synchronize configuration data and media files between them. The website can also be used to manage any number of systems allowing any size organization to manage every phone system in their organization from a single interface.
Method and apparatus for speech behavior visualization and gamification
The disclosed system and methods provide real-time information about how agents and customers sound as they are speaking, allowing a supervisor to continuously monitor how agents are doing. The system allows agents to visualize their own speech behavior performance during and after a conversation while viewing important comparative information about prior conversations, and gamifies conversations in real-time by providing visual comparison between the live conversations and various target metrics. The visualization in-turn enhances agent interactive skills such as active listening and mirroring, as well as decision-making skills based on observations of customer engagement and distress levels.
Method and apparatus of supporting wireless femtocell communications
A method and apparatus of routing a call in a femtocell network are disclosed. In one example call routing method, a call is originated from the mobile station via a femtocell access point and the call is transmitted to a femtocell gateway, a mobile switching center and a carrier gateway server and onto an enterprise gateway server to obtain policy information. A routing policy is determined based on the obtained policy information and the call is routed to its destination based on the routing policy. The call may be routed via local media from a femtocell access point directly to the enterprise gateway server. The call routing procedures may implement the Iuh protocol and/or the session initiation protocol (SIP) for call signaling in the femtocell network. Call routing may be performed in a wireless cellular communications network or an enterprise network environment.
METHOD FOR AUTOMATED AUTHENTICATION OF VOIP PHONE
A method for automated authentication of a user VoIP phone supported by a Private Branch eXchange (PBX) configuration server is provided. A VoIP phone or a VoIP supported device is configured for an automated authentication by a vendor. The authentication method does not require manual entry of authentication data by a user. The unique VoIP phone authentication data can be provided by the vendor in a form of a MAC address. Additionally, the vendor can assign a digital certificate (containing public and private encryption keys) signed by the vendor to the VoIP phone. In this case, the VoIP phone vendor serves as a trusted authority. Thus, the VoIP phone automatically connects with the configuration server and the authentication transformation server (ATS) and the address where the VoIP phone sends the authentication data upon connection to the network is determined by the ATS.
Analysis of call metrics for call direction
Processing of communications routed by an IPBX server are disclosed. At least one processing circuit is communicatively coupled to an Internet-Protocol Private Branch Exchange (IPBX) server that is configured and arranged to route calls for a plurality of agents in a communications/call center. The processing circuit is configured to receive communications event messages from the IPBX server for communications routed by the IPBX server, generate, during a communication to a first agent of the plurality of agents, a set of data metrics including communications summary metrics based on the communications event messages; and redirect, during the communication to the first agent, the communication to a second agent of the plurality of agents in response to the set of data metrics satisfying a set of criteria indicated in a policy.
Method and apparatus for routing voice calls over voice over internet protocol networks
A method is provided for improving voice quality of Voice over IP networks in which a highest-quality routing protocol is interposed between a local IP PBX exchange and a cloud-based Internet service provider server to which calls are to be routed, wherein the highest-quality routing protocol detects the quality of the voice channel between the local IP PBX exchange and cloud-based Internet service provider servers and routes voice calls to that cloud-based Internet service provider server exhibiting the highest voice call quality, with the highest voice quality server connection determined by detecting lost packets and packet delay between the local IP PBX exchange and a server.
Transition from a Legacy PBX System to an Advanced IP-Enabled System
A method is provided for replacing an enterprise legacy PBX with an advanced IP-enabled system, comprising: (i) providing configuration data of the legacy PBX; (ii) analyzing the data provided, and detecting missing details from among the legacy PBX configuration data; and detecting conflicts that exist among the legacy PBX configuration data; (iii) retrieving information associated with missing details by approaching data source(s) other than the source for the legacy PBX configuration data, and resolving detected conflicts; (iv) converting data associated with the users of the legacy PBX for use by a system implementing an advanced IP-enabled solution; and (v) configuring the advanced IP-enabled system accordingly.
Template-based management of telecommunications services
Certain aspects of the disclosure are directed to template-based management of telecommunications services. According to a specific example, a VoIP server is provided comprising one or more computer processor circuits configured to interface with a remotely-situated client entity using a first programming language. The VoIP server includes a call control engine that is configured to provide a private branch exchange (PBX) for the client entity, and identify at least one call control template written in a second programming language. The call control engine is further configured to control call routing by the PBX and for the VoIP telephone call by executing the call control template to identify at least one data source that corresponds to a call property for the VoIP telephone call, retrieve data from the data source, and implement one or more call processing functions specified by the call control template as being conditional upon the retrieved data.