root/freepbx/branches/2.1/CHANGES

2.1.3
- Fix the broken security fixes in 2.1.2 (sigh)
- Fix the newinstall.sql file from 2.1.2 to work with older MySQLs
- Add a noload for app_trunkisavail.so (This will be taken out as soon as
  we start using it 8)
- Fix typo in welcome page
- Fix warning on upgrade about AMPDBENGINE
- Add "+" as a valid dial prefix
- Add ability to force a refresh of modules.xml
- Add 'transmit_silence_during_record=yes' as a default, as having it on
  doesn't cause problems, and having it off can.

2.1.2
- Changed the default so that a new extension has Call Waiting on by default
- Changed the online module repository to point to mirror.freepbx.org,
  as we were having problems with sourceforge
- Start of support for pgsql and sqlite
- Added the option to set CID to 'hidden' to force no CID being sent
- Security fixes in CDR
- Fixed newinstall.sql so that Core is enabled by default, added some
  help text to point new users in the right direction

2.1.1
- Rob Thomas (xrobau@gmail.com) takes over stewardship of freePBX project from Coalescent Systems
- Clean up harmless warnings in recordingcheck (r1927 and r1940)
- SIP Anonymous wasn't working when language was not set to 'en' (r1932)
- Fixed unfortunate loop when more than 10 trunks defined (r1942)
- Voicemail changes weren't immediately visible (r1945)
- Various fixes to clean up parser errors in extensions.conf, and rewrite of a couple of macros (r1949, r1950, r1953, r1957)
- Various minor text cleanups (r1960, r1962)
- Show fatal error message when cannot read /etc/amportal.conf file (r1971)
- Add simple script for A@H users to restore their non-standard modules (r1972)

2.1

- Modules not packacked with FreePBX
- Included interface used to download/install/upgrade modules
- Inbound Routing based on (analog) zap channel (ie: no DID available)
- Russian and Portuguese 
- ModuleHooks system allows modules to interact with eachother
- dialparties completely re-written in PHP - eliminating dep for asterisk-perl
- General Option to allow unauthenticated SIP calls into the system
- Define different "Dial()" options for outbound calls
- Direct DID->Extension config
- New modules, including FeatureCodes, Callback, PinSets, and others

2.0

- AMP is now "freePBX"
- New module system allows for drop-in functionality
- Requires Asterisk 1.2.x
- All previous AMP functionality ported to new module system
- Added Modules: Conferences, Time Conditions, Asterisk CLI, Online Support 
- GUI improvements
- FOP .24
- ARI 00.08.03 - now with AJAX!
- Outbound Routes can now use an Authenticate Password File
- Queue Static Agents can have penalties applied
- Using native music on hold support - no more mpg123!!
- Default is to use freePBX database authentication.  New installs create a new user.
- Initial sqlite support!
- Much improved form validation for all modules
- Inbound routes can set ALERT_INFO variable for SIP devices
- Ability to force Emergency Caller ID for devices using an Emergency Outbound Route.

1.10.010

- Tested with Asterisk 1.2 (beta)
- Tested with PHP 5
- Removed all the sound files from AMP archive, instead depend on asterisk-sounds
- Ability to execute a script after applying changes in the AMP interface
  (see amportal.conf in source archive)
- Allow accountcode for IAX devices (again)
- Show custom extensions in FOP
- Allow mailbox setting for device to be set manually (for shared mailboxes)
- HINT extensions are now created for both FIXED and ADHOC devices
- Display AMP version in footer
- Support for remote mysql database
- ARI upgrade adds i18n and user settings
- Remove Play Next option from voicemail options and default to
  play next when deleting or saving voicemails
- Lots'o'bug fixes

1.10.009

- Asterisk Recording Interface (ARI).  ARI is a php interface to Voicemail and monitor recordings. (written by littlejohnconsulting.com)
- Queues can now play a "welcome" message to callers upon joining.
- DID Routes re-written as Inbound Routing.  This allows for DID specific fax emails and call answering options.
- RingGroups now use strategies: Ring All (default), Hunt, Memory Hunt
- Optional separation of Devices and Users (AMPEXTENSIONS option added to amportal.conf).  Devices are endpoints (ie: phones), and can be Fixed to a user, or Adhoc.  Users are extensions, with options like voicemail.  A user can log in to Adhoc devices by dialing *11, and log-off by dialing *12.
- Custom device technology support
- HINT priorities for FIXED devices
- Interface translated to French, German, Italian, Spanish
- FOP .21
- FOP button layout can now be sorted by last name or extension number

1.10.008

- Backup/Restore (schedule and restore backups)
- Extension Call Recording (inbound and outbound calls)
- Queue Call Recording (inbound to agents)
- Custom Trunks (use any Asterisk supported technology as a trunk)
- Remote Agents (join a Queue from any endpoint on a trunk)
- Outbound Route Password (require a password for certain outbound patterns)
- i18n (web interface can now be translated)
- ZAP trunk channels no longer hard-coded in retrieve_op_conf_from_mysql.pl
- *<exten> dials direct to voicemail()

1.10.007

- Added cvs2cl generated ChangeLog (see this for all changes and bug fixes)
- Added AMP Users (multi-department, multi-tenant)
- Added incremental upgrade script (install_amp)
- Use /etc/amportal.conf to tweak AMP to your environement (MySql credentials, web root, ip address, etc).  Apply changes with apply_conf.sh
- New Outbound Routes page to control trunks used for outbound calls based on dial patterns
- LCR using Outbound Routes
- Trunks page redone to support routing, adds dial rules to modify numbers per-trunk before dialing
- ENUM Trunks
- Queues support added
- Support for ZAP extensions
- More voicemail options added
- New AGI-based directory application to support both first and last name lookups and return to operator
- provide customization points for all AMP generated extension contexts.
- Upgrade to Flash Operator Panel 0.20
- Upgrade Asterisk-Stat to v2.0


1.10.006

- Use extensions_custom.conf for customizations.  Sample included.
- Add option to define outbound CallerID on trunks
- Add option to define outbound CallerID for extensions
- Create extensions without voicemail and directory
- Web voicemail (vmail.cgi) will automatically log in users linking from email notication. NOTE: see the new vm_email.inc.template for new URL format
- Add Call-Forward on Busy application (enable: *90<destination>, disable: *91)
- Upgrade FOP to 0.19.  AMP now writes out op_buttons_additional.conf
- Include AMP version on admin welcome page
- Rework extensions admin
- Add 'allow','disallow' settings for SIP and IAX extensions
- Add 'pickupgroup','callgroup' settings for SIP extensions
- Digital Receptionist voice menus can now be named
- Allow custom goto for Call Groups
- Digital Receptionist wizard check for proper format on custom goto
- Fixed bug which limited AMP to 10 Digital Receptionist menus
- Default outbound numbers now dial via a macro
- Increase verbosity of mysql connection errors
- Fixed upload wav for Ditial Receptionist
- Fix Trunks admin so that it writes FOP config

1.10.005

- Add "Advanced Edit" qualify= option for NEWLY created extensions
- Add support for custom applications in Digital Receptionist admin
- Prevent creation of multiple DIALOUTIDS variables in Trunks admin
- Allow for long 'register' sting in Trunks admin (for new installs only)
- Don't allow an extension number to be changed in Extension admin (force delete/re-create extension)
- Fix counter bug in Digital Receptionist admin

1.10.004

- Added Call Group CID Name prefixing
- Renamed parking.conf to features.conf
- Added condition to dialparties.agi that prevents potential pinning of the CPU
- Allow Digital Receptionist voice recordings to be uploaded in AMP admin
- Added new AMP logo
- Added AMP process control script "amportal"
- Write meetme configuration for IAX and SIP extensions
- Added IAX2 and SIP trunking
- Added "DID Routing"

1.10.003

- Added support for IAX clients
- Upgraded to FOP 0.17