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translating---geekpi
How to install Asterisk on the Raspberry Pi
============================================================
> Are you looking for a phone system for your small business or home office?
![How to install Asterisk on the Raspberry Pi](https://opensource.com/sites/default/files/styles/image-full-size/public/images/life/life-raspberrypi_0.png?itok=wxVxQ0Z4 "How to install Asterisk on the Raspberry Pi")
>Image credits : Dwight Sipler on [Flickr][8]
Are you looking for a phone system for your small business or home office? I've been interested in a scalable VoIP (Voice over IP) solution, and that's when I came across an implementation of [Asterisk][9] on the Raspberry Pi.
My curiosity was piqued and I was determined to give it a try, so I [downloaded][10] the software from [Asterisk][11] and then set about building the server using my Raspberry Pi 3.
### Getting started
First, I burned the downloaded image onto a MicroSD card; the suggested minimum is 4GB. After transferring the image to the MicroSD card and inserting it into the appropriate slot on the Raspberry Pi, I connected an Ethernet cable to the Pi and to an Ethernet port on my home router.
More on Raspberry Pi
* [Our latest on Raspberry Pi][1]
* [What is Raspberry Pi?][2]
* [Getting started with Raspberry Pi][3]
* [Send us your Raspberry Pi projects and tutorials][4]
Next, I opened a terminal on my Linux computer and entered **ssh root@192.168.1.8**, which is the IP address of my server. I was prompted to log in as root on the **raspbx**. The default password is "raspberry." (For security reasons, be sure to change your passwords from the default settings if you plan to do more than just try it out.)
Once I was logged into the shell on the **raspbx,** I then needed to prepare the server for use. Following the [documentation][12] provided on the site, I created new host keys as directed by entering **regen-hostkeys** at the shell prompt. Then, I configured the time zone for the server by entering **configure-timezone** at the shell prompt. I configured the locale setting by entering, **dpkg-reconfigure locales** at the prompt. I also installed [Fail2Ban][13] to provide security on my server.
Now I was ready to test my configuration.
### Testing
I logged out of the **raspbx** shell and then opened a browser and pointed it at the IP address of my server. Loading the server IP address into the browser, I was presented with a lovely login page.
[FreePBX][14] provides a very nice web-based, open source graphical user interface, which I used to control and configure Asterisk (find on [GitHub][15]). (FreePBX is licensed under the GPL.) I used it to complete the rest of the configuration. The default login for FreePBX is**username:admin;password:admin**.
![FreePBX_Login_Screen](https://opensource.com/sites/default/files/freepbx_login_screen.png "FreePBX_Login_Screen")
Once in, I navigated to the Application Menu, which is located at the upper left of the display. I clicked on the menu link and selected the second option, which is Applications, and selected the fourth option, which is labeled Extensions. From there I created a **New Chan_Sip** extension.
![](https://opensource.com/sites/default/files/add_a_new_chan_sip_extension.png)
I configured a **Sip** extension user with a password. Passwords are either automatically generated or you can elect to create your own.
Now that I had a functioning extension I was anxious to try out my new VoIP server. I downloaded and installed [Yate Client][16], which I discovered in the process of building the server. After installing [Yate][17], I wanted to test the connectivity with the server. I discovered that I could connect to the server for an echo test using Yate and entering ***43**. I was really excited when I heard the instructions through the client.
![](https://opensource.com/sites/default/files/echotest.png)
I decided to create another **Sip** extension so that I could test the voicemail capabilities of the system. Once I completed that I used the Yate client to call that extension and leave a brief voice message. Then using Yate again, I called that extension and entered ***97** and retrieved the voice message. Then I wanted to see if I could use my new server to call an outside line. Returning to the menu, I chose the Connectivity option and added a Google Voice line.
![Google_Voice_Connectivity](https://opensource.com/sites/default/files/google_voice_connectivity.png "Google_Voice_Connectivity")
Then I returned to the Connectivity menu and added Google Voice to the Outbound routes.
![Google_Voice_outbound_route](https://opensource.com/sites/default/files/google_voice_outbound_route.png "Google_Voice_outbound_route")
### Completing a call
Returning to the Yate client, I entered an outside line and successfully completed that call.
I'm convinced that this particular VoIP solution could easily work for a small office. According to the [Frequently Asked Questions][18] section of the RasPBX website, a typical Raspberry Pi system could support up to 10 concurrent calls on a Raspberry Pi 1.
Asterisk has many nuances and the FreePBX software easily leveraged them.
_For more information about the Asterisk on Raspberry Pi project, follow their[blog][5]. You can find additional information about [FreePBX source code][6] on their website._
--------------------------------------------------------------------------------
作者简介：
Don Watkins - Educator, education technology specialist, entrepreneur, open source advocate. M.A. in Educational Psychology, MSED in Educational Leadership, Linux system administrator, CCNA, virtualization using Virtual Box. Follow me at @Don_Watkins .
----------
via: https://opensource.com/article/17/4/asterisk-raspberry-pi-3
作者：[Don Watkins][a]
译者：[译者ID](https://github.com/译者ID)
校对：[校对者ID](https://github.com/校对者ID)
本文由 [LCTT](https://github.com/LCTT/TranslateProject) 原创编译，[Linux中国](https://linux.cn/) 荣誉推出
[a]:https://opensource.com/users/don-watkins
[1]:https://opensource.com/tags/raspberry-pi?src=raspberry_pi_resource_menu
[2]:https://opensource.com/resources/what-raspberry-pi?src=raspberry_pi_resource_menu
[3]:https://opensource.com/article/16/12/getting-started-raspberry-pi?src=raspberry_pi_resource_menu
[4]:https://opensource.com/article/17/2/raspberry-pi-submit-your-article?src=raspberry_pi_resource_menu
[5]:http://www.raspberry-asterisk.org/blog/
[6]:https://www.freepbx.org/development/source-code/
[7]:https://opensource.com/article/17/4/asterisk-raspberry-pi-3?rate=zM9tOp0HEPyOUq31Np__W0QNnuAfWATkdkixOdSysDY
[8]:http://www.flickr.com/photos/photofarmer/272567650/
[9]:http://www.asterisk.org/
[10]:http://download.raspberry-asterisk.org/raspbx-28-01-2017.zip
[11]:http://www.raspberry-asterisk.org/downloads/
[12]:http://www.raspberry-asterisk.org/documentation/
[13]:http://www.raspberry-asterisk.org/documentation/#fail2ban
[14]:https://www.freepbx.org/
[15]:https://github.com/asterisk/asterisk/blob/master/LICENSE
[16]:http://yateclient.yate.ro/index.php/Download/Download
[17]:https://en.wikipedia.org/wiki/Yate_(telephony_engine)
[18]:http://www.raspberry-asterisk.org/faq/
[19]:https://opensource.com/user/15542/feed
[20]:https://opensource.com/article/17/4/asterisk-raspberry-pi-3#comments
[21]:https://opensource.com/users/don-watkins

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如何在树莓派中安装 Asterisk
============================================================
> 你是否在为小型企业或家庭办公室寻找电话系统？
![How to install Asterisk on the Raspberry Pi](https://opensource.com/sites/default/files/styles/image-full-size/public/images/life/life-raspberrypi_0.png?itok=wxVxQ0Z4 "How to install Asterisk on the Raspberry Pi")
>图片版权： Dwight Sipler 的 [Flickr][8]
你是否在为小型企业或家庭办公室寻找电话系统？我一直对可扩展 VoIP（Voice over IP）解决方案感兴趣，我在树莓派上找到 [Asterisk][9] 的一个实现。
我的好奇心被激起了，我决心尝试一下，所以我从 [Asterisk][11] 官网[下载][10]了它，然后使用我的树莓派 3 构建服务器。
### 准备开始
首先，我将下载的镜像刻录到 MicroSD 卡上。建议的最小值是 4GB。将镜像传输到 MicroSD 卡并插到树莓派上的相应插槽中后，我将网线连接到树莓派和家庭路由器上的以太网端口中。
更多关于树莓派的
* [我们最新的树莓派教程][1]
* [什么是树莓派？][2]
* [开始使用树莓派][3]
* [给我们发送你的树莓派项目和教程][4]
接下来，我在 Linux 上打开一个终端，并输入 **ssh root@192.168.1.8**，这是我的服务器的IP地址。我被提示以 root 用户身份登录到 **raspbx** 上。默认密码是 “raspberry”。 （出于安全考虑，如果你打算再多试试，请务必更改默认密码。）
当我登录到了 raspbx 上的 shell 后，接下来我需要准备配置了。根据网站上提供的[文档][12]，我在 shell 下输入 **regen-hostkeys** 来创建新的主机密钥。然后输入 **configure-timezone** 来配置服务器的时区。我通过在提示符下输入 **dpkg-reconfigure locales** 来配置区域设置。我也安装了 [Fail2Ban][13] 来提供服务器的安全性。
现在我准备测试我的配置。
### 测试
我从 **raspbx** shell 中登出，然后打开浏览器并输入我的服务器的 IP 地址。将服务器 IP 地址加载到浏览器中，我看到了一个可爱的登录页面。
[FreePBX][14] 提供了一个非常好的基于 Web 的开源图形用户界面，我用它来控制和配置 Asterisk（在 [GitHub][15] 上找到）。（FreePBX 是 GPL 许可的）。我用它来完成其余的配置。FreePBX 的默认登录账号为 **用户名：admin； 密码：admin**。
![FreePBX_Login_Screen](https://opensource.com/sites/default/files/freepbx_login_screen.png "FreePBX_Login_Screen")
登录之后，我进入位于显示屏左上方的 “Application Menu”。点击菜单链接并选择了第二个选项，即 “Applications”，接着选择了第四个选项，其中标有 “Extensions”。从那里我选择创建一个 **New Chan_Sip** 分机。
![](https://opensource.com/sites/default/files/add_a_new_chan_sip_extension.png)
我使用密码配置了一个 **sip** 分机用户。密码是自动生成的，也可以选择创建自己的密码。
现在我有了一个完整的分机，我急于尝试我的新的 VoIP 服务器。我下载并安装了[ Yate 客户端][16]，这是在构建服务器的过程中发现的。安装 [Yate][17] 之后，我想测试与服务器的连接。我发现我可以使用 Yate 连接到服务器并输入 **43** 进行回声测试。当我听到客户端指示时，我感到很激动。
![](https://opensource.com/sites/default/files/echotest.png)
我决定创建另外一个 **sip** 分机，这样我就可以测试系统的语音信箱功能。 在完成后，我使用 Yate 客户端来呼叫这个分机，并留下了简短的语音留言。然后再次使用 Yate 呼叫该分机并输入 **97** 来检索语音留言。然后我想看看我是否可以使用我的新服务器来呼叫外线。返回到菜单，选择 “Connectivity” 选项，并添加了 Google Voice 号码。
![Google_Voice_Connectivity](https://opensource.com/sites/default/files/google_voice_connectivity.png "Google_Voice_Connectivity")
接着我返回到 “Connectivity” 菜单将 Google Voice 添加到出站路由中。
![Google_Voice_outbound_route](https://opensource.com/sites/default/files/google_voice_outbound_route.png "Google_Voice_outbound_route")
### 完成一个电话
回到 Yate 客户端，我呼叫了一个外线并成功完成了这个电话。
我相信这个特定的 VoIP 解决方案可以轻松地为一个小型办公室工作。根据 RasPBX 网站的[常见问题][18]部分，典型的树莓派系统可以在树莓派 1 上支持多达 10 个并发呼叫。
Asterisk 有很多细微差别的功能，FreePBX 则可以很容易地利用它们。
_关于树莓派上的 Asterisk 的更多信息，请参考他们的[博客][5]。你可以在他们的网站上找到有关[ FreePBX 源代码][6]的其他信息。
--------------------------------------------------------------------------------
作者简介：
Don Watkins - 教育家、教育技术专家、企业家、开源倡导者。教育心理学硕士、教育领导硕士、Linux 系统管理员、CCNA、使用 Virtual Box 虚拟化。关注我 @Don_Watkins。
----------
via: https://opensource.com/article/17/4/asterisk-raspberry-pi-3
作者：[Don Watkins][a]
译者：[geekpi](https://github.com/geekpi)
校对：[校对者ID](https://github.com/校对者ID)
本文由 [LCTT](https://github.com/LCTT/TranslateProject) 原创编译，[Linux中国](https://linux.cn/) 荣誉推出
[a]:https://opensource.com/users/don-watkins
[1]:https://opensource.com/tags/raspberry-pi?src=raspberry_pi_resource_menu
[2]:https://opensource.com/resources/what-raspberry-pi?src=raspberry_pi_resource_menu
[3]:https://opensource.com/article/16/12/getting-started-raspberry-pi?src=raspberry_pi_resource_menu
[4]:https://opensource.com/article/17/2/raspberry-pi-submit-your-article?src=raspberry_pi_resource_menu
[5]:http://www.raspberry-asterisk.org/blog/
[6]:https://www.freepbx.org/development/source-code/
[7]:https://opensource.com/article/17/4/asterisk-raspberry-pi-3?rate=zM9tOp0HEPyOUq31Np__W0QNnuAfWATkdkixOdSysDY
[8]:http://www.flickr.com/photos/photofarmer/272567650/
[9]:http://www.asterisk.org/
[10]:http://download.raspberry-asterisk.org/raspbx-28-01-2017.zip
[11]:http://www.raspberry-asterisk.org/downloads/
[12]:http://www.raspberry-asterisk.org/documentation/
[13]:http://www.raspberry-asterisk.org/documentation/#fail2ban
[14]:https://www.freepbx.org/
[15]:https://github.com/asterisk/asterisk/blob/master/LICENSE
[16]:http://yateclient.yate.ro/index.php/Download/Download
[17]:https://en.wikipedia.org/wiki/Yate_(telephony_engine)
[18]:http://www.raspberry-asterisk.org/faq/
[19]:https://opensource.com/user/15542/feed
[20]:https://opensource.com/article/17/4/asterisk-raspberry-pi-3#comments
[21]:https://opensource.com/users/don-watkins