Creating this as a megathread for speculating what TangoPBX might be… 
(and what users want it to be)
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Or maybe input from what people would like to see it be???
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Updated the post to reflect that suggestion
Me personally, I think TangoPBX will be a PBX GUI for Asterisk like FreePBX, maybe even built on top of FreePBX…
And if it is, I hope they add a Call Recording Reports module free, because it feels like it shouldn’t be a paid module like Sangoma has. (not hating on Sangoma)
Great kick-off @tpglitch!
Some things that come to my mind:
STATIC SORTING:
The ability to have the sort options options for modules to be static when moved to alphabetization, or better yet the ability to statically custom organize any items created within the modules and more ease for customizing the general layout.
UNDO BUTTON:
Prior to any DB reload/apply config an undo button.
EASY TO READ RECENT WORKLOG:
Easily know who/what/when changes were made.
USER MANAGEMENT/AD SYNC:
Optionally updates or resets changes respective to the phone system side such as Extension name,VM reset, etc.
BRANDING:
I completely support and agree to retaining any and all credit/branding to all party’s that are part of any project but I also believe the ability to add personal logos and certain areas with labeling specific to the business of the phone system would be great, to make it feel like “home”.
VARIABLES KEY/CUSTOM:
An easy to reference key for variables as well as the ability to define own.
END USER FAQ/TUTORIAL:
UCP has the ability to link an end user easily to a FAQ and or Basic Guide custom to the specific PBX.
Just a few that come to mind right now but I know I have more, interested to see what others have thought about.
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Maybe I’m lacking imagination but basically a CIP Distro to replace FreePBX. I don’t have a lot of complaints about FreePBX itself as much as all the chaos around it and the uncertainty.
@tonyclewis, we’ve been working together since 2022 and honestly I’ve never had anything but a great experience with everyone at CIP. A full distro from you guys would be amazing and pull all the anxiety out of my thoughts for the future of working with PBX’s.
I know that’s a tall ask but if we’re posting wish lists….
If that wish were to come true:
I know it’s not an easy task based on the way call data is recoded in asterisk but an easier way to pull the kind of CDR info from the PBX that is catered more towards what end user want to see. Simply who called/answered when and how long the call lasted.
Perhaps a more robust built in admin alert functionality? Automated email/text alerts for:
Endpoints dropping offline
Firewall status change
Trunk status change
Unusually high call traffic
Admin user login
Processor/memory/disk usage alerts
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I made a Python GUI and have a super long (but functional) SQL query to join some tables to give me IVR statistics (our current Alcatel system has one and it’s really handy). Basically shows:
-The calling number
-What IVR / DID they called
-How long they were on the menu
-What option they chose / what extension they were sent to from there
-Whether the caller hung up or the IVR hung up (due to timeout, etc.)
I was really proud of what I did, but it would be cool if there was a built-in module under Reports like “IVR Statistics” or something.
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Being able to change DND and CF for extension directly in the GUI. I always thought it was strange FreePBX showed you the check boxes but offered only UCP as an option to change them.
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A PBX platform with reliable support would be priority 1 for me. It’s scary having so many systems in production with no reliable way of getting support from Sangoma.
As far as features go. I’d love to have an extension template system, or at the very least a way to modify the default extension settings.
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You can sort of manipulate extension settings if you use the bulk export module, update applicable extensions in the spreadsheet, and then do a bulk import.
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Yeah, just a pain to do it that way. I’d like to be able to configure a deployment so it has the settings we want for that particular customer (SRTP, SIP TLS/TCP/UDP, codec, call waiting, voicemail, etc…), then when a new employee is onboarded, we can just run through the quick extension wizard, and all those settings are dialed in. Busting out a spreadsheet every time we want to add a new extension is pretty tedious.
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I set up a system the way I want it then take a FreePBX backup and save it. Then as I deploy a new system, I restore the backup and then make any other customizations required. Since extension numbers vary by customer and DID range, I keep a spreadsheet and modify it as required and then upload it for the extensions. This is my base template for any system.
The other alternative is if the OSS Enpoint manager is working, you can work with templates for your extensions.
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Another idea for a module: After Hours On-Call Group
I have a lot of Dr, Dentists, Vets etc that have call forwarding to a Dr on call after hours. Usually it’s an IVR item to select for issues “requiring a Dr’s immediate attention”
Most small practices have agreements with other Dr’s to cover for these calls at night/weekend/vacations etc. Usually it’s more than one other Dr and they take turns.
While this can be fairly easily done with an IVR and Call Flow, what’s difficult is the human interaction… The users/staff just don’t get the concept or forget to change the call flow etc…
A great module would be one where you could have any number of "contacts’ and a single calendar that you could select the contact and enter the days/hours they are “on call”
Then from an IVR or DID just route to that module after hours.
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I’ve been doing as @kenn10 for years.
Majority of my installs are small (10 or less extensions) and a spreadsheet isn’t needed.
I have a customer with a home office (15 extensions and 3 trunks). Nine remote offices (3 extensions and 1 trunk each). They all have 3 digit inter office dialing using Eugene Blanchard’s iax2 trunk configuration template
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I have several suggestions to improve performance and minimize footprint:
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Replace Apache with NGINX and php-fpm.
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Employ message queues for async updates, perhaps with RabbitMQ?
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Employ a test module that tests for SIP-ALG, proper RSP configuration,etc.
One HUGE repeated request from clients is a configuration that mimics line-key PBX systems.
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+1 for shared call or line appearances but the OSS community has maligned the idea for years. It seems that only proprietary systems have embraced this functionality. We’re told to live with call park buttons instead.
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I am always curious when people start saying they want to mimic a KTS but in reality they want a Hybrid system. Perhaps, even in the end just a PBX.
A true KTS doesn’t have the same functionality of a PBX, it has a very limit set of features. If you had six lines for your KTS, you had phones that had six line buttons active on it. Each of those six lines were associated with a DID. You had no control over the outbound CallerID presented, that was set at the carrier level. If you wanted hunting so if Line 1 was busy an incoming call would roll to Line 2, the carrier had to provide all that for you…at a cost. A KTS basically had calls, hold, transfer and intercom features. All other advanced routing features had to be supplied by the carrier. It wasn’t even until the late 70’s/early 80’s that a KTS had voicemail features or even CallerID support. Company’s used to connect their KTS to a PBX when they needed more advanced internal control of their calls.
When KTS started to incorporate more PBX features, they were called a Hybrid. It basically meant you gave each stations a Call Appearance for each line you wanted them to have. Additionally, picking up Line 3 didn’t mean the call went out Line 3…the system could choose which line to make the outbound call over. It also meant you started having features like “you have to dial 9 to open a line” or IVRs or parking and ring groups.
So in the end you have a KTS, KTS-Hybrid (KTS with some PBX features) and a PBX. The latter two do Call/Line Appearances differently than the first.
This is not correct. Asterisk has supported Shared Line Appearance for ages now. It originally was designed around DAHDi (for trunks) and Chan_SIP (for stations). It even allowed for Chan_SIP for both trunks and stations. Conceptually it should work with Chan_PJSIP but it seems that no one, in the last decade, has found the real need to work out and confirm SLA with Chan_PJSIP. If someone has, they haven’t shared it with the rest of the community in any way.
So all someone who wants SLA in Asterisk needs to do is roll up their sleeves and start banging away at the current SLA framework in Asterisk. It would require work in three core Asterisk configs, extensions.conf, sla.conf and pjsip.conf (or their counterparts in FreePBX).
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A feature I think would be super neat as well is multi-tenant support. If TangoPBX is based on Asterisk + FreePBX, this may be very difficult to do and would be low priority, but it would be a cool addition.
another +1 for shared call or line appearances
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It is true that the Asterisk team had a rudimentary shared line appearance that partially worked. I never saw it work for shared call appearance. It only worked on speific phones and was never really tested out beside a basic alpha concept. It only worked on certain telephones and never really got issues worked out. I inquired about the concept and was told it was “iffy” because they needed to use the Broadsoft SLA function on phones and there was risk of copyright infringement. I don’t know if that was actually the case as I see a number of SIP phones that line keys can be set for SCA (Broadsoft).
Small customers with a few analog lines and a few phones prefer the simplicity of a key system setup but with hybrid features like IVR and voice mail. Large customers who have lived with Avaya, Nortel and Cisco expect shared call appearances in work groups, boss/secretary setups, and other scenarios that Asterisk/FreePBX cannot address in a particularly elegant manner.
Somehow, those proprietary PBX’s figured out how to implement SIP telephones and still maintain shared call appearances.
I’m sure a subsystem that has to keep track of BLA’s, talk paths, which phone wants which line, etc., can probably be CPU intensive on a larger scale but we already have busy lamp on the system.
In any case, there is still demand for such functionality but apparently not much interest in the OSS community. This is one reason why there are still thousands of Avaya Merlin, Partner & Spirit systems, Nortel BCM’s, and tiny Cisco UCM’s out there.
I freely admit I’m not smart enough to try to program anything like it myself.
As pointed out in another thread, that would require the system to be completely overhauled to do such a thing.
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