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Configuration File Parameters for SIP Provisioning

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Revision as of 15:38, 5 January 2017 by Asle (talk) (Call Parameters)

This article is applicable to all IP Stations (Turbine and INCA stations) operating in SIP mode.


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 both files, the value from the device specific file takes precedence.

General Parameters

auto_update_interval
  • Required: No. If this parameter is 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.

SIP Parameters

nick_name
  • Required: No. Defaults to sip_id.
  • Description: The nickname for the station can be used to assign a logical name to the station. For example, a station belonging to James may be assigned the nickname “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 of 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 fallback) 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 fallback) domain. If the station loses connection to the primary SIP domain, it will switch over to the secondary domain.
  • Values: IP address given in regular dot notation, e.g. 10.5.2.100


sip_domain3
  • Required: No
  • Description: This is the tertiary (or fallback) domain. If the station loses connection to the secondary SIP domain, it will switch over to the tertiary domain.
  • 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 is 5060.
  • Description: The UDP port on the SIP 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

Call Parameters

ringlist_loop
  • Description: Decides whether the ringlists should start at the beginning again when the ringlist has reached the end.
  • Values: Integer value, 1 is on, 0 is off


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


ringlist_max_ring_time
  • Required: No. Defaults to 0
  • 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


ringlist1_value1
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed and ringlist is used, i.e. the telephone number of the receiving party. Next value on the Ringlist 1 is ringlist1_value2, ringlist1_value3 etc.
  • Values: String value


ringlist2_value1
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed and ringlist is used, i.e. the telephone number of the receiving party. Next value on the Ringlist 2 is ringlist2_value2, ringlist2_value3 etc.
  • Values: String value


ringlist3_value1
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed and ringlist is used, i.e. the telephone number of the receiving party. Next value on the Ringlist 3 is ringlist3_value2, ringlist3_value3 etc.
  • Values: String value


input1_value
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed is used, i.e. the telephone number of the receiving party.
  • Values: String value


input1_option
  • Required: Yes
  • Description: Decides which ringlist the button use.
  • Values: Integer value. -1 = no ringlist, 0 = ringlist 1, 1 = ringlist 2, 2 = ringlist 3


input2_value
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed is used, i.e. the telephone number of the receiving party.
  • Values: String value


input2_option
  • Required: Yes
  • Description: Decides which ringlist the button use.
  • Values: Integer value. -1 = no ringlist, 0 = ringlist 1, 1 = ringlist 2, 2 = ringlist 3


input3_value
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed is used, i.e. the telephone number of the receiving party.
  • Values: String value


input3_option
  • Required: Yes
  • Description: Decides which ringlist the button use.
  • Values: Integer value. -1 = no ringlist, 0 = ringlist 1, 1 = ringlist 2, 2 = ringlist 3


dak1_value
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed is used, i.e. the telephone number of the receiving party.
  • Values: String value


dak1_option
  • Required: Yes
  • Description: Decides which ringlist the button use.
  • Values: Integer value. -1 = no ringlist, 0 = ringlist 1, 1 = ringlist 2, 2 = ringlist 3


dak2_value
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed is used, i.e. the telephone number of the receiving party.
  • Values: String value


dak2_option
  • Required: Yes
  • Description: Decides which ringlist the button use.
  • Values: Integer value. -1 = no ringlist, 0 = ringlist 1, 1 = ringlist 2, 2 = ringlist 3


dak3_value
  • Required: Yes
  • Description: This is the SIP ID for the extension to be called when the first call button is pressed is used, i.e. the telephone number of the receiving party.
  • Values: String value


dak3_option
  • Required: Yes
  • Description: Decides which ringlist the button use.
  • Values: Integer value. -1 = no ringlist, 0 = ringlist 1, 1 = ringlist 2, 2 = ringlist 3


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


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 from remote station. Send DTMF * to talk and # to listen.
  • Values: Integer. 0 = disabled, 1 = enabled.


auto_answer_mode
  • Required: No.
  • Description: Enables auto-answer 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 auto-answer.
  • Values: Integer. 0 ≤ delay ≤ 30

Relay Parameters

SNMP Parameters

Example Configuration File