Stentofon Turbine Ip Station (SDV3)

Description

This resource configures a Stentofon Turbine IP Station

Release notes

Resource configuration interface

Resource configuration parameters

Relays Settings

• Digit for Relay On : Dtmf to active the relay. Select one of the following values #,,0-9*
• Digit for Relay Off : Dtmf to deactive the relay. Select one of the following values #,,0-9*
• Digit for Timed Relay : Dtmf to activate the relay for X seconds. Select one of the following values #,,0-9*
• Timed relay duration : Duration (in seconds) to keep the relay active. The default value is 10.
• Outgoing ringing : Which state the relay should be set to during outgoing ringing.
  • On: the relay will be activated
  • Off: the relay will be deactivated
  • Slow flash: the relay will flash slowly
  • Fast flash: the relay will flash fast.
• Incoming ringing : Which state the relay should be set to during incoming ringing.
  • On: the relay will be activated
  • Off: the relay will be deactivated
  • Slow flash: the relay will flash slowly
  • Fast flash: the relay will flash fast.
• Outgoing call : Which state the relay should be set to during outgoing call.
  • On: the relay will be activated
  • Off: the relay will be deactivated
  • Slow flash: the relay will flash slowly
  • Fast flash: the relay will flash fast.
• Incoming call : Which state the relay should be set to during incoming call.
  • On: the relay will be activated
  • Off: the relay will be deactivated
  • Slow flash: the relay will flash slowly
  • Fast flash: the relay will flash fast.

Direct access key Settings

• Direct Access Type : Decides either to call a unique number (which can be a queue) when the call button is pressed or use the ringlist. The default behavior is to use the number.
• Number to Call : The number to be called when the call button is pressed. Set this if the Direct Access Type has been set to number.
• Ringlist Loop : Decides whether the ringlists should start at the beginning again when the ringlist has reached the end. This value is used when the Direct Access Type has been set to ringlist.
• Ringlist Max Ring Time (s) : This parameter sets the time (in seconds) to wait ringing until step to next value in the ringlist. This value is used when the Direct Access Type has been set to ringlist.
• Ringlist 1 - value [1 - 14] : This is the extension to be called when the call button is pressed and Ringlist is used, i.e. the telephone number of the receiving party. Next value on the Ringlist 1 is Ringlist 1 Value 2, Ringlist 1 Value 3 etc. These values are used when the Direct Access Type have been set to ringlist.
• Direct Access Key In call: This defines the action to perform when the call button is pressed. Default value is Do nothing.

Audio Settings

• Speaker Volume : This parameter sets the volume of the station's speaker. The default value is set to 5.
• Microphone Sensitivity : This parameter adjusts the microphone sensitivity. The default value is set to 5.
• Automatic Volume Control : Enable or Disable automatic volume control that is adjusted according to the noise level
• Noise Reduction Level : This parameter will be set to 2 by default. The higher the noise reduction level the more deterioration there is in audio quality.

Call Settings

• Auto answer : This parameter enable or disable the auto answer. The default value is Disable.
• Auto answer delay (s) (max 30 sec) : This parameter define the number of second after what the call will be automatically answered (when the auto answer is enabled). The maximum value is 30 seconds. The default value is 0.
• Max Ringing Time (s): This parameter sets the max time of the ringing/alerting phase of the call
• Max Conversation Time (s): Maximum number of seconds for a conversation

Firmware settings

• Auto update interval: Number of minutes between two check for firmware update on the TFTP server. If empty, no check is done

SIP server feature

These technical settings if improperly configured can have impact on the availability of the phone

• SIP re-invite: allow the voice media to be transferred directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
  • Default: Keep default settings as defined in the Telephony server module (for example Asterisk 1.2.x)
  • Yes: Allow the server to perform the re-invite procedure.
  • No: Do not allow the server to perform the re-invite procedure.
• Qualify SIP devices registration: In order to monitor the SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
  • Default: Keep the default setting defined in the Telephony server module (for example Asterisk 1.2.x)
  • No: Disable Qualify SIP registration for this device
  • Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
  • Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
• Force specific codec: If this parameter is set to 'Yes', the SIP server will only accept the 3 codecs defined in this resource with the specified preference.
• First Codec, Second Codec, Third Codec: Ordered list of codec

SNMP Settings

• Enable all traps: If set to yes, enables to read MIBS in SNMP v1 or v2
• Trap receiver: The IP address of the server receiving SNMP traps
• Inform receiver: The IP address of the server receiving SNMP informs

Extra Settings

• In this text area, raw configuration parameters can be provided. Please refers to Vendor's documentation for a list of existing parameters. The format must respect the ini file format with the section name as seen in the configuration example provided by the vendor. If a settings is already provided by another configuration of the resource, it will be ignored.