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Difference between revisions of "Configuration File Parameters for SIP Provisioning"

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{{S}}This article describes the parameters in the configuration file used for TFTP Provisioning. See [[Turbine_Configuration_-_SIP_mode#TFTP_Provisioning|TFTP Provisioning]] for how to configure the station.
+
{{S}}This article describes the parameters in the configuration file used for TFTP Provisioning. See [[TFTP Provisioning (SIP)|TFTP Provisioning]] for how to configure the station for this function.
  
TFTP Provisioning is supported by all IP Stations ([[:Category:Turbine|Turbine]] and [[:Category:INCA Stations|INCA stations]]), operating in '''SIP mode'''.
+
TFTP Provisioning is supported by all [[:Category:Stations|IP Stations]] when operating in '''SIP mode'''.
  
  
 
+
==Remote Provisioning using TFTP==
== Remote Provisioning using TFTP ==
 
  
 
An IP station may be set up to automatically poll configuration from a TFTP server. The IP address of this TFTP server can be obtained using DHCP procedures or be manually configured.
 
An IP station may be set up to automatically poll configuration from a TFTP server. The IP address of this TFTP server can be obtained using DHCP procedures or be manually configured.
  
 
The IP station will first try to download the global configuration file:
 
The IP station will first try to download the global configuration file:
ipst_config.cfg
+
{{code|ipst_config.cfg}}
 
Then the IP station will download a device specific configuration file:
 
Then the IP station will download a device specific configuration file:
ipst_config_01_02_03_04_05_06.cfg
+
{{code|ipst_config_01_02_03_04_05_06.cfg}} where '''01_02_03_04_05_06''' is the MAC address of the IP station. Or:
where '''01_02_03_04_05_06''' is the MAC address of the IP station.
+
{{code|ipst_config_10_9_8_7.cfg}}
 +
where '''10_9_8_7''' is the IP address of the IP station.  
  
 
If the same parameter is found in several files, then the precedence is as following:
 
If the same parameter is found in several files, then the precedence is as following:
 +
 
#.MAC address file
 
#.MAC address file
 
#IP address file
 
#IP address file
 
#Global file
 
#Global file
  
== General Parameters ==
+
{{note|
 +
* ''The last line in the configuration file must be terminated with <'''Enter'''> (CR-LF), else the configuration will not be applied.''
 +
* ''If there are changes in the configuration file that require a reboot to take effect, the station will do a reboot when the cofiguration file is downloaded and changes are applied''
 +
* ''If there are no changes in the downloaded configuration file compared to the previous downloaded file, the station will ignore the new file'' }}
 +
 
 +
 
 +
==Detailed description of general parameters==
 +
===auto_update_interval===
 +
 
 +
*Required: No. If this parameter's not set in the file, the function will be disabled
 +
*Description: This parameter enables the station to automatically look for software updates on the TFTP server
 +
*Values: Number of minutes to wait between each server request. Value must be between 1 and 999
 +
 
 +
===auto_update_image_type===
 +
 
 +
*Required: If auto_update_interval is set
 +
*Description: The name of the software image file to be uploaded
 +
*Values: Text giving the name of the software image file. The full name of the file, including extension, is required. This parameter must be set if the auto update function is enabled
 +
 
 +
===auto_update_image_crc===
 +
 
 +
*Required: If auto_update_interval is set
 +
*Description: The CRC checksum calculated for the software image file specified by the auto_update_image_type parameter. This is used to check the integrity of the software file before updating the station
 +
*Values: Hexadecimal value
 +
 
 +
===use_last_known_ip (INCA stations only)===
 +
 
 +
*Required: No. Default is 0.
 +
*Description: Use Last IP On DHCP failure.
 +
*Values: Integer Value. 0 = disabled, 1 = enabled
 +
 
 +
===turbine_frontboard (deprecated from VS-IS 7.1)===
 +
 
 +
*Description: Configures the turbine frontboard type
 +
*Values:
 +
**0 = KIT
 +
**1 = TCIS-1, TCIS-2, TCIS-3
 +
**2 = TCIS-6
 +
**3 = TCIS4, TCIS-5
 +
**4 = TFIE-1, TFIX-1
 +
**5 = TFIE-2, TFIX-2
 +
**6 = TFIX-3
 +
**7 = ECPIR-P
 +
**8 = EAPII-1, EAPFX-1
 +
**9 = EAPII-6, EAPFX-6
 +
**10 = ECPIR-3P
 +
**11 = EAPIR-8
 +
**13 = TFIE-6
 +
**45 = IP-LCM
 +
**51 = TCIV-2, TCIV-3
 +
**52 = TCIV-6
 +
**53 = TMIS1, TMIS-2
 +
**54 = TCIV-5
 +
 
 +
==Detailed description of Relay parameter==
 +
 
 +
*The following relay keys are supported:
 +
**relay1 and relay2 for station physical relays
 +
**gpio1 to gpio6 for I/O pins configured as outputs
 +
**e_relay1 and e_relay2 for TA-10 relay module
 +
*To configure specific relay rename the parameter to contain the proper relay key, i.e. instead of "relay1_dtmf_activate" use "gpio1_dtmf_activate".
 +
 
 +
===relay1_dtmf_activate===
 +
 
 +
*Description: Dtmf value to send for activating the relay
 +
*Values: Valid values is 0-9, * and #. The character - means off.
 +
 
 +
===relay1_dtmf_deactivate===
 +
 
 +
*Description: Dtmf value to send for deactivating the relay
 +
*Values: Valid values is 0-9, * and #. The character - means off.
 +
 
 +
===relay1_dtmf_flashing_slow===
 +
 
 +
*Description: Dtmf value to send for setting the relay to flashing slow
 +
*Values: Valid values is 0-9, * and #. The character - means off.
 +
 
 +
===relay1_dtmf_flashing_fast===
 +
 
 +
*Description: Dtmf value to send for setting the relay to flashing fast
 +
*Values: Valid values is 0-9, * and #. The character - means off.
 +
 
 +
===relay1_dtmf_toggle===
 +
 
 +
*Description: Dtmf value to send for toggling the relay
 +
*Values: Valid values is 0-9, * and #. The character - means off.
 +
 
 +
===relay1_dtmf_timed_relay===
 +
 
 +
*Description: Dtmf value to send for activating the relay for X seconds
 +
*Values: Valid values is 0-9, * and #. The character - means off.
 +
 
 +
===relay1_dtmf_timed_relay_duration===
 +
 
 +
*Description: Duration to activate relay
 +
*Values: Integer. 0 means activate relay forever.
 +
 
 +
===relay1_event_out_ringing===
 +
 
 +
*Description: When the station is ringing in an outgoing call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
===relay1_event_inc_ringing===
 +
 
 +
*Description: When the station is ringing in an incoming call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
===relay1_event_inc_call===
 +
 
 +
*Description: When the station is in an incoming call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
===relay1_event_out_call===
 +
 
 +
*Description: When the station is in an outgoing call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
===relay1_event_group_call===
 +
 
 +
*Description: When the station is in a group call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
===relay1_event_idle===
 +
 
 +
*Description: When the station is idle, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
===relay1_event_error===
 +
 
 +
*Description: When the station is error, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
 +
*Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
 +
 
 +
==Detailed description of Tone parameter==
 +
===enabled===
 +
 
 +
*Description: Enables tone test
 +
*Values: Integer Value. 0 = disabled, 1 = enabled
 +
 
 +
===time_between_tonetest===
 +
 
 +
*Description: Time between tone tests
 +
*Values: Integer Value.
 +
 
 +
===sound_pressure_level (obsolete for SW > 6.1.3.0)===
 +
 
 +
*Description:  Minimum sound pressure level between silence and tone
 +
*Values: Integer Value. Only odd values. 53 ≤ sound_pressure_level ≤ 97
 +
 
 +
===volume (obsolete for SW > 6.1.3.0)===
 +
 
 +
*Description: Volume of the tone test
 +
*Values: Integer Value. 0 ≤ volume ≤ 7
 +
 
 +
===auto_set_sound_pressure_level (obsolete for non-INCA stations)===
 +
 
 +
*Description: Only available on INCA stations, station tries to calculate the parameter 'Minimum sound pressure level' (sound_pressure_level).
 +
*Values: Integer Value. 0 = disabled, 1 = enabled
 +
 
 +
===tone_selection (SW > 6.1.3.0)===
 +
 
 +
*Description: Defines frequency groups for the tone test.
 +
*Values: Integer value. 1 <= tone_selection <= 8
 +
 
 +
==Detailed description of SIP parameters==
 +
===nick_name===
 +
 
 +
*Required: No. Defaults to sip_id
 +
*Description: The nick name for the station can be used to assign a logical name to the station. E.g. a station belonging to James may be assigned the nick name "James", or "James' station"
 +
*Values: Text string. Max length is 64 characters.
 +
 
 +
===sip_id===
  
=== auto_update_interval ===
+
*Required: Yes
* '''Required:''' No. If this parameter's not set in the file, the function will be disabled
+
*Description: This is the identification of the station in the SIP domain, i.e. the phone number for the station
 +
*Values: Integer value. Max length is 64 characters.
  
* '''Description:''' This parameter enables the station to automatically look for software updates on the TFTP server
+
===sip_domain===
  
* '''Values:''' Number of minutes to wait between each server request. Value must be between 1 and 999
+
*Required: Yes
 +
*Description: SIP domain is a server that uses SIP (Session Initiation Protocol) to manage real-time communication among SIP clients. The sip_domain parameter specifies the primary domain for the station, as opposed to sip_domain2 which specifies the secondary (or fall back) domain. The IP address for the SIP domain server (e.g. Asterisk or Cisco Call Manager) should be defined in this section
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100
  
=== auto_update_image_type ===
+
===sip_domain2===
* '''Required:''' If ''auto_update_interval'' is set
 
  
* '''Description:''' The name of the software image file to be uploaded
+
*Required: No
 +
*Description: This is the secondary (or fall-back) domain. If the station loses connection to the primary SIP domain, it will switch over to the secondary.
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100
  
* '''Values:''' Text giving the name of the software image file. The full name of the file, including extension, is required. This parameter must be set if the auto update function is enabled
+
===sip_domain3===
  
=== auto_update_image_crc ===
+
*Required: No
* '''Required:''' If ''auto_update_interval'' is set
+
*Description: This is the tertiary (or fall-back) domain. If the station loses connection to the primary and secondary SIP domain, it will switch over to the tertiary.
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100
  
* '''Description:''' The CRC checksum calculated for the software image file specified by the ''auto_update_image_type'' parameter. This is used to check the integrity of the software file before updating the station
+
===registration_method===
  
* '''Values:''' Hexadecimal value
+
*Required: No
 +
*Description: Registration method which the station will use if fall-back domains are available.
 +
*Values: parallel = Parallel registration to fall-back domains, serial = Serial registration to fall-back domains
  
=== turbine_frontboard ===
+
===auth_user===
* '''Description:''' Configures the turbine frontboard type
 
  
* '''Values:''' 
+
*Required: Only if the SIP server requires authentication
* 0 = KIT
+
*Description: The authentication user name used to register the station to the SIP server.
* 1 = TKIE-1, TCIS-1, TCIS-2, TCIS-3
+
*Values: Text string.
* 2 = TCIS-6
 
* 3 = TCIS4, TCIS-5
 
* 4 = TFIE-1, TFIX-1
 
* 5 = TFIE-2, TFIX-2
 
* 6 = TFIX-3
 
* 7 = ECPIR-P
 
* 8 = EAPII-1, EAPFX-1
 
* 9 = EAPII-6, EAPFX-6
 
* 10 = ECPIR-3P
 
* 11 = EAPIR-8
 
* 51 = TCIV-2, TCIV-3
 
* 52 = TCIV-6
 
* 53 = MINI
 
  
== SIP Parameters ==
+
===auth_pwd===
  
=== nick_name ===
+
*Required: Only if the SIP server requires authentication
* '''Required:''' No. Defaults to ''sip_id''
+
*Description: The authentication user password used to register the station to the SIP server.
 +
*Values: Text string.
  
* '''Description:''' The nick name for the station can be used to assign a logical name to the station. E.g. a station belonging to James may be assigned the nick name "James", or "James' station"
+
===sip_outbound_proxy===
  
* '''Values:''' Text string. Max length is 64 characters.
+
*Required: Optional
 +
*Description: Configures an outbound-proxy server that receives all initiating request (INVITE and SUBSCRIBE) messages.
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100
  
=== sip_id ===
+
===sip_outbound_proxy_port===
* '''Required:''' Yes
 
  
* '''Description:''' This is the identification of the station in the SIP domain, i.e. the phone number for the station
+
*Required: If proxy server is defined. Default 5060.
 +
*Description: The UDP port used for SIP on the proxy server.
 +
*Values: Integer.
  
* '''Values:''' Integer value. Max length is 64 characters.
+
===register_interval===
  
=== sip_domain ===
+
*Required: No. Defaults to 600 seconds
* '''Required:''' Yes
+
*Description: This parameter specifies how often the station will register, and reregister, in the SIP domain. This parameter will affect the time it takes to discover that a connection to a SIP domain is lost
 +
*Values: Number of seconds. 60 ≤ register_interval ≤ 999999
  
* '''Description:''' SIP domain is a server that uses SIP (Session Initiation Protocol) to manage real-time communication among SIP clients. The ''sip_domain'' parameter specifies the primary domain for the station, as opposed to ''sip_domain2'' which specifies the secondary (or fall back) domain. The IP address for the SIP domain server (e.g. Asterisk or Cisco Call Manager) should be defined in this section
+
===fail_interval (since 4.9.3.2)===
  
* '''Values:''' IP address given in regular dot notation, e.g. 10.5.2.100
+
*Required: No. Defaults to 60 seconds
 +
*Description: In case Primary and both Backup servers are failing with SIP INVITEs, the device should go into failure mode, and immediately start sending REGISTER requests to all SIP servers, in time periods using this failure interval.
 +
*Values: Number of seconds. 5 ≤ fail_interval ≤ 999999
  
=== sip_domain2 ===
+
===playback_gain (RS amplifier only)===
* '''Required:''' No
 
  
* '''Description:''' This is the secondary (or fall-back) domain. If the station loses connection to the primary SIP domain, it will switch over to the secondary.
+
*Required: No.
 +
*Description: This parameter specifies the gain on a output channel
 +
*Values: dB.  -40 ≤ playback_gain ≤ 0
  
* '''Values:''' IP address given in regular dot notation, e.g. 10.5.2.100
+
===recorder_gain (RS amplifier only)===
  
=== auth_user ===
+
*Required: No.
* '''Required:''' Only if the SIP server requires authentication
+
*Description: This parameter specifies the gain on a input channel
 +
*Values: dB.  0 ≤ recorder_gain ≤ 40
  
* '''Description:''' The authentication user name used to register the station to the SIP server.
+
==Detailed description of Call parameters==
  
* '''Values:''' Text string.
+
*DAK, input and special keys configuration have the same parameters available, the exact name of the parameter will depend on the desired key ID and the name must start with that key ID.
  
=== auth_pwd ===
+
*Depending on the station type, the following key IDs will be available:
* '''Required:''' Only if the SIP server requires authentication
+
**input1 to input6
 +
**dak1 to dak10
 +
**offhook
 +
**onhook
 +
**ptt
 +
*For example, parameters name would then look like this:
 +
**input2_function
 +
**dak1_function
 +
**offhook_function
 +
**onhook_function
 +
**dak_function
 +
*This document will use "dak1" as an example for configuration parameters.
  
* '''Description:''' The authentication user password used to register the station to the SIP server.
+
===dak1_value===
  
* '''Values:''' Text string.
+
*Required: Yes for input1_function values 0 (default, call to), 3 (forward call) and4 (group call).
 +
*Description: This is the SIP ID for the extension to be called when the first DAK button is pressed, i.e. the telephone number of the receiving party. It's also used if input1_function is set to 3 (forward call to SIP ID) or 4 (group call to SIP ID).
 +
*Values: String value
  
=== sip_outbound_proxy ===
+
===dak1_option===
* '''Required:''' Optional
 
  
* '''Description:''' Configures an outbound-proxy server that receives all initiating request (INVITE and SUBSCRIBE) messages.
+
*Required: No
 +
*Description: Used in combination with call to function, configures if ringlist will be used or not.
 +
*Values: Integer. -1 is no ringlist. 0 is ringlist 1. 1 is ringlist 2. 2 is ringlist 3.
  
* '''Values:''' IP address given in regular dot notation, e.g. 10.5.2.100
+
===dak1_text (DualDisplay stations only)===
  
=== sip_outbound_proxy_port ===
+
*Required: No
* '''Required:''' If proxy server is defined. Default 5060.
+
*Description: Text which will be visible on the display next to DAK button 1. Note that this option will do nothing on inputs.
 +
*Values: String value
  
* '''Description:''' The UDP port used for SIP on the proxy server.
+
===dak1_function===
  
* '''Values:''' Integer.
+
*Required: No. Default value is 0 (call to)
 +
*Description: This decides what input button 1 will do when outside calls.
 +
*Values: Integer. 0 is call to.1 is do nothing. 3 is forward call (Turbine only). 4 is group call (Turbine only). 5 is conversation mode (Turbine only). 6 is volume control (Turbine only).
  
=== register_interval ===
+
===dak1_in_call_function===
* '''Required:''' No. Defaults to 600 seconds
 
  
* '''Description:''' This parameter specifies how often the station will register, and reregister, in the SIP domain. This parameter will affect the time it takes to discover that a connection to a SIP domain is lost
+
*Required: No. Default value is 1 (do nothing)
 +
*Description: This decides what input button 1 will do when in calls.
 +
*Values:  Integer. 0 is answer/end call.1 is do nothing. 2 is send DTMF. 3 is send text. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer
  
* '''Values:''' Number of seconds. 60 ≤ ''register_interval'' ≤ 999999
+
===dak1_dtmf_on===
  
=== playback_gain (RS amplifier only) ===
+
*Required: No. Only used with in call function "Send DTMF".
* '''Required:''' No. 
+
*Description: Which DTMF value will be sent when DAK button 1 is pressed.
 +
*Values: String. Supported values are from 0 to 9, and *, #, ! (DTMF Flash), O (Do Nothing).
  
* '''Description:''' This parameter specifies the gain on a output channel
+
===dak1_dtmf_off===
  
* '''Values:''' dB.  -40 ≤ ''playback_gain'' ≤ 0
+
*Required: No. Only used with in call function "Send DTMF".
 +
*Description: Which DTMF value will be sent when DAK button 1 is released.
 +
*Values: String. Supported values are from 0 to 9, and *, #, ! (DTMF Flash), O (Do Nothing).
  
=== recorder_gain (RS amplifier only) ===
+
===dak1_text_on===
* '''Required:''' No. 
 
  
* '''Description:''' This parameter specifies the gain on a input channel
+
*Required: No. Only used with in call function "Send Text".
 +
*Description: Which text value will be sent when DAK button 1 is pressed.
 +
*Values:  String.
  
* '''Values:''' dB.  0 ≤ ''recorder_gain'' ≤ 40
+
===dak1_text_off===
  
== Call Parameters ==
+
*Required: No. Only used with in call function "Send Text".
 +
*Description: Which text value will be sent when DAK button 1 is released.
 +
*Values:  String.
  
=== input1_value ===
+
===ringlist1_value1===
* '''Required:''' Yes
 
  
* '''Description:''' This is the SIP ID for the extension to be called when the first input button is pressed, i.e. the telephone number of the receiving party.
+
*Required: No
 +
*Description: This is the SIP ID for the extension to be called when ringlist 1 is used and it is at first entry. The next numbers in the ringlist is then ringlist1_value2, ringlist1_value3 etc.
 +
*Values: String value
  
* '''Values:''' String value
+
===ringlist1_wp1===
  
=== input1_in_call_function ===
+
*Required: No, default value is 0 (disabled)
* '''Description:''' This decides what input button 1 will do when in calls.
+
*Description: Call with previous parameter for the first entry in the ringlist 1. The next numbers in the ringlist is then ringlist1_wp2, ringlist1_wp3 etc.
 +
*Values: Integer. 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 is end call. 1 is do nothing.
+
===ringlist2_value1===
  
=== input2_value ===
+
*Required: No
* '''Required:''' Yes
+
*Description: This is the SIP ID for the extension to be called when ringlist 2 is used and it is at the first entry. The next numbers in the ringlist is then ringlist2_value2, ringlist2_value3 etc.
 +
*Values: String value
  
* '''Description:''' This is the SIP ID for the extension to be called when the second input button is pressed, i.e. the telephone number of the receiving party.
+
===ringlist2_wp1===
  
* '''Values:''' String value
+
*Required: No, default value is 0 (disabled)
 +
*Description: Call with previous parameter for the first entry in the ringlist 2. The next numbers in the ringlist is then ringlist2_wp2, ringlist2_wp3 etc.
 +
*Values: Integer. 0 = disabled, 1 = enabled
  
=== input2_in_call_function ===
+
===ringlist3_value1===
* '''Description:''' This decides what input button 1 will do when in calls.
 
  
* '''Values:''' Integer. 0 is end call. 1 is do nothing.
+
*Required: No
 +
*Description: This is the SIP ID for the extension to be called when ringlist 3 is used and it is at the first entry. The next numbers in the ringlist is then ringlist3_value2, ringlist3_value3 etc.
 +
*Values: String value
  
=== input3_value ===
+
===ringlist3_wp1===
* '''Required:''' Yes
 
  
* '''Description:''' This is the SIP ID for the extension to be called when the third input button is pressed, i.e. the telephone number of the receiving party.
+
*Required: No, default value is 0 (disabled)
 +
*Description: Call with previous parameter for the first entry in the ringlist 3. The next numbers in the ringlist is then ringlist3_wp2, ringlist3_wp3 etc.
 +
*Values: Integer. 0 = disabled, 1 = enabled
  
* '''Values:''' String value
+
===ringlist_max_ring_time===
  
=== input3_in_call_function ===
+
*Required: No. Defaults to 4
* '''Description:''' This decides what input button 1 will do when in calls.
+
*Description: This parameter sets the time to wait ringing until step to next value in the ringlist
 +
*Values: Number of seconds. 0 ≤ ringlist_max_ring_time ≤ 999999
  
* '''Values:''' Integer. 0 is end call. 1 is do nothing.
+
===ringlist_loop===
  
=== dak1_value ===
+
*Required: No. Defaults to 0
* '''Required:''' No
+
*Description: This parameter enable the loop so list is repeated until answer.
 +
*Values: Integer. 0 to disable, 1 to enable
  
* '''Description:''' This is the SIP ID for the extension to be called when the first dak button is pressed, i.e. the telephone number of the receiving party.
+
===ringlist_max_conv_time===
  
* '''Values:''' String value
+
*Required: No. Defaults to 4
 +
*Description: This parameter sets the max time of the conversation when followed ringlist
 +
*Values: Number of seconds. 0 ≤ ringlist_max_conv_time ≤ 999999
  
=== dak2_value ===
+
===conv_time===
* '''Required:''' No
 
  
* '''Description:''' This is the SIP ID for the extension to be called when the second dak button is pressed, i.e. the telephone number of the receiving party.
+
*Required: No. Defaults to 3600
 +
*Description: This parameter sets the max time of the conversation
 +
*Values: Number of seconds. 0 ≤ conv_time ≤ 999999
  
* '''Values:''' String value
+
===ring_time===
  
=== dak2_value ===
+
*Required: No. Defaults to 120
* '''Required:''' No
+
*Description: This parameter sets the max time of the ringing/alerting phase of the call
 +
*Values: Number of seconds. 0 ≤ ring_time ≤ 999999
  
* '''Description:''' This is the SIP ID for the extension to be called when the third dak button is pressed, i.e. the telephone number of the receiving party.
+
===poe_audio (obsolete)===
  
* '''Values:''' String value
+
*Description: This parameter sets whether to disable maximum speaker output
 +
*Values: Set to 0 to enable full audio output. Set to 1 to disable full audio output.
  
=== ringlist1_value1 ===
+
===speaker_volume===
* '''Required:''' No
 
  
* '''Description:''' This is the SIP ID for the extension to be called when ringlist 1 is used and it is at first entry. The next numbers in the ringlist is then ringlist1_value2, ringlist1_value3 etc.
+
*Required: No. Defaults to 4
 +
*Description: This parameter sets the volume of the station's speaker
 +
*Values: Integer. 0 ≤ speaker_volume ≤ 7
  
* '''Values:''' String value
+
===mic_sensitivity===
  
=== ringlist2_value1 ===
+
*Required: No. Defaults to 5
* '''Required:''' No
+
*Description: This parameter adjusts the microphone sensitivity
 +
*Values: Integer. 0 ≤ mic_sensitivity ≤ 7
  
* '''Description:''' This is the SIP ID for the extension to be called when ringlist 2 is used and it is at the first entry. The next numbers in the ringlist is then ringlist2_value2, ringlist2_value3 etc.
+
===noise_reduction===
  
* '''Values:''' String value
+
*Required: No. Defaults to 0
 +
*Description: This parameter adjusts the noise reduction level
 +
*Values: Integer. 0 ≤ noise_reduction ≤ 7
  
=== ringlist3_value1 ===
+
===echo_parameter (INCA stations only)===
* '''Required:''' No
 
  
* '''Description:''' This is the SIP ID for the extension to be called when ringlist 3 is used and it is at the first entry. The next numbers in the ringlist is then ringlist3_value2, ringlist3_value3 etc.
+
*Required: No. Defaults to 0
 +
*Description: This parameter adjusts the echo parameter
 +
*Values: Integer. 0 ≤ echo_parameter ≤ 7
  
* '''Values:''' String value
+
===rtp_timeout===
  
=== ringlist_max_ring_time ===
+
*Required: No. Defaults to 0
* '''Required:''' No. Defaults to 4
+
*Description: Cancels a call if the station does not receive rtp.
 +
*Values: Integer value: 0-9999 seconds. 0 = RTP timeout disabled.
  
* '''Description:''' This parameter sets the time to wait ringing until step to next value in the ringlist
+
===remote_controlled_volume_override_mode===
  
* '''Values:''' Number of seconds. 0 ≤ ''ringlist_max_ring_time'' ≤ 999999
+
*Required: No.
 +
*Description: Acts as a simplex mode after first DTMF * or # is received. At remote station: send DTMF * to talk and # to listen.
 +
*Values: Integer. 0 disabled, 1 enabled.
  
=== ringlist_loop ===
+
===speech_mode===
* '''Required:''' No. Defaults to 0
 
  
* '''Description:''' This parameter enable the loop so list is repeated until answer.
+
*Required: No.
 +
*Description: Set the conversation mode.
 +
*Values: Integer. 0 = Full Open Duplex, 1 = Push To Talk, 2 = Half Duplex Switching, 3 = Open, 4 = Robust Duplex.
  
* '''Values:''' Integer. 0 to disable, 1 to enable
+
===ptt_mode===
  
=== ringlist_max_conv_time ===
+
*Required: No.
* '''Required:''' No. Defaults to 4
+
*Description: Set the PTT mode, only active if Conversation Mode is set to PTT.
 +
*Values: ptt_mic_and_speaker = Mic and speaker is controlled by PTT button, ptt_mic_only = Mic is controlled by PTT button.
  
* '''Description:''' This parameter sets the max time of the conversation when followed ringlist
+
===auto_answer_mode===
  
* '''Values:''' Number of seconds. 0 ≤ ''ringlist_max_conv_time'' ≤ 999999
+
*Required: No.
 +
*Description: Enables autoanswer after a set number of seconds.
 +
*Values: Integer. 0 disabled, 1 enabled.
  
=== conv_time ===
+
===auto_answer_delay===
* '''Description:''' This parameter sets the max time of the conversation
 
  
=== ring_time ===
+
*Required: No. Defaults to 0.
* '''Description:''' This parameter sets the max time of the ringing/alerting phase of the call
+
*Description: The number of seconds to delay the autoanswer
 +
*Values: Integer. 0 ≤ delay ≤ 30
  
=== poe_audio (full audio output) ===
+
===accessory===
* '''Description:''' This parameter sets whether to disable maximum speaker output
 
* '''Values''': Set to 0 to enable full audio output. Set to 1 to disable full audio output.
 
  
=== speaker_volume ===
+
*Required: No. Defaults to 0.
* '''Required:''' No. Defaults to 4
+
*Description: Which accessory to use
 +
*Values: 0 = unused/default, 1 = handset, 2 = microphone w/ptt, 3 = headset, 4 = handset w/offhook, 5 = headset auto detect, 6 = handset w/offhook normally closed
  
* '''Description:''' This parameter sets the volume of the station's speaker
+
===input_as_key_matrix===
  
* '''Values:''' Integer. 0 ≤ ''speaker_volume'' ≤ 7
+
*Required: No. Defaults to 0.
 +
*Description: Use inputs as a key matrix. Requires that gpio is configured as gpi/input
 +
*Values: 0 = no inputs as key, 1 = means 1 input as as key matrix (1 dak), 2 = means 2 inputs as as key matrix (3 daks), 3 = means 3 inputs as as key matrix (7 daks)
  
=== mic_sensitivity ===
+
===fast_blink_pattern===
* '''Required:''' No. Defaults to 5
 
  
* '''Description:''' This parameter adjusts the microphone sensitivity
+
*Required: No. Defaults to 111000111000111000111000
 +
*Description: Customize fast blink pattern
 +
*Values: 1 = gpo high, 0 = gpo low
  
* '''Values:''' Integer. 0 ≤ ''mic_sensitivity'' ≤ 7
+
===slow_blink_pattern===
  
=== noise_reduction ===
+
*Required: No. Defaults to 111111111111000000000000
* '''Required:''' No. Defaults to 0
+
*Description: Customize slow blink pattern
 +
*Values: 1 = gpo high, 0 = gpo low
  
* '''Description:''' This parameter adjusts the noise reduction level
+
===open_duplex_dtmf===
  
* '''Values:''' Integer. 0 ≤ ''noise_reduction'' ≤ 7
+
*Required: No. Defaults to -
 +
*Description: Forces the station in Open Duplex when configured DTMF is received
 +
*Values: - = off, valid range: 0-9
  
=== echo_parameter ===
+
===override_remote_ptt===
* '''Required:''' No. Defaults to 0
 
  
* '''Description:''' This parameter adjusts the echo parameter
+
*Required: No. Defaults to 0
 +
*Description: If 2 stations call each other and Override Remote PTT is enabled, then conversation mode is switched to open duplex.
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 ≤ ''echo_parameter'' ≤ 7
+
===dtmf_style===
  
=== rtp_timeout ===
+
*Required: No. Defaults to 0
* '''Required:''' No. Defaults to 0
+
*Description: Choose how to send DTMF
 +
*Values: 0 = SIP INFO, 1 = RFC2833
  
* '''Description:''' Cancels a call if the station does not receive rtp.
+
===tone_volume===
  
* '''Values:''' Integer value: 0-9999 seconds. 0 = RTP timeout disabled.
+
*Required: No. Defaults to 0
 +
*Description: Control tone volume
 +
*Values: -1 = no tones, 0 default volume, 1-4 increases volume
  
=== remote_controlled_volume_override_mode ===
+
===codec_g729_pri===
* '''Required:''' No.
 
  
* '''Description:''' Acts as a simplex mode after first DTMF * or # is received. At remote station: send DTMF * to talk and # to listen.
+
*Required: No
 +
*Description: Set the priority of g729 codec
 +
*Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority
  
* '''Values:''' Integer. 0 disabled, 1 enabled.
+
===codec_g722_pri===
  
=== auto_answer_mode ===
+
*Required: No
* '''Required:''' No.
+
*Description: Set the priority of g722 codec
 +
*Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority
  
* '''Description:''' Enables autoanswer after a set number of seconds.
+
===codec_g711u_pri===
  
* '''Values:''' Integer. 0 disabled, 1 enabled.
+
*Required: No
 +
*Description: Set the priority of g711u codec
 +
*Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority
  
=== auto_answer_delay ===
+
===codec_g711a_pri===
* '''Required:''' No. Defaults to 0.
 
  
* '''Description:''' The number of seconds to delay the autoanswer
+
*Required: No
 +
*Description: Set the priority of g711a codec
 +
*Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority
  
* '''Values:''' Integer. 0 ≤ ''delay'' ≤ 30
+
===io_pin1===
  
=== accessory ===
+
*Required: No
* '''Required:''' No. Defaults to 0.
+
*Description: Will the I/O pin 1 behave as an input or as an output.
 +
*Values: 0 = Input, 1 = Output
  
* '''Description:''' Which accessory to use
+
===io_pin2===
  
* '''Values:''' 0 = unused/default, 1 = handset, 2 = microphone w/ptt, 3 = headset, 4 = handset w/offhook, 5 = headset auto detect, 6 = handset w/offhook normally closed
+
*Required: No
 +
*Description: Will the I/O pin 2 behave as an input or as an output.
 +
*Values: 0 = Input, 1 = Output
  
=== input_as_key_matrix ===
+
===io_pin3===
* '''Required:''' No. Defaults to 0.
 
  
* '''Description:''' Use inputs as a key matrix. Requires that gpio is configured as gpi/input
+
*Required: No
 +
*Description: Will the I/O pin 3 behave as an input or as an output.
 +
*Values: 0 = Input, 1 = Output
  
* '''Values:''' 0 = no inputs as key, 1 = means 1 input as as key matrix (1 dak), 2 = means 2 inputs as as key matrix (3 daks), 3 = means 3 inputs as as key matrix (7 daks)
+
===io_pin4===
  
=== fast_blink_pattern ===
+
*Required: No
* '''Required:''' No. Defaults to 111000111000111000111000
+
*Description: Will the I/O pin 4 behave as an input or as an output.
 +
*Values: 0 = Input, 1 = Output
  
* '''Description:''' Customize fast blink pattern
+
===io_pin5===
  
* '''Values: 1''' = gpo high, 0 = gpo low
+
*Required: No
 +
*Description: Will the I/O pin 5 behave as an input or as an output.
 +
*Values: 0 = Input, 1 = Output
  
=== slow_blink_pattern ===
+
===io_pin6===
* '''Required:''' No. Defaults to 111111111111000000000000
 
  
* '''Description:''' Customize slow blink pattern
+
*Required: No
 +
*Description: Will the I/O pin 6 behave as an input or as an output.
 +
*Values: 0 = Input, 1 = Output
  
* '''Values:''' 1 = gpo high, 0 = gpo low
+
==Detailed description of SNMP parameters==
 +
===trap_receiver===
  
=== open_duplex_dtmf ===
+
*Required: No.
* '''Required:''' No. Defaults to - 
+
*Description: The IP address of the server receiving SNMP traps.
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100
  
* '''Description:''' Forces the station in Open Duplex when configured DTMF is received
+
===inform_receiver===
  
* '''Values:''' - = off, valid range: 0-9
+
*Required: No.
 +
*Description: The IP address of the server receiving SNMP informs.
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100
  
=== override_remote_ptt ===
+
===network===
* '''Required:''' No. Defaults to 0 
 
  
* '''Description:''' If 2 stations call each other and Override Remote PTT is enabled, then conversation mode is switched to open duplex.
+
*Required: No.
 +
*Description: Used, together with the network mask, to determine the allowed network for reading the MIB on the IP station.
 +
*Values: IP address given in regular dot notation, e.g. 10.5.2.100. For example with an allowed network 10.5.2.0 and a network mask of 24, anyone with IP address 10.5.2.0 to 10.5.2.255 can access the MIB.
  
* '''Values:''' 0 = disabled, 1 = enabled
+
===network_mask===
  
=== poe_audio ===
+
*Required: No.
* '''Required:''' No. Defaults to
+
*Description: The mask used to determine the allowed network for reading the MIB.
 +
*Values: Integer. 0 ≤ network_mask ≤ 32. For example with an allowed network 10.5.2.0 and a network mask of 24, anyone with IP address 10.5.2.0 to 10.5.2.255 can access the MIB.
  
* '''Description:''' Enables full audio output. In case of PoE switch then the station might reboot if too much power is used.
+
===community===
  
* '''Values:''' 0 = disabled, 1 = enabled
+
*Required: No.
 +
*Description: An text staring used as password for authentication.
 +
*Values: String.
  
=== dtmf_style ===
+
===enable_v1===
* '''Required:''' No. Defaults to 0 
 
  
* '''Description:''' Choose how to send DTMF
+
*Required: No.
 +
*Description: Enables reading of MIB using SNMP version 1
 +
*Values: Integer. 1 enabled. 0 disabled
  
* '''Values:''' 0 = SIP INFO, 1 = RFC2833
+
===enable_v2c===
  
=== tone_volume ===
+
*Required: No.
* '''Required:''' No. Defaults to 0 
+
*Description: Enables reading of MIB using SNMP version 2c
 +
*Values: Integer. 1 enabled. 0 disabled
  
* '''Description:''' Control tone volume
+
===enable_ipsStarted===
  
* '''Values:''' -1 = no tones, 0 default volume, 1-4 increases volume
+
*Required: No. Defaults to 1
 +
*Description: If enabled, the station will send an SNMP trap when the station application is started
 +
*Values: 0 = disabled, 1 = enabled
  
== Relay Parameters ==
+
===enable_sipRegistered===
  
=== relay1_dtmf_activate ===
+
*Required: No. Defaults to 1
* '''Description:''' Dtmf value to send for activating the relay
+
*Description: If enabled, the station will send an SNMP trap when successfully registered in the SIP domain
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Valid values is 0-9, * and #. The character - means off.
+
===enable_sipRegisterFailed===
  
=== relay1_dtmf_deactivate ===
+
*Required: No. Defaults to 1
* '''Description:''' Dtmf value to send for deactivating the relay
+
*Description: If enabled, the station will send an SNMP trap if registration in the SIP domain failed
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Valid values is 0-9, * and #. The character - means off.
+
===enable_callConnect===
  
=== relay1_dtmf_flashing_slow ===
+
*Required: No. Defaults to 1
* '''Description:''' Dtmf value to send for setting the relay to flashing slow
+
*Description: If enabled, the station will send an SNMP trap when a call is connected
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Valid values is 0-9, * and #. The character - means off.
+
===enable_callConnectFailed===
  
=== relay1_dtmf_flashing_fast ===
+
*Required: No. Defaults to 1
* '''Description:''' Dtmf value to send for setting the relay to flashing fast
+
*Description: If enabled, the station will send an SNMP trap if a call to the station fails to connect for any reason (busy etc.)
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Valid values is 0-9, * and #. The character - means off.
+
===enable_callDisconnect===
  
=== relay1_dtmf_toggle ===
+
*Required: No. Defaults to 1
* '''Description:''' Dtmf value to send for toggling the relay
+
*Description: If enabled, the station will send an SNMP trap when a call is disconnected
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Valid values is 0-9, * and #. The character - means off.
+
===enable_buttonPressed===
  
=== relay1_dtmf_timed_relay ===
+
*Required: No. Defaults to 1
* '''Description:''' Dtmf value to send for activating the relay for X seconds
+
*Description: If enabled, the station will send an SNMP trap when an input button has been pressed
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Valid values is 0-9, * and #. The character - means off.
+
===enable_buttonReleased===
  
=== relay1_dtmf_timed_relay_duration ===
+
*Required: No. Defaults to 1
* '''Description:''' Duration to activate relay
+
*Description: If enabled, the station will send an SNMP trap when an input button has been released
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 means activate relay forever.
+
===enable_dakPressed===
  
=== relay1_event_out_ringing ===
+
*Required: No. Defaults to 1
* '''Description:''' When the station is ringing in an outgoing call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
+
*Description: If enabled, the station will send an SNMP trap when a DAK button has been pressed
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
+
===enable_dakReleased===
  
=== relay1_event_inc_ringing ===
+
*Required: No. Defaults to 1
* '''Description:''' When the station is ringing in an incoming call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
+
*Description: If enabled, the station will send an SNMP trap when a DAK button has been released
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
+
===enable_relayActivated===
  
=== relay1_event_inc_call ===
+
*Required: No. Defaults to 1
* '''Description:''' When the station is in an incoming call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
+
*Description: If enabled, the station will send an SNMP trap when a relay has been activated
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
+
===enable_relayDeactivated===
  
=== relay1_event_out_call ===
+
*Required: No. Defaults to 1
* '''Description:''' When the station is in an outgoing call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
+
*Description: If enabled, the station will send an SNMP trap when a relay has been deactivated
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
+
===enable_buttonHanging===
  
=== relay1_event_idle ===
+
*Required: No. Defaults to 1
* '''Description:''' When the station is idle, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
+
*Description: If enabled, the station will send an SNMP trap when a button is hanging (pressed for more than 10 seconds).
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast
+
===enable_soundTestSuccess===
  
=== relay1_event_error ===
+
*Required: No. Defaults to 1
* '''Description:''' When the station is error, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
+
*Description: If enabled, the station will send an SNMP trap when a sound test has been successfull
 +
*Values: 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast<br>
+
===enable_soundTestError===
  
== Tone Parameters ==
+
*Required: No. Defaults to 1
 +
*Description: If enabled, the station will send an SNMP trap when the tone test could not be carried out because the reference value ("silence") is not stable. This happens when measuring the "silence" several times, and the measured values are too different.
 +
*Values: 0 = disabled, 1 = enabled
  
=== enabled ===
+
===enable_soundTestFailed===
* '''Description:''' Enables tone test
 
  
* '''Values:''' Integer Value. 0 = disabled, 1 = enabled
+
*Required: No. Defaults to 1
 +
*Description: If enabled, the station will send an SNMP trap when a sound test has failed because the microphone didn’t get loud enough tone from the speaker.
 +
*Values: 0 = disabled, 1 = enabled
  
=== time_between_tonetest ===
+
==Detailed description of Audio Files parameters==
* '''Description:''' Time between tonetests
+
===tftpserver===
  
* '''Values:''' Integer Value.
+
*Required: No.
 +
*Description: IP Address of the TFTP server where audio files are located.
 +
*Values: String Value
  
=== sound_pressure_level ===
+
===ringing===
* '''Description:''' Time between tonetests
 
  
* '''Values:''' Integer Value. Only odd values. 53 ≤ ''sound_pressure_level'' ≤ 97
+
*Required: No.
 +
*Description: Wav file to be played during outgoing calls.
 +
*Values: String Value
  
=== volume ===
+
==Detailed description of Time parameters==
* '''Description:''' Volume of the tone test
+
===enabled===
  
* '''Values:''' Integer Value. 0 ≤ ''volume'' ≤ 7<br>
+
*Required: No.
 +
*Description: If NTP is enabled or not.
 +
*Values: Integer, 0 = disabled, 1 = enabled
  
== SNMP Parameters ==
+
===hostname===
  
=== trap_receiver ===
+
*Required: No.
* '''Required:''' No.
+
*Description: IP Address of hostname of NTP Server.
 +
*Values: String Value
  
* '''Description:''' The IP address of the server receiving SNMP traps.
+
===time_region (Turbine stations only)===
  
* '''Values:''' IP address given in regular dot notation, e.g. 10.5.2.100
+
*Required: No.
 +
*Description: Selected time region.
 +
*Values: String Value
  
=== network ===
+
===time_zone (Turbine stations only)===
* '''Required:''' No.
 
  
* '''Description:''' Used, together with the network mask, to determine the allowed network for reading the MIB on the IP station.
+
*Required: No.
 +
*Description: Selected time zone.
 +
*Values: String Value
  
* '''Values:''' IP address given in regular dot notation, e.g. 10.5.2.100. For example with an allowed network 10.5.2.0 and a network mask of 24, anyone with IP address 10.5.2.0 to 10.5.2.255 can access the MIB.
+
===plus_minus (INCA stations only)===
  
=== network_mask ===
+
*Required: No.
* '''Required:''' No.
+
*Description: If difference from GMT time is + or -.
 +
*Values: String, + = positive difference, - = negative difference
  
* '''Description:''' The mask used to determine the allowed network for reading the MIB.
+
===zone_hour (INCA stations only)===
  
* '''Values:''' Integer. 0 ≤ network_mask ≤ 32. For example with an allowed network 10.5.2.0 and a network mask of 24, anyone with IP address 10.5.2.0 to 10.5.2.255 can access the MIB.
+
*Required: No.
 +
*Description: Hour difference from GMT time
 +
*Values: Integer Value
  
=== community ===
+
===zone_minute (INCA stations only)===
* '''Required:''' No.
 
  
* '''Description:''' An text staring used as password for authentication.
+
*Required: No.
 +
*Description: Minutes difference from GMT time
 +
*Values: Integer Value
  
* '''Values:''' String.
+
===summertime (INCA stations only)===
  
=== enable_v1 ===
+
*Required: No.
* '''Required:''' No.
+
*Description:  Automatically adjust clock for 'Daylight Saving Time'.
 +
*Values: Integer, 0 = disabled, 1 = enabled.
  
* '''Description:''' Enables reading of MIB using SNMP version 1
+
==Detailed Description of DAVC parameters==
  
* '''Values:''' Integer. 1 enabled. 0 disabled
+
*NB! "avc_player_remotesLength" or "avc_target_receiversLength" have to be defined.
 +
*The user have to specify how many AVC source/receiver devices they want to add by define the "avc_player_remotesLengt" or "avc_target_receiversLength", otherwise the DAVC configs won't be applied to the station.
  
=== enable_v2c ===
+
===enable_davc===
* '''Required:''' No.
 
  
* '''Description:''' Enables reading of MIB using SNMP version 2c
+
*Required: No.
 +
*Description: *detail needed
 +
*Values: Integer, 0 = disabled, 1 = enabled
  
* '''Values:''' Integer. 1 enabled. 0 disabled
+
===avc_algorithm_enable===
  
=== enable_ipsStarted ===
+
*Required: No.
* '''Required:''' No. Defaults to 1
+
*Description: Enable AVC algorithm, and must be enabled on either all AVC Sources or all AVC Receivers.
 +
*Values: Integer, 0 = disabled, 1 = enabled
  
* '''Description:''' If enabled, the station will send an SNMP trap when the station application is started
+
===avc_algorithm_adjustmode===
  
* '''Values:''' 0 = disabled, 1 = enabled
+
*Required: No.
 +
*Description: Changing mode in Adjustment Mode.
 +
**Positive Mode: Gain adjustment is from zero at AVC threshold level and adjusted postive. AVC Receiver should have a low base gain.
 +
**Negative Mode: Gain adjustment is from zero at AVC max level and adjusted with negative gain downwards. AVC Receiver should have a high base gain.
 +
*Values: String, "positive" or "negative"
  
=== enable_sipRegistered ===
+
===avc_algorithm_lowerthreshold===
* '''Required:''' No. Defaults to 1
 
  
* '''Description:''' If enabled, the station will send an SNMP trap when successfully registered in the SIP domain
+
*Required: No.
 +
*Description: Mic signal level where AVC starts to work. Below this level no adjustment is done, and Audio Receivers should work on default base gain.
 +
*Values: Integer, Values between 30 dBA - 80 dBA, Have to be less than upper threshold value, default = 65 dBA.
  
* '''Values:''' 0 = disabled, 1 = enabled
+
===avc_algorithm_upperthreshold===
  
=== enable_sipRegisterFailed ===
+
*Required: No.
* '''Required:''' No. Defaults to 1
+
*Description: Mic signal level where AVC stops to work. Above this level, no adjustment is done. The diff between upper and lower thresholds also defines the working range for AVC gain adjustments. In negative AVC mode this also defined the ambient level where configured base output gain is reached (offset adjustment is zero)
 +
*Values: Integer, Values between 60 - 120 dBA, Have to be less than upper threshold value, default = 100 dB. (On firmware ver. 8.x and earlier the Values was between 80 - 120 dBA).
  
* '''Description:''' If enabled, the station will send an SNMP trap if registration in the SIP domain failed
+
===avc_algorithm_attackrate===
  
* '''Values:''' 0 = disabled, 1 = enabled
+
*Required: No.
 +
*Description: Determines how quickly the AVC adjusts gain on raising ambient audio level
 +
*Values: Float with 1 decimal, Values between 0.1 dB/sec to 200 dB/sec, default = 10.0 dB/sec.
  
=== enable_callConnect ===
+
===avc_algorithm_decayrate===
* '''Required:''' No. Defaults to 1
 
  
* '''Description:''' If enabled, the station will send an SNMP trap when a call is connected
+
*Required: No.
 +
*Description: Determines how quickly the AVC adjusts gain on falling ambient audio level
 +
*Values: Float with 1 decimal, Values between 0.1 dB/sec to 200 dB/sec, default = 1.0 dB/sec.
  
* '''Values:''' 0 = disabled, 1 = enabled
+
===avc_algorithm_hysteresis===
  
=== enable_callConnectFailed ===
+
*Required: No.
* '''Required:''' No. Defaults to 1
+
*Description: Hysteresis around previous set ambient audio level before doing adjustments
 +
*Values: Float with 2 decimal, Values between 0.00 dB to 200 dB, default = 3.00 dB.
  
* '''Description:''' If enabled, the station will send an SNMP trap if a call to the station fails to connect for any reason (busy etc.)
+
===avc_algorithm_farend_lockout_time===
  
* '''Values:''' 0 = disabled, 1 = enabled
+
*Required: No.
 +
*Description: When playing audio in AVC Zone (far-end-signal) all AVC adjustments is locked. When there is a pause in far-end-signal, adjustment commences after this lockout-time
 +
*Values: Float with 1 decimal, Values between 0.0 sec to 1.0 sec , default = 0.1 sec.
  
=== enable_callDisconnect ===
+
===avc_target_receiversLength===
* '''Required:''' No. Defaults to 1
 
  
* '''Description:''' If enabled, the station will send an SNMP trap when a call is disconnected
+
*Required: No.
 +
*Description: Define how many AVC receivers will be add, and the device is set to AVC Source
 +
*Values: Integer
  
* '''Values:''' 0 = disabled, 1 = enabled
+
===avc_target_receiver1_ipaddr===
  
== Example Configuration File ==
+
*Required: No.
 +
*Description: IP address of the first receivers, (For more receivers add "avc_algorithm_receiver3_ipaddr" and "avc_algorithm_receiver3_ipaddr" up to the length of receivers)
 +
*Values: String
  
 +
===avc_player_remotesLength===
  
 +
*Required: No.
 +
*Description: Define how many AVC source devices it will be and the device is set to AVC Receiver
 +
*Values: Integer
  
 +
===avc_player_remote_selection_strategy===
  
 +
*Required: No.
 +
*Description: Define the source selection strategy
 +
*Values: String, "highest", "averge" and "averge_mid", default = "highest"
  
 +
===avc_player_remote1_id===
  
 +
*Required: No.
 +
*Description: Source number of the first source (For more sources add "avc_player_remote2_id" etc)
 +
*Values: String
  
 +
===avc_player_remote1_ch1===
  
 +
*Required: No.
 +
*Description: Activate the audio channel 1 of the first source  with "out_1", and delete this parameter for deactivate it.
 +
*Values: String, "out_1" for activate
  
 +
===avc_player_remote1_ch2===
  
 +
*Required: No.
 +
*Description: Activate the audio channel 2 of the first source  with "out_2", and delete this parameter for deactivate it.
 +
*Values: String, "out_2" for activate
  
  
 +
==Detailed Description of Multicast Paging==
 +
===mcast_enabled===
  
 +
*Required: No.
 +
*Description: Activate Multicast Paging
 +
*Values: Integer, 0 = disable, 1 = enable
  
 +
===order_enabled===
  
 +
*Required: No.
 +
*Description: Activate Order Priority in multicast paging
 +
*Values: Integer, 0 = disable, 1 = enable
  
 +
===mcast_stream1_label===
  
 +
*Required: No.
 +
*Description: First Paging label ( "mcast_stream2_label" is the second paging label and etc)
 +
*Values: String.
  
 +
===mcast_stream1_address===
  
 +
*Required: No, have to define when mcast_stream1_label is defined.
 +
*Description: Listening address of First Paging label ( "mcast_stream2_address" is the listening address for second paging )
 +
*Values: String.
 +
 +
===mcast_stream1_port===
 +
 +
*Required: No, have to define when mcast_stream1_label is defined.
 +
*Description: Port number of the first paging label ( "mcast_stream2_port" is the port for second paging)
 +
*The ports must be within the port range 61020-61250, recommended range is 61080-61250. Using odd numbers is not recommended for ports. Listening address and port must be defined.
 +
*Values: Integer
 +
 +
===mcast_stream1_codec===
 +
 +
*Required: No, have to define when mcast_stream1_label is defined.
 +
*Description: Which codec is used for  the first paging label ( "mcast_stream2_codec" is the codec for second paging)
 +
*Values: String, Choose between "g729", "g729", "g711a", "g711u" and "L16x48D".
 +
 +
===mcast_stream1_priority===
 +
 +
*Required: No, have to define when mcast_stream1_label is defined.
 +
*Description: Set the priority for the first paging ( "mcast_stream2_priority" is the codec for second paging)
 +
*Values: String, Choose between "low", "normal", "high" and "emergency".
 +
 +
===mcast_stream1_order===
 +
 +
*Required: No, have to define when mcast_stream1_label is defined.
 +
*Description: Order of the first paging ( "mcast_stream2_order" is the order for second paging)
 +
*Values: Integer, Choose between 0 - 9.
 +
 +
===mcast_stream1_linemask===
 +
 +
*Required: No, have to define when mcast_stream1_label is defined.
 +
*Description: Select which Channels (Only for amplifiers)
 +
*Values: Integer, 1 = Channel 1, 2 = Channel 2, 3 = Channel 1 and Channel 2
 +
 +
==Intercom Example==
 
  [general]
 
  [general]
 
  auto_update_interval=10
 
  auto_update_interval=10
auto_update_image_type=tsi-3.0.1.50
 
Only for Turbine (8121)
 
 
  auto_update_image_type=A100G80200.01_10_1_2.bin
 
  auto_update_image_type=A100G80200.01_10_1_2.bin
Only for 8020-8024
 
 
  auto_update_image_crc=C1466499
 
  auto_update_image_crc=C1466499
  Only for 8020-8024
+
  [tone]
 +
volume=2
 +
time_between_tonetest=100
 +
enabled=0
 +
sound_pressure_level=65
 +
[relays]
 +
relay1_dtmf_activate=1
 +
relay1_dtmf_deactivate=2
 +
relay1_dtmf_flashing_slow=3
 +
relay1_dtmf_flashing_fast=4
 +
relay1_dtmf_toggle=-
 +
relay1_dtmf_timed_relay=8
 +
relay1_dtmf_timed_relay_duration=3
 +
relay1_event_out_ringing=1
 +
relay1_event_inc_ringing=0
 +
relay1_event_inc_call=2
 +
relay1_event_out_call=2
 +
relay1_event_idle=2
 +
relay1_event_error=4
 
  [sip]
 
  [sip]
 
  nick_name=Testname
 
  nick_name=Testname
Line 584: Line 974:
 
  sip_outbound_proxy=10.5.2.138
 
  sip_outbound_proxy=10.5.2.138
 
  sip_outbound_proxy_port=5060
 
  sip_outbound_proxy_port=5060
  register_interval=600
+
  register_interval=600                                       Value: 60 < seconds < 999999
Value: 60 < seconds < 999999
+
[sip_ch1]                                                  Amplifier channel 1 configuration
 +
playback_gain=-21
 +
sip_id=1006
 +
sip_domain=10.5.2.209
 +
[sip_ln1]                                                  Amplifier line in 1 configuration
 +
recorder_gain=15
 +
sip_id=1006
 +
sip_domain=10.5.2.209
 
  [call]
 
  [call]
  dak1_value=1008
+
  ringlist_max_conv_time=200
dak1_option=1
+
  ringlist_max_ring_time=30
dak2_value=1009
 
input1_value=1010
 
input2_value=1011
 
ringlist1_value1=1000
 
ringlist1_value2=1004@169.254.1.100
 
ringlist1_value3=
 
ringlist1_value4=
 
ringlist1_value5=
 
ringlist1_value6=
 
ringlist1_value7=
 
ringlist1_value8=
 
ringlist1_value9=
 
ringlist2_value1=1002
 
ringlist2_value2=1003@169.254.1.101
 
ringlist3_value1=1005
 
ringlist3_value2=1006@169.254.1.102
 
  ringlist_max_ring_time=60
 
 
  ringlist_loop=1
 
  ringlist_loop=1
  ringlist_max_ring_time=30
+
noise_reduction=2
  speaker_volume=4
+
echo_parameter=3
Value: 0 < level < 7.
+
input1_value=1000
  mic_sensitivity=5
+
input1_in_call_function=0                                      Input 1 will end current call if pressed during a call
Value: 0 < level < 7.
+
input2_value=1004@169.254.1.100
  rtp_timeout=60
+
input2_in_call_function=1                                      Input 2 will do nothing if pressed during a call
Value: 0 < seconds < 9999. 0 = RTP timeout disabled.
+
input3_value=
  remote_controlled_volume_override_mode=1
+
dak1_value=2000
Accepted values 0 or 1.
+
dak2_value=
  auto_answer_mode=1
+
dak3_value=
Accepted values 0 or 1.
+
ringlist_loop=0                                                Ringlists will not start at the beginning after trying to call all entries
  auto_answer_delay=10
+
ringlist_max_conv_time=600                                    Max conversation time of a call started with ringlist is 600 seconds
Value: 0 < seconds < 30
+
  ringlist_max_ring_time=50                                      Max ringing time of a call started with ringlist is 50 seconds
  [relays]
+
ringlist1_value1=1001                                          Ringlist 1 entry 1 will call to number 1001
relay1_dtmf_timed_relay=5
+
ringlist1_wp1=1                                                Ringlist 1 entry 1 will call at the same time as the previous entry
relay1_dtmf_timed_relay_duration=10
+
ringlist1_value2=1002                                          Ringlist 1 entry 2 will call to number 1002
relay1_event_inc_ringing=1
+
ringlist1_wp2=1                                                Ringlist 1 entry 2 will call at the same time as the previous entry
  relay1_event_inc_call=2
+
ringlist1_value3=1003
relay1_event_idle=1
+
ringlist1_value4=1004
 +
ringlist2_value1=1001
 +
ringlist2_value2=1002
 +
ringlist2_value3=1003
 +
ringlist2_value4=1004
 +
ringlist2_value5=1005
 +
ringlist3_value1=2001
 +
ringlist3_value2=2001
 +
  speaker_volume=4                                             Value: 0 < level < 7.
 +
  mic_sensitivity=5                                           Value: 0 < level < 7.
 +
  rtp_timeout=60                                               Value: 0 < seconds < 9999. 0 = RTP timeout disabled.
 +
  remote_controlled_volume_override_mode=1                     Accepted values 0 or 1.
 +
  auto_answer_mode=1                                           Accepted values 0 or 1.
 +
  auto_answer_delay=10                                         Value: 0 < seconds < 30
 +
  disable_disconnect_by_button=1                               Accepted values 0 or 1.
 +
   
 
  [snmp]
 
  [snmp]
 
  trap_receiver=10.5.2.219
 
  trap_receiver=10.5.2.219
 
  network=10.5.2.0
 
  network=10.5.2.0
 
  network_mask=24
 
  network_mask=24
 +
network_ipv6=fec0::
 +
network_ipv6_length=64
 
  community=public
 
  community=public
  enable_v1=1
+
  enable_v1=1                                                 Accepted values 0 or 1.
  Accepted values 0 or 1.
+
  enable_v2c=1                                                Accepted values 0 or 1.
  enable_v2c=1
+
  enable_ipsStarted=1                                         Accepted values 0 or 1.
  Accepted values 0 or 1.
+
  enable_sipRegistered=1                                      Accepted values 0 or 1.
  enable_ipsStarted=1
+
  enable_sipRegisterFailed=1                                 Accepted values 0 or 1.
  Accepted values 0 or 1.
+
  enable_callConnect=1                                        Accepted values 0 or 1.
  enable_sipRegistered=1
+
  enable_callConnectFailed=1                                 Accepted values 0 or 1.
  Accepted values 0 or 1.
+
  enable_callDisconnect=1                                    Accepted values 0 or 1.
  enable_sipRegisterFailed=1
+
  enable_buttonPressed=1                                     Accepted values 0 or 1.
  Accepted values 0 or 1.
+
  enable_buttonReleased=1                                    Accepted values 0 or 1.
  enable_callConnect=1
+
  enable_relayActivated=1                                     Accepted values 0 or 1.
  Accepted values 0 or 1.
+
  enable_relayDeactivated=1                                  Accepted values 0 or 1.
  enable_callConnectFailed=1
+
  enable_buttonHanging=1                                     Accepted values 0 or 1.
  Accepted values 0 or 1
+
  enable_soundTestSuccess=1                                  Accepted values 0 or 1.
  enable_callDisconnect=1
+
  enable_soundTestError=1                                     Accepted values 0 or 1.
  Accepted values 0 or 1
+
  enable_soundTestFailed=1                                    Accepted values 0 or 1.
       
+
 
        [[Category: Turbine Configuration]]
+
[audio_files]
     
+
tftpserver=10.8.25.200
       
+
ringing=ringing.wav
        [[Category:INCA Station Configuration Guide]]
+
 
 +
 
 +
[time]
 +
enabled=1
 +
hostname=10.8.25.200
 +
time_region=America
 +
time_zone=Denver
 +
 
 +
==Amplifier Example==
 +
[sip]
 +
sip_id=0203
 +
sip_domain=10.5.11.75
 +
nick_name=CCP03
 +
auth_user=0203
 +
auth_pwd=Ashley77
 +
 
 +
[call]
 +
# Use 3 GPI as key matrix for DAK1-7
 +
input_as_key_matrix=3
 +
io_pin1=0
 +
io_pin2=0
 +
io_pin3=0
 +
io_pin4=1
 +
io_pin5=1
 +
io_pin6=1
 +
fast_blink_pattern=1011111
 +
slow_blink_pattern=0000001000000
 +
# handset w/offhook - normally closed
 +
accessory=6
 +
# Allow speech mode to be overriden
 +
override_remote_ptt=1
 +
# use DTMF 9 go to open duplex
 +
open_duplex_dtmf=9
 +
# Allow maximum audio output
 +
poe_audio=1
 +
# Use RFC2833 to send DTMF
 +
dtmf_style=1
 +
# Disable tones
 +
tone_volume=-1
 +
# Use PTT as default speech mode
 +
speech_mode=1
 +
# auto answer enabled
 +
auto_answer_mode=1
 +
# reduced mic sensitivity
 +
mic_sensitivity=4
 +
# onhook send dtmf 8 in call
 +
onhook_in_call_function=2
 +
onhook_dtmf_on=8
 +
# Call 301
 +
dak1_value=0401
 +
dak1_in_call_function=0
 +
dak2_value=401
 +
dak2_in_call_function=0
 +
dak3_value=501
 +
dak3_in_call_function=0
 +
dak4_value=203
 +
dak4_in_call_function=0
 +
dak5_value=510
 +
dak5_in_call_function=0
 +
dak6_value=502
 +
dak6_in_call_function=0
 +
 
 +
[relays]
 +
gpio3_dtmf_activate=2
 +
gpio3_dtmf_deactivate=0
 +
gpio3_dtmf_flashing_slow=1
 +
gpio4_dtmf_activate=5
 +
gpio4_dtmf_deactivate=3
 +
gpio4_dtmf_flashing_slow=4
 +
gpio5_dtmf_activate=7
 +
gpio5_dtmf_deactivate=6
 +
Example - Amplifier
 +
[sip_ch1]
 +
nick_name=amp1_ch1_marius
 +
sip_id=0491
 +
auth_user=0491
 +
auth_pwd=Ashley77
 +
sip_domain=10.5.11.75
 +
playback_gain=-10
 +
 
 +
[sip_ch2]
 +
nick_name=Amp1_ch2_marius
 +
sip_id=0492
 +
auth_user=0492
 +
auth_pwd=Ashley77
 +
sip_domain=10.5.11.75
 +
playback_gain=-15 
 +
 
 +
  [[Category: Turbine Configuration]]
 +
 
 +
               
 +
  [[Category:INCA Station Configuration Guide]]
 +
 
 +
 
 +
  [[Category: SIP intercom - Configuration]]

Latest revision as of 09:32, 14 October 2024

SIP Icon 300px.png

This article describes the parameters in the configuration file used for TFTP Provisioning. See TFTP Provisioning for how to configure the station for this function.

TFTP Provisioning is supported by all IP Stations when operating in SIP mode.


Contents

Remote Provisioning using TFTP

An IP station may be set up to automatically poll configuration from a TFTP server. The IP address of this TFTP server can be obtained using DHCP procedures or be manually configured.

The IP station will first try to download the global configuration file:

ipst_config.cfg


Then the IP station will download a device specific configuration file:

ipst_config_01_02_03_04_05_06.cfg


where 01_02_03_04_05_06 is the MAC address of the IP station. Or:

ipst_config_10_9_8_7.cfg


where 10_9_8_7 is the IP address of the IP station.

If the same parameter is found in several files, then the precedence is as following:

  1. .MAC address file
  2. IP address file
  3. Global file
Note icon
  • The last line in the configuration file must be terminated with <Enter> (CR-LF), else the configuration will not be applied.
  • If there are changes in the configuration file that require a reboot to take effect, the station will do a reboot when the cofiguration file is downloaded and changes are applied
  • If there are no changes in the downloaded configuration file compared to the previous downloaded file, the station will ignore the new file



Detailed description of general parameters

auto_update_interval

  • Required: No. If this parameter's not set in the file, the function will be disabled
  • Description: This parameter enables the station to automatically look for software updates on the TFTP server
  • Values: Number of minutes to wait between each server request. Value must be between 1 and 999

auto_update_image_type

  • Required: If auto_update_interval is set
  • Description: The name of the software image file to be uploaded
  • Values: Text giving the name of the software image file. The full name of the file, including extension, is required. This parameter must be set if the auto update function is enabled

auto_update_image_crc

  • Required: If auto_update_interval is set
  • Description: The CRC checksum calculated for the software image file specified by the auto_update_image_type parameter. This is used to check the integrity of the software file before updating the station
  • Values: Hexadecimal value

use_last_known_ip (INCA stations only)

  • Required: No. Default is 0.
  • Description: Use Last IP On DHCP failure.
  • Values: Integer Value. 0 = disabled, 1 = enabled

turbine_frontboard (deprecated from VS-IS 7.1)

  • Description: Configures the turbine frontboard type
  • Values:
    • 0 = KIT
    • 1 = TCIS-1, TCIS-2, TCIS-3
    • 2 = TCIS-6
    • 3 = TCIS4, TCIS-5
    • 4 = TFIE-1, TFIX-1
    • 5 = TFIE-2, TFIX-2
    • 6 = TFIX-3
    • 7 = ECPIR-P
    • 8 = EAPII-1, EAPFX-1
    • 9 = EAPII-6, EAPFX-6
    • 10 = ECPIR-3P
    • 11 = EAPIR-8
    • 13 = TFIE-6
    • 45 = IP-LCM
    • 51 = TCIV-2, TCIV-3
    • 52 = TCIV-6
    • 53 = TMIS1, TMIS-2
    • 54 = TCIV-5

Detailed description of Relay parameter

  • The following relay keys are supported:
    • relay1 and relay2 for station physical relays
    • gpio1 to gpio6 for I/O pins configured as outputs
    • e_relay1 and e_relay2 for TA-10 relay module
  • To configure specific relay rename the parameter to contain the proper relay key, i.e. instead of "relay1_dtmf_activate" use "gpio1_dtmf_activate".

relay1_dtmf_activate

  • Description: Dtmf value to send for activating the relay
  • Values: Valid values is 0-9, * and #. The character - means off.

relay1_dtmf_deactivate

  • Description: Dtmf value to send for deactivating the relay
  • Values: Valid values is 0-9, * and #. The character - means off.

relay1_dtmf_flashing_slow

  • Description: Dtmf value to send for setting the relay to flashing slow
  • Values: Valid values is 0-9, * and #. The character - means off.

relay1_dtmf_flashing_fast

  • Description: Dtmf value to send for setting the relay to flashing fast
  • Values: Valid values is 0-9, * and #. The character - means off.

relay1_dtmf_toggle

  • Description: Dtmf value to send for toggling the relay
  • Values: Valid values is 0-9, * and #. The character - means off.

relay1_dtmf_timed_relay

  • Description: Dtmf value to send for activating the relay for X seconds
  • Values: Valid values is 0-9, * and #. The character - means off.

relay1_dtmf_timed_relay_duration

  • Description: Duration to activate relay
  • Values: Integer. 0 means activate relay forever.

relay1_event_out_ringing

  • Description: When the station is ringing in an outgoing call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

relay1_event_inc_ringing

  • Description: When the station is ringing in an incoming call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

relay1_event_inc_call

  • Description: When the station is in an incoming call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

relay1_event_out_call

  • Description: When the station is in an outgoing call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

relay1_event_group_call

  • Description: When the station is in a group call, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

relay1_event_idle

  • Description: When the station is idle, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

relay1_event_error

  • Description: When the station is error, turn the relay to state: activated, deactivated, flashing slow, flashing fast, do nothing
  • Values: Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast

Detailed description of Tone parameter

enabled

  • Description: Enables tone test
  • Values: Integer Value. 0 = disabled, 1 = enabled

time_between_tonetest

  • Description: Time between tone tests
  • Values: Integer Value.

sound_pressure_level (obsolete for SW > 6.1.3.0)

  • Description: Minimum sound pressure level between silence and tone
  • Values: Integer Value. Only odd values. 53 ≤ sound_pressure_level ≤ 97

volume (obsolete for SW > 6.1.3.0)

  • Description: Volume of the tone test
  • Values: Integer Value. 0 ≤ volume ≤ 7

auto_set_sound_pressure_level (obsolete for non-INCA stations)

  • Description: Only available on INCA stations, station tries to calculate the parameter 'Minimum sound pressure level' (sound_pressure_level).
  • Values: Integer Value. 0 = disabled, 1 = enabled

tone_selection (SW > 6.1.3.0)

  • Description: Defines frequency groups for the tone test.
  • Values: Integer value. 1 <= tone_selection <= 8

Detailed description of SIP parameters

nick_name

  • Required: No. Defaults to sip_id
  • Description: The nick name for the station can be used to assign a logical name to the station. E.g. a station belonging to James may be assigned the nick name "James", or "James' station"
  • Values: Text string. Max length is 64 characters.

sip_id

  • Required: Yes
  • Description: This is the identification of the station in the SIP domain, i.e. the phone number for the station
  • Values: Integer value. Max length is 64 characters.

sip_domain

  • Required: Yes
  • Description: SIP domain is a server that uses SIP (Session Initiation Protocol) to manage real-time communication among SIP clients. The sip_domain parameter specifies the primary domain for the station, as opposed to sip_domain2 which specifies the secondary (or fall back) domain. The IP address for the SIP domain server (e.g. Asterisk or Cisco Call Manager) should be defined in this section
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100

sip_domain2

  • Required: No
  • Description: This is the secondary (or fall-back) domain. If the station loses connection to the primary SIP domain, it will switch over to the secondary.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100

sip_domain3

  • Required: No
  • Description: This is the tertiary (or fall-back) domain. If the station loses connection to the primary and secondary SIP domain, it will switch over to the tertiary.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100

registration_method

  • Required: No
  • Description: Registration method which the station will use if fall-back domains are available.
  • Values: parallel = Parallel registration to fall-back domains, serial = Serial registration to fall-back domains

auth_user

  • Required: Only if the SIP server requires authentication
  • Description: The authentication user name used to register the station to the SIP server.
  • Values: Text string.

auth_pwd

  • Required: Only if the SIP server requires authentication
  • Description: The authentication user password used to register the station to the SIP server.
  • Values: Text string.

sip_outbound_proxy

  • Required: Optional
  • Description: Configures an outbound-proxy server that receives all initiating request (INVITE and SUBSCRIBE) messages.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100

sip_outbound_proxy_port

  • Required: If proxy server is defined. Default 5060.
  • Description: The UDP port used for SIP on the proxy server.
  • Values: Integer.

register_interval

  • Required: No. Defaults to 600 seconds
  • Description: This parameter specifies how often the station will register, and reregister, in the SIP domain. This parameter will affect the time it takes to discover that a connection to a SIP domain is lost
  • Values: Number of seconds. 60 ≤ register_interval ≤ 999999

fail_interval (since 4.9.3.2)

  • Required: No. Defaults to 60 seconds
  • Description: In case Primary and both Backup servers are failing with SIP INVITEs, the device should go into failure mode, and immediately start sending REGISTER requests to all SIP servers, in time periods using this failure interval.
  • Values: Number of seconds. 5 ≤ fail_interval ≤ 999999

playback_gain (RS amplifier only)

  • Required: No.
  • Description: This parameter specifies the gain on a output channel
  • Values: dB. -40 ≤ playback_gain ≤ 0

recorder_gain (RS amplifier only)

  • Required: No.
  • Description: This parameter specifies the gain on a input channel
  • Values: dB. 0 ≤ recorder_gain ≤ 40

Detailed description of Call parameters

  • DAK, input and special keys configuration have the same parameters available, the exact name of the parameter will depend on the desired key ID and the name must start with that key ID.
  • Depending on the station type, the following key IDs will be available:
    • input1 to input6
    • dak1 to dak10
    • offhook
    • onhook
    • ptt
  • For example, parameters name would then look like this:
    • input2_function
    • dak1_function
    • offhook_function
    • onhook_function
    • dak_function
  • This document will use "dak1" as an example for configuration parameters.

dak1_value

  • Required: Yes for input1_function values 0 (default, call to), 3 (forward call) and4 (group call).
  • Description: This is the SIP ID for the extension to be called when the first DAK button is pressed, i.e. the telephone number of the receiving party. It's also used if input1_function is set to 3 (forward call to SIP ID) or 4 (group call to SIP ID).
  • Values: String value

dak1_option

  • Required: No
  • Description: Used in combination with call to function, configures if ringlist will be used or not.
  • Values: Integer. -1 is no ringlist. 0 is ringlist 1. 1 is ringlist 2. 2 is ringlist 3.

dak1_text (DualDisplay stations only)

  • Required: No
  • Description: Text which will be visible on the display next to DAK button 1. Note that this option will do nothing on inputs.
  • Values: String value

dak1_function

  • Required: No. Default value is 0 (call to)
  • Description: This decides what input button 1 will do when outside calls.
  • Values: Integer. 0 is call to.1 is do nothing. 3 is forward call (Turbine only). 4 is group call (Turbine only). 5 is conversation mode (Turbine only). 6 is volume control (Turbine only).

dak1_in_call_function

  • Required: No. Default value is 1 (do nothing)
  • Description: This decides what input button 1 will do when in calls.
  • Values: Integer. 0 is answer/end call.1 is do nothing. 2 is send DTMF. 3 is send text. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer

dak1_dtmf_on

  • Required: No. Only used with in call function "Send DTMF".
  • Description: Which DTMF value will be sent when DAK button 1 is pressed.
  • Values: String. Supported values are from 0 to 9, and *, #, ! (DTMF Flash), O (Do Nothing).

dak1_dtmf_off

  • Required: No. Only used with in call function "Send DTMF".
  • Description: Which DTMF value will be sent when DAK button 1 is released.
  • Values: String. Supported values are from 0 to 9, and *, #, ! (DTMF Flash), O (Do Nothing).

dak1_text_on

  • Required: No. Only used with in call function "Send Text".
  • Description: Which text value will be sent when DAK button 1 is pressed.
  • Values: String.

dak1_text_off

  • Required: No. Only used with in call function "Send Text".
  • Description: Which text value will be sent when DAK button 1 is released.
  • Values: String.

ringlist1_value1

  • Required: No
  • Description: This is the SIP ID for the extension to be called when ringlist 1 is used and it is at first entry. The next numbers in the ringlist is then ringlist1_value2, ringlist1_value3 etc.
  • Values: String value

ringlist1_wp1

  • Required: No, default value is 0 (disabled)
  • Description: Call with previous parameter for the first entry in the ringlist 1. The next numbers in the ringlist is then ringlist1_wp2, ringlist1_wp3 etc.
  • Values: Integer. 0 = disabled, 1 = enabled

ringlist2_value1

  • Required: No
  • Description: This is the SIP ID for the extension to be called when ringlist 2 is used and it is at the first entry. The next numbers in the ringlist is then ringlist2_value2, ringlist2_value3 etc.
  • Values: String value

ringlist2_wp1

  • Required: No, default value is 0 (disabled)
  • Description: Call with previous parameter for the first entry in the ringlist 2. The next numbers in the ringlist is then ringlist2_wp2, ringlist2_wp3 etc.
  • Values: Integer. 0 = disabled, 1 = enabled

ringlist3_value1

  • Required: No
  • Description: This is the SIP ID for the extension to be called when ringlist 3 is used and it is at the first entry. The next numbers in the ringlist is then ringlist3_value2, ringlist3_value3 etc.
  • Values: String value

ringlist3_wp1

  • Required: No, default value is 0 (disabled)
  • Description: Call with previous parameter for the first entry in the ringlist 3. The next numbers in the ringlist is then ringlist3_wp2, ringlist3_wp3 etc.
  • Values: Integer. 0 = disabled, 1 = enabled

ringlist_max_ring_time

  • Required: No. Defaults to 4
  • Description: This parameter sets the time to wait ringing until step to next value in the ringlist
  • Values: Number of seconds. 0 ≤ ringlist_max_ring_time ≤ 999999

ringlist_loop

  • Required: No. Defaults to 0
  • Description: This parameter enable the loop so list is repeated until answer.
  • Values: Integer. 0 to disable, 1 to enable

ringlist_max_conv_time

  • Required: No. Defaults to 4
  • Description: This parameter sets the max time of the conversation when followed ringlist
  • Values: Number of seconds. 0 ≤ ringlist_max_conv_time ≤ 999999

conv_time

  • Required: No. Defaults to 3600
  • Description: This parameter sets the max time of the conversation
  • Values: Number of seconds. 0 ≤ conv_time ≤ 999999

ring_time

  • Required: No. Defaults to 120
  • Description: This parameter sets the max time of the ringing/alerting phase of the call
  • Values: Number of seconds. 0 ≤ ring_time ≤ 999999

poe_audio (obsolete)

  • Description: This parameter sets whether to disable maximum speaker output
  • Values: Set to 0 to enable full audio output. Set to 1 to disable full audio output.

speaker_volume

  • Required: No. Defaults to 4
  • Description: This parameter sets the volume of the station's speaker
  • Values: Integer. 0 ≤ speaker_volume ≤ 7

mic_sensitivity

  • Required: No. Defaults to 5
  • Description: This parameter adjusts the microphone sensitivity
  • Values: Integer. 0 ≤ mic_sensitivity ≤ 7

noise_reduction

  • Required: No. Defaults to 0
  • Description: This parameter adjusts the noise reduction level
  • Values: Integer. 0 ≤ noise_reduction ≤ 7

echo_parameter (INCA stations only)

  • Required: No. Defaults to 0
  • Description: This parameter adjusts the echo parameter
  • Values: Integer. 0 ≤ echo_parameter ≤ 7

rtp_timeout

  • Required: No. Defaults to 0
  • Description: Cancels a call if the station does not receive rtp.
  • Values: Integer value: 0-9999 seconds. 0 = RTP timeout disabled.

remote_controlled_volume_override_mode

  • Required: No.
  • Description: Acts as a simplex mode after first DTMF * or # is received. At remote station: send DTMF * to talk and # to listen.
  • Values: Integer. 0 disabled, 1 enabled.

speech_mode

  • Required: No.
  • Description: Set the conversation mode.
  • Values: Integer. 0 = Full Open Duplex, 1 = Push To Talk, 2 = Half Duplex Switching, 3 = Open, 4 = Robust Duplex.

ptt_mode

  • Required: No.
  • Description: Set the PTT mode, only active if Conversation Mode is set to PTT.
  • Values: ptt_mic_and_speaker = Mic and speaker is controlled by PTT button, ptt_mic_only = Mic is controlled by PTT button.

auto_answer_mode

  • Required: No.
  • Description: Enables autoanswer after a set number of seconds.
  • Values: Integer. 0 disabled, 1 enabled.

auto_answer_delay

  • Required: No. Defaults to 0.
  • Description: The number of seconds to delay the autoanswer
  • Values: Integer. 0 ≤ delay ≤ 30

accessory

  • Required: No. Defaults to 0.
  • Description: Which accessory to use
  • Values: 0 = unused/default, 1 = handset, 2 = microphone w/ptt, 3 = headset, 4 = handset w/offhook, 5 = headset auto detect, 6 = handset w/offhook normally closed

input_as_key_matrix

  • Required: No. Defaults to 0.
  • Description: Use inputs as a key matrix. Requires that gpio is configured as gpi/input
  • Values: 0 = no inputs as key, 1 = means 1 input as as key matrix (1 dak), 2 = means 2 inputs as as key matrix (3 daks), 3 = means 3 inputs as as key matrix (7 daks)

fast_blink_pattern

  • Required: No. Defaults to 111000111000111000111000
  • Description: Customize fast blink pattern
  • Values: 1 = gpo high, 0 = gpo low

slow_blink_pattern

  • Required: No. Defaults to 111111111111000000000000
  • Description: Customize slow blink pattern
  • Values: 1 = gpo high, 0 = gpo low

open_duplex_dtmf

  • Required: No. Defaults to -
  • Description: Forces the station in Open Duplex when configured DTMF is received
  • Values: - = off, valid range: 0-9

override_remote_ptt

  • Required: No. Defaults to 0
  • Description: If 2 stations call each other and Override Remote PTT is enabled, then conversation mode is switched to open duplex.
  • Values: 0 = disabled, 1 = enabled

dtmf_style

  • Required: No. Defaults to 0
  • Description: Choose how to send DTMF
  • Values: 0 = SIP INFO, 1 = RFC2833

tone_volume

  • Required: No. Defaults to 0
  • Description: Control tone volume
  • Values: -1 = no tones, 0 default volume, 1-4 increases volume

codec_g729_pri

  • Required: No
  • Description: Set the priority of g729 codec
  • Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority

codec_g722_pri

  • Required: No
  • Description: Set the priority of g722 codec
  • Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority

codec_g711u_pri

  • Required: No
  • Description: Set the priority of g711u codec
  • Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority

codec_g711a_pri

  • Required: No
  • Description: Set the priority of g711a codec
  • Values: unused = Not Used, low = Low Priority, medium = Medium priority, high = High Priority

io_pin1

  • Required: No
  • Description: Will the I/O pin 1 behave as an input or as an output.
  • Values: 0 = Input, 1 = Output

io_pin2

  • Required: No
  • Description: Will the I/O pin 2 behave as an input or as an output.
  • Values: 0 = Input, 1 = Output

io_pin3

  • Required: No
  • Description: Will the I/O pin 3 behave as an input or as an output.
  • Values: 0 = Input, 1 = Output

io_pin4

  • Required: No
  • Description: Will the I/O pin 4 behave as an input or as an output.
  • Values: 0 = Input, 1 = Output

io_pin5

  • Required: No
  • Description: Will the I/O pin 5 behave as an input or as an output.
  • Values: 0 = Input, 1 = Output

io_pin6

  • Required: No
  • Description: Will the I/O pin 6 behave as an input or as an output.
  • Values: 0 = Input, 1 = Output

Detailed description of SNMP parameters

trap_receiver

  • Required: No.
  • Description: The IP address of the server receiving SNMP traps.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100

inform_receiver

  • Required: No.
  • Description: The IP address of the server receiving SNMP informs.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100

network

  • Required: No.
  • Description: Used, together with the network mask, to determine the allowed network for reading the MIB on the IP station.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100. For example with an allowed network 10.5.2.0 and a network mask of 24, anyone with IP address 10.5.2.0 to 10.5.2.255 can access the MIB.

network_mask

  • Required: No.
  • Description: The mask used to determine the allowed network for reading the MIB.
  • Values: Integer. 0 ≤ network_mask ≤ 32. For example with an allowed network 10.5.2.0 and a network mask of 24, anyone with IP address 10.5.2.0 to 10.5.2.255 can access the MIB.

community

  • Required: No.
  • Description: An text staring used as password for authentication.
  • Values: String.

enable_v1

  • Required: No.
  • Description: Enables reading of MIB using SNMP version 1
  • Values: Integer. 1 enabled. 0 disabled

enable_v2c

  • Required: No.
  • Description: Enables reading of MIB using SNMP version 2c
  • Values: Integer. 1 enabled. 0 disabled

enable_ipsStarted

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when the station application is started
  • Values: 0 = disabled, 1 = enabled

enable_sipRegistered

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when successfully registered in the SIP domain
  • Values: 0 = disabled, 1 = enabled

enable_sipRegisterFailed

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap if registration in the SIP domain failed
  • Values: 0 = disabled, 1 = enabled

enable_callConnect

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a call is connected
  • Values: 0 = disabled, 1 = enabled

enable_callConnectFailed

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap if a call to the station fails to connect for any reason (busy etc.)
  • Values: 0 = disabled, 1 = enabled

enable_callDisconnect

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a call is disconnected
  • Values: 0 = disabled, 1 = enabled

enable_buttonPressed

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when an input button has been pressed
  • Values: 0 = disabled, 1 = enabled

enable_buttonReleased

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when an input button has been released
  • Values: 0 = disabled, 1 = enabled

enable_dakPressed

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a DAK button has been pressed
  • Values: 0 = disabled, 1 = enabled

enable_dakReleased

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a DAK button has been released
  • Values: 0 = disabled, 1 = enabled

enable_relayActivated

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a relay has been activated
  • Values: 0 = disabled, 1 = enabled

enable_relayDeactivated

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a relay has been deactivated
  • Values: 0 = disabled, 1 = enabled

enable_buttonHanging

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a button is hanging (pressed for more than 10 seconds).
  • Values: 0 = disabled, 1 = enabled

enable_soundTestSuccess

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a sound test has been successfull
  • Values: 0 = disabled, 1 = enabled

enable_soundTestError

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when the tone test could not be carried out because the reference value ("silence") is not stable. This happens when measuring the "silence" several times, and the measured values are too different.
  • Values: 0 = disabled, 1 = enabled

enable_soundTestFailed

  • Required: No. Defaults to 1
  • Description: If enabled, the station will send an SNMP trap when a sound test has failed because the microphone didn’t get loud enough tone from the speaker.
  • Values: 0 = disabled, 1 = enabled

Detailed description of Audio Files parameters

tftpserver

  • Required: No.
  • Description: IP Address of the TFTP server where audio files are located.
  • Values: String Value

ringing

  • Required: No.
  • Description: Wav file to be played during outgoing calls.
  • Values: String Value

Detailed description of Time parameters

enabled

  • Required: No.
  • Description: If NTP is enabled or not.
  • Values: Integer, 0 = disabled, 1 = enabled

hostname

  • Required: No.
  • Description: IP Address of hostname of NTP Server.
  • Values: String Value

time_region (Turbine stations only)

  • Required: No.
  • Description: Selected time region.
  • Values: String Value

time_zone (Turbine stations only)

  • Required: No.
  • Description: Selected time zone.
  • Values: String Value

plus_minus (INCA stations only)

  • Required: No.
  • Description: If difference from GMT time is + or -.
  • Values: String, + = positive difference, - = negative difference

zone_hour (INCA stations only)

  • Required: No.
  • Description: Hour difference from GMT time
  • Values: Integer Value

zone_minute (INCA stations only)

  • Required: No.
  • Description: Minutes difference from GMT time
  • Values: Integer Value

summertime (INCA stations only)

  • Required: No.
  • Description: Automatically adjust clock for 'Daylight Saving Time'.
  • Values: Integer, 0 = disabled, 1 = enabled.

Detailed Description of DAVC parameters

  • NB! "avc_player_remotesLength" or "avc_target_receiversLength" have to be defined.
  • The user have to specify how many AVC source/receiver devices they want to add by define the "avc_player_remotesLengt" or "avc_target_receiversLength", otherwise the DAVC configs won't be applied to the station.

enable_davc

  • Required: No.
  • Description: *detail needed
  • Values: Integer, 0 = disabled, 1 = enabled

avc_algorithm_enable

  • Required: No.
  • Description: Enable AVC algorithm, and must be enabled on either all AVC Sources or all AVC Receivers.
  • Values: Integer, 0 = disabled, 1 = enabled

avc_algorithm_adjustmode

  • Required: No.
  • Description: Changing mode in Adjustment Mode.
    • Positive Mode: Gain adjustment is from zero at AVC threshold level and adjusted postive. AVC Receiver should have a low base gain.
    • Negative Mode: Gain adjustment is from zero at AVC max level and adjusted with negative gain downwards. AVC Receiver should have a high base gain.
  • Values: String, "positive" or "negative"

avc_algorithm_lowerthreshold

  • Required: No.
  • Description: Mic signal level where AVC starts to work. Below this level no adjustment is done, and Audio Receivers should work on default base gain.
  • Values: Integer, Values between 30 dBA - 80 dBA, Have to be less than upper threshold value, default = 65 dBA.

avc_algorithm_upperthreshold

  • Required: No.
  • Description: Mic signal level where AVC stops to work. Above this level, no adjustment is done. The diff between upper and lower thresholds also defines the working range for AVC gain adjustments. In negative AVC mode this also defined the ambient level where configured base output gain is reached (offset adjustment is zero)
  • Values: Integer, Values between 60 - 120 dBA, Have to be less than upper threshold value, default = 100 dB. (On firmware ver. 8.x and earlier the Values was between 80 - 120 dBA).

avc_algorithm_attackrate

  • Required: No.
  • Description: Determines how quickly the AVC adjusts gain on raising ambient audio level
  • Values: Float with 1 decimal, Values between 0.1 dB/sec to 200 dB/sec, default = 10.0 dB/sec.

avc_algorithm_decayrate

  • Required: No.
  • Description: Determines how quickly the AVC adjusts gain on falling ambient audio level
  • Values: Float with 1 decimal, Values between 0.1 dB/sec to 200 dB/sec, default = 1.0 dB/sec.

avc_algorithm_hysteresis

  • Required: No.
  • Description: Hysteresis around previous set ambient audio level before doing adjustments
  • Values: Float with 2 decimal, Values between 0.00 dB to 200 dB, default = 3.00 dB.

avc_algorithm_farend_lockout_time

  • Required: No.
  • Description: When playing audio in AVC Zone (far-end-signal) all AVC adjustments is locked. When there is a pause in far-end-signal, adjustment commences after this lockout-time
  • Values: Float with 1 decimal, Values between 0.0 sec to 1.0 sec , default = 0.1 sec.

avc_target_receiversLength

  • Required: No.
  • Description: Define how many AVC receivers will be add, and the device is set to AVC Source
  • Values: Integer

avc_target_receiver1_ipaddr

  • Required: No.
  • Description: IP address of the first receivers, (For more receivers add "avc_algorithm_receiver3_ipaddr" and "avc_algorithm_receiver3_ipaddr" up to the length of receivers)
  • Values: String

avc_player_remotesLength

  • Required: No.
  • Description: Define how many AVC source devices it will be and the device is set to AVC Receiver
  • Values: Integer

avc_player_remote_selection_strategy

  • Required: No.
  • Description: Define the source selection strategy
  • Values: String, "highest", "averge" and "averge_mid", default = "highest"

avc_player_remote1_id

  • Required: No.
  • Description: Source number of the first source (For more sources add "avc_player_remote2_id" etc)
  • Values: String

avc_player_remote1_ch1

  • Required: No.
  • Description: Activate the audio channel 1 of the first source with "out_1", and delete this parameter for deactivate it.
  • Values: String, "out_1" for activate

avc_player_remote1_ch2

  • Required: No.
  • Description: Activate the audio channel 2 of the first source with "out_2", and delete this parameter for deactivate it.
  • Values: String, "out_2" for activate


Detailed Description of Multicast Paging

mcast_enabled

  • Required: No.
  • Description: Activate Multicast Paging
  • Values: Integer, 0 = disable, 1 = enable

order_enabled

  • Required: No.
  • Description: Activate Order Priority in multicast paging
  • Values: Integer, 0 = disable, 1 = enable

mcast_stream1_label

  • Required: No.
  • Description: First Paging label ( "mcast_stream2_label" is the second paging label and etc)
  • Values: String.

mcast_stream1_address

  • Required: No, have to define when mcast_stream1_label is defined.
  • Description: Listening address of First Paging label ( "mcast_stream2_address" is the listening address for second paging )
  • Values: String.

mcast_stream1_port

  • Required: No, have to define when mcast_stream1_label is defined.
  • Description: Port number of the first paging label ( "mcast_stream2_port" is the port for second paging)
  • The ports must be within the port range 61020-61250, recommended range is 61080-61250. Using odd numbers is not recommended for ports. Listening address and port must be defined.
  • Values: Integer

mcast_stream1_codec

  • Required: No, have to define when mcast_stream1_label is defined.
  • Description: Which codec is used for the first paging label ( "mcast_stream2_codec" is the codec for second paging)
  • Values: String, Choose between "g729", "g729", "g711a", "g711u" and "L16x48D".

mcast_stream1_priority

  • Required: No, have to define when mcast_stream1_label is defined.
  • Description: Set the priority for the first paging ( "mcast_stream2_priority" is the codec for second paging)
  • Values: String, Choose between "low", "normal", "high" and "emergency".

mcast_stream1_order

  • Required: No, have to define when mcast_stream1_label is defined.
  • Description: Order of the first paging ( "mcast_stream2_order" is the order for second paging)
  • Values: Integer, Choose between 0 - 9.

mcast_stream1_linemask

  • Required: No, have to define when mcast_stream1_label is defined.
  • Description: Select which Channels (Only for amplifiers)
  • Values: Integer, 1 = Channel 1, 2 = Channel 2, 3 = Channel 1 and Channel 2

Intercom Example

[general]
auto_update_interval=10
auto_update_image_type=A100G80200.01_10_1_2.bin
auto_update_image_crc=C1466499
[tone]
volume=2
time_between_tonetest=100
enabled=0
sound_pressure_level=65
[relays]
relay1_dtmf_activate=1
relay1_dtmf_deactivate=2
relay1_dtmf_flashing_slow=3
relay1_dtmf_flashing_fast=4
relay1_dtmf_toggle=-
relay1_dtmf_timed_relay=8
relay1_dtmf_timed_relay_duration=3
relay1_event_out_ringing=1
relay1_event_inc_ringing=0
relay1_event_inc_call=2
relay1_event_out_call=2
relay1_event_idle=2
relay1_event_error=4
[sip]
nick_name=Testname
sip_id=1003
sip_domain=10.5.2.209
sip_domain2=10.5.2.138
auth_user=1003
auth_pwd=1003pass
sip_outbound_proxy=10.5.2.138
sip_outbound_proxy_port=5060
register_interval=600                                        Value:  60 < seconds < 999999
[sip_ch1]                                                  Amplifier channel 1 configuration
playback_gain=-21
sip_id=1006
sip_domain=10.5.2.209
[sip_ln1]                                                  Amplifier line in 1 configuration
recorder_gain=15
sip_id=1006
sip_domain=10.5.2.209
[call]
ringlist_max_conv_time=200
ringlist_max_ring_time=30
ringlist_loop=1
noise_reduction=2
echo_parameter=3
input1_value=1000
input1_in_call_function=0                                      Input 1 will end current call if pressed during a call
input2_value=1004@169.254.1.100
input2_in_call_function=1                                      Input 2 will do nothing if pressed during a call
input3_value=
dak1_value=2000
dak2_value=
dak3_value=
ringlist_loop=0                                                Ringlists will not start at the beginning after trying to call all entries
ringlist_max_conv_time=600                                     Max conversation time of a call started with ringlist is 600 seconds
ringlist_max_ring_time=50                                      Max ringing time of a call started with ringlist is 50 seconds
ringlist1_value1=1001                                          Ringlist 1 entry 1 will call to number 1001
ringlist1_wp1=1                                                Ringlist 1 entry 1 will call at the same time as the previous entry
ringlist1_value2=1002                                          Ringlist 1 entry 2 will call to number 1002
ringlist1_wp2=1                                                Ringlist 1 entry 2 will call at the same time as the previous entry
ringlist1_value3=1003
ringlist1_value4=1004
ringlist2_value1=1001
ringlist2_value2=1002
ringlist2_value3=1003
ringlist2_value4=1004
ringlist2_value5=1005
ringlist3_value1=2001
ringlist3_value2=2001
speaker_volume=4                                             Value: 0 < level < 7.
mic_sensitivity=5                                            Value: 0 < level < 7.
rtp_timeout=60                                               Value: 0 < seconds < 9999. 0 = RTP timeout disabled.
remote_controlled_volume_override_mode=1                     Accepted values 0 or 1.
auto_answer_mode=1                                           Accepted values 0 or 1.
auto_answer_delay=10                                         Value: 0 < seconds < 30
disable_disconnect_by_button=1                               Accepted values 0 or 1.

[snmp]
trap_receiver=10.5.2.219
network=10.5.2.0
network_mask=24
network_ipv6=fec0::
network_ipv6_length=64
community=public
enable_v1=1                                                 Accepted values 0 or 1.
enable_v2c=1                                                Accepted values 0 or 1.
enable_ipsStarted=1                                         Accepted values 0 or 1.
enable_sipRegistered=1                                      Accepted values 0 or 1.
enable_sipRegisterFailed=1                                  Accepted values 0 or 1.
enable_callConnect=1                                        Accepted values 0 or 1.
enable_callConnectFailed=1                                  Accepted values 0 or 1.
enable_callDisconnect=1                                     Accepted values 0 or 1.
enable_buttonPressed=1                                      Accepted values 0 or 1.
enable_buttonReleased=1                                     Accepted values 0 or 1.
enable_relayActivated=1                                     Accepted values 0 or 1.
enable_relayDeactivated=1                                   Accepted values 0 or 1.
enable_buttonHanging=1                                      Accepted values 0 or 1.
enable_soundTestSuccess=1                                   Accepted values 0 or 1.
enable_soundTestError=1                                     Accepted values 0 or 1.
enable_soundTestFailed=1                                    Accepted values 0 or 1.
[audio_files]
tftpserver=10.8.25.200
ringing=ringing.wav


[time]
enabled=1
hostname=10.8.25.200
time_region=America
time_zone=Denver

Amplifier Example

[sip]
sip_id=0203
sip_domain=10.5.11.75
nick_name=CCP03
auth_user=0203
auth_pwd=Ashley77
 
[call]
# Use 3 GPI as key matrix for DAK1-7
input_as_key_matrix=3
io_pin1=0
io_pin2=0
io_pin3=0
io_pin4=1
io_pin5=1
io_pin6=1
fast_blink_pattern=1011111
slow_blink_pattern=0000001000000
# handset w/offhook - normally closed
accessory=6
# Allow speech mode to be overriden
override_remote_ptt=1
# use DTMF 9 go to open duplex
open_duplex_dtmf=9
# Allow maximum audio output
poe_audio=1
# Use RFC2833 to send DTMF
dtmf_style=1
# Disable tones
tone_volume=-1
# Use PTT as default speech mode
speech_mode=1
# auto answer enabled
auto_answer_mode=1
# reduced mic sensitivity
mic_sensitivity=4
# onhook send dtmf 8 in call
onhook_in_call_function=2
onhook_dtmf_on=8
# Call 301
dak1_value=0401
dak1_in_call_function=0
dak2_value=401
dak2_in_call_function=0
dak3_value=501
dak3_in_call_function=0
dak4_value=203
dak4_in_call_function=0
dak5_value=510
dak5_in_call_function=0
dak6_value=502
dak6_in_call_function=0
 
[relays]
gpio3_dtmf_activate=2
gpio3_dtmf_deactivate=0
gpio3_dtmf_flashing_slow=1
gpio4_dtmf_activate=5
gpio4_dtmf_deactivate=3
gpio4_dtmf_flashing_slow=4
gpio5_dtmf_activate=7
gpio5_dtmf_deactivate=6
Example - Amplifier
[sip_ch1]
nick_name=amp1_ch1_marius
sip_id=0491
auth_user=0491
auth_pwd=Ashley77
sip_domain=10.5.11.75
playback_gain=-10
  
[sip_ch2]
nick_name=Amp1_ch2_marius
sip_id=0492
auth_user=0492
auth_pwd=Ashley77
sip_domain=10.5.11.75
playback_gain=-15