DPMA Configuration

globalpin

integer; e.g. 10101019

If userlist_auth is set to globalpin, sets the PIN that must be entered on a phone, at boot, in order to retrieve the list of available phone configurations. If config_auth is set to globalpin, 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 globalpin 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. If group_pin is used for phones, and userlist_auth and config_auth are set to globalpin, then entry of the globalpin will show all phones for all groups

userlist_auth

disabled, globalpin

Specifies the kind of authentication required to retrieve the list of available phone profiles from the provisioning server. Options are "disabled," where no authentication is required to retrieve the list of phone profiles; and "globalpin," where entry of the Global PIN or a group_pin is required in order to retrieve a list of phone profiles. If the globalpin is entered by a phone user to retrieve the user list, all phones configured in the DPMA will be shown. If, instead, a group_pin is entered, only the phones with matching group_pins will be shown.

config_auth

mac, pin, globalpin, disabled

Specifies the kind of authentication required to retrieve a phone's configuration from the provisioning server. Options are "mac," where the MAC address of the device requesting the configuration must match the phone profile, and if it does, the phone will be automatically provisioned with the matching phone profile; "pin" where the entered PIN must match the phone profile; "globalpin," where the entered PIN must match the Global PIN or a group_pin; and disabled, where profiles are served without authentication - with this setting, any phone can pull any phone profile defined in this configuration file with no authentication challenges.

mdns_address

IP address or Hostname

The registration address provided by DPMA to Avahi. This is the address to which phones discovering DPMA by using mDNS will contact for SIP communications

mdns_port

Port, as an Integer; e.g. 5060

The registration port provided by DPMA to Avahi. This is the port to which phones discovering DPMA by using mDNS will connect for SIP communications

service_name

string, e.g. Digium Phones Config Server

The name of the service to be used in Avahi service discovery. Defaults to "Digium Phones Config Server" when service_discovery_enabled, mdns_address and mdns_port are set.

service_discovery_enabled

yes / no

If set to yes, will advertise the config server using Avahi. Defaults to yes. The Discovery Service will only be enabled when registration_address and registration_port are explicitly configured.

file_directory

path, e.g. /var/lib/asterisk/digium_phones

Specifies a folder from which phones can retrieve Display Rules XML files. Defaults to /var/lib/asterisk/digium_phones

pjsip_message_context

A dialplan context addressable by PJSIP

The dial plan context used for routing PJSIP messages so that conflicts found in pjsip.conf can be avoided. Defaults to dpma_pjsip_message_context

alias

String, e.g. MyNetwork

The name for the network, viewable in the phone's preferences menu

cidr

CIDR noted network, e.g. 198.51.100.1/24

A CIDR-formatted network address

registration_address

IP address or Hostname

Allows for explicit definition of the address to which phones should register.

registration_port

Port, as an Integer; e.g 5060

The port on which the registration server is running – the same port on which SIP is running on the Asterisk instance.

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.

file_url_prefix

URL, e.g. http://10.10.10.10/file_package_directory

Specifies the URL prefix the phone module should use to tell the phones where to retrieve firmware, logo, ringtone, contacts and blf items files. Digium phones support basic authentication, so a username and password may be passed in the URL line, e.g. http://user:[email protected]

public_firmware_url_prefix

URL

Specifies the URL prefix the phone should use, if public_firmware for a firmware profile is set to yes, to download firmware if the phone fails to retrieve a given firmware from the location specified in its firmware profile. This will default to a publicly accessible site hosted by Digium (address: http://dphone.dl.digium.com/firmware/asterisk/) if it isn't specified manually. When using firmwares from a public firmware repository, the path will always use the following pattern: public_firmware_url_prefix/VERSION/VERSION_MODEL_firmware.eff

ntp_server

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

Defines the NTP server to which phones will synchronize themselves

syslog_server

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

Defines a remote server to which syslog message are sent. Defaults to none.

syslog_port

port, as an integer, e.g. 514

Port to which syslog messages are sent. Defaults to 514.

syslog_level

debug, error, warning, information

Logging level for syslog messages

network_vlan_discovery_mode

NONE, MANUAL, LLDP

Sets use of none, manual, or LLDP discovered VLAN and, if MANUAL, defines the network; defaults to LLDP

sip_qos

Integer (0-7)

Sets the SIP QoS level. Defaults to 3

rtp_qos

Integer (0-7)

Sets the RTP QoS level. Defaults to 6

network_vlan_id

Integer (0-4095)

Sets the VLAN identifier of the network port

pc_vlan_id

(Integer (0-4095)

Sets the VLAN identifier of the PC port

pc_qos

Integer (0-7)

Sets the QoS level for PC port traffic. Defaults to 0.

sip_dscp

Integer (0-63)

Specifies the DSCP field of the DiffServ byte for SIP Signaling QoS, defaults to 24

rtp_dscp

Integer (0-63)

Specifies the DSCP field of the DiffServ byte for RTP Media QoS, defaults to 46

network

entity defined as "network" type in res_digium_phone.conf.

Assigns a network profile to a phone. More than one network can be assigned to a phone by using multiple network lines

firmware

entity defined as "firmware" type in res_digium_phone.conf

Assigns a firmware to a phone. More than one firmware can be assigned to a phone, provided that each firmware is for a different model, by using multiple firmware lines

mac

A MAC address, e.g. 0123456789ab

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

pin

integer, e.g. 10101019; or "voicemail"

Optional. When set, and when the general config_auth option requires PIN, one must enter this PIN on the phone before being able to pull the phone's configuration. If set to "voicemail" will tie the phone's pin to the voicemail account password, from voicemail.conf, as defined for the SIP peer, for flat-file configurations without externally maintained passwords only, used for the phone's primary internal line.

group_pin

integer; e.g. 101010109

Optional. When set, and when the general config_auth and userlist_auth options are set to globalpin, assigns this phone a group pin. When the group pin is input by the user, all phones with a matching group_pin will appear in the list of available phone profiles

line

entity defined as line in res_digium_phone.conf

Maps directly to a PJSIP endpoint entry. More than one line may be defined for a phone configuration. The first line entry defined is adopted as the phone's primary line.

external_line

entity defined as external_line in res_digium_phone.conf

Maps directly to an external_line defined in this configuration file. External line are lines not defined by SIP peers in sip.conf and generally do not register to this instance of Asterisk. If an external line is defined as the primary extension for a phone, many of the advanced phone application features will be disabled

application

Application section name identifier

Applies application-specific settings to a phone configuration. Multiple application options can exist in a single phone configuration

config_file

file, e.g. mycustomconfig.xml

The phone module automatically generates a configuration file for each phone based on the lines assigned to it; but, it is possible to supply a custom configuration file instead. Using this option will direct the DPMA to serve up the specified file, as found in the file_directory defined directory, to the phone. Note that using a custom configuration file, as opposed to the provisioning generated by the DPMA, precludes the phone's use of DPMA-specific applications, e.g. voicemail, parking, user status, etc. This option allows users to make use of the DPMA's mDNS provisioning capabilities, providing a simpler alternative to HTTP and Option 66 provisioning, but sacrifices the DPMA-specific features.

full_name

string, e.g. Bob Johnson

The full name of the person who will be using this phone, and what will appear in the user list that the phone pulls

contact

file, e.g. contacts.xml

An XML file, retrievable from the file_url_prefix, containing a list of contacts to serve to the phone. Multiple contact options can exist in a single phone configuration.

blf_contact_group

string, a group from contacts

Every contacts xml file will have at least one group defined in it. This setting controls which loaded group the phone should subscribe to for its Rapid Dial (BLF) keys. If this setting is blank, the phone will not subscribe for any device state or presence updates and LED indicators will not light.

contacts_max_subscriptions

timezone

Timezone String, e.g. America/Chicago

Sets the timezone used for the clock on this phone.

ntp_resync

seconds as integer, e.g. 86400

Defines the interval between NTP synchronization

parking_exten

extension, e.g. 700

Sets the extension used for parking calls. When this option is set, and the phone has an in-progress call, it will display a "Park" softkey, allowing for one touch parking.

parking_transfer_type

blind / attended

The type of transfer to perform when parking a call using the "Park" softkey.

ringtone

context name of a type=ringtone identifier

Loads ringtones onto a phone. More than one ringtone may be loaded onto a phone.

active_ringtone

Alarm, Chimes, Digium, GuitarStrum, Jingle, Office2, Office, RotaryPhone, SteelDrum, Techno, Theme, Tweedle, Twinkle, Vibe or the context name of a type=ringtone identifier that has been loaded onto the phone using the ringtone option

Sets the active ringtone for the phone, defaults to Digium.

web_ui_enabled

no / yes

By default, when using the Digium Phone Module for Asterisk, the phone's built-in Web UI is disabled. To override this and enable the Web UI anyway, which may result in unpredictable behavior if user settings conflict with the settings provided by the DPMA, enable this option.

record_own_calls

no / yes

When this option is enabled, and the phone has an in-progress call, it will display a "Record" softkey, allowing for one touch call recording. Disabled by default.

alert

Alert section name identifier

Defines Alerts to be applied to this phone profile. More than one Alert may be applied by specifying additional alert lines.

blf_unused_linekeys

no / yes

Digium phones, by default, place BLF keys on the sidecar, not on unused line keys. To disable this behavior and allow BLF keys to start mapping from the next available unused line key, enable this option.

send_to_vm

yes / no

Disabling this option hides the "Send VM" softkey during incoming calls and the "Transfer VM" softkey inside the Contacts application during call transfers. Enabled by default.

d40_logo_file

string

The idle screen image for a D40 model phone in PNG format, 150x45 pixels, 8-bit depth, a color type without alpha transparency and less than 10k in size. Retrieved from the file_url_prefix. The D40, D45 and D50 screen size is the same; therefore it is permissible to re-use the same logo file for each.

d50_logo_file

string

The idle screen image for a D50 model phone in PNG format, 150x45 pixels, 8-bit depth, a color type without alpha transparency and less than 10k in size. Retrieved from the file_url_prefix. The D40, D45 and D50 screen size is the same; therefore it is permissible to re-use the same logo file for each.

d70_logo_file

string

The idle screen image for a D70 model phone in PNG format, 205x85 pixels, 8-bit depth, a color type without alpha transparency and less than 10k in size. Retrieved from the file_url_prefix.

ehs

auto, plantronics, jabra_iq

Sets the electronic hookswitch device type.

lock_preferences

yes / no

Enabling this option locks phone preference settings to the DPMA supplied settings. Enabling this option also hides phone preference menus for menu items that are set in the Phone profile. Disabled by default

login_password

Integer, e.g. 789

Sets the Admin Password for logging into Web UI or Admin Settings Section on Phone Menu, defaults to 789

accept_local_calls

any, host

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

display_mc_notification

yes / no

Enables / Disables display of missed calls on the phone, defaults to Enabled

brightness

Integer, 0-10

Sets the LCD screen brightness, defaults to 5

contrast

Integer, 0-10

Sets the LCD screen contrast, defaults to 5

dim_backlight

yes, no

Enable backlight dimming. When enabled, dims the screen after backlight timeout has been reached and phone is otherwise idle. Defaults to enabled

backlight_timeout

Integer (0-3200)

Time, in seconds, before backlight is set to backlight_dim_level while phone is idle; setting to 0 disables backlight timeout, defaults to 0

backlight_dim_level

Integer (0-10)

Brightness level dims to when when dim_backlight is enabled, defaults to 2.

active_locale

de_DE, en_AU, en_CA, en_GB, en_NZ, en_US, es_ES, es_MX, fr_BE, fr_CA, fr_FR, it_IT, nl_BE, nl_NL, pt_BR, pt_PT, ru_RU

Specifies the locale used by the phone, defaults to en_US. ru_RU applies only to D6x models of phones.

ringer_volume

Integer (0-10)

Sets the ringer volume, defaults to 5

speaker_volume

Integer (0-10)

Sets the speaker volume, defaults to 5

handset_volume

Integer (0-10)

Sets the handset volume, defaults to 5

headset_volume

Integer (0-10)

Sets the headset volume, defaults to 5

Sets the gain, in negative dBs, for sidetone presented on the phone's handset. Defaults to -25. Digium cautions against changing this value.

reset_call_volume

yes / no

If enabled, volume changes made during a call do not persist to the next call, defaults to disabled

headset_answer

yes / no

Sets whether to use the headset, rather than the speaker, for answering all calls, defaults to disabled

name_format

first_last, last_first

Formats the display of contact names, defaults to first_last

Sets the method of 802.1X authentication for the phone, defaults to null (none).

8021x_identity

null, string

Sets the 802.1X authentication identifier (username), defaults to null (none).

8021x_anonymous_identity

null, string, PHONE_MAC

8021x_password

null, string

Sets the 802.1X authentication password, defaults to null (none)

exten

string, e.g. 1000

When the PJSIP endpoint name is different than the actual extension used to contact this line, this option should be set to the line's dialable extension. By default it is assumed that the PJSIP endpoint is actual dialable extension, which is true for most Asterisk distributions such as FreePBX and AsteriskNOW, but is not considered a best practice for use of generic Asterisk.

digit_map

Digit mapping, see #dialplans

The digit mapping to use for this line.

line_label

string, e.g. MD 123

The line label to display on the phone for this line's line key.

mailbox

mailbox from voicemail.conf

The voicemail box associated with the line. When not set, this will default first to the mailbox defined for the PJSIP endpoint associated with the line, and second to a mailbox entry in voicemail.conf matching the name of the line. If neither of these are found and this option is not set, the line does not have a mailbox and visual voicemail will not be enabled. If PJSIP endpoints are stored using Sorcery rather than the flat pjsip.conf file, then the mailbox to which the PJSIP endpoint is assigned must be specified here, as it cannot be retrieved by the DPMA from Sorcery .

voicemail_uri

string, sip:[email protected]

If the phone's Msgs button should dial a SIP URI rather than opening the visual voicemail application, this option specifies what URI the Msgs button should dial. Setting this option on a phone's primary lie will disable visual voicemail.

outboundproxy_address

Host / IP address

The Outbound SIP proxy address this line should use

outboundproxy_port

Port as integer

The port of the Outbound SIP proxy; defaults to 5060 when outboundproxy_address is set

transport

udp, tcp, tls

SIP transport method this line should use. If not specified, the network transport is preferred. Using tls or tcp as a transport for phones attached to DPMA requires Asterisk 13.11.0 or greater.

reregistration_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

secret

string, e.g. mymagicpassword

If PJSIP endpoints are stored using Sorcery rather than the flat pjsip.conf file, then the secret for the PJSIP endpoint mapped to this line must be specified so that the Digium phone can be passed the correct PJSIP endpoint credentials.

context

Asterisk dialplan context

If PJSIP endpoints are stored using Sorcery rather than the flat pjsip.conf file, then the dialplan context to which the PJSIP endpoint is assigned must be specified so that dialplan hints can be properly created by DPMA.

callerid

caller id string, e.g. "Bob Jones" <1234>

If PJSIP endpoints are stored using Sorcery rather than the flat pjsip.conf file, then the caller ID for the PJSIP endpoint mapped to this line should be specified so that the Digium phone can be provided with a proper Caller ID.

subscribecontext

Asterisk dialplan context

Allows for overriding of the line's subscription context. If defined here will override the setting from Asterisk's PJSIP configuration

server_address

string, e.g. otherpbx.othercompany.com

The address this line should contact for registration and outbound calls

server_port

integer, e.g. 5060

The port this 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

secondary_server_address

string, e.g. backuppbx.othercompany.com

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

secondary_server_port

integer, e.g. 5060

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

secondary_server_transport

tcp or udp

The transport type used for registration and calling to/from the secondary server

userid

string, e.g. bob1234

This line's SIP username. Defaults to the line's name

authname

string, e.g. bob1234

SIP authorization name if different than userid. If blank, defaults to userid.

secret

string, e.g. mymagicpassword

The SIP secret this line should used

register

yes / no

Indicates whether or not this line should register

callerid

string

Caller ID field to use for this line

application

queue, status, parking, voicemail

The application type name

queue

string

Asterisk's queue identifier, as configured in queues.conf

membername

string

Defaults to 'fullname' from the phone type configuration. Allows customization of the queue membername as viewed in the Queues application

location

Asterisk queue member location, e.g. Local/[email protected]/n

Defines the location of the member, from app_queue.so's perspective. For setups not involving local channels, this may not be required.

member

yes / no

Defines whether or not the user of this application is also a member of the queue that will receive calls. Defaults to yes.

permission

status, overview, details

Sets the permission level for this user. Status provides only login/out/pause capabilities. Overview also provides statistical information about a queue. Details also provides information about waiting callers and on-call members.

login_exten

Asterisk [email protected], e.g. *[email protected]

If defined, sets the extension to be dialed when a user of the application executes a log in command. If not set, the DPMA will use the QueueAdd functionality directly.

logout_exten

Asterisk [email protected], e.g. *[email protected]

If defined, sets the extension to be dialed when a user of the application eecuted a log out command. If not set, the DPMA will use the QueueRemove functionality directly.

status

available, dnd, away, xa, chat, unavailable

Sets the base presence type from one of the available six types

substatus

string

Optional. Sets a substatus for a particular status, e.g. "At the movies" for the away status would result in "Away - At the Movies"

send486

yes / no

Defaults to no. If set, when this Status is set on a phone, the phone will return a 486 reject to Asterisk when called.

parkinglot

string

Sets the name of a parking lot context as defined by Asterisk in features.conf

translation

identifier of a type=voicemail section

Sets the translation set for the application

name

string

The name of the custom application, e.g. JasonParkerApp; also used for the idlescreen_softkey label

filename

string

A named .zip file containing the user's application, as reachable from the file_url_prefix.

autostart

yes / no

Sets whether or not the custom application should load when the phone boots, or the first time a user opens the application

key

value

Optional key=value parameters may be passed to the application

alias

string

An identifier for the ringtone, e.g. FancyRinger. This identifier will be present in the phone's preferences menu and in its web menu

filename

string

A named 16-bit 16kHz mono raw audio file, not a .wav file, as reachable from the file_url_prefix.

alert_info

string

The Alert-Info header that the phone should expect when this Alert is to be used.

ring_type

normal, answer, ring-answer, visual

The type of ringing behavior that the phone should present when this Alert is called.

ringtone

Alarm, Chimes, Digium, GuitarStrum, Jingle, Office2, Office, RotaryPhone, SteelDrum, Techno, Theme, Tweedle, Twinkle, Vibe, or one of the context names where type=ringtone

Defines the ringtone to be used for this Alert. If blank, then the phone will not play back a ringing tone, instead present silence.

model

D40, D50, D70

Defines the phone model for this firmware. A firmware applies to only one model; thus, to handle all 3 (currently) models of phones, 3 different firmware definitions must be created and used

version

version string identifier of firmware

Sets the version string identifier for a particular firmware. The string is the version of the firmware, not counting the final underscore, model, underscore, "firmware," and ".eff" suffix.

file

string

Sets the file location of the firmware, to be retrieved by the phone and respecting the file_url_prefix network option

public_firmware

yes, no

Whether or not we will include a fallback address (based on the network's public_firmware_url_prefix) to retrieve this firmware if the phone can not reach the file specified here. Default is no.

key

value as a string

Sets the value to be substituted for the key

address

IPv4 address string

The address at which to expect multicast RTP

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.

interrupt

0, 1

When enabled, places any in-progress calls on hold before playing back audio, and ignore the phone's local volume setting, playing back the audio at full volume. When disabled, in-progress calls will have their audio played over. Defaults to 0.