WO2018120756A1 - 一种自动挂断通话的方法和系统 - Google Patents

一种自动挂断通话的方法和系统 Download PDF

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Publication number
WO2018120756A1
WO2018120756A1 PCT/CN2017/092919 CN2017092919W WO2018120756A1 WO 2018120756 A1 WO2018120756 A1 WO 2018120756A1 CN 2017092919 W CN2017092919 W CN 2017092919W WO 2018120756 A1 WO2018120756 A1 WO 2018120756A1
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Prior art keywords
call
parties
condition
server
module
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PCT/CN2017/092919
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English (en)
French (fr)
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陈文迪
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深圳有麦科技有限公司
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Publication of WO2018120756A1 publication Critical patent/WO2018120756A1/zh

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/82Line monitoring circuits for call progress or status discrimination
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72448User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72448User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
    • H04M1/72463User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions to restrict the functionality of the device

Definitions

  • the present invention relates to the technical field of communication, and more particularly to a method and system for automatically hanging up a call.
  • the present invention is implemented in such a manner that a method for automatically hanging up a call includes the following steps:
  • the server acquires information of the called party of the calling party, and the two parties make a call, and notify the server through the telecommunication network;
  • the two parties are returned to make a call.
  • the setting condition includes whether the audio signal is present or not and the volume.
  • step of the server acquiring information of the called party of the called party by the primary calling party includes the following specific steps:
  • the server obtains information that the primary caller calls the called party and the called party connects the call of the primary calling party;
  • the two parties make a call
  • the server receives the primary calling terminal and the called terminal respectively transmits their respective audios.
  • the current call condition satisfies the set condition
  • the server further includes:
  • the main caller is called to call the called party, and the two parties perform a call.
  • the step of automatically hanging up the call of both parties includes the following specific steps:
  • the server blocks the audio transmission of both parties
  • the server sends a hang up command to the primary calling terminal and the called terminal.
  • the invention also provides a system for automatically hanging up a call, comprising a call unit, a judging unit and a hanging unit;
  • the call unit is configured to acquire, by the server, information of the called party of the calling party, and notify the server through the telecommunication network to connect with the determining unit;
  • the determining unit is configured to determine whether the current call condition meets the set condition, and is respectively connected to the hanging unit and the calling unit;
  • the hanging unit is configured to automatically hang up the conversation between the two parties.
  • the call unit includes a call module, a call module, and a sending module
  • the calling module is configured to acquire, by the server, information that the primary calling party calls the called party and the called party connects the call of the primary calling party, and is connected to the calling module;
  • the call module is configured to perform a call between the two parties, and is respectively connected to the determining unit and the sending module;
  • the sending module is configured to receive, by the server, the primary calling terminal and the called terminal respectively send respective audios, and connect with the determining unit.
  • the determining unit includes an audio signal determining module and a volume determining module
  • the audio signal judging module is configured to determine whether the audio signals of the two parties satisfy the audio amount when no call is in progress, and are respectively connected to the volume judging module, the sending module, and the volume judging module;
  • the volume determining module is configured to determine whether the current volume level is lower than the set volume value, and is respectively connected to the calling module and the hanging unit.
  • system for automatically hanging up a call further includes a setting determining unit and a setting unit;
  • the setting determining unit is configured to determine whether a pre-set condition is required, and is respectively connected to the setting unit and the calling module;
  • the setting unit is configured to set a volume value, and is stored in the server and connected to the calling module.
  • the present invention provides a method for automatically hanging up a call, by monitoring the audio signal and the volume of the two parties during the call, when the audio signal and the volume meet the set value.
  • the call is automatically hanged up, and when the two parties do not make a sound for a long time, the two parties' calls are automatically hanged, the useless charge is reduced, and the missed connection of the important call can be avoided.
  • FIG. 1 is a flowchart of a method for automatically hanging up a call according to Embodiment 1 of the present invention
  • FIG. 3 is a specific flowchart of determining whether a current call condition satisfies a set condition according to Embodiment 1 of the present invention
  • FIG. 5 is a structural block diagram of a system for automatically hanging up a call according to Embodiment 1 of the present invention.
  • FIG. 6 is a structural block diagram of a method for automatically hanging up a call according to Embodiment 2 of the present invention.
  • the method for automatically hanging up a call provided by this embodiment can be used in the process of communication, and when the two parties do not emit sound for a long time, the two parties automatically hang up the call, thereby reducing the generation of useless deduction, and avoiding important The missed connection of the phone occurred.
  • FIG. 1 it is a first embodiment, a method for automatically hanging up a call, including the following specific steps:
  • the server acquires information of the called party of the calling party, and the two parties make a call, and notify the server through the telecommunication network;
  • the two parties are returned to make a call.
  • the server obtains the information of the called party of the calling party, and the two parties make a call, and notify the server through the telecommunication network, specifically through the communication network of the communication carrier (SMS center, USSD, service website, voice platform, etc.). Receive access information.
  • a fixed communication network including NGN
  • NGN NGN
  • 2G mobile communication network GSM/CDMA
  • TD-CDMA 3G mobile communication network
  • the signaling involved in this process is SS7, SIGTRAN, ISDN, SIP, and H.323 signaling.
  • the server acquires information of the called party of the calling party, and the steps of the two parties to perform the call include the following specific steps:
  • the server acquires information that the primary calling party calls the called party and the called party connects the call of the primary calling party.
  • the server receives the primary calling terminal and the called terminal respectively send respective audios.
  • step S123 the step of the server receiving the main calling terminal and the called terminal respectively transmitting respective audios is to monitor the respective transmitted audio, thereby providing a judgment condition for automatically hanging up the call.
  • step S13 it is determined whether the current call condition satisfies the set condition, and the set conditions herein include whether the audio signal is present or not.
  • the step of determining whether the current call condition satisfies the set condition specifically includes the following specific steps:
  • the current call condition satisfies the set condition.
  • step S131 the step of determining whether the respective audio signals of the two parties satisfy the audio amount when there is no call, wherein the audio signal in a certain period of time is the amount of audio when the call is not satisfied, and the audio signal is zero indicates that both sides are No sound.
  • step S133 if the respective audio signals are not satisfying the amount of audio when no call is in progress, it is determined whether the current volume is lower than the set volume value, and it is judged from the magnitude of the volume that the current volume can be in a chaotic environment.
  • the chaotic micro-sounds did not make a sound on both sides.
  • step S13 determines whether the current call condition meets the set condition, and can be implemented in two ways:
  • the isup is taken as an example. After receiving the ANM signaling, If there is no REL signaling in the T2 period, it can be used as an abnormal call, and the switch end is notified to check whether the PCM line is hanged and automatically hang up.
  • the step of automatically hanging up the two parties' calls includes the following specific steps:
  • the server blocks audio transmission between the two parties
  • the server sends a hang up command to the primary calling terminal and the called terminal.
  • the two parties monitor the audio signal and the volume of the volume during the call, and automatically hang up the call when the audio signal and the volume meet the set conditions.
  • the two parties' calls are automatically hanged up, the useless charge is reduced, and the leakage of important calls can be avoided.
  • FIG. 5 it is a system for automatically hanging up a call provided by the first embodiment, including a call unit, a judging unit, and a hanging unit;
  • a call unit configured to acquire, by the server, information of the called party of the primary calling party, and notify the server through the telecommunication network to connect with the determining unit;
  • the determining unit is configured to determine whether the current call condition satisfies the set condition, and is respectively connected to the hanging unit and the calling unit.
  • the call unit includes a call module 30, a call module 31, and a sending module 32;
  • the calling module 30 is configured to acquire, by the server, information that the primary calling party calls the called party and the called party connects the call of the primary calling party, and is connected to the calling module 31;
  • the calling module 31 is configured to perform a call between the two parties, and is respectively connected to the determining unit and the sending module 32;
  • the sending module 32 is configured to receive, by the server, the primary calling terminal and the called terminal respectively send respective audios, and connect with the determining unit.
  • the determining unit includes an audio signal determining module 40 and a volume determining module 41;
  • the audio signal determining module 40 is configured to determine whether the audio signals of the two parties satisfy the audio amount when no call is in progress, and are respectively connected to the hanging unit, the sending module 32, and the volume determining module 41;
  • the volume determining module 41 is configured to determine whether the current volume level is lower than the set volume value, and is respectively connected to the calling module 31 and the hanging unit.
  • a system for automatically hanging up a call further includes a setting determining unit 10 and a setting unit 20;
  • the setting determining unit 10 is configured to determine whether a pre-set condition is required, and is respectively connected to the setting unit 20 and the calling module 30;
  • the setting unit 20 is configured to set a volume value, and is stored in the server and connected to the calling module 30.
  • the hangup unit includes a blocking module 50 and a sending instruction module 51;
  • the blocking module 50 is configured to prevent the audio transmission of both parties by the server, and is connected to the audio signal determining module 40 and the volume determining module 41;
  • the sending instruction module 51 is configured to send a hang up command to the main calling terminal and the called terminal by the server, and is connected to the blocking module 50.
  • the call unit is provided for both parties to monitor the audio signal and the volume of the two parties during the call, and hang up when the audio signal and the volume level satisfy the set condition.
  • the unit automatically hangs up the call, and when the two parties do not make a sound for a long time, the call between the two parties is automatically hanged, the useless charge is reduced, and the missed connection of the important call can be avoided.
  • FIG. 6 it is a second embodiment, a method for automatically hanging up a call, including the following specific steps:
  • the server acquires information of the called party of the calling party, and the two parties make a call, and notify the server through the telecommunication network;
  • the two parties are returned to the call.
  • the above S24 step mainly determines that the user needs to pause the current call because of something, and does not want to hang up the call, thereby improving the integrity of the entire judgment.

Abstract

本发明涉及一种自动挂断通话的方法和系统,该方法包括服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话,并通过电信网通知服务器;判断当前的通话条件是否满足设定的条件;若当前的通话条件满足设定的条件,则自动挂断双方的通话;或者,若当前的通话条件不满足设定的条件,则返回所述双方进行通话。本发明通过双方在通话的过程中,对双方的音频信号以及音量的大小进行监测,当音频信号与音量大小满足设定的条件时,则自动挂断通话,实现双方长时间未发出声音的时候,自动挂断双方的通话,减少无用扣费的产生,并且能够避免重要电话的漏接情况发生。

Description

一种自动挂断通话的方法和系统
技术领域
本发明涉及通讯的技术领域,尤其是一种自动挂断通话的方法和系统。
背景技术
目前在通话的过程中,有时候由于双方都很忙碌,因此会忘记挂断通话,这样,容易被扣费,并且,长时间未挂断容易导致后续电话的接入,造成重要电话的漏接情况发生。
因此,有必要设计一种自动挂断通话的方法,实现双方长时间未发出声音的时候,自动挂断双方的通话,减少无用扣费的产生,并且能够避免重要电话的漏接情况发生。
发明内容
本发明的目的在于克服现有技术的缺陷,提供一种自动挂断通话的方法和系统。
本发明是这样实现的,一种自动挂断通话的方法,包括以下步骤:
服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话,并通过电信网通知服务器;
判断当前的通话条件是否满足设定的条件;
若当前的通话条件满足设定的条件,则自动挂断双方的通话;
或者,若当前的通话条件不满足设定的条件,则返回所述双方进行通话。
进一步地,所述判断当前的通话条件是否满足设定的条件的步骤中,所述设定条件包括音频信号的有否以及音量的大小。
进一步地,所述服务器获取主呼叫方呼叫被呼叫方的信息的步骤,包括以下具体步骤:
服务器获取主呼叫方呼叫被呼叫方且被呼叫方接通主呼叫方的呼叫的信息;
双方进行通话;
服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频。
进一步地,
判断双方各自的音频信号是否满足无通话时的音频量;
若各自的音频信号为满足无通话时的音频量,则当前的通话条件满足设定的条件;
若各自的音频信号不为满足无通话时的音频量,则判断当前的音量是否低于设定的音量值;
若是,则当前的通话条件满足设定的条件;
若不是,则当前的通话条件不满足设定的条件。
进一步地,所述服务器获取主呼叫方呼叫被呼叫方的信息的步骤之前,还包括:
判断是否需要预先设定条件;
若需要,则设定音量值,并保存于服务器内;
若不需要,则进入所述主呼叫方呼叫被呼叫方,双方进行通话的步骤。
进一步地,所述若当前的通话条件满足设定的条件,则自动挂断双方的通话的步骤,具体包括以下具体步骤:
服务器阻止双方的音频输送;
服务器发送挂断指令至主呼叫终端以及被呼叫终端。
本发明还提供了一种自动挂断通话的系统,包括通话单元、判断单元以及挂断单元;
所述通话单元,用于服务器获取主呼叫方呼叫被呼叫方的信息,并通过电信网通知服务器,与所述判断单元连接;
所述判断单元,用于判断当前的通话条件是否满足设定的条件,分别与所述挂断单元以及所述通话单元连接;
所述挂断单元,用于自动挂断双方的通话。
进一步地,所述通话单元包括呼叫模块、通话模块以及发送模块;
所述呼叫模块,用于服务器获取主呼叫方呼叫被呼叫方且被呼叫方接通主呼叫方的呼叫的信息,与所述通话模块连接;
所述通话模块,用于双方进行通话,分别与所述判断单元以及发送模块连接;
所述发送模块,用于服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频,与所述判断单元连接。
进一步地,所述判断单元包括音频信号判断模块以及音量判断模块;
所述音频信号判断模块,用于判断双方各自的音频信号是否满足无通话时的音频量,分别与所述音量判断模块、所述发送模块以及所述音量判断模块连接;
所述音量判断模块,用于判断当前的音量大小是否低于设定的音量值,分别与所述通话模块以及所述挂断单元连接。
进一步地,所述一种自动挂断通话的系统还包括设定判断单元以及设定单元;
所述设定判断单元,用于判断是否需要预先设定条件,分别与所述设定单元以及所述呼叫模块连接;
所述设定单元,用于设定音量值,并保存于服务器内,与所述呼叫模块连接。
与现有技术相比,本发明提供的一种自动挂断通话的方法,通过双方在通话的过程中,对双方的音频信号以及音量的大小进行监测,当音频信号与音量大小满足设定的条件时,则自动挂断通话,实现双方长时间未发出声音的时候,自动挂断双方的通话,减少无用扣费的产生,并且能够避免重要电话的漏接情况发生。
附图说明
图1是本发明实施例一提供的一种自动挂断通话的方法的流程图;
图2是本发明实施例一提供的双方进行通话的具体流程图;
图3是本发明实施例一提供的判断当前的通话条件是否满足设定的条件的具体流程图;
图4是本发明实施例一提供的自动挂断双方的通话的具体流程图;
图5是本发明实施例一提供的一种自动挂断通话的系统的结构框图;
图6是本发明实施例二提供的一种自动挂断通话的方法的结构框图;
具体实施方式
为了使本发明的目的、技术方案及优点更加清楚明白,以下结合附图及实施例,对本发明进行进一步详细说明。应当理解,此处所描述的具体实施例仅仅用以解释本发明,并不用于限定本发明。
以下结合具体实施例对本发明的实现进行详细的描述。
参照图1~6,为本发明提供的较佳实施例。
本实施例提供的一种自动挂断通话的方法,可以运用在通讯的过程中,实现双方长时间未发出声音的时候,自动挂断双方的通话,减少无用扣费的产生,并且能够避免重要电话的漏接情况发生。
如图1所示,是实施例一,一种自动挂断通话的方法,包括以下具体步骤:
S10、判断是否需要预先设定条件;
S11、若需要,则设定音量值,并保存于服务器内;
S12、或者,若不需要,服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话,并通过电信网通知服务器;
S13、判断当前的通话条件是否满足设定的条件;
S14、若当前的通话条件满足设定的条件,则自动挂断双方的通话;
或者,若当前的通话条件不满足设定的条件,则返回所述双方进行通话。
上述的S12步骤,服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话,并通过电信网通知服务器,具体通过通信运营商的通信网络(短信中心、USSD、业务网站、声讯平台等),接收获取信息。
另外,还可以使用固定通信网(含NGN)信令网、2G移动通信网(GSM/CDMA)信令网、3G移动通信网(TD-CDMA)信令网来采集信息。
这个过程涉及的信令有SS7、SIGTRAN、ISDN、SIP和H.323信令。
如图2所示,上述的S12步骤,所述服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话的步骤,包括以下具体步骤:
S121、服务器获取主呼叫方呼叫被呼叫方且被呼叫方接通主呼叫方的呼叫的信息;
S122、双方进行通话;
S123、服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频。
具体地,S123步骤,所述服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频的步骤,是为了对各自发送的音频进行监测,从而为自动挂断通话提供判断条件。
S13步骤,判断当前的通话条件是否满足设定的条件的步骤,这里的设定的条件包括音频信号的有否以及音量的大小。
更进一步的,如图3所示,上述的S13步骤,判断当前的通话条件是否满足设定的条件的步骤具体包括以下具体步骤:
S131、判断双方各自的音频信号是否满足无通话时的音频量;
S132、若各自的音频信号满足无通话时的音频量,则当前的通话条件满足设定的条件;
S133、若各自的音频信号不为满足无通话时的音频量,则判断当前的音量是否低于设定的音量值;
S134、若不是,则当前的通话条件不满足设定的条件;
或者,若是,则当前的通话条件满足设定的条件。
上述的S131步骤,判断双方各自的音频信号是否满足无通话时的音频量的步骤,这里是在一定时间段内的音频信号为满足无通话时的音频量,并且音频信号为零时表示双方都没有发出声音。
另外,S133步骤,若各自的音频信号不为满足无通话时的音频量,则判断当前的音量是否低于设定的音量值,从音量的大小来判断当前可以在混乱的环境中,存在一些混乱的微量声音,双方并没有发出声音。
具体地,S13步骤判断当前的通话条件是否满足设定的条件,可以通过两种方式实现:
1). 安装在用户手机终端上的软件:
软件检测用户的通话语音信号, 在满足条件(如超过周期T1=5分钟)
后,是否有语音信号, 如果没有,则向手机系统报告挂断此次通话;
2). 网络端的判断:
对于通话超长(T2=5小时), 通过信令监测,以isup为例,在收到ANM信令之后, T2周期没有REL信令,则可以作为异常的呼叫,通知交换机端检查是否PCM线路挂死,自动挂断处理。
更进一步的,如图4所示,上述的S14的步骤,若当前的通话条件满足设定的条件,则自动挂断双方的通话的步骤,具体包括以下具体步骤:
S141、服务器阻止双方的音频输送;
S142、服务器发送挂断指令至主呼叫终端以及被呼叫终端。
上述的一种自动挂断通话的方法,通过双方在通话的过程中,对双方的音频信号以及音量的大小进行监测,当音频信号与音量大小满足设定的条件时,则自动挂断通话,实现双方长时间未发出声音的时候,自动挂断双方的通话,减少无用扣费的产生,并且能够避免重要电话的漏接情况发生。
如图5所示,是本实施例一提供的一种自动挂断通话的系统,包括通话单元、判断单元以及挂断单元;
通话单元,用于服务器获取主呼叫方呼叫被呼叫方的信息,并通过电信网通知服务器,与所述判断单元连接;
判断单元,用于判断当前的通话条件是否满足设定的条件,分别与所述挂断单元以及所述通话单元连接。
挂断单元,用于自动挂断双方的通话。
更进一步的,通话单元包括呼叫模块30、通话模块31以及发送模块32;
呼叫模块30,用于服务器获取主呼叫方呼叫被呼叫方且被呼叫方接通主呼叫方的呼叫的信息,与所述通话模块31连接;
通话模块31,用于双方进行通话,分别与所述判断单元以及发送模块32连接;
发送模块32,用于服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频,与所述判断单元连接。
另外,判断单元包括音频信号判断模块40以及音量判断模块41;
音频信号判断模块40,用于判断双方各自的音频信号是否满足无通话时的音频量,分别与挂断单元、所述发送模块32以及所述音量判断模块41连接;
音量判断模块41,用于判断当前的音量大小是否低于设定的音量值,分别与所述通话模块31以及所述挂断单元连接。
更进一步的,一种自动挂断通话的系统还包括设定判断单元10以及设定单元20;
所述设定判断单元10,用于判断是否需要预先设定条件,分别与所述设定单元20以及所述呼叫模块30连接;
所述设定单元20,用于设定音量值,并保存于服务器内,与所述呼叫模块30连接。
具体地,挂断单元包括阻断模块50以及发送指令模块51;
阻断模块50,用于服务器阻止双方的音频输送,与音频信号判断模块40以及音量判断模块41连接;
发送指令模块51,用于服务器发送挂断指令至主呼叫终端以及被呼叫终端,与所述阻断模块50连接。
上述的一种自动挂断通话的方法,通过通话单元供双方在通话的过程中,对双方的音频信号以及音量的大小进行监测,当音频信号与音量大小满足设定的条件时,则挂断单元自动挂断通话,实现双方长时间未发出声音的时候,自动挂断双方的通话,减少无用扣费的产生,并且能够避免重要电话的漏接情况发生。
如图6所示,是实施例二,一种自动挂断通话的方法,包括以下具体步骤:
S20、判断是否需要预先设定条件;
S21、若需要,则设定音量值,并保存于服务器内;
S22、或者,若不需要,服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话,并通过电信网通知服务器;
S23、判断当前的通话条件是否满足设定的条件;
S24、若当前的通话条件满足设定的条件,则判断在满足设定的条件之前,其中一方是否提及暂停通话的信息;
S25、若有,自动挂断双方的通话;
或者,若当前的通话条件不满足设定的条件或者若其中一方没有提及暂停通话的信息,则返回所述双方进行通话。
上述的S20、S21、S22、S23、S25与实施例一的S10、S11、S12、S13、S14步骤一致,此处不再详细阐述。
上述的S24步骤,主要是判断当用户因为有事,需要暂停当前的通话,又不想挂断通话的情况,提高整个判断的完整性。
以上所述仅为本发明的较佳实施例而已,并不用以限制本发明,凡在本发明的精神和原则之内所作的任何修改、等同替换和改进等,均应包含在本发明的保护范围之内。

Claims (10)

  1. 一种自动挂断通话的方法,其特征在于,包括以下步骤:
    服务器获取主呼叫方呼叫被呼叫方的信息,双方进行通话,并通过电信网通知服务器;
    判断当前的通话条件是否满足设定的条件;
    若当前的通话条件满足设定的条件,则自动挂断双方的通话;
    或者,若当前的通话条件不满足设定的条件,则返回所述双方进行通话。
  2. 根据权利要求1所述的一种自动挂断通话的方法,其特征在于,所述判断当前的通话条件是否满足设定的条件的步骤中,所述设定条件包括音频信号的有否以及音量的大小。
  3. 根据权利要求1所述的一种自动挂断通话的方法,其特征在于,所述服务器获取主呼叫方呼叫被呼叫方的信息的步骤,包括以下具体步骤:
    服务器获取主呼叫方呼叫被呼叫方且被呼叫方接通主呼叫方的呼叫的信息;
    双方进行通话;
    服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频。
  4. 根据权利要求2所述的一种自动挂断通话的方法,其特征在于,所述判断当前的通话条件是否满足设定的条件的步骤,包括以下具体步骤:
    判断双方各自的音频信号是否满足无通话时的音频量;
    若各自的音频信号为满足无通话时的音频量,则当前的通话条件满足设定的条件;
    若各自的音频信号不为满足无通话时的音频量,则判断当前的音量是否低于设定的音量值;
    若是,则当前的通话条件满足设定的条件;
    若不是,则当前的通话条件不满足设定的条件。
  5. 根据权利要求1至4任一项所述的一种自动挂断通话的方法,其特征在于,所述服务器获取主呼叫方呼叫被呼叫方的信息的步骤之前,还包括:
    判断是否需要预先设定条件;
    若需要,则设定音量值,并保存于服务器内;
    若不需要,则进入所述主呼叫方呼叫被呼叫方,双方进行通话的步骤。
  6. 根据权利要求5所述的一种自动挂断通话的方法,其特征在于,所述若当前的通话条件满足设定的条件,则自动挂断双方的通话的步骤,具体包括以下具体步骤:
    服务器阻止双方的音频输送;
    服务器发送挂断指令至主呼叫终端以及被呼叫终端。
  7. 一种自动挂断通话的系统,其特征在于,包括通话单元、判断单元以及挂断单元;
    所述通话单元,用于服务器获取主呼叫方呼叫被呼叫方的信息,并通过电信网通知服务器,与所述判断单元连接;
    所述判断单元,用于判断当前的通话条件是否满足设定的条件,分别与所述挂断单元以及所述通话单元连接;
    所述挂断单元,用于自动挂断双方的通话。
  8. 根据权利要求7所述的一种自动挂断通话的系统,其特征在于,所述通话单元包括呼叫模块、通话模块以及发送模块;
    所述呼叫模块,用于服务器获取主呼叫方呼叫被呼叫方且被呼叫方接通主呼叫方的呼叫的信息,与所述通话模块连接;
    所述通话模块,用于双方进行通话,分别与所述判断单元以及发送模块连接;
    所述发送模块,用于服务器接收主呼叫终端以及被呼叫终端分别发送各自的音频,与所述判断单元连接。
  9. 根据权利要求8所述的一种自动挂断通话的系统,其特征在于,所述判断单元包括音频信号判断模块以及音量判断模块;
    所述音频信号判断模块,用于判断双方各自的音频信号是否满足无通话时的音频量;分别与所述音量判断模块、所述发送模块以及所述音量判断模块连接;
    所述音量判断模块,用于判断当前的音量大小是否低于设定的音量值,分别与所述通话模块以及所述挂断单元连接。
  10. 根据权利要求9所述的一种自动挂断通话的系统,其特征在于,所述一种自动挂断通话的系统还包括设定判断单元以及设定单元;
    所述设定判断单元,用于判断是否需要预先设定条件,分别与所述设定单元以及所述呼叫模块连接;
    所述设定单元,用于设定音量值,并保存于服务器内,与所述呼叫模块连接。
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