root/freepbx/trunk/CHANGES

Revision 3223, 8.8 kB (checked in by qldrob, 7 years ago)

Uncomment '##' attended transfer in features.conf, to avoid confusion when trying to send # or * to a caller.

  • Property svn:eol-style set to native
  • Property svn:keywords set to Author Date Id Revision
Line 
1 2.2
2 - New Modules: phpagiconf, printextensions, customerdb, inventorydb, announcement, blacklist,
3   cidlookup, customerdb, dictate, inventorydb, parking, pbdirectory, phonebook, printextensions,
4   speeddials, ZoIP
5 - New option in amportal.conf for remote backups (as well as significant backup fixes)
6 - Changed Call Recordings to user MixMontior, better performance and more reliable.
7 - Fixed prefix lookup to use localcallingguide.com XML interface
8 - Fix potential security hole in CALLERID(name) and CALLERID(num) (see r2076)
9 - Redo front end with the new look, Thanks to Steven Fischer for the template
10 - Using new redirect() call, so the back button on the web browser is usable again
11 - New module management, including progress of downloads
12 - Added ability to 'lock' a trunk to only use a specified CID ('KEEPCID' patches)
13 - Add support for Hebrew (RTL) text formatting
14 - dialparties.agi now written in PHP
15 - Went rummaging around through the old sourceforge forums and found some patches
16   that had been lost in the move
17 - FOP now using the latest version, .26
18 - Huge number (200+) of minor bug fixes
19 - Policy change with relation to releases. There is now a 'base' and a 'withmodules'
20   package. The 'withmodules' pack is useful for machine that don't have easy internet
21    access, and contains all the modules currently available at the time of the release.
22   This is also useful for new installations, too.
23 - Changed default '#' and '*' features (transfer and disconnect) to '##' and
24   '**'. Note, bugs in asterisk prior to 1.2.13 cause this to misbehave.
25
26 *KNOWN ISSUES*
27
28 CID Lookup and Pinsets (both modules that hook into Inbound Routes) do not display their changes immediately. After
29 you change a CID or Pinset, click on the route _again_ - the changes will then be displayed. This is due to how the
30 old module hooks were being processed, and isn't easily fixable.
31
32 2.1.1
33 - Rob Thomas (xrobau@gmail.com) takes over stewardship of freePBX project from Coalescent Systems
34 - Clean up harmless warnings in recordingcheck (r1927 and r1940)
35 - SIP Anonymous wasn't working when language was not set to 'en' (r1932)
36 - Fixed unfortunate loop when more than 10 trunks defined (r1942)
37 - Voicemail changes weren't immediately visible (r1945)
38 - Various fixes to clean up parser errors in extensions.conf, and rewrite of a couple of macros (r1949, r1950, r1953, r1957)
39 - Various minor text cleanups (r1960, r1962)
40 - Show fatal error message when cannot read /etc/amportal.conf file (r1971)
41 - Add simple script for A@H users to restore their non-standard modules (r1972)
42
43 2.1
44
45 - Modules not packacked with FreePBX
46 - Included interface used to download/install/upgrade modules
47 - Inbound Routing based on (analog) zap channel (ie: no DID available)
48 - Russian and Portuguese
49 - ModuleHooks system allows modules to interact with eachother
50 - dialparties completely re-written in PHP - eliminating dep for asterisk-perl
51 - General Option to allow unauthenticated SIP calls into the system
52 - Define different "Dial()" options for outbound calls
53 - Direct DID->Extension config
54 - New modules, including FeatureCodes, Callback, PinSets, and others
55
56 2.0
57
58 - AMP is now "freePBX"
59 - New module system allows for drop-in functionality
60 - Requires Asterisk 1.2.x
61 - All previous AMP functionality ported to new module system
62 - Added Modules: Conferences, Time Conditions, Asterisk CLI, Online Support
63 - GUI improvements
64 - FOP .24
65 - ARI 00.08.03 - now with AJAX!
66 - Outbound Routes can now use an Authenticate Password File
67 - Queue Static Agents can have penalties applied
68 - Using native music on hold support - no more mpg123!!
69 - Default is to use freePBX database authentication.  New installs create a new user.
70 - Initial sqlite support!
71 - Much improved form validation for all modules
72 - Inbound routes can set ALERT_INFO variable for SIP devices
73 - Ability to force Emergency Caller ID for devices using an Emergency Outbound Route.
74
75 1.10.010
76
77 - Tested with Asterisk 1.2 (beta)
78 - Tested with PHP 5
79 - Removed all the sound files from AMP archive, instead depend on asterisk-sounds
80 - Ability to execute a script after applying changes in the AMP interface
81   (see amportal.conf in source archive)
82 - Allow accountcode for IAX devices (again)
83 - Show custom extensions in FOP
84 - Allow mailbox setting for device to be set manually (for shared mailboxes)
85 - HINT extensions are now created for both FIXED and ADHOC devices
86 - Display AMP version in footer
87 - Support for remote mysql database
88 - ARI upgrade adds i18n and user settings
89 - Remove Play Next option from voicemail options and default to
90   play next when deleting or saving voicemails
91 - Lots'o'bug fixes
92
93 1.10.009
94
95 - Asterisk Recording Interface (ARI).  ARI is a php interface to Voicemail and monitor recordings. (written by littlejohnconsulting.com)
96 - Queues can now play a "welcome" message to callers upon joining.
97 - DID Routes re-written as Inbound Routing.  This allows for DID specific fax emails and call answering options.
98 - RingGroups now use strategies: Ring All (default), Hunt, Memory Hunt
99 - 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.
100 - Custom device technology support
101 - HINT priorities for FIXED devices
102 - Interface translated to French, German, Italian, Spanish
103 - FOP .21
104 - FOP button layout can now be sorted by last name or extension number
105
106 1.10.008
107
108 - Backup/Restore (schedule and restore backups)
109 - Extension Call Recording (inbound and outbound calls)
110 - Queue Call Recording (inbound to agents)
111 - Custom Trunks (use any Asterisk supported technology as a trunk)
112 - Remote Agents (join a Queue from any endpoint on a trunk)
113 - Outbound Route Password (require a password for certain outbound patterns)
114 - i18n (web interface can now be translated)
115 - ZAP trunk channels no longer hard-coded in retrieve_op_conf_from_mysql.pl
116 - *<exten> dials direct to voicemail()
117
118 1.10.007
119
120 - Added cvs2cl generated ChangeLog (see this for all changes and bug fixes)
121 - Added AMP Users (multi-department, multi-tenant)
122 - Added incremental upgrade script (install_amp)
123 - Use /etc/amportal.conf to tweak AMP to your environement (MySql credentials, web root, ip address, etc).  Apply changes with apply_conf.sh
124 - New Outbound Routes page to control trunks used for outbound calls based on dial patterns
125 - LCR using Outbound Routes
126 - Trunks page redone to support routing, adds dial rules to modify numbers per-trunk before dialing
127 - ENUM Trunks
128 - Queues support added
129 - Support for ZAP extensions
130 - More voicemail options added
131 - New AGI-based directory application to support both first and last name lookups and return to operator
132 - provide customization points for all AMP generated extension contexts.
133 - Upgrade to Flash Operator Panel 0.20
134 - Upgrade Asterisk-Stat to v2.0
135
136
137 1.10.006
138
139 - Use extensions_custom.conf for customizations.  Sample included.
140 - Add option to define outbound CallerID on trunks
141 - Add option to define outbound CallerID for extensions
142 - Create extensions without voicemail and directory
143 - 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
144 - Add Call-Forward on Busy application (enable: *90<destination>, disable: *91)
145 - Upgrade FOP to 0.19.  AMP now writes out op_buttons_additional.conf
146 - Include AMP version on admin welcome page
147 - Rework extensions admin
148 - Add 'allow','disallow' settings for SIP and IAX extensions
149 - Add 'pickupgroup','callgroup' settings for SIP extensions
150 - Digital Receptionist voice menus can now be named
151 - Allow custom goto for Call Groups
152 - Digital Receptionist wizard check for proper format on custom goto
153 - Fixed bug which limited AMP to 10 Digital Receptionist menus
154 - Default outbound numbers now dial via a macro
155 - Increase verbosity of mysql connection errors
156 - Fixed upload wav for Ditial Receptionist
157 - Fix Trunks admin so that it writes FOP config
158
159 1.10.005
160
161 - Add "Advanced Edit" qualify= option for NEWLY created extensions
162 - Add support for custom applications in Digital Receptionist admin
163 - Prevent creation of multiple DIALOUTIDS variables in Trunks admin
164 - Allow for long 'register' sting in Trunks admin (for new installs only)
165 - Don't allow an extension number to be changed in Extension admin (force delete/re-create extension)
166 - Fix counter bug in Digital Receptionist admin
167
168 1.10.004
169
170 - Added Call Group CID Name prefixing
171 - Renamed parking.conf to features.conf
172 - Added condition to dialparties.agi that prevents potential pinning of the CPU
173 - Allow Digital Receptionist voice recordings to be uploaded in AMP admin
174 - Added new AMP logo
175 - Added AMP process control script "amportal"
176 - Write meetme configuration for IAX and SIP extensions
177 - Added IAX2 and SIP trunking
178 - Added "DID Routing"
179
180 1.10.003
181
182 - Added support for IAX clients
183 - Upgraded to FOP 0.17
Note: See TracBrowser for help on using the browser.