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Not really, just that your phone is receiving the status change packets when a BLF actually changes.

Lol, no problem. Knew you were gonna kick yourself when I replied.

Odd, I would like to suggest something. But it's just a fundamental part of sip_sofia. Lights for call flows, dnd, etc are set in /etc/freeswitch/autoload_configs/lua.conf.xml but just normal BLF? I would recommend a packet capture either from the PBX using sngrep, or from the Yealink phones...

Ensure you have put the public IP address you get from Amazon in the sip and rtp address section in the variables. Otherwise it will bind to the internal address and Amazon won't use an ALG to fix it.

Enable those agent_authorized lines.

https://www.fusionpbx.com/support Probably best to speak to FusionPBX themselves.

We run the whole gamut of Yealink's in the 4 series, and have never had to do anything particularly special to get call park lights working. We either: - BLF = park+*5901 - Call Park = park+*5901 (Some of the older lads use BLF for EVERYTHING...) It is worth noting the the phones with dual...

destination_number(fo359937ip279332_00) is the clue in the log file here. Look for this default setting, you probably want the ${sip_to_user} option. Fairplay to Plusnet for filling RPID full of nonsense though! Real helpful from them. :rolleyes: Once you've changed this, you'll have to edit...

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.