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:
The features outlined here are available in the 188.8.131.52 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.
- Correct missing trailing parenthesis in dial plan
- Re-implement BLF keys on unused line keys feature for D50 and D70 models
- 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
- Integrated licensing code, removing need for separate digium_addons FreePBX Add-on.
- Embed license register code
- Address FREEPBX-9054, prevent license registration when DPMA is not installed
- Address FREEPBX-9976, support for multiple parking lots as provided by Parking PRO
- 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
- Add support for PJSIP configurations
- Update for new FreePBX 13 core functionality
- Remove en_US translations
- Address FREEPBX-10931
- Correct error on empty array
- Now requires DPMA 2.2+ and firmware 2.0+
- Add additional fixes to detect proper Astman
- Address FREEPBX-9880
- Address items in preparation of FreePBX 13
- Add input sanitizing
- Added options for default_fontsize and call_waiting_tone
- Address locale and device_id issues
- Fix a class loading problem
- 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
- Update to GPLv3 license and move all copyright into proper functions.inc.php file
- Firmware, Ringtones and Custom Applications have been moved into a separate directory that prevents them from being destroyed on module upgrade.
- Abstract some internal functions
- When firmware download errors occur, provide improved error reporting
- When reloading, explicitly skip astman
- Improvements to speed up retrieve_conf have been made
- A missing firmware directory no longer causes the add-on to fail
- Fixes for Custom Apps, External Line edit, Download and select firmware, Queue members and Status entries
- Data was not being saved on the General settings page
- Drag and drop for phonebooks and networks now works properly
- Data was not being saved on some pages, this addresses http://issues.freepbx.org/browse/FREEPBX-6747
- Module moved to FreePBX repository
- Styling and formatting changes to conform to FreePBX standards
- Address PC Port QoS errata from 184.108.40.206
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
- Firmware wasn't being stored in the proper location; this has been addressed
- Massive rewrite and reorganization from previous versions made available with AsteriskNOW 2.x.
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:
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:
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 - https://www.sangoma.com/asterisk/software/digium-phone-module-for-asterisk/. 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:
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.
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.
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:
Yes / No
Sets whether the Digium Phones FreePBX Addon is operating in Easy Mode or Not. Defaults to Yes.
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
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:
|Active Locale||None, 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_PT||If 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 Transport||UDP, TCP, TLS||Sets 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 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:
string, e.g MyExternalLine
A descriptive name for the External Line, used in other tabs to reference this external line configuration
string, e.g. bob1234
This external line's SIP username.
string, e.g. bob1234
SIP authorization name.
string, e.g. mymagicpassword
The SIP secret this external line should use
string, e.g. otherpbx.example.com
The address this external line should contact for registration and outbound calls
integer, e.g. 5060
The port this external line should contact for registration and outbound calls. Defaults to 5060.
UDP, TCP, TLS
The transport type used for registration and calling to/from the server. Defaults to UDP.
string, e.g. Robert Paulson
Caller ID field to use for this external line
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:
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.
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:
string, e.g. My Fancy Network
A named identifier for the Network
A CIDR-formatted network address
IP address or Hostname
Sets the address to which phones should register when this Network is active
integer, e.g. 5060
Sets the port that the phones will use when contacting the registration server when this Network is active.
|Transport||UDP, TCP, TLS||Sets 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:
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 Transport||UDP, TCP, TLS||Optional. Sets the transport the phone will use when contacting the alternate registration server when this Network is active.|
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
Disabled, Debug, Warning, Error, Info
Enables syslog-ing for phones on this network and sets the level. Defaults to Disabled.
hostname, IP address, e.g. syslog.example.com
If Syslog is enabled, sets the server address to which log messages are sent
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
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
If Network VLAN configuration is set to Manual, sets the VLAN QoS
|PC VLAN ID||integer||Sets the VLAN ID that the phone will use for the PC port|
|PC QoS||integer (0-7)||Sets the PC port VLAN QoS|
integer, e.g. 24
Sets the DSCP bit for the phone's SIP signaling packets
integer, e.g. 46
Sets the DSCP bit for the phone's RTP Media packets
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:
A named identifier for the Logo
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.
The Phone Applications tab contains controls for the built-in phone Queues and Status applications as well as for loading and controlling Custom applications.
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:
The named identifier for the Queue. This is culled from the FreePBX name of the Queue and cannot be edited here.
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.
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:
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
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.
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.
Allows for definition and assignment of Sub Statuses to the Status
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:
Custom Application Name
A named identifier for the custom application
Upload Selection Box
The .zip file containing the application, uploaded via browser
Automatically Start Application
Defines whether or not the application is started by the phone at boot. Defaults to No.
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.
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.
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:
When adding or editing a custom Ringtone, the following options are available:
A named identifier for the ringtone.
Choose File selection button
Displays a file upload box
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:
A named identifier for the Alert
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>"
normal, answer, ring-answer, visual
Sets the type of ringing to be used when the Alert is triggered.
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.
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:
|A named identifier for this listener, which will show in the phone's status bar during audio playback|
IPv4 address string
|The address at which to expect multicast RTP|
port (1-65536) as integer
The port at which, combined with the address, to expect multicast RTP
The priority for this listener. Lower (1) priorities take precedence over higher (10) priorities.
|Interrupt||0, 1||Whether or not to interrupt and place on hold any in-progress calls when audio is received by this listener|
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:
|A named identifier for this 802.1X Configuration|
|Enables or disables 802.1X pass-through support from the PC port to the LAN port of the phone|
EAPOL on Disconnect
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
none, EAP-MD5, EAP-TLS, PEAP-MSCHAP, PEAP-GTC, TTLS-MSCHAP, TTLS-GTC
Sets the 802.1X authentication method for this configuration
|Identity||string||Sets the identity|
|Anonymous Identity||string||Sets the anonymous identity|
|Password||string||Sets the password|
|Client Cert URL||URL of the client certificate||Specifies the URL at which the phone can retrieve its 802.1X client certificate|
|Client Cert Value||Local name for the client certificate||Specifies a local value the phone should use for naming the client certificate|
|Root Cert URL||URL of the CA file||Specifies the URL at which the phone can retrieve its 802.1X CA certificate|
|Root Cert Value||Local name for the CA file||Specifies a local value the phone should use for naming the CA certificate|
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:
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
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.
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.
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.
When adding an External Entry, the following options are available:
extension, e.g. 1234, or SIP URI, e.g. sip:email@example.com
The dialstring to be dialed for this contact
string, e.g. The Kitchen
The label to be applied to this contact
Sets whether or not to subscribe to the contact; depends on the phonebook being part of a phone's Rapid Dial keys
string, e.g. *123*400
Optional. Sets the subscription URL
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.
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:
A named identifier for the phone. This will show up in the boot menu if the user is choosing phones from the phone list
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.
FirstName LastName, LastName FirstName
Sets the name format for the rapid dial keys and the phone's Contact's application. Defaults to FirstName LastName
Drop-down selection box
Sets the phone's timezone
|802.1X||Drop-down selection box||Sets 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 Ringtones||Left / Right Selection Box||Sets 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 Pages||Left / Right Selection Box||Sets the Multicast Page listeners assigned to the phone|
|Select Parking Lot||Drop-down Selection Box||Sets the Parking Lot to which calls will be parked using the phone's Park soft key|
|Visible Parking Lot||Multi-Select Box||Selects which Parking Lots will be visible to the phone's Parking application|
Enable Call Recording
Enables or Disables the phone's one-touch call recording capability. Enabled by default.
Enable Send to Voicemail
Enables or Disables the pone's one-touch Send to Voicemail key. Enabled by default.
|Require PIN for Voicemail||Disabled, Enabled||When enabled, the user's phone PIN is required in order to open the voicemail app on the phone.|
Rapid Dials on Unused Line Keys
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
Enables or Disables the phone's Web UI. Disabled by default. This should not be enabled unless directed by Support
A firmware as culled from the Firmware tab
Sets the firmware package to be used for the phone
|Active Locale||The listing of the phone's built-in locales||Sets the phone's active locale|
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
Sets whether to accept calls from any source or only from hosts to which the phone is registered. Enabled by default.
|Enable Call Waiting Tone||Enabled (Default), Disabled||When 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
Disabling this option locks phone preference settings to the DPMA supplied settings. Enabled by default.
Sets the phone's brightness level.
Sets the phone's contrast level.
Dim Backlight After Timeout
If enabled, will dim the phone's backlight after the Backlight Timeout setting's time has expired. Disabled by default.
Backlight Dim Level
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
integer, as a number of seconds
Sets the activity timeout associated with the phone's backlight dimming
|Default Font Size||integer||Sets the default font size for the phone. Does not apply to D80.|
Display Missed Call Notifications
Specifies whether or not the phone should display missed call notifications
Sets the phone's ringing volume
Sets the phone's handsfree speaker volume
Sets the phone's handset volume
Sets the phone's headset volume
|Handset Sidetone||value in dB||Sets the volume of the phone's handset's sidetone|
|Headset Sidetone||value in dB||Sets the volume of the phone's headset's sidetone|
Call Volume Persistent Across Calls
Sets whether or not the phone's call volumes are reset between calls. Disabled by default.
Prefer Handset to Headset
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:
string, e.g. Malcolm D 123
Sets the line label displayed on the phone for this line
Digit Mapping, See Dial Plans
The digit mapping to use for this line.
SIP URI, e.g. sip:firstname.lastname@example.org
If defined, disabled Visual Voicemail for phones and instead configures phones Msgs button to dial the configured SIP URI instead.
UDP, TCP, TLS
Sets the SIP transport used for this line. Defaults to UDP.
|Media Encryption||none, sdes||If set to SDES, instructs the phone to use SDES SRTP for media encryption|
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:
Sets the prefix title for the contact, e.g. Mr.
Specifies the first name for the contact
Specifies the middle name for the contact
Specifies the last name for the contact
Specifies a suffix for the contact's name, e.g. Jr.
Sets an organization for a contact
Sets the job title for a contact
Sets the location for a contact
Sets the e-mail address for a contact
Allows for provision of notes about a contact, e.g. A Right Moody Geezer.
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.