Where ever possible, it would be useful to have test options not unlike the feature in CID Superfecta. So as an example, Sys Admin/Email Settings could have a button to test. Also, call routing settings modules could allow entry of numbers and display the resulting number format and route, and so on.
Some community members not associated with Sangoma discourage use but Sangoma does not anymore.
Zulu 3 uses pjsip moving forward. That’s how committed we are to it.
2FA for admin access.
Speech recognition. Just tell asterisk who you want to call.
Poly Voice to Text message email integration.
Maybe Watson too. But I don’t like the way Watson types out a phone number. It spells it out. Not good for a cell phone redial.
Watson has meta options. I have played with a few of the api’s it is really cool stuff
This would be wild difficult to setup, but I would love to see a visual display of a dial plan.
–> Trunk In --> Call Flow (with audio bindings) --> Time Conditions --> IVR (with all breakouts) --> Ring Groups / Queues (with all failovers).
I keep a manual flow chart for each customer. Would love to see this as a dashboard module to see exactly how a call would flow. Visual debug tool.
Trunk in to a destination doesn’t equate in freepbx since everything matches through inbound routes
Everything else is surprisingly easy.
I would love to see options to automatically delete call recordings and CDR records after a specified amount of time.
Also an option to force HTTPS for the web interface would be nice.
4000 minutes free, Then stop any CO calls
I think this is what you are saying, but the integrated ability to redirect from http to https would be great.
I know iv’e already posted a few suggestions, but i have more…
I want to talk a minute about migration:
I appreciate the efforts the team is putting in now to make restoring easier on FreePBX 15.
But I would also appreciate if you guys can come up with some tools to help migrating from non Asterisk Servers to FreePBX.
We’ve migrated many clients from Panasonic, Grandstream, Toshiba and other PBX servers to FreePBX, most of them were not systems crucial for use 24/7 so we were able to do the migration after hours or on a weekend day.
As mentioned in my first post, we used bulk handler for bigger systems which made the process a lot easier, and hopefully this module will get some attention by the developers and contributors.
What got us tied was, the critical systems that couldn’t afford any downtime so we did the following:
We created all extensions on the FreePBX let’s say 1xx and 2xx with a follow me that added a 5, so Extension 101 had a follow me of 5101, 107 had 5107 and so on. We had a Adtran in place that had Trunks to the FreePBX and old server, we configured that any 5xxx call should dial the Adtran which sent the call to the old PBX which called the appropriate 3 digit extension and vice versa so if you dialed 5101 on the old server you reached 101 on FreePBX (I don’t remember exactly what we did, but i think we stripped the 5 on the Adtran so we didn’t have to create a inbound route for each extension when the call was received on each PBX. or maybe we only had to create inbound routes on FreePBX? i can’t recall…)
Now, once we had a new phone connected to the FreePBX, we disabled follow me on FreePBX and enabled 5xxx follow me on the old Server which as explained called the extension on FreePBX, (we had to disable the follow me on all live extensions as soon as it was online, or at least once we created follow me on the old server, otherwise it would be a follow me loop…) this is basically how it was done for almost 200 phones: Replace Phone, Disable follow me on FreePBX, Create follow me on old server, repeat… No. i wouldn’t go back to that day
So… i believe that there is some way to make migration easier, it’s not always about cool features, sometimes it’s just being able to switch to FreePBX.
Taking that in mind, most of the time when migrating, some people want new extensions. It would be great having an option that allows to play “Please note that extension 112 has been changed to 305” and either return to the main ivr or call that party. When you have 5-20 Extensions you can do this with announcements, but it’s a no go when dealing with triple digits. (We’ve accomplished this partially with custom code, and I will be more than happy if I can help by providing this to FreePBX)
Thanks for looking at us migration workers…
Now back to feature requests…
I recently noticed this button which allows to override the softkeys for each phone, would be nice to have these for the Sangoma phones. (Maybe there is already? i couldn’t fine it)
Routing based on CID
Was posted many times here and some managed to make it work with some code, but it would be nice to see a supported module which allows you to specify a destination on calls that matches this CID name or number, or if it contains this name… we’ve done this.
Have the option to use a time group (not just calendar)
Would be nice to have a virtual extension feature that has just 3 options. Extension number, [External] number to call, Call Confirm Yes/No. (you can technically do this by using Misc Applications and Misc Destinations, but it would be a lot easier to have these under your extension list, which will also allow to add to Queues etc)
We have had many times that a client asked to disable an extension from an employee that had left the company.
Sometimes we just delete the extension, wipe the phone and no one can use this phone
But what happens to the voicemails? Or what happens if the manager wants this phone back online again NOW?..
Which got me thinking, why isn’t there just a simple disable box that will virtually kill this extension, no calls from or to this extension, internally or externally. This will give us another level of management.
Honestly, i didn’t play much with this yet. But would love to see such an idea for schools and collage campuses with some web portal for students to pay account balance, listen to voicemail etc.
We NEED a supported panel that works and is supported by FreePBX. (we are tired of portals that literally eat up your memory, or aren’t well supported)
I have more, but not more time now…
A GUI for configuring Asterisk 15’s SFU. Video added to the UCP softphone that can optionally use the SFU for multi-user conferencing.
This is a trivial one-liner in the apache config. That said, it should also be trivial to add as a check-box in the GUI.
I don’t agree with this statement. Cisco and Avaya are doing 1 jump a year.
yes, it is a very good feature. Asterisk-15 with SFU is a must:grinning:
Isn’t one of the main ideas of FreePBX so you can manage everything from a nice interface?
This is probably more of an asterisk thing, but I would love an option to accept wildcard certs from my SIP trunk provider. I know they are explicitly prohibited in the RFC, but the certs are valid and the alternative is to simply turn off host validation (severely crippling any protection secure trunking offers you).
Also, I would be interested to know why GraphQL was chosen over a more traditional REST API. I know it’s the new hotness, but I image you would get more developer buy in and extensions written on a more widely used platform.
Thanks, as always, for the fantastic software.
We decided on graphql not because it’s the new hotness but because of its flexibility to change and morph. If we went with a traditional rest api we would have to figure out versioning. Freepbx has over 127 modules. It would be a nightmare. Graphql allows the api to be flexible when each module changes or adds new functionality. Each module can also link into to other modules.
I think if you saw a demo of how it worked yoy would be impressed by it over restapi
Also users have been able to write extensions/modules since version 2.1. Long before even rest apis were popular and we haven’t seen an influx of modules.