In Twilio’s documentation (found here) for setting up SIP connections, they mention sending SIP Options messages from my PBX to the Twilio SIP Trunk. Here is what is says:
Optionally set-up your Communications Infrastructure to issue SIP OPTIONS messages as a ping mechanism to your Elastic SIP Trunk (Send the Message Request To: Termination URI you created (
example.pstn.twilio.com)); the Twilio platform will respond appropriately. Please maintain the Ping lower than 1 SIP OPTIONS every 10-15 seconds to avoid your requests from being banned by our Platform.
A bit further down in the same page in the “Deploying behing a NAT” (here), it mentions:
If you’re deploying behind a NAT without a Session Border Controller, it’s important to keep open the NAT translation binding.
For Signaling, when using UDP, this may be achieved by periodically sending SIP OPTIONS to Twilio, which will respond with a 200OK.
Where is this set up in FreePBX?
We have intermittent outages that end up resolving after a couple call in/out attempts, and I’m hoping this will help that issue.