91 results about "IP PBX" patented technology

A comprehensive solution for providing secure mobile communication is provided. The system includes techniques for authentication and control of communication end-points; chain of trust to ensure devices are certified as authentic; contact list management; peer-to-peer encrypted voice, email, and texting communication; and a technique for bypassing an IP PBX to ensure high levels of security. The system is able to support use of commodity mobile communication devices (e.g., smart phones, laptops) over public carrier networks.

A method is provided that includes receiving a request from a communication device to establish a communication session with a mobile station, the mobile station being operable to roam between a private and a public network. The mobile station is signaled via a cellular data network that a call is being initiated for the mobile station. Signaling information may be exchanged with a voice gateway such that one or more voice circuits are established. A signaling pathway may be established between an Internet protocol private branch exchange (IP PBX) and the mobile station via the cellular data network. The establishment of the signaling pathway is substantially concurrent with the establishment of one or more of the voice circuits. One or more features associated with a private network are delivered to the mobile station during the communication session as an end user moves between the public and private networks.

A system and method for configuring a communication device such as IP phone to operate in central and remote networks is disclosed. A central network includes an Internet Gateway Device, an Internet Protocol Private Branch Exchange (IP PBX) and a communication device. The communication device downloads and stores a configuration file including the externally-visible IP address or unique host name of the IP PBX. Upon installation in a remote network, the communication device uses its configuration file to locate the IP PBX and configure itself automatically for use in the remote network.

A serverless telephony system (STS) establishes and maintains a voice, data and video exchange between telephony devices using Internet Protocol (IP). The STS in a preferred embodiment comprises software running on a telephony device with a standard packet network connection over a local area network (LAN) or a wide area network (WAN), or a combination, which may include the Internet network. The new system enables one telephony device to establish a connection with another telephony device without intervention by any third device such as a switch or a server. In one embodiment an IP-PBX (Internet Protocol-Private Branch Exchange) application is provided having no server presented in the network for establishing a call between any two IP phones on a Local Area Network (LAN), providing substantially all services typically expected from a standard PBX, and yet functions in compliance with one or more of a H.323, a MGCP, a SIP protocol or any one of other standard protocols.

An IP PBX system that serves enterprise telephones via a landline IP network connection is be expanded to serve cellular wireless communication devices (WCD) via a cellular wireless carrier's radio access network (RAN). Calls to and from the cellular WCD are connected through the cellular carrier's RAN and the IP PBX system, so that the IP PBX system can control and manage the calls just as the IP PBX would control and manage calls involving other extensions on the IP PBX. A cellular WCD thereby becomes an IP PBX client station, i.e., an extension on the IP PBX system. As such, the cellular WCD can seamlessly benefit from many of the same IP PBX features that other more conventional IP PBX client stations (e.g., desk phones) enjoy.

A Voice over Internet Protocol (VoIP) proxy server is described. The VoIP proxy server is placed in a DMZ of a company's secure internal communication network. The VoIP proxy server receives VoIP control messages and VoIP media streams and based on the MAC address and source IP address contained in the control message, pushes a policy change to the internal network's external firewall to open call control protocol ports and Real Time Protocol (RTP) ports only for packets from the source IP address. The VoIP proxy server manages the hiding of the company's internal network address and directs incoming VoIP packets to an IP-PBX connected to the company's internal network.

An Internet Protocol telephony exchange system and a call control method thereof. The exchange system integrates an IP terminal and a legacy terminal in a single exchange system, controls a call of the IP terminal in the same manner as an extension subscriber, and contains at least one IP terminal connected to the IP network. The IP terminal contains a user interface for generating user information or converting received user interface information into information of a user-identifiable state, and performs a VoIP function in compliance with one or more of a H.323, a MGCP and an SIP protocol procedure. The IP network is connected to an IP driver and an IP-PBX (Internet Protocol-Private Branch Exchange). The IP driver controls an IP end point at which the IP terminals are positioned, upon receiving IP terminal connection information and call control information based on H.323 / MGCP / SIP protocol procedure.

A standard-SIP (session initiation protocol) protocol control module and an expanded sip protocol control module are provided so as to correspond to respective Sip protocol corresponding terminals, such as wireless LAN terminals compliant with the SIP. A signal (a request) from each Sip protocol corresponding terminal is converted into a signal according to a protocol of am IP-PBX and is sent to a PBX basic-connection-processing providing module. A signal (a response for the request or a request to the Sip protocol corresponding terminal) from the PBX basic-connection-processing providing module is converted into an Sip protocol signal and is sent to the Sip protocol corresponding terminal. A location-information management module collects a basic service set identifier (BSSID) from each wireless LAN network and stores the BSSID information into a database.

A switching apparatus for switching packetized voice traffic between a plurality of communication devices, the switching apparatus comprises a multi-layer switch, a plurality of communication ports, control means and ingress processing means, said packetized voice traffic comprises call control packets and medium packets which are exchanged between the communication devices via said communication ports, wherein medium packet traffic from a first communication device to a second communication device is split into a first call segment and a second call segment, the first call segment originates from said first communication devices and terminates at said switching apparatus, the second call segment originates from said switching apparatus and terminates at said second communication device, each medium packet from said first communication device is processed by said ingress processing means of said switching apparatus before onward transmission to said second communication device.

There is disclosed a private branch exchange in which an SIP terminal is capable of receiving great variety of services provided by the private branch exchange without upgrading version of software of the SIP terminal itself. A standard SIP protocol control module 17 and an expanded SIP protocol control module 18 are disposed for SIP corresponding terminals, convert signals from an SIP system protocol corresponding terminal into signals of a protocol inside an IP-PBX 1 to transmit the signals to a PBX basic connection process providing module 12, and convert signals from the PBX basic connection process providing module 12 into signals of an SIP system protocol to transmit the signals to the SIP system protocol corresponding terminal. A PBX additional service providing module 11 and the PBX basic connection process providing module 12 provide additional services in response to an hooking operation from the SIP corresponding terminal or an input of a number for starting the service.

A method is provided that includes receiving a request from a communication device to establish a communication session with a mobile station. The request is responded to by signaling the mobile station via a cellular data network that a call is being initiated for the mobile station. Signaling information may be exchanged with a voice gateway such that one or more voice circuits are established to accommodate voice data that may propagate between the communication device and the mobile station. A signaling pathway may be established between an Internet protocol private branch exchange (IP PBX) and the mobile station via the cellular data network. The establishment of the signaling pathway is substantially concurrent with the establishment of one or more of the voice circuits such that one or more features associated with a private network are delivered to the mobile station during the communication session.

Disclosed is an apparatus and a method for placing calls between legacy terminals and IP terminals in an IP-PBX system. Features that were once only used in traditional PBX telephone systems, like group ring services and station group services are now made available for IP terminals in an IP-PBX system. Methods of providing station group service and ring group service to both legacy terminals and IP terminals in an IP-PBX system is disclosed.

A standard-SIP (session initiation protocol) protocol control module and an expanded sip protocol control module are provided so as to correspond to respective Sip protocol corresponding terminals, such as wireless LAN terminals compliant with the SIP. A signal (a request) from each Sip protocol corresponding terminal is converted into a signal according to a protocol of an IP-PBX and is sent to a PBX basic-connection-processing providing module. A signal (a response for the request or a request to the Sip protocol corresponding terminal) from the PBX basic-connection-processing providing module is converted into an Sip protocol signal and is sent to the Sip protocol corresponding terminal. A location-information management module collects a basic service set identifier (BSSID) from each wireless LAN network and stores the BSSID information into a database.

An Internet protocol (IP)-private branch exchange (PBX) system and a method for linking the system with a Voice over Internet Protocol (VoIP) terminal. The IP-PBX system and method allow the VoIP terminal to be registered at an extension of the IP-PBX system, the VoIP terminal supporting a protocol, e.g., H.323 or session initiation protocol (SIP), standardized by a standardization organization, e.g., the International Telecommunication Union (ITU), the Internet Engineering Task Force (IETF), or the European Telecommunications Standards Institute (ETSI).

The invention discloses an automatic registration system of an IP telephone of an IP-private branch exchange (IP-PBX). The system is characterized by comprising an automatic registration server and an IP telephone automatic request session initiation protocol (SIP) registration client of an IP-PBX system, and a dynamic host configuration protocol (DHCP) server needing third-party supported option66. The system can effectively fuse user number management and IP telephone registration on the IP-PBX system together; any configuration on the IP telephone is not needed, namely the telephone is automatically registered after power on or restart, and the related configuration information of the IP telephone is automatically acquired; and a user can immediately dial the IP telephone, so the problems of telephone registration and number management are effectively solved.

A system, an apparatus and related method for roaming in DECT-VoIP network are provided. The system for roaming in DECT-VoIP network includes at least a DECT-VoIP handset, and at least a DECT-VoIP apparatus. The system can be connected to the Internet through connecting the DECT-VoIP apparatus to appropriate internet access device, such as, ADSL modem or IP PBX so as to make and receive phone calls through VoIP. With presetting the VoIP client account information at the DECT-VoIP handset, the DECT-VoIP handset can roam from the coverage of a first DECT-VoIP apparatus previously registered with to the coverage of a second DECT-VoIP apparatus sharing the same ID with the first DECT-VoIP apparatus without interruption to the connection.

The invention discloses an IP-PBX system which can realize a system cluster and improves the capacity of the system and the efficiency of the system through function separation of a softswitch service and a media gateway. To solve the problem, the technical proposal of the IP-PBX system of the invention is that: the IP-PBX system at least comprises more than two servers, wherein, the servers comprise a central server and a media server; the central server comprises a softswitch service module and a database; the media server comprises a media gateway module; a communication record of the media server is recorded in the database on the central server. The system can also comprise more than one local softswitch servers, each server comprises a local softswitch service module and a built-in gateway; the local softswitch service module is used to process local softswitch services and process the call control of the built-in gateway.

The invention provides a voice communication scheduling command system which comprises a scheduling server and an internet protocol-private branch exchange (IP-PBX), wherein the scheduling server is used in the voice communication process to achieve scheduling control of call services, and the IP-PBX is connected to the scheduling server to achieve call control in the voice communication process. Due to the technical scheme, the call control and scheduling services can be separated, and thus ability of load balancing of a system is enhanced, and expansion of the scheduling services and flexibility of configurations are promoted.

A method for establishing communication via a VoIP network bypasses the IP PBX component conventionally used to obtain address information. Instead of obtaining the IP address from a location register of the IP PBX, the method involves use of a server configured to assign and provide to the caller's communication device a unique address (IP address / port) of a proxy. The caller then sends a Short Message Service (SMS) text message to the callee with the assigned address of the proxy. Thereafter, the caller and the callee connect at the assigned address of the proxy, thereby forming a communication path. Preferably, the devices operated by the parties are conventional smart phones. According to the preferred embodiment of the present invention, the method further comprises the steps of: determining whether to encrypt the communication; and encrypting the communication, if it is determined that the communication is to be encrypted. Preferably, the Station-to-Station (STS) protocol is used to encrypt the data packets.

A system and method for integrating a legacy IP-PBX to a session initiation protocol (SIP) network supporting one or more SIP sets. A SIP-PBX proxy server coupled to the legacy IP-PBX and the SIP network maintains a logical IP set for each SIP set on the SIP network. The logical IP set is configured to maintain IP-based signaling and media connectivity with the legacy IP-PBX. The SIP-PBX proxy server receives media directed to a logical IP set associated with a particular SIP set, and redirects the media to the particular SIP set. In doing so, the SIP-PBX converts messages that adhere to a private, vendor-specific protocol, to messages that adhere to the SIP protocol.

A telecommunications system has been developed that includes an on-premise IP PBX and an offsite hosted IP PBX. The system includes logic that automatically and / or manually migrates system configuration data from the on-premise IP PBX to the offsite IP PBX during a failure of the on-premise IP PBX.

The invention provides a method and a system for bonding fixed-line extension telephones and mobile terminals. The method comprises the following steps of: establishing SIP UA (Session Initiation Protocol User Agent) connections between soft switch equipment and an IP (Internet Protocol) private exchange according to the requirement of a voice concurrency amount and assigning a soft switch number for each SIP UA connection as an outside line number of a fixed-line telephone; setting fixed-line extension telephone numbers under the outside line number of the fixed-line telephone, wherein each fixed-line extension telephone number corresponds to a fixed-line extension telephone; configuring mobile terminals to be bonded to the fixed-line extension telephones, storing a corresponding relation between the mobile terminal numbers and the fixed-line extension telephone numbers in the IP private exchange and storing a corresponding relation between the mobile terminal numbers and the outside line number of the fixed-line telephone in an application server; sending registration information to the soft switch equipment by the IP private exchange in an SIP UA mode; and configuring business subscription information for the bonded the mobile terminal numbers in a network to which the mobile terminals belong. The invention realizes the access of an IP PBX (Private Branch Exchange) into a telecommunication network and the deep integration application of the telecommunication network and an enterprise network.

The present invention relates to a call processing message converter and message converting method in an IP-PBX system, capable of controlling IP terminals equally to legacy terminals by transceiving user information with IP terminals. The message converter works best especially for the IP-PBX system, wherein the system includes: a hub connected to an Internet protocol (IP) network, having a plurality of LAN communication ports; user information interface connected to the LAN communication port of the hub for generating user information and for outputting the generated user information or for converting received exchange user information to an identifiable state to users; at least one IP terminal for performing VoIP function in conforming to a pre-designated data communication protocol procedure; IP driver connected to the LAN communication port of the hub for controlling Internet protocol endpoints; and IP-PBX, mounted with a database where telephone numbers of a legacy terminal and the plurality of IP terminals and corresponding terminal information, and an exchange interface and a gateway connected to the LAN communication port of the hub, respectively, for conducting call processing services corresponding to user information that is sent to the exchange interface and call connection information that is generated from the legacy terminal. The call processing message converter is built in the IP driver, and the converter analyzes an ARQ (Admission request) message transported from a terminating IP terminal, transports port information and digit information of the IP terminal to a call processing module inside of the IP-PBX, and transports a termination seizure message transported from the call processing module and user control information of a terminating IP terminal to a terminating terminal.

The IP-PBX 1 includes a multimedia gateway controller 2 in which a call control data master table stores call control data for all terminals falling under the control of the IP-PBX and protocol handlers which are installed on a system bus 101 or expansion system bus 102 in a sufficient number for serving the number of circuits accommodated by the IP-PBX. On each of the protocol handlers, a microprocessor performs processing tasks corresponding to a load shared on it in interfacing with a given number of terminals, using a call control data slave table in its memory, call control protocol conversion function, fault-supervisory-processing function, call control protocol retransmission function, call control protocol transfer function, and terminal supervisory function.

The IP-PBX includes a multimedia gateway controller in which a call control data master table stores call control data for all terminals falling under the control of the IP-PBX. In the event of failure of a PH accommodating IP phones and the like via the LAN, the multimedia gateway controller transfers the contents of the call control data slave table existing on the PH to the call control data slave table on another PH.

An IP-PBX has a software-based built-in control unit, a card slot into which a control card is plugged, and a data bus connecting the software-based control unit and the card slot. The software-based control unit includes a first management unit managing up to a first number of IP terminals, and an IP terminal registering unit connected to the first management unit and the card slot. The control card includes a second management unit managing up to a second number of IP terminals. The IP terminal registering unit compares a current number of IP terminals managed by the first management unit with the first number when an IP terminal requests communication. The IP terminal has an ID number and an IP address. If the current number is smaller than the first number, the IP terminal registering unit associates the first management unit with the ID number and the IP address, and the first management unit manages the IP terminal. If the current number is equal to the first number, the IP terminal registering unit associates the second management unit with the ID number and the IP address, and the second management unit manages the IP terminal.