92 results about "Busy tone" patented technology

A Service Control Point (SCP) has a call list associated with the main telephone number of a called party. When a calling party attempts to contact the called party by calling the main number, but cannot reach the main number, a Service Switching Point (SSP) fires a trigger. One trigger is a T_No_Answer trigger, which is fired in response to a call to the main number that is not answered within a predefined number of rings. Another trigger is a T_Busy trigger that is fired in response to a call to the main number that is made when the main number is busy. Each trigger causes the SCP to access the call list and to cause the SSP to ring an alternative telephone number associated with the main telephone number. If this attempt to reach the called party also fails, then the SSP again fires the trigger and an attempt is made to connect the calling party to the next number on the call list. If the call list is exhausted before the calling party is connected to a number on the called party's call list, then a message can be played to the calling party. Alternatively, the calling party may be connected to a voice mailbox associated with the called party or may receive a busy tone.

The invention discloses a VANET (vehicular ad-hoc network) media access control method. The VANET media access control method comprises the following steps of: distributing a special code path, namely a broadcasting code path, for broadcasting traffic safety information; dividing a busy-tone channel from a physical channel, so that a node transmits a busy-tone signal to notify a neighbor node that a current broadcasting code channel is occupied through the busy-tone channel, therefore, the problem of a hidden terminal on the broadcasting code channel is solved; simultaneously, placing a traffic safety information broadcast on a preferential position so as to ensure the timeliness of the traffic safety information broadcast; avoiding the increase of a window selection region by selecting K time slots as an advanced preferential competition time slot, and reducing the probability of selecting windows with the same size through a plurality of nodes so as to reduce confliction probability and reduce node access delay; simultaneously, integrally planning the time to be periodical control frames and data frames so that node-associated information can be periodically updated and can better adapt to the change of a vehicle network dynamic topologic structure; and finally, effectively improving the network throughput rate of the VANET by utilizing a DS-CDMA (direct sequence -code division multiple access) multi-address access advantage.

Disclosed are systems and methods which provide a busy tone or signal for use in communication collision avoidance in a wireless network, such as a WLAN, a cellular network, or other point-to-multipoint wireless network. An intelligent busy tone generator, upon the proper conditions, transmits a busy tone or signal to one or more wireless device which may otherwise be hidden from particular communications. According to embodiments the busy tone mimics a carrier sequence of an actual transmission to facilitate the wireless devices recognizing the channel as busy. Because the original uplink packet may be interfered by transmission of a busy tone carrier-sense sequence, embodiments are provided with interference cancellation capability. Interference cancellation circuitry of embodiments may operate in a plurality of modes, including a training mode (including an online training mode, an offline training mode, and a hybrid or combined mode) and an operating mode.

The business-customized user is as a called party (DP). When DP receives the call access request from the calling party (GP), he asks via business platform GP sending a Video Message (VM) to indicate his ID. According to VM, DP selects whether to answer this call or not. This completes the ID attestation. In case DP cannot be reached or a busy tone is returned, VM recorded by GP is sent to DP's Mail Box as a Video Mail for future read. This business system consists of the IP multi-media sub-system and videophone pass-card VPP business platform. This platform includes the application server AS, the media resource function server MRF, the media resource server, the database server and the gate net station Portal. The invention offers a forward active inquire mechanism against the strange calling and asks GP to record and send video ID recognition info. According to the Video Message, the called party selects whether to answer this call or not so as to prevent annoy calls effectively.

The invention relates to a multi-channel collaboration multi-address access method and a system. The method comprises the following steps: a source node selects a data channel from a channel state table and sends a CRTS packet on a control channel; when a target node which receives the CRTS packet judges that the data channel is available, the target node sends a CCTS packet on the control channel; when a neighbor node judges that the data channel is available and meets the collaboration conditions; the neighbor node sends a HA packet on the control channel and sends an RTH packet as a collaborative node; when the source node and the target node receive the RTH packet on the control channel, the source node and the target node respectively send an SCA packet and a DCA packet on the control channel, and the source node sends the data packet to the target node through the collaborative node on the data channel; and the target node sends a busy tone after receiving the data packet. On the one hand, the invention can solve the technical defects of low transmission reliability and the like of the existing data transmission due to conflicts, on the other hand, the invention can greatly reduce the application complexity and the cost.

The invention relates to a method and system for achieving incoming call filtering. The method is characterized in that a virtual extension is arranged, two paths of access telephone wires are arranged with one path connected to the virtual extension and the other to a delayer, the delayer is used for guaranteeing that the virtual extension intercepts and captures incoming call signals first, and the virtual extension classifies all caller IDs into four classes, namely a white list, a grey list, a black list and other IDs and controls a telephone line to be in a fully connected state, or in a selectively connected state, or in a call forwarding state according to the four classes of IDs. According to the method and system for achieving incoming call filtering, due to the adoption of the virtual extension technique, rejecting (the opposite side hears a busy tone), or forwarding to a voice mail, or answering and triggering a ringtone can be decided according to a strategy involved with time and the caller IDs.

The invention discloses a visible light full duplex continuous transmission random access method based on a channel reservation mechanism. Different from the CSMA / CA protocol in the existing IEEE802.15.7 standard, after a user executes idle channel estimation and detects that the channel is in an idle state, the user does not directly send a data frame, but firstly sends a RTS frame to reserve the channel, and sends the data frame after successfully reserving the channel. A full duplex visual light access point needs to broadcast busy tone to eliminate the hidden nodes when receiving the RTS frame or the data frame. Moreover, the user can directly execute CCA without executing random evasion again after successfully transmitting the data frame, therefore the conflict of the data frame is prevented, the time slot resources consumed by the random evasion mechanism are reduced, and the throughput and delay performance are improved. By adoption of the visible light full duplex continuous transmission random access method, the visible light hidden node problem is effectively solved, and meanwhile, the throughput and the delay performance of the visible light CSMA / CA protocol is optimized at the same time.

The invention provides a cooperation multiple access method and system for guaranteeing service quality and fairness. The multiple access method comprises the steps of respectively giving corresponding weight to and executing weighting operations on a transmitting queue service priority, transmitting queue head service continuous competition channel failure times and transmitting queue to-be-transmitted service quantity of each candidate sending party node, obtaining a combined priority of each transmitting queue of each candidate sending party node, obtaining a combined priority of each candidate sending party node from the combined priorities of the transmitting queues, converting the combined priorities of the candidate sending party nodes into binary codes, using the candidate sending party nodes to send busy tones or conduct monitoring on corresponding micro time slots of a plurality of micro time slots according to the binary codes, and therefore selecting one or more candidate sending party nodes with the highest priority. The number of the micro time slots is determined by the stage number of the combined priorities.

Method and apparatus for performing transmission data rate allocation in a high speed wireless communications network. A macro control loop with the network of base stations on one side and all the subscriber stations on the other side. Subscriber station selects a rate based on the amount of data queued for transmission (100). Adjusts this rate based on the available power headroom of the subscriber station (102). This adjusted transmission rate is then adjusted again to account for protection of base stations in the candidate set of the subscriber station (104). This rate is then adjusted in accordance with busy tone signals indicative of the loading conditions of active set base stations of the subscriber station (108). The base stations react to these action by refreshing measurements of their instantaneous traffic load and providing feedback in the form of soft busy tones. The algorithm is named Closed Loop Resource Allocation.

The present invention discloses a composite voice service terminal apparatus which includes a variety of voice call service functions to enable a user or subscriber to selectively use the services on his / her call purpose, and a method for using the same. The composite voice service terminal apparatus includes a Subscriber Line Interface Circuit / Subscriber Line Audio processing Circuit (SLIC / SLAC), a voice digital signal processor (DSP), a processor, an asymmetric digital subscriber line digital signal processor (ADSL DSP), an asymmetric digital subscriber line analog front end (ADSL AFE), a multiplexer (MUX) for outputting data to one of the channels from the digital tone generating unit and the voice digital signal processor, a digital tone generating unit for including sources of a dial tone, a ring back tone, a busy tone, and guide tones of voice over Internet protocol (VoIP), voice over digital subscriber line (VoDSL), and voice over multi-service broadband networks (VoMBN), generating a tone corresponding to a selected mode, and transmitting the tone to the Subscriber Line Interface Circuit / Subscriber Line Audio processing Circuit through the multiplexer, and a relay switch for maintaining an on state so that the user can use a general phone, and receiving an off signal from the processor and interrupting a general phone line when the user has pressed a specific digit for digital network services. The voice digital signal processor and the asymmetric digital subscriber line digital signal processor perform a voice processing procedure for satisfying the voice over Internet protocol, voice over digital subscriber line, and voice over multi-service broadband networks standards. The composite voice service terminal apparatus enables the user to compare and selectively use the composite voice services by call charges and quality. As a result, the user can easily make a phone call and use all the services including general phone call services, voice over Internet protocol (VoIP) services, voice over digital subscriber line (VoDSL) services, and voice over multi-service broadband networks (VoMBN) services, with one terminal at a lower price.

The invention relates to a MAC protocol communication method and device for a synchronous wireless sensor network. According to the method, by utilizing a periodic scheduling work mode in the node life period, each cycle period of a node is divided into synchronization time period, busy tone monitoring / sending time period, channel appointment time period and sleep time period, so that the node is allowed to carry out different operation in each time period according to the partial monitoring results; node leisure monitoring time is reduced; data multihop appointment continuous transmission in a cycle period is realized; requirement for data sending timeliness is adapted to; node period monitoring / sleeping duty ratio changes dynamically with the change of the service; node energy consumption is greatly saved; and meanwhile, through relay node selection in the channel appointment time period, channel appointment and route selection are combined together, and thus control overhead is reduced and the realization is simple.

A method for operating a telephony device accounts for variations in international standards for certain telephony operations, including hook flash functionality, pulse dialing operations, tone dialing operations, jitter buffer size, debounce characteristics, transmitter and receiver gain, caller identification functionality, pulse signaling operations, and ringing operations. The method includes defining each of the plurality of values as a management information base object; receiving a plurality of downloadable values downloaded to the telephony device over an Internet Protocol network as a management information base table; storing the plurality of downloadable values in a memory in the telephony device; and using the stored plurality of downloadable values to control operations of the telephony device, including hook flash functionality, pulse dialing operations, tone dialing operations, jitter buffer size, debounce characteristics, transmitter and receiver gain, caller identification functionality, pulse signaling operations, tone operations, call progress tones (e.g., busy tone, ringback, tone, dial tone, etc.) and ringing operations. Various ranges and default values are established to provide a wide range of operating capability.

In a related art mobile telephone system, the generation of an RBT (ring back tone) and a busy tone is realized by transmitting data registered in a network system to a calling telephone. In a mobile telephone system according to this invention, RBT data and the like are previously registered in a called telephone, and when a calling telephone originates a call to this called telephone, data corresponding to an operating state of the called telephone is transmitted to the calling telephone. Accordingly, an RBT and the like can be easily changed with a user's original melody or the like, whereby it is possible to improve service for the user.

The embodiment of the invention relates to a method, a device, a system and a terminal for call rejecting and forwarding. The method for call rejecting and forwarding comprises the following steps of: receiving a call request from a calling terminal, and transmitting the call request to a called terminal; receiving a reject response message of the called terminal to the call request, wherein the reject response message carries reject processing indication information; analyzing the reject processing indication information to acquire a reject service type, and inquiring whether the called terminal registers a reject service corresponding to the reject service type; if so, carrying out corresponding service treatment according to the reject service type. If the called party is inquired to register the call reject forward, the method, the device, the system and the terminal for call rejecting and forwarding can forward the call of busy tone which is caused by the rejection of the called party, therefore, the demands of users can be satisfied.

In a mobile telephone system of the prior art, generation of an RBT (Ring Bach Tone)and a busy tone is realized by transmitting date registered in a network-side device to a calling-side telephone. In a mobile telephone system according to the invention, the RBT data and the likes is registered in advance in the called-side telephone. When this called-side telephone is called, the data corresponding to the operating state of the called-side telephone is transmitted to the calling side telephone. As a result, the RBT or the like can be easily changed into an original melody of the user thereby to improve the service to the user.

A method and apparatus for performing transmission data rate allocation in a high-speed wireless communication network. Macro control loop of a network with base stations on one side and all user stations on the other side. The user station selects the data rate (100) based on the amount of data queued for transmission. The rate is adjusted (102) based on the peak power reserve available at the subscriber station. Subsequently, the adjusted transmission rate is adjusted again taking into account the protection of the base stations in the user station candidate group (104). The rate is then adjusted (108) based on busy tones indicative of load conditions on the subscriber station's working group base station. The base station responds to these operations by updating measurements of the subscriber station's instantaneous traffic load and providing feedback in the form of soft busy tones. This algorithm is called closed-loop resource allocation.

The invention provides a dynamic distributed cooperation on demand accessing method and system. The dynamic distributed cooperation on demand accessing method includes that each cooperation node competes according to the data transmission rate of the cooperation node and a transmitting party node and a receiving party node, whether having self data grouping transmitting and the times of grouping of already transmitted data of the cooperation node so that the successfully competing cooperation nodes are selected, wherein when one cooperation node has self data grouping transmitting, the winning cooperation node is selected according to the principles of first transmitting busy tones, having the longest busy tone transmitting time and not hearing other busy tones in a tiny slot time after busy tone transmitting. The dynamic distributed cooperation on demand accessing method and system reduce competition conflicts and expanses of the cooperation nodes and improve the data grouping transmission rate and the cooperation accessing control efficiency.

Disclosed are systems and methods which provide a busy tone or signal for use in communication collision avoidance in a wireless network, such as a WLAN, a cellular network, or other point-to-multipoint wireless network. An intelligent busy tone generator, upon the proper conditions, transmits a busy tone or signal to one or more wireless device which may otherwise be hidden from particular communications. According to embodiments the busy tone mimics a carrier sequence of an actual transmission to facilitate the wireless devices recognizing the channel as busy. Because the original uplink packet may be interfered by transmission of a busy tone carrier-sense sequence, embodiments are provided with interference cancellation capability. Interference cancellation circuitry of embodiments may operate in a plurality of modes, including a training mode (including an online training mode, an offline training mode, and a hybrid or combined mode) and an operating mode.

The invention discloses a channel reservation method based on a double busy tone mechanism in a wireless self-organizing network. Aiming at three conditions that network nodes (comprising transmitting nodes and receiving nodes) transmit and receive non-real-time data packets or transmit and receive a first real-time data packet, the network nodes transmit or receive the second and the following real-time packets, and the nodes monitor the real-time busy tone of transmission (RBTt) or real-time busy tone of reception (RBTr) when the adjacent node is started, channel reservation information is broadcast by the double busy tone mechanism, so that the real-time traffic can acquire the right to use the channel in advance, thus improving the success rate of channel reservation, greatly reducing the frame error rate of real-time packet transmission, and ensuring the service quality of the real-time traffic.

The invention discloses a heterogeneous wireless network cooperation coexistence method based on coordination busy tone intelligent protection. Targeting a heterogeneous network coexistence problem, the heterogeneous wireless network cooperation coexistence method designs an intelligent coordination busy tone protection strategy, and covers multiple dimensions like frequency, time, space, power, etc. On one hand, the heterogeneous wireless network cooperation coexistence method is compatible with TDMA, CSMA and a mixed access mechanism through early warning and parallel protection, and, on the other hand, the heterogeneous wireless network cooperation coexistence method avoids a monitoring dead zone problem of the coexistence network, expands monitoring ranges of multiple kinds of coexistence networks and network cooperation optimized information quantity; based on an internetwork disturbance classification method of disturbance decoupling and based on a standard that whether the various kinds of networks can monitor a counter party, the heterogeneous wireless network cooperation coexistence method avoids problems of thinking too much or not considering all about the co-existing interference; a busy tone channel is chosen according to a two-end rule and a left-right rule; and on the basis of low expenditure interference signal characteristic extraction, the heterogeneous wireless network cooperation coexistence method performs reasonable protection on a weak network in a coexistence environment and improves comprehensive performance of the coexistence network.

The invention discloses a magnet telephone converter. In the magnet telephone converter, a power supply module is respectively connected to a ring-current module, a ringing controlling module, a busy-tone detecting module, a feeding detecting module, a subscriber loop module, and a single-chip computer controlling circuit. An FXO interface circuit is connected to the single-chip computer controlling circuit through the busy-tone detecting module, the feeding detecting module, and the subscriber loop module. A magnet interface circuit is connected to subscriber loop module through the ringing controlling module and a ring-current detecting module. The ring-current module is connected to the ringing controlling module. The magnet interface circuit, an interface converting circuit and the FXO interface circuit are sequentially connected. With the converter provided by the invention, in a railway section, cable communication is replaced by optical fiber communication; a telephone communication network based on VOIP is adopted; magnet telephone connection can be used on site in the rail way section, and magnet dedicated line connection can also be realized on a station side.

This invention discloses a method for realizing internal voice calls of ONU (Optical Network Unit) devices in a passive optical network. The method comprises the following steps: configuring a local IP (Internet Protocol) and a Port of an OLT (Optical Line Terminal), a local IP and a Port of a calling ONU; powering on the calling ONU and setting up a socket; sending registration information to the OLT; after hooking off and dialing, getting into the dialing state; when dialing is finished, sending a called number and parameters of a calling SDP (Session Description Protocol) to the OLT; after the OLT matches and finds the called ONU, sending the calling number and the parameters of the calling SDP to the called ONU; if the called ONU can continue, sending the called number and the parameters of the called SDP to the calling ONU; playing a ring-back tone by the calling ONU; after the called ONU hooks off, sending a call establishment to the calling ONU so as to set up the call; if the called ONU cannot continue, sending a reject message to the calling ONU, and playing a busy tone by the calling ONU. By adopting the method of the invention, after the ONU is powered on, the socket is set up by using the local IP and local Port communicated with the OLT; thus, the messages are transmitted between the ONU and the OLT to realize the internal voice service; the messages are simply controlled between the ONU and the OLT, and the interaction is less; and the interaction process between the ONU and the OLT is greatly simplified.

A radio communication system, a radio communication control method, a radio communication apparatus, a control method for a radio communication apparatus, and a computer program make it possible to minimize the chance of co-channel interference in a multi-cell environment, and also permit free adjustment of an uplink / downlink ratio. The carrier sense region of a terminal wanting an uplink shares the region used as a directional busy tone region for a terminal wanting an uplink, and the region used as a reference signal transmission region of a terminal wanting a downlink. Furthermore, the carrier sense region of a terminal wanting a downlink shares the region used as a directional busy tone region for a terminal wanting a downlink, and the region used as a reference signal transmission region of a terminal wanting an uplink. If a result of the carrier sense indicates an idle mode, then a reference signal is transmitted.

The invention discloses a hidden node concurrent receiving method based on a busy tone for removing an ACK (acknowledgement character) frame interference, and mainly solves the problem that a hidden node cannot be used for concurrent communication in the prior art. During a communication process of a primary link, a neighbor node per se can be determined to be the hidden node when a CTS (clear to send) frame is received by the neighbor node; if an RTS (ready to send) frame is sent to the hidden node at the moment, a busy tone sending / receiving module in a starting system sends a continuous busy tone signal to a sending node of the hidden node for noticing the sending node to send a data frame, and the continuous busy tone signal can be replied to the sending node after the data frame is completely received by the hidden node; and whether the hidden node is interfered by an ACK frame or not during a data receiving process can be determined by the hidden node according to the ACK frame obtained from the CTS frame information, the removing for the ACK frame can be processed if the hidden node is interfered by the ACK frame, and the removing for the ACK frame is not needed to be processed if not. According to the invention, since the receiving for the data frame of the hidden node is realized, the concurrent degree of a network is increased, thus the throughput of the network is increased, and the hidden node concurrent receiving method can be used for a wireless Ad hoc network.

The invention, which belongs to the technical field of automatic measuring method of relay input busy signal parameters in a loop of voice access device of VoIP (Voice over Internet Protocol) communication system, discloses a method for detecting relay busy tone of a switch loop. The method comprises the steps of connecting media processing chip resource with a relay port link of a loop receivingbusy signals; collecting detection parameters of busy tone on-off time; setting comparative parameter group of busy signals; collecting busy tone comparison result and calculating busy signal parameters according to the result to obtain an actual frequency of busy signals of a front-end processor. In the method provided in the invention, the existing software and hardware resource of the relay access device of the VoIP system voice loop is utilized to realize the effect of obtaining the most precise onsite busy tone parameter detection data easily and rapidly.

The invention discloses a busy tone detecting method which includes the steps that A. the frequency of speech signals is tested to judge whether the frequency of the speech signals is the audio frequency of the busy tone or not and an indication signal of a judgment result is output; B. the response period and the duty ratio of the speech signal are detected by using the indication signal of the judge result; C. according to the response time and the duty ratio detected in the step B, whether the speech signal is a busy tone or not is determined and the indication signal of the busy tone is output if the speech signal is the busy tone. The invention also discloses a device for realizing the method at the same time. The method has the advantages of easy realization, low cost and reliable performance.

A distinguishing and treating method for two dial faults of a soft-exchanger user comprises following steps: (1) after detecting device removal of an access user, a gateway reports the device removal affair to the soft-exchanger by use of Notify messages; (2) the soft-exchanger sends the Notify messages to the gateway and directs the gateway to draw back the dial number from the user, and the Notify messages include number charts and timers for number drawing back; (3) when faults occurred upon user dialing are detected, the gateway sends the Notify messages to the soft-exchanger and reports the dial completion affair, the Notify messages takes following parameters: Observed Events: dd / ce, ds, wherein the Method adopts two different parameter values: 'FM' and 'PM' to distinguish the two faults of the user: not-dialing for a long time and wrong dialing of the first number; (4) The soft exchanger distinguishes the faults based on the empty number strands in the Notify messages and the values of the Method, and takes treating measures: sending the Notify messages to the gateway and directing the gateway to play busy tone for the user. This method can distinguish and treat the two dial faults of the soft-exchanger user accurately and duly with high efficiency, simple performance and ensuring of operation service quality.

The invention discloses an automatic alarm device for non-ring-off, comprising a telephone line access inlet, a telephone line outlet, a busy tone alarming module and a power supply, wherein the telephone line access inlet is connected with the telephone line outlet, the telephone line access inlet is connected with a telephone outside line, the telephone line outlet is connected with a telephone set by the telephone line, the input end of the busy tone alarming module is connected between the telephone line access inlet and the telephone line outlet, the busy tone alarming module is connected with the power supply. When the ring-off of the telephone set fails or a receiver drops, the telephone set is in the off-hook state for a long time, outside line can not enter, while the master do not know that, the invention can automatically alarm and remind the people of ringing off. The invention has simple structure, convenient use and lower cost, and can be matched with any of telephone set, the original telephone set can not be changed when in use; a single power supply, a single loudspeaker and a multiple-color light emitting diode combination are used, the alarming way is diversified and the alarming effect is good.