Asterisk 安装和配置

CONFIGURING AN ASTERISK SERVERFrom Standard ERP's T elephony module, it is possible to fully configure an Asterisk server, whether it is installed locally on the same server as Standard ERP (only applicable on a Linux-based server) or on a separate remote server.It is also possible to integrate Standard ERP with an existing Asterisk server without managing its configuration directly from Standard ERP. Below are the steps required for a full integration, and will not explain what steps are required for local, remote, and existing servers.PBX ConnectionThe basic setting for integrating your Standard ERP system with an Asterisk server is a PBX Connection. A PBX Connection represents a connection to a unique actual VoIP Server.T o start with, you should create a new PBX Connection from the T elephony module in the PBX Connection register.A PBX Connection is defined primarily by:Code: The PBX unique identifier.Comment: A descriptive text.Type: The type of VoIP server to connect to; this can be chosen from a variety of Asterisk versions, TrixBox, Digium Switchvox, VPBX and 3CX Phone System 12 etc.Hostname: The hostname for the server being used.Host IP address: The IP address for the server being used.Port: The port through which the Asterisk's Management Interface (AMI) can be reached (5038 by default for Asterisk)Username: Username to connect to the AMI.Password:Password to connect to the AMI.Closed: A PBX Connection can be closed when it is no longer in use or valid.In the case of a locally or remotely installed Asterisk server, the T ype should be Asterisk 1.8 (AMI). The username and password can be freely set as they will be configured in the server for you.In case you are connecting to an existing server, the port, username, and password should match the content of your server's manager.conf file. Here is an example of manager configuration usable with Standard ERP:[general]enabled=yesport=5038bindaddr=0.0.0.0allowmultiplelogin=yesdisplayconnects=yestimestampevents=yes[myadmin]secret=passwordxyzdeny=0.0.0.0/0.0.0.0permit=1.2.3.4read=system,call,originatewrite=system,call,originateIn this case Port would be 5038, username myadmin and password passwordxyz. You should replace 1.2.3.4 by the IP address of your Standard ERP server.A PBX Connection has other fields organised in four tabs, and which are used when managing a local or remote server entirely from Standard ERP. Administrators using an existing server fully managed by some external means should skip over to the Contact records section.Dialling SettingsUnder this tab, you can configure the various telephony prefixes in use in your system.International Dial Prefix: T o dial out of your country.Country Code: T o dial in your country.Default Area Code: T o dial in your local area.External Line Prefix: T o dial out of your organisation.Skip Digit for International Calls:As its name indicates, you may also define digits to skip when dialling internationally.These parameters are used to place and receive calls, and to identify contacts based on their caller ID, including when using IAX (see below).RulesThis T ab allows to define a number of rules from various types:Ignore SIP Channel:In case of an existing server, this will ignore possible intermediary SIP channels to handle calls and instead only care about the end points. The Identifier is the name of the intermediary SIP channel to ignore.No Act For Calls Between Extensions Shorter Than: Disables the automatic Activity creation for internal calls (detected by the short length of internal extensions). This is only applicable for PBX Connections of the T ype Digium SwitchVox. The Identifier is the maximum length.Track number: Not used.Unique callers only: With this option, only one call will be displayed in Communicator even if there is more than one call from or to the number configured in Identifier.Remote AdministrationFrom this tab, you can define the following:Remote User: Linux user that will be used to copy the Asterisk configuration files to a remote serverRemote Configuration Directory: The path where to copy said configuration files. As such, it is important that the directory is writable by the Linux user and that your Standard ERP server has been set up to be able to connect directly to the remote Asterisk server without needing to enter a password (namely set up a Public Key Authentication between both servers).AsteriskFrom this final tab, you can enable the connection of your Asterisk server to other Asterisk server using IAX (Inter-Asterisk eXchange).Enable Inter-Asterisk eXchange(IAX):By ticking this option, you will allow all other PBX Connections configured in Standard ERP and set to use IAX to connect to this particular server as well as allow this server to connect to all other servers enabled for IAX and configured in Standard ERP.IAX Password: The password used by this server to connect to other IAX servers.Sending configuration to the serverThis section only applies to the local and remote servers situations.After completing the above configuration of a PBX Connection, you can already send the configuration to an Asterisk server.If you are running a local server, you can dump the configuration files by using the Local Asterisk Server settings from the T elephony module.First, you will need to install the Asterisk server by selecting Setup Asterisk Server in the Operation Menu. This will give you a warning pop-up reading “Starting download and installation of Asterisk server”. Click OK. This will download the binaries for the Asterisk Server from the HansaWorld servers and install them on your local server. The server will then be started. You should never have to use Setup Asterisk Server again after this.If you close the Local Asterisk Inspect window and reopen it, you will see the path where your server is installed.From the Operation Menu, you can also select “Update Asterisk Server Configuration” (which will dump the current configuration on the Asterisk server configuration directory and restart the Asterisk server so that the configuration is applied), Start Asterisk Server, and Stop Asterisk Server (which should both only rarely be used, for instance for external maintenance purposes).If you are running a remote server which is fully configured using Standard ERP, you should instead head to the Asterisk SIP Configuration Files maintenance in the Routines of the T elephony module.Use Paste Special in the PBX Connection to select the server you want to update, and tick Send Files to Server before running. If you do not tick this option, then the files will only be generated locally on your Standard ERP server.Note that this will only work if you have properly setup your PBX Connection and the Linux environment of your Asterisk server (see above).You can also select from the following other Maintenance Routines:!Asterisk SIP trunks.!Asterisk Users.These routines will generate respectively only the configuration files for the SIP trunks of a PBX Connection, or for its users, instead of regenerating all the files.Note that the files are only sent to the server but not applied. An administrator needs to connect manually to the Asterisk server and reload them. For instance by issuing a 'core reload' command from Asterisk's command line interface.Asterisk usersThe next step in setting up your Asterisk server is to create a number of users.This section is applicable for local and remote servers.From the T elephony module, you can create new Asterisk Users for your employees or partners.For each user, you can define:PBX Connections: One or more servers on which the user will be created and allowed to connect to. Leave blank to create the user on all PBX Connections configured.Name: A descriptive name.Username: Will be used to configure their SIP client.Password:Will be used to configure their SIP client.Group: No longer used.Caller ID number: The display number that might be shown to the party this user is calling. Note that this can easily be overridden by the configuration of a SIP client or SIP trunk. Especially when dialling out to international telephone numbers, Caller ID numbers are likely to get lost.Caller ID name: The display name that might be shown to the party this user is calling. Note that this can easilybe overridden by the configuration of a SIP client or SIP trunk.Especially when dialling out to a mobile or landline telephone number, plain text display names will be lost.Closed: A closed user will simply not be configured on the server and as such, it will not be possible to connect to the server using that user.No Queue Fallback: If an Asterisk User is marked as not being part of Queue Fallback, then that user will not be called when a queue is not staffed but is receiving a phone call.Note that after creating one (or more) Asterisk users, it is necessary to send the configuration to the server, as described above.Contact recordsThis section is applicable for all types of servers: local, remote, and existing.As of now, Asterisk users and Standard ERP users (Persons), and their contact cards are not connected and as such, Contact cards for your users will need to be filled in manually with their SIP contact details.In the SIP field of the contact record pertaining to your Asterisk user should be filled in as username@host name. Where username comes from the Asterisk User record, and hostname from the PBX Connection record.SIP TrunksThis section is applicable for local and remote servers.At this point of the configuration, you can place calls between users of your Asterisk server. T o reach out to the outside world, you will need a SIP trunk or VoIP trunk. Each country usually has several providers that can help you get started. As Asterisk is a commonly used VoIP server platform, it is easy to get help from your provider in general.A simple Internet search should allow you to find a number of SIP providers for your country.Using the information provided by your subscriber, you will be able to fill in the SIP Trunk record necessary for you to place calls to the rest of the world. A SIP provider will usually be able to sell you the usage of one, or more phone numbers that your contacts will be able to call to reach you. In some cases, your SIP provider might also allow you to place outgoing calls. Make sure to carefully select the SIP provider that is able to provide you with the capabilities you need to run your business smoothly.Setting up a SIP trunk comes with a wide array of technical possibilities, a number of which are supported inStandard ERP. We will detail some of those here but it is not possible to list all the possible technical configurations one can encounter.Code: Select a unique code for your SIP trunk.PBX Connection: Paste Special the PBX connection on which you want this SIP trunk to be terminated.Host: Fill in the host name or IP address provided by your SIP provider here. It might be that host and domain have the same value.Domain: Fill in the domain name provided by your SIP provider here. It might be that host and domain have the same value.Username: Fill in the username provided by your SIP provider here.Password: Fill in the information provided by your SIP provider here.Skip Digit for International Calls: This parameters operates similarly to that set in the PBX Connection but will apply to calls using the SIP trunk.Country Code:This parameters operates similarly to that set in PBX Connection but will apply to calls using the SIP trunk.Caller ID: The caller ID of your SIP trunk provider (optional).Allow anonymous calls: Lets the system accept anonymous calls coming from your providers.Allowed IPs: Only incoming calls coming from these IP addresses will be allowed. Please check with your SIP provider to only open the minimum number of addresses. (optional but important security point).Inbound phone numbers: A SIP provider may very well provide you several telephone numbers using the same SIP trunks. In certain cases, you will be given unique identifiers for each one of them. They should be filled in here. It might be that the usernames and passwords are the same as above.Trunk type:Set to Outbound calls only if you intend to input a separate configuration for Inbound Phone Numbers in the matrix as described just above. Set to In- and outbound calls if you do not have a separate configuration for Inbound Phone Numbers.IAX: Select this if your SIP trunk provider is providing you services using an Asterisk IAX trunk.Queues and MenusThis section is applicable for local and remote servers.Most of us are familiar with the telephony lines operated by large companies. A welcome Menu plays when you call into the support line of a company, after pushing a few digits on your phone and listening to a few more voice Menu messages, you are placed in a Queue. Thanks to Standard ERP's integration with Asterisk, your company can easily benefit from such technology.In Standard ERP's terminology, a Menu is used to select between different queues or menus; and a Queue is used to put in relation agents answering calls and external callers. Queues and Menus share a number of settings (Phone Numbers, Opening Times) and capabilities (Playing a sound upon arrival, when closed, etc.).Instructions for users to use queues can be found earlier in the document.A Queue contains the following information:Code: A unique identifier in Standard ERPQueue ID: A unique identifier in Asterisk which will be used by your employees to connect to the queue and start answering calls.SIP Trunk: The SIP trunk from where the calls will be arriving.Description: A free-text comment.Phone Number: (Optional) in the case where you want a direct number for callers to reach the queue without going through a Menu. Note: you can play a greetings message even in the case where callers go straight toa queue. You do not need a Menu to play a welcome message.Fallback number: An optional number to call in the case i) no agent is available in the queue AND ii) no one is logged in to the Asterisk server or everyone who is logged in is marked with “No Queue Fallback”.Open from/until: it is possible to define two sets of opening hours (to include the possibility of a lunch break for instance). In case only one set of opening hours is needed, use the first pair of “Open from”/”Open until”fields and leave the second pair blank.A Menu contains the following information:Code: A unique identifier.SIP Trunk: T he SIP trunk from where the calls will be arriving.Description: A free-text comment.Phone Number: The phone number for your contacts to dial in order to access the Menu. Optional in case the Menu is accessed via another Menu.Open from/until:It is possible to define two sets of opening hours (to include the possibility of a lunch break for instance). In case only one set of opening hours is needed, use the first pair of “Open from”/”Open until”fields and leave the second pair blank.Repeat every (s): The number of seconds between repeats of the message explaining to the caller his or her possible choices.A matrix finally allows you to configure the different Menus and Queues reachable from this Menu:Number: The digit to press for the user to enter the selected Queue or Menu. Note that in the case pressing the digit leads to entering a Queue, the digit need not be the same as the Queue ID defined in the Queue.Queue: Paste Special to an existing Queue (note, if you select this, you should not select a Menu as well).Menu: Paste Special to an existing Menu (note, if you select this, you should not select a Queue as well).Comment: A free-text comment as a reminder of what the selected Queue or Menu might be.Using Menus, you can cascade multiple levels of Menus. However, once a caller has joined a Queue, he or she will not be able to go back to another Queue or Menu.The last remaining part of the configuration is now to assign sound files to be played to guide your callers through your Menus and Queues.Whereas all the previous configuration was done in Registers of the T elephony module, sounds will be configured from the Settings of the T elephony module. More precisely, from the PBX Sounds setting.First, create a new PBX Sound. Then in Event, use Paste Special to select the type of Event that will trigger the sound file to play. The Event you select will affect whether you are selecting a Queue or a Menu in the following field. Available Events are:Initial Queue Message: Played as an initial greeting when a caller reaches a Queue.Line Busy: Played after 30 seconds of a caller waiting in a Queue.Menu Closed: Played whenever a caller arrives to a Menu outside of the defined opening hoursMenu Message: Played as an initial greeting when a caller enters a Menu (should also describes the options available from the Menu and the digits associated with each function).Music on Hold:Music to play while the caller is waiting in a Queue.Queue Closed: Played whenever a caller arrives to a Queue outside of the defined opening hours.Once an Event is selected, use Paste Special to select the Queue/Menu where the sound file should be used. Only one Queue or Menu can be selected. After Saving the Record, you can now attach a file to the Record following the usual way of dragging and dropping the file over the paperclip icon or into the Document Manager window which you can open by double-clicking the paperclip icon.Note: the attached sound file must be a mono.wav file, sampled at 8kHz.Remember to send the configuration to the server once done. The sound files will be copied during that stage a s well.。

23.4 Asterisk的
未来
23.5**的未 来
作者介绍
这是《Asterisk权威指南（原书第5版）》的读书笔记模板，暂无该书作者的介绍。
读书笔记
这是《Asterisk权威指南（原书第5版）》的读书笔记模板，可以替换为自己的心得。
精彩摘录
这是《Asterisk权威指南（原书第5版）》的读书笔记模板，可以替换为自己的精彩内容摘录。
5.3配置Asterisk
5.4进行测试以确保 设备已经注册成功
5.5用于设备测试的 基本dialplan
5.6透过现象看本质： 第一次通话
5.7小结
6.1 dialplan语法
6.2一个简单的 dialplan
6.3构建交互式 dialplan
6.4小结
7.1中继基础
7.2用于外线连接的 基本dialplan
15.8 ACD队列的数 据库集成
15.7存储呼叫详情 记录
15.9小结
16.1 IVR的组成
16.2 IVR设计的考 虑因素
16.3 Asterisk的 IVR模块
16.4使用CURL()创 建简单的IVR
16.6语音识别和文 本语音转换
16.5录制提示音的 IVR函数
16.7小结
1
17.1呼叫文件
10.9小结
11.2驻留和寻呼
11.3高级会议 11.4小结
12.1创建一个 1
简单的ACD队 列
2
12.2队列成员
3
12.3高级队列
4 12.4队列统计：
queue_log文 件
5
12.5小结
13.2使用hint指令 的分机状态
13.1设备状态信息

centos6.4安装Asterisk+freepbx2013-07-26 11:02:36分类： LINUXCentos 6.4 上安装Asterisk 11.3+FreePBX 2.11最近项目需要用asterisk来做一下研究，顺手记录一下安装过程，官方文档都太旧了，地址 https:///wiki/display /AST/Asterisk+Packages现在asterisk版本都已经出到了11.3.0，所以以下记录都是针对最新版本在Centos 6.4上的安装。

1. 安装asteriskasterisk官方为RedHat/Centos/Ubuntu等主流发行版提供好了相应的包，直接用yum/apt就可以完成安装，还是相当方便的。

sudo rpm -Uvh/centos/6/current/x86 _64/RPMS/asterisknow-version-3.0.0-1_centos6. noarch.rpmsudo yum -y update这是目前的最新版本，如果asterisk有更新的版本，可以先去/centos/里找对应的那个包的地址，注意区分centos版本和i386/x86_64架构然后先更新一把系统，重启一下（因为可能会更新内核版本）好了，可以开始装asterisk，目前版本是11.xsudo yum install asterisk asterisk-configs --enablerepo=asterisk-11安装需要的一些模块sudo yum install dahdi-linux dahdi-tools libpri验证安装如果一切正常的话，到这里就安装完成了，可以用以下命令检查dahdi 和asterisk是否已经运行sudo service dahdi statussudo service asterisk status如果没有运行的话，用sudo service dahdi start 和sudo service asterisk start 来启动如果dahdi正常运行，用lsmod|grep dahdi 应该可以看到加载了一些相应的模块如果asterisk正常运行，在命令行下用sudo asterisk -r 可以连接到控制台，到这里验证安装完成。

asterisk,mysql,freepbx完整安装手册1 Install Asterisk server1.1 Install CentOS 5.3Description :CentOS is an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. CentOS conforms fully with the upstream vendors redistribution policy and aims to be 100% binary compatible.●Need install Web Server , Mail Server , MySQL Database , Development T ools●Enable W eb server, Mail server, MySql Database, Development tools.●Disable firewall, SElinux.●Config your network, make sure you can connect to the internet.1.2 Update your system#yum -y update1.3 Install dependencies and extra packages# yum install e2fsprogs-devel keyutils-libs-devel krb5-devel libogg libselinux-devel libsepol-devel libxml2-devel libtiff-devel gmp php-pear php-pear-DB php-gd php-mysqlphp-pdo kernel-devel ncurses-devel audiofile-devel libogg-devel openssl-devel mysql-devel zlib-devel perl-DateManip sendmail-cf1.4 Install lame-3.97Description :LAME is an educational tool to be used for learning about MP3 encoding. The goal of the LAME project is to use the opensource model to improve the psycho acoustics, noise shaping and speed of MP3. LAME is not for everyone - it is distributed as source code only and requires the ability to use a C compiler. However, many popular ripping and encoding programs include the LAME encoding engine.# cd /usr/src# wget /doc/945198382.html,/sourceforge/lame/l ame-3.97.tar.gz# tar zxvf lame-3.97.tar.gz# cd lame-3.97# ./configure# make# make install1.5 Install libpriDescription :libpri is a C implementation of the Primary Rate ISDN specification. It wasbased on the Bellcore specification SR-NWT-002343 for National ISDN. As ofMay 12, 2001, it has been tested work with NI-2, Nortel DMS-100, andLucent 5E Custom protocols on switches from Nortel and Lucent.# cd /usr/src# wget /doc/945198382.html,/pub/libpri/libpri-1.4-current.tar.gz# tar zxvf libpri-1.4-current.tar.gz# cd libpri-1.4.10/# make# make install1.6 Install mpg123The mpg123 is make for the “music on hold” module to up custom *.mp3 or *.wav file to PBX.#tar xvf mpg123-1.9.0.tar.bz2#./configure#make#make install1.7 Install Dahdi / Dahdi-toolsDescription :Dahdi is a short for ZAPata TELephony.This packet is the drive of the Digital Cards# cd /usr/src# download dahdi-linux-current.tar.gz dahdi-tools-current.tar.gz# tar zxvf dahdi-linux-2.2.0.2# cd zaptel-1.4.12.1# make# make install# cd .# . tar zxvf dahdi-tools-current.tar.gz# ./configure# make# make install# make config/etc/init.d/dahdi start# echo "/etc/init.d/dahdi start" >> /etc/rc.d/rc.local# dahdi_genconf //check the TDM400P board automatic#copy dahdi_channels.conf content to chan_dahdi.conf# dahdi_cfg –vv1.8 Install asterisk1.7.1 Install asterisk-1.6.1.4Description :Asterisk is the world's leading open source PBXi, telephony engine, and telephony applications toolkit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions...for free.# cd /usr/src# useradd -c "Asterisk PBX" -d /var/lib/asterisk asterisk# mkdir /var/run/asterisk# mkdir /var/log/asterisk# chown -R asterisk:asterisk /var/run/asterisk# chown -R asterisk:asterisk /var/log/asterisk# chown -R asterisk:asterisk /var/lib/php/session/# sed -i "s/User apache/User asterisk/" /etc/httpd/conf/httpd.conf# sed -i "s/Group apache/Group asterisk/" /etc/httpd/conf/httpd.conf# sed -i "s/AllowOverride None/AllowOverride All/" /etc/httpd/conf/httpd.conf# download asterisk-1.6.1.4.tar.gz# tar zxvf asterisk/asterisk-1.6.1.4.tar.gz# cd asterisk-1.6.1.4# ./configure# make# make install# make samples# make config1.7.2 Install asterisk-addons-1.6.1.1Description :This package contains additional modules for Asterisk which are, for one reasonor another, not included in the normal base distribution. Many of thesemodules are experimental.# cd /usr/src# download asterisk-addons-1.6.1-current.tar.gz# tar zxvf asterisk-addons-1.6.1-current.tar.gz# cd asterisk-addons-1.6.1.1# ./configure# make# make install# make samples1.7.3 Install asterisk-soundsDescription :This packet is not for hardware dependency. It’s just the necessary packet (Include many sound file) for asterisk server.Download the packet asterisk-sounds-1.2.1.tar.gz# cd /usr/src# tar zxvf asterisk-sounds-1.2.1.tar.gz# cd asterisk-sounds-1.2.1# make install1.9 Install FreePBXDescription :FreePBX is for both developers and people searching for a Business Phone System (or a really fancy home one).# cd /usr/src# wget/doc/945198382.html,/sourceforge/ampo rtal/freepbx-2.3.1.tar.gz# tar zxvf freepbx-2.5.0.tar.gz# cd freepbx-2.5.0# service httpd start# service mysqld start# chkconfig httpd on# chkconfig mysqld on1.9.1 Config MySql# mysqladmin create asterisk# mysqladmin create asteriskcdrdb# mysql asterisk < SQL/newinstall.sql# mysql asteriskcdrdb < SQL/cdr_mysql_table.sql# mysql# GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY 'occvoip';# GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY 'occvoip';# mysqladmin -u root password 'occvoip'1.9.2 Install freepbx# vi /etc/asterisk/asterisk.conf++++++++++++++++++++++++++++++++Modify [directories](!) ==> [directories]Modify /var/run ==> /var/run/asterisk++++++++++++++++++++++++++++++++# cd /usr/src/freepbx-2.5.0# ./start_asterisk start# ./install_amp --username=asteriskuser --password=occvoip# echo /usr/local/sbin/amportal start >> /etc/rc.localOpen browser to http://ipaddressofpbx#Click the FreePBX Administration to config the FreePBX. Then click Apply configuration changes.Download modules.: click Module Admin to download this modules:Feature codeVoicemailPhonebook Directory toolPhonebookSpeed dialDay night modeIVRFollow meRing groupCall ForwardCall WaitingCallbackConferencesDo-Not-Disturb (DND)Info ServicesannouncementMisc ApplicationsMisc DestinationsMusic on HoldPIN SetsParking lotAsterisk Info toolCustom Applications tool# reboot1.10 Open browser to http://ipaddressofpbxClick the FreePBX Administration to config the FreePBX.。