1. Freepbx Flash Operator Panel
2. Install Flash Operator Panel Asterisk Meaning
3. Flash Operator Panel
4. Freepbx Operator Panel

Install Flash Operator Panel Asterisk. There are two basic types of three way points. One has three frog wires and the more. The frog polarity wiring is shown below for both. The frog polarity switching and wiring is the same for dc as it is for DCC. Some 3 way points are supplied with two wires from the frogs as shown in the top. If so how can admins monitor their systems without Flash Operator Panel? Bruce_ferguson 2012-11-08 16:22:02 UTC #2 I am far from an expert here, but we have tried FOP2 which you may like. Install Flash Operator Panel Asterisk Determine the maximum capacity of an Asterisk PBXSometimes, determining of the maximum processing capacity of an Asterisk server is a mandatory requirement. In most cases, the maximum processing capacity signifies the maximum number of calls that a certain server (in a specific hardware and software configuration) can support. The Flash Operator Panel was the first truly multiplatform realtime display for a PBX enabling drag&drop transfers and actions and it is the only one GPL'd. The project has more than two years in existance. I have put countless hours of work. Please consider making a donation or hiring me for custom work, sponsored development, etc.

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2.9.0 (Highlights)

* see overview and full list of tickets available at:

http://www.freepbx.org/trac/milestone/2.9

2.8.0 (Highlights)

* see overview and full list of tickets available at:

http://www.freepbx.org/trac/milestone/2.8

2.7.0 (Highlights)

* FAX module changes to support FFA and change the way FAX detection works

* Different CID Options for Follow-Me Module

* Different CID Options for Ring Group Module

* Some enhanced functionality in Queues and improved dynamic agent abilities

* Setting Penalties for Dynamic Agents

* Restricting a queue to only specific dynamic agents

* Advanced mode to specify static devices vs. extensions

* Some improvements to Backup

* per backup set FTP and SCP options for remote storage of backup sets

* per session additional directories to backup (and restore if needed)

* Language option for incoming routes

* Increased handling of HANGUPCAUSE codes

* Outbound Route Specific CIDs

* Force Trunk CIDs and remove CNAM option on trunks

* CF Unconditional add support for DEVSTATE

* per device hints created with BLF support

* toggle option created designed to work with BLF

* BEEPONLY support added to minimize messages played

* Advanced Outbound Route Selection

* allows routes to be chosen based on dialed number and CID/extension number or pattern

* Add MoH Class choice for Conferences

* Allow MoH directory to be specified in amportal.conf

* Add ability for Module Admin to reinstall the same version or and 'older' version (with many caveats)

* Move all of 'recordingcheck' AGI script into dialplan

* Add optional and experimental 'macro-dial-one' that can be used to replace 'macro-dial' for single

extension only dialing (no ringgroups, followme, etc.). Requires special setup, see: #4068.

2.6

- Added Extended Repository to allow more contributed modules not part of main

project, some extended modules include:

- Bulk Extension Add/Delete/Edit

- Voicemail Admin

- Set CID

- Route Permissions

- Moved the following modules to the extended repository:

- Customer DB

- Inventroy DB

- Gabcast

- Added new modules:

- Asterisk SIP Settings

- Asterisk IAX Settings

- Outbound Route Messages

- Phone Restart

- Weak Password Checks (back ported to 2.5 also)

- Several Enhancements to Queue Module

- Enhancements to Print Extensions

- Performance Enhancements to Paging (helps large page groups)

- Added Virtual Extension support

- Added Pinless Dialing exception to Extension/User GUI

- More improvemenmts to Directed Call Pickup for Asterisk 1.4+ systems

- New version of mindTerm (used in Java SSH module); has new licensing

options (and restrictions). See

http://www.appgate.com/index/products/mindterm/ for more info.

- Added fields for Publisher and License in module.xml

- Added ability to put dependencies on PHP versions and PHP components in

module.xml

- Changed database mode passwords form clear text to encrypted passwords

- Changed internal schema of trunks to add proper sql tables

- Eliminated dialparties.agi accessing AMI when EXTENSION_STATE() is avail

2.5.1

- Biggest changes from 2.5.0 to 2.5.1 were many loose ends to handle localization

translations through out the code.

- Added support to recognize Asterisk Business Edition versions and work properly

as if they were 1.2 or 1.4.

2.5.0 Added in final

- When using database mode there is a new option to allow a non-admin user to Add

extensions or devices. By default they can not add which means users who previously

existed will need to have the additional permission added to them if you want them

to be able to add extensions or devices. They can still edit existing ones.

2.5.0 Added in rc2

- Add queue weights setting and autfill setting per queue. Set persistentmember=yes

in queues general section to apply to all queues.

- Added ability in IVR to have voicemail system return calls to the IVR after leaving

or checking messages as well as returning to the IVR if line is busy (and user has

not voicemail)

- Added option to incoming routes allowing a CID only route to take priority over a

DID only route. This means that the CID route will route the call for calls that

come to that DID with the specified CID. Default behavior would always route the

call to the DID only route based on how Asterisk sorts routes.

- Split the framework 'module' into framework, fw_fop and fw_ari so that FOP and

ARI updates could be split from other framework updates in order to allow people

with highly customized FOP and ARI changes to pull framework updates easier.

- Added Streaming categories to MoH in addition to downloaded files

2.5.0 Added before rc1

WARNING: The separation of directdid and other incoming routes has been removed.

this has resulted in the obsoletion of the following API call:

function core_directdid_list()

function core_users_directdid_get($directdid=')

These API calls will now always return empty arrays. You should use the

core_did_list() and core_did_get() function calls in their place. See the source

code for specifics about these calls.

WARNING: MoH has been changed to convert MP3 into WAV format using mpg123 and

sox. If you do not have one or both of these installed you should install them.

You can revert to the previous behavior by setting: AMPMPG123=false in the

amportal.conf file.

WARNING: If testing with sqlite3 prior to rc2, you will have to change the field

size for the globals table as there is no conversion script in the upgrades directory

since sqlite3 is a pain to do such schema changes and there is no existing installed

base to convert.

AMPORTAL CONF NEW SETTINGS:

USEDEVSTATE = true false

DEFAULT VALUE: false

If this is set, it assumes that you are running Asterisk 1.4 or higher and want

to take advantage of the func_devstate.c backport available from Asterisk 1.6

which allows custom hints to be created to support BLF for server side feature

codes such as daynight, followme, etc.

MODULEADMINWGET=true false

DEFAULT VALUE: false

Module Admin normally tries to get its online information through direct file

open calls to URLs that go back to the freepbx.org server. If it fails, typically

because of content filters in firewalls that don't like the way PHP formats the

requests, the code will fall back and try a wget to pull the information. This

will often solve the problem. However, in such environemnts there can be a

significant timeout before the failed file open calls to the URLs return and

there are often 2-3 of these that occur. Setting this value will force FreePBX

to avoid the attempt to open the URL and go straight to the wget calls.

AMPDISABLELOG=true false

DEFAULT VALUE: true

Whether or not to invoke the freepbx log facility

AMPSYSLOGLEVEL=LOG_EMERG LOG_ALERT LOG_CRIT LOG_ERR LOG_WARNING LOG_NOTICE

LOG_INFO LOG_DEBUG LOG_SQL SQL

DEFAULT VALUE: LOG_ERR

Where to log if enabled, SQL, LOG_SQL logs to old MySQL table, others are passed

to syslog system to determine where to log

AMPENABLEDEVELDEBUG=true false

DEFAULT VALUE: false

Whether or not to include log messages marked as 'devel-debug' in the log system

AMPMPG123=true false

DEFAULT VALUE: true

When set to false, the old MoH behavior is adopted where MP3 files can be loaded

and WAV files converted to MP3 The new default behavior assumes you have mpg123

loaded as well as sox and will convert MP3 files to WAV. This is highly recommended

as MP3 files heavily tax the system and can cause instability on a busy phone system.

AMPVMUMASK

DEFAULT VALUE: 077

Allows setting a umask for Asterisk to control the voicemail file permissions

Special Case configuration variables for the CDR reports to pull data from remote

databases:

CDRDBHOST: hostname of db server if not the same as AMPDBHOST

CDRDBPORT: Port number for db host

CDRDBUSER: username to connect to db with if its not the same as AMPDBUSER

CDRDBPASS: password for connecting to db if its not the same as AMPDBPASS

CDRDBNAME: name of database used for cdr records

CDRDBTYPE: mysql or postgres mysql is default

CDRDBTABLENAME: Name of the table in the db where the cdr is stored cdr is default

DASHBOARD_STATS_UPDATE_TIME=integer_seconds

DEFAULT VALUE: 6

DASHBOARD_INFO_UPDATE_TIME=integer_seconds

DEFAULT VALUE: 20

These can be used to change the refresh rate of the System Status Panel. Most of

the stats are updated based on the STATS interval but a few items are checked

less frequently (such as Astersisk Uptime) based on the INFO value

ZAP2DAHDICOMPAT=true false

DEFAULT VALUE: false

If set to true, FreePBX will check if you have chan_dadhi installed. If so, it will

automatically use all your ZAP configuration settings (devices and trunks) and

silently convert them, under the covers, to DAHDI so no changes are needed. The

GUI will continue to refer to these as ZAP but it will use the proper DAHDI channels.

This will also keep Zap Channel DIDs working.

HIGHLIGHTS:

A detailed list of changes is available on the 2.5 Mileston:

http://freepbx.org/trac/milestone/2.5

Where you can review the summmary as well as the link to all tickets associated

with this Milestone.

- New module queueprio that allows priorities to be assigned to callers that will

effect their position in any queue they drop into.

- New module dundicheck, allows the extension registry to detect duplicate

extension conflicts between DUNDi branch systems. Also provides a simple lookup

for extensions on the configured cluster.

- Timecondition module changed with the addition of Time Groups to allow multiple

times to be considered in a single timecondition. The timegroups are abstracted

and available for other modules to take advantage of in the future. This was

a merge of the timegroups module in the contributed modules directory.

- Day/Night Mode module modified to hook into Time Conditions and allow any Time

Condtion to be directly linked to the stated of a Day/Night mode feature code.

This avoids the need for adding a Day/Night mode module into the call flow and

allows a single Day/Night mode module to change multiple Time Conditions at once.

- Direct DIDs have been merged with incoming routes. Any incoming route that goes

to an extension/user will appear under that user. New directdids can be added

on the user screen but all detailed configuration of that did must be configured

on its corresponding incoming route page. Conenient links are introduced to

navigate between a user/extension and the incoming routes quickly. Filters have

also been introduced on the incoming routes page to see directdids only, all but

direct dids only, or unassigned dids (with no destinations). Unassigned dids are

not generated in the dialplan. (So if there is a catchall defined they will end

there instead of a hangup because of the lack of a destination.

- Users page (only viewable in devicesandusers mode) now has links to each fixed

device as well as each adhoc device who's default user is this user. And the

Device page has a direct link back to the fixed or default user if specified.

- Introduced the optional usage of BLF on many feature codes. This requires the

inclusion of the Asterisk function func_devstate.c which is backported from

Asterisk 1.6 but available on Asterisk 1.4 and has been stable for a long time.

By setting the value 'USEDEVSTATE=true' in amportal.conf, the dialplan will be

generated to take advantage of this. This allows functions like DND, Day/Night,

Follow-Me, Meetme and others to have BLF settings so phone buttons can recognize

the states.

- Follow-Me feature code added to enable/disable Follow-Me as is available in

the FreePBX GUI or ARI.

- Caller screening configurable per user for external calls, requiring a caller

to announce themselves and then providing the called user the option of

listening to who the announced caller is and choosing whether or not to take

the call, with options to send to voicemail, or other alternatives.

- System Recordings has been enhanced so that recordings can have a dedicated

feature code assigned to them that allows them to re-record the specific recording.

Recordings that use built-in recordings or that are constructed from multiple

concatenated recording segments can not have a feature code created. This allows

a customer to easily modify a recording that may be associated with an IVR (or

anything else) without having to do anything with the GUI.

- Queues have been modified with an optional filter to control what dynamic agent

callback numbers are acceptable to be entered when a user logs in. This is done

through the introduction of an optioal REGEX filter for each queue. This can

allow a queue to be limited to a range of extensions, block external numbers, or

any other filtering that can be expressed through a regex expression to test

the validity of the entered agent number.

Also added a CID prepend option to add the Queue Wait time for a caller to be

presneted to the agent when ringing their phone.

- Delete and Add icons have been added to many of the links on most modules that use

links instead of buttons for these actions.

- Optional Module Admin configuration file has been added, freepbx_module_admin.conf,

that allows any module to be filtered out of the Module Admin GUI.

- Module Admin Changelog displays have added auto-generated links to referenced

tickets or changesets.

- Module Admin has been modified to fall back to using wget if it can't reach the

online server through direct file read commands that sometimes get blocked by

firewall content filters.

- Optional Feature Codes configuration file has been added, freepbx_featurecodes.conf,

that allows the default values normally hardcoded by each module to be specified.

These default values can still be overridden in the Feature Code panel as usual.

- We have tried to introduce logical 'tabindex' settings to all the pages so that

tabbing through a form logically progresses through the fields as one might hope.

- Paging & Intercom control beep and more

- Skip Busy Agents feature has been added to Ring Groups (was on Queues), as well

as Ignore CF Settings, allowing a Ring Group to ignore and block any agent's CF

settings (CF, CFU, CFB) whether they are server or device side settings.

- Added VmX Locater GUI to FreePBX so admin and user can make changes, also enabled

0 option even with VmX disabled so it can be used by admin to redirect 0 out on

voicemail without requiring VmX to the user.

- IVR enhanced to allow the annoucement message to be changed in the event of a

timeout or ivalid extension chosen.

- Throughout the modules all references to system recordings by a module are done so

with an id so that recording changes are reflected with a relad.

- Sqlite3 support has been added.

2.4.1

Mainly a maintenance release that is all available through the Framework update, the

bugs addressed are listed below as per the Framework Changelog. The biggest change

is with FOP that had included the newest version of FOP in order to accomdate the

incompatability with Flash Player 9.0.124.0 and higher.

2.4.0.1: #2843, #2701, #2818, #2784, #2604, #2766, #2798, #2809, #2799, #2685, #2676

2.4.1.0: #2862, #2855, #2782 FOP update to make flash player 9.0.124.0 and newer happy

2.4.0

WARNING: changes were made to some of the core_did_XXXX() API calls that could effect

any custom applications that were depending on these.

WARNING: changes were made to context ordering wrt to ext-did-catchall and

from-did-direct. Previously, if you had not ext-did-catchall you might be in a

situation where you were reveiving direct DID calls to your extensions even though

not configured since there was no catchall route. If you then made a catchall route

you would suddenly stop receiving those calls and would have to add the dids in a

route or as a direct did. With this change, it is now deterministic but the behavior

of an existing system could change (they could suddenly start receiving DIDs). This

can be easily corrected though by intercepting those DIDs with an inbound route (with

pattern matching if need be).

- Implementation of a distributed Extension and Destination Registry through callbacks

in all modules and supporting APIs in framework. The Extension Registry provides the

needed information and APIs to detect and allow a module to block the creation of an

extension number that is used elsewhere. The Destination Registry provides a

mechanism for a module to detrmine if any of it's entities are being used as a

destination by other modules so it can provide warnings or feedback about the impact

of deleting such entities. Both registries are checked when reloading a configuration

and any inegrity issues are supplied to the notification panel. All supported modules

should be instrumented to use these once updated.

- Addition of Custom Applications Module. Provides a place to register custom extension

numbers as well as custom destinations that are to be used in FreePBX. Replaces the

old Custom Destinations choice that was available in each module.

- Moved vmblast form contributed modules to supported module after significant changes

and fixes as it never worked form the original contributor. Add additional features

to it and added a default vmblast group option to be used with extensions/user add

and edit.

- Custom destinations will no longer show up under the destination selections unless there

is already one configured or an unknown destination is detected (which are one and the

same). To use a custom destination in FreePBX, it will have to be registered with this

module to appear as a choice to other modules. (Similar to adding a destination to the

Misc Dests module).

- Module admin changed so that 'problem' modules that have dependency issues will not

block other modules from being downloaded and/or installed. A warning is still generated

but the action is allowed to proceed with any modules that have all their dependencies

met.

- Removed Channel Routing from 'Inbound Routes.' Added 'Zap Channel DIDs' to core modules

to assign DIDs to Zap Channels which can then use 'Inbound Routes' to route them with

all the same flexibility that is there today and without some of the issues that the

previous Channel routing implementation provided. Existing Channel routes will be

converted and entries inserted into the 'Zap Channel DIDs' tables.

- Ringgroups, Queues and Follow-Me have been enhanced with a Quick Pick utilitlity that

allows extensions to be added into the the ring list.

- Several changes and enhancements have been made to improve the usability of Users/Devices

mode particularyly around Adhoc devices. Some highlights:

- Default user information is retained and the device returned to that user upon a logout

- Editing devices in FreePBX will no longer erase current logged in device information

- Hints are initially generated properly for Adhoc devices

- Hints are dynamically added/deleted as part of the logon/logoff process

- There are still issues if reloading from the CLI. A script and some instructions will

be supplied on ways to address this until a more permanent solution can be determined.

- Pulled some agi scripts and macro calls out of dialout-trunk / dialout-enum into the outbound

route code so they would only be called once when the call sequence has to try multiple

trunks.

- Added reload option to CLI module_admin to peform same task as the reload bar.

- Added support in macro-user-callerid to support per-user/extension language changes.

- Significant changes within Paging & Intercom Module for 2.4 version of Module. Highlights:

- Intercom works properly when User is logged into multiple devices and will intercom them all

- Explicit Allow and Deny options to control who can/can't intercom you

- AstDB flag that can be set for a specific extension to block it from intercoming anyone

- designate a group as default for add/edit at extension/device creation/edit time

- Significant improvments in Auto-Answer ability for more phone support:

- Defaults pulled from database which can be changed by an advanced user

- Defaults can be overode for specific phone useragents based on information in

database, for advanced users and to allow new phones to be supported once details

are reported to the FreePBX team.

- Abilility to trigger custom macros for phones based on useragent info or on a per-device

basis with information stored in AstDB for that device, for advanced users.

- Queues Module has been updated to remove its dependency from the old legacy extensions table

and the current queues table is replaced with queues_config and queues_details table.

- Queues and the SIP, IAX2 and ZAP conf file generation has been replaced with proper queues_conf

and core_conf classes

- Added partial support for DUNDi via a DUNDi trunk, dundi.conf configuration is still manual

- Support Asterisk 1.6 to the extent that it can be supported as it is in beta at the time of

2.4 release. But we will try to keep on top of 1.6 issues.

- Misc other bug fixes and some feature requests that can be obtained through the SVN log.

2.3.1

- Module Admin previously exploded new module tarball updates ontop of the existing earlier

versions. It has been changed to replace the entire module directory with the new tarball

contents. Removed files as well as any other files in the directory will be removed.

- #2335 Module Admin can now be disabled in database mode.

- module_admin (cli version) has new reload option (same as pressing orange bar)

- FOPRUN now defaults to true in amportal.conf for new installs

- retrieve_conf will look for and include a script called retrieve_conf_post_custom if found

in AMPLOCALBIN which must be defined in amportal.conf. This can be used for custom actions

and configuration upon reloads after dialpans and conf files have been generated.

- macro-dialout-trunk has been enhanced with a call to macro-dialout-trunk-predial-hook that

can be used to make custom changes (e.g. add SIP header information) or even bypass the trunk

if a macro is defined by the user.

- #2412 fixed by r5096 was creating javascript validation in several modules to fail

- apply_conf.sh improved to handle all password formats and manager user login name changes

2.3.0

- Final release is almost all bug fixes, see change logs in framework

- Changed several categories

- Linked Help tab into online freepbx.org help system

Added in Beta2:

- WARNING:

amportal has been changed to call freepbx_engine so that the framework can update that

script if necessary. fop_start, fop_stop and fop_restart has also been added to freepbx_engine

as new commands. If you are upgrading through install_amp then you will receive all these

changes. If you were beta testing during Beta1 and were upgraded through the Framework updgade

you will have to manually update the amportal script that lives under /usr/sbin normally,

or run an install_amp upgrade. You can do this by changing to root and copying the file from

amp_conf/sbin/amportal to /usr/sbin/amportal or where ever your AMPBIN directory is located.

- WARNING:

ARI split out into several modules. There may be some old ARI modules that are left over since

the install script does not to delete the previous modules if they are still there. You can

look in the install directory under amp_conf/htdocs/recordings/modules to see what gets packaged

with the install. You can safetly remove any modules not listed there from the install

directory, typically /var/www/html/recordings/modules is where they would be.

- New Dashboard Index page - shows notifications from the system and vital system statistics

- New Logos and styling

- FOP 0.27 upgrade

- Added CID prefix and description to inbound routes

- Added CW enable/disable to core extensions/users

- Segregated ARI into multiple ARI modules and added CW and DND.

- Removed followme destinations, and changed Core destinations to Extensions, Voicemail and

Terminate Call. Extensions will go to followme if enabled and present consistent with normal

dialing behavior. Voicemail can choose busy or unavail. Terminate call included the Hangup and

related core destinations.

- New notification framework added to allow all notifications and errors to be consolidated

and used by different systems like the dashboard.

- New crontab manager added to allow modules to install crontab type entries run by the manager.

Checks hourly and modules can indicate how frequently they want something run. Initially created for

online update checking.

- Automatic Online Update checks with notification through the dashboard or email.

- Framework updates modified to handle full upgrades using the same upgrades directory to

apply schema changes. Shared by install_amp.

- FOP upgrading added to Framework

- New FOPRUN variable in amportal.conf, if set to false, FOP won't be started by default

- Added support in amportal.conf for AMPMANAGERPROXY, to use if running with a astmanproxy style proxy

- libfreepbx.install.php split out from install_amp to consolidate common functions needed by framework

- version array removed from install_amp upgrade script, it will now derive the version from the last

upgrade direcotry and use the upgrade directories to run though the installs.

- added 'hidden' make-links-devel flag to install_amp, to install with symbolic links if running

out of an svn tree

- retrieve_conf instrumented to provide notifications to the dashboard on failures

- fixed several dependency logic bugs in the online module infastructure

- improved the amportal.conf parser and modified retrieve_conf to use the main parser

Added in Beta1:

- To Get Full Details - look at the SVN logs of changes since the previous

release. These are only higlights.

- WARNING:

Removed Follow-Me destinations and changed how 'Core Extension' destinations

work. This has been an area of confusion and inconsistency. Under all calling

conditions, if you call someone and they have an enabled Follow-Me, that is

where the call goes. If not, it goes to their extension. Now the Core destination

of an extension works the same way. There is no longer a Follow-Me destination

to choose from. All settings should be migrated automatically.

- WARNING:

Changed default behavior of Call Waiting state when extensions are created. It is

now enabled by default (r4175) and you must set ENABLECW=no to keep the preivous

behavior

- MOVED CORE MODULES to the module repository, meaning they can now be updated online

like other modules.

- ADDED Framework Module, which provides a facility to update all the rest of FreePBX

through the Online Module Admin System

- VmX Locater and its intergration with FollowMe. This is a new feature that allows

each VoiceMail extension to have the option of having a 'personal' IVR so the caller

can have choices like call them on their cell, optionally try their Follow-Me (which

can otherwise be disabled), etc. You check the box down with Voicemail and then

the user controls the rest from the ARI.

- Added enable/disable of Follow-Me without having to delete. In disabled mode, VmX

can still send calls to Follow-Me.

- ARI control of Follow-Me, VmX Locater, all CF modes, and a handful of other

ARI bugs addressed. (ARI is still in EOL mode - but since no user portal is ready

yet, it still servers as a user interface).

- Inbound MoH classes based on DID routing or Direct DID routing.

- Outbound MoH clases based on the outbound route selected.

- New ring strategies for FollowMe and Ringgroups (ringallv2, firstavalable, firstonphone)

- Per-Extension Ring Times to override the global setting in General

- Sipname alias (that can be non-numeric) to provide user friendly sip dialing

information if you accept annonymous sip calls.

- Internal calling CID Number Masquerading, to allow your internal extension appear

as a different number when making internal calls. (For example, a support team can

all masquerade with the number of a queue so that people who call them back call the

queue instead of their personal extension.

- CWINUSEBUSY=yes/no in amportal.conf, to have calls to an occupied multi-line phone (or

CW enabled phone) end up in the voicemail busy greeting instead of the unavailable

greeting.

- Asterisk 1.4 support

- Sqlite3 support (deprecate sqlite2)

- Day/Night Control Module

- Recording Module with playback ability

- Re-Introduced bad-number context, to play friendly message if a bad number is dialed. Added

from-internal-xfer as TRANSFER_CONTEXT in global variables. This keeps the previous error

of transfering a user to a bad number and dropping the transfered user into the bad-number

context.

2.2.3

- #2025 fix bug that blocks the editing of an extension that has a directdid

with an alert box saying the directdid is already in use.

- #1747 add South Africa indications.

- changed from auto-symlink to auto-copy of agi-bin scripts packaged with a

module. The symlinks create issues on some systems. To keep the coying from

overwriting files in the real agi-bin, make them read only permission to

astersik.

- Fixed several module version dependency checking bugs

- #1841: don't strip '+' from directdid

- added unique unidentifiable tracking id for online system auditing

2.2.2

- To Get Full Details - look at the SVN logs of changes since the previous

release. These are only higlights.

- WARNING:

merge ext-did and ext-did-direct all into ext-did context, and create

new context, ext-did-cacthall when creating an ANY/ANY route. The inclusion

of ext-did-catchall is in the extensions.conf file so if any customizations

have been done, make sure this is included.

The purpose of this change allows directdids specified with the extension

to properly co-exist with those create with inbound routing. In addition,

error checking has been added to keep the same did from being used two places.

However, you can use a did on an extension as a directdid, and then included

the same did+CID info on inbound routing and that is legal, and will now work

properly instead of being ignored as was the case in the past.

- WARNING:

sip.conf file changes that MUST be adopted. The inclusion of sip_registrations.conf

and sip_registrations_custom.conf have been added to sip.conf. In the past the

registrations were put at the very top of sip_additional.conf which made it really

easy to break things if you put a custom sip context into sip_custom.conf.

- javascript warning when users try to use the 'r' option in the

'Asterisk Outbound Dial command options' of the 'General' tab.

- allow the '=' character on the right side of an assignment in the trunk specification

section. This was a common error propblem if a secret included an '=' sign, for

instance. There are other settings that require '=' there also.

- fix bug in ringgroups and followme when DND was enabled on the first extension of a

ringgoup, the others would not be tried. This behavior is correct if the ring

strategy includes the '-prim' postfix but was doing it to all strategies.

- Added Israel and India Indications to General tab

- Added MailBoxExists and WaitExten asterisk commands to extensions.class.php to enable

some bug fixes requiring those commands forthcoming to the IVR and Voicemail modules.

2.2.1

- Fix ENUM lookup bug in 2.2.0 - r3546

- Convert MP3 MOH files to SLN (Asterisk Native) format - r3548

- module_install() now returns true for already installed modules - r3569

- Allow null and blank values to be put into astdb - r3576

- don't propogate dnd behavior and not ring other phones if this was not

a prim mode strategy - r3580

- Apply fix for #1361 (patch #1667), mailbox field not being propogated in in

deviceanduser mode. - r3584

- Fix typo in extensions.conf, when pushing '0' for oper and not having an

opereration extension defined, would pass a bad Dial string. - r3585

- added warning on save of trunk if user context left blank and user details

filled in that details will not be saved #1666 - r3631

- limit rnav width #1647

fixed panel displaying extensions over 9999 as trunks - ticket #1710

List device technology on page when editing Ticket #1711

fixed trunks stripping AMP: which removed ANY occurance of the letters

A,M,P,: from the beginning of all trunks, also unified the display on

the routing page - partially noted in #1713

CFB when dialparties.agi decides not to - offhook, user hits ignore/reject,

etc. - patch #1681 - (Backport from trunk) r3643 naftali5

- now module_admin works even for 'broken' modules, running from every

directory - r3678

- do not display warnings about password when not using mysql/pgsql - r3679

- make the cdr page links a bit nicer - r3689

- fix typo in sip.conf - r3691

- keep rtone from being set in queues_additional.conf #1635 - r3697

- fix queues retrieve conf bug part of #1659 - r3744

2.2

- IMPORTANT CHANGE - Asterisk 'transfer' featurecode is now defaulted to '##', rather than '#'.

This was changed to avoid issues with sending a '#' to an externally called party. Note

that this is a SIGNIFICANT CHANGE, and you should be aware of it.

- 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

• Copy lines
• Copy permalink

This will build a container for FreePBX - A Voice over IP Manager for Asterisk. Upon starting this image it will give you a turn-key PBX system for SIP calling.

• Latest release Version 15
• Compiles and Installs Asterisk 16
• Choice of running embedded database or Modifies to support external MySQL Database and only require one DB.
• Supports Data Persistence
• Fail2Ban installed to block brute force attacks
• Debian Stretch Base w/ Apache2
• NodeJS 11.x
• Automatically Installs User Control Panel and displays at first page
• Option to Install Flash Operator Panel 2
• Customizable FOP and Admin URLs

This Container uses tiredofit/debian:stretch as a base.

If you are presently running this image when it utilized FreePBX 14 andAsterisk 14 and can no longer use your image, please see this post

• Introduction
• Configuration
• Maintenance

This image assumes that you are using a reverse proxy such asjwilder/nginx-proxy and optionally the Let's Encrypt ProxyCompanion @https://github.com/JrCs/docker-letsencrypt-nginx-proxy-companionin order to serve your pages. However, it will run just fine on it's own if you map appropriate ports.

You will also need an external MySQL/MariaDB Container, athough it can use an internally provided service (not recommended).

Automated builds of the image are available on Docker Hub and is the recommended method of installation.

The following image tags are available:

• 15 - Asterisk 16, Freepbx 15 - Debian Stretch (latest build)
• 14 - Asterisk 14, Freepbx 14 - Debian Stretch (latest build)
• latest - Asterisk 16, Freepbx 15 - Debian Stretch (Same as 15)

Freepbx Flash Operator Panel

You can also visit the image tags section on Docker hub to pull a version that follows the CHANGELOG.

• The quickest way to get started is using docker-compose. See the examples folder for a working docker-compose.yml that can be modified for development or production use.

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• Set various environment variables to understand the capabilities of this image.

• Map persistent storage for access to configuration and data files for backup.

• Make networking ports available for public access if necessary

The first boot can take from 3 minutes - 30 minutes depending on your internet connection as there is a considerable amount of downloading to do!

Install Flash Operator Panel Asterisk Meaning

Login to the web server and enter in your admin username, admin password, and email address and start configuring the system!

Flash Operator Panel

Data-Volumes

Freepbx Operator Panel

The container supports data persistence and during Dockerfile Build creates symbolic links for /var/lib/asterisk, /var/spool/asterisk, /home/asterisk, and /etc/asterisk. Upon startup configuration files are copied and generated to support portability.

The following directories are used for configuration and can be mapped for persistent storage.

Directory	Description
/certs	Drop your Certificates here for TLS w/PJSIP / UCP / HTTPd/ FOP
/var/www/html	FreePBX web files
/var/log/	Apache, Asterisk and FreePBX Log Files
/data	Data Persistence for Asterisk and Freepbx and FOP
/assets/custom	OPTIONAL - If you would like to overwrite some files in the container, put them here following the same folder structure for anything underneath the /var/www/html directory

Environment Variables

Along with the Environment Variables from the Base image, below is the complete list of available options that can be used to customize your installation.

Parameter	Description
ADMIN_DIRECTORY	What folder to access admin panel - Default /admin
DB_EMBEDDED	Allows you to use an internally provided MariaDB Server e.g. TRUE or FALSE
DB_HOST	Host or container name of MySQL Server e.g. freepbx-db
DB_PORT	MySQL Port - Default 3306
DB_NAME	MySQL Database name e.g. asterisk
DB_USER	MySQL Username for above Database e.g. asterisk
DB_PASS	MySQL Password for above Database e.g. password
ENABLE_FAIL2BAN	Enable Fail2ban to block the bad guys - Default TRUE
ENABLE_FOP	Enable Flash Operator Panel - Default TRUE
ENABLE_SSL	Enable HTTPd to serve SSL requests - Default FALSE
ENABLE_XMPP	Enable XMPP Module with MongoDB - Default FALSE
HTTP_PORT	HTTP Listening Port - Default 80
HTTPS_PORT	HTTPS Listening Port - Default 443
FOP_DIRECTORY	What folder to access FOP - Default /fop
RTP_START	What port to start RTP Transmissions - Default 18000
RTP_FINISH	What port to start RTP Transmissions - Default 20000
UCP_FIRST	Load UCP as web frontpage TRUE/FALSE - Default TRUE
TLS_CERT	TLS Certificate to drop in /certs for HTTPS if no reverse proxy
TLS_KEY	TLS Key to drop in /certs for HTTPS if no reverse proxy
WEBROOT	If you wish to install to a subfolder use this. Example: /var/www/html/pbx Default '/var/www/html'

ADMIN_DIRECTORY and FOP_DIRECTORY may not work correctly if WEBROOT is changed or UCP_FIRST=FALSE

Networking

The following ports are exposed.

Port	Description
80	HTTP
443	HTTPS
4445	FOP
4569	IAX
5060	PJSIP
5160	SIP
8001	UCP
8003	UCP SSL
8008	UCP
8009	UCP SSL
18000-20000/udp	RTP Ports

• There seems to be a problem with the CDR Module when updating where it refuses to update when using an external DB Server. If that happens, simply enter the container (as shown below) and execute upgrade-cdr, which will download the latest CDR module, apply a tweak, install, and reload the system for you.

• When installing Parking Lot or Feature Codes you sometimes get SQLSTATE[22001]: String data, right truncated: 1406 Data too long for column 'helptext' at row 1. To resolve login to your SQL server and issue this statement: alter table featurecodes modify column helptext varchar(500);
• If you find yourself needing to update the framework or core modules and experience issues, enter the container and run upgrade-core which will truncate the column and auto upgrade the core and framework modules.

Shell Access

For debugging and maintenance purposes you may want access the containers shell.