Topic: [ANNOUNCE] smeserver-freepbx

[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.