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New install with updates on Debian 13. It was working OK with one number. Direct Dial is disabled, instead I put the extension number as an option. The Destination Action sends it to the IVR. The user enters the extension number. I want to replace the number with a different number. I changed...

I am setting up the IVR menu for direct dial. One extension retry is allowed and then I want it to hangup. But after multiple wrong extension number entries it replays the main long message. Greet Long > Recording > please_enter_the_extension Timeout 3000 Exit Action Hangup Direct Dial Enabled...

I cannot set or clear any permission in Group Manager heading XML CDR. It is the same for anything under anything about Voice Mail. Saving resets it to how it was. I can successfully change permissions for Access Controls and other heading items at the top of the list. Everything under...

Latest install of FusionPBX with FreeSWITCH v1.11.0 on new AWS EC2 instance of Debian 13. If I try to save anything in Advanced > Variables it gives: Fatal error: Uncaught TypeError: fwrite(): Argument #1 ($stream) must be of type resource, bool given in...

My system was working then I rebooted it. Now calling in to an extension it immediately goes to callcentric voicemail. The fs_cli log has this line: Dialplan: sofia/external/CALLER_NUMBER@sip.callcentric.net Action log(WARNING [inbound routes] 404 not found ${sip_network_ip} destination_number...

All incoming calls are getting routed directly to the first extension. Incoming routes are bypassed. In Destinations -> Destination I have tried the Callcentric trunk number, the DID number (11 digits). Incoming route is bypassed. Even if I put a nonsense number in Destinations -> Destination...

I have 2 Callcentric DID numbers, associated with two extensions. With Callcentric there is the primary extension 100 and a secondary extensions. On FusionPBX I have setup the two extensions and the two destinations (inbound). When I call the second DID number it should go to the second...

I set up two gateways for two sip providers. One sip provider is UDP and the other is TLS/SRTP. The TLS/SRTP gateway worked first. For the UDP gateway, I copied the external sip profile set up for TLS/SRTP, and I disabled the rtp-secure-media and media-encryption in it. I assigned each gateway...