What would you like to see added in FreePBX 15?

my 2 cents:

Observation field in the outboundroutes to write aditional info
https://issues.freepbx.org/browse/FREEPBX-12820

Be able to route anonymoud inbound call’s to DID’s
https://issues.freepbx.org/browse/FREEPBX-8516

Beeing able to set an inbound route to a ring group, that has external numbers in the format #999555111
so that the numbers are call’s simultaneasly

get rid of the requirement for some modules/features (few) that are stick to redhat/centos based distros, I use debian but this should be generic.

We have a anonymous inbound route, it doesn’t work for you? Set the CID to anonymous and point to whichever destination you want.

We have this working on Asterisk 11, I’m not sure about higher versions but i assume it should work as well.
Inbound route > Ring Group:
12124442222#
12086668855#
19057572211#
17779993366#

Works perfectly.

Here goes:

  1. One of the most requested features that I have been asked for by users is the ability to have a Directory/Phonebook on their handset. I realise that different manufacturers use different formats of handset phonebooks but it would be good to be able to construct a phone book of internal extensions and external contacts. I know the contact manager goes some way to doing this and that Sangoma handsets already have this feature via the Contacts PhoneApp but this is so cumbersome to use that I don’t know anyone who can be bothered to use it.
  2. A template feature to build templates for extensions, queues, ring groups, etc. So when adding an extension or other entity you could just simply use a pre-configured template.
  3. I would second what some others have said about the new UCP. Some great enhancements but I’ve found that first time uses are confused by it and it would be good to have a “standard” template that could be built for new UCP users.
  4. More for Zulu that for FreePBX but include synchronisation between Outlook calendar events and presence.

There are probably more but I will post them as I think of them later.

Lee

This already exists in freepbx 14

I’ve never noticed that. Is it a function of Zulu? is there any documentation?

A trunk balancing module, like the one linked here would be an awesome feature to add to the FreePBX distro!

Thanks!

The Contacts App does not go ‘some way to doing this’, it does exactly that. I find the app perfectly fine for a small number of contacts, but I agree that the UI is not great for large directories. I honestly don’t see any way of improving it. If you have suggestions for improving usability, please open feature requests.

Check our Freeobx’s calendar module. It has some new calendar synchronization features.

Hard to understand why there is a concern about http vs https for public information such as code distributions. https doesn’t increase code security. Hash verification might. https for your logins and personal data, yes, and definitely for administering your PBX. (it should be a default for the admin and ucp sites) Code download and public wikis? ehh…

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It’s not just about confidentiality (though that’s a nice benefit - see previous update pinning example!) it’s also about the integrity of the data that comes through the pipe (i.e. you get what you requested).

Sure, one could presumably check hashes from a secure site. Do they? I wonder what the ratio of binary downloads to hash downloads is for those who provide both.

Why not help provide both confidentiality and integrity by default while requiring no extra steps of you consumers instead of hoping they won’t run a binary from non-secure http without taking steps to ensure it’s genuine? It’s easy enough, costs nothing other than time, and quite cheap computationally.

Thanks for the explanation. I appreciate that.

Any thoughts on enabling wildcard certs in SIP? I know it’s a weird thing, but I have heard definitively from Twilio that they will not stop using wildcard certs despite what the RFC says. Since their service seems to be fairly popular, it would be great if we could still connect securely to it.

Maybe it could be a third option in the TLS/SSL settings for PJSIP: Yes, No, Yes (Flexible)

That’s not something we can control. You’d have to ask Digium.

The options asterisk allows here are “Yes” and “No”.

Understood. I wasn’t sure if you made any tweaks to asterisk when you compile it for use in the distro or if you try to leave the source alone. I will see about opening a feature request with Digium. Thanks again.

There are only two things we compile into Asterisk. Lazy Members patch for app_queue and res_blf_mwi. We try not to patch Asterisk as much as possible so that our bug reporting to Digium is a clean as possible.

Gotcha. That makes sense. Thanks.

It’s an open source project, anyone can contribute patches to add functionality. It doesn’t have to be Digium. There are many non-Digium contributors to the project - have a look over at https://gerrit.asterisk.org to see some of the recent patch submissions and code merges. :slight_smile:

Sure. But you don’t accept 100% of all patches everyone ever submits to you. No project does. I also am not a representative of Digium in any capacity so I will always forward people to the right place.

I stand by my “ask Digium” statement. But I will expand it “ask Digium [if they do accept wildcard certs and/or if they would be willing to accept a patch for such a feature] [in their forums]”.

I would like an easy way to disable an extension. There are times when I want to turn a specific extension off so the user can not make/receive calls (ie: they don’t pay their bill, etc…). Its needs to be an easy enable/disable setting in “application->extensions” and endpoint “manager->extension mapping” just like you can turn a trunk on/off.

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+1 for this. (I think i actually mentioned this here, but the more the merrier… :wink: )

I would like as way to restrict who can call an extension. For example the only person who should phone the CEO extension is their personal assistant.

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