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

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*'''Description:''' This decides what input button 1 will do when in calls.
 
*'''Description:''' This decides what input button 1 will do when in calls.
  
*'''Values:''' Integer. 0 is end call. 1 is do nothing.
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*'''Values:''' Integer. 0 is end call. 1 is do nothing. 9 is push to talk.
  
 
===input2_value===
 
===input2_value===
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*'''Description:''' This decides what input button 1 will do when in calls.
 
*'''Description:''' This decides what input button 1 will do when in calls.
  
*'''Values:''' Integer. 0 is end call. 1 is do nothing.
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*'''Values:''' Integer. 0 is end call. 1 is do nothing. 9 is push to talk.
  
 
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===input3_value===
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*'''Description:''' This decides what input button 1 will do when in calls.
 
*'''Description:''' This decides what input button 1 will do when in calls.
  
*'''Values:''' Integer. 0 is end call. 1 is do nothing.
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*'''Values:''' Integer. 0 is end call. 1 is do nothing. 9 is push to talk.
  
 
===dak1_value===
 
===dak1_value===

Revision as of 12:37, 14 June 2019

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.

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

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 end call. 1 is do nothing. 9 is push to talk.

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 end call. 1 is do nothing. 9 is push to talk.

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 end call. 1 is do nothing. 9 is push to talk.

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

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_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

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.

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

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_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 tonetests
  • Values: Integer Value.

sound_pressure_level

  • Description: Time between tonetests
  • Values: Integer Value. Only odd values. 53 ≤ sound_pressure_level ≤ 97

volume

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

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

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