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Yes

No, don't do it, You DON'T want to overwrite the caller ID. I'm asking questions as I don't know how you've configured your PBX. Sometimes people put their DDI in to the below field and then don't realise they are overwriting the caller ID

Applications -> Call Flow Create a call flow to point at the ring group and the alternate to a dummy extension that you can set a divert on. Then just toggle the call flow with the feature code you set.

Dialplan -> Destinations

Did you overwrite the caller ID in the destination?

Check the logs in the usual places: Journal: # journalctl Website: # cat /var/log/nginx/access.log # cat /var/log/nginx/error.log # cat /var/log/php7.4-fpm.log

It's an inherent problem with SIP and Mobile Phones. SIP = Requires constant registration, continually updating port and IP addresses to contact. Mobile Phones = Go to sleep to preserve battery, can go for hours without enabling the network card. To fix this, you need a service hosted on the...

I can recommend Acrobits. White labelling all through a web based editor, can enable and disable features that you want to distribute. Will even build and publish the apps to Apple and Google store fronts.

The author of this thread and the error messages: 2023-04-08 19:41:31.011432 97.20% [DEBUG] sofia_glue.c:3422 IP 111.111.111.83 is on local network, not seting NAT mode. 2023-04-08 19:41:31.311356 97.20% [DEBUG] sofia_glue.c:3422 IP 111.111.111.84 is on local network, not seting NAT mode.

No, that would require some heavy modification to call_forward.lua to include ring_groups. What you need to do is use a Call Flow to allow users to toggle a call flow at the desired times.

rfc1918.auto - RFC1918 Space nat.auto - RFC1918 Excluding your local lan. localnet.auto - ACL for your local lan. loopback.auto - ACL for your local lan. Your external profile probably has localnet.auto in for the local-network-acl, and from your subnetting design that is...

Yeh *720800111111, or just enable (*72), disable(*73) or toggle(*74). You can modify the request_id=false line to set it to true, so anyone as long as you have the voicemail pin can enable a call forward.

Think you might also want: From User: <Your Username> From Domain: xxxxxx.phonestar.ch I think that's why it's defaulting to 'FreeSwitch'

You're usually looking for 'VAD' - Voice Activity Detection. Personally, I turn it off where possible. It usually results in clipping of voices at the start and end of sentences. You can force things with Comfort Noise Generation sometimes...

Autofill how? One up from the last one you defined?

First things first, remove the domain name from the ACL. You rarely ever want that there. Only the CIDR IP Addresses of the providers SBCs for IP Authentication. I don't think you want username authentication, that's what it looks like your trunk is doing with the username of 'FreeSWITCH' and...

Usually now is about time to look at: 1. The logs in /var/log/freeswitch/freeswitch.log, can you see the INVITE attempts? 2. Running sngrep from a root terminal, can you see the SIP invites, is FusionPBX responding? If you can't see either, then you know it's a network issue still. Unlikely...

Only in the interests of keeping this forum the premier location for FusionPBX knowledge! First few nights the support desk were not logging out of their phones like instructed so it got very confusing seeing yesterdays calls on the wallboard at 8am!

Think you'll want this on your outbound route: https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Dialplan/Variables-Archive/x__DTMF-Variables_16352562/#dtmf_type <action application="set" data="dtmf_type=rfc2833"/> The FreeSwitch documentation is a bit shitty for DTMF, so I'm...

Add the user to the ring group. As for forwarding via star code *72,*73,*74 are call forward enable, call forward disable and call forward toggle respectively.