WO2009065308A1 - A method and apparatus for calling user to join conference - Google Patents

A method and apparatus for calling user to join conference Download PDF

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Publication number
WO2009065308A1
WO2009065308A1 PCT/CN2008/071366 CN2008071366W WO2009065308A1 WO 2009065308 A1 WO2009065308 A1 WO 2009065308A1 CN 2008071366 W CN2008071366 W CN 2008071366W WO 2009065308 A1 WO2009065308 A1 WO 2009065308A1
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WIPO (PCT)
Prior art keywords
call
conference
user
voice stream
subscriber number
Prior art date
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PCT/CN2008/071366
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French (fr)
Chinese (zh)
Inventor
Zhengkai Yin
Original Assignee
Huawei Technologies Co., Ltd.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
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Publication date
Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2009065308A1 publication Critical patent/WO2009065308A1/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00Data switching networks
    • H04L12/02Details
    • H04L12/16Arrangements for providing special services to substations
    • H04L12/18Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
    • H04L12/1813Arrangements for providing special services to substations for broadcast or conference, e.g. multicast for computer conferences, e.g. chat rooms
    • H04L12/1818Conference organisation arrangements, e.g. handling schedules, setting up parameters needed by nodes to attend a conference, booking network resources, notifying involved parties
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42204Arrangements at the exchange for service or number selection by voice
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/56Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2201/00Electronic components, circuits, software, systems or apparatus used in telephone systems
    • H04M2201/40Electronic components, circuits, software, systems or apparatus used in telephone systems using speech recognition

Definitions

  • the present invention relates to a method and apparatus for calling a user to join a conference, and more particularly to a method and apparatus for calling a user to join a conference during a conference.
  • VoIP Voice over IP
  • VoIP voice over the network
  • NGN Next Generation Communication Network
  • the network system is based on packet-switched data networks and directly targets multi-service convergence.
  • the service provides requirements for integration, personalization, and openness.
  • the technical system meets the requirements of layering, standardization, and openness, and represents the direction of network development.
  • IP PBX The core product of the enterprise-class IP telephony system is called IP PBX, which realizes the function of dedicated PBX voice service with open LAN packet switching technology, and seamlessly integrates with existing LAN. It not only integrates the internal data network and the voice network, but also meets the full range of needs to keep the enterprise running and flourishing. From complex call control to integrated voicemail and computer/telephone integration; from caller identification to multiple, multi-level auto attendants, to the easy addition, movement and change of terminal phones; from LAN and WAN (LAN/WAN) communication, connectivity to the Internet, etc., these technically While the advancement reduces the associated overall costs, it also enables companies to access all functions in an economical, manageable manner.
  • FIG. 1 The system structure of the IP PBX is shown in Figure 1, which includes:
  • Call Controller As the intelligent part of the enterprise-class IP telephony system, it is responsible for providing the services provided by all the centers of traditional PBX systems (PBX switches). Responsible for controlling all call setup and call management, providing most basic services (such as call hold, call forwarding, and call waiting), as well as configuring the phone's Extension Number, feature buttons, call management, and routing decisions. . In addition, it controls all PC virtual phone features such as voicemail, unified messaging, desktop call control, auto attendant, interactive voice response, and automatic call distribution.
  • Media gateway In the IP PBX system, the media gateway is used to realize the communication between the IP network and the traditional circuit switching network, and is responsible for transferring the call to the PSTN network to complete the incoming and outgoing calls of the heterogeneous network. In addition to having the ability to connect to the called party, it can also connect calls from the PSTN to the enterprise IP telephony system. Media gateways play a key role in the entire IP PBX system. It not only enables the IP PBX system to connect to PSTN (Public Service Telephone Network, PSDN) users, but also increases the reliability of the entire system, enabling the system to handle emergency calls. The media gateway also needs to complete voice code conversion, communication protocol conversion and LAN-PSTN (local area network to PSTN) call setup/demolition.
  • PSTN Public Service Telephone Network
  • LAN access module Provides the connection between the components of the IP PBX system.
  • the module can be a LAN switch or a function module of the call controller.
  • Control Center Includes resource management system, billing system, network management system, voice mail, etc.
  • the resource management system mainly performs functions such as address translation, authority control, and route management
  • the billing system mainly performs billing and settlement functions
  • the network management system completes maintenance and management of network and IP PBX operations.
  • Signaling Gateway Provides interworking functions with other signaling networks. It can support H.323, SS7 (Signaling System 7, Signaling No. 7, SS7), SIP (Session Initiation Protocol). All or part of the functions of SIP), ⁇ 248, MGCP (Media Gateway Control Protocol, MGCP) can be implemented on a single device, in multiple devices, or embedded in multiple devices. Implemented in the call controller.
  • Application Server Provides value-added application services for IP PBX systems.
  • IP phone terminal including terminal part connection software PC Phone, PC Phone+, IP-Phone, etc.:
  • PC Phone multimedia PC, or PC + sound card + phone handle that does not provide dialing function
  • PC Phone+ has the basic functions, appearance and usage of ordinary phone, it transmits voice information by means of sound card on PC , through the serial port to complete dialing and other controls;
  • IP-Phone The telephone directly connected to the IP network can be directly connected to the network through the RJ45 connector.
  • Data terminal A computer used for data communication purposes.
  • Analog terminal Traditional analog equipment such as ordinary telephones and fax machines.
  • IP PBX In addition to providing all the traditional C4/C5 and PBX services for enterprise users, the IP PBX solution provides a wealth of new services and applications.
  • the conference server function is one of them.
  • the system structure of the conference system is shown in Figure 2.
  • the conference system is based on the IP PBX system.
  • the MCU (Mul t ipoint Contro l Uni t, multi-point control unit, MCU) is added to the IP PBX system for conferences. Control and management, and use IP networks to combine video, audio and data to achieve mutual communication and negotiation among multiple people.
  • the MCU consists of three parts:
  • Conference service management component responsible for the control and management of high-level attributes such as policies and services of the conference;
  • signaling control component responsible for the transmission, control, interaction and other functions of signaling;
  • Media processing component responsible for receiving, encoding, negotiating, processing, etc. of the media.
  • the conference establishment process is shown in FIG. 3, and the conference establishment process is divided into two phases:
  • An object of the embodiments of the present invention is to provide a method and apparatus for a calling user to join a conference, to receive and identify a conference invitation in a voice manner, and to call a new conference member to join the conference.
  • an embodiment of the present invention provides a method for a calling user to join a conference, including:
  • Receiving an initiating call prompt message prompting to receive a call voice stream of the participant; receiving a call voice stream of the participant;
  • An invitation to join the conference is issued to the called subscriber number.
  • a device for calling a user to join a conference including:
  • a call prompt information receiving module configured to receive a call prompt information, and activate a call voice stream receiving module
  • the call voice stream receiving module is configured to receive a call command voice stream of the participant
  • a call voice stream identification module configured to identify the call command voice stream, and obtain a called user number
  • the invitation sending module of the conference is configured to send an invitation to join the conference to the called subscriber number according to the called subscriber number obtained by the call flow voice stream identification module.
  • the embodiment of the present invention implements receiving a conference invitation in a voice manner by receiving a call voice stream of the participant, and identifying the acquired voice stream by identifying the call voice stream. Calling the user number to implement the meeting invitation, and calling the new meeting member to join the meeting, so that the meeting starting process is more flexible and flexible.
  • FIG. 1 is a schematic structural diagram of a system of a prior art IP PBX
  • FIG. 2 is a schematic structural diagram of a system of a prior art conference system
  • FIG. 4 is a flowchart of a method for a calling user to join a conference according to Embodiment 1 of the present invention
  • FIG. 5 is a signaling flowchart of a mode 1 of a conference creation process according to Embodiment 1 of the present invention.
  • FIG. 7 is a schematic structural diagram of an embodiment of an apparatus for calling a user to join a conference according to the present invention. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
  • a flowchart of Embodiment 1 of the method for calling a user to join a conference, as shown in FIG. 4, includes the following steps:
  • Step 1 01 The conference server receives an initiation call prompt message
  • Step 102 The conference server receives a call command voice stream of the participant.
  • Step 1 03 The conference server identifies the voice stream of the call command, and obtains the called subscriber number.
  • the conference server may obtain the called subscriber number by using the following two methods:
  • the call command voice stream includes the called subscriber number information
  • the conference server identifies the call command voice stream and obtains a call command; the conference server obtains the subscriber number from the call command.
  • the call voice stream includes the called subscriber number query information;
  • the conference server queries the subscriber number information table according to the called subscriber query information, and obtains the called subscriber number.
  • the user number information table may be stored in the enterprise telephone server, or may be a conference user list configured by the conference chairperson when the conference server creates the conference call.
  • Step 104 Send an invitation to join the conference to the called subscriber number.
  • the voice recognition of the conference server can be started through an operation manner such as a command line, a web interface, or a special service number, as follows:
  • the conference server starts voice recognition, and recognizes the voice input by the specific terminal; if through the web interface, you can Any computer terminal logs in to the web page of the conference server, clicks the option to start the conference invitation and voice recognition, and the conference server starts the voice recognition module to identify the voice input by the specific terminal; if the special service number is adopted, the conference is added after the conference The conference terminal dials the special service number, and the conference server starts the voice recognition module to identify the voice input by the specific terminal.
  • the conference server can prompt the user to say the name of the person to be invited by playing the prompt tone, and end the voice input with a certain button. When all participants are added to the conference, the conference chairperson can turn off voice recognition through the command line, web interface, and dialing the service number.
  • the conference creation process can be as follows:
  • the user first dials the conference access number through the telephone terminal to enter the conference.
  • the conference server will actively call the user in the conference user list stored therein (in which the information of the members participating in the conference and the corresponding user terminal are recorded). Enter the meeting.
  • Each user in the meeting member address book can set several phone numbers, and the conference server will dial the phone number under the user name one by one until it is connected and confirms that the corresponding phone terminal of the user establishes a communication connection with the conference.
  • the difference from the first method is that the user logs in to the conference server through a web page or a GUI (Graphic User Interface, GUI).
  • GUI Graphic User Interface
  • the operation triggers the conference startup process. Further, you can edit the information in the conference user list through interface operations, and add or delete conference members.
  • the specific process is as follows:
  • the user logs in to the conference server through the web page or GUI, opens the conference user list, selects some users, and starts the conference.
  • the conference server actively calls these users to the specified conference.
  • the user can load himself into the conference user list, let the conference server call himself to the conference, or dial the conference number through the telephone terminal to access the conference.
  • the conference startup mode provided in this embodiment can accurately select the users who need to participate in the conference, and has great flexibility.
  • a privileged user (such as an administrator or other user who has permission to create a conference) can log in to the conference server through a web page or a GUI to create a conference.
  • the conference server allocates corresponding conference resources for the conference, and performs setting of conference members and the like.
  • the conference timing mode is introduced, and the conference server directly initiates the conference initiation process. As shown in Figure 6, the specific process is as follows:
  • the conference member address book and the conference start time are configured.
  • the conference server dials the telephone number of the user terminal according to the user information of the conference participant configured in the conference user list, and actively calls the user. The user joins the meeting.
  • the method further includes: returning a user verification prompt, which may be a voice prompt, etc., requiring the user to input relevant verification information, and the verification information may be preset for the conference chairperson. a password, etc.
  • a user verification prompt which may be a voice prompt, etc., requiring the user to input relevant verification information
  • the verification information may be preset for the conference chairperson. a password, etc.
  • step 102 Receive authentication information sent by the user and verify the identity of the user. If the verification is passed, step 102 is continued; otherwise, the information of the authentication failure is returned, and the process of the first embodiment is ended.
  • FIG. 7 is a schematic structural diagram of an embodiment of an apparatus for calling a user to join a conference according to the present invention, including:
  • the call prompt information receiving module 1 is configured to receive the call prompt information and initiate the call voice flow. Receiving module
  • a call voice stream receiving module 2 configured to receive a call command voice stream of the participant
  • a call voice stream identification module 3 configured to identify a call command voice stream, and obtain a called user number
  • the invitation sending module 4 of the conference is configured to send an invitation to join the conference to the called subscriber number according to the called subscriber number obtained by the call command voice stream identification module.
  • the call voice stream identification module may specifically include:
  • a call command extraction module configured to acquire a call command from the call voice stream
  • the user number extraction module is configured to obtain a user number from a call command obtained by the call command extraction module.
  • the call voice stream identification module may further include:
  • the called subscriber number query information extraction module is configured to be obtained from the call voice stream
  • the user number information table querying module is configured to query the user number information table according to the called user number query information, and obtain the called user number.
  • the user number information table may be located in the enterprise telephone server; the user number information table may also be a conference user list.
  • the device in this embodiment may further include a user verification module, configured to return a user verification prompt, receive verification information sent by the user, and verify the identity of the user.
  • a user verification module configured to return a user verification prompt, receive verification information sent by the user, and verify the identity of the user.
  • the storage medium may be a magnetic disk, an optical disk, a read-only memory (ROM), or a random access memory (RAM).

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • General Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Telephonic Communication Services (AREA)

Abstract

A method and an apparatus for calling user to join conference are disclosed, said method comprises: receiving call speech stream of conference participant; recognizing the call speech stream, obtaining number of called user; sending invitation of joining conference to the called user's number. Said apparatus comprises: call prompt message receiving unit for receiving prompt message of initiating call and starting call speech stream receiving unit; call speech stream receiving unit for receiving speech stream of call command of conference participant; call speech stream recognition unit for recognizing speech stream of call command and obtaining the called user's number; conference invitation sending unit for sending invitation of joining conference to the called user according to the number of the called user obtained by call command speech stream recognition unit. With the method and the apparatus, conference invitation in the form of speech can be received and recognized for calling new conference member to join.

Description

一种呼叫用户加入会议的方法及装置  Method and device for calling user to join conference
本申请要求于 2007 年 11 月 23 日提交中国专利局、 申请号为 200710178010.4、 发明名称为"一种呼叫用户加入会议的方法及装置"的中国 专利申请的优先权, 其全部内容通过引用结合在本申请中。 技术领域  This application claims priority to Chinese Patent Application No. 200710178010.4, entitled "A Method and Apparatus for Calling a User to Join a Conference" on November 23, 2007, the entire contents of which are incorporated by reference. In this application. Technical field
本发明涉及一种呼叫用户加入会议的方法及装置, 尤其是一种在会议过程 中呼叫用户加入会议的方法及装置。 背景技术 随着 IP技术的广泛普及, 目前众多企业都开始迈向 VoIP ( Voi ce over IP, 网络上传送话音, 筒称 VoIP ) , 通过采用新的业务和应用, 以达到提高生产效 率和降低通信成本的目的。 近几年出现的 IP PBX ( IP pr ivate branch exchange, IP专用分组交换机, 筒称 IP PBX )技术, 就是企业专网 VoIP技术的代表。 在 企业 VoIP市场日益兴旺的同时, 运营商的 NGN ( Next Genera t ion Network , 下 一代通信网络, 筒称 NGN ) VoIP技术也在短短的几年时间在全国开展起来。 由 于其先进的开放理念, 以软交换为核心的 NGN技术一出现, 就引起了全 3½营 商、 设备供应商和软件开发商的高度重视, 一些新的业务不断出现。 该网络体 系基于包交换数据网络, 直接面向多业务融合, 业务提供满足综合化、 个性化、 开放性等要求, 技术体系上满足层次化、 标准化、 开放性的要求, 代表了网络 发展的方向。  The present invention relates to a method and apparatus for calling a user to join a conference, and more particularly to a method and apparatus for calling a user to join a conference during a conference. BACKGROUND With the widespread popularity of IP technologies, many enterprises are now moving toward VoIP (Voice over IP, voice over the network, called VoIP), and by adopting new services and applications, to improve production efficiency and reduce communication. The purpose of the cost. In recent years, the IP PBX (IP pr ivate branch exchange, IP PBX) technology is the representative of enterprise private network VoIP technology. While the enterprise VoIP market is booming, the carrier's NGN (Next Generation Communication Network, NGN) VoIP technology has also been launched nationwide in just a few years. Due to its advanced open concept, the emergence of NGN technology with softswitch as the core has attracted the attention of all 31⁄2 business, equipment suppliers and software developers, and some new businesses are emerging. The network system is based on packet-switched data networks and directly targets multi-service convergence. The service provides requirements for integration, personalization, and openness. The technical system meets the requirements of layering, standardization, and openness, and represents the direction of network development.
企业级 IP电话系统的核心产品称为 IP PBX, 即以开放的局域网包交换技术 实现专用 PBX语音服务的功能, 与现有的局域网无缝集成。 它不仅可以使内部 的数据网和话音网合一, 还可以满足保持企业持续运作和蓬勃发展的全方位需 求。 从复杂的呼叫控制, 到集成的语音邮件以及计算机 /电话集成功能; 从呼叫 者身份识别到多重、 多级自动值机员, 再到终端话机的轻松添加、 移动和改变; 从局域网和广域网 (LAN/WAN )通信, 到 Internet的连接能力等, 这些技术上 的进步降低相关的整体费用的同时, 还使得使企业能够以经济的、 可管理的方 式获得所有功能。 The core product of the enterprise-class IP telephony system is called IP PBX, which realizes the function of dedicated PBX voice service with open LAN packet switching technology, and seamlessly integrates with existing LAN. It not only integrates the internal data network and the voice network, but also meets the full range of needs to keep the enterprise running and flourishing. From complex call control to integrated voicemail and computer/telephone integration; from caller identification to multiple, multi-level auto attendants, to the easy addition, movement and change of terminal phones; from LAN and WAN ( LAN/WAN) communication, connectivity to the Internet, etc., these technically While the advancement reduces the associated overall costs, it also enables companies to access all functions in an economical, manageable manner.
IP PBX的系统结构如图 1所示, 其中包括:  The system structure of the IP PBX is shown in Figure 1, which includes:
1 )呼叫控制器: 作为企业级 IP 电话系统的智能部分, 它负责提供一切传 统的 PBX系统的中心(PBX交换机)所能提供的服务。 负责控制所有的呼叫建立 和呼叫管理, 能提供大部分基本服务(如呼叫保持、 呼叫转移及呼叫等待等) , 以及配置电话的分机号码 ( Extension Number ) 、 功能按钮、 通话管理和路由 决策等功能。 此外, 它还控制所有的 PC虚拟电话功能, 如语音邮件、 统一消息、 桌面呼叫控制、 自动话务员、 交互话音响应和自动呼叫分配等。  1) Call Controller: As the intelligent part of the enterprise-class IP telephony system, it is responsible for providing the services provided by all the centers of traditional PBX systems (PBX switches). Responsible for controlling all call setup and call management, providing most basic services (such as call hold, call forwarding, and call waiting), as well as configuring the phone's Extension Number, feature buttons, call management, and routing decisions. . In addition, it controls all PC virtual phone features such as voicemail, unified messaging, desktop call control, auto attendant, interactive voice response, and automatic call distribution.
2)媒体网关: 在 IP PBX系统中, 媒体网关用来实现 IP网络和传统电路交 换网的通信, 负责把呼叫转接到 PSTN网, 完成异种网络的电话呼入和呼出。 它 除了具有接通被叫的功能外, 还可以把来自 PSTN的呼叫接续到企业级 IP电话 系统。 媒体网关在整个 IP PBX 系统中起着非常关键的作用。 它不仅使 IP PBX 系统能够连接 PSTN ( Public Service Telephone Network, 公共业务电话网, 筒称 PSDN)用户, 而且能够增加整个系统的可靠性, 使系统具有处理紧急呼叫 的能力。 媒体网关还需要完成语音编码转换, 通信协议转换及 LAN-PSTN (局域 网到 PSTN)呼叫建立 /拆除等功能。  2) Media gateway: In the IP PBX system, the media gateway is used to realize the communication between the IP network and the traditional circuit switching network, and is responsible for transferring the call to the PSTN network to complete the incoming and outgoing calls of the heterogeneous network. In addition to having the ability to connect to the called party, it can also connect calls from the PSTN to the enterprise IP telephony system. Media gateways play a key role in the entire IP PBX system. It not only enables the IP PBX system to connect to PSTN (Public Service Telephone Network, PSDN) users, but also increases the reliability of the entire system, enabling the system to handle emergency calls. The media gateway also needs to complete voice code conversion, communication protocol conversion and LAN-PSTN (local area network to PSTN) call setup/demolition.
3)局域网接入模块: 提供 IP PBX 系统各组件之间的连接, 在局域网的环 境中, 该模块可以是局域网交换机, 也可作为呼叫控制器的一个功能模块。  3) LAN access module: Provides the connection between the components of the IP PBX system. In the environment of the LAN, the module can be a LAN switch or a function module of the call controller.
4 )控制中心: 包括资源管理系统、 计费系统、 网管系统、 语音信箱等。 其 中, 资源管理系统主要执行地址翻译、 权限控制、 路由管理等功能; 计费系统 主要完成计费及结算功能; 网管系统完成对网络及 IP PBX运行的维护管理。  4) Control Center: Includes resource management system, billing system, network management system, voice mail, etc. Among them, the resource management system mainly performs functions such as address translation, authority control, and route management; the billing system mainly performs billing and settlement functions; the network management system completes maintenance and management of network and IP PBX operations.
5 )信令网关: 提供和其它信令网络的互通功能, 可以支持 H.323、 SS7 ( Signaling System 7, 7号信令,筒称 SS7 )、SIP( Session Initiation Protocol , 会话初始协议, 筒称 SIP) 、 Η· 248、 MGCP (Media Gateway Control Protocol, 媒体网关控制协议, 筒称 MGCP)的全部或者部分功能, 可以在单个设备上实现, 也可以在多个设备中实现, 也可以内嵌在呼叫控制器中实现。 6 )应用服务器: 为 IP PBX系统提供增值应用服务。 5) Signaling Gateway: Provides interworking functions with other signaling networks. It can support H.323, SS7 (Signaling System 7, Signaling No. 7, SS7), SIP (Session Initiation Protocol). All or part of the functions of SIP), Η248, MGCP (Media Gateway Control Protocol, MGCP) can be implemented on a single device, in multiple devices, or embedded in multiple devices. Implemented in the call controller. 6) Application Server: Provides value-added application services for IP PBX systems.
7 ) IP话机终端: 包括终端部分接续软件 PC Phone、 PC Phone+、 IP-Phone 等:  7) IP phone terminal: including terminal part connection software PC Phone, PC Phone+, IP-Phone, etc.:
a ) PC Phone: 多媒体 PC机, 或 PC机 +声卡 +不提供拨号功能的电话手柄; b ) PC Phone+: 具有普通电话的基本功能、 外观和使用方法, 它借助于 PC 机上的声卡传输语音信息, 通过串口完成拨号和其他控制;  a) PC Phone: multimedia PC, or PC + sound card + phone handle that does not provide dialing function; b) PC Phone+: has the basic functions, appearance and usage of ordinary phone, it transmits voice information by means of sound card on PC , through the serial port to complete dialing and other controls;
c ) IP-Phone: 直接接入 IP网络的电话机, 可以通过 RJ45接头直接接入网 络使用。  c) IP-Phone: The telephone directly connected to the IP network can be directly connected to the network through the RJ45 connector.
8 )数据终端: 用于数据通信目的的计算机。  8) Data terminal: A computer used for data communication purposes.
9 )模拟终端: 普通电话, 传真机等传统模拟设备。  9) Analog terminal: Traditional analog equipment such as ordinary telephones and fax machines.
IP PBX解决方案除提供给企业用户所有传统的 C4/C5和 PBX业务外, 还提 供丰富的新业务和应用, 会议服务器功能就是其中一个。  In addition to providing all the traditional C4/C5 and PBX services for enterprise users, the IP PBX solution provides a wealth of new services and applications. The conference server function is one of them.
目前会议系统的系统结构如图 2所示, 会议系统基于 IP PBX 系统, 在 IP PBX系统中加入了 MCU ( Mul t ipoint Contro l Uni t , 多点控制单元, 筒称 MCU ) , 用来进行会议控制与管理, 并利用 IP 网络, 结合视频、 音频和数据等多种途 径, 实现多个人之间的相互交流、 协商。 其中, MCU包括三部分:  The system structure of the conference system is shown in Figure 2. The conference system is based on the IP PBX system. The MCU (Mul t ipoint Contro l Uni t, multi-point control unit, MCU) is added to the IP PBX system for conferences. Control and management, and use IP networks to combine video, audio and data to achieve mutual communication and negotiation among multiple people. Among them, the MCU consists of three parts:
会议业务管理组件:负责会议的策略、 业务等高层属性的控制和管理; 信令控制组件:负责信令的传输、 控制、 交互等职能;  Conference service management component: responsible for the control and management of high-level attributes such as policies and services of the conference; signaling control component: responsible for the transmission, control, interaction and other functions of signaling;
媒体处理组件:负责媒体的接收、 编码、 协商、 处理等工作。  Media processing component: responsible for receiving, encoding, negotiating, processing, etc. of the media.
现有技术中, 会议建立流程如图 3所示, 会议的建立流程分为两阶段: In the prior art, the conference establishment process is shown in FIG. 3, and the conference establishment process is divided into two phases:
1 )会议的创建: 需要管理员或者有权限的用户预先创建会议(这里的创建 会议, 主要是预先申请会议资源也可以进一步预订会议时间等, 在会议的创建 过程中, MCU会为该会议分配相应的会议资源), 会议创建后便可以进行会议发 起流程。 1) Creation of a conference: An administrator or a user with permission is required to create a conference in advance (the conference is created here, mainly in advance, and the conference time can be further reserved. During the conference creation process, the MCU allocates the conference. The corresponding conference resource), after the conference is created, the conference initiation process can be performed.
2 )会议的启动: 每个需要参加会议的用户, 通过其用户终端拨打会议接入 号与会议服务器连接, 会议服务器对该终端进行验证后, 会议服务器返回确认 响应, 允许该用户终端加入会议。 在实现本发明的过程中, 发明人发现在上述现有技术的会议启动方法中, 存在如下缺陷: 2) Start of the conference: Each user who needs to participate in the conference dials the conference access number through the user terminal to connect with the conference server. After the conference server verifies the terminal, the conference server returns an acknowledgement response, allowing the user terminal to join the conference. In the process of implementing the present invention, the inventors have found that in the above-described prior art conference starting method, there are the following drawbacks:
每个用户都需要拨打会议接入号才能加入会议, 导致会议启动流程繁瑣, 先进入会议的用户需要等待后进入的用户, 并且对于用户来说, 加入过程比较 繁瑣, 操作容易出错, 不容易一次性成功的加入会议。 并且, 无法以语音的方 式向会议服务器发出会议邀请。 发明内容  Each user needs to dial the conference access number to join the conference. This causes the conference startup process to be cumbersome. The user who enters the conference first needs to wait for the user to enter, and for the user, the joining process is cumbersome, the operation is easy to make mistakes, and it is not easy to Sexually successful to join the meeting. Also, you cannot send a meeting invitation to the conference server in voice. Summary of the invention
本发明实施例的目的是提供呼叫用户加入会议的方法及装置, 实现以语音 的方式接收并识别会议邀请, 并呼叫新的会议成员加入该会议。  An object of the embodiments of the present invention is to provide a method and apparatus for a calling user to join a conference, to receive and identify a conference invitation in a voice manner, and to call a new conference member to join the conference.
为实现上述目的, 本发明实施例提供了一种呼叫用户加入会议的方法, 包 括:  To achieve the above objective, an embodiment of the present invention provides a method for a calling user to join a conference, including:
接收一个发起呼叫提示信息, 提示准备接收与会者的呼叫语音流; 接收与会者的呼叫语音流;  Receiving an initiating call prompt message, prompting to receive a call voice stream of the participant; receiving a call voice stream of the participant;
对所述呼叫语音流进行识别, 获得被叫用户号码;  Identifying the call voice stream to obtain a called subscriber number;
向所述被叫用户号码发出加入所述会议的邀请。  An invitation to join the conference is issued to the called subscriber number.
一种呼叫用户加入会议的装置, 包括:  A device for calling a user to join a conference, including:
呼叫提示信息接收模块, 用于接收发起呼叫提示信息, 并启动呼叫语音流 接收模块;  a call prompt information receiving module, configured to receive a call prompt information, and activate a call voice stream receiving module;
所述呼叫语音流接收模块, 用于接收与会者的呼叫命令语音流;  The call voice stream receiving module is configured to receive a call command voice stream of the participant;
呼叫语音流识别模块, 用于对所述呼叫命令语音流进行识别, 获得被叫用 户号码;  a call voice stream identification module, configured to identify the call command voice stream, and obtain a called user number;
会议的邀请发送模块, 用于根据所述呼叫命令语音流识别模块获得的被叫 用户号码, 向该用于向所述被叫用户号码发出加入所述会议的邀请。  The invitation sending module of the conference is configured to send an invitation to join the conference to the called subscriber number according to the called subscriber number obtained by the call flow voice stream identification module.
由上述技术方案可知, 本发明实施例通过接收与会者的呼叫语音流实现了 以语音的方式接收会议邀请, 并通过对呼叫语音流进行识别, 向识别获取的被 叫用户号码实现会议邀请, 并呼叫新的会议成员加入该会议, 从而使会议启动 流程更加筒单灵活。 According to the foregoing technical solution, the embodiment of the present invention implements receiving a conference invitation in a voice manner by receiving a call voice stream of the participant, and identifying the acquired voice stream by identifying the call voice stream. Calling the user number to implement the meeting invitation, and calling the new meeting member to join the meeting, so that the meeting starting process is more flexible and flexible.
下面通过附图和实施例, 对本发明的技术方案做进一步的详细描述。 附图说明  The technical solution of the present invention will be further described in detail below through the accompanying drawings and embodiments. DRAWINGS
图 1为现有技术的 IP PBX的系统结构示意图;  1 is a schematic structural diagram of a system of a prior art IP PBX;
图 2为现有技术的会议系统的系统结构示意图;  2 is a schematic structural diagram of a system of a prior art conference system;
图 3为现有技术的会议建立的信令流程图;  3 is a signaling flowchart of a conference setup established in the prior art;
图 4为本发明实施例一的呼叫用户加入会议的方法的流程图;  4 is a flowchart of a method for a calling user to join a conference according to Embodiment 1 of the present invention;
图 5为本发明实施例一的会议创建过程方式一的信令流程图;  FIG. 5 is a signaling flowchart of a mode 1 of a conference creation process according to Embodiment 1 of the present invention;
图 6为本发明实施例一的会议创建过程方式三的信令流程图;  6 is a signaling flowchart of mode 3 of a conference creation process according to Embodiment 1 of the present invention;
图 7为本发明呼叫用户加入会议的装置的实施例的结构示意图。 具体实施方式 实施例一  FIG. 7 is a schematic structural diagram of an embodiment of an apparatus for calling a user to join a conference according to the present invention. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
本发明呼叫用户加入会议的方法的实施例一的流程图, 如图 4 所示, 包括 如下步骤:  A flowchart of Embodiment 1 of the method for calling a user to join a conference, as shown in FIG. 4, includes the following steps:
步骤 1 01、 会议服务器接收一个发起呼叫提示信息;  Step 1 01: The conference server receives an initiation call prompt message;
步骤 102、 会议服务器接收与会者的呼叫命令语音流;  Step 102: The conference server receives a call command voice stream of the participant.
步骤 1 03、会议服务器对所述呼叫命令语音流进行识别,获得被叫用户号码; 会议服务器具体可以通过如下两种方式获得被叫用户号码:  Step 1 03: The conference server identifies the voice stream of the call command, and obtains the called subscriber number. The conference server may obtain the called subscriber number by using the following two methods:
方式一  method one
呼叫命令语音流包括被叫用户号码信息;  The call command voice stream includes the called subscriber number information;
会议服务器对呼叫命令语音流进行识别, 获得呼叫命令; 会议服务器从呼 叫命令中获得用户号码。  The conference server identifies the call command voice stream and obtains a call command; the conference server obtains the subscriber number from the call command.
方式二  Way two
呼叫语音流包括被叫用户号码查询信息; 会议服务器根据被叫用户查询信息查询用户号码信息表, 并获得被叫用户 号码。 The call voice stream includes the called subscriber number query information; The conference server queries the subscriber number information table according to the called subscriber query information, and obtains the called subscriber number.
其中, 用户号码信息表可以存储在企业电话服务器中, 也可以是会议服务 器创建电话会议时, 会议主席配置的会议用户列表。  The user number information table may be stored in the enterprise telephone server, or may be a conference user list configured by the conference chairperson when the conference server creates the conference call.
步骤 104、 向被叫用户号码发出加入所述会议的邀请。  Step 104: Send an invitation to join the conference to the called subscriber number.
在本实施例中, 可以通过命令行、 web界面、 特服号等操作方式启动会议服 务器的语音识别, 具体如下:  In this embodiment, the voice recognition of the conference server can be started through an operation manner such as a command line, a web interface, or a special service number, as follows:
如果通过命令行启动语音识别, 那么需要直接在服务器上操作或者远程登 陆到服务器上, 输入相应的命令, 会议服务器启动语音识别, 对特定的终端输 入的语音进行识别; 如果通过 web界面, 可以在任意电脑终端登陆到会议服务 器的 web 页面, 点击启动会议邀请、 语音识别的选项, 会议服务器启动语音识 别模块, 对特定的终端输入的语音进行识别; 如果通过特服号的方式, 加入会 议后在会议终端上拨打特服号, 会议服务器启动语音识别模块, 对特定的终端 输入的语音进行识别; 会议服务器可以通过播放提示音提醒用户说出要邀请的 人员名称, 以某个按键结束语音输入。 当把所有的与会者加入会议后, 会议主 席可以通过命令行、 web界面、 拨打特服号来关闭语音识别。 的会议创建过程可以采用如下方式:  If voice recognition is started through the command line, then you need to operate directly on the server or remotely log in to the server, input the corresponding command, the conference server starts voice recognition, and recognizes the voice input by the specific terminal; if through the web interface, you can Any computer terminal logs in to the web page of the conference server, clicks the option to start the conference invitation and voice recognition, and the conference server starts the voice recognition module to identify the voice input by the specific terminal; if the special service number is adopted, the conference is added after the conference The conference terminal dials the special service number, and the conference server starts the voice recognition module to identify the voice input by the specific terminal. The conference server can prompt the user to say the name of the person to be invited by playing the prompt tone, and end the voice input with a certain button. When all participants are added to the conference, the conference chairperson can turn off voice recognition through the command line, web interface, and dialing the service number. The conference creation process can be as follows:
方式一  method one
如图 5 所示, 用户首先通过电话终端拨打会议接入号码进入会议, 会议服 务器将主动把其存储的会议用户列表中 (其中记载了参与会议的成员及对应的 用户终端的信息) 的用户呼入到该会议中。 会议成员通信录中的每个用户都可 以设置几个电话号码, 会议服务器将逐个拨打该用户名下的电话号码, 直到接 通并确认该用户对应的电话终端与该会议建立了通信连接为止。  As shown in Figure 5, the user first dials the conference access number through the telephone terminal to enter the conference. The conference server will actively call the user in the conference user list stored therein (in which the information of the members participating in the conference and the corresponding user terminal are recorded). Enter the meeting. Each user in the meeting member address book can set several phone numbers, and the conference server will dial the phone number under the user name one by one until it is connected and confirms that the corresponding phone terminal of the user establishes a communication connection with the conference.
方式二  Way two
与方式一的不同之处在于, 用户通过网页(web )方式或者 GUI ( Graphi ca l Us er Interface , 图形用户接口, 筒称 GUI )方式登录会议服务器, 通过界面 操作触发会议启动流程, 进一步地可以通过界面操作编辑会议用户列表中的信 息, 添加或删除会议成员。 具体流程如下: The difference from the first method is that the user logs in to the conference server through a web page or a GUI (Graphic User Interface, GUI). The operation triggers the conference startup process. Further, you can edit the information in the conference user list through interface operations, and add or delete conference members. The specific process is as follows:
用户通过网页或者 GUI 方式登录会议服务器, 打开会议用户列表, 选中某 些用户, 启动会议, 会议服务器主动将这些用户呼叫到指定会议中。 用户可以 把自己加载到会议用户列表中, 让会议服务器把自己呼叫到会议中, 或者自己 通过电话终端主动拨打会议号码接入到会议中。 本实施例中提供的会议启动模 式可以精确的选择需要参加会议的用户, 具有很大的灵活性。  The user logs in to the conference server through the web page or GUI, opens the conference user list, selects some users, and starts the conference. The conference server actively calls these users to the specified conference. The user can load himself into the conference user list, let the conference server call himself to the conference, or dial the conference number through the telephone terminal to access the conference. The conference startup mode provided in this embodiment can accurately select the users who need to participate in the conference, and has great flexibility.
对于此种方式, 用户操作的灵活性较大, 进一步地来说, 有权限的用户 (例 如管理员或者其他有权限创建会议的用户) , 可以通过网页或者 GUI 方式登录 会议服务器, 进行会议创建, 会议服务器为该会议分配相应的会议资源, 并且 进行会议成员的设定等。  In this way, the user has greater flexibility in operation. Further, a privileged user (such as an administrator or other user who has permission to create a conference) can log in to the conference server through a web page or a GUI to create a conference. The conference server allocates corresponding conference resources for the conference, and performs setting of conference members and the like.
方式三  Way three
引入了会议定时模式, 由会议服务器直接发起启动会议流程。 如图 6所示, 具体流程如下:  The conference timing mode is introduced, and the conference server directly initiates the conference initiation process. As shown in Figure 6, the specific process is as follows:
在预定会议时, 配置会议成员通信录和会议开始时间, 当到达和会议开始 时间时, 会议服务器按照会议用户列表中配置的参与会议的用户信息, 拨打用 户终端的电话号码, 主动呼叫用户, 将用户加入到会议中。  When the conference is scheduled, the conference member address book and the conference start time are configured. When the conference and the conference start time are reached, the conference server dials the telephone number of the user terminal according to the user information of the conference participant configured in the conference user list, and actively calls the user. The user joins the meeting.
实施例二  Embodiment 2
与上述实施例一不同之处在于, 加入了验证机制, 在步骤 102之前还包括: 返回用户验证提示, 具体可以是语音提示等, 要求用户输入相关的验证信 息, 验证信息可以为会议主席预先设定的密码等;  The difference from the first embodiment is that the verification mechanism is added. Before the step 102, the method further includes: returning a user verification prompt, which may be a voice prompt, etc., requiring the user to input relevant verification information, and the verification information may be preset for the conference chairperson. a password, etc.
接收用户发送的验证信息, 并对用户身份进行验证。 如果验证通过, 则继 续执行步骤 102 , 否则, 返回身份验证失败的信息, 结束实施例一的流程。  Receive authentication information sent by the user and verify the identity of the user. If the verification is passed, step 102 is continued; otherwise, the information of the authentication failure is returned, and the process of the first embodiment is ended.
实施例三  Embodiment 3
如图 7所示, 其为本发明呼叫用户加入会议的装置的实施例的结构示意图, 包括:  FIG. 7 is a schematic structural diagram of an embodiment of an apparatus for calling a user to join a conference according to the present invention, including:
呼叫提示信息接收模块 1 , 用于接收发起呼叫提示信息, 并启动呼叫语音流 接收模块; The call prompt information receiving module 1 is configured to receive the call prompt information and initiate the call voice flow. Receiving module
呼叫语音流接收模块 2 , 用于接收与会者的呼叫命令语音流;  a call voice stream receiving module 2, configured to receive a call command voice stream of the participant;
呼叫语音流识别模块 3 , 用于对呼叫命令语音流进行识别, 获得被叫用户号 码;  a call voice stream identification module 3, configured to identify a call command voice stream, and obtain a called user number;
会议的邀请发送模块 4 ,用于根据呼叫命令语音流识别模块获得的被叫用户 号码, 向该被叫用户号码发出加入所述会议的邀请。  The invitation sending module 4 of the conference is configured to send an invitation to join the conference to the called subscriber number according to the called subscriber number obtained by the call command voice stream identification module.
呼叫语音流识别模块可以具体包括:  The call voice stream identification module may specifically include:
呼叫命令提取模块, 用于从所述呼叫语音流中获取呼叫命令;  a call command extraction module, configured to acquire a call command from the call voice stream;
用户号码提取模块, 用于从呼叫命令提取模块获取的呼叫命令中获取用户 号码。  The user number extraction module is configured to obtain a user number from a call command obtained by the call command extraction module.
呼叫语音流识别模块还可以具体包括:  The call voice stream identification module may further include:
被叫用户号码查询信息提取模块, 用于从呼叫语音流中获取;  The called subscriber number query information extraction module is configured to be obtained from the call voice stream;
用户号码信息表查询模块, 用于根据被叫用户号码查询信息查询用户号码 信息表, 并获得被叫用户号码。  The user number information table querying module is configured to query the user number information table according to the called user number query information, and obtain the called user number.
其中, 用户号码信息表可以位于企业电话服务器中; 用户号码信息表也可 以为会议用户列表。  The user number information table may be located in the enterprise telephone server; the user number information table may also be a conference user list.
在本实施例的装置中还可以包括用户验证模块, 用于返回用户验证提示, 接收用户发送的验证信息, 并对用户身份进行验证。  The device in this embodiment may further include a user verification module, configured to return a user verification prompt, receive verification information sent by the user, and verify the identity of the user.
本领域普通技术人员可以理解实现上述实施例方法中的全部或部分流 程, 是可以通过计算机程序来指令相关的硬件来完成, 所述的程序可存储于 一计算机可读取存储介质中, 该程序在执行时, 可包括如上述各方法的实施 例的流程。其中,所述的存储介质可为磁碟、光盘、只读存储记忆体( Read-Only Memory, ROM ) 或随机存^ ^己忆体 ( Random Access Memory, RAM ) 等。  A person skilled in the art can understand that all or part of the process of implementing the above embodiment method can be completed by a computer program to instruct related hardware, and the program can be stored in a computer readable storage medium. In execution, the flow of an embodiment of the methods as described above may be included. The storage medium may be a magnetic disk, an optical disk, a read-only memory (ROM), or a random access memory (RAM).
最后应说明的是: 以上实施例仅用以说明本发明的技术方案而非对其进行 限制, 尽管参照较佳实施例对本发明进行了详细的说明, 本领域的普通技术人 员应当理解: 其依然可以对本发明的技术方案进行修改或者等同替换, 而这些 修改或者等同替换亦不能使修改后的技术方案脱离本发明技术方案的精神和范 围。  It should be noted that the above embodiments are only intended to illustrate the technical solutions of the present invention and are not to be construed as limiting the embodiments of the present invention. The technical solutions of the present invention may be modified or equivalently substituted, and the modified technical solutions may not deviate from the spirit and scope of the technical solutions of the present invention.

Claims

权 利 要 求 Rights request
1、 一种呼叫用户加入会议的方法, 其特征在于, 包括:  A method for calling a user to join a conference, which is characterized by comprising:
接收一个发起呼叫提示信息, 提示准备接收与会者的呼叫语音流; 接收与会者的呼叫语音流;  Receiving an initiating call prompt message, prompting to receive a call voice stream of the participant; receiving a call voice stream of the participant;
对所述呼叫语音流进行识别, 获得被叫用户号码;  Identifying the call voice stream to obtain a called subscriber number;
向所述被叫用户号码发出加入所述会议的邀请。  An invitation to join the conference is issued to the called subscriber number.
2、 如权利要求 1所述的方法, 其特征在于:  2. The method of claim 1 wherein:
呼叫语音流包括被叫用户号码信息;  The call voice stream includes the called subscriber number information;
所述对所述呼叫语音流进行识别, 获得被叫用户号码的步骤包括: 对所述呼叫语音流进行识别, 获得呼叫命令;  The step of identifying the call voice stream and obtaining the called subscriber number includes: identifying the call voice stream, and obtaining a call command;
从所述呼叫命令中获得用户号码。  A subscriber number is obtained from the call command.
3、 如权利要求 1所述的方法, 其特征在于:  3. The method of claim 1 wherein:
呼叫语音流包括被叫用户号码查询信息;  The call voice stream includes the called subscriber number query information;
所述对所述呼叫语音流进行识别, 获得被叫用户号码的步骤包括: 所述对所述呼叫语音流进行识别, 获得被叫用户号码查询信息; 根据所述被叫用户查询信息查询用户号码信息表, 并获得被叫用户号码。 The step of identifying the call voice stream and obtaining the called subscriber number includes: identifying the call voice stream, obtaining the called subscriber number query information; and querying the subscriber number according to the called subscriber query information Information sheet, and get the called user number.
4、 如权利要求 3所述的方法, 其特征在于, 所述用户号码信息表位于企业 电话服务器中。 4. The method of claim 3, wherein the subscriber number information table is located in an enterprise telephone server.
5、 如权利要求 3所述的方法, 其特征在于, 所述用户号码信息表为会议用 户列表。  The method according to claim 3, wherein the user number information table is a conference user list.
6、 如权利要求 5所述的方法, 其特征在于, 所述方法还包括:  The method of claim 5, wherein the method further comprises:
当会议创建者创建会议时, 接收会议创建者提供的会议用户列表; 存储所述会议用户列表。  When the meeting creator creates a meeting, receiving a meeting user list provided by the meeting creator; storing the meeting user list.
7、 如权利要求 1所述的方法, 其特征在于, 接收与会者的呼叫命令语音流 步骤前还包括:  7. The method according to claim 1, wherein the step of receiving the call command voice stream of the participant further comprises:
返回用户验证提示; 接收用户发送的验证信息, 并对用户身份进行验证。 Return user verification prompts; Receive authentication information sent by the user and verify the identity of the user.
8、 一种呼叫用户加入会议的装置, 其特征在于, 包括:  8. A device for calling a user to join a conference, comprising:
呼叫提示信息接收模块, 用于接收发起呼叫提示信息, 并启动呼叫语音流 接收模块;  a call prompt information receiving module, configured to receive a call prompt information, and activate a call voice stream receiving module;
所述呼叫语音流接收模块, 用于接收与会者的呼叫命令语音流;  The call voice stream receiving module is configured to receive a call command voice stream of the participant;
呼叫语音流识别模块, 用于对所述呼叫命令语音流进行识别, 获得被叫用 户号码;  a call voice stream identification module, configured to identify the call command voice stream, and obtain a called user number;
会议的邀请发送模块, 用于根据所述呼叫命令语音流识别模块获得的被叫 用户号码, 向该用于向所述被叫用户号码发出加入所述会议的邀请。  The invitation sending module of the conference is configured to send an invitation to join the conference to the called subscriber number according to the called subscriber number obtained by the call flow voice stream identification module.
9、 根据权利要求 8所述的装置, 其特征在于, 所述呼叫语音流识别模块包 括:  9. The apparatus according to claim 8, wherein the call voice stream identification module comprises:
呼叫命令提取模块, 用于从所述呼叫语音流中获取呼叫命令;  a call command extraction module, configured to acquire a call command from the call voice stream;
用户号码提取模块, 用于从呼叫命令提取模块获取的呼叫命令中获取用户 号码。  The user number extraction module is configured to obtain a user number from a call command obtained by the call command extraction module.
10、 根据权利要求 8所述的装置, 其特征在于, 所述呼叫语音流识别模块 包括:  The device according to claim 8, wherein the call voice stream identification module comprises:
被叫用户号码查询信息提取模块, 用于从所述呼叫语音流中获取; 用户号码信息表查询模块, 用于根据所述被叫用户号码查询信息查询用户 号码信息表, 并获得被叫用户号码。  The called user number query information extraction module is configured to be obtained from the call voice stream; the user number information table querying module is configured to query the user number information table according to the called user number query information, and obtain the called user number .
11、 根据权利要求 1 0所述的装置, 其特征在于, 所述用户号码信息表位于 企业电话服务器中。  11. Apparatus according to claim 10 wherein said subscriber number information table is located in an enterprise telephone server.
12、 根据权利要求 1 0所述的装置, 其特征在于, 所述用户号码信息表为会 议用户列表。  12. The apparatus according to claim 10, wherein the subscriber number information table is a conference subscriber list.
1 3、 根据权利要求 8所述的装置, 其特征在于, 还包括用户验证模块, 用 于返回用户验证提示, 接收用户发送的验证信息, 并对用户身份进行验证。  The device according to claim 8, further comprising a user verification module, configured to return a user verification prompt, receive verification information sent by the user, and verify the identity of the user.
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