I really don’t understand why people are complaining about documentation.
Please find me something which is poorly documented… The devs here are constantly, constantly updating Wiki’s.
If you are hanging out here a lot, you can probably tell that probably 50% of the “how to” questions are well documented.
Queue Wallboard - This module wasn’t released yet, but there’s already documentation for this at least two months.
These are ASTERISK things you are inquiring about. Those can be found at http://wiki.asterisk.org. I really don’t think Sangoma needs to re-write existing documents that Digium (Asterisk) has already provided. Not to mention that Chan_SIP is as old as Asterisk itself, hell in most cases voip-info.org has all the settings and their meanings in regards to Chan_SIP because it really hasn’t changed much in the last 10+ years.
In addition to this addition, Linux is Linux. Outside of some specific things like “apt-get” vs “yum” and some Distro based commands, Linux commands are the same in each OS. Sangoma hasn’t made any qualms about the FreePBX distro being CentOS/RHEL based OS. So that should give you a good jumping point about what OS documents you should refer to.
I’m not inquiring about anything. I know all about these things.
Asterisk is a critical component of a FreePBX installation. The PEER details is a field contained in the GUI in FREEPBX. You and I know all about the PEER details, and the fact that Asterisk is a separate project from Digium. But, new users don’t. Documentation is how they learn.
And that’s precisely what the documentation should say!!
Sure, anybody can go searching the internet for the answer to a question about CentOS. That’s certainly how I did it when Iearned how to set-up my FreePBX system.
But doing that wouldn’t be necessary if FreePBX released comprehensive documentation. It is true that Linux is Linux. Asterisk is Asterisk. Postfix is Postfix. CentOS is CentOS (or SchmoozeOS in this case). You could say that about every component of FreePBX and every component of Linux and every component of Asterisk.
If knowledge of Linux, or Asterisk, or Postfix, or something else is necessary to setup and use FreePBX, it should be documented. New users shouldn’t have to scour the internet to find the answers.
I agree that we all went through the hard work of learning Linux and learning Asterisk in order to use FreePBX.
But, if FreePBX is going to succeed in an environment where users can download 3CX and run their PBX on a windows server, FreePBX is going to have to provide documentation for all of the components necessary to allow a user to get their system up and running.
Please go find me another Asterisk based PBX project that took all of the existing documents over at the Asterisk Wiki and then made an entirely new wiki for that. You will be hard pressed.
Also, you’re worrying about Chan_SIP. Outside of the fact it is highly documented across the Internet, it’s also dead. Chan_SIP hasn’t been touched by Digium since Aug 2014. It’s 100% community driven now. All current and future development of Asterisk for SIP revolves around Chan_PJSIP. It’s also the default SIP channel driver for FreePBX. Has been for a few years now.
There should be no need for Chan_SIP documentation now or in the future on the FreePBX wiki as it’s not a supported driver. No new features or bug fixes. The best you’re going to see is if a major security hole was found. Digium might patch that.
To be clear, I’m not saying that Sangoma is a 100% on the documentation but the Wiki is open to everyone. So if there are certain documents that you think need fixing, updating or created then you are free to do so anytime you want. Sangoma will happily provide your community account on the Wiki edit/add privileges so you can fix/edit/create documentation to help the FreePBX community.
If you feel so strongly about this and since this is a FOSS based project at its core, you could contribute documentation to the project. It’s what I’ve done in the past.
Actually no, Asterisk comes with two voicemail applications, the one everyone uses is Comedian Mail, which is a monolithic “c language” application that predates Asterisk, it is NOT amenable to any changes outside the documented behavior,
So you want FreePBX to be no better than any other Asterisk-based PBX project? That’s a low bar.
No, I’m not worrying about it at all. I was asked to give an example, and I gave one. If I spent hours combing through the Wiki, I would have listed 30 examples. I don’t have that kind of time.
The fact that Chan_sip isn’t being developed is totally irrelevant. FreePBX supports it, in that you can set-up and use Trunks with Chan_sip… Chan_sip works, and it works well. If it has bugs, I haven’t found any in years.
The reason that Digiium isn’t developing it is because it works and they’ve moved on. As long as Chan_sip is included in FreePBX, it should be documented by FreePBX.
The notion that we should not document any component that is no longer in active development, even if it is supported in FreePBX (meaning that you can use it, make, it work, etc.), is a horribly low bar.
Keep up that attitude and we’ll see FreePBX die a slow and painful death.
Elastix = Dead/Bought by 3CX. Also was a fork of FreePPBX
PBX In a Flash = Dead/Bought by 3CX. Also was a fork of FreePBX
Trixbox = WAAAAY Dead. Also was a fork of FreePBX
IncrediblePBX = Still around-ish. Offshoot of PBX In a Flash. Was forked from FreePBX
I’ve been using FreePBX for almost 8-10 years, it was around before that. This is the longest slow and painful death I’ve ever seen.
Do we need to lock this thread and put you guys in a time out? Please this same thing happened between Martin and Andrew yesterday and now its Martin and Tom who has nothing to do with Sangoma. Maybe Martin you need to take a moment and chill out and stop coming in here to argue with everyone who does not agree with you. Take off your lawyer hat for a bit and quit trying to just argue for the sake of arguing. Your point a view is not the only one and your stance on things is not the only one that matters.