摘要:
Measures for use in operating a network node in a telecommunications network. At a network node located in a second part of the network, signaling messages transmitted between a server and end devices in an at least a first group are monitored. Each monitored signaling message includes a telephone dialing number for a given end device in the at least first group and an extension number for the given end device. On the basis of the monitoring, mapping data is stored which includes mappings between telephone dialing numbers and extension numbers for end devices in the at least first group. In response to detecting a loss of connectivity between a first part of the telecommunications network and the second part of the telecommunications network, extension number dialing request messages received from end devices in the first group are processed according to a detected loss of connectivity operating mode.
摘要:
The present invention discloses a method for implementing Wide Area Centrex (WAC), wherein the long number and short number corresponding relationship of the WAC users is set and saved. The method includes: routing the call initiated by a user to a soft-switch; receiving by the soft-switch the call information the calling user sends; after determining that the calling user is a WAC user based on the calling number, the soft-switch determining the route to the called user based on the call information and completing the call. The WAC, covering the Public Switched Telephone Network (PSTN)/Public Land Mobile Network (PLMN) user and the Next Generation Network (NGN) user of different physical networks, can be implemented by using the method of the present invention without any software or hardware change to the switches in the existing network.
摘要:
A communication system with a common channel for selectively coupling a subscriber line port of a telephone system to one of a plurality of voice communication devices including a fixed network communication module, an internet-based phone module, and a wireless internet-based phone module is provided. When a phone unit of the telephone system dials a called end phone number, the telephone system is connected to the common channel, a ring current generated by the ring current generator is then supplied to the phone unit, the called end phone number is decoded by the tone decoder, and thereby the microprocessor in correspondence to the called end phone number couple the telephone system to one of the fixed network communication module, the internet-based phone module, or the wireless internet-based phone module to establish a voice communication therebetween according to a preset data table stored in the memory.
摘要:
The Simulated Facility Group System manages a plurality of IP Centrex Groups within a GR303 interface, where each IP Centrex Group is assigned to a unique Simulated Facility Group. A Centrex Feature Gateway interconnects the IP Centrex Access Facilities that serve each IP Centrex Group to a GR303 interface. The Simulated Facility Group System calculates the IP Centrex Access Facility bandwidth requirement of each IP Centrex call set up request, based on a Call Type Parameter received from the Centrex Feature Gateway. On a per call basis, the Simulated Facility Group System checks the available bandwidth on the IP Centrex Access Facility that serves the Centrex Group from which the call originated. The Simulated Facility Group System determines if the call set up request requires a timeslot on the GR303 interface to the central office switch. If a timeslot is required, the Simulated Facility Group System determines if a timeslot is available on the GR303 interface. In addition, the Simulated Facility Group System enables the completion of intra-IP Centrex Group intercom calls without the use of IP Centrex Access Facility bandwidth.
摘要:
A telecommunication group is formed by a multiplicity of private telecommunication facilities configured to connect calls between clients services by those facilities and subscribers of those clients based upon facility identification numbers AID and client identification numbers KID determined by dispatcher numbers from E.164 call numbers.
摘要:
An intelligent network for providing access to an information network (80) has a number of central office switches (64, 82). Each central office switch (64, 82) is capable of receiving a call to a centrex telephone number and transmitting a query. A service control point (70) is coupled to the central office switches (64, 82) by a SS7 signal link (66). The service control point receives the query and transmits a response to the central office switch (64, 82). A hub switch (75) is coupled to the central office switches (64, 82) and receives the call. The hub switch (75) contains a digital trunk service (76) that combines the call with a number of other calls to form a data stream. The data stream is then transmitted to an information network node (80).
摘要:
The Simulated Facility Group System manages a plurality of IP Centrex Groups within a GR303 interface, where each IP Centrex Group is assigned to a unique Simulated Facility Group. A Centrex Feature Gateway interconnects the IP Centrex Access Facilities that serve each IP Centrex Group to a GR303 interface. The Simulated Facility Group System calculates the IP Centrex Access Facility bandwidth requirement of each IP Centrex call set up request, based on a Call Type Parameter received from the Centrex Feature Gateway. On a per call basis, the Simulated Facility Group System checks the available bandwidth on the IP Centex Access Facility that serves the Centrex Group from which the call originated. The Simulated Facility Group System determines if the call set up request requires a timeslot on the GR303 interface to the central office switch. If a timeslot is required, the Simulated Facility Group System determines if a timeslot is available on the GR303 interface. In addition, the Simulated Facility Group System enables the completion of intra-IP Centrex Group intercom calls without the use of IP Centrex Access Facility bandwidth.
摘要:
Telephone lines are moved from one or more switching modules (105 and 107) to one or more new switching modules (201) with minimal outage for customers whose lines are affected. Pseudo-lines are created (303) and the new switching module (201) and line data that supports customer features and routing of calls is programmed into the new switching module(s) (201). Connections are moved from the old module(s) (105 and 107) to the new module(s) (201), and the telephone numbers, i.e., routing information, is exchanged (309) between the modules in a single transaction.
摘要:
A method for dialing a telephone number indirectly, including obtaining an alphanumeric identifier that is assigned to a telephone subscriber and associating the identifier with a destination telephone number of the telephone subscriber to form a mapping therebetween. The method further includes storing the mapping in a lookup table, dialing the identifier using an originating telephone, using the identifier as a pointer to the lookup table so as to recover the destination telephone number from the stored mapping, and establishing a connection between the originating telephone and a destination telephone via the destination telephone number.
摘要:
Stimulus signalling protocols and message protocols are methods for transferring information over a communications network. Stimulus signalling protocols (e.g. P-Phone) are typically used for connecting simple terminals to a more powerful host whereas message protocols (e.g. H.323 standard) are typically used to connect between such hosts. Equipment has typically been developed and manufactured for use with either but not both systems and this has led to the need for protocol conversion devices which are complex and time consuming to develop and maintain. Voice data is sent via an H.323 channel over a data network and associated P-Phone signalling data is sent via a T.120 channel as part of the H.323 call. This enables stimulus signalling protocol equipment (e.g. business telephony handsets) to be used in message protocol systems without the need for protocol conversion.