- From the top menu click Settings
- From the drop down click Asterisk Sip Settings
Allow Anonymous inbound SIP Calls
Allowing Inbound Anonymous SIP calls means that you will allow any call coming in from an unknown IP source to be directed to the 'from-pstn' side of your dialplan. This is where inbound calls come in. Although FreePBX severely restricts access to the internal dialplan, allowing Anonymous SIP calls does introduce additional security risks. If you allow SIP URI dialling to your PBX or use services like ENUM, you will be required to set this to Yes for Inbound traffic to work. This is NOT an Asterisk sip.conf setting, it is used in the dialplan in conjunction with the Default Context. If that context is changed to something custom, this setting may be rendered useless as well as if 'Allow SIP Guests' is set to no.
This should display your externally public facing IP address. You can select "Detect Network Settings" to have the PBX detect its External and Local networks, adjust accordingly if neccessary.
Local network settings in the form of “ip/mask” such as, “192.168.1.0/255.255.255.0.” For networks with more than one LAN subnet, such as VPN network, us the “Add Local Network” button to add more fields. Blank fields will be removed upon submitting.
Check the desired codecs and drag to reorder. All others will be disabled unless explicitly enabled in a device or trunk configuration. Note that some codecs, such as g729, require commercial licensing.
Click Submit to save