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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...

Not entirely sure, out of the box I don't get the same issue. Might be a question for the Signalwire FreeSwitch forums.

What's the conference profile look like? I'm guessing it's the default one. I've seen this occasionally when MOH is changed, but the rate isn't the same as the MOH

That is for CDR purposes. It is a CALLEE entry and not the CALLER. The $1 means the callee_id_number is set to whatever you dialled without the plus.

Ok. What is set on the ring group?

I don't know where else you would be able to change it if the flow is: Destination -> Extension -> Registered Softphone Unless you pointed it at something other than an extension. You can overwrite the caller ID on a Ring Group.

Remove 'Main Number' from the Caller ID Name

https://docs.fusionpbx.com/en/latest/applications/call_flows.html Does this help? I don't know how to simplify it further.

As long as it's not the caller ID number field.

This is inbound. Outbound calls are set on the extension or outbound routes.