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Dialplan Manager and then edit the two rules.

It's in my comment above: phonebook.data.1.url Youll have to replace the variables manually if you aren't using provisioning.

Well, that's what media_mix_inbound_outbound_codec does. I do it in reverse for OPUS. Did you restart the SIP Profiles?

Sorry, I don't touch Grandstreams. From your picture it looks like the field is right there under XML server path. Check the Grandstreams templates to see if the URL is any different.

Is the extension registered? What does it say in the logs when it happens?

You using PostgreSQL for the core? SQLite on a RAM drive would also be good.

If it's an outbound call, surely it's your clients that sending it like that? Or is it that your carrier is sending a SIP UPDATE with the new destionation number in?

Did you modify the dialplan entries for both to your requirements?

I would only really set them in the Variables: Set: global_codec_prefs = PCMU,PCMA outbound_codec_prefs = G726,PCMU,PCMA media_mix_inbound_outbound_codec = true Internal profile should only then allow PCMU and PCMA, and the external will transcode G726 if it needs to.

It's already in the templates: https://github.com/fusionpbx/fusionpbx/blob/master/resources/templates/provision/yealink/t46u/y000000000108.cfg {if isset($yealink_remote_phonebook_1_url) } remote_phonebook.data.1.url = {$yealink_remote_phonebook_1_url} {else} remote_phonebook.data.1.url =...

Well a Direct Dial In number needs to be routed to a ring group or an extension to forward it if you wish to control it with a star code. You might be able to create a Call Flow to do a similar thing. Extension Forwarding is the *72, *73 and *74.

Historically, the Contacts page address book. You needed to assign a contact to a group or user, and then the device to a user. That can be disabled now however, so it's pretty much on the web interface. We have a few customers that don't utilise the frontend at all.

Never tested it myself, this extension appears to call the click to call API: https://github.com/gizmobin/FusionPBX-SelectToCall Extension is on the Chrome web store.

Depends on the router that network is on. You might find it more reliable to switch the client to TCP, routers handle it better. This is all implying the PBX and the phones are on different networks.

How odd, Debian 11? Is this a fresh, brand new install?

It's not built into FusionPBX, but with some modification to the conference module (Conference Controls/Profile) and a dialplan entry (Dialplan Manager), you can get something similar, thought I don't know how well it would work with a group call, might be fine on Enterprise ring groups with...

*75 = Ring Group Forward. Modify the dialplan entry in the Dialplan Manager to remove/alter the PIN number and set the ring group and destination, or let it prompt for it. *72, *73, and *74 are enable, disable and toggle call forwarding. You can also modify the dialplan entry in the Dialplan...

That's not wrong, SIP clients can use any ports they wish using rport. Otherwise you would only be allowed to use one SIP client per IP address. Does that network have some sort of SIP ALG enabled? If so disable it. Also arbitrary port blocking rules for UDP? If so, whitelist the PBX address.

Had a look myself at the member documentation, doesn't seem to state which level. Probably best giving them a call: https://www.fusionpbx.com/support

This should do it, this is in Dialplan Manager -> user_record, right at the bottom.