root/archive/quickform/CHANGES

2.2 Final
- Fixed trunk 'locking' (Never Override CallerID) checkbox to work properly. This allows a
  trunk to restrict outbound CallerID settings to those of the trunk or defined in an
  extensions outboundcid settings. WARNING: THIS WILL BREAK 2.2.0rc1 WORKAROUNDS. The logic
  was reversed, you had to check the box in rc1 to get it to allow such foreign callerid's.
  That was a bug (#1501). By fixing the logic, the behavior is now opposite and you may
  need to go back to your trunks and change it.

2.2
- New Modules: phpagiconf, printextensions, customerdb, inventorydb, announcement, blacklist,
  cidlookup, customerdb, dictate, inventorydb, parking, pbdirectory, phonebook, printextensions,
  speeddials, ZoIP
- New option in amportal.conf for remote backups (as well as significant backup fixes)
- Changed Call Recordings to user MixMontior, better performance and more reliable.
- Fixed prefix lookup to use localcallingguide.com XML interface
- Fix potential security hole in CALLERID(name) and CALLERID(num) (see r2076)
- Redo front end with the new look, Thanks to Steven Fischer for the template
- Using new redirect() call, so the back button on the web browser is usable again
- New module management, including progress of downloads
- Added ability to 'lock' a trunk to only use a specified CID ('KEEPCID' patches)
- Add support for Hebrew (RTL) text formatting
- dialparties.agi now written in PHP
- Went rummaging around through the old sourceforge forums and found some patches
  that had been lost in the move 
- FOP now using the latest version, .26
- Huge number (200+) of minor bug fixes
- Policy change with relation to releases. There is now a 'base' and a 'withmodules'
  package. The 'withmodules' pack is useful for machine that don't have easy internet
   access, and contains all the modules currently available at the time of the release. 
  This is also useful for new installations, too.
- Changed default '#' and '*' features (transfer and disconnect) to '##' and 
  '**'. Note, bugs in asterisk prior to 1.2.13 cause this to misbehave.

*KNOWN ISSUES*

CID Lookup and Pinsets (both modules that hook into Inbound Routes) do not display their changes immediately. After
you change a CID or Pinset, click on the route _again_ - the changes will then be displayed. This is due to how the
old module hooks were being processed, and isn't easily fixable. 

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