Use FreePBX as SIP Trunk authenticator + CTS - need solution assistance

Is there a method of using FreePBX to take a call termination service, and split individual DIDs off as standalone SIP Trunks, with the authentication method as a username + password?

To give you some context, we have a large number of customers with plain-simple SIP Trunks. They get given a telephone number, and to authenticate the SIP Trunk on their ATA or telephone system, they use the provided username + password + SIP proxy URL. We provide these 3 parameters to them. They also can log in to a customer portal which allows them to perform basic voicemail setup, forward calls when busy, or forward all calls.

Is there a method of establishing a similar setup to this? I understand this is not the intended purpose of FreePBX, but I’m told the underlying Asterisk software is limitless in its ability to do all things voice and would think surely someone else would be using FreePBX for this purpose.

Any suggestions or feedback is welcome, thanks!

Asterisk is not a SIP Proxy, but rather a back to back user agent. If this is only for calls towards your customers, what they require is a registrar URI; if they also can make calls, they will need a UAS URI, which will, typically include the same domain part, with the user part determined by the called number. Technically they also need an address of record, but in simple cases the user part of that will be the same as the user for authentication, and I don’t think Asterisk checks the domain part.

Unless you are offering other services, my feeling is that FreePBX will be over-engineering, and that a raw Asterisk solution will be better.

Actually, if you don’t have other services, a combined real SIP Proxy and registrar, such as I believe is the case for Kamailio (based on documentation, not experience), may be an even better solution.

Especially if you are handling calls from the customers, you should familiarise yourself with current and pending legislation in your jurisdiction regarding resellers of telephony services. Many US sellers are dropping out because the cost of compliance is unaffordable.

One of the things that makes ITSP setup instructions difficult to interpret, is that they often provide examples, without explaining the protocol roles of the various settings. That is why it is necessary to be clear about the roles of various parameters. These are often degenerate, but that degeneracy should be made explicit.


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