“FreePBX 12 中的NAT 设置”的版本间的差异

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（创建页面，内容为“'''NAT Configuration FreePBX 12''' '''NAT issues''' Some of the biggest problems that plague people such as "one way audio" or "Calls dropping after XX Seconds" ar...”）
 
 
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'''NAT Configuration FreePBX 12'''
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'''FreePBX 12 以上版本NAT 用户配置指南'''
  
 
'''NAT issues'''
 
'''NAT issues'''
  
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Some of the biggest problems that plague people such as "one way audio" or "Calls dropping after XX Seconds" are caused by NAT not being correctly setup.
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VoIP 最大的问题就是通常人们说的单通问题，或者几秒钟以后就掉线的问题。这些问题通常就是NAT配置不当导致。
  
 
'''Make sure you have a resolvable address on the Internet.'''
 
'''Make sure you have a resolvable address on the Internet.'''
  
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If you don't want to pay a few bucks to get a static IP address, and are served by an ISP that periodically changes your IP address, then get an account with a dynamic DNS service such as [http://www.dyndns.com/services/dns/dyndns/ DynDNS] . Your router may already have built-in support for one or more of these services, if so, use one that your router supports and then configure your router to automatically update your dynamic address when your ISP changes your IP address. Failing that, you can set up an updater program such as inadyn, there are instructions for doing that at [http://michigantelephone.wordpress.com/2006/06/23/a-quick-way-for-freepbxelastixtrixbox-users-to-auto-update-dyndns-maybe/ this blog page]
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如果用户没有购买静态IP地址的话，，用户可能经常会修改IP地址。建议设置一个动态的DNS服务。
  
 
'''Adding NAT information in  FreePBX '''
 
'''Adding NAT information in  FreePBX '''
  
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All of your settings will be under Settings > Asterisk SIP settings
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在高级配置菜单，点击 Asterisk SIP settings
  
 
http://wiki.freepbx.org/download/attachments/24051965/SipRightMenu.png
 
http://wiki.freepbx.org/download/attachments/24051965/SipRightMenu.png
  
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Next Click Chan SIP in the right menu
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然后点击SIP 配置
  
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[[image:Picture 2|165x73px|C:\aef3702e49caad622478b08b9c1f3e09]]
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右边菜单支持的是 FreePBX 12。 FreePBX 11 以下版本是主界面配置
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VERSION SPECIFIC
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This right menu is specific to FreePBX 12. In 2.11 all settings are on the main page
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'''Static IP from your ISP '''
 
'''Static IP from your ISP '''
  
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Select "Static IP" and enter your external IP
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选择"Static IP" ，输入外部IP地址
  
 
http://wiki.freepbx.org/download/attachments/24051965/settings_nat_static.png
 
http://wiki.freepbx.org/download/attachments/24051965/settings_nat_static.png
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'''Dynamic IP Updated through dynamic IP service'''
 
'''Dynamic IP Updated through dynamic IP service'''
  
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Select "Dynamic IP" and put the Full host name in such as &nbsp;"<nowiki>foo.dyndns.net</nowiki>"
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选择 "Dynamic IP" ，输入主机名称全称，例如"<nowiki>foo.dyndns.net</nowiki>"
  
 
http://wiki.freepbx.org/download/attachments/24051965/settings_nat_dynamic.png
 
http://wiki.freepbx.org/download/attachments/24051965/settings_nat_dynamic.png
  
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REMEMBER
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完成配置以后，要点击 "submit" 和点击"APPLY" 按钮。
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Icon
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Whenever you make a change in the UI you need to "submit" the changes then click "APPLY" at the top
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&nbsp;
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After clicking "submit changes" and the Red Apply click "General SIP Settings" on the right menu
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[[image:Picture 5|165x73px|C:\aef3702e49caad622478b08b9c1f3e09]]
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'''Local Networks'''
 
'''Local Networks'''
  
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Under "NAT" you will see a box for "Local Networks"&nbsp;
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在 "NAT"下，用户会看到一个勾选值 "Local Networks"。
  
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In these boxes you will put your LAN information with the IP in the first box and the SUBNET in the second box
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这里输入内网IP地址和子网地址。
  
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If your IP is 192.168.0.254 you would put 192.168.0.0 / 255.255.255.0
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如果用户内网地址是192.168.0.254，那么需要设置为192.168.0.0 / 255.255.255.0
  
 
http://wiki.freepbx.org/download/attachments/24051965/settings_nat_localnet.png
 
http://wiki.freepbx.org/download/attachments/24051965/settings_nat_localnet.png
  
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Click "Submit changes" And the red "APPLY" button.
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点击提交，然后点击''APPLY Config''
  
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&nbsp;
 
  
 
'''RTP Port Range'''
 
'''RTP Port Range'''
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'''Open the SIP and RTP ports to your Asterisk server'''
 
'''Open the SIP and RTP ports to your Asterisk server'''
  
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You must make sure that you open the correct UDP ports in your router's firewall and pointed at your Asterisk server. For SIP protocol, open UDP (NOT TCP) port 5060 (SIP) AND ports 10000-20000 (RTP, which must also be defined in /etc/asterisk<nowiki>/rtp.co</nowiki>nf, see below). All these ports are UDP, opening the TCP ports will NOT help anything and may expose your system needlessly. While you are in your firewall configuration, you may as well also open UDP port 4569 (IAX), since sooner or later you'll probably want to accept IAX connections.
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用户必须确认开启了正确的端口，检查防火墙UDP端口设置，设置转发端口到用户的Asterisk端口。 对于SIP协议，用户需要开启UDP(不是TCP) 端口5060 (SIP) ，并且开启端口 10000-20000 来传输语音(RTP 在系统文件/etc/asterisk<nowiki>/rtp.co</nowiki>nf中定义)。 所有这些端口都是UDP端口，开启TCP不会对NAT有任何帮助。 配置防火墙时，用户可能还要打开 UDP port 4569 (IAX)，用户有时可以通过IAX连接到asterisk服务器。
  
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You can see the actual range under the "General SIP Settings" page.
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用户可以看到实际开启的端口
  
 
http://wiki.freepbx.org/download/attachments/24051965/RTPSettings.png
 
http://wiki.freepbx.org/download/attachments/24051965/RTPSettings.png
  
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&nbsp;
 
  
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If the port values are any different, change them. &nbsp;These MUST match what you opened in your firewall
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如果实例的端口和本地应用有什么不同，用户需要检查自己的防火墙和相应的实际端口配置。
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Warning
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Icon
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You may need to set this to start with 10001, as port 10000, conflicts with usage in Webmin. This only matters if you have installed Webmin
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Some people feel the need to open fewer than 10,000 ports. I don't recommend this because six months from now when you start having audio problems you may not remember that you opened fewer than the recommended number of ports, and may spend hours troubleshooting the issue. But if you are simply obsessive about open ports, remember that each open SIP connection may require as many as FOUR concurrent ports, so don't cut it down to some ridiculously small number. For the non-paranoid, I suggest sticking with the recommendations above (and remember, if a hacker is looking at ports on your system, he's going to scan ALL of them, so having fewer UDP ports open really doesn't make you any more secure).
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用户可能需要开启端口从10001开始，10000端口可能和Webmin端口冲突。如果没有安装Webmin则无需担心冲突。

2015年11月3日 (二) 14:42的最新版本

FreePBX 12 以上版本NAT 用户配置指南

NAT issues

VoIP 最大的问题就是通常人们说的单通问题，或者几秒钟以后就掉线的问题。这些问题通常就是NAT配置不当导致。

Make sure you have a resolvable address on the Internet.

如果用户没有购买静态IP地址的话，，用户可能经常会修改IP地址。建议设置一个动态的DNS服务。

Adding NAT information in  FreePBX 

在高级配置菜单，点击 Asterisk SIP settings

SipRightMenu.png

然后点击SIP 配置

右边菜单支持的是 FreePBX 12。 FreePBX 11 以下版本是主界面配置

 

Set NAT as yes

Static IP from your ISP 

选择"Static IP" ，输入外部IP地址

settings_nat_static.png

Dynamic IP Updated through dynamic IP service

选择 "Dynamic IP" ，输入主机名称全称，例如"foo.dyndns.net"

settings_nat_dynamic.png

完成配置以后，要点击 "submit" 和点击"APPLY" 按钮。

Local Networks

在 "NAT"下，用户会看到一个勾选值 "Local Networks"。

这里输入内网IP地址和子网地址。

如果用户内网地址是192.168.0.254，那么需要设置为192.168.0.0 / 255.255.255.0

settings_nat_localnet.png

点击提交，然后点击APPLY Config


RTP Port Range

Open the SIP and RTP ports to your Asterisk server

用户必须确认开启了正确的端口，检查防火墙UDP端口设置，设置转发端口到用户的Asterisk端口。 对于SIP协议，用户需要开启UDP(不是TCP) 端口5060 (SIP) ，并且开启端口 10000-20000 来传输语音(RTP 在系统文件/etc/asterisk/rtp.conf中定义)。 所有这些端口都是UDP端口，开启TCP不会对NAT有任何帮助。 配置防火墙时，用户可能还要打开 UDP port 4569 (IAX)，用户有时可以通过IAX连接到asterisk服务器。

用户可以看到实际开启的端口

RTPSettings.png


如果实例的端口和本地应用有什么不同，用户需要检查自己的防火墙和相应的实际端口配置。

用户可能需要开启端口从10001开始，10000端口可能和Webmin端口冲突。如果没有安装Webmin则无需担心冲突。

取自“http://freepbx.org.cn/wiki/index.php?title=FreePBX_12_中的NAT_设置&oldid=518”