The modules are listed along the left-hand side of the web interface, and are divided into two sections, "Setup" and "Tools." Once you're at a specific module's page, you can hover your mouse over the title of each entry and get instructions on what the entry does and how to configure it.
For more details about each module, click on the name of the module, below.
Overview of the Modules
Overview
Everything in FreePBX is a module, so you need to enable all the modules you want to use. If you do enable something you don't use, it won't matter - it'll just make clicking the red 'Reload Bar' take a little longer, as it goes through and asks every modules 'What would you like done?. If you're using a lower powered system (Say, a Piii 500 or slower) you might want to be a bit selective with the modules you use, but it will _only_ affect the speed of FreePBX, not the speed of Asterisk itself.
The modules are listed along the left-hand side of the GUI, and are divided into two sections, "Setup" and "Tools." Once you're at a specific module's page, you can hover your mouse over the title of each entry and get instructions on what the entry does. Generally, you'll want to configure the modules in this order:
Trunks- Trunks are the PBX equivalent of a phone line. They are how your system makes calls to the outside world and receives calls from the outside world. Without a trunk, you can't call anyone. You can configure a trunk to connect with any VOIP service provider (such as FreePBX's SipStation), with a PSTN/Media Gateway (which allows you to make and receive calls over standard telephone lines from your local telephone company), or to connect directly to another PBX.
Most reputable VOIP providers will give instructions on how to configure a FreePBX trunk with their service. The vast majority of VOIP providers provide their service using SIP Trunks. A few also support IAX Trunks.
Zap trunks are used to connect with phone line cards that are installed into your PBX computer.
An ENUM trunk interfaces with e164.org and allow you to make and receive free calls over the internet to anyone who has registered their number with e164.org. ENUM's usefulness is limited by the fact that your machine must be exposed to the internet to receive ENUM calls, and doing so presents a security risk.
The Dialed Number Manipulation Rules section lets you redirect calls to certain numbers to other numbers. For example, if someone dials 411, FreePBX can be configured to change that to 1-800-FREE-411. Or you could make 611 call your grandmother. Hover your mouse over the words "Dialed Number Manipulation Rules" for more details.
Outbound Routes- Outbound Routes are how you tell your PBX which Trunks (phone lines) to use when people dial certain telephone numbers. A simple installation will tell the PBX to send all calls to a single trunk. However, a complex setup will have an outbound route for emergency calls, another outbound route for local calls, another for long distance calls, and perhaps even another for international calls. You can even create a "dead trunk" and route prohibited calls (such as international and 976 calls) to it.
Hover your mouse over each of the fields for more details on what each field controls.
Extensions (or Devices and Users)- Extensions are where you set-up devices (telephones) and users (extensions) on your system.
To create one, click "add extension" on the right-hand side of the screen, and then select "generic SIP extension." Although there are a lot of fields available, most of them can be left blank or at the default setting. The only required fields are: User Extension (set the extension number here), Display Name (give the extension a name, usually a location or a person), and secret (the password used to register a phone to the extension).
All of the available fields have help available right on the Web GUI. Just hover your mouse over the field and a pop-up will tell you what it does.
Follow-me- The follow-me module allows you to create a more complicated method of routing calls that are placed to a specific extension. Using this module, you can make a call to one extension ring several other extensions, or even outside phone numbers. You can also make calls to one-extension end in the voicemail of another extension.
For example, using "follow-me," you could make a call to extension 10 actually ring extension 10, extension 11, and extension 12, and call someone's cellular phone, for 15 seconds, and then, if nobody answers, go the voicemailbox for extension 17.
Ring Groups- Ring Groups allow you to create a single extension number (the Ring Group Number) that will call more than one person.
For example, you could make a Ring Group so that when any user dials extension 601, extensions 10, 11, 12, and 13 ring for 15 seconds, and then the call goes to the voicemail for extension 17.
Inbound Routes- The Inbound Routes module is where you tell the PBX how to handle incoming calls. Typically, you tell the PBX the phone number that outside callers have called ("DID Number" or "Direct Inward Dial Number") and then indicate which extension, Ring Group, Voicemail, or other destination the call will go to.
Parking Lot- A parking lot allows anyone who has received a call to park the call on an extension that anyone else can access. Typically, you receive the call, transfer it to extension 70, and then listen as the system tells you where you can pick up the call (usually extension 71). Then, anyone else on your PBX can dial 71 to pick-up the call.
Feature Codes- This module allows you to set the special codes that users dial to access various features. You can also disable features if you don't want users to be able to access them.
General Settings-: This module has several important features you may want to consider changing:
. Dial Dial Voicemail Prefix: This feature allows you to directly dial an extension's voicemailbox by dialing the prefix listed on this page. So, for example, if you dial * (the default) and then an extension number, you'll skip ringing the extension and go straight to their mailbox. This is useful when you wish to transfer someone directly to voicemail. If you leave this at the default of *, and you use two digit extensions, the direct dial voicemail function will confict with certain feature codes. You may wish to change the direct dial voicemail prefix from the default of "*" to something else, such as *86.
. Optional Voicemail Recording Gain: If you find that the voicemail messages you receive are quiet compared to the system recordings, you might want to change this from the default of 0 to 5.
. Do Not Play "please leave message after tone" to caller- If you'd prefer that asterisk just play the outgoing message and then beep, then check this box.
. Operator Extension- You can specify the extension number/ring group number that people will get transferred to when they dial 0 while leaving a voicemail. If you leave this blank, the caller will return to whatever ring group they came from before reaching voicemail.
Paging and Intercom- By default, you dial *80 plus the extension number to intercom a specific user. The Paging and Intercom module allows you to define numbers you can dial to page a group of devices at once. For example, in a small office, you might define a paging group that allows any user to dial 00 to page the entire office.
Conferences- This module allows you to create an extension number that people can dial into in order to have a conference call.
For example, any user could dial extension 800 and they would be in a conference call.
IVR- This is the module where you configure an auto attendant to answer calls and direct them.
System Recordings- This is the module where you record the messages for use on your auto-attendant.
DISA ("Direct Inward System Access")- This module allows you to create a destination that allows people to call in from an outside line and reach a system dial tone. This is useful if you want people to be able to take advantage of your lower rate for toll calls, or if you want outside callers to be able to use the paging or intercom features of the system. Always password protect this feature, if you use it at all.
Backup and Restore- This module allows you to backup and restore the settings and recordings made by FreePBX/Asterisk. After they are made, you can find the backups by typing the following at your command prompt:
cd /var/lib/asterisk/backups
ls -l
Time Conditions/Time Groups- These modules allow you to define time periods and then choose where calls will go during those time periods. For example, you could set up a time period for the days and hours when a business is open, and then set the system to send calls to all extensions during business hours and straight to voicemail after hours.
PIN Sets- Allows you to set PINS (i.e., passwords) that you can make the system require users to enter before putting calls through. The PINs are defined in this module, and then selected in the Outbound Routes module.
Additional FreePBX modules
You will find some additional modules that have been contributed to the FreePBX community here:
http://mirror.freepbx.org/modules/release/contributed_modules/
Third-Party Unsupported Modules
Contributed Modules
These modules are unsupported — they have been known to cause happiness and glee, or dizziness, confusion, and frustration.
The modules can be obtained from:
http://mirror.freepbx.org/modules/release/contributed_modules/
Instructions for installing third-party modules that are not yet included in the Third-Party module repository (e.g. recently contributed modules) can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/modules/README.t...
Please understand that these modules were written by other users of FreePBX and not the FreePBX development team. Issues such as what version or versions they will work with, problems with installation, etc. need to be directed to the authors of these modules and not the FreePBX development team.
If you have installed any of these modules and are about to upgrade your system from one version to a newer version it is possible that right after the upgrade your system might not work properly. If that is the case please first disable these third party modules to verify that they are not causing the issue. Some authors don't upgrade versions as quickly as others do and might not know there is a problem.
It is also possible that when you upgrade that suddenly you might not have a working GUI to disable a module. If that becomes the case, you can do the following at a Linux prompt (assuming standard install defaults):
/var/www/html/admin/modules/framework/bin/module_admin disable {module name}
If you are an author of any module(s) here we request that you please keep it updated, and in your description include what version(s) it has been designed and tested on, so that in the future people will know how current the posted module is when they look at it.
Boss-Secretary
Boss-Secretary
The boss-secretary module creates a special ring group which includes one or more "bosses" and one or more "secretaries". When someone calls the boss' extension, the secretary (or secretaries) extension will ring too, allowing the secretary to answer his or her boss' call.
Additionally one may define one or more chiefs, who may call the boss directly, without ringing the secretary's extension.
The module includes codes for activating, deactivating and toggling the groups' state. For example, when a secretary ends her working day, she may turn off the boss-secretary group dialing *255<ext number>, so her boss will receive calls directly.
The module generates the appropriate hints to have ip phones show the groups state by subscribing to the *255<ext number> extension.
Note: this module send an alert info of type alert-group to ip phones, so we recommend to set up the boss' phone so its ring tone is silent or very quiet when receiving an alert info of type alert-group; this way, the boss won't be distracted by phone calls which are being processed by his or her secretary. Calls from the secretary to the boss will ring normally.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/modules
Bulk DIDs
Bulk DIDs
Manage DIDs in bulk using CSV files.
Start by downloading the Template CSV file or clicking the Export DIDs button.
Modify the CSV file to add, edit, or delete DIDs as desired. Then load the CSV file. After the CSV file is processed, the action taken for each row will be displayed.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Bulk Extensions
Bulk Extensions
Use CSV files to add, edit, or delete one or more extensions. Export extensions feature now available. Improved interface and documentation.
This module is a replacement and upgrade for the importextensions module. After installation there will be a Bulk Extensions entry under Third Party Addon on the Tools tab on the left side menu. The Bulk Extensions page allows you to download a template CSV file. It also allows you to upload a CSV file for processing. Almost all the options shown on the FreePBX 2.4 Extensions interface can be specified in the CSV file. The template CSV file has examples of adding a new extension, editing an existing extension, and deleting an existing extension.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Note that versions starting with 0. can be used with FreePBX 2.4 (and possibly earlier), while versions starting with 2.5. may only be used with FreePBX 2.5.
Caller ID Popup (post answer for use with Ring Group)
CID Popup
Caller ID Popup (post answer for use with Ring Group)
This specialized module allows you to specify a destination IP Address of FQDN to be associated with various AGI Scripts that can be launched as part of a post answer action in a ringgroup. The scripts are specialized to deal with various destination CRM systems such as SugarCRM and other future system to provide push based CID PoPup and other CRM data to the agent who answers the call. Once you make an entry including the relevant information, these instances will be available within ringgroups to optionally associated a ringgroup with one of the configured servers so that such CRM data can be displayed to its agents.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Caller ID Superfecta
Caller ID Superfecta
Purpose:
This module installs the Caller ID Superfecta (a utility program which adds incoming CallerID name lookups to your Asterisk system using several different sources: AsteriDex, the Google Phonebook, AnyWho, and WhitePages, to name only a few) as a FreePBX Module. As a Module, the Configuration items can be changed from the Web UI.
Notes on version 2.2.2:
Caller ID Superfecta is an easy to install module designed for use with almost any Asterisk/FreePBX/MySQL PBX distribution. It is user interface driven, and requires no special technical capabilities to install and configure.
Version 2.2.2 of the caller ID Superfecta provides worldwide caller ID lookup from multiple sources, and provides support for international caller ID formats of all varieties.
This most recent release of the Caller ID Superfecta includes 27 different data sources, and supports multiple caller ID schemes, that allow the PBX administrator unparalleled flexibility in configuration of inbound caller ID functions. More data sources are added frequently. And just the make sure your Superfecta keeps working at its peak, all data sources can be updated and new sources added online with the click of a mouse using Caller ID Superfecta live data source update – and its all built right into the module.
Conditions/Prerequisites:
This module depends up the Asterisk DB AMP user ID and password being set at their default values. The module script may be edited to reflect your actual id and passwords if you have changed them.
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBX In A Flash
Elastix *See Special Installation Steps
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_Caller_ID_S...
The principal community discussion thread for the module is located here.
http://pbxinaflash.com/forum/showthread.php?t=4387
Release Announcement:
http://www.freepbx.org/forum/freepbx/users/caller-id-superfecta-module-f...
Theory of Operation v 2.0.0:
http://projects.colsolgrp.net/documents/show/2
Capture Groups
Capture Groups
This module allow the administrator to quickly create and administrate capture groups. It will configure every extension's capturegroup and pickupgroup automatically.
Additionally it will generate a virtual extension number which, which notifies users (phones) of calls in the capture group.
Using the following asterisk 1.4.x patch (File asterisk-1.4-pickupbycallid.patch)
one can subscribe phones to the virtual extension generated by this module and receive notifications of all calls received by the groups' members, and may pickup the incoming call by pressing the subscribed button (tested with snom phones and firmware >= 7.1.35).
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/modules
Also see Ticket #3910
CDRCost (a.k.a. Call Cost)
CDRCost (a.k.a Call Cost)
This downloadable FreePBX plug-in allows you to setup the Call Cost parameters, categorizing each call and assessing a cost for each. From this data you can generate call costs by create a new table called cdrcost in the asteriskcdrdb (which contains the cost and the used rate of each calls which is not 0). Afterwards you can generate many kinds of statistics from this (to see which extension, group etc. called which direction, and how much the cost was for you).
You can define the following parameters:
Zone Group
These are a collection of zones which are only used for grouping of Zones (for the UI). You can put each Zone into one Zone Group.
Parameters:
- You can give a name for each Zone Groups.
Example of Zone Groups:
- Local
- Long Distance
- Mobile
- International, etc.
Zone
These are the definition of the Zones.
Parameters:
- You can give a name for each Zone,
- Assign it to the Zone Group to which it belongs (choose from the list),
- Define which pattern is used for this Zone. This pattern is a regular expression which will be fitted on the destination number (i.e. Do not use the Asterisk style patterns NXZ!!!). Example:
^1888[0-9]{7}$
Schedule
A Schedule is a collection of Schedule Parts.
Parameters:
- You can give a name for each Schedule.
A Schedule Part define an interval(s).
Parameters:
- You can give a name for each Schedule Part.
- Assign it to the Schedule which it belongs to,
- Weekday of this Part 0-7 (both 0 and 7 means Sunday) or -1 in case it's applied for all days,
- From which time valid (format is: 'hh:mm:ss'),
- Until which time valid (format is: 'hh:mm:ss').
Rate
This defines the cost parameters from which the cost can be calculated.
Parameters:
- You can give a name for each Rate.
- The accountcode of the call when this rate can be applied,
- From when this rate is valid (format is: 'yyyy-mm-dd hh:mm:ss'),
- Until when this rate can be used (format is: 'yyyy-mm-dd hh:mm:ss'),
- The outbound Trunk prefix of the call (eg. Zap),
- Zone for this rate is valid,
- Rate is the call cost per minutes,
- Minimum duration which will be charged in seconds,
- Block size of the call duration (step size) in seconds,
- Cost of the established connection, connection fee,
- Disconnection cost,
- The schedule in which rate is valid.
Maintenance
In this tab you can run a few maintenance operations on the cdrcost table.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Config Editor
Config Editor
Purpose:
This module installs a "protected" version of the Config Edit Program. The "protected" version obfuscates several configuration files, specifically, those that should never be edited by hand.
Conditions/Prerequisites:
This module will co-reside with the Advanced "un-protected" (non obfuscating) Config Editor module.
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBXIAF
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_Config_Edit...
Config Editor (Advanced)
Config Editor (Advanced)
Purpose:
This module installs an "unprotected" version of the Config Edit Program. The "unprotected" version does not obfuscate any of the configuration files, including those that should never be edited by hand.
Notice: Anything which is put in the xxx_additional.conf files will be overwritten by FreePBX.
Don't use this tool unless you are intimate with the workings of FreePBX.
Conditions/Prerequisites:
This module will co-reside with the standard "protected" (obfuscating) Config Edit module.
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBXIAF
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_Config_Editor_(Advanced)
CustomContexts
Currently this is an unofficial module that must be manually installed. It can be downloaded from:
http://mirror.freepbx.org/modules/release/contributed_modules
choosing the latest version of the customcontext module. The easiest way to install it is to dowload it to your desktop and then choose "Upload Modules" in FreePBX Module Admin and install the module.
Then in FreePBX, click on the Tools tab, Module Admin, and Custom Contexts, select Install, click on Process, and then the red bar to complete installation.
Possible Uses
- Restrict access to certain outbound routes or feature codes by a particular extension or group of extensions.
- Give particular extension(s) priority access to certain outbound routes, such as a particular emergency route associated with their geographic location.
- Give certain outbound routes top priority for use during "free" or low cost calling periods, while making those same routes lower priority (or disallowing access entirely) during higher cost time periods.
- Disallow access to outbound routes (with possible exception of Emergency access) to certain (or all) extensions during particular time periods (don't let night cleaning crew make long distance calls, or disallow outgoing night calls from telephones in children's rooms, while still allowing emergency number calls).
- Allow two or more families/companies/organizations to use the same FreePBX box, while still allowing each to have access only to "their" outgoing routes and trunks.
- If you have a SIP provider that does not send DID (normally a pain to handle because you can't create a normal Inbound Route), set up a new custom context (call it idiot-provider), give them no access to anything (deny all), and then specify where you want their calls to go in the Failover Destination. Then put context=idiot-provider in that provider's trunk user details.
What This Module Is NOT Intended For
- This module is not intended to provide an alternative way to access code that is found in, or might normally be placed in extensions_custom.conf. You probably want the DialplanInjection module if that is what you are trying to achieve.
- It's also not intended to give you simplified access to existing features, applications, or destinations (e.g. from an IVR) - you probably want to use Misc. Applications and/or Misc. Destinations (or possibly a Custom Extension) for that.
Known Incompatibility
Please be aware that if you you have installed both this module and HUDlite on the same system, HUDlite will allow users to bypass any restrictions placed upon them by this module. Therefore, restricted users should not be given access to HUDlite. Also, a savy user can bypass the system. As soon as they transfer a call, the current dialplan puts them in a new context which is effectively from-internal taking away any restrictions.
Description
One feature which was a bit lacking in Asterisk/FreePBX was the ability to easily create multiple tenants.
This module creates custom contexts which can be used to allow limited access to dialplan applications.
Now allows for time restrictions on any dialplan access!
This can be very useful for multi-tenant systems.
Inbound routing can be done using DID or zap channel routing, this module allows for selective outbound routing.
House/public phones can be placed in a restricted context allowing them only internal calls.
Custom contexts can now be used as destinations. An IVR menu, Time Condition, etc. can now send a caller into a custom context. This feature requires FreePBX 2.2.0rc2 (or the latest SVN version if prior to the release of rc2)
(The following are the module author's comments, "I" refers to the module author, not the original creator of this wiki page).
A number of improvements have been made to freePbx to handle multiple tenants.
1) inbound routing based on zap channel - i used to have to hack it by putting each zap channel in its own context.
2) authtype = database allows for dividing extension ranges
the main problem for me was outbound routing...
I wanted some extensions to dial out one route, and others out another route.
I had to create a custom context for each, then place each in their own custom context, then include all of the contexts which they should have access to. This became a nuisance as each module added its own context to from-internal-additional which could not be included as it also contains outbound-allroutes.
The purpose of this module is to dynamically list all contexts included in any contexts you choose, and allow you to create custom contexts which can include any of these all without config editing.
As an added bonus, I added a select list to the devices/extensions page to allow you to easily select any of your custom contexts to place the device in.
Version 0.1.1 - Now has optional Time Groups which allows you to name a set of times to enable the user to not only deny or allow access to certain dialplan contexts, but to control access to each context by time, date or day also.
Version 0.1.2 - Changes
Bugfixes- deleted routes, etc. now are removed.
Context tests for spaces and illegal chars.
Moved admin to tools to reduce confusion.
Added option to allow entire internal dialplan. (Useful for time limit on everything)
Made description for outbound-allroutes clearer that allowing overrides to allow all routes.
Version 0.1.3 - Made it obvious when allowing one include may allow another entire context.
Version 0.2.0 - Added priority feature to allow the user to control in what order the allowed contexts are included.
Version 0.2.1 - Added Duplicate Context option to easily copy an entire set of rules.
Version 0.2.2 - bugfix
Version 0.3.0 - New Features:
Allow or Deny based on pattern matching.
Failover Destination (one for regular extension, one for failed feature codes)
Bugfixes:
Adjusted Gui, Duplicate context, now duplicates the description too.
Version 0.3.1 - New Features:
Now prompts on delete. After duplicate you are editing new context.
It is now possible to rename contexts.
Version 0.3.2 - New Features:
Optional PIN to protect failover destination.
Contexts can now be used as destinations. An IVR menu, Time Condition, etc. can now send a caller into a custom context.
Version 0.3.3 -
New Feature: Added Set All option to quickly allow/deny all.
Fixed bug which caused routes to be denied after rename/sort/or delete other route.
Version 0.3.4 -
Fix for compatibility issues with FreePBX version 2.3.1.3.
Installation of Beta version
Download the latest Beta version using the instructions in the first paragraph.
If you did not use the instructions for getting and installing the module using wget, then expand the .tgz file into the /var/www/html/admin/modules directory - it will create a new directory called customcontexts. Make sure the group and owner of that directory are asterisk and that the permissions match that of the other module subdirectories.
Browse to FreePBX, Tools | Module Administration. You should see an entry for Custom Contexts. Click on it, click install, then click process and the red bar as usual.
Usage Instructions
Most users will not need to do anything in the Custom Contexts Admin section (now found under the Tools tab) - that is for advanced users. When you "add" or "remove" contexts from the Admin, you are not really adding or removing anything, you are just telling the module where to find all of the includes to list. By default there are three includes which should be sufficient for most users: from-internal, from-internal-additional, and outbound-allroutes. So, skip the Custom Contexts Admin section until you feel comfortable making changes there.
The first thing that you will want to create is time groups, if you plan to use those. The reason for doing this first is so that they will become available in the drop down selections when you create your custom contexts. For each group you create, you can decide which times it should be available. You can define multiple times within one named group, and then each named group then becomes available along with allow/deny for each choice under a custom context (this will become clearer further down), so you can allow, deny, or choose your time group to allow only at specific times/dates/days.
One thing to bear in mind when creating time groups is that this module will not forcibly end calls in progress. So if, for example, you have "free" calling on a particular route from 9:00 PM to 7:00 AM, you probably don't want to set the end time right at 7:00 AM, because then someone could make a call at 6:59 and talk for several minutes into the non-free period.
Now, to actually create a Custom Context, you go to the Custom Contexts page, and add a context - note that the context name may NOT contain spaces. Then add a description (spaces are okay here) and submit.
We'll talk about Dial Rules later - in many cases you will want to leave the Dial Rules blank.
Once the context is created, you can edit it to allow or disallow the features and routes you want a particular extension (or group of extensions) to have access to. There is a "Set All" option to set all the features and routes to Allow or Deny - this is useful when you want to start out with all of the dropdowns in one state, so that you only need to change the exceptions. Then choose "Allow" or "Deny" for each application or route - for example, you may wish to allow all, except for the items you specifically wish to restrict (for example, you probably want to restrict ChanSpy and ZapBarge!). If you have created any time conditions, it will also be possible to select those, to allow a feature or route to be accessed only during certain times. If you have any Dial Rules, you can choose to "Allow Rules" (allow the feature or route only if a Dial Rules pattern is matched) or "Deny Rules" (deny the feature or route only if a Dial Rules pattern is matched).
Certain items are in bold red letters, such as "ENTIRE Basic Internal Dialplan" and "ALL OUTBOUND ROUTES." If you allow ALL OUTBOUND ROUTES, it will override the individual route selections in the following section. So if you want users of this context to have access to all outbound routes, you can just allow outbound-allroutes and ignore the individual route sections (leave them all set to "deny"). But if you want to select routes individually, then make sure that outbound-allroutes is set to "Deny". Of course, you could also use non-overlapping time conditions for outbound-allroutes and individual routes.
If you allow "ENTIRE Basic Internal Dialplan", then it overrides every other selection on the page. You would normally only use this with a time rule, to allow your unaltered dialplan to be used for a portion of the day. Allowing the "ENTIRE Basic Internal Dialplan" without using a time rule is usually pointless. If you want control over individual items, deny "ENTIRE Basic Internal Dialplan", and allow only what you want.
Associated with each item a "Priority" dropdown. All priorities are set to 50 by default (so you can easily make any item higher or lower in priority). The best use of these is in the Outbound Routes section - you WILL want to make sure that any Outbound Routes that you allow are ordered by priority, otherwise your outbound calls may not be routed as you expect. Normally you will want to mirror the priority of the existing routes - the easiest way to do that is add 50 to the number at the start of the route, so for example if you have a route called "outrt-001-Emergency" you could add 50 to the "001" and use 51 as the priority. But note that you do not have to mirror the default priority of routes, which could become useful in certain situations.
For example, let's suppose you have an emergency route that goes to a an emergency answering point in your local area, but you also have another emergency route that goes to an emergency answering point in a community where you have a remote office. You could create two emergency routes going to the two different answering points and let the one going to the local point be higher in priority normally, but create a custom context for your remote extensions and in that custom context, make their community's emergency answering point higher in priority.
One more note about priorities - you can hide the display of the priority dropdowns by clicking on "Hide Sort Option" at the top of any custom context page. BUT - if you click on the "Submit" button while you have the priorities hidden, all the priorities on that page will be reset to the default (50)! So use this option with care!!
Note: This option is no longer available as its purpose was to clean up the page when priorities were listed below each context. Now that the display was fixed, the "hide priorities" option was removed.
At the bottom of the page, you can select a Failover Destination and a Feature Code Failover Destination. The Failover Destination is used when the called number does not start with a * and does not match on any route, while the Feature Code Failover Destination is used when the called number begins with a * and does not match any feature code. Be careful here, because it's possible to send a caller to a destination that gives them access to destinations that you don't intend for them to be able to access. Either or both of the Failover Destinations can be PIN protected, that is, you can enter a numeric PIN to require authentication before continuing to the destination.
Regarding Dial Rules, these can be used when you want to further allow or restrict access based on the number dialed. For example, you could give an internal caller access to a particular route only if 911 was called, or if a local number was called, while restricting their ability to place other calls on the same route. It's also possible to use the | character to strip off initial digits. For example, if you had a dial plan that included something like 90210|1NXXNXXXXXX you could set an outbound route to "Allow Rules" and it would generally restrict access to that route, except for those callers that know that they must dial 90210 prior to the 1+area code+number.
Sometimes you will want to create a new custom context that is very similar to an existing custom context you have already created - perhaps you only want to modify one or two items in the new context. The easiest way to do that is to go into the existing context, then click on "Duplicate Context ..." at the top of the page. This will create a duplicate of the existing context that you can edit as you
wish.
Finally, you need to go to your Extensions page and select each extension for which you wish to use a custom context. On each individual extension page, you should now see a dropdown to allow you to select a custom context. This drop-down is simply a convenient way to fill in the correct context in the "context" textbox. When you click on a custom context, it replaces whatever is currently in the "context" textbox with your new selection - if you choose "Default", it resets the extension back to the default "from-internal" context. Don't forget to click "Submit", and then click the red bar when you are all finished making changes.
NOTE that if you disable or uninstall the Custom Contexts module, you MUST reset all the extensions back to the default "from-internal" context. If you delete a time group, anyone who had that time limitation becomes "Allow" with no time restrictions. If you add a new outbound route, by default that route is set as "Deny" in the Custom Contexts, so you should go into each context and set it to "Allow" (or use a time condition) where appropriate.
One more caveat. After you add an outbound route, it is not available until you reload.
Dialplan Injection
Currently this is an unofficial module that must be manually installed. It can be downloaded from this unofficial repository. See this FreePBX module tutorial if you need help understanding how to install it.
Alternately, here is how to get and install this file (version 0.1.1) using wget:
cd /var/www/html/admin/modules
wget http://www.zelie.com/~n3glv/asterisk/dialplaninjection-0.1.1.tgz
tar -xzvf dialplaninjection-0.1.1.tgz
rm -f dialplaninjection-0.1.1.tgz
Description
This unofficial FreePBX module allows you to create short custom dial plan fragments. While such fragments can also be added to extensions-custom.conf, the advantage of creating them in this module is that the resulting dial plan fragment can be directly selected as a destination in modules that use destinations. Optionally, the dial plan fragment can also be accessed directly by calling an extension number. Just about anything that could be put into extensions_custom.conf can be placed in a Dialplan Injection.
Version history
Version 0.0.1: Initial version
Version 0.0.2:
- Fixed extension bug
- Allowed patterns in extension (allows a Dialplan Injection to be accessed by a group of extensions defined by a pattern)
- Commands are one big text area now
Version 0.0.3:
- Added ability to add line labels. Labels also become available as destinations for other modules.
- Each direct dial injection is now in its own context to allow individual inclusion in other contexts.
- Version now display correctly on screen.
Version 0.1.0:
- Removed unique constraint on direct dial extension.
- Added templates for most dialplan apps.
Version 0.1.1: Fixed a few templates and version display bug.
Installation of Beta version
Download the latest Beta version using the instructions in the first paragraph.
If you did not use the instructions for getting and installing the module using wget, then expand the .tgz file into the /var/www/html/admin/modules directory - it will create a new directory called dialplaninjection. Make sure the group and owner of that directory are asterisk and that the permissions match that of the other module subdirectories.
Browse to FreePBX, Tools | Module Administration. You should see an entry for Dialplan Injection. Click on it, click install, then click process and the red bar as usual.
Usage Instructions
To create a Dialplan Injection, click on Dialplan Injection and then on "Add Injection" (if you are not already on that page).
Enter a short description for your Injection - this should contain letters and numbers only.
Optionally, you may enter an extension number for direct access, which will allow dialing this injection directly. You may leave the extension field blank if you only plan to access the injection indirectly (such as from an IVR menu choice) OR if you plan to use Misc. Applications to create one or more extensions (or feature codes) for entry point(s). The extension may be a pattern (such as would be allowed in a route dial plan) to match (for example) a range of extensions. Also, you may use a pipe | to strip the preceding digits, as would be allowed in a route dial plan pattern.
Under Destination, choose a destination for use when all the lines in your Dialplan Injection have executed. For example, you could select Core: Hangup if you simply plan to play a message to the caller and disconnect. Or, you could send a caller to an IVR to make another selection.
Click on "Submit" to create the Dialplan Injection. Do not click the Red Bar yet.
Now select the Injection you just created from the list of Injections at the right. When you bring it up, you should see a text box where you can enter the actual lines of your injection. Remember, at this point you are playing the role of computer programmer and if you write bad code, your Injection wont work as you intend. Garbage in, garbage out. So check what you write very carefully.
As with code you might write in an actual context, you can use line labels to allow for conditional or unconditional jumps, or to permit multiple entry points to your code (for example, you might write a routine that returns certain information about an extension - if entered at one point, it might give the information about the extension the user is calling from, whereas if entered at a different point, it might prompt the caller to enter an extension and then give the report about that extension). To use a label, simply enclose it in parenthesis and put a comma between it and the statement, like this:
(label1),NoOp(This is a line with a label)
.....
GotoIf(somecondition?label1)
Each label you use can be selected as an entry point from other applications (for example, in Misc. Applications you will see a radio button and dropdown for Dialplan Injections, and in that dropdown you'll be able to select any labeled statement as an entry point, in addition to the normal entry at the top of the code). For example, if your Dialplan Injection was named My Injection and contained the above code fragment, you'd be able to use "My Injection" as a destination, and also "My Injection-label1".
There's also a "New Command" box that contains some commonly used commands, in order to help you recall the syntax of these commands. You don't have to select anything here but if you do, whatever you select will be pasted into your code.
Remember to put Ringing on a line by itself (usually as the first line of the code) if you want your caller to hear a Ringing signal until the call is answered. Also, you need to use Answer on a line by itself before playing any significant information to the caller (any message not having to do with call progress) in order to comply with legal requirements and to make sure that billing commences at the proper point. Obviously this does not apply for injections that can only be reached by internal callers, or can only be reached after the call has already been answered (by an IVR, for example).
When you are finished writing your code, click on "Submit", and only now should you click the Red Bar to enable use of your Dialplan Injection.
There is one other point that needs to be mentioned: If you have also installed the Custom Contexts module, you can individually allow or deny each Dialplan Injection. Here is how to do that (please note that nothing in the next three paragraphs will make sense to you if you have never used the Custom Contexts module):
Click on the Tools tab, Custom Contexts Admin, and Add context. Put ext-injections in the Context field (this must be entered exactly as shown), and give it a description (e.g. "Dialplan Injections"). Submit the page and click on the Red Bar.
(Optional but recommended): Go back to Tools, Custom Contexts Admin, and click on the description you just created (e.g. "Dialplan Injections"). When the page comes up, give each of the injections a meaningful name, rather than the default ext-injection-number. Note that the numbers at the end of each default injection name match the numbers in angle brackets following each injection name on the Dialplan Injections page. At present the Custom Contexts Admin tool doesn't seem to pick up the "friendly" name of the Dialplan Injections automatically.
After doing the above, you can go to the Setup tab, Custom Contexts, and edit your contexts (this assumes you've already created some custom contexts) as follows: Deny ext-injections, which should now be in red text, unless you want the context to allow ALL Dialplan Injections. Then, allow only those injections you wish to allow in each custom context.
Examples
Here are some actual Dialplan Injections to give you an idea of how simple a Dialplan Injection can be:
Play Music On Hold from the default context to the caller for up to 9999 seconds:
SetMusicOnHold(default)
WaitMusicOnHold(9999)
Inform a caller that no 911 service is available on the line by playing an appropriate recording three times, separated by one second of silence (note this should never actually be used as a substitute for 911 service, it's just an example):
Playback(no-911-2&silence/1&no-911-2&silence/1&no-911-2,noanswer)
The ,noanswer means that if the call has not already been answered it will not be, since this is considered a call progress message (Playback, unlike some other methods of playing audio, defaults to answering the line and requires the ,noanswer appendage if you don't want the call answered).
In both of the above examples, you would probably use Core: Hangup as the Final Destination.
Before ringing a particular line, play a recording to the caller containing some sort of notice. Here we'll play one second of silence (optional, but useful if some of your callers are calling from a phone with a dial in the handset - it gives them time to get the phone to their ear), then the system recording that says "This call may be monitored or recorded":
Playback(silence/1&this-call-may-be-monitored-or-recorded)
Ringing
Then use Core: and the desired extension (or, if you prefer, a ring group) as the final destination. You can then use this Injection as a selection from your main IVR menu, so that callers that select this extension or department hear the recording first.
In a way this is a trivial example, because when creating a Ring Group you can specify an announcement to be played before ringing commences, and System Recordings lets you concatenate multiple recordings into one (so you don't really need to use a Dialplan Injection if that is all you want to do). BUT, suppose you want to play some audio that is dynamically generated by an AGI script, rather than system recordings? For example, you could call an AGI script that plays some information about current conditions (e.g. system status, the weather, or whatever you might be monitoring), then returns to an IVR as a final destination.
Inform caller that parking lot slot is empty (example pattern usage) - Let's say you have a parking lot for parked calls with eight slots, which can be numbered 901-908. If a caller tries to pick up a parked call and it's no longer there, you want to inform them of that fact. So, you would create a Dialplan Injection and use a pattern for the extension:
_90[1-8]
(Note: the underscore as the first character of a pattern is not required, as the module will insert it in the dialplan if it detects a pattern).
Then for the actual injection, simply play one second of silence, followed by an appropriate system recording:
Playback(silence/1&pbx-invalidpark,noanswer)
If there is a parked call, it takes precedence when someone dials the appropriate parking lot extension, otherwise the Dialplan Injection kicks in and plays the message to the caller.
Modified Speaking Clock routine with labels and multiple entry points - Finally, here's a more complex example - a modified Speaking Clock routine, that can give the time in either of two time zones (in this example, U.S. Eastern and U.S. Central time, but you can change these to any standard Unix time zones). This example uses labels, AND has two entry points (you can add more). We'll show the complete instructions to implement this:
1) Go to Dialplan Injections, Add Injection. Give it a description (e.g. "Speaking Clock") but do NOT give it an extension (with the code as shown you actually could add the extension for the Eastern Time Zone entry point, but for demonstration purposes we won't give it an extension here). Make the Destination Core: Hangup. Click Submit.
2) Re-Enter your new Injection, and paste the following code into the "Command" textbox:
(est),NoOp(Speaking Clock for Eastern Time Zone)
Set(TimeZn=EST5EDT)
goto(scstart)
(cst),NoOp(Speaking Clock for Central Time Zone)
Set(TimeZn=CST6CDT)
(scstart),Ringing
Set(FutureTime=$[${EPOCH} + 8])
Set(FutureTimeMod=$[${FutureTime} % 10])
Set(FutureTime=$[${FutureTime} - ${FutureTimeMod}])
Set(MaxConnectTime=$[${FutureTime} + 180])
(scringsomemore),Set(FutureTimeMod=$[${FutureTime} - ${EPOCH}])
GotoIf($["${FutureTimeMod}" < "0"]?scanswer:scwaitasec)
(scwaitasec),wait(1)
goto(scringsomemore)
(scanswer),Answer
(scplayagain),Set(FutureTime=$[${FutureTime} + 10])
Set(FutureTimeMod=$[${FutureTime} % 60])
wait(1)
playback(at-tone-time-exactly)
SayUnixTime(${FutureTime},${TimeZn},IM)
GotoIf($["${FutureTimeMod}" = "0"]?scexactmin:scsaysecs)
(scexactmin),SayUnixTime(${FutureTime},${TimeZn},p)
goto(scwaittobeep)
(scsaysecs),playback(and)
SayUnixTime(${FutureTime},${TimeZn},S)
playback(seconds)
(scwaittobeep),Set(FutureTimeMod=$[${FutureTime} - ${EPOCH}])
GotoIf($["${FutureTimeMod}" < "1"]?scplaybeep:scwaitsectobeep)
(scwaitsectobeep),wait(1)
goto(scwaittobeep)
(scplaybeep),playback(beep)
Set(FutureTimeMod=$[${MaxConnectTime} - ${EPOCH}])
GotoIf($["${FutureTimeMod}" < "1"]?scthatsall:scplayagain)
(scthatsall),GotoIf($["x${IVR_CONTEXT}" = "x"]?app-blackhole,hangup,1:${IVR_CONTEXT},return,1)
3) Click Submit after entering the above.
4) Now, because we want multiple entry points, go to Misc. Applications, Add Misc. Application. Give it a description (such as "Speaking Clock-Eastern") and a feature code number (an unused one, or you can use *60 if you have disabled FreePBX's default speaking clock under Feature Codes). For the Destination, select Dialplan Injection and in the dropdown select "Speaking Clock-est". Submit.
5) Repeat step 4, except make the description different (e.g. "Speaking Clock-Central" and assign a different feature code. In the dropdown, select "Speaking Clock-cst". Submit.
6) Click the red bar. Now you can use one extension or feature code to get the time in one time zone, and the other extension or feature code to get the time in the other.
Alternately, if you have also installed the Custom Contexts module, you could use the same feature code number in steps 4 and 5, but then set up custom contexts in such a way that any particular extension only gets access to one time zone or the other.
Module Author: naftali5
ENUMPlus
ENUMPlus
ENUMPlus is a community effort whose goal is to simplify the use of ENUM. Major features include :
• Immediate Phone Verification
• SIP URI Testing
• Instantaneous record lookup.
• Open Source (GPL v.3)
• Nameserver redundancy
• Additional ENUM Lookup Sources
ENUMPlus project page: http://enumplus.org/
Blog post introducing the module with additional details: http://geekhut.org/enumplus/
Related thread in PBX in a Flash forum: http://pbxinaflash.com/forum/showthread.php?t=4375
Extended Routing
Currently this is an unofficial module that must be manually installed. It can be downloaded from this unofficial repository. See this FreePBX module tutorial if you need help understanding how to install it.
Alternately, here is how to get and install this file (version 0.0.1) using wget:
cd /var/www/html/admin/modules
wget http://www.zelie.com/~n3glv/asterisk/extendedrouting-0.0.1.tgz
tar -xzvf extendedrouting-0.0.1.tgz
rm -f extendedrouting-0.0.1.tgz
Description
This unofficial FreePBX module adds Extended Routing capabilities to FreePBX. It adds a failover destination to outbound routes, and also allows you to choose an outbound route as a destination from other parts of the dialplan.
Some possible uses for this module (just as examples, there are many others):
Use #1 - controlling costs.
Suppose that on a particular route, you have some free or low-cost trunks, and one trunk that costs (more) money to use, and you want to fall through to it only as a last resort, and you want to know when you are using that expensive trunk. You don't want to have a different route with a different dial pattern, since that would be a nuisance (i.e. dial... oops "all circuits are busy"... hang up and dial the expen$ive route). So you set the costly trunk as the last trunk in your standard route, but the problem is that up to now, you have had no way of knowing when you are talking on that trunk, other than by watching the CLI.
Enter extended routing.
You set up two routes, with the same dial patterns. The second is your high-cost route that includes the expen$ive trunk, and because of its priority it will normally never get hit (unless someone is in a custom context that only has access to the more expensive route). Add a failover destination on the first route that goes on to the second, and put a pin on the second. You now have a very simple method of trying multiple routes with the SAME dial pattern, and by requiring a pin the caller must affirmatively choose to use that route.
Alternatively you can fail the first route to a custom sound, and then continue to the second route without a pin. In this case it will simply warn you that you're on a more costly call, but are not required to input a pin. But, note that in version 0.0.1, you cannot use a dialplan injection as the sound source (see "Limitations" section below).
Use #2 - routes as a destination.
You have a few people in a restrictive custom context. But, you have another Asterisk box on which you don't mind them having unlimited access. You have an IAX2 trunk set up between the two. You can't set up an outbound route that allows "everything" (a dot as the pattern), or else all of your calls may start going out via that trunk. So instead, set up the route and give NO ONE access to it. Then you can fail over any custom context to that route, and anything they don't have permission for will try that route (this one can also be accomplished using priorities in a custom context, but this is probably safer.)
Use #3 - failover for routes.
You have DISA set up, and you don't want Allison to tell you that all circuits are busy and then hang up. You would rather have your outbound routes fail to the "all circuits busy" message, but then go to your IVR so you can reenter the DISA (thereby avoiding the need for you to hang up and call back).
Limitations in version 0.0.1: Dialplan Injections currently mess up the dialed number, and therefore should not send to an outbound route as a destination. For example, they cannot be used as a "middleman" to generate the sound mentioned in "Use #1" because they will lose the dialed number. This will be fixed soon. Also, when using outbound routing as a destination, it has the same rules as when using a custom context as a destination. You do NOT have the chance to dial another number (it is not DISA), it simply takes the dialed number and tries it out that route.
Installation of Beta version
Download the latest Beta version using the instructions in the first paragraph.
If you did not use the instructions for getting and installing the module using wget, then expand the .tgz file into the /var/www/html/admin/modules directory - it will create a new directory called extendedrouting. Make sure the group and owner of that directory are asterisk and that the permissions match that of the other module subdirectories.
Browse to FreePBX, Tools | Module Administration. You should see an entry for Extended Routing. Click on it, click install, then click process and the red bar as usual.
Module Author: naftali5
Extension Settings
Extension Settings
Allows the easy viewing and changing of the following settings for each extension:
DND (Do Not Disturb)
Call Waiting
Call Forward All
Call Forward Busy
Call Forward No Answer
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
(Filename begins with extcfg)
External Audio (Paging Interface)
The "External Audio" Module for FreePBX
This module provides a public address interface for paging
Operation
This module provides a destination that can be selected from another module such as the miscellaneous application module. This destination connects to the audio line interfaces on the PBX hardware so allowing paging over a PA system.
The extaudio module also controls the audio mixer to set volume levels for interfacing to a public address system.
Normally an external audio input (such as from a radio) is passed through to the external audio output (the PA system). If there is a call to the extaudio destination then the audio input is muted and instead the caller's voice is output to the PA system. At the end of the call the normal audio (such as from a radio) is resumed.
Preconditions
This module expects the alsa mixer to exist on the local system with "Line", "PCM" and "Capture" audio interfaces. (The alsa mixer is available in the alsa-utils package.)
This module requires one of the asterisk console modules to be loaded - either chan_oss.so or preferably chan_alsa.so . (Remove corresponding noload command from /etc/asterisk/modules.conf)
Open Issues
This module is not compatible with some implementations of live/streaming music on hold (i.e. those implementations that cannot coexist with chan_oss.so/chan_alsa.so)
Author
nick.lewis[-at-]atltelecom.com
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Feature Panel
Currently this is an unofficial module that must be manually installed. It can be downloaded from this unofficial repository. See this FreePBX module tutorial if you need help understanding how to install it.
Alternately, here is how to get and install this file (version 0.0.2) using wget:
cd /var/www/html/admin/modules
wget http://www.zelie.com/~n3glv/asterisk/featurepanel-0.0.2.tgz
tar -xzvf featurepanel-0.0.2.tgz
rm -f featurepanel-0.0.2.tgz
Description
This unofficial FreePBX module allows you to see the status of certain features on extensions that are normally activated/deactivated using *xx feature codes. The status of those features can be modified from within this module's web page.
Features that can be checked or modified currently include:
- Call Forward All
- Call Forward Busy
- Call Forward No Answer/Unavailable
- Call Waiting
- DND (Do Not Disturb)
- User Intercom
Note: This module can only be used to view or change settings that are set within FreePBX/Asterisk. Some features may be activated at the device level (IP phone or VoIP adapter) and this module cannot interact with those settings. If you want to be able to control these settings from this module, you should turn off these features in the device configuration settings of phones and VoIP adapters, so that they will be controlled by Asterisk and FreePBX only.
Version history
Version 0.0.1: Initial version
Version 0.0.2: Allows selecting external or non-standard extensions for feature settings
Installation of Beta version
Download the latest Beta version using the instructions in the first paragraph.
If you did not use the instructions for getting and installing the module using wget, then expand the .tgz file into the /var/www/html/admin/modules directory - it will create a new directory called featurepanel. Make sure the group and owner of that directory are asterisk and that the permissions match that of the other module subdirectories.
Browse to FreePBX, Tools | Module Administration. You should see an entry for Feature Panel. Click on it, click install, then click process and the red bar as usual.
Module Author: naftali5
Gabcast
Gabcast
Gabcast is a social broadcasting platform that offers virtual communities, individuals, and organizations an easy way to create and distribute audio content.
Visit www.gabcast.com for more info.
This module allows you to:
• Link extensions to Gabcast channels. It creates a feature code, which defaults to *422 'gab' (you can change this in Feature Code Admin) which allows you to log directly into your Gabcast account. This is ideal for personal podcasting!
• Define a Gabcast channel as a Destination for other modules. For example, you can direct a DID or IVR menu option directly to Gabcast. This is ideal for group and public podcasting!
You must have a Gabcast account & channel to use this feature. Visit www.gabcast.com to sign up. It's a free service!
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Hotel Style WakeUp Calls
Hotel Style WakeUp Calls
Purpose:
This module installs the Hotel Style Wake Up Calls software as a FreePBX Module.
As a Module, the Feature Code may be managed by FreePBX Feature Code page.
Conditions/Prerequisites:
This Module requires php v 5.x on the platform, due to the use of db classes introduced in that version.
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBX In A Flash
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php?title=FreePBX_Module:_Hotel...
Additional/alternate instructions and download link (may have older version of the software):
http://nerdvittles.com/?p=589
iSymphony
iSymphony
This module is only useful if you are using the i9technologies iSymphony software.
"The iSymphony module for FreePBX will replicate the configuration data for extensions, queues and conference rooms. It's simple, download the module from the downloads section. Then browse to the FreePBX module admin section and upload the iSymphony module package. Once installed and activated simply click on the iSymphony module on the left menu and follow the remaining setup instructions. Finally, ease your mind knowing you no longer have to manually update iSymphony to match the information within FreePBX."
The latest release can be found at the i9technologies site in the download section or at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Keylock
Keylock
This module allows the user to lock or unlock his or her extension by dialing the appropriate code and a pin. When the extension is locked, only calls destined to numbers specified in the module's configuration can be made.
The module generates the appropriate hints to have ip phones show the keylock state by subscribing to the <toggle code><ext number> extension.
The first time the user tries to lock his or her extension the module will ask for a new password, which will be used thereafter to lock or unlock the extension.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/modules
LDAP Caller ID Lookup
LDAP Caller ID Lookup
Allows Caller ID Lookup of incoming calls against different sources (MySQL, HTTP, ENUM, Phonebook Module and LDAP)
Significant updates to cidlookup module to support LDAP, including ldap lookup AGI script.
Please note, this is a replacement for CIDLOOKUP which lives in the same name space. Do not install both at the same time.
(This is excerpted from http://www.freepbx.org/forum/freepbx/users/ldap-support-for-caller-id-lo...):
I've modified the cidlookup module significantly to provide support for LDAP lookup.
I now use it extensively to perform caller id lookups against active directory.
You can find it in ticket #2389. It supports an area-code prefix and a number format option.
If you set prefix to 417, and format to (XXX) XXX-XXXX
then a caller by the number of 4173161234 will be searched as
4173161234
3161234
and
(417) 316-1234
This should cover most number formats, though if you need more (like '316-1234' to match a formatted local number) I'll have a look.
Note that its called 'ldapcidlookup' until the changes are (if ever) integrated into the real cidlookup module.
Please don't enable cidlookup and ldapcidlookup at the same time. They use the same database and naming convention.
Blacky
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
OSSEC Module for FreePBX
OSSEC Module for FreePBX
Purpose:
This module installs the OSSEC client interface as a FreePBX Module. According to the OSSEC web site, "OSSEC is an Open Source Host-based Intrusion Detection System. It performs log analysis, file integrity checking, policy monitoring, rootkit detection, real-time alerting and active response."
Conditions/Prerequisites:
This Module is a helper module for use with the OSSEC software installed by default on the following distributions:
Fonicatec PABX
Foncordiax
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_OSSEC
Also see the "Install OSSEC" section on the FonicaPABX-Install page.
Outbound Route Permissions
Outbound Route Permissions
This module allows you to block access to certain routes from specified extensions. You can do bulk changes (for a range of extensions) on the module's main page, or you can individually change access to routes on each extension's page.
You can also pick a Default Destination if a call is denied (so you can send the caller to a recording, etc.). If you wish to use a different destination for denied calls in a particular situation, see the usage tip below.
Note that Asterisk is incapable of having two identical routes and trying to force calls to use the other route if one of them is banned by this module. It will not work. You must have unique outbound routes for the proper selection to work. N.B. Just having different trunk selections does NOT make the routes non-identical!
If you wish to emulate this functionality, you can use the 'Redirect' function. Any number you type in the 'Redirect' range will be PREPENDED to the number dialed, and the call will then be sent through the dialplan again (specifically, it will be sent back to the from-internal context). For example:
• Route 1: Zap/1 matches 0|.
• Route 2: Sip/Foo matches 1|.
If you wanted to stop extension 100 from using Zap/1 at all, and send all his calls through Sip/Foo, you would need to DENY 100 access to Route1, and create a NEW route, Route3:
• Route 3: Sip/Foo matches 9990|.
In the 'Redirect' field, type '999'. When extension 100 dials 0123456, they match Route 1. Route 1 FAILS, and then system invisibly changes the number dialed to be 9990123456 (note the '0' he dialled originally is preserved, and you then strip 9990 from the front in Route 3), which matches Route 3 and the call is then sent via Sip/Foo.
Redirect rules are only checked if the route is DENIED.
You can set a Default Destination if calls are denied. If you wish to use something other than the default in a specific instance, you can use a Redirect prefix and a Misc. Application. Example: set the redirect prefix to 000123, then create a Misc. Application and set the Feature Code to _000123. (note the underscore at the start and the period at the end of the Feature Code - both are necessary), then make the destination of the Misc. Application whatever you wish.
Caveats: If you already have a large dialplan, see How to increase the execution time and/or memory allowed for "orange bar" reloads - you may need to increase one or both of those values. Also, you probably should be running the SVN version of FreePBX (however, it appears to work with FreePBX 2.5.1.2).
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
(the filename is routepermissions-version.tgz)
Usage tips:
Apparently some users seem to be having problems because they don't understand that you need to have one route where the number called by the user is matched exactly. For example, if you want different groups of users to access different trunks whenever 1NXXNXXXXXX is called, you must have a route that has the pattern 1NXXNXXXXXX (or some acceptable variation - see next paragraph) in the Dial Patterns textbox, and that should be set to use the trunks that you want the majority of your users to be accessing. You allow access to that route for those users, then for the "exceptions" you disallow access to that route and (optionally) use a Redirect Prefix to redirect the call to another outbound route. What you cannot do is use is use a Redirect Prefix in front of the pattern on ALL your outbound routes. One route should contain a pattern that exactly matches whatever the caller dials. Then you can have other routes that include the same pattern, but with a Redirect Prefix.
Note that saying that the pattern must be matched exactly does not mean that you cannot add or strip digits in your primary route - it just means that one route must have a pattern that exactly matches what the user actually dials. So if your users dial 1NXXNXXXXXX but all of your providers only want to see the last ten digits, you could use 1|NXXNXXXXXX in your primary route, and then if you want to use a Redirect Prefix of 00009 with some extensions, you'd have another route with the pattern 000091|NXXNXXXXXX (note that in real-world use, it's probably better to strip the leading "1" at the trunk level, because different providers have different requirements. But this is just an example to clarify what is permissible).
Another point: When a call is denied and a prefix is prepended, the call is then sent back to the from-internal context, as if the user had dialed the call with the prefix prepended. While it's normally expected that the system will try to match the modified number using a route, so that the call will be sent out on a different trunk (or group of trunks) for that particular user, that doesn't necessarily have to be the case. Here's a trivial example:
Let's say you have to pay a charge for directory assistance calls. You have a route set up that handles nothing but directory assistance calls (matching 411, 1NXX5551212, and perhaps other patterns associated with the service). You have one user that refuses to look numbers up online or in the telephone directory, and has run up hundreds of dollars in directory assistance changes. You want to block his calls to directory assistance, but you also want to play a recording of the boss telling him that if he runs up one more cent in directory assistance charges he's fired!
So you block the calls and use a redirect prefix (I always suggest using redirect prefixes that start with several zeroes because generally speaking, no "normal" dialing pattern would ever start with more than about two zeroes). So let's say you make the redirect prefix 0000034733 (34733=FIRED on a phone keypad, it's just an example here). Now you create a Misc. Application and set the Feature Code to _0000034733. (note the underscore at the start and the period at the end - the underscore specifies that this is a pattern and the period that there will be additional digits after the prefix). Make the destination of the Misc. Application the Announcement that corresponds to the boss's recording.
Now, whenever he makes a call to directory assistance, the number he called will have the prefix prepened, and then the call will be sent back to from-internal where the Misc. Application will catch it. And note that you can use a Misc. Application in this way to send the call to almost any system destination. Coupled with a Misc. Destination, you could even reroute calls from a particular user to a particular number, to go to a different particular number. This potentially makes this module a very powerful tool in routing calls from a particular extension.
Here's another example: You can have speed dial codes that are specific to a group of extensions. For example, let's say you want to make 222 a universal speed dial on your system for the home telephone number of the department manager, but you have several departments, each with their own manager. You could make a Misc. Destination for each manager's home phone (with the manager's number in the "Dial" field), then make a Misc. Application for each (making the destination the Misc. Destination you just created), but in the Feature Code textbox use a unique prefix in front of the 222 (first manager would be 00001222, second would be 00002222, third would be 00003222, etc.).
Also create a "catch all" Misc. Application that goes to Terminate Call: Congestion, or to a "Sorry, you call cannot be completed" recording or something of that nature, and assign the Feature Code 0000000000 to it (this will only be used if a caller dials 222 from a phone that is not part of a department with a manager).
Then make a CUSTOM trunk with the following Custom Dial String (this is the only field you need to fill in): Local/0000000000@from-internal
After creating the trunk, create a new Outbound Route with 222 as the only entry in the Dial Patterns textbox, and select the Local/0000000000@from-internal trunk (created in the previous paragraph) as the only trunk choice for that route.
Finally, in the Outbound Route Permissions section of each extension's configuration page (for every extension that is part of a group with a manager that should be reachable by dialing 222), check "No" for the 222 Outbound Route, then enter the appropriate prefix of the correct manager in the Redirect Prefix text box (00001, 00002, 00003, etc.). Alternately, you can make bulk changes to entire groups of extensions at once from the Outbound Route Permissions page. Now, when a user in a department dials 222, the call to the "222" route will be disallowed, but the correct prefix will be prepended and then the call will flow through the Misc. Application/Misc. Destination pair and call out to the appropriate manager. Should someone dial 222 from a phone not part of a department, use of the Outbound Route will be allowed (unless you simply disallow it and don't specify a redirect prefix), but it will go to the Custom Trunk which will send it to the "catch all" Misc Application.
Why it doesn't work when you try to use the same dial pattern in two different routes:
Some people try to make two routes that contain identical dial patterns and wonder why the second route in never used, even when access to the first is denied.
Perhaps it will help if I explain it this way. When you place a call, irregardless of what you may or may not have done in routepermissions, Asterisk goes through your routes one by one and tries to match the number called to the patterns in your routes. It stops searching on the FIRST match it finds, and that's it - under no circumstances will it look at any other route once it's found a match. Only AFTER it has found a match (actually, only after it's already sent the call to a trunk) does it check to see if the user has permission to use that route. If yes, the call goes through, but if no, the call stops dead in its tracks (and if you haven't supplied a redirect prefix, it goes to the default destination).
Let's say you have a second route with identical dial patterns as the first. Your outbound calls will never use it, no matter what you do. Remember: Asterisk stops searching on the FIRST match, and it doesn't check to see if the user is allowed to use the route until AFTER it's made that match.
So that's the point of the redirect prefix. Let's say your first route has the pattern 1NXXNXXXXXX (not something I'd recommend unless you want to allow some really high-cost calls to the Caribbean, but it's just an example here). If in your second route you also put 1NXXNXXXXXX, that pattern will never be matched. It HAS to be unique. So what I might do is instead use something like 0001|1NXXNXXXXXX for the second route. Then when you deny access to that first route, you put the 0001 prefix in the "redirect" text entry box. Now let's say you make a call to 1-800-555-1212 from an "alternate route" extension:
● User dials 1-800-555-1212
● 18005551212 is sent to from-internal context which begins looking for a match on the number in the route dial patterns.
● A match for 18005551212 is found in a route, the one and only route that will ever be used for the number 18005551212.
● The call is then sent to the first trunk in the list associated with that route.
● One of the first things the trunk does is to determine if the user (identified by Caller ID number) is allowed to place calls via the route that the call just came from (which is still available in a variable). In this case it finds that no, the user is NOT allowed to place a call on this route, BUT that a redirect prefix of 0001 has been supplied
● The called number is then modified to be 000118005551212
● 000118005551212 is sent back to the from-internal context which begins looking for a match on that number in the route dial patterns.
● A match for 000118005551212 is found in a route (hopefully NOT the same one that would match 18005551212), the one and only route that will ever be used for the number 000118005551212.
● Because of the bar character in the dial pattern, the digits 0001 are removed from the called number - note that at this point the route has already been selected - so the number again becomes 18005551212 before being passed to the trunk.
● The call is then sent to the first trunk in the list associated with that route.
● One of the first things the trunk does is to determine if the user (identified by Caller ID number) is allowed to place calls via the route that the call just came from (which is still available in a variable). In this case it finds that yes, the user IS allowed to place a call on this route.
● The call then goes out the selected trunk - or if that trunk is busy, it will try any other trunks associated with that route.
I hope that helps you understand why you can't use the same pattern in two different routes and expect it to work. You must use a redirect prefix on at least one of the patterns so that it will be recognized as unique.
Panel (Operator Panel Layout)
The "Operator Panel Layout" Module for FreePBX
This module provides layout control of the Flash Operator Panel (FOP)
Operation
This module populates a 'panel' database table with layout information relating to FOP.
Some versions of retrieve_op_conf_from_mysql.pl will detect the existence of the 'panel' database table and use the layout information to generate the FOP
Preconditions
This module requires support for the 'panel' database table in retrieve_op_conf_from_mysql.pl . Please see Ticket #2989 for details.
Open Issues
The layout preview is crude. It gives an indication of the positioning of the layout areas but it does not attempt to simulate the FOP appearance
Author
nick.lewis[-at-]atltelecom.com
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
phpMyAdmin
phpMyAdmin
Purpose:
This module installs phpMyAdmin as a FreePBX Module.
Conditions/Prerequisites:
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBXIAF
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_phpMyAdmin
Set CallerID
Set CallerID
Adds the ability to change the CallerID within a call flow.
Set CallerID allows you to change the caller id of the call and then continue on to the desired destination. For example, you may want to change the caller id from "John Doe" to "Sales: John Doe". Please note, the text you enter is what the callerid is changed to. To append to the current callerid, use the proper asterisk variables, such as "${CALLERID(name)}" for the currently set callerid name and "${CALLERID(num)}" for the currently set callerid number.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Silent Monitor with Whisper
Silent Monitor with Whisper
This module adds feature codes to allow supervisors or administrators to spy on a user. Includes additional feature codes for whisper and private whisper modes if running Asterisk 1.4 or higher.
See Ticket 2441 for more information.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Sys Info
Sys Info
Purpose:
This module installs Sys Info as a FreePBX Module.
Conditions/Prerequisites:
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBXIAF
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_Sys_Info
Temporary Extensions
Temporary Extensions
Author's description:
I was looking around for a particular use case where
* employee visits abroad and carries a cell/mobile phone. * employee wants an extension forwarded to that phone while he is out of office. * Admin can allocate an extension and set the destination number as his call/mobile. * Admin can also set an expiry date on the extension.
Once the extension is created, calling in to the system or calling internally to that extension will forward the call using specified international routes (that are used when calling from an internal phone). When you dial the extension past the expiry date, "The extension you dialed, has expired!" message is played back to the caller (this is to stop people abusing the system).
In this case the trunk cannot be set for this particular extension alone. All calls will go through the trunks/outbound routes that are defined for calling from an internal phone. If this module proves useful for folks around, I can spend some more time and improve it as per suggestions/feedback.
Module is presently available at Ticket #3624 module submission page:
http://freepbx.org/trac/ticket/3624
U.S. Weather by Zip Code
U.S. Weather by Zip Code
Purpose:
This module installs the U.S. Weather by Zip Code program by Ward Mundy.
Conditions/Prerequisites:
This module is compatible with the security models used in the following distributions:
Fonicatec PABX
Foncordiax
PBXIAF
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/FreePBX_Module:_U.S._Weathe...
Usersets
The "Usersets" Module for FreePBX
This module provides user based access control for outbound routes
Operation
If the use of a userset is specified by an outbound route then the route will not be accessible unless the caller is listed in the userset.
If not listed the caller will hear the audible prompt "Cancelled" and the call will terminate
Within a userset there are two types of users:
(i) Users that are trusted - These users need to provide no authentication. The fact that they are calling from a trusted extension number gives them access to the outbound route.
(ii) Users that need authentication - These users need to provide authentication to demonstrate that they are who they claim to be. These users are prompted for their voicemail password before being given access to the outbound route.
If this module is enabled then it hooks into the outbound routes page (in the same way as the pinsets module). All existing usersets are displayed in a list box on the page.
Preconditions
This module expects the ext_vmauthenticate class to be in extensions.class.php as per FreePBX Ticket #2777.
If not the modules functions.inc.php will need to be modified to generate the VMAuthenticate dialplan command itself e.g.
$command = "VMAuthenticate(" .($mailbox ? $mailbox : ) .($context ? '@'.$context : ) .($options ? '|'.$options : ) .")"
Open Issues
A caller's number is tested in turn against each entry in the userset. For large usersets this can be a slow process. More time sensitive users should be put near the top of a userset list.
Author
nick.lewis[-at-]atltelecom.com
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Web MeetMe Support
Web MeetMe Support
Purpose:
This module creates the Feature Code for the Web MeetMe function in FreePBX, and provides for access to the Web MeetMe user interface from inside FreePBX.
Prerequisites:
Before this module can be used, Web MeetMe must be installed on your PBX. As of this writing, there are scripts to install Web MeetMe in three distributions:
Fonicatec PABX (Pre Installed)
Foncordiax (Pre Installed)
PBX In A Flash
Full installation instructions and a link to download the latest release can be found at:
http://www.fonicaprojects.com/wiki/index.php/WebMeetMe_Support_Module
