Whats the current state of STIR/SHAKEN in freepbx? I see a notice that there is a deadline for June/2022 related to this. I currently use Telnyx who says they sign all my calls but do I need to get a certificate and do I need to start doing something? I don’t want to wait until calls stop working to get on this.

Is anyone currently using this in a FreePBX environment? What are the trials and tribulations you had tyo go through. I currently have 8 PBX systems with about an even mix between V14 and V15, will I have to upgrade to 16 to support this or will it be added to 14, 15?

Calls need to be signed, and generally that is done by your provider. If your provider tells you that you must sign the calls yourself, that’s when you need to investigate the Stir/Shaken signing process. I’m aware of a few people signing calls directly using FreePBX by means of a third party provider, and there are contributed documents in the FreePBX wiki covering this.

The official response from Telnyx was “we sign every call but this doesn’t relive you of your obligation to FCC regulations…”. Not sure what that means and what I have to do. Can you point to a specific section of WIKI for this? I have read several articles but so far I have not seen anything that said, do this…I contacted TransNexus for a cert but I dont even know what I will have to do with when I get it?

The wiki does have a search option. Just enter STIR/SHAKEN and you will be presented with results.

thanks, I did find a page specifically discussing “TransNexus”

1 Like

I went with 16 and Asterisk 18.9…converted to PJSIP which is mandatory. Working with Sansay, there is a script they provide which hooks a call to a validation server of mine that I setup under VMWare. Since I provide the server, the cost with Sansay is $250 per month…Transnexus is $500 if you provide the server. I hired Thomas Lynch, per Voip Innovations recommendations, to help do the paper work with the FCC and various entities. There is a little learning curve on the Sansay server setup, but Sansay’s docs and support team can get you going pretty quick. Around $3k to get everything done for startup (not including the server which I already had).

Another option if you have several PBX that you host would be to set up a Session Border Controller to handle the STI-AS signatures. (And also Proxy the SIP traffic between your providers and your tenant PBXs)
dSIPRouter is as a frontend to Kamailio, and has a nice interface for TransNexus. You can also “roll your own” with Kamailio or OpenSIPS by themselves.

1 Like

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.