This article describes advertising your public IP in your SIP signaling on Asterisk-based PBXs. If your Asterisk-based PBX is behind NAT or is advertising a private IP, such as 192.168.0.2, in your SIP signaling, then you'll need to make the following changes to prevent intermittent one-way audio and other call issues.
NOTE: The following example assumes that your Asterisk PBX uses the IP address 192.168.1.x. |
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You will need two firewall policies for Flowroute's Direct Media set up on your router to prevent call and audio issues.
- Go to /etc/asterisk/.
- Using a text editor, open sip.conf.
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Modify the file with the following:
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externip=xxx.xxx.xx.xx
where
xxx.xxx.xx.xx
is your public IP address. This can be found on the router's administration web page or by going to www.whatismyip.com, which will read and display your IP address. -
localnet=192.168.1.0/255.255.255.0
This must match your local subnet.
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nat=yes
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Restart Asterisk.
Consult your PBX documentation on how to perform this. Many PBXs can restart Asterisk by running the following commands:
service restart asterisk
sudo /etc/init.d/asterisk restart
- Verify that the firewall rules were set up.