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Overview

The Digium Phones Add-on for FreePBX (DPAF) provides a simple solution for users wanting to configure Digium phones and DPMA with FreePBX. This addon is available from the FreePBX module repository and when installed is visible under the Connectivity category, labeled as Digium Phones:

Version

The features outlined here are available in the 13.0.7.1 release of the Digium Phones Addon for FreePBX (DPAF) and with DPMA version 3.4.1.

Note that some versions may only currently be available when running FreePBX with the Edge repository enabled.

Changes

 Changes

13.0.7.1

  • Correct issue with logos for D80
  • Correct issue with Send to VM permission
  • Correct issues with VM forward permission
  • Correct issues with Monitor permission
  • Correct issues with Intercom permission
  • Introduce dynamic BLF Items generation to improve behavior for all phones, and to provide Favorites support for D80

13.0.6

  • Integrated licensing code, removing need for separate digium_addons FreePBX Add-on.

13.0.5.1

  • Embed license register code

13.0.5

  • Address FREEPBX-9054, prevent license registration when DPMA is not installed

13.0.4

  • Address FREEPBX-9976, support for multiple parking lots as provided by Parking PRO

13.0.3

  • Add support for configuring TCP/TLS transports
  • Add support for D6x model phones
  • Add handles to manipulate:
    • SDES SRTP
    • D6x Logos
    • Multicast Page Listeners
    • 802.1X authentication

13.0.2

  • Add support for PJSIP configurations

13.0.1

  • Update for new FreePBX 13 core functionality

2.11.3.2

2.11.3.1

  • Remove en_US translations

2.11.3.0

2.11.2.9

  • Correct error on empty array

2.11.2.8

  • Now requires DPMA 2.2+ and firmware 2.0+

2.11.2.6

  • Add additional fixes to detect proper Astman

2.11.2.5

2.11.2.4

2.11.2.3

  • Add input sanitizing

2.11.2.2

  • Added options for default_fontsize and call_waiting_tone
  • Address locale and device_id issues

2.11.2.1

  • Fix a class loading problem

2.11.2.0

  • Support DPMA 2.1, D45 custom logos, tie phone PIN to voicemail option, and allow users to download any firmware package (not just the latest). 
  • Address bug with adding custom ringtones to a phone exclusive of any special alerts

2.11.1.1

  • Update to GPLv3 license and move all copyright into proper functions.inc.php file

2.11.1.0

  • Firmware, Ringtones and Custom Applications have been moved into a separate directory that prevents them from being destroyed on module upgrade.

2.11.0.10

  • Abstract some internal functions

2.11.0.9

  • When firmware download errors occur, provide improved error reporting

2.11.0.8

  • When reloading, explicitly skip astman

2.11.0.7

  • Improvements to speed up retrieve_conf have been made

2.11.0.6

  • A missing firmware directory no longer causes the add-on to fail

2.11.0.5

  • Fixes for Custom Apps, External Line edit, Download and select firmware, Queue members and Status entries

2.11.0.4

  • Data was not being saved on the General settings page
  • Drag and drop for phonebooks and networks now works properly

2.11.0.3

2.11.0.1

  • Module moved to FreePBX repository

2.10.0.9.3

  • Styling and formatting changes to conform to FreePBX standards

2.10.0.9.1

  • Address PC Port QoS errata from 2.10.0.9

2.10.0.9

  • Added support for setting Phone Locale

  • Added support for manipulating VLAN tagging of phone's PC port on Networks tab (requires phone firmware 1.3.2 or greater)
  • Fixed bug with VLAN Discovery Type setting
  • Known Errata:
    • PC Port QoS setting is not properly set for phones

2.10.0.8.1

  • Firmware wasn't being stored in the proper location; this has been addressed

2.10.0.8

  • Massive rewrite and reorganization from previous versions made available with AsteriskNOW 2.x.

 

QuickStart

Install AsteriskNOW or FreePBX

Installation instructions for AsteriskNOW are available on the Installing AsteriskNOW wiki page. The steps there will guide you through the installation and login process, as well as remind you to change your password from the default admin/admin.

Load DPAF to check for DPMA

Recent versions of DPAF will check for the presence of DPMA before allowing you to proceed.  If DPMA is already installed on your system, you may skip this instructional section and move on.  If DPMA is not installed, you will see a screen similar to the following:

In the event that you see this screen, open a root terminal to your FreePBX system and perform the following:

yum install asterisk13-res_digium_phone

If your FreePBX system indicates that asterisk13-res_digium_phone cannot be found in the yum repository, please follow the manual instructions for DPMA Installation.  Take care to install the correct version of DPMA.  Recent versions of FreePBX utilize the "-bundled" version of PJSIP, so you should use the "-bundled" version of DPMA.

You will then need to restart Asterisk.  To restart Asterisk from the console, perform:

fwconsole restart

Register DPMA 

Once you browse to DPAF and it has checked and ensured that DPMA is installed, you will see the following screen:

To use DPMA, registration is required. To register DPMA, a free, no-cost license key is required and may be obtained from Digium directly.  You may either click the "Get Free License" link from within DPAF as shown above, or you may go directly to the Digium store - http://store.digium.com/productview.php?category_id=196&product_code=804-00032. After signing in to your Digium account, adding a DPMA license to your cart and checking out, you will receive an e-mail that contains the key, such as:

Hello Digium Customer,

The following is your key for Digium Phone Module for Asterisk:

Key:

DPMA-000000000000


Instructions for using your key may be found at:

         http://downloads.digium.com/pub/telephony/res_digium_phone

If you have followed all of the instructions in the README file and are unsuccessful or have other installation issues, please open a case with Digium support via the web:

         http://digium.com/logcase

Thank you for your business.

- Order: W0000000 -

Once you have received this e-mail, you may input the key, along with other details as required into DPAF.  Upon inputting all of the necessary details and clicking the submit button, you will see a license agreement.  Please read it, and if acceptable, click the Accept button.  Upon doing so, you will see the following screen:

Congratulations!  DPAF is now registered and ready to use!

 

Add an Extension

Next, add a PJSIP extension to FreePBX using the Applications->Extensions tool:

Choose an extension, e.g. 101, and a display name, e.g. Malcolm Davenport, on the General tab:

Then, under the Voicemail tab, enable it and set a voicemail password:

Then, under the Other tab, in the Digium Phones Contacts Options heading, you can, optionally, input more specific contact information about the user, such as their prefix; first, middle, last, and suffix names; their organization, job title and location; their e-mail address and any note to attach to the contact.

Also on the Other tab, under the Digium Phones Line options, you can enter a Line Label (what will appear beside the line key button on the phone) and a Digit Mapping for the line:

Finally, at the bottom of the page, press the submit button.

Now, to Apply your changes, press the Red Button to when it appears:

and then boot your Digium phone.

Upon booting, your Digium phone will attempt to contact your Asterisk server using mDNS discovery. If the phone is able to find the server using mDNS discovery, and there are no other provisioning servers on the network, the phone will contact the server and display a listing of available phone extensions, including the one that you made by following the above steps:

Press the "Select" button, and your phone will be configured as the selected extension, e.g.:

For more information about configuring Digium phones using the FreePBX configuration tool as well as an explanation of all its options, please read other sections of this wiki page.

Concepts

Easy Mode

The Digium Phone FreePBX Addon operates in Easy Mode On, or Easy Mode Off. When Easy Mode is enabled, the Addon will create and remove Phone Configurations automatically, based on the presence of Extensions within the system. Every time a new Extension is added or removed to/from FreePBX, if Easy Mode is enabled, the Digium Phones Addon will add or remove a Phone Configuration. Phone Configurations added will show up in the available list of phones from the phone's boot menu. Phone Configurations removed will not show in the list, and any phones actively using a Phone Configuration that has been removed, because the underlying extension was removed, will drop their configurations.

Easy Mode also automatically assigns Phone Contacts and Rapid Dial keys. When Easy Mode is enabled, every phone has every other phone in its Contacts list and on its Rapid Dial keys.

Easy Mode, as a configuration, is most useful for small systems that only require basic functionality. If any advanced configuration is required, Easy Mode cannot be used. When Easy Mode is disabled, additional features are available and additional requirements on the Administrator are levied.

When Easy Mode is disabled, Phone Configurations are not automatically tied to the availability of a FreePBX Extension. Unlike Easy Mode On, Easy Mode Off does not automatically add a new Phone Configuration every time a FreePBX Extension is added; nor does it delete a Phone Configuration every time a FreePBX Extension is removed. The administrator is responsible for adding, deleting, and setting the options for Phone Configurations as they see fit.

Easy Mode Disabled allows for additional phone authentication methods - MAC Address and Phone PIN, both of which are configured in expanded Phone Configuration options while Easy Mode is Disabled.

Disabling Easy Mode does not disable the Internal Phonebook. An internal phonebook of all available extensions on the system is still maintained and updated. But, disabling Easy Mode does allow configuration of customized phonebooks as well as assignment of individual and multiple phonebooks to Phone Configurations.

General Settings

The General Settings tab contains settings that apply to DPMA's configuration, notably its authentication methods as well as its broadcast address for service discovery. The General Settings tab also maintains the Easy Mode toggle.

The following options are available for configuration on the General Settings tab:

Option

Values

Description

Easy Mode

Yes / No

Sets whether the Digium Phones FreePBX Addon is operating in Easy Mode or Not. Defaults to Yes.

Global PIN

Integer; e.g. 10101019

If "Require Global PIN for user list" is set to "Yes," sets the PIN that must be entered on a phone, at boot, in order to retrieve the list of available phone configurations. If "Phone Authentication Method" is set to "Global PIN," sets the PIN that must be entered on a phone, at the userlist screen, in order to request a particular Phone Configuration - note that if the Global PIN has already been entered to authenticate to retrieve the list of available Phone Configurations, that it will not required a second time in order to request a particular phone configuration. No default.

Require Global PIN for user list?

Yes / No

Defines whether or not the Global PIN must be entered on a phone in order to retrieve the list of available Phone Configurations. Defaults to No.

Phone Authentication Method

None, Phone MAC Address, Phone PIN, Global PIN, None

Defines the method by which a Phone with authenticate. When Easy Mode is enabled, the available options are None, where no authentication is required and Global Pin, where the Global PIN must be entered. When Easy Mode is disabled, the Phone MAC Address and Phone PIN options are also made available. Use of these two options requires entry of a Phone MAC Address or a Phone PIN into the Phone Configuration in the Phones tab. Defaults to None.

Internal Phonebook Sort Order

Extension, Name

Defines the sorting order of Rapid Dial key contacts for the Internally created phonebook. Defaults to Extension.

The following options are available under the Advanced Options section of the General Settings tab:

Option

Values

Description

Active LocaleNone, de_DE, en_AU, en_CA, en_GB, en_US, es_ES, es_MX, fr_BE, fr_CA, fr_FR, it_IT, nl_BE, nl_NL, pt_BR, pt_PTIf set other than none, will default all phones to the configured locale settings.

mDNS Service Name

String, e.g. Asterisk

Sets the multicast service name that phones discovering DPMA over Multicast will see on their boot menu

mDNS Discovery Address

IPv4 Address or Hostname

Sets the contact address for this server that will be advertised over Multicast. This should be set to the address on which this server responds to SIP requests.

mDNS Discovery Port

Port, as an Integer; e.g. 5060

Sets the port for this server that will be advertised over Multicast. This should be set to the port on which this server responds to SIP requests.

mDNS Discovery TransportUDP, TCP, TLSSets the transport type that will be advertised over Multicast. This should be set to a transport type on which this server responds to SIP requests.

External Lines

External Lines are SIP configurations that point to a server other than this one. This differs from normal Lines, that are SIP configurations local to this server. External Line configurations are created in this tab and then later applied to a particular phone in the Phones tab. An External Line configuration may exist on only one Phone at a time.

The External Lines tab allows for viewing, editing, and deleting of External Line configurations. When editing or adding an External Line, the following options are available:

Option

Values

Description

Line Name

string, e.g MyExternalLine

A descriptive name for the External Line, used in other tabs to reference this external line configuration

User ID

string, e.g. bob1234

This external line's SIP username.

Auth Name

string, e.g. bob1234

SIP authorization name.

Secret

string, e.g. mymagicpassword

The SIP secret this external line should use

Server Address

string, e.g. otherpbx.example.com

The address this external line should contact for registration and outbound calls

Server Port

integer, e.g. 5060

The port this external line should contact for registration and outbound calls. Defaults to 5060.

Transport

UDP, TCP, TLS

The transport type used for registration and calling to/from the server. Defaults to UDP.

Caller ID

string, e.g. Robert Paulson

Caller ID field to use for this external line

Register

Yes, No

Whether or not to send REGISTER for this external line. Defaults to Yes.

The following options are available under the Advanced Options section when adding or editing an External Line:

Option

Values

Description

Alternate Server Address

string, e.g. backuppbx.example.com

The address this external line should contact for registration and outbound calls if the primary server is not available

Alternate Server Port

integer, e.g. 5060

The port this external line should contact for registration and outbound calls if the primary server is not available.

Alternate Server Transport

UDP, TCP, TLS

The transport type used for the alternate server. Defaults to UDP.

Networks

Digium phones have a concept of Networks. When a Digium phone boots, and it receives its IP address from a DHCP server, if the address were 192.168.0.5, the phone would consider that a separate network than if the phone received an address of 10.0.0.5. The concept of Networks allows Digium phones to be configured to contact the server differently, depending on the Network on which it might find itself.

Whereas a phone booting to a 10.0.0.5 address might be inside of a company, where the PBX address is 10.0.0.1; the phone might, when taken offsite to a home location, receive an address of 192.168.0.5 from a home router, and contact the PBX via a different address, i.e. external-pbx.example.com.

The Networks tab allows for creation and manipulation of Networks, that can be applied to phones on the Phones configuration tab. A Default Network, with a CIDR of 0.0.0.0/0 is always assigned to phones so that if the phone comes up on an otherwise undefined network, it has a Network definition to fall back upon.

When editing or adding a Network, the following options are available:

Option

Values

Description

Network Name

string, e.g. My Fancy Network

A named identifier for the Network

Network CIDR

CIDR network

A CIDR-formatted network address

Registration Address

IP address or Hostname

Sets the address to which phones should register when this Network is active

Registration Port

integer, e.g. 5060

Sets the port that the phones will use when contacting the registration server when this Network is active.

TransportUDP, TCP, TLSSets the transport the phone will use when contacting the registration server when this Network is active

The following options are available under the Advanced Options section when adding or editing a Network:

Option

Values

Description

File URL Prefix

URL, e.g. http://pbx.example.com/files

Specifies the URL prefix the phone module should use to tell the phones where to retrieve files. For AsteriskNOW, this should be http://\[server IP]/admin/modules/digium_phones/firmware_package

Alternate Registration Address

IP address or Hostname

Optional. If defined, the address to which phones will maintain a backup registration. If the primary server becomes unavailable, calls will be directed to this alternate host.

Alternate Registration Port

Port, as an Integer; e.g. 5060

Optional. If defined along with alternate_registration_address, the port to be used for the backup registration.

Alternate TransportUDP, TCP, TLSOptional. Sets the transport the phone will use when contacting the alternate registration server when this Network is active.

NTP Server

hostname, IP address, e.g. ntp.example.com

Defines the NTP server to which phones will synchronize themselves. Digium maintains an NTP server that can be used by Digium phones at 0.digium.pool.ntp.org

Syslog Level

Disabled, Debug, Warning, Error, Info

Enables syslog-ing for phones on this network and sets the level. Defaults to Disabled.

Syslog Server

hostname, IP address, e.g. syslog.example.com

If Syslog is enabled, sets the server address to which log messages are sent

Syslog Port

port, as an integer; e.g. 514

If Syslog is enabled, sets the port to which log messages are sent

Network VLAN Discovery

LLDP, None, Manual

Sets the VLAN discovery method for the phone's network (LAN) port. VLAN can be assigned by LLDP (Default), manually, or disabled.

Network VLAN ID

integer

If Network VLAN Discovery is set to Manual, sets the VLAN ID to which the phone will bind its network (LAN) port

Network VLAN QoS

integer (0-7)

If Network VLAN configuration is set to Manual, sets the VLAN QoS

PC VLAN IDintegerSets the VLAN ID that the phone will use for the PC port
PC QoSinteger (0-7)Sets the PC port VLAN QoS

Signaling DSCP

integer, e.g. 24

Sets the DSCP bit for the phone's SIP signaling packets

Media DSCP

integer, e.g. 46

Sets the DSCP bit for the phone's RTP Media packets

Logos

Digium phones support custom idle screen logos. By default, and in Easy Mode, the Asterisk logo is displayed. Custom logos may be uploaded when Easy Mode is disabled in this configuration tab and assigned to phones in the Phones configuration tab.

Logos for D40, D45, D50 and D70 phones should be in PNG format, 8-bit, not transparent, less than 10k in file size.
For D40, D45 and D50 phones, the logo should measure no more than 150x45 pixels.
For D70 phones, the logo should measure no more than 205x85 pixels.

Logos for D60, D62, and D65 phones should be in PNG format, not transparent, less than 10k in file size and no more than 205x85 pixels.

Logos for D80 phones should be in PNG format, color, not transparent, and no more than 800x1280 pixels.

When editing or adding a Logo, the following options are available:

Option

Values

Description

Logo Name

string

A named identifier for the Logo

Phone Model

D40, D45, D50, D60, D62, D65, D70, D80

The phone model to which the logo will apply.

The Choose File form button brings up a file-select dialog that, when Save'd, will upload the logo.

Phone Applications

The Phone Applications tab contains controls for the built-in phone Queues and Status applications as well as for loading and controlling Custom applications.

Queues

Digium phones, when used with DPMA, have a built-in Queues application that allows for interaction with Asterisk's app_queue queue application as used in FreePBX.

The application provides 3 levels of permission: status, overview and details; each of which encompasses the previous permission's capabilities. The status view allows agents to manipulate their Pause status and their login/logout status on a per-queue basis or for all queues. The overview level allows agents or managers to also view, per queue, the number of members logged in / total assigned to a queue, the number of members on a call, the number of callers waiting, the total number of calls into the queue, the number of answered calls, and the number of abandoned calls. Additional statistical information within the overview section of the application is only available when Digium phones are used with Switchvox. The details level allows agents or managers to also view, per queue, the total number of waiting calls, the number of available/total members, the callers waiting in a queue with position and wait time, and the members on-call status.

When editing a Queue, the following options are available:

Option

Values

Description

Queue Name

string

The named identifier for the Queue. This is culled from the FreePBX name of the Queue and cannot be edited here.

Member Permission

Details, Overview, Status

The permission level for the member. Members are culled from the FreePBX list of members on the FreePBX Queue editing page

Available Devices / Managers

Drag / Drop Select Box

Contains a listing of all Digium phones that are not members of the call Queue. These devices may be made managers of the queue, where they will maintain Details-level permissions, but they won't receive calls, as they are not members of the queue.

Status

When used with DPMA, Digium Phones are capable of seeing both device state and user status . Device state is simply the device state one can subscribe to over SIP SUBSCRIBE, that maps directly to a hint in the diaplan. User Status is an entirely new concept to Asterisk, and expands upon the usage of dialplan hints, allowing them to represent both device state and user presence at the same time. Digium Phones not connected to DPMA are capable of only Available and DND (Phone returns 486 to Asterisk) status. Digium Phones using DPMA are capable of much more, with a Status application that allows users to change their presence on the server, opening up new methods for call routing based on user-presence, and not merely device presence.

New to FreePBX is the Presence State capability that builds upon Asterisk's own internal support for Presence.  Each Status is defined as one of five primary types: Available, Do Not Disturb, Away, Extended Away, or Prefer Chat. Additionally, each status can be assigned a number of Sub Statuses, that are represented in the phone's Status application and can be seen by other phones subscribed to one's own presence.  If this module is present within FreePBX and is enabled, the Digium Phones Addon for FreePBX will read presence settings from it and make those available to Digium phones.  In this case, a screen like the following will be seen:

It is highly recommended to use the built-in Presence State capabilities of FreePBX.  If the module is not present and enabled within FreePBX, there will be no linkage of statues to dialplan actions, and custom Statuses, that are phone-specific, can be configured manually.  I

When adding or editing a Status, only where the FreePBX Presence State module is not installed, the following options are available:

Option

Values

Description

Status Name

string

A named identifier for the status. This is not reflected on the phone, it is only for use within FreePBX for assignment to phones on the Phones tab

Status Type

Available, Do Not Disturb, Away, Extended Away, Prefer Chat

The actual status type for this Status. This affects the status icon associated with the Status.

Send-486

Yes, No

If enabled, directs the phone to return a 486 to Asterisk when called if it is in this Status. Within FreePBX it's a good idea to enable this for DND or any other status where you'd like the call to traverse normal FreePBX "unavailable" dialplan logic if the phone is in said status.

Sub Status

String

Allows for definition and assignment of Sub Statuses to the Status

Custom

Digium Phones are rich development platforms for building custom applications that administrators can load onto phones using the Custom tab. For more information about writing your own applications for Digium phones, visit https://wiki.asterisk.org/wiki/display/DIGIUM/Phone+API+Documentation to see examples and read about the API.

When adding or editing a Custom application, the following options are available:

Option

Values

Description

Custom Application Name

string

A named identifier for the custom application

File

Upload Selection Box

The .zip file containing the application, uploaded via browser

Automatically Start Application

No, Yes

Defines whether or not the application is started by the phone at boot. Defaults to No.

Custom Key

string

Sets a custom key, matched with a Custom Value, that is passed as an argument to the application when it's loaded. Optional. More than one Custom Key/Value pair may be passed.

Custom Value

string

Sets a custom value, matched with a Custom Key, that is passed as an argument to the application when it's loaded. Optional. More than one Custom Key/Value pair may be passed.

Ringtones

Digium phones support custom ring tones. Custom ring tones uploaded via this tab can be applied to phones using the Phones tab.

Ringtones should be 16-bit, 16kHz raw signed linear audio files. .wav files are not acceptable. If a .wav file is used, distortion may be heard in the audio as the phone plays back the ringtone. In order to generate a raw signed linear audio file from a .wav file, the sox utility, or any other audio manipulation utility that can handle .wav and raw signed linear files can be used.

To convert a .wav file into an appropriate signed linear file, using the sox utility, the following command-line example should be used:

sox -V myfile.wav -t raw -r 16000 -s -w -c 1 myfile.sln
Icon

Downsampling higher bitrate files, e.g. 48kHz, to lower bitrate files, e.g. 16kHz as used for ringtones, is acceptable. Upsampling lower bitrate files to higher bitrates, e.g. 8kHz to 16kHz, will produce files that do not sound pleasing.

When adding or editing a custom Ringtone, the following options are available:

Option

Values

Description

Ringtone Name

string

A named identifier for the ringtone.

Upload Ringtone

Choose File selection button

Displays a file upload box

Alerts

Digium phones can be configured with custom Alerts. Alerts define an Alert-Info header, a ringing type and a ringtone to be used by the phone. Alerts allow complete customization of the phone's ringing behavior. Within FreePBX there isn't a great deal of flexibility for setting up alerts except for Paging.

Alerts defined on this tab are applied to phones on the Phones tab.

When adding or editing an Alert, the following options are available:

Option

Values

Description

Alert Name

string

A named identifier for the Alert

Alert Info

string, e.g. intercom

The string passed as part of the Alert-Info header that the phone should expect. If the Alert-Info is "intercom" then the phone would expect an Alert-Info header like: "Alert-Info: <intercom>"

Ringing Type

normal, answer, ring-answer, visual

Sets the type of ringing to be used when the Alert is triggered.

Ringtone

Alarm, Chimes, Digium, GuitarStrum, Jingle, Office2, Office, RotaryPhone, SteelDrum, Techno, Theme, Tweedle, Twinkle, Vibe, or one of the custom Ringtones

Sets the ringtone to be used when the Alert is triggered.

Multicast Page

Digium phones can be configured to listen to multicast addresses for overhead paging audio.  To configure these listeners, use the Multicast Page tab of the Add-on.

When adding or editing a Multicast Page listener, the following options are available:

Option

Values

Description

Page Name

string

A named identifier for this listener, which will show in the phone's status bar during audio playback

Multicast Address

IPv4 address string

The address at which to expect multicast RTP 

Multicast Port

port (1-65536) as integer

The port at which, combined with the address, to expect multicast RTP

Priority

integer, 1-10

The priority for this listener. Lower (1) priorities take precedence over higher (10) priorities.

Interrupt0, 1Whether or not to interrupt and place on hold any in-progress calls when audio is received by this listener

802.1X

Digium phones can be configured for 802.1X pass-through and auto-logoff as well as various types of authentication that they can perform.  This is managed on the 802.1X tab.

When adding or editing an 802.1X configuration, the following options are available:

Option

Values

Description

Name

string

A named identifier for this 802.1X Configuration

Passthrough

boolean

Enables or disables 802.1X pass-through support from the PC port to the LAN port of the phone

EAPOL on Disconnect

boolean

When enabled, the phone will watch for EAPOL requests from PC-port attached devices and will send a logoff upstream on their behalf when they disconnect

Method

none, EAP-MD5, EAP-TLS, PEAP-MSCHAP, PEAP-GTC, TTLS-MSCHAP, TTLS-GTC

Sets the 802.1X authentication method for this configuration

IdentitystringSets the identity
Anonymous IdentitystringSets the anonymous identity
PasswordstringSets the password
Client Cert URLURL of the client certificateSpecifies the URL at which the phone can retrieve its 802.1X client certificate
Client Cert ValueLocal name for the client certificateSpecifies a local value the phone should use for naming the client certificate
Root Cert URLURL of the CA fileSpecifies the URL at which the phone can retrieve its 802.1X CA certificate
Root Cert ValueLocal name for the CA fileSpecifies a local value the phone should use for naming the CA certificate

Phonebooks

Digium phones come with a Contacts application that reads Phonebooks. Phonebooks are a collection of contacts, a named entity that is dialable via some number. Phonebooks are applied to phones on the Phones tab.

Phonebooks are first added and named. Once phonebooks exist, entries, either Internal or External may be added to the phonebook.

When adding an Internal Entry, the following options are available:

Option

Values

Description

Extension

A drop-down selector that contains all of the FreePBX extensions

An extension local to this system and that is contained in the FreePBX Extensions listing

Has Voicemail

Yes, No

Defines whether or not this extension has a voicemail box. If this contact has a mailbox, then it will display a "...vm" key when viewed. FreePBX does not currently allow for appropriate dialplan construction to deal with this key. Defaults to No.

Can Intercom

Yes, No

Defines whether or not this contact can be Intercommed. If this contact can be Intercommed, and if this contact is part of the phone's Rapid Dial keys, then when the contact's details are viewed, and when the contact is not on the phone, an Intercom action will be displayed. The Digium Phones Addon does not currently implement FreePBX dialplan logic to support this option. Defaults to No.

Can Monitor

Yes, No

Defines whether or not this contact can be Monitored. If this contact can be Monitored, and if this contact is part of the phone's Rapid Dial keys, then when the contact's details are viewed, and when the contact is on the telephone, a Monitor action will be displayed. The Digium Phones Addon does not currently implement FreePBX dialplan logic to support this option. Defaults to No.

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For more information about dialplan construction to properly handle the Has Voicemail, Can Intercom, and Can Monitor capabilities, see https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+phones+when+used+with+the+DPMA#DigiumphoneswhenusedwiththeDPMA-DigiumphonesusedwithorwithouttheDPMA

 

When adding an External Entry, the following options are available:

Option

Values

Description

Extension

extension, e.g. 1234, or SIP URI, e.g. sip:123@server.example.com

The dialstring to be dialed for this contact

Label

string, e.g. The Kitchen

The label to be applied to this contact

Subscribe

Checkbox

Sets whether or not to subscribe to the contact; depends on the phonebook being part of a phone's Rapid Dial keys

Subscription URL

string, e.g. *123*400

Optional. Sets the subscription URL

Firmware

Firmware for Digium phones is distributed in packages. One package contains firmware for models D40, D45, D50, D60, D62 and D70. When the "Check for Updates" button is pressed on the Firmware tab, Digium is contacted for updated firmware.  Available firmware packages can then be downloaded.

Firmware, once downloaded, is applied to phones on the Phones tab.

The Info button provides more information about the firmware package, including listing which phones are currently using a particular firmware package.

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When Checking for Updates and downloading the updates, note that a firmware package consists of firmware for all models of phone, which means it is quite large, ~200+ megabytes; so, downloading the update will take some time, depending on your Internet connection. Don't panic if you don't see anything for a while

Phones

The Phones tab provides both visibility into phone information and presence, the ability to issue reconfigure messages to phones, as well as the phone-specific configuration options.

The top part of the page contains a table of all existing Phone configurations. This table lists the phone name, any line assignments, information about known phones (model, IP address, MAC address), the current presence state of the phone if known, actions (edit, delete, reconfigure), and a notes field.

 

When adding a new phone or editing an existing phone (options that are only exposed when Easy Mode is disabled), an array of options are available at the bottom of the page. Those options that can be configured for a phone, when Easy Mode is disabled include:

Option

Values

Description

Phone Name

string

A named identifier for the phone. This will show up in the boot menu if the user is choosing phones from the phone list

Phone PIN

integer, e.g. 101010109, or Voicemail PIN

Optional. When set, and when the General configuration option requires a phone PIN for authorization, one must enter this PIN on the phone before being able to pull the phone's configuration. If "Use Voicemail PIN" is enabled, the phone's PIN will be synchronized to the voicemail PIN of the phone's primary line.

Phone MAC Address

MAC address, e.g. 0123456789ab

Optional. When set, and when the General configuration option requires MAC for authentication, locks a phone configuration to a device matching this MAC address.

Available / Assigned Extensions

Left / Right and Up / Down Selection Box

Sets the lines (internal extensions or External Lines) and ordering applied to those lines for the phone

Available / Assigned Phonebooks

Left / Right Selection Box

Sets the phonebooks assigned to a Phone

Rapid Dial Key Phonebook

Drop-down selection box

Sets the phonebook assigned to a phone's Rapid Dial Keys.

Name Format

FirstName LastName, LastName FirstName

Sets the name format for the rapid dial keys and the phone's Contact's application. Defaults to FirstName LastName

Timezone

Drop-down selection box

Sets the phone's timezone

802.1XDrop-down selection boxSets the 802.1X configuration for a phone

Available / Assigned Networks

Left / Right Selection Box

Sets the Network(s) assigned to a phone

Available / Assigned Logos

Left / Right Selection Box

Sets the logo assigned to the phone. Note that only one logo for each model type should be assigned to a phone

Available / Assigned Alerts

Left / Right Selection Box

Sets the Alerts assigned to the phone

Available / Assigned RingtonesLeft / Right Selection BoxSets the Ringtones assigned to the phone

Available / Assigned Statuses

Left / Right Selection Box

Sets the Statuses assigned to the phone

Available / Assigned Custom Apps

Left / Right Selection Box

Sets the Custom applications assigned to the phone

Available / Assigned Multicast PagesLeft / Right Selection BoxSets the Multicast Page listeners assigned to the phone
Select Parking LotDrop-down Selection BoxSets the Parking Lot to which calls will be parked using the phone's Park soft key
Visible Parking LotMulti-Select BoxSelects which Parking Lots will be visible to the phone's Parking application

Enable Call Recording

Enabled, Disabled

Enables or Disables the phone's one-touch call recording capability. Enabled by default.

Enable Send to Voicemail

Enabled, Disabled

Enables or Disables the pone's one-touch Send to Voicemail key. Enabled by default.

Require PIN for VoicemailDisabled, EnabledWhen enabled, the user's phone PIN is required in order to open the voicemail app on the phone.

Rapid Dials on Unused Line Keys

Enabled, Disabled

Sets whether Rapid Dial keys begin at the next unused line key or at the first sidecar key. Disabled by default.

Seconds between NTP sync

seconds as integer, e.g. 86400

Defines the interval between NTP synchronization

Enable Web UI

Disabled, Enabled

Enables or Disables the phone's Web UI. Disabled by default. This should not be enabled unless directed by Support

Select Firmware

A firmware as culled from the Firmware tab

Sets the firmware package to be used for the phone

Active LocaleThe listing of the phone's built-in localesSets the phone's active locale

Default Ringtone

The listing of the phone's built-in ringtones and any custom ringtones

Specifies the phone's standard, default ringing tone

Phone Admin Password

Sets the phone's admin password used to access the web interface of the Admin settings from the phone's preferences menu. Defaults to 789.

Accept Only Local Calls

Enabled, Disabled

Sets whether to accept calls from any source or only from hosts to which the phone is registered. Enabled by default.

Enable Call Waiting ToneEnabled (Default), DisabledWhen Enabled, the phone will present a call waiting tone if another call rings while the phone is already on-call. If disabled, the phone will not provide audible indication of a waiting call.

Phone Can Override Preferences

Enabled, Disabled

Disabling this option locks phone preference settings to the DPMA supplied settings. Enabled by default.

Brightness Level

0-10

Sets the phone's brightness level.

Contrast Level

0-10

Sets the phone's contrast level.

Dim Backlight After Timeout

Enabled, Disabled

If enabled, will dim the phone's backlight after the Backlight Timeout setting's time has expired. Disabled by default.

Backlight Dim Level

0-10

If Dim Backlight After Timeout is enabled and the Backlight Timeout has been reached, sets the dim level to which the backlight will be set

Backlight Timeout

integer, as a number of seconds

Sets the activity timeout associated with the phone's backlight dimming

Default Font SizeintegerSets the default font size for the phone. Does not apply to D80.

Display Missed Call Notifications

Enabled, Disabled

Specifies whether or not the phone should display missed call notifications

Ringer Volume

0-10

Sets the phone's ringing volume

Speaker Volume

0-10

Sets the phone's handsfree speaker volume

Handset Volume

0-10

Sets the phone's handset volume

Headset Volume

0-10

Sets the phone's headset volume

Handset Sidetonevalue in dBSets the volume of the phone's handset's sidetone
Headset Sidetonevalue in dBSets the volume of the phone's headset's sidetone

Call Volume Persistent Across Calls

Enabled, Disabled

Sets whether or not the phone's call volumes are reset between calls. Disabled by default.

Prefer Handset to Headset

Enabled, Disabled

If a headset is installed, specifies whether the Handset is preferred to the Headset for answering calls. Disabled by default

Digium Phones Line Options

Line configuration options are specified in the FreePBX Extension configuration utility, on the Other tab, in a section called "Digium Phones Line Options"

Within this section, the following Line configuration options are available:

Option

Values

Description

Line Label

string, e.g. Malcolm D 123

Sets the line label displayed on the phone for this line

Digit Map

Digit Mapping, See Dial Plans

The digit mapping to use for this line.

Voicemail URI

SIP URI, e.g. sip:899@mypbx.example.com

If defined, disabled Visual Voicemail for phones and instead configures phones Msgs button to dial the configured SIP URI instead.

Transport

UDP, TCP, TLS

Sets the SIP transport used for this line. Defaults to UDP.

Media Encryptionnone, sdesIf set to SDES, instructs the phone to use SDES SRTP for media encryption

Re-registration Timeout

integer, e.g. 300

The number of seconds before re-registering

Registration Retry Interval

integer, e.g. 25

The number of seconds to wait before retrying to register after registration fails.

Registration Max Retries

integer, e.g. 5

The number of times the phone will attempt to retry registering after registration fails

Digium Phones Contacts Options

FreePBX does not offer a facility for rich contact information as is possible with Digium phones' Contacts application, so additional fields to contain this information are provided in the FreePBX Extensions configuration tool under the Other tab in the "Digium Phones Contacts Options" heading.

Within this section, the following Contact configuration options are available:

Option

Values

Description

Prefix

string

Sets the prefix title for the contact, e.g. Mr.

First Name

string

Specifies the first name for the contact

Middle Name

string

Specifies the middle name for the contact

Last Name

string

Specifies the last name for the contact

Suffix

string

Specifies a suffix for the contact's name, e.g. Jr.

Organization

string

Sets an organization for a contact

Job Title

string

Sets the job title for a contact

Location

string

Sets the location for a contact

E-mail Address

string

Sets the e-mail address for a contact

Notes

string

Allows for provision of notes about a contact, e.g. A Right Moody Geezer.

Reconfiguring Phones

When changes are made to a phone's configuration, e.g. new lines, new phonebooks, preferences settings, etc. from FreePBX, those changes are not affected on the phone. In order to affect those changes on the phone, the phone must be provided with a reconfigure notice.

This notice is provided to the phone using the Reconfigure button on the Phones tab. Phones may be reconfigured individually using the Reconfigure button in the Actions field for a specified phone, or all phones may be reconfigured using the Reconfigure All button at the top of the Phones tab.

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