We have our inbound providers route calls to our PBX using SIP URIs configured on the Inbound Routes page of our PBX. We have found this more reliable and easier to configure than using SIP registrations for inbound calls. Unfortunately, SIP dialers tend to run across our system and attempt to dial into our PBX using random numbers/URIs filling our logs with “junk.”
Is there a way to configure FreePBX so that the “Allow Anonymous Inbound SIP Calls” feature under “Asterisk SIP Settings” only allows anonymous calls from certain IP addresses (e.g. our inbound calling provider(s)). Alternatively, could we implement FreeSWITCH-like configuration where we only allow “Allow Anonymous Inbound SIP Calls” be processed on a certain port (e.g. 5080)?