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Is there any way I can create a custom dial string in outbound routes, similar to what exists in Asterisk and FreePBX? EXAMPLE: My end user dials 15149999999 from his registered fusionpbx extension and the custom outbound route makes a SIP call to 12121111111@mydomain.com and passes the...

Can I restrict the user Dashboard to only show calls of that particular user? The system stats he currently sees are for the entire system

I am offering a bounty of $400 to integrate DIDww with the SMS app. If anyone is interested please PM me. All my DIDs are with them and I don't want to use another provider just for SMS.

I installed the fusionpbx SMS app more than a year ago and see that changes were posted a while ago. Do I have to run git clone https://github.com/fusionpbx/fusionpbx-apps to update the SMS app, or will it update automatically when I run the GUI upgrade script?

A DID on our fusionpbx system is forwarded to an external voicemail system. When we enter digits they are malformed at the endpoint and the caller cannot navigate the IVR. It seems the carrier is sending a mix of rfc2833 info and/or inband and the voicemail IVR system appears to be listening...

This is a major issue for us. Please help if you can. Thanks in advance. We have a toll-free DID on our system that is forwarded to an external local number which is a company PBX system. When people call, they are prompted to press 1 for English and 2 for French etc... but any digits entered...

When I try to upgrade my FusionPBX Debian 11 installation, it says The following packages have been kept back: postgresql-client-common postgresql-common 0 upgraded, 0 newly installed, 0 to remove and 2 not upgraded. apt list --upgradable shows I am running version 241 and could upgrade to 248...

"destination_country_code is required" https://github.com/fusionpbx/fusionpbx/commit/26b0988075419b2fc0a9daff3c71af8ff73a76de so 2 days ago an update to DESTINATIONS was committed to GitHub that made the previously optional country code field mandatory. Never mind that it killed my whole...

Recently I noticed that my fusionpbx GUI got a lot slower. I added about 70 extensions and now it takes more than 2 minutes for the "Advanced / Default Settings" page to load. I constantly get the popup "This page isn't responding, Wait or Exit page". If I wait long enough the page loads ok but...

My fax server can receive faxes but it can't SEND faxes. Everytime I send a fax it fails and the FreeSWITCH log says: [WARNING] mod_dptools.c:4401 Can't find user [18884732963@mypbx.com] [NOTICE] switch_ivr_originate.c:3039 Cannot create outgoing channel of type [user] cause...

I have one extension that simply does not want to register. The username and passcode are correct and I can see the auth challenge in the log but it does not tell me why the registration is failing. I already turned off or restarted fail2ban, iptables and freeswitch just to see if anything...

Is there any way to do a 3-way conference in fusionpbx or at least to simulate that functionality? I need a simple method similar to 3-way calling from a POTS line. Thanks. e.g. call to person A, some * code to put him on hold call to person B some * code to connect the 3 callers

I am looking for the variable "vm-disk-quota" that is referenced in the fusionpbx docs Voicemail Options It's supposed to be in Dialplan -> global-variables but I can't find it. Do I have to add it?

Any reason why this defaults to FALSE? Does fusionpbx not support UDP transport unless this is set to TRUE?

I am switching from FreePBX to FusionPBX and we have a ton of Grandstream HT286 among our end users that still work perfectly fine with Asterisk. However, I can not get them to register with Fusionpbx. I entering the SIP credentials manually and I can see the SIP auth challenge in the log but...

Does anyone know how to provision a SPA 2102 from Fusionpbx? I followed the instructions and added the profile rule below and rebooted the SPA but it did not provision. [--uid provision-user --pwd provision-password]http://mydomain.com/app/provision/?mac=$MA Typically I use the syntax below to...

I am trying to provision a Grandstream HT/701. After setting everything up on the fusionpbx side, I opened the Grandstream GUI and entered the config server path, http user name and password. If I set provisioning to HTTP, it works. HTTPS does not work. We use Let's Encrypt for SSL. Is there...

I installed Let's Encrypt with the dehydrated script and was under the impression that it would somehow auto-renew but I always have to renew manually. Is there a way to set it up so that it renews the SSL certificate automatically without any intervention? When I do it manually, I always have...

Anyone know how to setup an Anveo Gateway? The host is sbc.anveo.com Do I have to allow additional signaling IPs in the ACL list? In FreePBX everything works great but same settings in Fusionpbx fail. I keep getting CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]

My PBX has 11-digit Destinations (Inbound Routes) and 10-digit Extensions. If extension 1000 dials extension 5141234567 then the call is processed as an internal transfer. However, if extension 1000 dials Destination 15141234567 then the call is sent to an Outbound route and processed via an...