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FreePBX 2.3 Beta Program (now closed)

2.3.0 final has been released! If you are running an RC or Beta release of 2.3, we strongly urge you to upgrade

The 2.3 beta program was a great success, with over 3900 installs in 9 weeks, and hundreds of bugs found and fixed. The following page is available for reference only; with the launch of 2.3.0 the 2.3 Beta program is officially closed. Thanks to everyone that helped out!

IMPORTANT: IF YOU USED FRAMEWORK TO UPGRADE TO BETA2

If you upgraded from a pre-beta2 release using the online module system then there is one important step you are going to want to take. The online update system is not able to update the /usr/sbin/amportal script that is used to start and stop Asterisk and FOP, since this is owned by root. When installed with the tarball, Beta2 replaces this script with a very simple script that calls another script, freepbx_engine which the online system can maintain.

This change also engages the ability to selectively start and stop FOP when Asterisk is started. You can take the following simple steps to make the change to your installation which will then allow the online system to do everything for you.

su - # you must be root
wget http://mirror.freepbx.org/amportal
cp ./amportal /usr/sbin/amportal

With that change, you will no longer need to worry about installing tarballs.

Beta Program General Information

Thank you for being part of our Beta program. Your support will help assure the stable launch of FreePBX 2.3 once we have flushed out and fix as many bugs as possible. If you find a bug, or something that's not working as the documentation says it should, please do the following:

Search the bug tracker for existing bugs and possible fixes
Consider posting discussions prior to filing bugs in The FreePBX Forum
If you are testing with Asterisk 1.4 and the bug might be related to this:
- Try and reproduce the bug on an Asterisk 1.2 version if at all possible
- If the bug exists there also, don't mark it as an Asterisk 1.4 bug, mark it as 1.2
- If it appears to be related to Asterisk 1.4, make sure the ticket reflects that information
create a new ticket (Note - you need to create an account on the trac system.
Use this timeline to track recent activity (bugs being fixed, tickets opened closed, etc.)

A more detailed 2.3beta and release candidate Bug Report can be found at this link as well.

Known Issues

These are the known issues. The first are ones that we could use some help testing potential fixes before we close the bugs. At the end of the list is a link to a more comprehensive report of these same issues.

THESE NEED TESTING

2.3.x bugs (by component)

Closed Issues

#813: Link to recordings aren't working properly
#1237: Wrong headers for paging Aastra 480
#1329: Unable to add /var/lib/asterisk/sounds/custom/FindMe.wav - Can not read file!
#1401: IE6+ and Module Admin
#1442: PHP Info and Asterisk Logs broken
#1487: Cannot edit / delete Queue when queue number is 0
#1520: asterisk 1.4 "show version" is now called "core show version"
#1522: Change DeadAGI to AGI in extensions.conf
#1533: Caller ID Doesn't name...only Number
#1610: retrieve_conf with get_version error - Asterisk 1.4
#1612: Modules should store recording ID - not recording file list
#1626: permissions wrong on /etc/amportal.com /etc/asterisk/asterisk.conf after amportal chown
#1633: Deleting inbound routes needs to redraw the page
#1638: Blacklist mod context has same line twice
#1659: Patch to fix #1487 Queue number 0
#1670: Caller ID Prepend on queue replaces existing callerid (instead of prepending)
#1733: dialing non-default voicemial contexts broken sometimes
#1736: alert-info database fields are too short for some requirements
#1745: Misc Applications delete doesn't delete from db table
#1747: Indications.conf needs updating to add za
#1748: UTF-8 gets stripped in dialplans
#1759: Don't hide password field in ampusers if AUTHTYPE != database
#1770: Should miscapps and pinsets have uninstall.sql files?
#1774: timeconditions module does not install it's tables on sqlite
#1775: miscapps module does not install it's tables on sqlite
#1776: manager module does not install it's tables on sqlite
#1778: pinsets module does not install it's DB in sqlite3
#1785: ARI crypted file bug with apache/ubuntu
#1808: Should check vmcontext for a user for illegal values
#1813: Voicemail symlink not being created, MWI broken
#1820: IVR hangs up after announcement if auto fallthrough is set to "no".
#1826: Translation broken in left menus when some modules are enabled
#1836: Cleanup after deleting an extention needed
#1841: Can't add + to Direct DID for extension
#1855: calls coming in on pri with <blank> (null) calling number are not 'tagged' with ring group or followme prefix
#1857: Undefined variable: astman
#1861: intercom dialplan not added unless a paging group is created
#1866: Hard Coded Paths & Settin
#1878: Installation free pbx 2.21
#1882: retrieve_op_conf_from_mysql.pl sort order wrong
#1884: FreePBX and compatibility with Asterisk 1.4
#1889: CID Num Alias
#1900: System recordings being forgotton
#1902: Misc Application feature code not updating correctly
#1903: safe_opserver is a sh script but uses "source" command
#1906: Going to vm if vm module is not installed
#1907: 180 Ringing sent before 486 Busy Here and 503 Service Unavailable
#1921: voicemail module is not working
#1925: Module Update through squid non-transparent proxy
#1933: adding a did that already exists does not give an error
#1947: Rename Recordings to system recordings
#1961: findmefollow problem with extensions 0-leaded
#1963: mysql_real_escape_string not available when running on sqlite3 mode
#1965: prefix tag for call queue is added multiple times when looping
#1967: Security on Asterisk Logs Module
#1969: music on hold java validation scripts not working
#1970: Changes required for 1.4 support
#1973: Dissable Fax email field in Extensions if Fax Extension is set to "system"
#1982: depends version checking used by module admin is broken
#1983: New install isn't reading amportal.conf
#1984: callback agi script needs to be moved into the callback module
#1985: queues sql generation move to module
#1987: link images in retrieve_conf from modules
#1989: Long Lines Truncated in Trunks/Outgoing Settings/Peer Details
#1992: Backup Schema being created in core and not module
#1993: Dissable the ability to uninstall CORE
#1995: 2.3 Install error
#1997: include file before extensions_additonal.conf
#2007: latest svn update rejects asterisk 1.4.5
#2008: Syntax error in install_amp
#2009: callerid parameter not getting set correctly for created/updated extensions
#2010: Instructions spelt incorrectly
#2012: Garbled module string in module admin screen
#2016: fix AMPPLAYKEY to use amportal.conf default
#2017: Unable to check for module updates online.
#2019: Can't update extensions
#2020: symlinked agis can fail
#2021: dialplan.agi and asterisk 1.4 database get error
#2023: is_empty should be empty
#2024: Ring group needs duplicate checking
#2025: edit existing user with directdid compains because user had direct did
#2026: Incoming routes SQL error
#2027: can't save recordings made with *77
#2028: Directory does not work!
#2029: AMPWEBROOT not installing to /panel
#2031: change engine_getinfo() to return proper version numbers for Asterisk SVN branches
#2032: why are recordings a destination
#2035: MailBoxExists requires 'j' option to function on 1.4
#2036: FOP 0.27 console messages after FOP restart
#2037: Dialparties.agi symbolic link doesn't work
#2038: extesions additional does not update
#2039: try and report status of retreive_conf
#2040: Company directory doe not work
#2041: Core: 2.3.0beta1.4 module breaks fixlocalprefix
#2042: core agi scripts with hardcoded paths
#2044: ChanSpy
#2045: ChanSpy
#2046: VmX Doesn't work
#2047: Day and Night Swapped in Day/Night Module
#2048: iax2 route
#2049: Speed Dial issues with speeddial-clean macro
#2050: iax2 route
#2052: core 2.3.0beta1.6 broke outbound dial patterns fixlocaprefix
#2053: fixlocalprefix is broken
#2055: *99 ignores Language Selection
#2057: CID Prefix removed even if not prefix added
#2061: app-blacklist-add/last - Auto fallthrough on 1.4.6
#2070: FreePBX incompatible with php 5.2
#2078: remote clients will not register
#2079: Problem with Routes - (version 2.3.0-beta1)
#2080: Updating modules doesn't draw the screen properly
#2081: Error after latest SVN of 2.3
#2083: Online support: online documentation: link incorrect
#2084: Dashboard module
#2086: Undef vars in install_amp
#2087: Undef vars in install_amp
#2090: DeviceandUser mode problem with users who are not logged in
#2091: DeviceandUser mode problem with users who are not logged in
#2092: Unable to add /var/lib/asterisk/sounds/custom/filename.wav - Can not read file!
#2094: Dialplan injection do not work inside 2.3beta1
#2095: Patch 963 not included in 2.3 alpha
#2096: ALERT_INFO deprecated in Asterisk 1.4
#2111: log notification critical error on retrieve_conf failure
#2113: dialparties.agi error with ring groups and call forward
#2114: Removing a PHP5 warning
#2115: Asterisk Direcory Bug
#2116: Add new op_style.cfg to fop .027
#2117: No event handler for event 'peerstatus'
#2119: "Connecting to database..FAILED" error when installing 2.3 beta1 and 2.3 branch
#2120: retrieve_conf failed
#2122: No Web GUI for FreePbx 2.3 beta1 or branch svn4458 on CentOS 5
#2124: FOPWEBROOT path
#2125: ampwebroot with wrong reference
#2127: call-limit parameter needed for Asterisk 1.4 compatibility
#2128: AMPDBPASS not set on beta2
#2129: amportal chmod should be runned after ./install_amp
#2130: Online Core Upgrade 2.3.0beta1.9 to (2.3.0beta2.0) Fail
#2131: Dashboard (FreePBX System Status) Undefined Index errors
#2132: Safari Browser Problem (JQuery Issue)
#2133: 2.3.0Beta2 - draw_graph function is confused by numbers formatted with ',' - comma
#2136: Adding a freepbx admin to a department permanently locks the user to that department, even if removed
#2137: VmX Locator is disabled in ARI - even though its enabled for the extension.
#2138: pear DB library creating deprecated is_a() errors
#2139: FreePBX System Status - Failed to copy from module agi-bin
#2140: Warning: Division by zero
#2141: Fatal error: Call to a member function addError()
#2142: Phones Online (in FreePBX System Status) problem
#2143: Caller Lookup Source not Showing on Inbound Route (hooks broke)
#2144: Privacy Manager not working per extension
#2145: Macro: app-calltrace-perform - fails on Asterisk 1.4
#2146: Installign modules after a clean install
#2147: Link to FOP pointing to wrong URL
#2148: disa module not working properly
#2150: error in freepbx cron job output
#2151: Outbound route trunk sequence loses order
#2153: no folders in /var/spool/asterisk/voicemail/default/ voicemail dont work
#2154: function do_reload() / start bounce_op.sh problem
#2155: Description for Maximum Channels on Edit Trunk is incorrect.
#2156: Bug in page.queues.php with other languages than english
#2157: authtype=database - user must have access to all or nothing restricted accounts dont work
#2160: Administrators modules not working
#2161: Reload sometimes only partially shades the screen
#2162: logout of database mode user results in error
#2163: Syntax error in Phonebook module
#2165: Paging first time use displays on a blank page
#2166: Voicemail error after hitting #
#2168: Intercom dysfunctional with 1.4 backend as SIP settings are not inherited.
#2169: featurecodes delete method does not work
#2170: Styling Issues on Dashboard
#2171: The return of engine_getinfo is not evaluated in retrieve_conf
#2172: move from | to , in dialplan for asterisk 1.6 support
#2173: Error message after deleting a Queue
#2174: Flash Operator Panel layout setup don't refresh correctly
#2176: Extensions error
#2177: Error after clicking submit button
#2178: Flash Operator Panel don't follow language setting
#2180: Freepbx framework 2.3.0beta2.6 has wrong file op_style.cfg
#2185: MOH by Inbound Route breaks when forwarded to Queue
#2186: Call Forwarding on RI (new)
#2187: AsteriskInfo module has problems with Asterisk 1.4 (with patch)
#2188: Queue problems with MOH/ringing and recording (Asterisk 1.4.9)
#2189: [PATCH] Direct Dial to Voicemail message doesn't work at all with Asterisk 1.4.9
#2193: moh path hardcoded
#2194: FreePBX System Status (index page/dashboard) fails with ugly errors if Asterisk is not running
#2199: new extensions are not honoring the ENABLECW setting in amportal.conf
#2201: retrieve_conf failures overflow the orange box and could use better formating
#2202: call report does not work since 2.3 beta 2.7
#2206: Error: Did not receive valid response from server
#2208: intercom and Aastra phones
#2209: FreePBX Statistics Phones Online Count - Beta 2.3.0beta2.8
#2210: trunk dial patterns are being saved but not redisplayed on trunk screen
#2211: Add jquery scripts to CONTRIB.txt
#2212: Include queues_custom.conf after queues_additional.conf
#2213: Update notification
#2214: anonymous URI call get busy if originated from aastra 57i phones
#2216: Translation Appear to be broken
#2217: Could not reload the FOP operator panel error
#2218: FOP doesn't work if AMPWEBADDRESS includes a port number (:81)
#2219: amportal should check for ASTRUNDIR
#2222: Asterisk 1.4.9 hangup call on queue annonce if Announce Position or Announce Hold Time set to NO
#2224: HUDLite server
#2225: Tab graphics are broken
#2226: queue module post install error
#2227: reload bar doesn't show up after submit
#2228: Can't Enable Dictation Services
#2229: Dashboard 0.3 report an error when mysql is not local and on port 3306
#2230: Fatal error: Maximum execution time of 30 seconds exceeded in /var/www/html/admin/common/php-asmanager.php on line 187
#2231: astdb is not restored by System Restore -> Restore System Configuration
#2232: Dialing MyVoicemail asks for mailbox number
#2233: cdr not showing any calls
#2237: Misspelling in endpoint manager edit page
#2239: Web Server Time Out Issue
#2240: Queue problems with MOH/ringing and recording (Asterisk 1.4.9)
#2242: Strangeness inside Enum macro
#2243: trunks can be counted twice in Freepbx Status page
#2244: MOH wierdness
#2245: old framework files copied back to system
#2246: 'FreePBX Connections' heading is not visible on Dashboard status page
#2247: RC1 upgrade problem
#2248: Apply configuration Changes fails in mozilla browsers
#2249: callmonitor.module allows user to see all calls, not just their own
#2250: FreePBX System Status assumes standard port 22 for SSH
#2252: FOP op_style.cfg UPDATE
#2253: Remove jbenable and jbforce from sip.conf
#2254: Missing parameters in op_style.cfg
#2256: [PATCH] AJAX reload fails when proxy is being used
#2258: Cosmetic issue with Zork in Module Admin
#2259: Outbound dialing doesn't try next trunk if ENUM/ trunk fails (patch)
#2260: MOH doesn't work due to change in directory name (patch)
#2264: MySQL system status ERROR
#2265: Still DeadAGI in macro-record-enable of extensions.conf
#2266: [patch] Visualization problem with ARI and translations
#2267: restoreastdb.php fails to restore values to astdb that contain spaces
#2268: Backup and restore interface needs polishing
#2270: Undefined index and variables module_admin and freepbx_cronsheduler.php
#2276: Outbound routing problems
#2277: [PATCH] Queues "Voice Menu" playing caller position
#2278: Disks with long paths look ugly in System Status (dashboard)
#2284: FreePBX 2.3.0RC1.1 and Blind Transfer to Voice Mail
#2286: No OP panel
#2287: Remote Announce in Ring Group Not Announcing
#2288: CDR Reports Not Working
#2289: Flash Operator Panel Not Working
#2290: CDR Reports Not Working
#2291: Networks under System Statistics in FreePBX Admin Page Not Working
#2292: CDR Report Not Working
#2293: Flash Operator Panel Not Working
#2294: ARI Not Working
#2296: Privacy Manager does not work (on 1.4)
#2300: Online Support Page - Invalid Link to Documentation
#2305: Feature Status in Misc Applications don't change
#2306: System Recordings bugs or Imcomplete
#2309: Queue issues
#2310: Set maximum channels in SIP trunks broken in 1.4 ?
#2311: Can't backup NOW
#2336: Variable not defined in freepbx-cron-scheduler.php
#2363: Unable to manually dial sip/12345@notmydomain.com
#2399: Caller ID Lookup Source. Customization.
#2404: VMX options not updating (timeout, repeat & loop)

Downloading 2.3

Make sure you are logged in as root; get the tarball, unpack it and install it on your system with install_amp. Then make sure Asterisk is running.

      su -
      cd /usr/src/
      wget http://mirror.freepbx.org/freepbx-2.3.0rc1.tar.gz
      tar -zxvf freepbx-2.3.0rc1.tar.gz
      cd freepbx-2.3.0rc1
      ./install_amp --force-version 2.2.2 # required if you were running svn/trunk
      amportal start

Go to Module Admin and install or upgrade any modules that need attention
Start testing and reporting bugs as needed

Getting The Latest 2.3 beta and RC Fixes

FreePBX 2.3 now allows Core modules to be updated online, as well as the FreePBX frameworks itself, which is all the other files requires to build the application. As part of the beta program, we will be over exercising this ability which means we will provide many of the required bug fixes that are addressed during the program through these mechanisms. In order to maximize our testing efforts, we would request that you use this facility. If there is a critical bug that you require and it has not been made available yet, contact one of the developers (we are usually on #freepbx-dev on the IRC, or got to the Forum) and we will push out an upgradeable patch.

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