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What does the 200 OK (SDP) show for the second image? It almost looks like it's replied back with codecs that don't intersect. In the Freeswitch logs at /var/log/freeswitch.log this should show as a mismatch during the 'Audio Codec Compare' stage.

Nothing FreeSwitch about it. It's FusionPBXs provisioning app. You have to replace the variables with your own. No point me telling you my domain name or http_auth_password.

If you run an sngrep from the server, is it the client that is rejecting the call or the server? It might not be the codec, it might be SRTP or something similar.

I think you should be able to set: ignore_display_updates=true In your outbound routes https://developer.signalwire.com/freeswitch/Channel-Variables-Catalog/ignore_display_updates_16353145/

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...