freePBX" href="/wiki/index.php?page=freePBX">FreePBX®

THIS SITE SHOULD NO LONGER BE USED FOR freePBX" href="/wiki/index.php?page=freePBX">FreePBX DOCUMENTATION. ALL DOCUMENTATOIN IS NOW BEING MAINTAINED ON The freePBX" href="/wiki/index.php?page=freePBX">FreePBX Website

NOTE: For pages that were pre-existing on this site, the page formatting was not correctly transferred to the new site, making some articles harder to read. You can still view the original freePBX" href="/wiki/index.php?page=freePBX">FreePBX® Documentation pages here, but be aware that any updates to these pages might be added at the new site.

freePBX" href="/wiki/index.php?page=freePBX">FreePBX is the new name for Asterisk Management Portal, the Web GUI for Asterisk (and, potentially, other VoIP PBX's). If you find something wrong or missing, please feel free to correct it yourself - This is a 'Wiki', and it gives everyone the ability to enter whatever information they want, or feel is missing.

Installation

• FreePBX 2.2 Issues
• Upgrading from A@H or an older AMP
• CentOS 4.3
  • Using yum on RHEL instructions
• CentOS 4.4, Sangoma A200D, 3Ware 8006-2LP SATA RAID
• ClarkConnect
• Debian Linux
• FreeBSD
• Gentoo Linux
• PoundKey
• SuSE Linux
• Ubuntu Linux

Release Notes

If you're upgrading from a previous version, READ THESE FIRST.

• 2.1.1 - 2006-05-30
• 2.1beta1 - 2006-04-27
• INSTALL

Configuration

• Module Admin - Read this first!
• General Settings
• Administrators
• Asterisk CLI
• Backup and Restore
• Callback
• Call Cost
• Conferences
• Destinations
• Digital Receptionist
• DISA
• Extensions
• Feature Codes
• Follow Me
• Inbound Routes
• Misc Destinations
• On Hold Music
• Online Support
• Outbound Routes
• Paging and Intercom
• Parking Lot
• PIN Sets
• Print Extensions
• Queues
• Ring Groups
• System Recordings
• Time Conditions
• Trunks
• VMware
• ZoIP Read This before Installing Module!

Configuration of Unofficial Third-Party Modules

(These modules are not supported by the creators of freePBX" href="/wiki/index.php?page=freePBX">FreePBX)

• Custom Contexts
• Dialplan Injection
• Extended Routing
• Feature Panel

Fax

• Asterisk With NVFaxDetect If you want to try to receive incoming faxes with Asterisk and freePBX" href="/wiki/index.php?page=freePBX">FreePBX, here are the setup instructions
• Faxing with rxfax and txfax
• Installation of HylaFax with IAXmodem

Common Problems

• Invalid Conference Number when using Page or Conferences
• 'You are running freePBX and asterisk with the default manager pass.' warning appears
• 'You are running freePBX and mysql with the default password' warning appears
• YOU MUST ACCESS THE CDR THROUGH THE ASTERISK MANAGEMENT PORTAL! when viewing Call Records
• 'You Need to Upgrade' when trying to use online modules.

Hints and Tips

• Asterisk Tutorials Free video tutorials on Asterisk, TrixBox, and freePBX" href="/wiki/index.php?page=freePBX">FreePBX
• Custom feature codes to read back the feature status of extensions - Allows users to find out how features such as Call Waiting, Do Not Disturb, and the variations of Call Forwarding are presently set, simply by dialing a feature code
• Feature Codes - Numbers you dial for various things (Call Forwarding, Divert, etc)
• Follow-Me Function implemented using Queues tutorial at VOIPSpeak.net
• Hints on Route Dial Patterns and Trunk Dial Rules - Helps to clear up confusion for new freePBX" href="/wiki/index.php?page=freePBX">FreePBX users
• How to bypass Grand Central's requirement to press 1 to accept the call - Google's GrandCentral provides a way to get a free DID, and a BroadbandReport.com user reveals how you can use it with freePBX" href="/wiki/index.php?page=freePBX">FreePBX
• How to change incoming CallerID - Does your provider chop off or add digits to CallerID, so call return doesn't work? Here's a way to fix it (until we get a way to do it in the trunk definition)
• How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail) - Can't get an inbound route for a particular SIP trunk to work? Calls from one particular SIP provider won't come in at all, or only come in on your default route? Or, maybe you have two or more SIP trunks from the same provider, and Asterisk thinks all your incoming calls are from only one of the trunks? The answers might be here
• How to make multiple extensions use a common voicemail box
• How to make voicemail accessible from an outside line - You can allow your users to access their voicemail from anywhere there's a phone
• How to upgrade Asterisk - This is not necessarily something you should do, but if you feel you must, here's how
• I upgraded, but I can't do a restore
• MythTvOsd - Displaying caller ID info on MythTV's on-screen-display
• New freePBX" href="/wiki/index.php?page=freePBX">FreePBX users guide to diagnosing problems - Read this BEFORE you go into the #freepbx IRC channel
• Resolving Audio Problems - No audio? One-way audio? Here are some common fixes for audio issues - also read this page if you are trying to set up freePBX" href="/wiki/index.php?page=freePBX">FreePBX behind a NAT firewall
• Resolving freePBX" href="/wiki/index.php?page=freePBX">FreePBX and Sipura/Linksys Supplementary Service and Feature Code Conflicts
• Setting up a trunk and route for free Directory Assistance in the USA - Don't pay the phone company for looking up a number!
• Setup a Linksys/Sipura SPA-3000 with freePBX" href="/wiki/index.php?page=freePBX">FreePBX - Also applicable to SPA-3102
• Tested and working SIP provider configurations
• Trunk Hints - Configuration hints for various VSP's

Hardware examples

• Using freePBX with an AVM B1 Active card
• Using freePBX with an Eicon Diva 4BRI/V-4BRI server adapter

Dependencies

• List of sound files that freePBX requires