VoIP+DECT Phone Keypad Function List

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Unify 桌面电话说明书

Unify 桌面电话说明书

© Unify Software and Solutions GmbH & Co. KGMies-van-der-Rohe-Str. 6, 80807 Munich/GermanyAll rights reserved. 12/2015Reference No.: A31003-S2000-U146-5-7619Key Layout and OperationFunction Keys and Audio Keys – Default ConfigurationMode Keys and TouchSliderTouchGuideOpen Context MenuIf the context menu isn’t shown, you can access the contextmenu by pressing the right arrow key on the TouchGuide.The left arrow key escapes the selected option or function.Hold callTransfer callConferencep Turn muteon/offo Turn headset on/offn Turn speaker on/offt Home screenu Phonebooksw Call listsx Voicemailsv Service/Applications menuy Help functionRun finger over TouchSlider:Depending on context, set vo-lumes for ringer or speakerXShining blue: mode is activeShining white:for call lists: new entry in call listfor voicemails: new voicemailPress m key:- Scroll upwardsHold down:- Jump to beginningof listPress l key:- Scroll downwardsHold down:- Jump to end of listPress g key:- Call up the contextmenu- Go down a levelPress h key:- Cancel function- Delete characterleft of cursor- Go up a menu levelPress i key:- Confirm input- Perform actionRun finger over in-ner edge of ring:- Scroll through listsand menus- Adjust volumeUse g toopen contextmenu.Use h to can-cel and goback one level.Key Layout and OperationFunction Keys and Audio Keys – Symbol ConfigurationIcon OverviewDisplay Icons in Idle StateDisplay Icons during a CallIcon ExplanationYou have received one or more new messagesOne or more new entries have been addedto the call listsCall Forwarding is activated for all callsRing tone is deactivatedRemote maintenance is activatedDo not disturb is activatedPhone lock is activatedBluetooth connectivity is activatedA mobile user is logged on to the phoneIcon ExplanationCall is activeCall has been disconnectedYou have placed the call on holdYour call partner has placed the call on holdSecure voice connectionInsecure voice connectionq For future user Activate/deactivatecall forwardings End (disconnect) callp Turn muteon/offo Turn headset on/offn Turn speaker on/off{}OpenStage60/80SIPOpenScape VoiceSingle Line ConfigurationQuick Reference CardUsing your OpenStagePlace a Call•Lift handset, dial number and press i, or•dial number and lift handset, or•for handsfree call: dial number and press i.Answer a Call•Lift handset, or•for handsfree call: press n.End a Call•Hang up,or•to end a handsfree call: press n.Use the headset•Place a call: dial number and press i.•Answer or end a call: press o.Redial a Number1.Select "Redial" in the context menu and press i.2.Lift handset to use handset mode.Hold or Retrieve a Call•During a call press pre-programmed Hold key.•To retrieve a call: press pre-programmed Hold key.Make a Conference Call1.During a call with party A, press pre-programmed Conferencekey.2.Enter the phone number for party B and press i.3.Once connected with party B, press pre-programmed Confe-rence key.You are now connected in a conference with party A and B. Transfer a CallBlind transfer (no consultation):1.During the call with party A, select"Blind transfer" in the con-text menu and press i.2.Enter the phone number of party B and press i.Semi-attended transfer (transfer while ringing):1.During the call with party A, press pre-programmed Transferkey.2.Enter the phone number of party B and press i.3.When the phone starts to ring, select "Complete transfer"and press i.Attended transfer (with consultation):1.During the call with party A, press pre-programmed Transferkey.2.Enter the phone number of party B and press i.3.Announce the call to party B.4.Select "Complete Xfer" in the context menu and press ing your OpenStageProgramm Call Forwarding1.Press pre-programmed Call Forwarding Key.2.Select "Set a forwarding destiantion" and press i.3.Enter the destination number and press i.4.Select "Save&Exit" in the menu and press i.5.Press v to return to phone mode.Turn Call Forwarding on or off for All Calls•Press r for turning Call Forwarding on/off.Dial from the Call List1.Press w.2.The "Missed" tab will appear first, press w repeatedly for"Dialed", "Received", or "Forwarded" tabs.3.Select the desired phone number and press i.Activate Callback while calling1.Enter the destination phone number and press i.2.If there is no answer or the line is busy: select "Callback"from the context menu and press i.Deflect a Call while ringing1.Select "Deflect" from the context menu and press i.2.Enter a destination phone number and press i.Use Mute during a call•Press p for turning mute on or off.Switch to Speakerphone Mode during a Call•US mode:press n and hang up.•Default mode: hold down n until you hang up.Switch to Handset Mode during a Call•Lift handset.Save a Function to a Key1.Press and hold the desired programmable keyuntil a popup appears.2.Press i to confirm entering programming mode.3.Select "Normal" or "Shifted" and press i.4.Select desired function and press i.5.Define an appropriate key label and press i.6.In some cases: enter additional parameters and press i.7.Select "Save&Exit" in the context menu and press i.8.Press v to return to phone mode.List of Programmable Functions* only visible if provided by adminFunction ExplanationUnallocated Clears the keySelected dialing Dials a pre-defined numberRepeat dialing Calls the last dialed numberForward all calls Forwards all incoming callsForward no reply Forwards calls only if they are notansweredForward busy Forwards calls only when the line isbusyRinger off Switches the ringer off/onHold Places a call on holdAlternate Switches between two callsBlind transfer call Transfers a call without consultationTransfer call Transfers a call with consultationDeflect Deflects a call to another destinationShift Switches to the shifted key levelConference Places a conference callHeadset Answers a call using the headsetDo not disturb Switches the ringer off; callers hear thebusy signalGroup pickup Picks up a group callRepertory dial Dials pre-defined numbers and controlsequencesShow phone screen Toggles features hosted by OpenScapeVoiceMobility Login/Logoff for mobile usersDirected pickup Picks up another ringing phoneCallback Requests an automatic call back(busy/no answer)Cancel callbacks Cancels a callback requestConsultation Puts an active call on hold and providesa prompt for dialingDSS*Dials a pre-defined internal numberCall Waiting Notifies of a second incoming call whilein active callImmediate ring Ringing keyset line without delay(Executive/Assistant configuration)Preview Preview line details for shared linesAICS Zip tone*For headset operation only:auto answer and alert toneStart application Launches an application (short cut)Built in fwd Turns Call Forwarding on/off。

voip decoding errors -回复

voip decoding errors -回复

voip decoding errors -回复标题:VoIP解码错误:原因、影响及解决方案引言:VoIP(互联网语音通信)是一种通过互联网协议传输语音通信的技术,已经成为现代通信的主要方式之一。

然而,在VoIP通信过程中,我们可能会遇到解码错误的问题。

本文将探讨VoIP解码错误的原因、影响及解决方案。

一、VoIP解码错误的原因1. 网络问题:网络延迟、抖动或丢包等因素会影响VoIP信号的传输稳定性,导致解码错误的发生。

2. 低带宽问题:VoIP需要足够的带宽才能传输音频数据,如果网络带宽不足,可能会导致解码错误。

3. 压缩算法问题:VoIP中的音频通常使用压缩算法进行编码和解码,但不同的压缩算法会对音频数据产生不同的失真,从而导致解码错误。

4. 硬件和软件问题:网络设备、VoIP电话、VoIP软件等硬件和软件问题可能会导致解码错误。

二、VoIP解码错误的影响1. 音频质量下降:解码错误会导致音频质量下降,出现噪音、断续、失真等问题,影响通话质量。

2. 通话中断:严重的解码错误可能导致通话中断,使得交流无法进行。

3. 信息丢失:解码错误可能导致语音中的部分信息丢失,影响通话效果和交流内容的准确性。

三、解决VoIP解码错误的方法1. 改进网络稳定性:通过提升网络的稳定性和性能来减少VoIP解码错误的发生。

可以采用调整网络设置、增加带宽、优化网络设备等方法来改善网络环境。

2. 使用高质量的网络设备和软件:选择高性能的路由器、交换机等网络设备,以及经过充分测试和验证的VoIP电话和软件,有助于降低解码错误的风险。

3. 配置QoS(服务质量):通过配置网络设备的QoS功能,可以优先保障VoIP通信的带宽和网络资源,从而减少解码错误的发生。

4. 使用适当的编解码器:选择合适的音频编解码器,根据网络条件和通话质量需求进行配置,有助于提升解码效果和减少解码错误。

5. 定期监测和维护:定期监测网络和VoIP 设备的性能,及时发现和解决潜在问题,有助于减少解码错误的频率。

H3C VoIP配置

H3C VoIP配置
第 2 章 语音用户线配置...........................................................................................................2-1 2.1 信号音 ................................................................................................................................ 2-1 2.2 FXS语音用户线 .................................................................................................................. 2-1 2.2.1 FXS接口 .................................................................................................................. 2-1 2.2.2 FXS语音用户线的主叫号码识别及显示功能............................................................ 2-1 2.3 FXO语音用户线.................................................................................................................. 2-2 2.3.1 FXO接口.................................................................................................................. 2-2 2.3.2 FXO语音用户线的接收及发送主叫号码功能 ........................................................... 2-2 2.3.3 FXO用户线忙音检测功能 ........................................................................................ 2-2 2.4 E&M语音用户线 ................................................................................................................. 2-3 2.4.1 E&M接口 ................................................................................................................. 2-3 2.4.2 E&M的启动方式....................................................................................................... 2-4 2.5 语音用户线配置任务简介 ................................................................................................... 2-5 2.6 配置提示音模式.................................................................................................................. 2-5 2.6.1 配置准备 .................................................................................................................. 2-5 2.6.2 配置多国提示音模式................................................................................................ 2-6 2.6.3 配置自定义提示音模式 ............................................................................................ 2-6 2.7 配置语音用户线的基本功能................................................................................................ 2-7 2.7.1 配置准备 .................................................................................................................. 2-7 2.7.2 配置语音用户线的基本功能 ..................................................................................... 2-7 2.8 配置FXS语音用户线........................................................................................................... 2-7 2.8.1 配置准备 .................................................................................................................. 2-7 2.8.2 配置FXS语音用户线的主叫号码识别及显示功能..................................................... 2-8 2.8.3 配置FXS语音用户线的丢包补偿方式....................................................................... 2-9 2.9 配置FXO语音用户线 .......................................................................................................... 2-9 2.9.1 配置准备 .................................................................................................................. 2-9 2.9.2 配置FXO语音用户线的接收及发送主叫号码功能 .................................................... 2-9 2.9.3 配置FXO语音用户线的忙音检测功能 .................................................................... 2-10 2.9.4 配置FXO语音用户线的摘机方式............................................................................ 2-12 2.9.5 配置FXO语音用户线的其他功能............................................................................ 2-13 2.10 配置FXS&FXO 1:1 绑定................................................................................................. 2-13 2.10.1 配置准备 .............................................................................................................. 2-14

VOIP电话菜单设置翻译

VOIP电话菜单设置翻译

一、Menu:菜单↓(一)、status:(状态)↓1、IP:10.32.62.112、MAC:001565386764 物理地址3、Firmware:(固件版本)9.61.0.854、More:(更多)↓①Network:网络↓MAC:001565386764WAN Type:网络类型WAN IP:广域网的IP地址WAN Mask:广域网的子网掩码LAN Type: 局域网类型LAN IP: 局域网IPLAN Mask:局域网的子网掩码Gateway: 网关Pri.DNS: 首选域名服务器Sec.DNS:备选域名服务器②Phone:电话↓Model:模型Hardware: 硬件Firmware: 固件Product ID:产品ID③Accounts:账户↓990026:Registered(注册)Empty:空的(二)、Features:功能↓1、Forward:呼叫转移↓①Always:总是Always(Disable/Enable): 总是(禁用/可以)Forward to: 转发到On code:开启代码Off code:关闭代码②Busy:占线Busy(Disable/Enable): 占线(禁用/可以)③No Answer:不答复No Answer(Disable/Enable):不答复(禁用/可以)④After duratio: 通过时间2、Call waiting:呼叫等待↓①Call waiting(Disable/Enable):呼叫等待(禁用/可以)②Play Tone(Disable/Enable): 播放音(禁用/可以)3、Dss keys:可编程按键↓①Line key 1:关键线路1Type(line/BLF/BLF list/shared line/URL Record/ACD/key Event/Speed Dial/Intercom)类型(线路/监视/监视目录/共享线/地址记录/自动呼叫分配/项目关键/快速拨号/对讲机)Account ID(Line 1/Line 2/Auto):账户ID(1号口/2号口/自动):用户名Server:服务器②Line key 2: 关键线路2Type(line/BLF/BLF list/shared line/URL Record/ACD/key Event/Speed Dial/Intercom)类型(线路/监视/监视目录/共享线/地址记录/自动呼叫分配/项目关键/快速拨号/对讲机)Account ID(Line 1/Line 2/Auto):账户ID(1号口/2号口/自动):用户名Server:服务器4、Key as Send:拨号键↓①key as send(#/*/Disable):发送的关键(#/*/禁用)5、Hot Line:热线↓①Hot Number:热点数②Hot line Delay:热线延迟6、Anonym call:匿名电话↓①Line ID(Line 1/line 2):线路ID(线路1/线路2)②Anonym call(Disable/Enable):匿名电话(禁止/开启)③Anonym on:匿名电话开启④Anonym off:匿名电话关闭⑤Rejection(Disable/Enable):拒绝(禁止/开启)⑥Rejection on:拒绝开启⑦Rejection off:拒绝关闭7、Auto Redial:自动重拨↓①Auto Redial(Disable/Enable):自动重拨(禁止/开启)②Inter val:区间③Times:时间8、Intercom:自动重拨↓①Intercom Allo(Disable/Enable):内部分配(禁止/开启)②Intercom Mute(Disable/Enable):通话静音(禁止/开启)③Intercom Ton(Disable/Enable):通话音(禁止/开启)④Intercom Barg(Disable/Enable):对讲(禁止/开启)9、DND:对讲机↓①DND(Disable/Enable):DND(禁止/开启)②DND on code:DND代码打开③DND off code:DND代码关闭10、Call complet:呼叫完成↓①Call complet(Disable/Enable):呼叫完成(禁止/开启)11、History sett:历史设置↓①History Recor(Disable/Enable):历史记录(禁止/开启)(三)、Settings:设置↓1、Basic:基本↓①Language:语言②Time & Date:时间&日期SNTP(Time Zone/NTP serr/DST):网络时间协议(时区/服务器/夏令时)Manual(Date/Time):手册(日期/时间)Time & Date(clock 24hour/12hour):时间&日期(表24小时/12小时)DHCP Time(Disable/Enable):动态主机配置协议时间(禁止/开启)③Ring Tones:铃声④Phone Volume:手机体积Handset vol:手机音量Speaker vol:扬声器音量Headset vol:耳机音量Ringer vol:铃声音量2、Advanced:高级↓①Password:密码(四)、Messages:消息↓1、View Voice Ma:可视声音码↓①990026(0 new 0 old Mai):可视声音码②Account 2(Unregistered):账户2(未注册)2、Set Voice Mai:设置语音麦↓①Account 1 NO:账户1号②Account 2 NO:账户2号(五)、History:消息(六)、Directory:号码薄1、Local Directo:本地号码↓①Contact lis:联系信息②Black list:黑名单New Item:新项目二、电话按键翻译:1、HOLD:保存2、CONF:会议3、MESSAGE:通知(消息)4、HEADSET:耳机5、TRAN:传输6、RD:删除空的目录。

常用语音VoIP配置命令手册

常用语音VoIP配置命令手册

//Shows all voice port configurations in detail //Shows one voice port configuration in detail //Shows all voice port configurations in brief //Shows all DSP statuses
在 GW 上常用的配置命令: router(config)# voice service voip router(conf-voi-serv)# h323 router(conf-voi-serv)# no shutdown router(config)# interface loopback 0 router(config-if)# ip address 10.10.1.1 255.255.255.0 router(config-if)# h323-gateway voip interface router(config-if)# h323-gateway voip h323-id gw1 router(config-if)# h323-gateway voip bind srcaddr 10.10.1.1 在 GW 上定义编码配置: 定义编码集 router(config)# voice class codec 100 router(config-class)# codec preference 1 g711alaw router(config-class)# codec preference 2 g729br8 router(config)# dial-peer voice 500 voip 多编码协商 router(config-dial-peer)# voice-class codec 100 单信令编码协商 router(config-dial-peer)# codec g711alaw 在 GW 上配置 H323 参数 router(config)# voice class h323 600

VOIP基础ppt课件

VOIP基础ppt课件

➢ 量化(quantizing) 对采样结果赋予一个特定范围内的整数值
➢ 编码(Coding)
8bit
Nyquist定理:如果以最高频率的两倍速率采样,就可以将信号完整地恢 复到模拟形式。
一个话路的PCM信号速率:8000x8=64kbps
➢ PCM-u:北美,日本 ➢ PCM-A:欧洲及其他地区
11
网关通知呼叫代理,请求监视的某些事件已发生 获得某端点或一组端点的详细信息
检索某端点上某连接的信息
MG->CA
告之某端点或一组端点退出服务或投入服务
28
MGCP呼叫流程
User
MG
MGC
Off-hook
Dial tone Dialing
Ringback tone
1 RQNT ACK
2 NTFY ACK
➢ 价格低廉 ➢ 可提供的业务较多 ➢ 网络利用率高 ➢ 可以与Internet应用很好融合 ➢ 符合三网合一的发展方向
缺点:
➢ 服务质量:延迟和抖动较大 ➢ 稳定性:断电保护 ➢ 安全性:容易遭受攻击和窃听
5
VOIP应用方式
VOIP三种典型应用方式:
➢ PC机用户与PC机用户间通过IP网络进电话或传真通信,这是最初的应用; ➢ PC机用户与PSTN或PBX用户通过IP网络进行电话或传真通信; ➢ PSTN或PBX用户通过IP网络进行电话或传真通信;
3 RQNT ACK
4 NTFY ACK
5 CRCX ACK
7 RQNT ACK
9 MDCX ACK
Conversation
6 CRCX ACK
8 10
NTFY ACK MDCX ACK
Busy tone On-hook

IP PHONE功能热键

IP PHONE功能热键
VOIP IP PHONE功能热键 功能键 分机代答 按键 *8 按键 功能解释 同一分组内如果有分机震铃,其他话机按此键可以代替其应答 如*4001表示代4001接起 将当前的呼叫直接转移给另外一个人 将话务转给其它分机 你接起来一个电话,这个时候按此热键系统会将这个用户驻留在等待音 701-799 乐中并且告诉你一个号码,比如701,你这个时候可以挂掉电话,然后 走到任何一部话机按701可以将此电话转接过来 如果你觉得当前谈话十分重要,按此热键系统将在向你播放beep的一声 后开始录下你们的谈话内容; 500 501 *3 按此键,收听自己语音信箱中的留言 读出当前分机的号码 在通话过程中主叫按该热键可以使双方都进入到隐秘的会议室中,如果 需要继续邀请其它人可以使用热键0,邀请完成后按*0返回会议室
指定号码代答*+分机号码 电话盲转 **
话务机转接 *2
呼叫驻留
*7
一键录音
*1Leabharlann 信箱管理 读分机号码 热键会议

阿尔卡特朗讯4019型数字话机用户快速参考手册概要

阿尔卡特朗讯4019型数字话机用户快速参考手册概要

E n g l i s hOtherCustomize your terminalModifying your personal codeLock / unlock your telephonefollow informations displayed on thescreendepending the displayed informations, enter your password or confirm* SettingsOKOptionsORPhone OK Password OKLockOK2653a b c d e f k m noAdjusting the audio functions Adjusting screen brightnessUsing your telephoneIdentify the terminal you are on Making a calladjust ringing (melody, volume, mode,... by following instructions displayed on the screento return to the home page (long press increase or decrease thecontrastapplyto return to the home page (long pressdisplays your telephone numberto return to the home page (long presslift the receiver hands freeprogrammed linekeycalling from your personaldirectorydial directly the number for yourcallTo make an external call, dial the outside line access code before dialling your correspondent's number.SettingsOK Phone OKRinging OKSettingsOK Phone OKContrastOKOROKWhoamIOKOROROROKOR2653ab cde fk m no EnA l c a t e l -L u c e n t I P T o u c h 4018 HandsetAudio keysFunction keysHang-up key: to terminate a call.Hands-free/Loudspeaker Key: to make or answer a call without lif-ting the receiver (Alcatel-Lucent IP Touch 4008/IP Touch 4018 .•lit in hands-free mode or headset mode; (short press.•flashing in loudspeaker mode (long press.Intercom/Mute key:•During a conversation: press this key so that yourcorrespondent can no longer hear you.•Terminal idle: press this key to answer calls automaticallywithout picking up the receiver.To adjust the loudspeaker or handset volume up or downMessaging key to access various mail servicesIf the key flashes, a new voice message or a new text message has been received.'Redial' key : To access the 'Redial' function .NavigationUp-down navigator: used to navigate around the home page, through the menus or in a text zone when entering characters.The home page consists of:•Information on the status of the set (setlocked, call forward, etc.•Date and time•Set programming and configurationfunctionsOK key: used to validate your choices and options while programming or configu-ring.Back/Exit key: to return to previous menu (short press or return to first screen (long press. You can also use this key to correct a character you entered.WhoamILockSettingsOKFunction keys and programmable keysGuide key: Use to obtain information on the pre-programmed keys or to access the set programming or configuration.Phone book key : Phone book key to access your personal phone book (short press or to make a call by name (long press or press twice.Pre-programmed function keys and programmable key Lit when the function associated with the key is activated. The 6-key block consists of:two line keys,one 'immediate forwarding or cancel call-forward' key, one 'conference' key, one 'transfer' key,one direct call key to programme.Alphanumeric keypadTo avoid accidentally damaging the set telephone line connector, make sure you position the cord correctly in the compartment intended for this purpose.A l c a t e l -L u c e n t I P T o u c h 4008 / 4018 P h o n e & 4019 D i g i t a l P h on e w w w .a l c a t e l -l u c e n t .c o mReceiving a callRediallingMake a call-back request to a busy numberVoice message serviceConsulting your voice mailboxIf the key flashes, a new voice message or a new text message has been received.Divert your calls to another numberDivert your calls to another numberlift the receiverhands free (IP Touch 4008/4018press the key for the line thatis lit up(AIP Touch 4008/4018last number redialthe number you are calling isbusypress the key next to 'consult voice ortext message'follow informations displayed on the screenpress the 'call forward' function key dial the destination numberselect the function associated with 'immediate forwarding'apply*dial the destination number press the ok key if programming is not automatically recorded*OROR¤CallbackOKOK2653ab cde fk m no OR Forward OK OKOK2653ab cde fk m noOKDiverting calls to your voice mailboxWhen you return, cancel all diversionsDuring a conversationSending DTMF signalsDuring a conversation you sometimes have to send DTMF signals, such as with a voice server, an automated attendant or a remotely consulted answering machine. The function is automatically cancelled when you hang up.Calling a second person during a conversationDuring a conversation, to call a second person.Transferring a callDuring a conversation, to transfer the call to another number:select the function associated with'immediate forwarding'press the OK key if programming is not automatically recorded*.press the 'call forward' function keyif necessary, confirm cancellation of thediversionselect the function associated with 'emit in voicefrequencies'the first call is on holddial directly the number for your call number tobe called'transfer' programmed key ForwardOKOKOKORForward OK Cancelfwd.OK¤MFcodeORSend DTMF OK2653ab cde fk m no 2653ab cde fk m no ORTransfer OKThree-way conferenceDuring a conversation, a second call is on hold.cancel conference and return to first correspondentDirectoryUsing call by name*You enter your contact's name in predictive text mode. In this mode, enter each letter of the name by pressing only once on the key with this letter.Program the keys in the personal phone bookMake a call using the personal phone bookGuarantee and clausesThis document describes the services offered by the Alcatel-Lucent IP Touch4008/4018 Phone and 4019 Digital Phone connected to an Alcatel-Lucent OmniPCX Office or Alcatel-Lucent OmniPCX Enterprise Communication Server system. For more information, see the user manual for your Alcatel-Lucent IP Touch 4008/4018 Phone(IP set or 4019 Digital Phone (digital set. Contact your installer.Warning: never place your telephone in contact with water. To clean your telephone, you may however use a damp soft cloth. Never use solvents (trichlorethylene, acetone, etc. which may damage the plastic parts of your telephone. Never spray it with cleaning products.The ear piece and microphone area of the handset may attract metallic objects that may be dangerous for the ear.The wording is not contractual and may be subject to change. Some functions of your telephone are controlled by a software key and the configuration of the unit.EC countries: we, Alcatel-Lucent Enterprise , declare that the Alcatel-Lucent IP Touch 4008/4018 Phone and 4019 Digital Phone products comply with the essential demands of Directive 1999/5/CE of the European Parliament and Council. A copy of the original of this declaration of compliance can be obtained from your installer.* Depending on your telephone system, contact your installer or consult the appropriate User Guide.Collection of these products at the end of their product life must be done selectively.Alcatel, Lucent, Alcatel-Lucent and the Alcatel-Lucent logo are trademarks of Alcatel-Lucent. All other trademarks are the property of their respective owners. The informationpresented is subject to change without notice. Alcatel-Lucent assumes no responsibility for inaccuracies contained herein. Copyright © 2007 Alcatel-Lucent. All rights reserved.“conference” programmed key hang up on all corresponden t“conference” programmed keylong pressenter the first letters of the name select the name from thedisplayed listfollowinformations displayed on thescreenpress and release select anentry in the directory (0-9*press and release press the programmed key directly (0 to 9select the contact to callstart the callOR¤Conference OKOREnd conference OKOK*OK2653ab cde fk m no OROK‡3GV19008BSAB010728ÔÎ̉。

手机测试常用词汇的中英文对照

手机测试常用词汇的中英文对照

彩铃polyphonic ringtone和弦铃声chord music ringtone 对讲机Walkie-Talkie全球定位系统GPS (Global Positioning System)高保真high fidelity(常简写为hi-fi)移动梦网Monternet(Mobile+Internet)短信服务SMS(Short Message Service)彩信服务MMS(Multimedia Message service)客户身份识别卡SIM卡(Subscriber Identity Module)全球移动通信系统GSM (Global System For Mobile Communications) 储值卡pre-paid phone card语音提示voice prompt直板手机bar phone翻盖手机clamshell phone /flip phone滑盖手机slide phone翻盖接听flip answer按键keypad按键音keypad tone提示音warning tone手机充值cellular phone replenishing/recharging手机入网费initiation charges for mobile phone; mobile access fee漫游roaming service手机用户mobile phone user/subscriber短信short message; text message图片短信picture message手机费mobile phone fee关机power off手机铃音mobile phone ringtone振动vibrate手机实名制mobile phone identification policy双向收费two-way charging scheme彩屏color screen壁纸wallpaper待机模式standby mode操作菜单options菜单模式list view/ grid view快捷图标short-cut icon自动重拨automatic redial快速拨号speed dial语音拨号voice dial任意键应答any key answer限制呼叫fixed dial呼出通话outgoing call被叫通话incoming call近来的呼叫recent call呼叫转移call divert未接电话missed call已接电话received call不在服务区out of reach手机电路中常用的中英文对照A/D:模数转换。

融合VoIP与DECT的数字无绳电话解决方案

融合VoIP与DECT的数字无绳电话解决方案

融合VoIP与DECT的数字无绳电话解决方案关键字:DECT Skype数字增强型无绳电话标准开放数据接入协议ODAP数字增强型无线通讯标准(DECT)是整合数据应用的第一种无绳电话标准之一。

为了保证市场竞争力,DECT着力于创建一个可互操作的平台,该平台能够同时为住宅及商业市场提供话音及数据应用。

根据这一策略,DECT正为固定网络运营商提供一系列增强其竞争力的基础设施。

同时,用户也会得益于这一广泛的新型服务及应用。

他们的无绳系统不再仅能提供话音服务,同时也将为他们提供数据及商业应用方面的服务。

开放数据接入协议(ODAP)新近为欧洲电信标准协会(ETSI)所批准。

根据ODAP这一新的可互操作应用平台,DECT已为将来的革新奠定了基础。

工业控制、设备管理及存储管理仅为应用功能的一部分,而这些应用就能够帮助企业客户优化其商业绩效。

ETSI为全球100多个国家认可,由其制定的DECT标准是一个提供话音及数据应用的短距离无线通信技术。

它的主要优势在于:1. DECT的使用范围延伸至全球110多个国家;2. 现今市场上可购得的DECT产品超过200种;3. DECT数据产品已经开始运作,并且在市面上可购得;4. DECT满足国际电信联盟(ITU)对第三代移动通信系统的要求,并且DECT被认为是IMT2000无线通信接口家族成员之一。

DECT标准的特征DECT支持语音及数据通信,并且还有其它广泛的应用,例如,电话无线本地环路(WLL)应用、专用分组交换机(PBX)应用等等。

DECT投入应用前期主要用于住宅区内无绳话音应用。

目前,DECT已开始成功应用于数据通信中。

并且,DECT已被认为是IMT2000五个空中接口之一。

总之,DECT是一个发展中的、低成本、拥有大量市场的无线话音/数据通信技术。

此外,DECT 是唯一一个可同时应用于局域网(LAN)和公网的用于传输数据的无线接入技术。

在当前速度有限的情况下,在局域网中采用DECT是小型办公和家居办公(SOHO)应用的一个附加功能。

mtk平台函数

mtk平台函数

DIARYMTK 2010-06-07 14:07:18 阅读90 评论0 字号:大中小May.19LCD移植static const s_lcd_probe gLcdProbe[] = {...{"ILI9328", LCD_IsILI9328, &LCD_func_ILI9328},{"LP4948", NULL, &LCD_func_LP4948},}原来{"LP4948", NULL, &LCD_func_LP4948},放在{"ILI9328", LCD_IsILI9328, &LCD_func_ILI9328},前面导致开机白屏,原因是void LCD_FunConfigNew(void){#if 1kal_uint32 i;for (i = 0; i < LCD_PROBE_NUM; i ++) {if ((gLcdProbe[i].lcd_probe == NULL) || (KAL_TRUE == (gLcdProbe[i].lcd_probe)())) {break;}}gLcdSeq = i;MainLCD = gLcdProbe[gLcdSeq].lcd_func;#endif// MainLCD = &LCD_func_ILI9225;}当走到"LP4948"项时,由于其对应的lcd_probe为"NULL",导致退出循环,使得LCD相应的功能函数都指向"LP4948"对应的函数。

双卡改单卡。

SINGLE_SIM_MMI_ONLY某个菜单项STR_ID不显示。

原来是其子菜单个数与实际个数不相符。

键盘定义Custom/drv/Drv_tool/DrvGen.exeCustom/drv/misc_drv/MT6225_08A_GEMMI_BB/Codegen/codegen.dws分布式编译某个模块出问题了。

VI2010 VoIP Phone 说明书

VI2010 VoIP Phone 说明书

VI2010VoIP Phone User Manual1 Introduction (3)1.1 Hardware Overview (3)1.2 Software Overview (3)2 Keypad interface for IP Phone demo system (4)2.1 Keypad description (4)3 Setup the VoIP Phone by Web Browser (5)3.1 Login (5)3.2 System Information for the VOIP PHONE. (5)3.3 Network (6)3.4 SIP Settings (9)3.5 Update (10)3.6 Reboot (10)4 Appendix: (11)4.1 How to use the FXO port (11)4.2 Keypad Function and setting List (13)1 IntroductionThis user’s manual is for VoIP Phone. This user’s manual will explain the keypad instruction, web configuration and command line configuration for the VoIP Phone. Before using the VoIP Phone, some setup processes are required to make the VoIP Phone work properly. Please refer to the Setup Menu for further information. 1.1 Hardware OverviewThe VoIP Phone has the following interfaces for Networking, telephone interface, LED indication, and power connector.1.1.1 Two RJ-45 Networking interface, these two interfaces support 10/100Mps Fast Ethernet. you can connect one RJ-45 Fast Ethernet port to the ADSL or Switch, and connect the other one to your computer.1.1.2 LED Indication: There are some LED indicators in the VoIP Phone to show the functions, like speaker phone, .Rggister, …. 1.2 Software OverviewNetwork ProtocolTone• Ring Tone • Ring Back Tone • Dial Tone • Busy Tone • User Programming Tone• SIP v1 (RFC2543), v2(RFC3261) • IP/TCP/UDP/RTP/RTCP •IP/ICMP/ARP/RARP/SNTP • TFTP Client/DHCP Client/ PPPoE Client • Telnet/HTTP Server • DNS ClientPhone FunctionCodec• Volume Adjustment • Speed dial, Phone book • Flash • Speaker Phone IP Assignment• G.711: 64k bit/s (PCM) • G.723.1: 6.3k / 5.3k bit/s • G.726: 16k/ 24k / 32k / 40k bit/s (ADPCM) • G.729A: 8k bit/s (CS-ACELP) • G.729B: adds VAD & CNG to G.729 Voice Quality• Static IP • DHCP • PPPoE Security• HTTP 1.1 basic/digest authentication for Web setup • MD5 for SIP authentication (RFC2069/ RFC 2617) • VAD: Voice activity detection • CNG: Comfortable noisegenerator • LEC: Line echo canceller • Packet Loss Compensation • Adaptive Jitter BufferQoSCall Function• ToS field NAT Traversal • STUN • Call Hold • Call Waiting • Call Forward • Caller ID • 3-way conferenceConfigurationDTMF Function• Web Browser • Console/Telnet • Keypad • In-Band DTMF • Out-of Band DTMF • SIP Info SIP ServerFirmware Upgrade• Registrar Server (three SIP account) • Outbound Proxy • TFTP • Console • HTTP • FTP2 Keypad interface for IP Phone demo system2.1 Keypad descriptionKey Name Description1 ”, “!”, “?” ٫“1”, “-“, “2 “2”, “a”, “b”, “c”, “A”, “B”, “C”3 “3”, “d”, “e”, ”f”, “D”, “E”, “F”4 “4”, “g”, “h”, “I”, “G”, “H”, “I”5 “5”, “j”, “k”, “l”, “J”, “K”, “L”6 “6”, “m”, “n”, “o”, “M”, “N”, “O”7 “7”, “p”, “q”, “r”, “s”, “P”, “Q”, “R”, ‘S”8 “8”, “t”, “u”, “v”, “T”, “U”, “V”9 “9”, “w”, “x”, “y”, “z”, “W”, “X”, “Y”, “Z”0 “0”, “space”* “*”, “•”, “:”, “@”# Start dialing processFlash This is “Transfer” to the other phone numberREDIAL This is “REDIAL” the same number againHOLD This is “HOLD” functionMute This is “Mute” functionDND This is “Reject” functionOK This is “OK”, accept settingBackspace This is “Delete”, Delete word or phone numberUP/DOWN This is Up↑ and Down↓ keyMENU This is the “Menu” key to set the IP PhoneSPK This the Speaker PhoneLine1~Line2 When Call hold ,can switch the sessionM1~M9 This is the M1 to M9, this is 9 speed dial number.Conf This is three way conference functionCID This is Incoming call and going call listVolume -/+ This is volume settingPhone Book This is Phone Book list3 Setup the VoIP Phone by Web BrowserDefault the IP Phone’s NAT is enabled, WAN port is in DHCP Client Mode, LAN port is in DHCP Server Mode. You can connect you PC on LAN port, and set the IP type of your PC to DHCP ,then you will get an IP Address from the IP Phone.The IP Phone provides a built-in web server. You can use Web browser to configure the IP Phone. First please input the IP address http://192.168.123.1or http://192.168.123.1:9999 in the IE.3.1 Login.3.1.1 Please input the username and password into the blank field. The default setting is:1 For Administrator, the username is: root; and the password is: test. If you use the account login, you can configure all the setting.2 For normal user, the username is: system or user; and the password is: test. If you use the account login, but you can not configure the SIP setting.3.1.2 Click the “Login” button will move into the VOIP PHONE web based management information page.3.1.3 If you change the setting in the Web Management interface, please do remember to click the “Submit” button in that page. After you finished the change of the setting, click the “Save” function in the left side, and click the Save Button. When you finished the setting, please click the Reboot function in the left side, and click the Reboot button in that page. After the system restart, all the setting can work properly.3.2 System Information for the VOIP PHONE.3.2.1 When you login the web page, you can see the VOIP PHONE current system information like firmware version, company… etc in this page.3.2.2 Also you can see the function lists in the left side. You can use mouse to click the function you want to set up.3.3 Network3.3.1 In Network you can check the Network status, configure the WAN Settings, LAN Settings, DDNS settings and VLAN Settings.3.3.2 Network Status: You can check the current Network setting in this page.3.3.3 WAN Settings: In this page you can configure the IP Phone WAN port’s setting. The WAN port is for you to connect to the ADSL Router, Broadband Router. Also you can use PPPoE to get the WAN IP address from your ISP.3.3.3.1 The IP Phone’s default setting is NAT mode. If you don’t need to use the NAT Mode, you can chang toBridge Mode. If you change the setting to Bridge Mode, then the LAN setting will not effect and will be the same as WAN port.3.3.3.2 The WAN port default is DHCP Client mode, You can change the setting to Fixed IP Mode, or PPPoEMode.3.3.3.3 If you change the WAN port’s setting to Fix IP Mode, then you have to make sure the IP address. NetMask, Gateway, and DNS setting is suitable in your current network environment.3.3.3.4 If you change the WAN port’s setting to PPPoE Mode, you have to input a correct username/passwordto get the IP address from your Internet Service Provider.3.3.3.5 When you finished the setting, please click the Submit button.3.3.3.6 If there is nothing need to change, please click the Save Change Item in the left side, then click theSave button. The change you made will save into the system and the system will Reboot automatically.3.3.4 LAN Settings: In this page you can configure the IP Phone LAN port’s setting.3.3.4.1 The LAN port’s default IP address is 192.168.123.1, Net Mask is 255.255.255.0., and DHCP Serverenabled. The start IP address if 150, end IP adress is 200. It is not necessary to change the LAN settings.3.3.4.2 You can connect your PC to the LAN port, set your PC as DHCP Client mode, then you can get IPaddreess from the TA.3.3.4.3 When you finished the setting, please click the Submit button.3.3.4.4 If there is nothing need to change, please click the Save Change Item in the left side, then click theSave button. The change you made will save into the system and the system will Reboot automatically.3.4 SIP Settings3.4.1 In SIP Settings you can setup the Service Domain, Port Settngs, Codec Settings, Codec ID Settings, RTP Setting, RPort Setting and Other Settings. If the VoIP service is provided by ISP, you need to setup the related informations correctly then you can register to the SIP Proxy Server correctly.3.4.2 In Service Domain Function you need to input the account and the related informations in this page, please refer to your ISP provider. You can register three SIP account in the VoIP Phone. You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from these three SIP accounts.3.4.2.1 First you need click Active to enable the Service Domain, then you can input the following items:3.4.2.1.1 Display Name: you can input the name you want to display.3.4.2.1.2 User Name: you need to input the User Name get from your ISP.3.4.2.1.3 Register Name: you need to input the Register Name get from your ISP.3.4.2.1.4 Register Password: you need to input the Register Password get from your ISP.3.4.2.1.5 Domain Server: you need to input the Domain Server get from your ISP.3.4.2.1.6 Proxy Server: you need to input the Proxy Server get from your ISP.3.4.2.1.7 Outbound Proxy: you need to input the Outbound Proxy get from your ISP. If your ISP does notprovide the information, then you can skip this item.3.4.2.1.8 You can see the Register Status in the Status item. If the item shows “Registered”, then yourVoIP Phone is registered to the ISP, you can make a phone call direcly.3.4.2.1.9 If you have more than one SIP account, you can following the steps to register to the otherISP.3.4.2.1.10 When you finished the setting, please click the Submit button.3.5 Update3.5.1 In Update you can update the VoIP Phone’s firmware to the new one or do the factory reset to let the VoIP Phone back to default setting.3.5.2 In New Firmware function you can update new firmware via HTTP in this page. You can ugrade the firmware by the following steps:3.5.2.1 Select the firmware code type, Risc or DSP code.3.5.2.2 Click the “Browse” button in the right side of the File Location or you can type the correct path and thefilename in File Location blank.3.5.2.3 Select the correct file you want to download to the VoIP Phone then click the Update button.3.5.3 In Default Setting you can restore the VoIP Phone to factory default in this page. You can just click the Restore button, then the VoIP Phone will restore to default and automatically restart again.3.6 Reboot3.6.1 Reboot function you can restart the VoIP Phone. If you want to restart the VoIP Phone, you can just click the Reboot button, then the VoIP Phone will automatically.4 Appendix:4.1 How to use the FXO port (Just for VI2010)First, you can dial 0* switch IP to PSTN.You can use the call forward setting forward the call from IP to PSTN or from PSTN to IP.If you want forward the IP call to a PSTN number, you can first select forward the call to PSTN,Then assign the forward number. After above setting if the gateway receiving a call it can forward to the forward number.4.1.1 Call Forward function: you can setup the phone number you want to forward in this page. There are three type of Forward mode. You can choose All Forward, Busy Forward, and No Answer Forward by click the icon.4.1.1.1 All Forward: All incoming call will forward to the number you choosed. You can input the name and thephone number in URL/Number field. If you select this function, then all the incoming call will direct forward to the speed dial number you choose.4.1.1.2 Busy Forward: If you are on the phone, the new incoming call will forward to the number you choosed.You can input the name and the phone number in URL field.4.1.1.3 No Answer Forward: : If you can not answer the phone, the incoming call will forward to the numberyou choosed. You can input the name and the phone number in URL field. Also you have to set the Time Out time for system to start to forward the call to the number you choosed.4.1.1.4 When you finished the setting, please click the Submit button.You can set the Auto Answer function to answer the incoming call by the phone. If the call is come from the IP, then the TA can let user to redial the call to PSTN phone number. If the call is coming from PSTN, then the TA can let user to redial to IP Phone number. Auto Answer Counter is to set after the ring count meet the number you set then the auto answer will enable. For security issue, You’d better to set the PIN Code. If you have set the PIN code, you will hear a tone to inform you input the PIN Code then you can dial out.4.2 Keypad Function and setting List4.2.1 Phone Book4.2.1.1 Search:Search Phone Book.4.2.1.2 Add entry:Add new phone number to phone book.4.2.1.3 Speed dial:Add speed dial phone number to speed dial list.4.2.1.4 Erase all:Erase all phone number from Phone Book.4.2.2 Call history4.2.2.1 Incoming calls: Show all incoming call.4.2.2.2 Dialed numbers: Show all dialed call.4.2.2.3 Erase record: Delete call history.1 All: Delete all call history.2 Incoming: Delete all incoming call.3 Dialed: Delete all dialed out call.4.2.3 Phone setting4.2.3.1 Call forward4.2.3.1.1 All Forward.1 Activation: To Enabled/Disabled this function.2 Number: Forward to a Speed Dial Number.4.2.3.1.2 Busy Forward.1 Activation: To Enabled/Disabled this function.2 Number: Forward to a Speed Dial Number.4.2.3.1.3 No Answer Forward.1 Activation: To Enabled/Disabled this function.2 Number: Forward to a Speed Dial Number.4.2.3.1.4 Ring Timeout: Set the Ring times to start the no answer forward function, ex: 2 means after 2rings then forward to the dedicated number.4.2.3.2 Block Setting1 All: Block all incoming call.Time2 By3 Duration: Set the start time and end time to Block Setting.4.2.3.3 Date/Time setting: Date and Time Setting.4.2.3.3.1 Date & Time: Set the IP Phone Date and Time.4.2.3.3.2 SNTP setting4.2.3.3.2.1 SNTP : Enabled / Disable SNTP.4.2.3.3.2.2 Primary SNTP: Set Primary SNTP server IP address.4.2.3.3.2.3 Secondary SNTP: Set Secondary SNTP server IP address.4.2.3.3.2.4 Time zone: Set Time zone.4.2.3.3.2.5 Adjustment Time: Set adjustment time period.4.2.3.4 Volume and Gain4.2.3.4.1 Handset volume: Set Handset volume from 0~15 (max.) for you to hear.4.2.3.4.2 Speaker volume: Set Spearer phone volume from 0~15 (max.) for you to hear.4.2.3.4.3 Handset Gain: Set Handset Gain from 0~15 (max.) for the other site to haer.4.2.3.4.4 Speaker Gain: Set Spearer phone Gain from 0~15 (max.) for the other site to haer.4.2.3.5 Ringer4.2.3.5.1 Ringer volume: Ringer volume setting from 0~15 (max.).4.2.3.5.2 Ringer type: Ringer tone selection from 1~4.4.2.3.6 Auto Dial: Set Auto Dial time from 3~9 seconds.4.2.3.7 Auto Answer: This function will active on IP Phone with FXO interface. Set Auto Answr for user canre-dial a call from IP call to PSTN call or from PSTN call to IP call.4.2.3.8 Answer Counter: This function will active on IP Phone with FXO interface. Set Auto Answer will activeafter the numbers of ring.4.2.4 Network4.2.4.1 WAN Setup4.2.4.1.1 IP Type1 Fixed IP client2 DHCPclientclient3 PPPoE4.2.4.1.2 Fixed IP setting4.2.4.1.2.1 IP Address4.2.4.1.2.2 Subnet mask4.2.4.1.2.3 Default Gateway4.2.4.1.2.4 MAC address4.2.4.1.3 PPPoE setting4.2.4.1.3.1 User name4.2.4.1.3.2 Password4.2.4.2 LAN Setup4.2.4.2.1 Bridge4.2.4.2.2 NAT4.2.4.3 DNS4.2.4.3.1 Primary DNS4.2.4.3.2 Secondary DNS4.2.4.4 VLAN4.2.4.4.1 Activation4.2.4.4.2 VID: VID 2~40944.2.4.4.3 Priority: 0~74.2.4.4.4 CFI: 0~14.2.4.5 Status: Show WAN, LAN IP address and MAC address4.2.5 SIP Settings If you want to use Kaypad to set the SIP setting, you have to go to item 7 (Administrator)System Authent to input the password, or you can not change the SIP setting.4.2.5.1 Service domain4.2.5.1.1 First realm4.2.5.1.1.1 Activation4.2.5.1.1.2 User name4.2.5.1.1.3 Display name4.2.5.1.1.4 Register name4.2.5.1.1.5 Register password4.2.5.1.1.6 Proxy server4.2.5.1.1.7 Domain server4.2.5.1.1.8 Outbound proxy4.2.5.1.2 Second realm4.2.5.1.2.1 Activation4.2.5.1.2.2 User name4.2.5.1.2.3 Display name4.2.5.1.2.4 Register name4.2.5.1.2.5 Register password4.2.5.1.2.6 Proxy server Proxy4.2.5.1.2.7 Domain server4.2.5.1.2.8 Outbound proxy4.2.5.1.3 Third realm4.2.5.1.3.1 Activation4.2.5.1.3.2 User name4.2.5.1.3.3 Display name4.2.5.1.3.4 Register name4.2.5.1.3.5 Register password4.2.5.1.3.6 Proxy server Proxy4.2.5.1.3.7 Domain server: Domain4.2.5.1.3.8 Outbound proxy: Outbound Proxy4.2.5.2 Codec4.2.5.2.1 Codec typeuLaw1 G.711aLaw2 G.7113 G.7234 G.7295 G.726-166 G.726-247 G.726-328 G.726-404.2.5.2.2 VAD: Voice Active Detection Enable/Disable.4.2.5.3 RTP setting4.2.5.3.1 Outband DTMF: Outband DTMF4.2.5.3.2 Duplicate RTPduplicate1 Noduplicate2 Oneduplicate3 Two4.2.5.4 RPort Setting: RPort Enabled/Disabled4.2.5.5 Hold by RFC4.2.5.6 Status: Show the SIP Proxy register status. You can use UP/Down key to check each Realm’s status.Realm1 FirstRealm2 SecondRealm3 Third4.2.6 NAT Transversal4.2.6.1 STUN setting4.2.6.1.1 STUN: STUN4.2.6.1.2 STUN server4.2.7 Administrator4.2.7.1 Auto Config4.2.7.1.1 Config Mode: You can select Disable/TFTP/FTP to do the auto config function. This functionmust work with the Auto Config Server.4.2.7.1.2 TFTP server: Setting the TFTP server IP address.4.2.7.1.3 FTP server: Setting the FTP server IP address.4.2.7.1.4 FTP Login Name: Setting the login name to the FTP server.4.2.7.1.5 FTP Password: Setting the Password to the FTP server.4.2.7.2 Default setting: You can restore to the default setting..4.2.7.3 System Authentication: To do the SIP setting from Keypad, need to input the password first. Default is“test”.4.2.7.4 Version: This will show the system’s firmware version.4.2.7.5 Watch Dog: You can use this to enable Watch Dog function to do the debugging.4.2.7.6 Restart: You can use this function to restart your IP Phone.。

VoIP模拟手机系统的常用功能网络电话设计

VoIP模拟手机系统的常用功能网络电话设计

---------------------------------------------------------------范文最新推荐------------------------------------------------------ VoIP模拟手机系统的常用功能网络电话设计摘要网络电话是一种通过互联网或其他使用IP技术的网络,来实现新型的电话通讯。

其低通话成本、低建设成本、易扩充性及日渐优良化的通话质量等主要特点,被目前国际电信企业看成是对传统典型业务产生的有力的竞争点。

本课题论文类似网络电话,是基于网络通信技术与网络编程技术在计算机中用来模拟手机常用功能。

关于网络编程的模型,主要考虑到的是当下流行的C/S模型,为了模拟手机的最重要的通话功能,我们利用VoIP(Voice over Internet Protocol)技术,而其他的功能利用Delphi+SQL Server来实现整个软件功能。

10260关键词网络电话网络编程VoIP数据库毕业设计说明书(论文)外文摘要1 / 15TitleSimulating Common Functionally of Mobile System AbstractWeb phone, using the Internet or other networks that depends on IP technology, is a new type of telephone that can achieve telephone communications. Main features of the web phone, such as low call cost, low construction cost, easy expandability and increasingly excellent call quality, have a more powerful competitiveness to the traditional telecom business. Similar to the web phone, this task is to imitate common functions of mobile based on the technology of network communication and network programming. Concerning the model of network programming, we take the currently popular C/S model into account and use VoIP (Voice over Internet Protocol) technology in order to realize the most important phone call function. Moreover, we take the advantage of Delphi + SQL Server to realize other functions of the entire features of the software.---------------------------------------------------------------范文最新推荐------------------------------------------------------ 2相关技术简介2.1VoIP技术VoIP(Voice over Internet Protocol)是一种以IP电话为主,并推出相应的增值业务的技术。

Philips DECT Cordless Telephone 系列用户指南说明书

Philips DECT Cordless Telephone 系列用户指南说明书

Advertencia
• La red eléctrica está clasificada como peligrosa. La única manera de apagar el cargador es desenchufar la fuente de alimentación de la toma eléctrica, que debe ser siempre de fácil acceso.
durante 5 segundos. 4 Introduzca el PIN/código del sistema (0000). 5 Pulse para confirmar el PIN/código.
»»El registro se completará en menos de 2 minutos.
Permanece encendido al comprobar las llamadas entrantes respondidas en el registro de llamadas.
Indica que se ha realizado una llamada en la lista de rellamadas.
Llamadas desde la agenda
1 Pulse . 2 Seleccione un registro y pulse .
Memoria de acceso directo
Dispone de 2 memorias de acceso directo (teclas 1 y 2). Para marcar automáticamente el número guardado, mantenga pulsados los botones durante el modo de espera. Según el país, las teclas 1 y 2 están predeterminadas como [1_correo voz] (número de correo de voz) y [2_serv info] (número de servicio de información) del operador del servicio de red respectivamente (según la red).

VOIP基础课程专有名词补充解说

VOIP基础课程专有名词补充解说

VOIP基礎課程專有名詞解說:1.DTMF(Dual Tone Multi-Frequency)撥號方式有分為脈衝式(轉盤)與音頻式(按鍵),一般多使用音頻式按鍵機,它會產生出雙音複頻的聲音,則叫做DTMF按鍵音,共有12個鍵,每一個按鍵音是由一個低頻及一個高頻所組合的複頻聲音;每撥打一個鍵則複頻的聲音也會由話機將此號碼的DTMF訊號傳遞給交換機,交換機再透過高/低頻濾波器,將此訊號轉換成對應的電話號碼.@按鍵數字DTMF雙音複頻訊號頻率表:2.回音消除(Echo Cancellers):回音是由於傳輸時在電話回路上產生的信號反射而引起的。

回音消除功能解決語音不清、雙聲、通話不全及背景雜訊變化等問題,在IP網路上傳輸語音時因為回音往返一次的延時要遠遠大於10ms,這時人耳可以聽到明顯的回聲,因而採取回音消除技術是非常必要。

ITU正在制訂的G.165標準正是規定了回音消除技術所需要的性能指標。

回音消除器把從網路上接收到的語音資料和被發送的語音資料進行比較,通過在傳輸線路上設置數位篩檢程式進行回音消除。

3.靜音抑制技術:通常人進行會話是半雙工的,一方講話一方聽,其中的一些靜音階段在它們被作為語音包通過網路傳輸前需要被抑制。

靜音抑制可以採用數位語音插空技術(DSI)實現,抑制靜音可以節省大量的網路帶寬用於進行其他的語音和資料傳輸。

4.RTP(Real-Timer Transport Protocol)即時傳送協定進行語音傳輸會使用RTP做以下動作:(1).偵測傳輸中的封包是否有遺失.(2).提供封包傳輸的資訊.(3).接收端知道封包傳送延遲程度,並做適當的遞補.5.RTCP(Real-Timer Control Protocol)即時控制協定(1).目前傳輸的QoS(Quality of Service)情形.(2).偵測傳輸速度與頻寬是否要改變.PS.凡是與Video&Voice&Data的傳送都是透過RTP和RTCP進行的.6.QoS(Quality of Service)網路服務質量Qos為網路服務品質,它是指網路提供更高優先服務的一種能力,包括專用帶寬、抖動控制和延遲(用於即時和互動式流量情形)、丟包率的改進以及不同WAN、LAN 和MAN 技術下的指定網路流量等,同時確保為每種流量提供的優先權不會阻礙其他流量的進程。

Philips DECT5112S 无线电话说明书

Philips DECT5112S 无线电话说明书

Cordless telephoneDECT5112SThe smart choiceReliable quality coupled with practical features: this classic design with its modern touchmakes the DECT511 really special.Light up your calls•Display for numbers, letters, icons and drawings•LCD screen with backlighting for convenience in the darkMaximize your messaging experience•Text messaging (SMS) for short messages over a fixed line•Eatoni predictive text entry simplifies writing SMS messages•Create several SMS mail boxes that can be PIN-code protectedBenefit from easy-to-use technology•Clock alarm timer•Lock your keypad to avoid hitting the wrong keys•Pilot key & carousel menu navigates in a menu-in-cycle form•Screen incoming call and mute the ringer for more controlMulti-pack for ultimate convenience•Conference call shares a conversation on several handsets•Free intercom - Free internal calls between handsetsHighlightsFull graphic displayThis type of displays makes it easier to use your phone, thanks to the good quality and legibility of the characters. It also gives you more freedom in communication from your home. Now you can combine text with images or numbers, when sending an SMS for example. Backlight on screenBacklighting is a feature that illuminates the LCD screen or keypad for use in the dark. The two most common types of backlighting include LED and electroluminescent (EL). Remotes backlit with LEDs are typically bright but uneven with a yellow or green color, while EL panels are smooth with blue, white or green shades.Text messaging (SMS)The Short Message Service (SMS) is a mechanism for delivery of short messages which can have up to 160 characters of text. These 160 characters are plain text in nature and can be comprised of words, numbers or an alphanumeric combination. TheEatoni®LetterWise predictive text editor helps you to write text messages more quickly. With its unique SMS system you can receive up to 200 messages in 10 different mailboxes. Greeting to a loved one, a quick reminder about an important meeting, …all this can now be put in a SMS and sent over the fixed network.Eatoni text editorEatoni's LetterWise is an easy-to-use text-entry solution requiring just 1.16 taps per correct letter. It makes typing on phone keypads as comfortable, fast, and accurate asusing a typewriter keyboard. It is unique amongpredictive text entry solutions in its ability toeasily handle the entry of abbreviations, names,addresses, and URLs.Personal mail boxesChildren at a certain age need some privacy athome when sending SMS to their friends. Butpaying for several mobile phones quickly getsvery expensive. DECT phones provide up to 9personal SMS mail boxes which can beprotected by a PIN code - SMS can stay privatedespite collective phone usage. The user musttell his correspondents his mail box number sothat they can send short messages directly tothis personal mail box. Sending a SMS from apersonal mailbox makes it easy for theaddressee to reply directly to this mailboxwithout having to remember the mailboxnumber.Clock alarm timerClock alarm timerKeypad lockHow convenient it is to take your DECTphone with you whilst gardening. But in yourpocket it can be so jolted that a key might beaccidently pressed. To prevent inadvertentlycalling your cousin in the US, the keypad lockis a safe way to avoid bad surprises on yournext phone bill!Pilot key & carousel menuThe pilot key navigation system allows easynavigation and settings. The pilot key is a rolleron the left side of the phone that can bescrolled up and down and pressed in themiddle. It commands the Carousel menu whichis a circular loop of menu icons displayed onthe screen.Do-not-disturb functionTired of your children’s friends calling justwhen you’re about to serve dinner? The “Do-not-Disturb” function lets you screenunwanted calls and mute the ringer when youdo not wish to be disturbed, but keeps the lineopen to the family.Conference call functionThanks to the conference call, there is no needto travel to meet people. Especially useful in aprofessional environment where several workstations are connected to the same basestation, the conference call allows, forexample, two colleagues from Paris tocommunicate with a colleague in China. Aconference call allows two handsets to sharethe same external phone conversation in sucha manner that each handset is able tocommunicate with the two others.Free intercomNo need to stop what you are doing and go togive the handset to your son when his friendcalls: you can freely and easily call the handsetin his bedroom and transfer the call.Baby-sit functionWorried about your baby not sleeping? Noneed to constantly go and check in hisbedroom. The baby-sit mode allows you tokeep an ear on him wherever you are in thehouse.Issue date 2018-01-18 Version: 2.0.312 NC: 9961 400 03405 EAN: 87 10895 85233 3© 2018 Koninklijke Philips N.V.All Rights reserved.Specifications are subject to change without notice. Trademarks are the property of Koninklijke Philips N.V. or their respective owners.SpecificationsPicture/Display•Backlight•Backlight color: Orange•Diagonal screen size - tele: 4.0 cm•Effective viewing area: 31.5x21•Lines of text:4Sound•Handset ringers: Polyphonic•Volume Control: digitalConvenience•Auto out-of-range warning•Base Station keys: Paging key•Battery charging indication•Battery full indication•Battery low indication•Call Management: Call Counters, Call Forwarding, Call on Hold, Call Time, Call Waiting*, Caller ID*, Conference Call, Explicit Call Transfer, Microphone mute, Missed Calls, Received Calls •Caroussel Icons: Phonebook, SMS, Intercom, Network, Extra, Base, Handset, Call log•Drop proof: 120 cm•Ease of Use: Hands free mode, Hot Keys, Keypad Lock•Function: Do not disturb ringer mute, Call cost, Alarm clock, Babysit function, Microphone mute, Conference call, Chain dialling, Direct to Do Not Disturb mode, Short cut keys, Keypad lock, Free handset to handset call•Handset Keys: Line, Pilot key, Cancel / Recall, Dialling keypad, Loudspeaker, Call log, Phonebook •Lines of text:4•Multi handset capability: Up to 6 handsets •Text input: Eatoni•VIP group with own melody: Yes, 3 groups •Volume controlAccessories•AC/DC Adaptor•Batteries: 4 x AAA•Cables: RJ 11 cable•Charger: yes, 1 charger•User ManualRelated Products•Package contents: Handset, Base, 2 AAA NiMh Batteries, Telecom Line Cord, Power adapter, User Manual, Warranty leaflet, StickerGreen Specifications•Chemical composition: Nickel-Metal Hydride •Eco Designed•Halogen-free housing•Packaging material: Carton•Packaging type: Giftbox•PVC-free product and cables•User manual: Recycled paperDimensions•Antenna: Integrated•Base dimensions: 118x130x60•Base weight: 180g•Form Factor: Semi lying on base station •Handset color: silver shadow•Handset dimensions: 120mm•Handset weight: 100g•Packaging dimensions (W x H x D):262 x 90 x 217 mm•Relative humidity (operation): up to 95% at 40°C •Relative humidity (storage): up to 95% at 40°C •Temperature range (operation): 0°C to 40°C •Temperature range (storage): -25°C to 70°C Power•Ambient temperature: 5 °C to 40 °C•Battery Type: NiMH•Charging time: 14 hours hr•Kind of Battery: Rechargeable•Number of batteries: 2•Power supply base station: 230V•Radio RF power: <250mW•Standby time: 200 hours•Talk time: up to 15 hoursNetwork Features•Antenna: Fast Selection Technology, Integrated on base, Integrated on handset•Compatible: GAP, PABX•Dialing: Tone, Pulse•Messaging: Quick SMSOperator Requirements•SMS•Name and Caller ID•Caller ID on call waiting•Operators services phonebookMemory Capacity•Name & number shared phonebook: 65•Melody download capacity: -•SMS storage capacity: 25•Call log entries: 40。

VoIP+DECT+MID多媒体网络话机项目

VoIP+DECT+MID多媒体网络话机项目

VolP+DECT+MI[多媒体网络话机项目、产品外观产品外观参见图片:1、2、3、4图1:产品整体效果图片图2:产品分体效果图片图3可移动MID图片-舌牙前撇-TKW WIFtt txwist -WtlM图4: DEC■彩屏子机图片tim产品需求描述一、产品功能1.An droid多媒体网络电话机An droid多媒体网络电话,采用传统的固定电话方式,将传统固定电话的拨号按钮和液晶显示屏换成带触控功能的An droid 系统和LCD显示屏。

用户可通过触摸屏操作电话。

2.广告系统广告系统包含广告管理平台和An droid多媒体网络电话机两部分。

An droid多媒体网络电话机具有联网和广告播放功能,在an droid系统界面右侧有一小块区域用于广告显示。

广告管理平台用于完成广告内容录入、下发、统计信息回收、报表导出等功能。

二. 技术内容电话功能电话功能采用An droid系统自带拨号软件,支持拨号、常用联系人、通话记录等。

通讯录也采用An droid系统自带联系人软件,支持联系人头像设置,联系人固定电话、手机等设置,邮箱,备注,分组,名片模式等;VOIP功能可完成网络电话,采用现在已有第三方软件集成。

电话录音、留言等功能采用第三方软件集成。

桌面办公桌面办公分成:电子记事本、备忘录、手写录入系统、文字编辑、预约提醒、电子时钟、电子台历等;预约提醒、电子时钟、电子台历等均可使用An droid自带软件实现票务预订、计算器、股市等采用第三方软件集成。

网络功能具有网络功能,可浏览网页、登录QQ、MSN。

(直接安装第三方软件)多媒体功能桌面电子相册、音乐播放器、视频播放器;(An droid自带软件实现)终端广告植入系统终端以广告业务为主要收益来源,广告业务是终端的关键部分。

广告主要以图片、可点击互动广告等为主。

1.屏幕主界面布局(功能示意图)屏幕左侧区域为an droid 系统区域,右侧为广告播放区域。

VoIP电话机开发方案设计

VoIP电话机开发方案设计

VoIP电话机开发方案设计V oIP电话机开发方案设计话机图片话机按键功能介绍按键状态功能/显示Volume + 通话状态增加音量配置状态翻页(上翻)Volume - 通话状态减小音量配置状态翻页(下翻)Message 拨号状态听取话机的留言信息Speaker 通话状态手柄和免提之间的切换Mute 通话状态静音配置状态编辑或者删除Redial 拨号状态重拨上次的号码并且呼叫Hold 通话状态保持或者解除保持M1~M7 拨号状态快速拨号并呼叫Menu 待机状态短按进入快速键存储状态,长按进入话机菜单配置状态确认/进入下一级菜单Flash 配置状态退出/返回上一级菜单1 拨号状态“1”配置状态“1”, “space”, “@”, “_”, “-”, “/”, “%”2 拨号状态“2”配置状态“2”, “a”, “b”, “c”, “A”, “B”, “C”3 拨号状态“3”配置状态“3”, “d”, “e”, ”f”,“D”, “E”, “F”4 拨号状态“4”配置状态“4”, “g”, “h”, “I”, “G”, “H”, “I”5 拨号状态“5”配置状态“5”, “j”, “k”, “l”, “J”, “K”, “L”6 拨号状态“6”配置状态“6”, “m”, “n”, “o”, “M”, “N”, “O”7 拨号状态“7”配置状态“7”, “p”, “q”, “r”, “s”, “P”, “Q”, “R”, …S”8 拨号状态“8”配置状态“8”, “t”, “u”, “v”, “T”, “U”, “V”9 拨号状态“9”配置状态“9”, “w”, “x”, “y”, “z”, “W”, “X”, “Y”, “Z”0 拨号状态“0”配置状态“0”, “*”, “#”, “$”, “&”, “?”, “!”, “<”, “>”* 拨号状态“*”配置状态“*”, “.”# 拨号状态可以作为第一个号码拨出或相当于拨号结束标记产品特性2个网络接口WAN和LAN,支持PoE供电128x64图形点阵LCD屏幕可支持电容式触摸按键支持宽带编码,高品质语音保证如果有开发需求:TEL1800-3035-318李动态语音检测;舒适噪音生成;语音缓冲技术;可通过HTTP,FTP或TFTP方式升级;话机特性支持2条线路,热线,紧急号码呼出支持Hold,Transfer,Conference,Call Waiting, Call Park等功能来电显示,重拨,静音及免打扰(DND)。

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VoIP+DECT Phone Keypad Function List 1.1 机器图片
1.4 功能特点
产品特性
2个网络接口WAN和LAN,支持PoE供电
128x64图形点阵LCD屏幕
可支持电容式触摸按键
支持宽带编码,高品质语音保证
动态语音检测;舒适噪音生成;语音缓冲技术;
可通过HTTP,FTP或TFTP方式升级;
支持5个DECT子机,座机可与DECT子机实现对讲,转接及会议等功能。

话机特性
支持2条线路,热线,紧急号码呼出
支持Hold,Transfer,Conference,Call Waiting, Call Park等功能来电显示,重拨,静音及免打扰(DND)。

语音信箱,快速拨号
支持可编程多功能键。

通话记录:呼出、呼入、未接。

IP PBX 兼容功能
免打扰
Hold音乐
Call Waiting
Call Park
呼叫拒接
网络会议
呼叫规则, 立即呼出
音频特性
宽带编码: G.722
窄带编码:G.711μ/A, G723.1, G726, G.729AB
支持VAD、CNG、AEC、PLC、AJB、AGC等音频处理
全双工免提, 带自动回音消除
网络特性
支持SIP v1(RFC2543), V2(RFC3261)协议
支持IPv6
支持Prack, Rport,
支持DNS SRV(RFC3263)
冗余服务器支持(Failover模式和Redundant模式)
支持STUN内网穿透
支持三种DTMF模式:带内(In-band)、RFC2833、SIP INFO IP地址分配模式:Static/DHCP/PPPoE拨号
支持TFTP/DHCP/PPPoE客户端
支持Telnet/HTTP服务
支持WAN to LAN 镜像功能
配置管理
支持FTP/TFTP/HTTP/HTTPS/PnP方式的自动配置
支持三种配置模式:网页、话机及自动配置
自定义出厂配置文件
SSH管理功能
安全性
支持802.1x, VLAN QoS(802.1pq),
LLDP
物理特性
128x64点阵液晶屏幕
共计36个按键,包括10个可编程键
1个RJ-9(4P4C)手柄接口
2个RJ-45 10/100M以太网口
电源适配器:AV 100-240V输入,
DC 5V/2A输出
支持以太网供电(IEEE 802.3af)
功耗:2W
操作环境湿度:10-95%
工作环境温度:60°C以下
目前已测试的交换机:Asterisk,Avaya,Mitel等;。

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