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Sat 17 of May, 2008 [08:22 UTC]

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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.46s
  • Memory usage: 2.23MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 0.97

freePBX

Image


FreePBX Open Telephony Training Seminar in Las Vegas, NV - May 20-23rd, 2008


FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any "built in" programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.

FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. Some of the features that FreePBX supports out of the box are:

  • Unlimited number of Voicemail boxes
  • "Follow Me" functionality
  • Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)
  • Unlimited number of Conferences (limited by available CPU power - about 300 simultaneous users in conferences on a P4 3ghz - 600 with a dual core!)
  • Paging and Intercom functionality for man SIP phones that support it.
  • Music on Hold (via MP3s, or streamed off the internet)
  • Call Queues
  • And many other features


Some screenshots are available, to see what it looks like


FreePBX is built on the LAMPA™ stack (Linux, Apache, MySQL, PHP and Asterisk). It's a modular system, with click-to-install plugins downloadable over the internet from the online module repository.

FreePBX Features at a Glance:



  • Add or change extension and voicemail accounts in seconds
  • Native support of SIP, IAX, and ZAP clients (other endpoints are supported through custom extensions)
  • Supports all Asterisk supported trunk technologies
  • Reduce long distance costs with LCR
  • Route incoming calls based on time-of-day
  • Create interactive Digital Receptionist (IVR) menus
  • Design sophisticated call groups
  • Manage callers with Queues
  • Upload custom on-hold music (MOH)
  • Search company directory, based on first or last name
  • Detect and receive incoming faxes
  • Share administrative duties
  • Backup and Restore your system
  • Save audio recordings of calls
  • View call detail reporting with asterisk-stat
  • View extension and trunk status with Flash Operator Panel
  • View conversation recordings with Asterisk Recording Interface (ARI)

Project Sponsored by Atengo LLC

Resources



Notes


Created by jht2, Last modification by James Finstrom on Sat 19 of Apr, 2008 [15:45 UTC]

Comments

Comments Filter

Freepbx ubuntu install

by Johny Kadarisman on Wednesday 04 of October, 2006 [21:31:20 UTC]
Correction on ubuntu installation guide, you have to do svn checkout freepbx first before continue to update database privileges.

Hope this help. :)

Call Waiting

by Adam Sheridan on Tuesday 25 of April, 2006 [08:45:48 UTC]
This should be enabled by default.
I use grandstreams and have to do a *70 on each one of them before the lines will work.
I would have thought most people would want the chance of a second call coming in rather than straight to voicemail.
Maybe a choice in a generel config page for things like this, i.e. Call waiting, DND, ....etc.

Asterisk@Home Digital Receptionist

by Nick Kewney on Tuesday 04 of April, 2006 [15:30:52 UTC]
I installed Asterisk@Home 2.8 Beta to mess around with, but the digital assistant is throwing up the following error:

Warning: array_keys(): The first argument should be an array in /var/www/html/admin/modules/ivr/functions.inc.php on line 81

Warning: Invalid argument supplied for foreach() in /var/www/html/admin/modules/ivr/functions.inc.php on line 81

Anybody know anything about it?

Nick

FreeBSD

by muppie on Wednesday 24 of August, 2005 [11:46:32 UTC]
Does anyone have instructions on how to install AMP on FreeBSD? Thank you.

Re: security documentation REALLY needed

by davidcsi on Wednesday 17 of August, 2005 [22:38:26 UTC]
To change the maint and wwwadmin password, you must delete the users from the file: /usr/local/apache/passwd/wwwpasswd
and add the users again like this:
htpasswd /usr/local/apache/passwd/wwwpasswd maint
htpasswd /usr/local/apache/passwd/wwwpasswd wwwadmin


DVG

security documentation REALLY needed

by firestrm on Saturday 21 of May, 2005 [21:17:09 UTC]
(:evil:)
Like how do you change the #@#$%*& maint password!!

Help Wanted

by tokio69 on Saturday 12 of March, 2005 [16:18:02 UTC]
I'm using Asterisk @ Home and it has a buil in AMP.
How can I activate the Digital receptionist? How does it work.
I can not get it to work. tom@tokiogroup.com

doesn't support ZAP FXS extensions.

by jgabriels on Thursday 10 of February, 2005 [21:23:59 UTC]
You must manually add those by editing the
config files.
Edit

$200 documentation BOUNTY

by Anonymous on Friday 12 of November, 2004 [14:29:05 UTC]
There is currently a $200 bounty for writing a step by step documentation guide for the installation of AMP.

If you are interested in working on this please email me dean'at'collins.net.pr
Edit

cblevins

by Anonymous on Thursday 21 of October, 2004 [04:44:59 UTC]
(:biggrin:)
This is the bomb!
I have been evaluating many GUI managers and starting to get a bit discouraged. And this project snuck in!

Thanx so Much

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