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XML Configuration
This section describes the formatting and options available when creating XML-based configuration files for provisioning Digium phones. Users choosing this method of configuration forgo use of the DPMA, and instead are provisioning phones for use with Asterisk versions that do not support the DPMA.
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Setting Elements
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Each <setting> element represents at least an id and value pair of attributes. Some <setting> tags may have additional attributes.
General (Login)
Option | Values | Description |
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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 |
General (Time)
Option | Values | Description |
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time_zone | Timezone String, e.g. America/Chicago | Sets the time zone for the phone |
time_source | "ntp" | Sets the time source for the phone. Currently, the only option is "ntp" |
ntp_server | Hostname or IP address, e.g. 0.digium.pool.ntp.org | Sets the NTP server to which the phone will synchronize itself, defaults to 0.digium.pool.ntp.org |
ntp_resync | Seconds as integer, e.g. 86400 | Sets the interval between NTP synchronization |
General (SIP)
Option | Values | Description | Models |
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accept_local_calls | any / host | Sets whether to accept calls from any source or only from hosts to which the phone is registered | |
transport_udp_enabled | boolean | Sets whether to enable UDP transport, defaults to 1 | |
transport_udp_port | Valid integer for ports (1-65535) | Sets the local UDP SIP port, defaults to 5060 | |
transport_tcp_enabled | boolean | Sets whether to enable TCP transport, defaults to 1 | |
transport_tcp_port | Valid integer for ports (1-65535 ) | Sets the local TCP SIP port, defaults to 5060 | |
udp_ka_interval | integer, in seconds | Sets the UDP keep alive interval, at which the phone will send CR-LF to the registered server. Defaults to 0, never. |
Preferences (Idle Screen)
Option | Values | Description |
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logo_file | value as factory / user; path as location on disk of file - /factory_asterisk.png for default and /user_image.png for custom ; url as optional location to fetch a logo; md5 as optional when url is used to determine if logo has changed to avoid re-fetching | Sets the idle screen logo, defaults to factory-asterisk.png |
display_mc_notification | boolean | Disables / Enables display of missed calls on the phone, defaults to 1 |
Preferences (Display)
Option | Values | Description |
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brightness | integer (0-10) | Sets the LCD screen brightness, defaults to 5 |
contrast | integer (0-10) | Sets the LCD screen contrast, defaults to 5 |
enable_check_sync | boolean | Controls whether the phone will perform a reconfigure when sent a check-sync SIP NOTIFY Event from the server to which it is registered. Defaults to 1. |
dim_backlight | boolean | enable backlight dimming where 1 dims the screen after backlight timeout has been reached and phone is otherwise idle, defaults to 1 |
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 1, defaults to 2 |
default_fontsize | integer (10-14) | Sets the default font size for the phone. Caution should be exercised when using this option as larger sizes will cause labels to overrun their allowed space. D40, D45, and D50 default to 10. D70 defaults to 11. |
Preferences (Localization)
Option | Values | Description |
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locale | 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 | Specifies the locale used by the phone, defaults to en_US |
Preferences (Sounds)
Option | Values | Description |
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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 |
handset_sidetone_db | Integer, e.g. -25 | Sets the gain, in negative dBs, for sidetone presented on the phone's handset. Defaults to -25. Digium cautions against changing this value. |
headset_sidetone_db | Integer, e.g. -15 | Sets the gain, in negative dBs, for sidetone presented on the phone's headset. Defaults to -15. Digium cautions against changing this value. |
reset_call_volume | boolean | If 1, volume changes made during a call do not persist to the next call, defaults to 0 |
active_ringtone | Tone ID from <tones> provided to phone | Sets the current user-selected ringtone, defaults to Digium |
Preferences (Answering Calls)
Option | Values | Description |
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headset_answer | boolean | Sets whether to use the headset, rather than the speaker, for answering all calls, defaults to 0 |
ring_headset_only | boolean | Sets whether or not to play ringing tone out the headset, defaults to 0 |
call_waiting_tone | boolean | Enabled by default, if disabled, the phone phone will not playback the call-waiting tone |
ehs | auto, jabra_iq, plantronics | Defines the Electronics Hookswitch type to support |
Contacts
Option | Values | Description |
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enable_blf_on_unused_line_keys | boolean | If 1, assigns BLFs beginning with first empty line key. If 0, assigns BLFs beginning with first sidecar key. Defaults to false. |
contacts_max_subscriptions | integer, e.g. 40 | Sets the maximum number of SUBSCRIBEs the phone will perform for contacts |
name_format | first_last, last_first | Formats the display of contact names, defaults to first_last |
blf_contact_group | Any group_name from the loaded contacts | The group_name of the contact list group to use for the rapid dial list |
Network (IP Settings)
Option | Values | Description |
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network_enable_dhcp | boolean | Disable or Enable DHCP network configuration, defaults to 1 |
network_static_ip_address | IPv4 address | Defines the network address for the phone |
network_subnet_mask | IPv4 netmask | Defines the netmask for the phone |
network_default_gateway | IPv4 address | Defines the network gateway for the phone |
network_primary_dns_server | IPv4 address | Defines the primary DNS server for the phone |
network_secondary_dns_server | IPv4 address | Defines the secondary DNS server for the phone |
Network (Virtual LAN)
Option | Values | Description |
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network_vlan_discovery_mode | value of NONE, MANUAL, LLDP; network as IP mask | Sets use of none, manual, or LLDP discovered VLAN and, if MANUAL, defines the network; defaults to LLDP |
network_vlan_id | integer (0-4095) | Sets the VLAN ID |
pc_vlan_id | integer (0-4095) | Sets the VLAN ID of the PC port; untagged traffic from the PC port to the LAN port will be tagged with this VLAN ID |
Network (Interfaces)
Option | Values | Description |
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lan_port_mode | auto, 10hd, 10fd, 100hd, 100fd, 1000fd | Sets the port speed for the phone's LAN port. "auto" will perform auto-negotiation. |
pc_port_mode | auto, 10hd, 10fd, 100hd, 100fd, 1000fd, off | Sets the port speed for the phone's PC port. "auto" will perform auto-negotiation; "off" disables the port." |
Logging
Option | Values | Description |
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log_level | error, warning, debug, information | Sets the logging level, defaults to error |
log_server | IPv4 address of syslog server | Specifies remote syslog server |
log_port | port as integer | Specifies port of remote syslog server |
enable_logging | boolean | Disables or Enables remote syslog, defaults to 0 |
Miscellaneous
Option | Values | Description |
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web_ui_enabled | boolean | Disables, Enables the phone's web user interface, defaults to 1 (Enabled) |
sip_qos | integer (0-7) | Defaults to 3. Sets the SIP signaling QOS level |
rtp_qos | integer (0-7) | Defaults to 6. Sets the RTP media QOS level |
pc_qos | integer (0-7) | No default. Sets the QOS level for traffic from the PC port to the LAN port |
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 |
802.1X
Option | Values | Description | Models |
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8021x_passthrough | boolean | Enables or disables pass-through of EAPOL packets from the PC port to the LAN port, defaults to 0 | |
8021x_eapol_on_disconnect | boolean | Enables or disables sending of EAPOL disconnect on behalf of PC-port attached device when disconnected, defaults to 0 | |
8021x_method | null, eap-md5, peap-mschap, eap-tls, peap-gtc | Sets the method of 802.1X authentication for the phone, defaults to null (none). | D40, D45, D50, D70 - eap-md5 only |
8021x_identity | null, string | Sets the 802.1X authentication identifier (username), defaults to null (none). | |
8021x_anonymous_identity | null, string, PHONE_MAC | Sets the 802.1X anonymous authentication identifier (username), defaults to null (none), can be set to "PHONE_MAC" to pass phone's MAC address | |
8021x_password | null, string | Sets the 802.1X authentication password, defaults to null (none) | |
8021x_client_cert | null, http(s) or ftp(s) URI as "url" string as "value" | Sets the URL the phone will cURL its 802.1X client certificate from, and the local name the phone should use when storing the certificate. Phone will retrieve a new certificate when factory defaulted or when value changes. Defaults to null (none) | D6x models only |
8021X_root_cert | null, http(s) or ftp(s) URI as "url" string as "value" | Sets the URL the phone will cURL its 802.1X root certificate from, and the local name the phone should use when storing the certificate. Phone will retrieve a new certificate when factory defaulted or when value changes. Defaults to null (none) | D6x models only |
8021x_debug | null, -d, -dd | Sets the debug level to be used when troubleshooting 802.1X authentication errors. Phone will generate error report that can be utilized by Digium Support. Phone should not be configured to operate in this mode on an ongoing basis as it will generate excessive messages. Defaults to null (none) |
Keymap Elements
Keymap Elements control the functionality of softkeys during various phone states.
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Option | Values | Description |
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key | id as 0-3, action as "accept_call, reject_call, transfer_call, send_to_vmail" | Maps a softkey represented by id of 0-3, left-to-right, to an action. |
Contacts Element
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Option | Values | Description |
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contacts | url as file link, id as unique identifier, md5 as the md5sum of the xml file | Specifies the contacts XML file to be retrieved by the phone and identifies that file; more than one contacts parameter may be used. Digium phones support basic authentication, so a username and password may be passed in the URL line, e.g. http://user:pass@server.example.com |
Smart BLF Element
The Smart BLF element contains the BLF Items child element. BLF Items points to an XML sheet that defines the function and positioning of BLF keys, Contacts and Applications on the phone.
BLF Items: Child Element of <smart_blf>
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Option | Values | Description |
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blf_items | url as file link, network_id (optional) as network identifier for this element, md5 (optional) as md5 sum of referenced XML sheet | Specifies the BLF Items XML file to be retrieved by the phone. Digium phones support basic authentication, so a username and password may be passed in the URL line, e.g. http://user:pass@server.example.com |
Accounts Element
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Option | Values | Description |
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server_uuid | String | Specifies a unique identifier for the server |
index | integer (0-5) | Defines the line key to which the account will be mapped |
status | boolean | Enables the line key; if false, will not display the line on the phone or register with the primary host |
register | boolean | If 1, then this account will attempt to register with the primary host |
account_id | string | SIP username |
username | string | SIP username |
authname | string | SIP authname |
password | string | SIP password |
passcode | string | SIP password |
line_label | string | The text that shows up next to the line key for this account |
caller_id | Name <Number> | Outgoing caller id displayed for this account |
dial_plan | Digit mapping, see #dialplans | The dial plan / digit mapping for this account |
visual_voicemail | boolean | Only valid on account with index of 0. Only valid for phones provisioned by Switchvox or the DPMA. If this is set to 1 then the Msgs button action will open the voicemail app. Otherwise it will dial the voicemail extension. |
voicemail | digits or SIP URI | A SIP URI or extension to be dialed for voicemail pertaining to this account. |
needMwiSubscription | boolean | Defines whether or not a phone should subscribe for MWI on this account. |
outbound_proxy | IP address / Hostname | Outbound proxy for this account |
outbound_port | port | Port for the outbound proxy |
conflict | replace |
Host Primary: Child Element of <account>
Option | Values | Description | Models |
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server | Hostname or IPv4 Address | Sets the server to which calls for this account are directed | |
port | integer (1-65535) | Sets the server's SIP port | |
transport | udp, tcp, tls | Sets the transport type, UDP or TCP, TLS | TLS for D6x only |
reregister | integer in seconds | Sets the re-registration interval | |
retry | integer | Specifies the number of time to attempt re-registration if registration fails | |
num_retries | integer | Specifies the number of retries to attempt if registration fails |
Host Alternate: Child Element of <account>
Option | Values | Description |
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server | Hostname or IPv4 Address | Sets the server to which calls for this account are directed in the event that host_primary is unreachable |
port | integer (1-65535) | Sets the server's SIP port |
transport | udp, tcp | Sets the transport type, UDP or TCP |
reregister | integer in seconds | Sets the re-registration interval |
retry | integer | Specifies the number of time to attempt re-registration if registration fails |
num_retries | integer | Specifies the number of retries to attempt if registration fails |
Permission: Child Element of <account>
Defines line/account based permissions for various phone functions with an id and value pair.
Option | Values | Description |
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record_own_calls | boolean | If 1, allows the user to record their own calls using a soft-key. Note that this feature can only enabled when using the DPMA. Users manually provisioning Digium phones should set this to 0 in order to ensure that a non-functional (because the DPMA is not being used) call recording softkey does not appear. |
Networks Element
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Option | Values | Description |
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id | integer | Unique, numbered identifier for the network |
display_name | string | A named identifier for the network |
cidr | CIDR formatted address | A CIDR formatted network address, e.g. 10.0.0.0/8 |
Codecs Element
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Option | Values | Description |
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id | PCMU, PCMA, G722, G726-32, G729, L16, L16-256 | A codec supported by the phone |
priority | integer (1-255) | Priority of the codec where higher numbers mean the codec is more favored |
packetization | integer in 10ms increments per RFC codec guidelines | Packetization (ptime) rate for the specified codec, defaults to 20 |
jitter_min | integer in ms | Sets the minimum size of the codec jitter buffer |
jitter_max | integer in ms | Sets the maximum size of the codec jitter buffer |
jitter_target | integer in ms | Sets the target size of the codec jitter buffer |
enabled | boolean | Disables / Enables a codec |
Ringtones Element
This section has two primary child elements:
- tones, which are the actual sounds heard when a call is made
- alerts, which map to a tone and represent a certain call condition
Tones: Child element of <ringtones>
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Option | Values | Description |
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id | string | Internal Tone identifier |
display | string | External Tone Description |
url | URL string | Location from which to retrieve a 16-bit, 16kHz, mono raw signed linear sound file, less than 1MB in size |
md5 | md5sum | MD5 sum of the file to be retrieved |
type | phone, user | Indicates the tone's origin; tones that are type phone are embedded into the phone's firmware, tones that are type user are retrieved by URL |
Alerts: Child element of <ringtones>
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Option | Values | Description |
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alert_info | string | The alert_info header that, as received, applies to this alert |
ringtone_id | string | The id of the ring tone for this alert |
ring_type | normal, answer, ring-answer, visual | The type of call-answer to affect for this alert. |
Firmwares Element
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Option | Values | Description |
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model | D40, D50, D70, D45, D60, D62, D65 | Model number of the Digium phone |
version | string | Version string for the firmware. On boot, the phone will check the version string against an internal copy of the string, as previously loaded. If the strings differ, the phone will load the new firmware |
url | http URL string | URL location of the phone firmware. Digium phones support basic authentication, so a username and password may be passed in the URL line, e.g. http://user:pass@server.example.com |
Public Firmwares Element
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Option | Values | Description |
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model | D40, D50, D70, D45, D60, D62, D65 | Model number of the Digium phone |
version | string | Version string for the firmware. On boot, the phone will check the version string against an internal copy of the string, as previously loaded. If the strings differ, the phone will load the new firmware |
url | http URL string | URL location of the phone firmware. Digium phones support basic authentication, so a username and password may be passed in the URL line, e.g. http://user:pass@server.example.com |
Appconfig Element
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- Element lists the <appconfig> elements
Display Rules: Child Element of <appconfig> for contacts application
Option | Values | Description |
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id | unique id | A unique identifier for this display_rule, e.g. 0, 1, 2, etc. |
action_id | A valid action as defined in the phone's Contacts file | Sets the action_id for the display rule to act upon |
phone_state | idle, hold, transfer, incoming/transfer, incoming, connected, dialing, calling, failed | Defines the state of the local phone during which the rule will be acting. If not specified in a rule, all states are matched. Note that this list of states differs from the larger list of states available for BLF key action mappings. |
target_status | unknown, idle, on_hold, ringing, on_the_phone | Optional. Sets the status of the subscribed to contact that must be matched for this display rule to be in effect |
show | Boolean | If set to yes, shows a particular action; if set to no, hides the action. To hide an action for all states and only show it for some states, first declare the action to have a false show, then declare it to have a true show for only a particular state or states. |
The can_transfer_vm attribute controls the display of the "Transfer VM" softkey from within the Contacts application.
The name_format attribute controls the display order of names from within the Contacts application.
Multicastpage Element
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Option | Values | Description |
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id | string | A unique identifier for the listener, required |
name | string | A name to be provided in the phone's announcement status bar when audio is received over this listener, required |
addr | IPv4 address | Multicast address to which phone should subscribe for audio, required |
port | Valid integer for ports (1-65535) | Port, combined with address above, to which phone should subscribe for audio, required |
priority | integer, 1-10 | Prioritization level, lower given more priority, for playing back streams when more than one subscribed address is providing audio, required |
interrupt_callers | boolean | When enabled, places any in-progress calls on hold before playing back audio. When disabled, in-progress calls will have their audio played over. Defaults to 0. |
network_id attribute
The network_id attribute, in conjunction with the Networks element is used to provision multiple different options for a particular element, e.g. account address, so that when the Digium phone is located on different networks, the proper element for that network can be loaded by the phone. When the phone boots and discovers its IP address, it compares that against matching elements with network_id attributes and loads only those elements, rather than elements with non-matching network_ids.
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