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Is there is global way of setting up the voicemail feature so that any caller getting an extensions voicemail can press '0' to go to either operator or ivr? The only way I know how is by editing each users voicemail options.

Hey everyone, I migrated my fusionpbx server from debian 9 to 11 without any problems except for this. My fanvil phones won't SIP connect via TLS. I'm definitely not an expert when it comes to certs but I believe debian is trying to negotiate TLS1.3. I've changed freeswitch variables to...

I have a fusionpbx server 5.0.1 with a Fanvil X6U (firmware 2.4.5) On Fusionpbx Server Default dial plan valet_park (has not been modified) On Fanvil I have Caller ID Header: <FROM> DSSKEY set for Memory Key -> Value=park+*5901 -> Type= Call Park Call park works with no issues. Unparking the...

Does anyone have any documentation regarding the cloud phonebook feature on the X5U? I would like to try and get fusion to send directory information to the handset. Any help appreciated.

Hey everyone, hope some will be kind enough to give me some guidance in setting up a call center for this particular clients situation. It's a real estate office that has 5 remote branch offices that connect to a hosted Fusionpbx 4.4.5 Each office has their own direct DID. They would like to...

Currently the when a user receives a call, the phone rings and the caller will pick up the call. While the user is on the phone, a second call will come in. Instead of the phone ringing a beep tone will be heard from the handset. Is there any way of getting the phone to ring just like the...

I had a fusionpbx set up one time where in the operator panel you could transfer a call to a extensions name instead of their number. I don't know where or how this was set up but I can't seem to replicate it. Can someone help me please as it doesn't seem to work?

And what do I need to do? Trying to set up interop. We have added termination as well. Please use prefixes when sending traffic: 999001 A-Z/guaranteed termination 999002 On-Net/best effort termination

I'm pretty new at this so forgive the newb question. I've set up with a sip provider ip based authentication. I've got incoming working (allowed the providers ip in acl). Now how do I set up outgoing? Do I create a gateway with no authentication? Any help really appreciated.

Let me know if I'm thinking this out wrong but I had a problem with a particular sip provider. They had some issues with their services and consequently some of my clients couldn't dial out or take incoming calls. It was brief but got me thinking. Is there a way to set up a failover gateway so...

When trying to check why my phones weren't provisioning, I went to http://mydomain.com/app/provision/{mac}.cfg and get this error. Not sure where to look.

I didn't want to put these providers all in one thread so I'm asking separately (for future users wanting to search). Has anyone tried bandwidth.com? Are they good? Bad? Do you find their rates competitive? Any insight appreciated.

Does anyone have any experience with DIDWW? Are the good to deal with? Are their rates competitive? Any help appreciated.

I wanted to get anyone/everyone opinion on different voip providers. Would it be okay to mention their names and to possibly get some comments/suggestions/reviews?

I've ready the minimal documentation on this topic and can't seem to get my custom wav file to play. I added a GLOBAL music group called 'custom'. Added an 8khz wav file. Changed in variables hold_music=local_stream://custom Doesn't seem to change anything as the default music still plays.

I've created a user called OperatorPanel and assigned it the rights for both Apps and Call Detail Record in the menu manager. But the user gets 'Access Denied' when trying to select the option form their menu. Are there other rights that I need to assign?

Is there a quick users guide for FusionPBX. I've set up an office and they were asking for a quick tutorial on how to use the system so I will set out to create a quick manual but wanted to check to see if there wasn't one already. Any help appreciated.

I'm fairly new to fusion so I need some guidance with a particular office I just set up. Let me give you some background info: Real Estate office that has two receptionists. The office is using yealink handsets. They may get four or five calls come in at once and if one receptionist is on the...

I noticed that one of the parameters in the default settings 'line_sip_transport' is set for TCP. I was under the assumption that UDP is a better protocol for voip. Should this not be set to UDP? Please correct me if I'm wrong.

I have fusionpbx hosted in the cloud. If the remote office loses internet, is there a way of getting fusion to forward calls to another number?