摘要:
A telephone exchange system includes a connection interface connecting a communication line and a plurality of extension telephones, a database having a dial-in conversion table registering a received number in association with a number after conversion for a dial-in incoming call, and an extension number table registering an extension telephone in association with an extension number of the extension telephone, and a main controller connected to the connection interface and the database. When the dial-in incoming call comes from the communication line, the main controller retrieves an extension number which coincides with the received number from the extension number table. When there is an extension number which coincides with the received number, the main controller causes an extension telephone corresponding to the extension number to ring.
摘要:
A method may include determining a count of excess telephone numbers (TNs) for removal from a block of TNs associated with a customer. The method may further include determining an order of priority for removing the excess TNs from the block of TNs and marking the excess TNs for removal from the block of TNs in the determined order of priority.
摘要:
A method (400) and system (300) of indicating receipt of a call from a caller on a shared home network can include extracting (402) information from the calling party or the calling party's network indicating an originating network, sending (406) information of the originating network to a called party, and indicating (410) if the originating network belongs to the called party's shared home network. The indication can be an icon (11) or other visual indication on a mobile subscriber unit's user interface (13), or by sharing v-card or other user identification information between the calling party and the called party or by sending or providing an audible indication to the called party. The indication can be done before enabling the ability of the called party to answer a call. The step of extracting can optionally include extracting (404) information from billing information to indicate the originating network.
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 private branch exchange changes the load distribution according to the use condition by a main control unit and a sub-control unit. A line card comprises a first module group normally carrying out various operations in response to an instruction of the sub-control unit, a second module group carrying out various operations in response to the sub-control unit or the main control unit, a bus arbitration circuit, and an internal module bus selecting unit having a bypass selector for bypassing the bus arbitration circuit. The main control unit predicts the load on a system at the start of the system from the information stored in the incorporated line card, and determines, considering the prediction result, whether the control is made by the sub-control unit or the main control unit, for each module of each line card.
摘要:
A method for minimizing toll call costs for completing a call at an originating PBX within a network of connected PBXs to a remote destination, comprising the steps of generating within the originating PBX and communicating to each other PBX within the network a message calling for bids from each PBX for least cost routing of the call according to respective local routing plans, determining within the originating PBX a least cost direct route for completing the call, determining from the respective local routing plans the least cost routing for each other PBX and communicating the bids to the originating PBX, and comparing and in response selecting within the originating PBX a least costly one of the least cost direct routes and the least cost routing for each other PBX for completing the call to the remote destination.
摘要:
According to this invention, a digital key telephone system connected to an analog public network NW having a function of transmitting a ringing signal including identification information of a calling line through a subscriber line (CO line), accommodating a plurality of extension lines each connected to a digital key telephone (DKT) 2 or a standard telephone (STT) 4 as an extension terminal, and having a function of switching and connecting the subscriber line to the plurality of extension lines or the extension lines to each other includes a called party storage means storing, in advance, information representing the correlation between the calling line and the extension terminals 2 and 4 as a called terminal. When a ringing signal arrives from the analog communication network NW, calling line identification information (caller ID) contained in the ringing signal is detected by a calling line identification information interface unit (RCIU) 12. A control unit (RCTU) 16 determines the called extension terminal on the basis of the detected caller ID and the information stored in the storage means, so the extension terminal receives the call from the digital key telephone interface unit (RDKU) 13 or a standard telephone interface unit (RSTU) 15.
摘要:
The present invention proposes a device having a voice communication server structure comprising a rack called main rack including: a board called master board equipped with: a Central Processing Unit (CPU), a Digital Signal Processor (DSP) called master DSP for a telephonic application running on said master board, and having an access to a switching unit, a second DSP, distinct from said master DSP, for a telecommunication application and having an access to a switching unit, inter-DSP communication means arranged to allow in real time a direct exchange of information between said master and second DSP.
摘要:
An application module interface (AMI) is disclosed that allows one or more modules to access voice or data channels in a private branch exchange environment. The AMI provides a number of hardware signals to the telephone terminal and/or modules. The hardware control signals include module present, phone receiving OK and module receiving OK. The module present (MP) signal indicates that a module is plugged into the AMI electrical interface. The module receiving OK (MOD-OK) and phone receiving OK (PHONE-OK) signals indicates that the module and phone, respectively, is receiving valid AMI control channel messages. The hardware control signals permit the presence of a telephone terminal and/or a module to be detected, as well as to detect the disruption of an AMI connection. The hardware control signals are used to implement a start-up, recovery and tear-down mechanism. The disruption of the AMI connection can be detected if a telephone terminal or a module detects that its counterpart receiving OK (PHONE-OK or MOD-OK) signal has been turned off. In addition, the telephone terminal may detect that the module present (MP) signal has been turned off.
摘要:
A solution to the music-on-hold problem associated with audio conference calls. The music on-hold-problem occurs when a conferee having music-on-hold puts the conference call on hold, resulting in a continuous stream of music being transmitted to the other conferees. Such a conferee is called an offending conferee. The solution presented herein is to prevent music-on-hold signals emanating from an offending conferee from being passed through an audio conference bridge to the other conferees. This is accomplished, in particular embodiments, by directing a merging/summing subsystem of the audio conference bridge to temporarily stop combining audio emanating from the offending conferee from being combined or merged onto audio channels through which the other conferees communicate on the audio conference. Once the music-on-hold is terminated, the offending conferee can rejoin conference call by sending a signal that directs the merging/summing subsystem to resume the combining of audio signals emanating from the offending conferee onto the audio channels of the other conferees.