System and method for speeding call originations to a variety of devices using intelligent predictive techniques for half-call routing

A mobile application gateway configured to interconnect mobile communication devices on a cellular network with an enterprise network is provided. The mobile application gateway includes a voice and data signaling gateway configured to provide routing functionalities, service functionalities and admission control. A gateway GPRS support node (GGSN) is configured to establish a secure data session between one or more of the mobile communication devices and the enterprise network by establishing a GPRS tunneling protocol (GTP) tunnel between a carrier-hosted serving GPRS support node (SGSN) and the GGSN.

CALL FLOW SYSTEM AND METHOD FOR USE IN A LEGACY TELECOMMUNICATION SYSTEM

A method of operating a network server, such as a mobile application gateway, connect devices on a cellular or carrier network with individual networks, such as enterprise voice and data networks or residential networks. The effects of the present invention are far reaching in terms of transferring effective call control from the cellular network into the control of the individual network, such as the enterprise, and enabling new business models for the purchase of cellular service from a public cellular carrier by an enterprise. 1. A method , comprising:providing, by an enterprise gateway server, a first routing information to a carrier gateway server, the first routing information comprising at least one routing instruction based on at least one policy, wherein the at least one policy comprises enterprise session routing criteria for at least one of a restriction on times the session may be routed and a restriction on a destination where the session may be routed;transmitting from the carrier gateway server to a mobile switching network, a second routing information, the second routing information comprising at least one of destination information and session path information for routing the session to a destination; androuting, by the mobile switching network, the session to the destination.2. The method of comprising supporting claim 1 , by the enterprise gateway server claim 1 , session routing between a private branch exchange deployed in an enterprise network and a cellular network.3. The method of claim 2 , wherein the destination is the private branch exchange.4. The method of claim 1 , wherein the destination is the enterprise gateway server.5. The method of claim 2 , wherein the destination is the enterprise gateway server via the private branch exchange.6. The method of claim 2 , wherein the destination is the private branch exchange via the enterprise gateway.7. The method of claim 2 , wherein the enterprise gateway server routes to a final destination ...

SYSTEM AND METHOD FOR ENABLING VPN-LESS SESSION SETUP FOR CONNECTING MOBILE DATA DEVICES TO AN ENTERPRISE DATA NETWORK

A mobile application gateway configured to interconnect mobile communication devices on a cellular network with an enterprise network is provided. The mobile application gateway includes a voice and data signaling gateway configured to provide routing functionalities, service functionalities and admission control. A gateway GPRS support node (GGSN) is configured to establish a secure data session between one or more of the mobile communication devices and the enterprise network by establishing a GPRS tunneling protocol (GTP) tunnel between a carrier-hosted serving GPRS support node (SGSN) and the GGSN. 1. A method comprising:originating a first call leg between a subscriber and a carrier gateway server;identifying a call line preference via outgoing call digits from the subscriber specifying a call line to use for an outgoing call;originating the outgoing call to an intended destination on the specified call line to establish an outgoing call leg;connecting the outgoing call leg with the first call leg; andforwarding the call to an enterprise gateway server communicatively coupled to the carrier gateway server;wherein the call line preference is maintained for a predetermined period of time;wherein the call line preference and the predetermined period of time are stored in a subscriber profile which is accessed automatically based on at least one subscriber action.2. The method of claim 1 , further comprising selecting a primary line assigned to the subscriber to route the call.3. The method of claim 1 , further comprising:receiving additional outgoing call digits from a subsequent outgoing call from the subscriber; andforwarding the outgoing call on a default primary line or the specified line associated with the call line preference.4. The method of claim 3 , wherein the additional outgoing call digits do not specify a call line preference.5. The method of claim 1 , further comprising resetting the specified call line to a default call line once the predetermined ...

CALL CLASSIFICATION THROUGH ANALYSIS OF DTMF EVENTS

Systems, methods, and computer-readable media for call classification and for training a model for call classification, an example method comprising: receiving DTMF information from a plurality of calls; determining, for each of the calls, a feature vector including statistics based on DTMF information such as DTMF residual signal comprising channel noise and additive noise; training a model for classification; comparing a new call feature vector to the model; predicting a device type and geographic location based on the comparison of the new call feature vector to the model; classifying the call as spoofed or genuine; and authenticating a call or altering an IVR call flow. 1. A computer-implemented method to classify a call , the computer-implemented method comprising:receiving dual-tone multifrequency (DTMF) information from a call;determining a feature vector based on the DTMF information;comparing the feature vector to a model; andclassifying the call based on the comparison of the feature vector to the model.2. The computer-implemented method of claim 1 , further comprising:prompting, by an Interactive Voice Response (IVR) system, entry of the DTMF information.3. The computer-implemented method of claim 1 , further comprising:determining an ideal DTMF tone, wherein the determining the feature vector based on the DTMF information includes determining a feature based on the ideal DTMF tone.4. The computer-implemented method of claim 1 , further comprising:estimating channel noise in a DTMF tone, wherein the determining the feature vector based on the DTMF information includes determining a feature based on the channel noise.5. The computer-implemented method of claim 1 , further comprising:estimating additive noise in a DTMF tone, wherein the determining the feature vector based on the DTMF information includes determining a feature based on the additive noise.6. The computer-implemented method of claim 1 , wherein the feature vector is a vector of features ...

ENHANCED COMMUNICATION PLATFORM AND RELATED COMMUNICATION METHOD USING THE PLATFORM

Pre-authorized communication services and/or transactions are provided via a plurality of networks in response to a request received from a user to provide at least one of a communication service, a transaction and user account information via a plurality of networks of different types. Prior to processing the request, there is verification of the user's authorization to receive the at least one of the communication service, the transaction, and the user account information, and that an account associated with the user has a sufficient amount currently available for payment of the at least one of the communication service and the transaction. After verification, an authorized account associated with the user is charged in real time as the at least one of the communication service and the transaction is provided. 1. At least one non-transitory computer readable medium encoded with processing instructions performed by at least one computer platform performing a method of providing authorized communication services or transactions using a communication network , including at least networks of different types which are external to the platform , user devices communicating with the platform via the communication network , the method comprising:accepting and processing, by the platform, a request to provide at least one of a communication service, a transaction or user account information via one of the external networks;verifying at the platform that the requestor is authorized to receive the at least one of the communication service, the transaction, or the user account information, and that an account associated with the user has a sufficient amount currently available for payment of the at least one of the communication service or the transaction; andcharging at the platform by logging transactions on a real-time basis to an authorized account associated with the user, while the platform controls an element of a corresponding one of the external networks to provide at ...

SYSTEMS AND METHODS FOR TONE TO TOKEN TELECOMMUNICATIONS PLATFORM

Methods and systems are disclosed for enabling the generation of a token corresponding to a tone generated by a telephony system, comprising receiving one or more dual tone multi-frequency (DTMF) tones generated by a telephony system, generating a token based on the one or more DTMF tones; and transmitting the generated token to a merchant system. 120-. (canceled)21. A method for enabling the generation of a token corresponding to a tone generated by a telephony system , the method comprising:detecting, by a processor separate from a merchant system, a dual tone multi-frequency (DTMF) tone generated by a telephony system;intercepting, by the processor, the detected DTMF tone;tokenizing, by the processor, the intercepted DTMF tone to create a token; andprompting usage of the token to complete a transaction.22. The method of claim 21 , wherein the intercepted DTMF tone corresponds to cardholder data.23. The method of claim 21 , wherein the token comprises a low-value token.24. The method of claim 23 , further comprising:generating a high-value token based on the low-value token.25. The method of claim 24 , wherein the low-value token applies to one specific transaction and the high-value token applies to multiple transactions.26. The method of claim 24 , further comprising:prompting processing of a payment or transaction using the high-value token.27. The method of claim 21 , wherein the token is used for transactions including returns or price adjustments.28. A system for enabling the generation of a token corresponding to a tone generated by a telephony system claim 21 , the system comprising:a data storage device storing instructions for enabling the generation of a token corresponding to a tone generated by a telephony system; anda processor configured to execute the instructions to perform a method comprising:detecting, by a processor separate from a merchant system, a dual tone multi-frequency (DTMF) tone generated by a telephony system;intercepting, by the ...

CALL FLOW SYSTEM AND METHOD FOR USE IN A LEGACY TELECOMMUNICATION SYSTEM

A method of operating a network server, such as a mobile application gateway, connect devices on a cellular or carrier network with individual networks, such as enterprise voice and data networks or residential networks. The effects of the present invention are far reaching in terms of transferring effective call control from the cellular network into the control of the individual network, such as the enterprise, and enabling new business models for the purchase of cellular service from a public cellular carrier by an enterprise. 1. A method , comprising:providing, by an enterprise gateway server, a first routing information to a carrier gateway server, the first routing information comprising at least one routing instruction on how to route a session based on at least one policy;transmitting from the carrier gateway server to a mobile switching network, a second routing information, wherein the second routing information comprises a routing instruction informing the mobile switching network to route the session to the carrier gateway server; androuting, by the mobile switching network, the session through an enterprise network, via the carrier gateway server, to a destination.2. The method of comprising supporting claim 1 , by the enterprise gateway server claim 1 , session routing between a private branch exchange deployed in the enterprise network and a cellular network.3. The method of claim 2 , wherein the destination is the private branch exchange.4. The method of claim 1 , wherein the destination is the enterprise gateway server.5. The method of claim 2 , wherein the destination is the enterprise gateway server via the private branch exchange.6. The method of claim 2 , wherein the destination is the private branch exchange via the enterprise gateway.7. The method of claim 2 , wherein the enterprise gateway server routes to a final destination via the private branch exchange.8. The method of claim 1 , wherein the enterprise gateway server routes to a final ...

REMOTE DEVICE INTERFACE AND TELEPHONE ENTRY SYSTEM

Methods and systems for granting access to a building are disclosed. The system includes a telephone entry system associated with the building, a processor in communication with the telephone entry system, a non-transitory computer readable medium. The non-transitory computer-readable medium includes one or more programming instructions that when executed by the processor, cause the processor to receive a request including an access code for access to the building code, determine whether the access code is valid, and transmit to the telephone entry system associated with the building a sequence of dual-tone multi-frequency (DTMF) tones to cause the telephone entry system to grant access to the building in response to determining that the access code is valid. 1. A method for granting access to a building , the method comprising , by a processor:receiving a request for access to the building, wherein the request comprises an access code;determining whether the access code is valid; andin response to determining that the access code is valid, transmitting to a telephone entry system associated with the building a sequence of dual-tone multi-frequency (DTMF) tones to cause the telephone entry system to grant access to the building.2. The method of claim 1 , further comprising claim 1 , by the processor claim 1 , in response to determining that the access code is not valid claim 1 , denying access to the building.3. The method of claim 1 , wherein receiving the request for access to the building comprises receiving the request from a communication device associated with a user requesting access.4. The method of claim 2 , wherein the communication device is configured to transmit the request for access to the building to the processor upon occurrence of at least one of the following: receipt of a user request or detection of proximity to the building.5. The method of claim 1 , wherein receiving the request for access to the building comprises receiving the request from ...

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. 1. A method , comprising:receiving, by an enterprise gateway server, an instruction to redirect a call to a carrier network via a temporary routing number;routing, by the enterprise gateway server, the call to the carrier network via the temporary routing number; andnotifying, by the enterprise gateway server, a private branch exchange that received a request for the call, that a mobile station is ringing.2. The method of claim 1 , further comprising:transmitting, by the private branch exchange, a first call invite message to the enterprise gateway server;transmitting, by the private branch exchange, a second call invite message to a phone;forwarding, by the enterprise gateway server, the first call invite message to a carrier gateway server; andestablishing a secure tunnel between the enterprise gateway server and the carrier gateway server.3. The method of claim 2 , further comprising performing claim 2 , by the carrier gateway server claim 2 , a location request.4. The method of claim 3 , further comprising receiving claim 3 , by the enterprise gateway server claim 3 , the temporary routing number required to deliver the call to the mobile station.5. The method of claim 1 , further comprising receiving claim 1 , by the private branch exchange claim 1 , a call origination request of the call destined to a member having the mobile station and a phone communicably coupled to the private branch exchange.6. The method of claim 2 , further comprising forwarding claim 2 , by the carrier gateway server claim 2 , the temporary routing number to the enterprise gateway server.7. The method of claim 1 , further comprising receiving claim 1 , by the enterprise gateway server claim 1 , an indication that the mobile station is ringing.8. The method of claim 1 , further comprising inserting claim ...

SYSTEMS AND METHODS FOR TONE TO TOKEN TELECOMMUNICATIONS PLATFORM

Methods and systems are disclosed for enabling the generation of a token corresponding to a tone generated by a telephony system, comprising receiving one or more dual tone multi-frequency (DTMF) tones generated by a telephony system, generating a token based on the one or more DTMF tones; and transmitting the generated token to a merchant system. 120-. (canceled)21. A method for generating a token corresponding to sensitive data , the method comprising:receiving, by a processor separate from a merchant system, sensitive data from a user device via an interactive voice response system;tokenizing, by the processor, the received sensitive data to create a first token, wherein the first token applies to a specific transaction; andgenerating, by the processor, a second token based on the first token, wherein the second token applies to one or more transactions.22. The method of claim 21 , wherein the received sensitive data corresponds to cardholder data.23. The method of claim 21 , wherein the first token is valid for a predetermined period of time.24. The method of claim 21 , further comprising:prompting processing of a payment or transaction using the second token.25. The method of claim 21 , wherein the second token is used for transactions including returns or price adjustments.26. The method of claim 21 , further comprising:prompting suppression of sensitive data recording by the merchant system.27. The method of claim 21 , further comprising:generating or prompting a display corresponding to the received sensitive data.28. A system for generating a token corresponding to sensitive data claim 21 , the system comprising:a data storage device storing instructions for generating a token corresponding to sensitive data, the sensitive data generated by an interactive voice response system; anda processor configured to execute the instructions to perform a method comprising:receiving, by a processor separate from a merchant system, sensitive data from a user device via ...

CALL CLASSIFICATION THROUGH ANALYSIS OF DTMF EVENTS

Systems, methods, and computer-readable media for call classification and for training a model for call classification, an example method comprising: receiving DTMF information from a plurality of calls; determining, for each of the calls, a feature vector including statistics based on DTMF information such as DTMF residual signal comprising channel noise and additive noise; training a model for classification; comparing a new call feature vector to the model; predicting a device type and geographic location based on the comparison of the new call feature vector to the model; classifying the call as spoofed or genuine; and authenticating a call or altering an IVR call flow. 1. A computer-implemented method compromising:receiving, by a computer, a dual tone multi frequency (DTMF) tone associated with a phone call;generating, by the computer, an ideal DTMF tone corresponding to the received DTMF tone; andexecuting, by the computer, a machine learning model on the difference between the received DTMF tone and the ideal DTMF tone to classify the phone call.2. The computer-implemented method of claim 1 , further comprising:generating, by the computer, a feature vector based on the difference between the received DTMF tone and the ideal DTMF tone; and 'feeding, by the computer, the feature vector to the machine learning model.', 'wherein executing the machine learning model comprises3. The computer-implemented method of claim 1 , wherein the computer classifies the phone call as fraudulent or non-fraudulent.4. The computer-implemented method of claim 1 , wherein the received DTMF tone is received as an analog audio signal.5. The computer-implemented method of claim 1 , wherein the received DTMF tone is received as an audio packet.6. The computer-implemented method of claim 5 , wherein the audio packet is a Real-time Transport Protocol (RTP) audio packet.7. The computer-implemented method of claim 1 , further comprising:receiving, by the computer, a plurality of DTMF tones ...

TWO-WAY DUAL-TONE METHODS AND SYSTEMS FOR SYNCHRONIZING REMOTE MODULES

Methods and systems of correcting operation of multiple modules are described. The methods include generating a first dual-tone signal in a first module and transmitting the first dual-tone signal to a second module. A second dual-tone signal is generated in the second module and transmitted to the first module. The first module determines a first phase of the received second dual-tone signal and the second module determines a second phase of the received first dual-tone signal. Operation of one or both of the first and second modules is corrected based on the determined first phase and the determined second phase. 1. A method of correcting an operation of multiple modules , the method comprising:{'sub': 1', '2', '1, 'generating a first dual-tone signal using a first signal generator in a first module, the first dual-tone signal having a first frequency fand a second frequency fthat is different from the first frequency f;'}transmitting the first dual-tone signal from the first module to a second module;{'sub': 3', '4', '3, 'generating a second dual-tone signal using a second signal generator in the second module, the second dual-tone signal having a third frequency fand a fourth frequency fthat is different from the third frequency f;'}transmitting the second dual-tone signal from the second module to the first module;determining, at the first module, a first phase of the received second dual-tone signal;determining, at the second module, a second phase of the received first dual-tone signal; andcorrecting the operation of at least one of the first module and the second module based on the determined first phase and the determined second phase.2. The method of claim 1 , wherein correcting the operation comprises a phase correction.3. The method of claim 1 , further comprising:transmitting a first clock frequency from the first module to the second module; andmatching a second clock frequency of the second module with the transmitted first clock frequency.4. The ...

SYSTEM AND METHOD FOR SPEEDING CALL ORIGINATIONS TO A VARIETY OF DEVICES USING INTELLIGENT PREDICTIVE TECHNIQUES FOR HALF-CALL ROUTING

A mobile application gateway configured to interconnect mobile communication devices on a cellular network with an enterprise network is provided. The mobile application gateway includes a voice and data signaling gateway configured to provide routing functionalities, service functionalities and admission control. A gateway GPRS support node (GGSN) is configured to establish a secure data session between one or more of the mobile communication devices and the enterprise network by establishing a GPRS tunneling protocol (GTP) tunnel between a carrier-hosted serving GPRS support node (SGSN) and the GGSN. 1. A method , comprising: send a destination to an enterprise gateway server for a communication associated with destination routing information; and', 'set an enterprise network routing number, via the enterprise gateway server;, 'a carrier gateway server configured to accesses historical destination routing time data; and', 'use the historical destination routing time data, the destination routing information and the enterprise routing number to decide when to route a call leg towards the destination., 'wherein the enterprise gateway server is configured to2. The method of claim 1 , wherein the enterprise network routing number is stored in a mobile network home location register (HLR).3. The method of claim 1 , wherein the enterprise network routing number is associated with a mobile device.4. The method of comprising processing the call via at least one of a carrier gateway server (GS-C) and an enterprise gateway server (GS-E).5. The method of comprising automatically re-routing the call.6. The method of claim 3 , wherein the enterprise network routing number is a corresponding mobile network routing number associated with the mobile device operated by an enterprise user assigned to the enterprise network routing number.7. The method of claim 6 , wherein if the GS-E is not available to perform call routing functions then setting the enterprise network routing ...

ENHANCED COMMUNICATION PLATFORM AND RELATED COMMUNICATION METHOD USING THE PLATFORM

Pre-authorized communication services and/or transactions are provided via a plurality of networks in response to a request received from a user to provide at least one of a communication service, a transaction and user account information via a plurality of networks of different types. Prior to processing the request, there is verification of the user's authorization to receive the at least one of the communication service, the transaction, and the user account information, and that an account associated with the user has a sufficient amount currently available for payment of the at least one of the communication service and the transaction. After verification, an authorized account associated with the user is charged in real time as the at least one of the communication service and the transaction is provided. 1. At least one non-transitory computer readable medium encoded with processing instructions performed by at least one computer platform performing a method of providing authorized communication services or transactions using a communication network , including external networks of different types which are external to the at least one computer platform , user devices communicating with the at least one computer platform via the communication network , the method comprising:accepting and processing, by the at least one computer platform, a request from a user to provide at least one of a communication service, a transaction, or user account information via one of the external networks;verifying at the at least one computer platform that the user is authorized to receive the at least one of the communication service, the transaction, or the user account information, and that an account associated with the user has a sufficient value currently available for payment of the at least one of the communication service or the transaction; andcharging at the at least one computer platform by logging transactions on a real-time basis to an authorized account ...

SYSTEMS AND METHODS FOR INTERACTING WITH ANSWERING SYSTEMS

Systems, methods, and non-transitory computer-readable media can cause a voice communication between a computing device and an automated answering system to be established. A set of options that are available for selection through the automated answering system during the voice communication can be determined. The set of options can be provided for presentation on a display screen of the computing device, wherein at least some of the options in the set are labeled with information that visually describes the options. 1. A computer-implemented method comprising:causing, by a computing device, a voice communication between the computing device and an automated answering system to be established;determining, by the computing device, a set of options that are available for selection through the automated answering system during the voice communication; andproviding, by the computing device, the set of options for presentation on a display screen of the computing device, wherein at least some of the options in the set are labeled with information that visually describes the options.2. The computer-implemented method of claim 1 , wherein determining the set of options that are available for selection through the automated answering system further comprises:obtaining, by the computing device, information describing an automated menu used by the automated answering system; andcausing, by the computing device, the set of options to be labeled based at least in part on the information.3. The computer-implemented method of claim 1 , wherein determining the set of options that are available for selection through the automated answering system further comprises:determining, by the computing device, the set of options by analyzing audio being communicated by the automated answering system during the voice communication, wherein the audio is analyzed using one or more speech recognition techniques; andcausing, by the computing device, the set of options to be labeled based at ...

System and method for enabling vpn-less session setup for connecting mobile data devices to an enterprise data network

A mobile application gateway configured to interconnect mobile communication devices on a cellular network with an enterprise network is provided. The mobile application gateway includes a voice and data signaling gateway configured to provide routing functionalities, service functionalities and admission control. A gateway GPRS support node (GGSN) is configured to establish a secure data session between one or more of the mobile communication devices and the enterprise network by establishing a GPRS tunneling protocol (GTP) tunnel between a carrier-hosted serving GPRS support node (SGSN) and the GGSN.

Isdn exchanger's automatic response function

1. 청구범위에 기재된 발명이 속한 기술분야 1. TECHNICAL FIELD OF THE INVENTION 피호출측 가입자가 부재중인 경우에 호출측 가입자에게 피호출측 가입자가 부재중임을 안내하는 방법에 관한 것이다. The present invention relates to a method of informing the calling party that the called party is absent when the called party is absent. 2. 발명이 해결하려고 하는 기술적 과제 2. The technical problem to be solved by the invention 국설교환기에 접속된 호출측 가입자의 전화기상에 디지탈국선을 수용하는 종합정보통신망(ISDN)교환기에 접속된 피호출측 가입자가 부재중임을 나태내는 안내메시지가 표시되도록 한다. An indication message is displayed on the telephone of the calling party subscriber connected to the local exchange, indicating that the called party subscriber connected to the ISDN exchange receiving digital trunk line is absent. 3. 발명의 해결방법의 요지 3. Summary of Solution to Invention 본 발명은 디지탈국선을 통해 국설교환기에 접속되며, 액정표시기(LCD)와 다수의 기능버튼을 가지고 있는 디지탈전화기를 내선으로 접속하고 있는 ISDN교환기에서의 부재중 안내방법을 제공한다. The present invention provides an out-of-office guidance method in an ISDN exchanger connected to a local exchange via a digital trunk line and connecting a digital telephone having a liquid crystal display (LCD) and a plurality of function buttons to the extension. 본 발명에 따르면, ISDN교환기의 어느 한 내선이 호출될 시 이 호출된 내선이 부재중 안내모드로 설정되어 있는지 여부를 확인한 후 확인된 경우 상기 호출된 내선을 호출한 호출측으로 소정의 안내메시지를 송신하여 호출측 디지탈전화기의 LCD상에 상기 송신된 안내메시지가 디스플레이되도록 하여 피호출측 가입자가 부재중임을 안내한다. 상기 ISDN교환기는 가입자의 각종 상황에 따른 부재이유를 나타내는 다수의 안내메시지를 저장하고 있는 메모리를 더 포함하여 구성된다. 이에 따라 가입자는 부재하기 전에 메모리에 저장되어 있는 다수의 안내메시지중 부재이유를 나타내는 안내메시지를 미리 설정할 수 있다. 부재이유를 나타내는 안내메시지가 설정되고 부재중 안내모드로 설정된 디지탈전화기가 호출되면, ISDN교환기는 메모리에 저장되어 있는 다수의 안내메시지중 미리 설정된 안내메시지를 메모리로부터 읽어들인다. 그러한 후에 호출측으로 상기 읽어들인 안내메시지를 송신하여 호출측 디지탈전화기의 LCD상에 피호출 가입자의 부재이유가 디스플레이되도록 한다. According to the present invention, when an extension of an ISDN exchange is called, it is checked whether the called extension is set to the missed announcement mode, and if confirmed, by sending a predetermined announcement message to the calling party calling the called extension. The transmitted guidance message is displayed on the LCD of the calling digital telephone to ...

从综合业务数字网交换系统发送留言的方法

一种将ISDN交换系统的留言发送到与该系统连接的数字电话机的方法。当ISDN交换系统的分机用户被呼叫时，该系统检查被呼叫的分机用户是否被置为留言发送方式，如果是，则将存储的留言发送到主叫用户，以便显示在主叫用户数字电话机的LCD屏幕上。电话用户可在离开之前在多个代表其离开原因的留言中选择一个。因此，当该用户的数字电话机被呼叫时，ISDN交换系统从存储器中检索被选择的留言，并将该留言传送并显示在主叫用户数字电话机的LCD屏幕上。

Dialing method for long distance call, long distance telephone service station and carrier selection promotion method

(57)【要約】 【課題】 柔軟性のある長距離系キャリアの選択ができ る長距離通話の発呼方法等を提供する。 【解決手段】 長距離電話サービス局は、長距離通話に 対して長距離系キャリアを選択する補助したり、料金表 およびサービス・ロジックを備える。また、料金表メモ リには、様々な長距離系キャリヤが加入者に課す料金を 示す料金データ、加入者が好む長距離系キャリヤの１つ を示す加入者嗜好データを格納する。この料金データの 変更は定期的に検索され、料金表メモリに格納された り、加入者のローカル装置にダウンロードされる。

Apparatus and method for storing/calling telephone number in a mobile station

본 발명은 이동 단말기에서 통화 중에 입력/수신하는 전화번호를 최근번호로 등록하고, 통화 중 또는 통화 종료 후에 상기 최근번호를 이용하여 즉시 발신이 가능하도록 하는 장치 및 방법에 관한 것으로서, 이러한 본 발명은 이동 단말기에서 입력 또는 수신되는 전화번호를 이용한 통화연결 방법에 있어서, 이동 단말기 사용자 간에 호가 설정된 상태에서 소정 입력 정보의 발생여부를 체크하는 과정과, 상기 입력 정보가 발생하면 상기 입력 정보를 인식하여 전화번호를 추출하는 과정과, 상기 전화번호 추출 후 통화연결이 요청되면, 상기 전화번호에 상응하는 이동 단말기로 3자 통화 연결하는 과정과, 상기 전화번호 추출 후 통화연결이 종료되면, 상기 전화번호에 상응하는 이동 단말기로 통화 연결하는 과정을 포함한다. The present invention relates to an apparatus and method for registering a telephone number input / received during a call in a mobile terminal as a recent number, and enabling an immediate call using the latest number during or after a call. A call connection method using a phone number input or received by a mobile terminal, the method comprising: checking whether generation of predetermined input information occurs when a call is set between mobile terminal users, and when the input information occurs, recognizes the input information and makes a phone call. Extracting a number; and if a call connection is requested after extracting the phone number; connecting a three-way call to a mobile terminal corresponding to the phone number; and terminating the call connection after extracting the phone number; And connecting the call to the corresponding mobile terminal. 이동 단말기, 전화번호, 최근번호, 3자 통화, DTMF, 음성인식 Mobile terminal, phone number, recent number, 3-way call, DTMF, voice recognition

Dtmf signal detecting system

The circuit and method detect the dual tone multi-frequency (DTMF) signal without errors. The circuit includes a DTMF signal detector (20) for generating the parallel dial data by decoding input DTMF signal and for generating strobe signals, a same repeating data detector (40) for detecting the same repeating dial data, and a CPU (30) for analyzing input dial data generated by the DTMF signal detector and the same repeating data detector (40).

Character Information Service Method through Integrated Information Communication Network

본 발명은 종합정보통신망을 가지는 사설교환기에서 가입자의 요청으로 메뉴에 따른 문자 정보 서비스를 등록한 후 일정 시간 간격/필요시/항시 문자 정보를 수신하여 해당 단말기로 제공함으로서 해당 단말기에 문자 정보가 디스플레이 되도록 하는 종합정보통신망을 통한 문자정보 서비스 방법을 구현함으로서 다양한 부가 서비스를 제공할 수 있으며, 점차적인 서비스 추가 개선이 용이하다. The present invention registers a text information service according to a menu at the request of a subscriber in a private exchange having a comprehensive information communication network, and then receives text information at a predetermined time interval / if necessary / always and provides the text information to the corresponding terminal to display the text information on the corresponding terminal. By implementing a text information service method through a comprehensive information and communication network can provide a variety of additional services, it is easy to further improve the service.

How to guide the absence of an integrated information network exchange

1. 청구범위에 기재된 발명이 속한 기술분야 피호출측 가입자가 부재중인 경우에 호출측 가입자에게 피호출측 가입자가 부재중임을 안내하는 방법에 관한 것이다. 2. 발명이 해결하려고 하는 기술적 과제 국설교환기에 접속된 호출측 가입자의 전화기상에 디지탈국선을 수용하는 종합정보통신망(ISDN)교환기에 접속된 피호출측 가입자가 부재중임을 나타내는 안내메시지가 표시되도록 한다. 3. 발명의 해결방법의 요지 본 발명은 디지탈국선을 통해 국설교환기에 접속되며, 액정표시기(LCD)와 다수의 기능버튼을 가지고 있는 디지탈전화기를 내선으로 접속하고 있는 ISDN교환기에서의 부재중 안내방법을 제공한다. 본 발명에 따르면, ISDN교환기의 어느 한 내선이 호출될 시 이 호출된 내선이 부재중 안내모드로 설정되어 있는지 여부를 확인한 후 확인된 경우 상기 호출된 내선을 호출한 호출측으로 소정의 안내메시지를 송신하여 호출측 디지탈전화기의 LCD상에 상기 송신된 안내메시지가 디스플레이되도록 하여 피호출측 가입자가 부재중임을 안내한다. 상기 ISDN교환기는 가입자의 각종 상황에 따른 부재이유를 나타내는 다수의 안내메시지를 저장하고 있는 메모리를 더 포함하여 구성된다. 이에 따라 가입자는 부재하기 전에 메모리에 저장되어 있는 다수의 안내메시지중 부재 이유를 나타내는 안내메시지를 미리 설정할 수 있다. 부재이유를 나타내는 안내메시지가 설정되고 부재중 안내모드로 설정된 디지탈전화기가 호출되면, ISDN교환기는 메모리에 저장되어 있는 다수의 안내메시지중 미리 설정된 안내메시지를 메모리로부터 읽어들인다. 그러한 후에 호출측으로 상기 읽어들인 안내메시지를 송신하여 호출측 디지탈 전화기의 LCD상에 피호출 가입자의 부재이유가 디스플레이되도록 한다. 4. 발명의 중요한 용도 종합정보통신망(ISDN)의 디지탈 전화기

Multimodal telephone numbers

Data communication between a data sender and at least one data recipient is facilitated when the data recipient is associated with a telephone number that can not be used to access the user with a telephone call. An invention receives the telephone number associated with the data destination (data recipient) and receives an indication that data is to be communicated. The invention uses the telephone number to determine at least one characteristic of data destination/recipient. The invention also receives at least one characteristic of the data to be communicated that enable the data to be communicated to the data receiver/recipient.

Communications webs with personal communications links for PSTN subscribers

Systems for connecting telecommunications infrastructure lines to telephones, handsets, computers, telecopy machines and other end user interfaces or consumer electronics devices in a residence or business. Systems according to the present invention include Network Control Units which form the center of a star topology and which communicate via RF link with Wireless Access Units and handsets. Wireless Access Units feature an interface, such as, for example, a standard telephone jack, for accommodating a telephone, a fax machine, a compute modem or other device. Computers or other devices may also be accommodated by Wireless Access Units having other physical and virtual interfaces, including, for instance, serial ports or network interfaces. The Wireless Access Units may also be digital to accommodate ISDN or any other digital standard. Wireless control/monitoring accessories may also be employed to communicate with the Network Control Unit and provide additional functionality such as entrance monitoring, baby monitoring, HVAC control and other services. Personal Communication Links provide wireless handset and speakerphone services. The Personal Communication Links include independent talk/listen paths from the unit to the Network Control Unit. In addition, each Personal Communication Link includes circuitry for automatically adjusting from high-volume/high-microphone sensitivity for speakerphone operation to low-volume/low-microphone sensitivity for conventional handset operation.

Spectrum monitoring for PSTN subscribers

Systems for connecting PSTN lines to telephones, handsets, computers, telecopy machines and other end user interfaces or consumer electronics devices in a residence or business. Systems according to the present invention include Network Control Units which form the center of a star topology and which communicate via RF link with Wireless Access Units and handsets. Wireless Access Units feature an interface, such as, for example, a standard telephone jack, for accommodating a telephone, a fax machine, a compute modem or other device. Computers or other devices may also be accommodated by Wireless Access Units having other physical and virtual interfaces, including, for instance, serial ports or network interfaces. The Wireless Access Units may also be digital to accommodate ISDN for any other digital standard. Wireless control/monitoring accessories may also be employed to communicate with the Network Control Unit and provide additional functionality such as entrance monitoring, baby monitoring, HVAC control and other services. Such systems may monitor various bands of interest, process, store and change their operation according to information associated with signals or noise in such bands, and, if desired, report such information upstream to management nodes for various purposes.

Method for setting inter-office signalling system in the switching system

본 발명은 교환기가 복수의 국간신호방식을 동시에 수용할 수 있도록 국간신호운용체계를 설정하기 위한 교환기에 있어서 국간신호운용체계 설정방법에 관한 것으로, 본 방법은 교환기내에 구비되어 있는 라인프로세서들의 상태가 정상인지 비정상인지를 조사하고;조사한 결과, 라인프로세서들이 정상이면, 각 라인프로세서들의 라인타입을 확인하고;확인된 라인타입을 운용자에게 통보하고;통보 후, 운용자로부터 라인프로세서들에 대한 라인타입 변경명령이 인가되면, 변경하고자 하는 라인프로세서들을 관할하고 있는 메인프로세서의 출중계, 입중계 및 루트에서 트렁크상태를 확인하고;확인한 결과, 트렁크가 설정되어 있으면, 트렁크를 지우고, 각 라인프로세서들의 라인타입을 변경하고;확인한 결과, 트렁크가 설정되어 있지 않으면, 바로 라인프로세서들의 라인타입을 변경하고; 메인프로세서의 루트데이타, 트렁크 및 프리픽스 등을 생성하는 단계를 포함하여 수행된다.

CONTROL OF THE SPECTRUM FOR SUBSCRIBERS OF THE SWITCHED TELEPHONE NETWORK OF GENERAL USE

2000132201 А ко РОССИЙСКАЯ ФЕДЕРАЦИЯ ФЕДЕРАЛЬНАЯ СЛУЖБА (19) 22% РО (11) ко м“ АЦ (51) М’ Н 04 @ 11/04 х < \ < А \ у А < м 7 ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ, ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ 12 ИЗВЕЩЕНИЯ К ЗАЯВКЕ Н А ИЗОБРЕТЕНИЕ (21), (22) Заявка: 2000132201/09, 19.05.1999 (30) Конвенционный приоритет: 22.05.1998 Ц$ 09/083,464 (43) Дата публикации заявки: 27.11.2002 (85) Дата перевода заявки РСТ на национальную фазу: 22.12.2000 (86) Заявка РСТ: 0$ 99/11079 (19.05.1999) (87) Публикация РСТ: М/О 99/62290 (02.12.1999) Адрес для переписки: 129010, Москва, ул. Б.Спасская 25, стр.3, "Юридическая фирма Городисский и Партнеры", Егоровой Г.Б. (71) Заявитель(и): ХОУМ ВАЙРЛЕСС НЕТВОРКС, ИНК. (Ц$) (72) Автор(ы): СНЕЛЛИНГ Ричард К. (1$), МакИНТОШ П. Стаки (Ц$), ТЭЙЛОР Джон С. В. (Ц$)}, ТАКЕР Марк (1$) (74) Патентный поверенный: Егорова Галина Борисовна 000 (54) КОНТРОЛЬ СПЕКТРА ДЛЯ АБОНЕНТОВ КОММУТИРУЕМОЙ ТЕЛЕФОННОЙ СЕТИ ОБЩЕГО ПОЛЬЗОВАНИЯ ГАЗА - Признание заявки на изобретение отозванной в связи с непредставлением в установленный срок ходатайства о проведении экспертизы заявки по существу Дата, с которой заявка признана отозванной: 27.05.2008 Извещение опубликовано: 10.07.2008 БИ: 19/2008 Страница: 1 па 0$ 1000с У

Telecommunication commutator, which has programmable network protocols, method for its functioning, and method for development of programmable network protocols

FIELD: telecommunications, computer engineering. SUBSTANCE: programmable telecommunication commutator provides environment for development of signal transmission protocols. The environment provides possibility to process separate finite state automaton for each port. Protocols for various telecommunication applications and different levels of software may be developed. EFFECT: dynamic alteration of system protocol, facilitated development and application of individual system protocols. 21 cl, 21 dwg Гросс ПЧ ГЭ РОССИЙСКОЕ АГЕНТСТВО ПО ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ (19) (51) МПК? ВИ” 2 150 791 ‘' 13) С1 Н 04 29/06, Н 04 М 3/42 12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ РОССИЙСКОЙ ФЕДЕРАЦИИ (21), (22) Заявка: 96108957/09, 05.10.1994 (24) Дата начала действия патента: 05.10.1994 (30) Приоритет: 08.10.1993 1$ 08/134,122 (46) Дата публикации: 10.06.2000 (56) Ссылки: ЕР 0555997 А2, 18.08.1993. ЦЗ 5007080 А, 09.04.1991. 4$ 4980906 А, 25.12.1990. Ц$ 4221933 А, 09.09.1980. $4 1670801 АЛ, 15.08.1991. (85) Дата перевода заявки РСТ на национальную фазу: 05.05.1996 (86) Заявка РСТ: 1$ 94/11319 (05.10.1994) (87) Публикация РСТ: М/О 95/10912 (20.04.1995) (98) Адрес для переписки: 191186, Санкт-Петербург, а/я 230, "АРС-ПАТЕНТ", Рыбакову В.М. (71) Заявитель: ЭКСЕЛ СВИТЧИНГ КОРПОРЭЙШН (45$) (72) Изобретатель: Хеберт Марк П. (ЦЗ) (73) Патентообладатель: ЭКСЕЛ СВИТЧИНГ КОРПОРЭЙШН (453) (54) КОММУТАТОР ТЕЛЕКОММУНИКАЦИОННЫХ СИСТЕМ, СОДЕРЖАЩИЙ ПРОГРАММИРУЕМЫЕ СЕТЕВЫЕ ПРОТОКОЛЫ, СПОСОБ ФУНКЦИОНИРОВАНИЯ ЭТОГО КОММУТАТОРА И СПОСОБ РАЗРАБОТКИ ПРОГРАММИРУЕМЫХ СЕТЕВЫХ ПРОТОКОЛОВ (57) Реферат: Устройство ОТНОСИТСЯ К телекоммуникационным и в особенности к программируемым коммутаторам телекоммуникационных систем и управляемым внешним — компьютерам, поддерживающим различные телекоммуникационные приложения. Технический результат - обеспечение динамического — изменения — протокола сигнализации, возможность легкой разработки и внедрения индивидуальных протоколов сигнализации для приложений. Сущность изобретения ...

Telephone service implementation system, automatic branch exchange controller and CTI server

(57)【要約】 １以上の電話装置上において電話サービスを実施するシステムであって、該電話装置は１以上の私用自動枝交換器によって管理される数個の電話端末を有している。このためにシステムはコンピューターネットワークによりアクセスできるサーバータイプコンピューター設備と、１以上の中継器とを有している。該中継器はサービスまたは要請されたサービスに対応する制御信号を送ることができ、これにより１以上の自動枝交換器を駆動する。これらの中継器はこの目的のために自動枝交換器の電話ラインに接続されている。

Apparatus for generating dual tone- multi- frequency/call in private branch exchange

본 발명은 제어 신호를 발생하는 제어부 인터페이스부, 클럭 정보를 제공하는 클럭 인터페이스부, 호출자로부터 호출 번호가 수신되면, 국선 회로부에 호출 번호를 송출하기 위한 톤 데이터를 발생하고, 상기 클럭 인터페이스부로부터 전송된 클럭 정보를 이용하여 상기 발생된 톤 데이터의 동기를 맞추고, 상기 제어부 인터페이스부로부터 전송된 제어 신호를 이용하여 국선 회로부의 해당되는 PCM 번지를 찾아서 해당 국선의 버퍼부에 상기 톤 데이터를 저장한 후, 수신 하이웨이로 송출하는 스위치 회로부로 구성된 것으로서, DTMF/호출음 발생부가 스위치 회로부내에 있어서, 별도의 32회선 가입자 송신 하이웨이 리소스가 필요없게 되어 송신 하이웨이는 교환기의 가입자를 위해 사용할 수 있다. The present invention provides a control unit interface unit for generating a control signal, a clock interface unit for providing clock information, and when a call number is received from the caller, generates tone data for transmitting the call number to the trunk line circuit unit, and transmits the call number from the clock interface unit. Synchronizes the generated tone data using the generated clock information, finds a corresponding PCM address of a trunk line circuit unit by using a control signal transmitted from the controller interface unit, and stores the tone data in a buffer unit of the trunk line. In this case, the DTMF / calling sound generator is included in the switch circuit section so that a separate 32-line subscriber transmission highway resource is not required, and thus the transmission highway can be used for the subscriber of the exchange.

COMMUNICATION NETWORK FOR PSTN SUBSCRIBERS

2000132215 А ко РОССИЙСКАЯ ФЕДЕРАЦИЯ ФЕДЕРАЛЬНАЯ СЛУЖБА (19) 2 (11) за \ , (13) АА м“ \ № мах ЮО 3 & < 5 А {$ ХУ Ка <» (51) МК’ [04 М 1/72 ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ, ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ 12 ИЗВЕЩЕНИЯ К ЗАЯВКЕ Н А ИЗОБРЕТЕНИЕ (21), (22) Заявка: 2000132215/09, 19.05.1999 (30) Конвенционный приоритет: 22.05.1998 Ц$ 09/083,726 (43) Дата публикации заявки: 27.11.2002 (85) Дата перевода заявки РСТ на национальную фазу: 22.12.2000 (86) Заявка РСТ: 0$ 99/11078 (19.05.1999) (87) Публикация РСТ: М/О 99/62241 (02.12.1999) Адрес для переписки: 129010, Москва, ул. Б.Спасская 25, стр.3, "Юридическая фирма Городисский и Партнеры", Егоровой Г.Б. (71) Заявитель(и): ХОУМ ВАЙРЛЕСС НЕТВОРКС, ИНК. (Ц$) (72) Автор(ы): СНЕЛЛИНГ Ричард К. (1$), МакИНТОШ П. Стаки (Ц$), ТЭЙЛОР Джон С. В. (Ц$)}, ТАКЕР Марк (1$) (74) Патентный поверенный: Егорова Галина Борисовна 000 (54) СЕТЬ СВЯЗИ ДЛЯ АБОНЕНТОВ ТСОП ГАЭА - Признание заявки на изобретение отозванной в связи с непредставлением в установленный срок ходатайства о проведении экспертизы заявки по существу Дата, с которой заявка признана отозванной: 26.05.2008 Извещение опубликовано: 10.07.2008 БИ: 19/2008 Страница: 1 па 91$ 1000Сс У

Private telephone network telephone service provision having terminal central switching circuit routing and interfaces server driven transmitting command signaling services providing and telephone line connected.

The telephone service provision system has several telephone terminals and a central private switching circuit (20). The information equipment is a server accessible from an information network from several interfaces (300) which transmit command signaling corresponding to services and driving one of several switching circuits. The interfaces are connected to the telephone line (L1,Ln) switching circuits.

TELEPHONE, MULTIMEDIA TERMINAL AND SERVER

L'invention concerne un poste téléphonique associé à un terminal multimédia.Ce poste (PT1) comprend des moyens pour commander l'émission, vers un serveur (SERV) comportant une base de données (BD) dans laquelle sont stockés des identifiants de postes téléphoniques et des identifiants des terminaux multimédias (TM1, TM2) associés à ces postes téléphoniques, d'une requête sous la forme d'un message comprenant un identifiant de ce premier poste téléphonique et un identifiant d'un second poste téléphonique (PT2) avec lequel le premier poste est en cours de communication téléphonique.Le serveur comporte des moyens pour commander l'émission, vers un terminal multimédia associé au premier et/ou au second poste téléphonique, d'un identifiant de l'autre terminal, afin d'établir une communication multimédia, à travers une voie logique distincte de la voie de la communication téléphonique, entre ces terminaux multimédias. The invention relates to a telephone set associated with a multimedia terminal. This set (PT1) comprises means for controlling transmission to a server (SERV) comprising a database (BD) in which identifiers of telephone sets are stored. and identifiers of the multimedia terminals (TM1, TM2) associated with these telephone sets, a request in the form of a message comprising an identifier of this first telephone set and an identifier of a second telephone set (PT2) with which the first station is in the process of a telephone call. The server comprises means for controlling the transmission, to a multimedia terminal associated with the first and / or the second telephone station, of an identifier of the other terminal, in order to establish multimedia communication, through a logical channel separate from the telephone communication channel, between these multimedia terminals.

METHOD FOR PROVIDING SERVICES TO SUBSCRIBERS OF A TELEPHONE NETWORK

Telecommunication system which is controlled by computer

FIELD: communications. SUBSTANCE: device has central control computer, commutator, two data memory units, program memory unit, groups of system and peripheral interface devices for voice and data transmission. Commutator is designed for simplex voice transmission through four-wire communication line. Commutator is connected to system and peripheral interface devices using two wires. EFFECT: increased functional capabilities. 5 cl, 6 dwg зЗоо0эО0гс пы Го (19) РОССИЙСКОЕ АГЕНТСТВО ПО ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ ВИ “” 2 106 008 ' С 06 Е 15/00, 15/16 (51) МПК 13) СЛ 12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ РОССИЙСКОЙ ФЕДЕРАЦИИ (21), (22) Заявка: 93004619/09, 25.03.1993 (30) Приоритет: 25.03.1992 ОЕ Р 4210167.0 (46) Дата публикации: 27.02.1998 (56) Ссылки: 1. ОЕ, пат ент, 3923960, кл. Н 04 @ 3/54, 1989. 2. ОЕ, патент, 3442144, кл. Н 03 М 100, 1986. (71) Заявитель: ДеТеВе-Дойче Телефонверке Акциенгезельшафт унд Ко. (ОЕ) (72) Изобретатель: Рольф Баумайстер[ПЕ], Уве Думманн[ОЕ], Клаус-Петер Эрзель:[ОЕ], Хорст Йэкелы[ОЕ], Флориан Янковски[ОЕ], Гюнтер Кернер[ОЕ] (73) Патентообладатель: ДеТеВе-Дойче Телефонверке Акциенгезельшафт унд Ко. (ОЕ) (54) УПРАВЛЯЕМАЯ ОТ ЭВМ ТЕЛЕКОММУНИКАЦИОННАЯ УСТАНОВКА (57) Реферат: Изобретение касается управляемой от ЭВМ телекоммуникационной — установки, содержащий центральную управляющую ЭВМ, блок коммутации, два блока памяти данных, блок памяти программ, группы системных и —внесистемных устройств сопряжения для передачи речи и данных, причем блок коммутации выполнен с возможностью однонаправленной передачи речи через четырехпроводный канал связи, соединение блока коммутации с системными и внесистемными устройствами сопряжения выполнено двухпроводным. 4 з.п. ф-лы, 6 ил. ^, 1 м... ооьаввьь о > кк -> жк Г <- Ха. „. кк --+ Ка я а ж----------д--+ 4-4 ---- [_* Е 111111 Регуривхде [$35] {У 1 [4 .||зех42}1 раха11е1 5ег1е1}. раха].1е1 фиг.1 2106008 С1 КО зЗоо0эО0гс пы Го 9) тщу"” 2 406 008 “”С4 51) мЕ С° < 06 Е 15/00, 15/16 КУЗЗАМ АСЕМСУ ГОК РАТЕМТ$ АМО ТКАОЕМАКК ...

INTERFACE DEVICE AND METHOD FOR IMPLEMENTING AN INTERFACE IN PARTICULAR BETWEEN AN ANALOGUE LINK AND A DIGITAL LINK

Ce dispositif d'interface (10) est disposé entre une liaison analogique (a2 ) de télécomunications et une liaison numérique (d2 ) de télécommunications. Des codes de numérotation analogiques sur la liaison analogique (a2 ) sont convertis en codes de commande destinés à commander une interface numérique (12) de télécommunications, fixée à la liaison numérique (a2 ), pour assurer un accès à un réseau numérique de télécommunications (16, 18, 20). L'accès à la liaison (a2 ) de télécommunication est activé sélectivement par le dispositif d'interface (10) en réponse à un mot de passe codé sous forme d'une séquence de numérotation analogique sur la liaison analogique (a2 ). Le dispositif d'interface (10) retarde l'établissement de l'appel à l'aide d'un modem (6) connecté aux liaisons analogiques (a1 , a2 ) pour synchroniser la fin de l'établissement de l'appel avec une procédure d'établissement d'appel d'un liaison distante connectée au réseau (16, 18, 20). This interface device (10) is arranged between an analog telecommunications link (a2) and a digital telecommunications link (d2). Analog dial codes on the analog link (a2) are converted into control codes for controlling a digital telecommunications interface (12), attached to the digital link (a2), to provide access to a digital telecommunications network ( 16, 18, 20). Access to the telecommunications link (a2) is selectively activated by the interface device (10) in response to a password encoded as an analog dial sequence on the analog link (a2). The interface device (10) delays the establishment of the call using a modem (6) connected to the analog links (a1, a2) to synchronize the end of the establishment of the call with a call establishment procedure of a remote link connected to the network (16, 18, 20).

Dual frequency multiplexing device for use in repeater, has branch forming access port of access point, and each of set of other branches is connected to access point of one of band pass filters of set of band pass filters

The device (50) has four band pass filters (611-614) that are centered respectively on four frequencies (F1-F4). Each band pass filter is arranged with an access point and another access point. Each access point is connected to an access point of one band pass filter of the four band pass filters. A branch forms an access port of the latter access point, and each of a set of other branches is connected to the latter access point of one of the band pass filters of the set of band pass filters.

Directed telephone system

The Directed Telephone System (DTS) enhances the operation of group of telephones connected to a Central Office line. The system sends a brief pre-recorded message asking the caller to press one digit out of a menu of names and corresponding digits. On receiving that digit DTS directs the incoming call to a particular telephone set and alerts only the called person by a unique ring code. The DTS provides personalised intercom calls using the existing normal telephones and alert the called person with the same ring code. The system also connects a fax call directly to the fax machine. A DTS unit can be constructed on a circuit board that can be fitted inside each telephone set during manufacturing or in a little box connected to each telephone. Suitable for multiple personnel homes, rented room dwellings, farms, small businesses, etc..

Program controller

(57)【要約】 （修正有） 【課題】 同一市外局番を有する複数の電話機に順次電 話をかける場合の、ダイヤルボタン操作の手間の低減 と、交換接続時間の短縮を実現する交換機及びその交換 機を含む電話交換網を提供する。 【解決手段】 各交換機に、加入者電話機からダイヤル された“市外局番＋＊”を認識する手段（Ｓ４０２，Ｓ ４０３）を設け、その市外局番に対応する市外交換局ま でのルーティング実現する。市外交換機は、自局宛ての “市外局番＋＊”を受信すると第二発信音を送出し（Ｓ ４０５）、その後受信するダイヤル信号に応じて交換接 続を行う（Ｓ４０６〜Ｓ４０８）。また、各交換機は、 加入者電話機からのオフフック信号によっても直ちに、 終話処理を行わず、所定時間が経過した後（Ｓ４１ ２）、または所定時間内にオフフック及びオンフックを 検出した場合に、終話処理を行う。

Convergent communications platform and method for mobile and electronic commerce in a heterogeneous network environment

A method and apparatus for providing mobile and electronic commerce, customer care and communication services via networks, including receiving in a roaming network an identification number and a request for a service, forwarding to a home network the identification number, the request for the service, and a cost/rate of the service, verifying, by a convergent communication platform located on the home network, that the identification number relates to a valid user account, that a user device is authorized to receive the service, and that the valid user account has sufficient value, providing an authorization to the service provider, and charging the valid user account on a real time basis. The convergent communications system employs a rule set usable in determining at least one rule applicable for authorizing a transaction and debiting an account of the authorized user according to the at least one rule, in real time.

SYSTEM FOR IMPLEMENTING TELEPHONE SERVICES

Call classification through analysis of dtmf events

Systems, methods, and computer-readable media for call classification and for training a model for call classification, an example method comprising: receiving DTMF information from a plurality of calls; determining, for each of the calls, a feature vector including statistics based on DTMF information such as DTMF residual signal comprising channel noise and additive noise; training a model for classification; comparing a new call feature vector to the model; predicting a device type and geographic location based on the comparison of the new call feature vector to the model; classifying the call as spoofed or genuine; and authenticating a call or altering an IVR call flow.

DEVICE FOR CONNECTING ADAPTATION TO THE DIGITAL TELECOMMUNICATIONS

Method and apparatus for establishing a through path for a communication signal

A method for establishing a path for a communication signal is performed in a wireless communication system which includes a plurality of interconnected wireless network elements (510, 514, 515, 516), and a landline communications network which includes a plurality of interconnected landline network elements (305, 404, 508). The method includes establishing a first signaling link (501, 330) between a communication unit (206) and a landline network element (404), followed by establishing, by the landline network element (404), a second signaling link (522, 524) between the landline network element (404) and a wireless network element (514). The communication unit (206) transmits mobility-related messaging associated with the communication signal to the wireless network element (514) via the first and second signaling links.

DTMF Signaling on four-wire switched 56 Kbps Lines

Dual tone multi-frequency (DTMF) address signaling on four-wire switched 56 kilobits per second (Kbps) service lines is obtained by determining the proper byte boundary information. An iterative learning process is used with an existing digital switching network having a call progress indication capability. The learning process steps through each possible byte alignment until the call progress protocol indicates that a proper byte alignment has been achieved. Once a paper byte alignment has been determined, that byte alignment information is retained for subsequent use, thus avoiding the need to repeat the learning process.

Method of accessing service resource items that are for use in a telecommunications system

Convergent communications platform and method for mobile and electronic commerce in a heterogeneous network environment

A method and apparatus for providing mobile and electronic commerce, customer care and communication services via networks, including receiving in a roaming network an identification number and a request for a service, forwarding to a home network the identification number, the request for the service, and a cost/rate of the service, verifying, by a convergent communication platform located on the home network, that the identification number relates to a valid user account, that a user device is authorized to receive the service, and that the valid user account has sufficient value, providing an authorization to the service provider, and charging the valid user account on a real time basis. The convergent communications system employs a rule set usable in determining at least one rule applicable for authorizing a transaction and debiting an account of the authorized user according to the at least one rule, in real time.

Method and device for providing service features for mobile subscribers over a communication network

The process involves an external terminal appts. (KE1) logging on at a communication system (PBX) and a speech connection (SV) is set up between the external end unit and a teleworking participant connective group (T-BG) of the communication system. Control information input to the external unit (KE1) are transmitted via the speech connection in the form of DTMF signals. At the teleworking side, the signals are changed to signalising information according to a signalisation protocol (Cornet -TS) and are transferred to a control unit (STE) of the communication system. The teleworking participant connective group has a teleworking computer (TW-S) connected via a local area network. The computer controls the group. The control information received at the group are transmitted via the computer to the control unit.

Intelligent multimedia calls

A method of setting up and/or controlling a multimedia call involving an H.324 enabled user terminal and a circuit switched connection terminating at the user terminal and at a video gateway, the method comprising: sending DTMF control signals over the circuit switched connection within H.245 protocol control messages; and on the basis of said DTMF control signals, routing or re-routing the connection to an appropriate data source or mapping the connection to an appropriate data source.

Cooperative media applications using packet network media redirection

A system and method for a packet VRU which directly utilize packet network protocols, such as those of the H.323 standard, to provide enhanced services via a packet network, such as whisper mode communications, call recording, call broadcasting, advertisement broadcasting during calls, one-number service, and operator services. In a preferred embodiment of the present invention, the packet VRU establishes control of media streams, such as through the use of separate signaling channels associated with various communication devices, in order to provide desired enhanced call services.

Computer controlled telecommunication system and sub-systems of such telecommunication system

Distributed processing systems incorporating a plurality of cells which process information in response to single events

A distributed processing system includes a plurality of cells where each cell processes information in response to a type of event, and cells communicate with each other according to an (asynchronous) event-reaction protocol. Cells maintain mailing lists which are associated with events and which indicate the address(es) of cell(s) to whom a message is sent in response to the event. According to one embodiment, at least one cell is provided with the ability to read, write, and delete any address on any mailing list in the system. According to another embodiment, at least one cell keeps a list of events (subscription list) that are deemed important by the user where it is the cell's responsibility to insure that it remains subscribed to the mailing lists associated with these events. According to another embodiment, at least one cell serves more than one mailing list and preferably keeps a list of the mailing lists it serves. This cell also has the ability to move subscribers from one mailing list to another, to modify which mailing list is the currently active mailing list or to make a plurality of mailing lists simultaneously active. According to another embodiment, at least one cell has the ability to modify the subscription list of another cell. The invention is illustrated by example with reference to a PBX system.

TELEPHONE STATION, MULTIMEDIA TERMINAL AND SERVER

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.

Telecommunication switch having programmable network protocols and communications services

Intelligent, high-speed switch matrix

COMMUNICATION BETWEEN AN ELEVATOR INSTALLATION AND A REMOTE SERVER.

Un procédé de communication entre une installation d'ascenseur (1) et un serveur distant (17), le procédé comprenant - recevoir des données numériques de mesure (Sensor_data(n)) issues du au moins un capteur et/ou des données numériques de commande (Ctrl_data(n)) issues du serveur distant et destinées à un dispositif de gestion pilotant l'installation d'ascenseur, - réaliser un encodage DTMF encoder desdites données numériques de mesure ou de commande, et - transmettre vers le serveur distant ou le dispositif de gestion un signal analogique (DDTMF(t) ; EDTMF(t)) correspondant à cet encodage DTMF de données, ce signal analogique étant dans la bande de fréquences de téléphonie fixe classique. A method of communication between an elevator installation (1) and a remote server (17), the method comprising - receiving digital measurement data (Sensor_data (n)) from the at least one sensor and / or digital data of command (Ctrl_data (n)) from the remote server and intended for a management device controlling the elevator installation, - performing a DTMF encoding encoding said digital measurement or control data, and - transmitting to the remote server or the an analog signal management device (DDTMF (t); EDTMF (t)) corresponding to this data DTMF encoding, this analog signal being in the conventional fixed telephony frequency band.

Telecommunication switch having programmable network protocols and communications services

A telecommunications protocol development environment which enables a user to define a separate finite state machine for each port provided by a programmable telecommunications switch. Protocols for various telecommunications applications and software layers may be developed.

A method for exchanging signaling messages in two phases

Signaling messages are exchanged for a call between a calling party to a called party. A setup message for the call is exchanged through at least one gate controller. Network resources are reserved for the call based on the exchanged setup messages. An end-to-end message for the call is exchanged without the end-to-end message being routed through the at least one gate controller.

Systems and methods for managing messages

A Message Storage and Deliver System (10) (MSDS) is connected to the public switched telephone network (PSTN) and receives incoming calls with these calls being facsimile, voice, or data transmissions. The MSDS (10) detects the type of call and stores the message signal in a database. The MSDS (10) is also connected to the Internet and has a hyper-text transfer protocol deamon (HTTPD) for receiving requests from users. The HTTPD forwards requests for certain files or messages to a network server which transmits at least part of the message to the HTTPD and then to the user. In addition to request for certain documents, the HTTPD may also receive a request in the form of a search query. The search query is forwarded from the HTTPD to an application program for conducting the search of the database. The results of the search are forwarded through the HTTPD to the user. The user may then select one or more files or messages from the search results and may save the search for later reference.

Call flow system and method use in legacy telecommunication system

A method of operating a network server, such as a mobile application gateway, connect devices on a cellular or carrier network with individual networks, such as enterprise voice and data networks or residential networks. The effects of the present invention are far reaching in terms of transferring effective call control from the cellular network into the control of the individual network, such as the enterprise, and enabling new business models for the purchase of cellular service from a public cellular carrier by an enterprise.

Automatic call forwarding

A method of forwarding calls, comprises storing a list of possible forwarding destinations for each subscriber, maintaining a behaviour database for each subscriber indicating the probable location of the subscriber at any particular time, and forwarding incoming calls to the probable location determined from the behaviour database. The behaviour database is dynamically updated according to the success rate of the forwarded calls.

Communication network with hidden calling number capability

A method and apparatus are disclosed for delivering calling line information in a manner which preserves the privacy interests of the caller while providing useful information to the called party. The caller initiates a call from a caller terminal of a communication network. A destination central office or other network element receives the call along with a complete calling line number and delivers partial calling line information to the called party prior to connecting the call. The partial calling line information may include an NPA-NXX portion of an NPA-NXX-XXXX calling line number, geographical information such as state, county, city, township and zip code associated with the calling line, and/or an indication as to whether the call is from a residence, business, pay phone, cellular phone or other particular source such as a hospital or hotel. The central office or other network element may store the full calling line number in an incoming call log maintained for the called party, and a name selected by the called party may be associated with the calling line number. The central office will then deliver the selected name as part of the partial calling information when subsequent calls are placed from the calling line to the called party. This allows the called party to perform selective screening, call transfer and other functions without ever learning the full calling line number. The called party can also direct the central office to utilize the full calling line number stored therein to initiate a return call to the corresponding calling line.

Priority call on busy

A method and a system for setting a priority level of a call in which a request is received for changing a priority level of a call to one of a plurality of available call priority levels, the priority level of the call is changed based on the received request, and the call is handled based on the changed priority level of the call. The request for changing the priority level of the call can be a request for a central office service feature for changing the priority of the call, receipt a call routed to a predetermined telephone number, such as an 800, an 888, a 900 or a charge-to-caller number, or receipt of a DTMF signal indicating the requested priority level. A message can be generated providing a menu for changing the priority level of the call having at least one available priority level with an associated charge rate for each available priority level. When the priority level of the call is changed, the call can be advanced in a call holding queue, a lower priority call can be interrupted or the call can be routed on a high-priority trunk capacity available for highpriority calls.

Systems and method for storing, delivering, and managing messages

Systems and methods are provided for storing a mark-up language file and delivering the file from a network server to a user's computer via a packet switched data network using a hyper-text transfer protocol (HTTP). The mark-up language file is stored in a storage area accessible by the network server, wherein the file contains information personal to the user. A notification is transmitted to the user's computer, wherein the notification serves to notify the user of the availability of the file. In response to a user request made in response to the notification, the mark-up language file is transmitted from the network server to the user's computer, via the packet switched data network, using the HTTP.

Response service procedures and systems

Method and apparatus for providing user controlled call management services

A method of providing call screening to a subscriber of personal communication services facilitates the screening of incoming calls by associating the number dialed by a calling party with a function or role of the dialed number. This function or role is then sent to the subscriber when an incoming call is received. This way, the subscriber can decide whether to answer the incoming call based upon knowing the role or context associated with the dialed number. The subscriber is also provided with user controlled features which can be invoked by the use of DTMF codes. The subscriber can place a calling party on hold prior to answering the incoming call. In addition, the subscriber can proceed with a call transfer from a wireless telephone to a wired telephone and vice-versa, while the call is still in progress. Another feature allows the subscriber to implement a real-time call screening, while the calling party is leaving a message on voice mail. Yet another feature provides a wide area virtual extension to several subscribers such that an incoming call will be simultaneously routed to each subscriber in the group, as if their telephones were extensions of each other.

COMMUNICATION PROCEDURE WITH PERSON NUMBER

Method for establishing connections between mobile radio communication system and subscriber of private communication network

FIELD: communications engineering. SUBSTANCE: method for establishing connections between mobile radio communication system and terminal subscriber of private communication network involves use of mobile radio communication network access device wherein request for establishing connection is conveyed from mobile set to mobile radio communication network access device; the latter first initiates connection with desired terminal subscriber of private communication network and then, if connection is adequate, it also establishes connection with mobile set. EFFECT: enhanced reliability of connecting mobile radio communication network to private communication network. 5 cl, 6 dwg

Data interface apparatus and method

A data interface apparatus (10) provides an interface between an analog communications link (a2) and a digital communications link (d2). Analog dialling codes on the analog communications link (a2) are converted to control codes for controlling a digital communications interface (12) attached to the digital communications link (a2) for providing access to a digital communications network (16, 18, 20). Access to the digital communications link (a2) is selectively enabled by the data interface apparatus (10) in response to a password encoded as an analog dialling sequence on the analog communications link (a2). The data interface apparatus (10) delays call set-up with a modem (6) connected to the analog communications link (a1, a2) so as to synchronize completion of call set-up with another call set-up procedure performed on a remote analog link (a3, a4) connected to the digital communications network (16, 18, 20).

Presence information method and system

A method of operating an availability service in a telecommunication system (1) interconnecting a legacy telephone system (10) with at least one presence system (11), said legacy telephone system including a plurality of terminals (T1...T10), at least some (T1,T2) of which are terminals of subscribers to said availability service, said method including: (a) continuously monitoring each subscriber terminal (T1,T2) for a change of a call status of said subscriber terminal from the presence of a call to the absence of a call; (b) generating a first or "high availability" signal when said call status of said subscriber terminal changes from said presence of a call into said absence of a call; (c) monitoring an interval of time beginning with said first signal; (d) generating a second or "availability unknown" signal after a predetermined time period .tau.within said interval; and (e) making said availability signals accessible to any interconnected presence system.

Digital key telephone system

A digital key telephone system comprising a main unit connected to a communication network through an office line, accommodating a plurality of extension lines. When an operation mode change request arrives at a computer connection interface from a computer connected to the computer connection interface, transfers the change request to the main unit of the system. The main unit, when the mode change request is transferred, determines whether the mode change request is appropriate and supplies a determination result to the computer connection interface which has transferred the mode change request and the computer connection interface performs mode change processing in response to the mode change request from the computer on the basis of the determination result supplied by the main unit after transfer of the mode change request.

Method and system for termination blocking of message delivery service in a switch-based telecommunication system

A method for providing a messaging service in a telecommunication system ( 900 ) includes providing a calling party with an option to leave a voice message ( 1010 ). If the calling party accepts the option, determining screening criteria for the called communication station ( 1018 ). In response to the screening criteria, the call is selectively completed to a messaging service ( 172 ).

Call setup path for telecommunication system

Method of providing telecommunication services

Traditional IN (Intelligent Network) services in a PSTN use service logic and data that is accessible for use only by the PSTN, though provision may be made for users to change certain controlled parameters of the services. The present system has the service logic and data placed on a server (51) accessible over the Internet (50). This permits anyone to access useful telephone data such a time-of-day routing or diversion number information of a user. Thus a calling party (A) can determine before placing a call over the PSTN the best number to call by accessing the phone page (49) of the intended called party (B). This phone page (49) would remain accessible to the PSTN for service provision. In a preferred embodiment, the service logic and data is maintained by the user (B) independently of the PSTN on a server (51) of the user's choice; in this case, the PSTN would also access the user's service logic and data over the Internet (50).

Communications web for PSTN subscribers

用于将电信基础设施线连接到住宅或企业中的电话、手持机、计算机、电复印机和其它终端用户接口或用户电气设备的系统。按照本发明的系统包括网络控制单元,形成星形拓扑结构的中心,并且与无线接入单元和手持机通过RF链路通信。无线接入单元具有一接口,诸如标准电话插孔,用于接纳一电话、传真机、计算机调制解调器或其它设备。为了补偿一端公共电信网络和另一端本发明系统之间的线路损耗、失配和其它不兼容,本发明的新编码技术,与常规的使用8位编码字的μ定律编码和解码相比,采用使用10位编码字编码和解码的编解码器,产生的16位性能导致在本发明的系统的无线频率链路上主观上约90dB的信噪比。这种编码技术和10位码段能够容易地由当前处理能力适应,这在采用μ定律编码的时代,是不能使用的。

Enhanced communication platform and related communication method using the platform

An advanced intelligent communication system that provides subscriber-requested, advanced communication services through existing communication switches even in those circumstances in which the hardware communication switch is not configured to provide such communication services. The system supports the use of personal identification number access cards for use in fixed and mobile markets from any conmunication device located anywhere in the world and provides flexible call processing and switching services that deliver enhanced computer telephony capabilities utilizing standard communication equipment and operating systems.

Method for setting up connection between mobile radio network and destination cell number in private communication network

本发明涉及通过一个移动无线网接入装置建立从移动无线网至一个专用通信网的一个电话号码的方法，其中由一个移动终端传送一个连接请求至移动无线网接入装置，并且首先由移动无线网接入装置引导建立至所要专用通信网电话号码的连接，并在其完成后接着建立至移动终端的连接。

Double-IP telephone and method for tlelcomunicating with the same telephone

一种双模电话和使用该双模电话的通信方法，它具有ISDN电话和IP电话。该双模电话根据用户输入来识别ISDN模式、IP模式和外部连接模式中的至少一种，并根据所识别的模式来连接或断开ISDN电话和IP电话之间的话音路径，从而实现外部ISDN电话和IP电话之间的呼叫。

Personal number communications system

为用户传送通信的方法包括：指定用户个人号码；接收包括目的地表的通信路由信息，该表按照通信路由的用户优先权次序按级排列；接收发向个人号码的通信；从所述通信路由信息中选择第一目的地；通报在所述第一目的地接收所述通信；接收通信处理信息；根据通信处理信息借助于将所述的通信传送到所述第一目的地或借助于在所述目的地表选出下级目的地，处理所述通信，并且在所述的下一目的地重复执行上述这些步骤。

Communication services

Extension number switching method and apparatus and telephone exchange using the same

Method and apparatus for message communication

In a message communication in which communication of message data is effected between a calling terminal and a called terminal connected to each other through an integrated services digital network, control information indicative of message communication is added in the calling terminal to the call setting information transmitted to the integrated services digital network when originating a call to a called terminal; in the called terminal, the call setting information received from the integrated services digital network is analyzed; when it is determined that control information is not added to the call setting information, it switches to a voice mode to effect voice communication with the called terminal after responding to the call setting information; when control information is added to the call setting information, it switches to the message mode to automatically receive message data transmitted from the called terminal after responding to the call setting information.

User selectable overlap and enbloc dialing of ISDN line

An ISDN test set employs a user selectable dialing mechanism, that allows the test set user to select from multiple types of dialing—overlap dialing and enbloc dialing, through the use of a simplified functional dialing key set. The dialing key set, in combination with dialing software employed by the test set's supervisory control processor, provides the user with substantial ISDN dialing flexibility, while reducing interaction between the user/craftsperson and the test set, that minimizes dialing inaccuracies that often depend upon the skill and familiarity of the technician with ISDN communications. Enbloc redialing (of the most recently dialed number) or of a stored number in memory is provided only in ON-HOOK mode by the operation of a REDIAL or RECALL key. Otherwise (whenever OFF-HOOK) only overlap mode dialing is employed.

Method and device for arranging local calls by telephone

Bei einem Verfahren und einer Einrichtung zur telefonischen Vermittlung eines Ortsgesprächs zwischen einem rufenden und einem gerufenen Endsystem über ein wenigstens partiell proprietäres Kommunikationsnetz durch einen Vermittlungsdienst ist erfindungsgemäß vorgesehen, daß eine vom rufenden Endsystem angewählte Rufnummer danach geprüft wird, ob das gerufene Endsystem innerhalb oder außerhalb des Vorwahlbereichs des rufenden Endsystems liegt, und daß der Anruf bei Vorliegen des gleichen Vorwahlbereichs an einen Leitungsträger (Carrier) eines alternativen Vermittlungsdienstes (Provider) weitergeleitet und von diesem, unter Verwendung einer Vorwahlnummer, an das gerufene Endsystem vermittelt wird. Das vorgeschlagene Verfahren ermöglicht auch im Nahbereich die Bereitstellung von alternativen Vermittlungsdiensten und trägt dort zur Gebühren- bzw. Kostenminimierung bei. In a method and a device for the telephone switching of a local call between a calling and a called end system via an at least partially proprietary communication network by a switching service, the invention provides that a call number dialed by the calling end system is checked according to whether the called end system is inside or outside the Area code range of the calling end system lies, and that the call is forwarded to a line carrier of an alternative switching service (provider) when the same area code area is present, and is transferred by this, using a code number, to the called end system. The proposed method also enables the provision of alternative switching services in the short-range and helps to minimize fees and costs there.

Connection system between public telecommunications network and private companies

The company telephone exchanges (F-KVSt, HVSt, ZVSt and EVSt) function as conventional (public) Telekom systems, except that in the register (KRg or Rg) and translator (UmW), not only is the local network identifier (ONKZ) - without the prefix "0" - evaluated, but also a service identifier (DKZ), which is dialled first, and possibly the origin (Telekom or private company) are evaluated, so that the correct connection to a Telekom subscriber or a company subscriber can be made via a favourable connection. Additional junction line groupings connect the nodal exchanges of the private companies with those of the public Telekom. The line groupings are provided for incoming calls from Telekom, outgoing calls for Telekom terminal exchanges, and outgoing calls to Telekom nodal exchanges for connections via further Telekom exchanges e.g. for local or foreign calls. Repeaters perform the required adjustments including charge determination.

Teleconference device and method

Call classification through analysis of dtmf events

Systems, methods, and computer-readable media for call classification and for training a model for call classification, an example method comprising: receiving DTMF information from a plurality of calls; determining, for each of the calls, a feature vector including statistics based on DTMF information such as DTMF residual signal comprising channel noise and additive noise; training a model for classification; comparing a new call feature vector to the model; predicting a device type and geographic location based on the comparison of the new call feature vector to the model; classifying the call as spoofed or genuine; and authenticating a call or altering an IVR call flow.

Method and system for making a phonecall to an international destination using voip infrastructure

The method and system in accordance with the preferred embodiment for making a phonecall to an international destination using VoIP infrastructure allows a user to provide a simple and quick indication to the VoIP gateway that the user intends to initiate a phonecall to an international destination. The VoIP gateway can then advantageously be prepared to receive the string of digits and analyze the first few digits for a country code while ensuring that the succeeding numbers correspond to a subscriber number corresponding to that country code. The present quickdial method in accordance with the preferred embodiment requires the user in a VoIP setting to dial an asterisk, corresponding to the '*' symbol on most phone keypads before providing the country code and subscriber number. The dialing of the asterisk, is advantageously an indication to an originating gateway that the incoming string of digits corresponds to a phonecall to an international destination and not to be confused with a local subscriber number.

System and method for packet network media redirection

A system and method for a packet VRU which directly utilize packet network protocols, such as those of the H.323 standard, to provide enhanced services via a packet network, such as whisper mode communications, call recording, call broadcasting, advertisement broadcasting during calls, one-number service, and operator services. In a preferred embodiment of the present invention, the packet VRU establishes control of media streams, such as through the use of separate signaling channels associated with various communication devices, in order to provide desired enhanced call services.

Private Branch Exchange

In a private branch exchange having an optimum route selecting function of selecting an optimum route by taking the destination, day of the week, time zone, permissive class of the extension terminal and the like into consideration when a call originating request from the extension terminal is generated, adequately processing dial number and outputting the same to an outgoing line so as to use the route, the route is selected by taking usage rates of the respective routes into consideration. Further, in the route selecting process, the route is selected by taking communication time (estimated value) into consideration. In an exchange system which is constructed by connecting plural private branch exchanges and in which, when an optimum route connected to one exchange is busy, dial number for permitting use of the optimum route is output via another exchange, the optimum route is not immediately selected via another exchange, but a substantially optimum route or routes which are substantially the same as the optimum route and connected to the given exchange are selected so that the dial number will be output from the given exchange with maximum possibility.

Presence information method and system

A method of operating an availability service in a telecommunication system (1) interconnecting a legacy telephone system (10) with at least one presence system (11), said legacy telephone system including a plurality of terminals (T1...T10), at least some (T1,T2) of which are terminals of subscribers to said availability service, said method including: (a) continuously monitoring each subscriber terminal (T1,T2) for a change of a call status of said subscriber terminal from the presence of a call to the absence of a call; (b) generating a first or "high availability" signal when said call status of said subscriber terminal changes from said presence of a call into said absence of a call; (c) monitoring an interval of time beginning with said first signal; (d) generating a second or "availability unknown" signal after a predetermined time period .tau.within said interval; and (e) making said availability signals accessible to any interconnected presence system.

Digital key telephone system and extension terminal device thereof

DMP sending device and method in private exchange system

가. 청구범위에 기재된 발명이 속하는 기술분야 end. TECHNICAL FIELD OF THE INVENTION 사설 교환시스템에서 디티엠에프를 송출하는 장치 및 방법 Apparatus and method for transmitting DMP in private exchange system 나. 발명이 해결하려고 하는 기술적 과제 I. The technical problem that the invention is trying to solve 사설 교환시스템에서 가입자가 송출하고자 하는 디티엠에프를 저장하여 한꺼번에 송출하는 장치 및 방법을 제공한다. Provided are an apparatus and a method for storing and transmitting DTMs that a subscriber wants to transmit in a private exchange system. 다. 발명의 해결 방법의 요지 All. Summary of the Solution of the Invention 사설 교환시스템에서 가입자에 의해 다이얼링된 디티엠에프 데이터를 저장하는 저장부와, 상기 가입자로부터의 데이터 입력이 완료된 것을 검출하여 상기 저장된 디티엠에프 데이터의 송출을 제어하는 제어부와, 상기 제어부의 제어하에 상기 저장된 디티엠에프 데이터를 송출하는 송출부로 구성된다. A storage unit for storing DTP data dialed by a subscriber in a private switching system, a control unit for detecting the completion of data input from the subscriber and controlling the transmission of the stored DFM data, under the control of the control unit; And a transmitter for transmitting the stored DFM data. 라. 발명의 중요한 용도 la. Important uses of the invention 사설 교환시스템에서 디티엠에프 송신부의 점유 시간을 줄여서 디티엠에프 송신부를 효율적으로 사용할 수 있다. In a private switching system, it is possible to efficiently use the DTP transmitter by reducing the occupation time of the DTP receiver.

Call classification through analysis of DTMF events

Systems, methods, and computer-readable media for call classification and for training a model for call classification, an example method comprising: receiving DTMF information from a plurality of calls; determining, for each of the calls, a feature vector including statistics based on DTMF information such as DTMF residual signal comprising channel noise and additive noise; training a model for classification; comparing a new call feature vector to the model; predicting a device type and geographic location based on the comparison of the new call feature vector to the model; classifying the call as spoofed or genuine; and authenticating a call or altering an IVR call flow.

Method and apparatus for message communication

System and method to detect pbx-mobility call failure

Aspects relate to initiating voice calls to mobile devices from an enterprise network. In one example, a server can initiate a voice channel to a mobile device, and if the mobile device cannot be reached, an access network may redirect the call from the server to a number identified as a voicemail number; however, such number is reconfigured to instead be directed to a number located within the enterprise network. The server can monitor or receive an indication that a call is incoming, and attributable to the mobile device. If such a call is incoming or received, then the server determines that the call has failed. If the mobile device answers the call, the server can determine that the voice channel has been established, without waiting for a confirmation indication, such as a DTMF tone.

Modem and method of operating the same

모뎀 및 그 제어방법이 개시된다. 본발명의 모뎀은 가입자 단말기와 인터페이싱하기 위한 제1인터페이스와, 교환국 장치와 인터페이싱하기 위한 제2인터페이스, 가입자 단말기와의 통신 접속 상태를 검출하고, 검출 결과신호를 출력하는 가입자 단말기 접속상태 검출수단과, 가입자 단말기와 교환국장치 사이의 양방향 데이터 통신을 처리하며 가입자 단말기 접속상태 검출수단으로부터 비접속 신호가 입력되면 제2인터페이스의 동작이 중지되도록 제2인터페이스를 제어하는 제어부를 구비한다. 이러한 모뎀 및 그 제어방법에 의하면, 가입자 단말기와의 통신 접속상태가 비접속상태일 때 교환국장치와의 통신 채널을 해제하도록 동작함으로써, 불필요한 전력소모를 억제할 수 있고 기기의 수명을 연장시킬 수 있다. 또한, 이러한 가입자 모뎀의 동작은 다수의 가입자 모뎀에 대한 통신서비스를 지원하는 교환국 장치의 불필요한 회선 사용이 억제되어 교환국장치의 전력절감 및 전체 통신시스템의 데이터 처리속도를 향상시키는 파급 효과를 발생시키는 장점이 있다. A modem and a control method thereof are disclosed. Modem of the present invention comprises a first terminal for interfacing with the subscriber terminal, a second interface for interfacing with the switching center apparatus, the subscriber terminal connection state detection means for detecting a communication connection state with the subscriber terminal and outputting a detection result signal; And a control unit which processes the bidirectional data communication between the subscriber station and the switching center apparatus and controls the second interface to stop the operation of the second interface when a non-connection signal is input from the subscriber station connection state detecting means. According to such a modem and its control method, it is possible to suppress unnecessary power consumption and extend the life of the device by operating to release the communication channel with the switching center apparatus when the communication connection state with the subscriber station is not connected. . In addition, the operation of the subscriber modem is advantageous in that unnecessary use of the circuit of the switching center device supporting the communication service for a plurality of subscriber modems is suppressed, thereby generating a ripple effect of reducing the power of the switching center device and improving the data processing speed of the entire communication system. There is this.

Method and apparatus for accessing communication data relevant to a target entity identified by a number string

Service resource items ( 49 ) for use in call setup in a telephone system are held on servers ( 58 ) that are connected to a computer network which is logically distinct from the telephone system infrastructure; this computer network may, for example, make use of the Internet. Each service resource item ( 49 ) is locatable on the network at a corresponding URI and is associated with a particular telephone number. A mapping is provided between telephone numbers and the URIs of associated service resource items. When it is desired to access a service resource item ( 49 ) associated with a particular telephone number, this mapping is used to retrieve the corresponding URI (step 55 ) which is then used to access the desired service resource item ( 56, 57, 59 ).