Topic: [ANNOUNCE] smeserver-freepbx

[Daniel B.]

Hi everyone.

A new contrib is available: smeserver-freepbx. This contrib will integrate FreePBX on your SME server.
Basically, FreePBX is a very powerfull interface for the telephony server Asterisk. It's also very easy to use.
FreePBX is not compatible with SAIL, it's an alternative.

I've started a wiki page for the installation instructions:
http://wiki.contribs.org/FreePBX

For more documentation on FrePBX, please refere to the project site http://freepbx.org. I'll try to enhance the wiki page with more informations (how to configure some hardware cards, add AGI scripts to receive an email when a call fails etc...) when I have some time. Please, do not hesitate to correct/add things in the wiki.

Cheers

[Normando]

Thank you very much!

[Teviot]

I have tried to install this twice and found that both times it didn't work.

Even the removal script doesn't work


SAIL is the better choice for PBX in my opinion.

It installs first go and is easy to configure

Stick with SAIL

[David Harper]

Let's not diss FreePBX - it's just another option.

I would say that I agree that - for the moment - SAIL might be a better choice for business deployments, because it's commercially supported by a company with deep experience running on SME. Looking at FreePBX, a lot of the under-the-hood stuff seems to be handled by editing configuration files, rather than from a GUI. So if you're new to VoIP SAIL might be a better place to start.

On the other hand, competition encourages everyone to lift their game, and we can't have enough of it. SAIL still does some pretty weird stuff (e.g. not setting the localnet parameter at install time) and can be a bit buggy on occasion. And if you want to learn the ins and outs of how it Asterisk works, FreePBX seems to be more flexible and so forth.

[Daniel B.]

I have tried to install this twice and found that both times it didn't work.

Even the removal script doesn't work

SAIL is the better choice for PBX in my opinion.

It installs first go and is easy to configure

Stick with SAIL

Hi. FreePBX contrib is quite young. I use FreePBX on many production servers since 2 or 3 years. It's a great product. But the contrib itself may have some problem. Instead of saying "The install didn't work" and "Even the removal script doesn't work" (which I think both are wrong, I've installed this on several servers. It may just be a typo or something, or even a conflict with another contrib you have), you should explain what's not working. Open a bug and/or send me an email. This will make the product better.

And just one more thing: if you can't get it working, that doesn't mean it's a bad product.

Looking at FreePBX, a lot of the under-the-hood stuff seems to be handled by editing configuration files, rather than from a GUI.

Well, I can't really compare with SAIL as I only use FreePBX, but there's really not a lot to configure by hands. At the begining, you'll have to configure the hardware card, you may also have to enter your own language in sip and iax configuration file. Once this is done, quite everything can be controlled in the GUI, you just have to install additional modules. There's a lot of modules available in the module manager, directly from FreePBX GUI.

Cheers, Daniel

[Stefano]

I have tried to install this twice and found that both times it didn't work.
Even the removal script doesn't work
SAIL is the better choice for PBX in my opinion.
It installs first go and is easy to configure
Stick with SAIL

teviot.. IMO your post is rude and it does not help Daniel to solve your problem..

if you are interested in his contrib, then help him, us and yourself to solve the issues.. start writing what was broken/didn't work.. otherwise, please, simply don't post.

Thank you

Stefano

[Tib]

OK

I have setup a vmware freepbx setup

Installed SME 7.4 ... updated with the latest updates.

Followed the installation instructions.

Code: [Select]

yum --enablerepo=smecontribs install smeserver-freepbx
this code pulled in 14 files which included dahdi-tools dahdi-linux dahdi-linux-kmdl

then did a

Code: [Select]

signal-event freepbx-update
tried the

Code: [Select]

https://server.domain.tld/freepbx/admin
and there was nothing there

Not Found

The requested URL /freepbx/admin was not found on this server.

then I tried just

Code: [Select]

https://server.domain.tld/freepbx/
and I got

Forbidden

You don't have permission to access /freepbx/ on this server.

By the way ... I'm not interested in this contrib ... just thought I'd help test it.

Regards,

Tib

[soprom]

SAIL is the better choice for PBX in my opinion

There are a few differences between FreePBX and SAIL. I am not so inclined to say that SAIL is definitively better...

FreePBX has more options for those who are not using "custom apps" in SAIL. The support base (forums...) is wider. All those options have a cost: response time, complexity of the dialplan. There is a case where I can't install SAIL because they want those options. it's not easy to reproduce some of them in "custom apps" in SAIL.

SAIL has less options so it runs faster. This is noticable on sound quality and response time. It  also use an "engine" of it's own to process calls.  The codes on the phones to transfer calls etc are quite different from what is in use in americas (*73, *74...) and this is a downer.  SAIL is more straight forward to handle, easier after we know it. The main difference between both PBX, in a poweruser's perspective is certainly the "multi-tenant" functions. SAIL does it, FreePBX only dreams about it.

With "custom apps" in SAIL, transition from newbie to knowledge is quite something.

Even though I prefer SAIL, I still think that FreePBX is okay.

Thanks to Daniel for sharing his work with us!

And please! help him debuging and getting his contrib better instead of saying it's not that good...

[Daniel B.]

By the way ... I'm not interested in this contrib ... just thought I'd help test it.

Hi.
First, I'd like to thank you to take the time to test the contrib.
Unfortunately, it'd be quite hard to troubleshoot the issue in the forum. I've asked a component for freepbx in the bug tracker, it should be created soon. Waiting for this, can you please contact me by email (daniel AT firewall-services DOT com), and send me the install log (/root/freepbx_install.log)

Cheers, Daniel

[Tib]

Will do ... booting the VM as I'm writing this.

Regards,

Tib

[Teviot]

This is for those who think that this contrib works. 

First, I have tried to install it more than once.

second, I did try to get help with it ... see http://forums.contribs.org/index.php/topic,42811.0.html

third, Stefano a.k.a. nenonano ... you are the rude one.  If you frequent the forums (ALL OF THEM) not just one particular part of the forums you would see that I had already asked for help and got NO responce.

forth, Don't be like a number of other people that I have been victim to and witnessed by putting people down with out yourself doing some research.

I'm afraid that until I see some kind of real evidence that FreePBX on an SME server is working and worth a second look, I will continue to say that SAIL is the system of choice.

BTW this particular part of the forums is for SAIL. FreePBX shouldn't (In my opinion) have been posted here.

I had my say and will NOT discuss it further as from here it would only become a slinging match which I WILL NOT get involved with so please don't reply to this post. You won't get a responce.

Not that I need to explain myself to anybody, and this has nothing to do with SAIL, I used to use TRIXBOX but had to get a mate to configure it and make changes when needed.  SAIL, I configured it, I make the changes and my mate hasn't even logged in to see what I have done.  All he knows is it works.



VIP-ire
Don't stop developing it. I believe there is room for many contribs that when it is all said and done ... do the same thing only in different way. When you get this problem sorted, please let me know.

until then

LONG LIVE SAIL

[SARK devs]

OK guys...

Let's draw a line under the flaming.  We don't need it here.  You've both given as good as you got so let's leave it there.

The Forum is for all things VoIP and telephony on SME and FreePBX is welcome here.  Please everyone, this is a nice, courteous forum where people enjoy the learning experience.  Let's try to keep it that way and we'll all win.

Sophie - you never told us about the American standard feature codes or we would have changed them for you by now ( ).   

Can you send us over a list?  Is there such a thing?

Also can you expand some more on FPBX features that you don't believe are in SAIL?  We used to benchmark against FPBX but we stopped some time ago when we realised that SARK is in a completely different space to FreePBX.  Here in the UK we replace at least one or two failed FreePBX sites a month, however it isn't always the fault of FreePBX, it's sometimes the case that some space cowboy has offered a customer a PBX for 50 Cents and then done a runner when he realised he couldn't deliver. 





 

[soprom]

Thanks Selintra for helping this thread back to something productive...

about the American standard feature codes

For example, *97 and *98 are of frequent use to get voice mails, and call forwarding is done with *72 and *73. Since people are used to these codes, they have to learn a new set.

If you don't mind I'll insert a list right here (watch for the lines with "FreePBX:")

Code: [Select]

** FreePBX: Call Pickup (Can be used with GXP-2000)
*0 FreePBX: Speeddial prefix
*11 FreePBX: User Logon
*12 FreePBX: User Logoff
*16 Linksys/Sipura: Make all outbound calls secure
*17 Linksys/Sipura: Make all outbound calls not secure
*18 Linksys/Sipura: Make the next outbound call secure. This operation is redundant if all outbound calls are secure by default.
*19 Linksys/Sipura: Make the next outbound call not secure. This operation is redundant if all outbound calls are not secure by default.
*25 Linksys/Sipura: Enable Call Waiting Caller-ID generation
*26 Linksys/Sipura: Enable Distinctive Ringing
*30 FreePBX: Blacklist a number
*31 FreePBX: Remove a number from the blacklist
*32 FreePBX: Blacklist the last caller
*34 FreePBX: Perform dictation
*35 FreePBX: Email completed dictation
*43 FreePBX: Echo Test
*45 Linksys/Sipura: Disable Call Waiting Caller-ID generation
*46 Linksys/Sipura: Disable Distinctive Ringing
*52 FreePBX: Call Forward No Answer/Unavailable Activate
*53 FreePBX: Call Forward No Answer/Unavailable Deactivate
*54 FreePBX: User lntercom Allow
*55 FreePBX: User lntercom Disallow
*57 Linksys/Sipura: Disable Call Waiting on all calls
*60 FreePBX: Speaking Clock
*60 Linksys/Sipura: Block the last inbound call
*61 Trixbox: Weather
*62 Trixbox: Wakeup
*63 Linksys/Sipura: Forward the last inbound or outbound calls to the target specified after the activation code
*64 Linksys/Sipura: Accept the last outbound call. Let it ring through when DND or Call Forward All is in effect
*65 FreePBX: Speak Your Exten Number
*65 Linksys/Sipura: Enable Caller-ID Generation
*66 Linksys/Sipura: Callback when the last outbound call is not busy
*67 Linksys/Sipura: Block CID on all outbound calls
*68 Linksys/Sipura: Unblock CID on all outbound calls
*69 FreePBX: Call Trace
*69 Linksys/Sipura: Call the last caller.
*70 FreePBX: Call Waiting - Activate
*70 Linksys/Sipura: Disable Call Waiting for the next call
*71 FreePBX: Call Waiting - Deactivate
*71 Linksys/Sipura: Enable Call Waiting for the next call
*72 FreePBX: Call Forward All Activate
*72 Linksys/Sipura: Forward all calls to the target specified after the activation code
*73 FreePBX: Call Forward All Deactivate
*73 Linksys/Sipura: Cancel call forward all
*74 FreePBX: Call Forward All Prompting Deactivate
*74 Linksys/Sipura: Assign a speed dial number
*75 FreePBX: Set user speed dial
*77 FreePBX: Save Recording
*77 Linksys/Sipura: Block all anonymous calls
*78 FreePBX: DND Activate
*78 Linksys/Sipura: Enable Do Not Disturb
*79 FreePBX: DND Deactivate
*79 Linksys/Sipura: Disable Do Not Disturb
*80 FreePBX: Intercom prefix
*80 Linksys/Sipura: Cancel blocking of the last inbound call
*81 Linksys/Sipura: Block CID on the next outbound call
*82 Linksys/Sipura: Unblock CID on the next inbound call
*83 Linksys/Sipura: Cancel call forward last
*84 Linksys/Sipura: Cancel Accept Last
*85 Linksys/Sipura: Disable Call-ID Generation
*86 Linksys/Sipura: Cancel callback
*87 Linksys/Sipura: Unblock all anonymous calls
*90 FreePBX: Call Forward Busy Activate
*90 Linksys/Sipura: Forward busy calls to the target specified after the activation code
*91 FreePBX: Call Forward Busy Deactivate
*91 Linksys/Sipura: Cancel call forward busy
*92 FreePBX: Call Forward Busy Prompting Deactivate
*92 Linksys/Sipura: Forward no-answer calls to the target specified after the activation code
*93 Linksys/Sipura: Cancel call forward no-answer
*97 FreePBX: My Voicemail
*98 FreePBX: Dial Voicemail
*98 Linksys/Sipura: Blind transfer current call to the target specified after the activation code
*99 FreePBX: Check Recording
*99 Linksys/Sipura: Modem Line Toggle Code
They have a panel where all feature codes known to the PBX are listed by name with a field to define the code for each one.

Also can you expand some more on FPBX features that you don't believe are in SAIL?

Concerning other features...

Please keep in mind that they may be there in Selintra but I didn't figure out the way to activate them, and that I don't use easily custom apps. FreePBX allows a non-programmer to set some options, and that is what I'm discussing here:

Cell phones:

• Callback cell via a land line (TDM400) is something that Selintra was working on.
• Include cell in ring group (fallback)
  

The inclusion of cell phones is done with simple "custom extensions" that are then inserted into ring group. In Selintra, such a custom extension would be added in an alias extensions list. Calling extension 5432, for example, would ring a cell phone.

The Complex Follow-me:

This is something I can live without, but some seem to enjoy it...
See http://freepbx.org/news/2007-09-13/follow-me-or-vmx-locater-which-one-is-for-you

and VmX Locator with "Ring strategies" http://freepbx.org/2007/06/03/ring-group-and-follow-me-ring-strategies-1-of-2


And again, thanks for your work on Selintra!

[David Harper]

Sounds like SAIL has some new features coming its way

[soprom]

David, could you tell me where to get info on the coming features?
Thanks.

[SARK devs]

Thanks for the input Sophie - you're a star!

OK - functions first...

Callback - we've had that for some time and I think ours is better and safer than the FreePBX equivalent.  FPBX simply allows you to create an open number which will blindly callback any number that calls it.  SAIL callback will ONLY callback numbers it recognises as authorised to receive callback.  You can read about it here...  http://sarkpbx.com/twiki/bin/view/Main/DocChapterCallBack

Include cell in ring group.  - you've always been able to do this.  Just include the cell or national number in the ring group list...   5001 5002 07791234567 5006 5008 01924918076   and so on.   Also if you wish to "outcome" to a cellphone then simply create an alias with one entry (the cell number) and outcome to that alias.

Complex Follow me... This is where I think that we philosophically split with the FPBX guys.   We've always had simple follow me (effectively a remote or reverse call forward to wherever I am).  I never say never, but it is unlikely we'll ever do a complex follow me unless someone pays us to do it.  We have never been asked for this fature by a serious business phone user (maybe its different in North America).  To us, it just looks like one of those "just because we could" type features that go down a storm over at Nerd Vittles but have little or no relevance in the real World.  It's also a feature which any serious phone administrator would instantly ban.   So there you go - SAIL architect's worldview in one paragraph ( ). 

Regarding the feature codes - we'll take a serious look at implementing the standard ones.  Are these all Ma Bell compliant codes Sophie?

Lastly, we'll put up a roadmap on the Wiki showing some of the stuff we are working on.  Right now DAHDI is just rolling out (I think we've done a nice job with that - I'm pleased with it) and we are working on retrospective One Touch Recording (ROTR) -  we call it the TARDIS feature, but I don't know if that translates outside of the UK. We are also looking at record "ducking" to cover data protection issues in some (mainly European) countries.  These are specialised freatures mainly for call centres,  but along with our high availability platform and integration with third party products like QueueMetrics queue intelligence and AutoCab's GPS vehicle control system, they give us unique advantages in the professional VoIP arena.

Thanks again for your kind input Sophie. 

S

     

[Stefano]

This is for those who think that this contrib works.

at least it works for someone..

First, I have tried to install it more than once.

second, I did try to get help with it ... see http://forums.contribs.org/index.php/topic,42811.0.html

teviot.. you are a member of this community since 2004 and so I guess you are a linux & SME good fellow..

third, Stefano a.k.a. nenonano ... you are the rude one.  If you frequent the forums (ALL OF THEM) not just one particular part of the forums you would see that I had already asked for help and got NO responce.

FYI I read almost ALL the 3ads on the forums (like Cactus and other).. you can see from my profile.. that said, in this 3ad and in the other, you did not post anything usefull to help Daniel to solve your problems, nothing!

where are your logs? your install log? and what about looking in /var/log/messages?

forth, Don't be like a number of other people that I have been victim to and witnessed by putting people down with out yourself doing some research.

you are not a victim, so stop acting/writing in this way..

you should know that I (generally speaking) can help you ONLY if you tell me exactly what's going on.. and since you are in Australia [OT]the country where I would like to live in[/OT] and I'm in Italy, it's impossible for me to see your screen, your server, your logs.. HELP us to HELP you.. it's not so difficult.

I'm afraid that until I see some kind of real evidence that FreePBX on an SME server is working and worth a second look, I will continue to say that SAIL is the system of choice.

you will not see any evidence of it if all behave like you.. I repeat, if you are interested, then start testing it deeply and help Daniel to solve any issue REPORTING what's wrong.

BTW this particular part of the forums is for SAIL. FreePBX shouldn't (In my opinion) have been posted here.

IMO you are wrong.. this is the forum SME 7.x VoIP so is the only right place.

I had my say and will NOT discuss it further as from here it would only become a slinging match which I WILL NOT get involved with so please don't reply to this post. You won't get a responce.

again (it's the second time).. this is the worst attitude into a forum/community.. you are simply saying that You ARE RIGHT, the others no.. if you post in a pubblic forum like this one, you must be open to criticism

VIP-ire
Don't stop developing it. I believe there is room for many contribs that when it is all said and done ... do the same thing only in different way. When you get this problem sorted, please let me know.

HOW do you think Daniel could understand your problems and solve them if you don't say anything about them?

instead of writing this post and the previous one that do not add anything usefull, why did you not post your install log (for example)?

change your attitude, help the community and yourself.

Ciao
Stefano

[Daniel B.]

Indeed, there was a problem installing it from smecontribs, not because of the contribs itself, but because the version of speex was incompatible with asterisk, so asterisk couldn't be started, and so FreePBX couldn't start the install process. This is corrected now.
To install it, you just have to:

Code: [Select]

yum --enablerepo=smecontribs update speex
rmdir /opt/freepbx
signal-event freepbx-update

FPBX simply allows you to create an open number which will blindly callback any number that calls it. 

Sorry, this is wrong. Of course we can do it if we want, but you just have to create a inbound route with a special CID, and link this route to the callback. The callback will then only work with this special CID.

Cheers

[jmbac]

Thanks Daniel,

I will try reinstalling it again

My earlier attempt to install using smeserver-freepbx -0.12 failed using dahdi-linux & asterisk from smecontribs & atrpms repo .

I agree with you that the issue is asterisk not starting ..because I recompiled Trixbox srpms  and asterisk started ok and Smeserver-freepbx installed without a problem
. see my previous post

http://forums.contribs.org/index.php/topic,43529.0.html

Thanks for the excellent Contribs..

Thanks
Jon..

[SARK devs]

    FPBX simply allows you to create an open number which will blindly callback any number that calls it.

Sorry, this is wrong. Of course we can do it if we want, but you just have to create a inbound route with a special CID, and link this route to the callback. The callback will then only work with this special CID.

I stand by what I said Daniel.  I used the word allows.   SAIL does NOT allow you to create a blind callback.  It mandates the entry of an authorized number set.  With FreePBX you can have it happen unintentionally if you don't know what you are doing.  This would seem to us to be contrary to the notion of FreePBX as a tool for the novice PBX builder. 

Kind Regards

S



 

[Daniel B.]

Ok, sorry, I misunderstood what you meant. So yes, FreePBX allows it. And this is what I like in this product, it's very (maybe too much ) flexible. You can really do what you want with it. I think it's also very intuitive (I've started playing with VoIP with FreePBX, I've learned everything I know with this great tool)

[soprom]

Are these all Ma Bell compliant codes Sophie?

Thanks for your comments...

Here is a list of Bell "* Codes" (http://www.yashy.com/star.codes.html). It would be good to have a confirmation from someone who knows better than me. FreePBX and Trixbox are using most of these...

Code: [Select]

*03 DeActivate *66 Notification (One Time Only)
*12 Call Assitance (Police)
*30 Cancel Forwarding
*31 Automatic Forwarding
*32 Notify
*40 change forward-to number for customer programmable call forwarding busy-line
*41 six-way conference calling activation
*42 change forward-to number for customer programmable call forwarding don't answer
*43 drop last member of six-way conference call
*46 french voice activated network control
*47 override feature authorization
*48 override do not disturb
*49 - disable long distance special ring
*51 Intercom Ring I
*52 Intercom Ring II
*53 Intercom Ring III
*54 Extension Hold
*57 Call Trace
*60 Call Screen (Selective Call Rejection)
*61 Selective Distinct Alert
*62 Selective Call Acceptance
*63 Selective Call Forwarding
*65 ICLID Activation
*66 Busy Call Return
*67 Call Display Block
*68 Computer Access restriction
*69 Last Call Return
*70 Call Waiting Disable (Per Call)
*71 Three-way calling (Per Call)
*72 Activate Call Forwarding
*73 DeActivate Call Forwarding
*74 Speed Dialling (8 numbers)
*75 Speed Dialling (30 numbers)
*77 Anonymous Call Rejection
*80 Call Screen Disable
*81 Selective Distinct Disable
*82 Selective Acceptance Disable
*83 Selective Forwarding Disable
*85 ICLID Disable
*86 DeActivate Busy Call Return
*87 Cancel Anonymous Call Rejection
*89 DeActivate Last Call Return
*90 Re-Activate Call Forwarding Busy
*91 De-Activate Call Forwarding Busy
*92 Re-Activate Forwarding No Answer
*93 Call Forwarding No Answer (*91,*93 to completely disable)
*94 Ring Control (Call Answer)
*95 Message Monitor (Call Answer (link|flash) to interuppt/talk))
*97 check if you have messages, remotely (on=2beeps,off=4beeps)
*98 Call Answer

Include cell in ring group

Thanks, I'll put that to use right away!

Callback

I'll update my test and see...



As far as I am concerned, elaborate functions like the ones provided by FreePBX are requested by some users, but most people sleep well at night without them.

I think that Feature Codes adapted for the Americas are the main attraction here, like...

• *72 Linksys/Sipura: Forward all calls to the target specified after the activation code
• *73 FreePBX: Call Forward All Deactivate
• *97 FreePBX: My Voicemail
• *98 FreePBX: Dial Voicemail

This is like a feature request, I guess... and I'll sleep so much better at night!!!

[David Harper]

If the feature codes were virtualised in a database key, they could be re-programmed at will. That would be sweet, not only because of better support for local standards, but because SAIL could then be used as a drop-in replacement for a lot of legacy/proprietary systems.

[SARK devs]

my one big regret, and one of the few things that FreePBX does better than us, is the softkey mapping of feature keys.  Ours were based upon what we knew at the time, which was AT&T Definity (now Avaya).

Heigh ho


Best
S

[soprom]

my one big regret

Does it mean that I'll regret it too ?

[SARK devs]

No, we can do it, but it's going to be a long boring job for someone.

[churchtech]

Just installed using the Contrib, won't be able to test the actual PSTN connection until this weekend but the software install appears to have completed successfully. All expected web functionality is available. Thanks to Daniel B. for making this possible.

I do have a possible bug to report which I will do in the bug tracker...but there is no component listing in the 'SME Contribs' section of the tracker at this time. Though, I see Daniel's request for it to be added. Should I wait or post the bug somewhere else?

UPDATE: I reported the bug via email and the fix should be in the latest rpm.