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CPU free. I guess the FreeSWITCH people were sick of getting fault reports about people maxing out their hardware and complaining about lost calls? It makes no sense to me.

It only works with Google Firebase and Apple Push Service out of the box, and neither of those (if I recall) have the ability to wake up a random browser page and bring it to the foreground. You need to hook it into this service that triggers push notifications as you mentioned above. To...

If you haven't done a lot of crazy custom stuff, there's a schema upgrade tool in the upgrade area. This checks the DB schema and updates it. As always with server admin jobs, check your backups and ensure you can restore if anything goes wrong before doing this. As stated, this probably won't...

Have you triggered the Event Guard in your testing? Check it out and see if your IP address got blocked. Other than that, you can use 'sngrep' from the terminal to see if the SIP messages are making it to the PBX. You can also view the log files in /var/log/freeswitch/freeswitch.log or the...

[ERROR: column "call_block_direction" does not exist LINE 4: and call_block_direction = 'inbound' Your table is missing an entire column. I wonder if you missed an update schema some time ago. It might not affect functionality for this fault, but it would annoy the hell out of me. Also, your...

I don't think it's related at all. You have a big fat SQL error. What's happened to that table? It's either that or Fail2Ban/Event Guard is blocking due to an invalid REGISTER: 2023-03-28 14:08:07.377805 [WARNING] sofia_reg.c:1739 SIP auth failure (REGISTER) on sofia profile 'internal' for...

You probably want to look at something like Microsoft Bookings. https://learn.microsoft.com/en-us/microsoft-365/bookings/bookings-overview?view=o365-worldwide

SIP.js, is a SIP client, that triggers a WebRTC session for the media stream and sends the SIP over a websocket instead of a traditional network socket due to browser sandboxing limitations. It's a similar concept, only you would need to develop a proxy that speaks websocket to the SIP.js...

Yes, I think the only open source implementation there is, is Flexisip: https://www.linphone.org/technical-corner/flexisip

Yes.... by implementing an always online proxy service that converts SIP messages to Apple/Google push notifications and then proxies SIP messages through to the device. 3CX uses bespoke tunnelling to create a point-to-point VPN between app and server, historically you used to have to import a...

Registered to what? It's the client that registers to the server. The client sends information about contact port, source address, expiry time, etc. If no client is registered, where can FreeSWITCH send the SIP INVITE to?

I refer you back to the link for how one company does it: https://doc.acrobits.net/sipis/incoming_calls.html You need a service that proxies SIP to an endpoint after waking it using the push service. SIP alone can't do this, other than just forcing the app to run 24/7 wasting battery.

You mean when both the service and the client are registered at the same time? In the local extension dialplan rule, ensure that fail_on_single_reject=false Your intermediary service will proxy the SIP messages to the endpoint if it's not present. If it is present, then the call will be forked...

You will have to architect it as a service, but ultimately something will still have to be listening 24/7. Commonly how SIP phone developers do it is by creating their own service which is hosted on the internet which registers against the PBX, and then when a call is received, it uses the...

That looks like a normal off hook call, as in you weren't in the middle of a call? It's a bound digit action, so it shouldn't need to go through the dialplan. 1. Pick up call. 2. Without pressing hold or anything, dial *4 3. When you hear silence, type 101 4. Wait until it connects, speak to...

I still don't think you are sending them differently. It's trying to compare *4101 instead of *4, then processing 101 via DTMF. Press *4, send, then press 101. As for SIP INFO, you can enable liberal-dtmf in the relevant Sofia profile, I'm guessing internal...

Classic. ISPs, Router Manufacturers and Firewall manufacturers all to often think they need to 'fix' SIP by messing with 5060 UDP packets without realising phone systems fixed it years ago.

FreeSwitch can't speak Teams, so I wouldn't try too hard. Kamailio can, I think OpenSIPS can too, and DSIPRouter. So a lot of people slap these in front of FusionPBX to act as their SBC. As for the hardware, yeh. It's a shame. I don't think any community ever really kicked off. Something like...

What is it saying is happening in the FreeSwitch logs when you do this?

Other than the NDLB settings for FreeSwitch. You might be looking at one of those scenarios where you have to deploy a VPN on the FusionPBX server and creating a SIP profile for VPN connecting phones. No experience with Grandstream phones here, but OpenVPN has saved our bacon with Yealinks when...