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Hmm guess noone has this issue except me

Hi guys, so i have noticed on our cloud setup that when a user answers a call and then transfers that call the caller id of the inbound caller is not passed through, also if you transfer to an extension and it goes to voicemail the message info is from the person who transferred the call not the...

I don't think that's possible to be honest, and even if it was would be problematic as each fusionpbx instance is already multi tenant. I would just spin up multiple nodes and use domains as your way in, for example we have a cloud setup using fusion, each tenant has their own sub-domain and we...

As far as im aware freeswitch doesnt play well with windows at all. Im my opinion of fitting coms systems for 25 years i would never install any phone system even 3CX on a windoze box. Linux is native and much more stable.

That sounds like one of two things to me, either you have a permissions issue on the directory that the vm files are stored in or you have a broken install. I guess the first thing to ask is what OS you are using and how did you install it ? i found the easiest way is debian 8 and use the auto...

So if i set this under a key it wouldnt light the key up ?

As Daz says i would find this annoying and a bit pointless, however if you are simply transferring to a ring group just use the ring group greeting , i think that plays before the group actually rings..

How do you then ass this under a key though ? is the value entered under key the same ?

Fusion/Freeswitch blows the pants of asterisk in my opinion, however regarding this issue i would suggest as above you have a firewall issue, also did you set the username and password in the provisioning section correctly ?

Hi, your question has a multitude of possible answers, check the following: 1: Check your providers IP address is listed in the ACL section. 2: How are you connecting to your provider ? user/pass or IP auth ? 3: When you try in inbound call do you see anything at all in fs_cli 4: Does your...

That did it, thansk alot.

Hi Guys, has anyone managed to get *2 on demand call recording to work on fusionpbx ? i have set the bind_digit_action to true, i then made a call from my extension and when answered dialed *2 , all i heard was the tones and no recording was done, how does this work please Thanks k

That would be great, if it helps it doesn't seem to create the tables in the database properly however the page does work.

This doesnt appear to work any more, i can get it installed however when i add a number it doesnt add it, nothing appears on the page at all, no link no add, also nothing in the sql table.

^+(\d*)$ then in prefix set that as 00

Im pretty sure you coukd just do an outbound route and set the custom digit to + then in the same route tell ot to add 00. This means it will use + as tge dial oht code which it then deletes and adds in the 00. Works on mine i use it for locals as well.

set the codec on the phones, that enable you to limit as tpo where the phones are. you could also set it in the phones provision profile under settings if you know the setting on teh phone.

Hi rusguns, i wanted this fuction as well , i dont think its possible to achieve in fusionpbx, however what i have done is run my sip trunks through ASTPP, we use this for multitennat billing but it also ive the ability to limit inbound calls on a sip trunk. hope this helps, you just register...

Thankyou ...

Thanks alot i will give that a try. Do i need to enter *2 anywhere or just enable bind_digit_action ?