We have started the FreePBX 2.5.0 Beta program for early testers by providing the 2.5.0alpha1 tarball available for immediate download. We expect to move it to full beta as soon as we get some initial feedback on the alpha tarball although testing has show it to be quite stable. To see details of what has been put in the release, you can go to: http://freepbx.org/trac/milestone/2.5 and from there, you can see the overview and drill down into the bugs and tickets that have been addressed.
We are proud to announce FreePBX Version 2.3.0 available for immediate download. We believe this latest release is the most exciting and stable release of FreePBX ever and have allowed for an extended Beta program to maximize the testing efforts before final release. In the 9 weeks since the Beta Program was officially launched we have had over 3900 active installs estimated at well over 1 Million hours of testing! That is probably more beta sites then other commercial Asterisk based telephony systems have for their entire installed base! We have closed over 250 bugs that were reported on 2.3 or previous releases. Most of these were existing bugs on FreePBX 2.2 and prior releases making 2.3 the production version of choice.... read more
I apologise for the delay, but I've finally managed to get the announcement out on SourceForge. Most of the news announcements are now on the WordPress Blog on http://www.freepbx.org, and the best place to catch up with the latest news is there.
For those that haven't been reading, 2.2.0 has a new look, a pile of new modules, and over 200 bug fixes. There is a minor problem with ENUM lookup, and a single-line fix is available on the Wordpress Blog, if you do need to use it.... read more
This is our first Release Candidate for the 2.2 branch. As we've done a huge amount of changes since 2.1, we have had 3 betas, and we think we've got all the bugs licked. All the important issues are fixed, as far as we know, and everything seems to work properly now. This is the first release with a reasonably accurate Changelog, too!
** 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 in the current layout. We expect this to be resolved in the 2.3 branch.... read more
Huge number of changes - CHANGES file hasn't been updated in the release, but just download it and see. I'll update it fully prior to the release. The most visible difference is the great new facelift!
FreePBX 2.1.2 has just been released. This release contains a number of security related bugfixes, and we encourage all users to upgrade to 2.1.2 as soon as possible. It also resolves the the 'SSL' errors that were occuring in the Online Module section, even though they were harmless, they were very annoying!
Work is still going ahead full steam for a 2.2 release, as soon as possible!
Just a couple of minor fixups for core stuff that we can't fix with online modules:
- 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)
FreePBX 2.1 is now available.
This release features a new module admin that allows for downloading/installing/upgrading modules from a online repository. A fix to a module no longer requires upgrading the core freepbx, just "click" "click" and your up to date!.
- 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 ... read more
The Asterisk Management Portal (AMP) is now known as FreePBX.
FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces!
This exciting new release boasts a better user experience, additional functionality, and a new module system.
The module system is designed to be simple, powerful, and easy to use existing code with. It only imposes a minimal API, with just a few requirements to make it work. Please see the project wiki for more information (http://freepbx.org).... read more
We are pleased to announce the latest release of Asterisk Management
Portal (AMP 1.10.010). This version includes support for Asterisk 1.2,
an improved Device/User implementation, improved support for phones with
Busy Lamp Indicators (ie: HINT extensions), improved ARI, as well as
several bug fixes.
More information about AMP can be found at:
Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below).
The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find
links to the download, install guide, and documentation wiki.
As usual, please use amportal-users mailing list for discussions about AMP: https://sourceforge.net/mail/?group_id=121515
AMP 1.10.009 changes:... read more
Asterisk Management Portal 1.10.008 has now been released.
Please visit the AMP homepage for links to Downloads, Support, Documentation, and Trackers.
- 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() (the * prefix is configurable) ... read more
A couple of errors surfaced immediately following the release of 007. 1.10.007a corrects issues with Digital Receptionist not remembering past settings, and with possible permissions problems on Asterisk configuration scripts.
The "Secret Agent" final release of the Asterisk Management Portal is now available for download:
This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback!
- Added AMP Users (multi-department, basic 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)
- New Outbound Routes page to control trunks used for outbound calls based on dial patterns
- LCR using Outbound Routes
- Trunks page 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
- Added cvs2cl generated ChangeLog (see this for all changes and bug fixes)
New release of the Asterisk Management Portal:
- 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
We've just posted a new version of AMP to address some of the issues that have come up since the last release.
- 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
- Don't allow an extension number to be changed in Extension admin
(force delete/re-create extension)
- Fix counter bug in Digital Receptionist admin
- 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 Routes"
- Corrected fax-detect not working for call-groups
To satisfy the requests for an easier installation document, we have created a "newbie" installation manual.
This guide details every step necessary to get AMP operational on WhiteBox Enterprise Linux.
Available at http://amp.voxbox.ca
The Asterisk Management Portal has been updated to accommodate IAX clients, as well as SIP clients.
In addition, the included version of Flash Operator Panel has been updated to the latest 0.17.
A script has been included to streamline the upgrade on existing AMP installs.
AMP's INSTALL instructions and a few php files have been updated to make life easier for Debian users of AMP. Changes are in the CVS only.
If you've already got AMP installed and working, simply pull the latest CVS and run apply_conf.sh.