Difference between revisions of "Configuration File Parameters for SIP Provisioning"
From Zenitel Wiki
(added conversation and ptt modes) (Tag: Visual edit) |
(updated relay parameters) (Tag: Visual edit) |
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*'''Values:''' Integer. 0 disabled, 1 enabled. | *'''Values:''' Integer. 0 disabled, 1 enabled. | ||
− | === speech_mode === | + | ===speech_mode=== |
− | * '''Required:''' No. | + | *'''Required:''' No. |
− | * '''Description:''' Set the conversation mode. | + | *'''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 = Full Open Duplex, 1 = Push To Talk, 2 = Half Duplex Switching, 3 = Open, 4 = Robust Duplex. |
− | === ptt_mode === | + | ===ptt_mode=== |
− | * '''Required:''' No. | + | *'''Required:''' No. |
− | * '''Description:''' Set the PTT mode, only active if Conversation Mode is set to PTT. | + | *'''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. | + | *'''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=== | ===auto_answer_mode=== | ||
Line 557: | Line 557: | ||
==Relay Parameters== | ==Relay Parameters== | ||
+ | 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=== | ===relay1_dtmf_activate=== | ||
Line 623: | Line 630: | ||
*'''Values:''' Integer. 0 = do nothing, 1 = activate, 2 = deactivate, 3 = flash slow, 4 = flash fast | *'''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=== | ===relay1_event_idle=== |
Revision as of 11:51, 11 July 2019
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
- 1 Remote Provisioning using TFTP
- 2 General Parameters
- 3 SIP Parameters
- 4 Call Parameters
- 4.1 input1_value
- 4.2 input1_in_call_function
- 4.3 input2_value
- 4.4 input2_in_call_function
- 4.5 input3_value
- 4.6 input3_in_call_function
- 4.7 input4_value
- 4.8 input4_in_call_function
- 4.9 input5_value
- 4.10 input5_in_call_function
- 4.11 input6_value
- 4.12 input6_in_call_function
- 4.13 dak1_value
- 4.14 dak1_in_call_function
- 4.15 dak2_value
- 4.16 dak2_in_call_function
- 4.17 dak3_value
- 4.18 dak3_in_call_function
- 4.19 offhook_value
- 4.20 offhook_in_call_function
- 4.21 onhook_value
- 4.22 onhook_in_call_function
- 4.23 ptt_value
- 4.24 ptt_in_call_function
- 4.25 ringlist1_value1
- 4.26 ringlist2_value1
- 4.27 ringlist3_value1
- 4.28 ringlist_max_ring_time
- 4.29 ringlist_loop
- 4.30 ringlist_max_conv_time
- 4.31 conv_time
- 4.32 ring_time
- 4.33 poe_audio (full audio output)
- 4.34 speaker_volume
- 4.35 mic_sensitivity
- 4.36 noise_reduction
- 4.37 echo_parameter
- 4.38 rtp_timeout
- 4.39 remote_controlled_volume_override_mode
- 4.40 speech_mode
- 4.41 ptt_mode
- 4.42 auto_answer_mode
- 4.43 auto_answer_delay
- 4.44 accessory
- 4.45 input_as_key_matrix
- 4.46 fast_blink_pattern
- 4.47 slow_blink_pattern
- 4.48 open_duplex_dtmf
- 4.49 override_remote_ptt
- 4.50 poe_audio
- 4.51 dtmf_style
- 4.52 tone_volume
- 5 Relay Parameters
- 5.1 relay1_dtmf_activate
- 5.2 relay1_dtmf_deactivate
- 5.3 relay1_dtmf_flashing_slow
- 5.4 relay1_dtmf_flashing_fast
- 5.5 relay1_dtmf_toggle
- 5.6 relay1_dtmf_timed_relay
- 5.7 relay1_dtmf_timed_relay_duration
- 5.8 relay1_event_out_ringing
- 5.9 relay1_event_inc_ringing
- 5.10 relay1_event_inc_call
- 5.11 relay1_event_out_call
- 5.12 relay1_event_group_call
- 5.13 relay1_event_idle
- 5.14 relay1_event_error
- 6 Tone Parameters
- 7 SNMP Parameters
- 7.1 trap_receiver
- 7.2 inform_receiver
- 7.3 network
- 7.4 network_mask
- 7.5 community
- 7.6 enable_v1
- 7.7 enable_v2c
- 7.8 enable_ipsStarted
- 7.9 enable_sipRegistered
- 7.10 enable_sipRegisterFailed
- 7.11 enable_callConnect
- 7.12 enable_callConnectFailed
- 7.13 enable_callDisconnect
- 7.14 enable_buttonPressed
- 7.15 enable_buttonReleased
- 7.16 enable_dakPressed
- 7.17 enable_dakReleased
- 7.18 enable_relayActivated
- 7.19 enable_relayDeactivated
- 7.20 enable_buttonHanging
- 7.21 enable_soundTestSuccess
- 7.22 enable_soundTestError
- 7.23 enable_soundTestFailed
- 8 Examples
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.
If the same parameter is found in several files, then the precedence is as following:
- .MAC address file
- IP address file
- Global file
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
turbine_frontboard
- Description: Configures the turbine frontboard type
- Values:
- 0 = KIT
- 1 = TKIE-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
- 51 = TCIV-2, TCIV-3
- 52 = TCIV-6
- 53 = MINI
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
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
- 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
Call Parameters
input1_value
- 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.
- Values: String value
input1_in_call_function
- Description: This decides what input button 1 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
input2_value
- Required: Yes
- 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.
- Values: String value
input2_in_call_function
- Description: This decides what input button 1 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
input3_value
- 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.
- Values: String value
input3_in_call_function
- Description: This decides what input button 1 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
input4_value
- Required: Yes
- Description: This is the SIP ID for the extension to be called when the fourth input button is pressed, i.e. the telephone number of the receiving party.
- Values: String value
input4_in_call_function
- Description: This decides what input button 4 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
input5_value
- Required: Yes
- Description: This is the SIP ID for the extension to be called when the fifth input button is pressed, i.e. the telephone number of the receiving party.
- Values: String value
input5_in_call_function
- Description: This decides what input button 5 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
input6_value
- Required: Yes
- Description: This is the SIP ID for the extension to be called when the sixth input button is pressed, i.e. the telephone number of the receiving party.
- Values: String value
input6_in_call_function
- Description: This decides what input button 6 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 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_value
- Required: No
- 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.
- Values: String value
dak1_in_call_function
- Description: This decides what dak button 1 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
dak2_value
- 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.
- Values: String value
dak2_in_call_function
- Description: This decides what dak button 1 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
dak3_value
- Required: No
- 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.
- Values: String value
dak3_in_call_function
- Description: This decides what dak button 3 will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
offhook_value
- Required: No
- Description: This is the SIP ID for the extension to be called when the offhook button is pressed, i.e. the telephone number of the receiving party.
- Values: String value
offhook_in_call_function
- Description: This decides what offhook button will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
onhook_value
- Required: No
- Description: This is the SIP ID for the extension to be called when the offhook button is pressed, i.e. the telephone number of the receiving party.
- Values: String value
onhook_in_call_function
- Description: This decides what offhook button will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
ptt_value
- Required: No
- Description: This is the SIP ID for the extension to be called when the ptt button is pressed, i.e. the telephone number of the receiving party.
- Values: String value
ptt_in_call_function
- Description: This decides what ptt button will do when in calls.
- Values: Integer. 0 is answer/end call.1 is do nothing. 5 is end call. 6 is answer call. 7 is park call. 9 is push to talk. 10 is hold call. 11 is defer.
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
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
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
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
- Description: This parameter sets the max time of the conversation
ring_time
- Description: This parameter sets the max time of the ringing/alerting phase of the call
poe_audio (full audio output)
- 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
- 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
poe_audio
- Required: No. Defaults to 0
- Description: Enables full audio output. In case of PoE switch then the station might reboot if too much power is used.
- 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
Relay Parameters
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
Tone Parameters
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
- Description: Minimum sound pressure level between silence and tone
- Values: Integer Value. Only odd values. 53 ≤ sound_pressure_level ≤ 97
volume
- Description: Volume of the tone test
- Values: Integer Value. 0 ≤ volume ≤ 7
auto_set_sound_pressure_level
- 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
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 DAKbutton 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
Examples
Example Configuration File
[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 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.
Example 2
[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