Telefonní systém kosmické lodi Hipporion ze SKSP2019
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# SIP Default Generic Configuration File
# Image Version
image_version: P0S3-8-12-00
# Proxy Server (Can be dotted IP or FQDN)
proxy1_address: ""
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""
# Proxy Server Port (default - 5060)
proxy1_port: ""
proxy2_port: ""
proxy3_port: ""
proxy4_port: ""
proxy5_port: ""
proxy6_port: ""
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711alaw
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 0
# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always-always avt)
dtmf_outofband: avt
# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3
# SIP Timers
timer_t1: 500; Default 500 msec
timer_t2: 4000; Default 4 sec
sip_retx: 10; Default 10
sip_invite_retx: 6; Default 6
timer_invite_expires: 180; Default 180 sec
####### New Parameters added in Release 2.0 #######
# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: Dialplan
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: Phones/; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: ""; SNTP Server IP Address
sntp_mode: unicast; anycast (default), unicast, multicast, or directedbroadcast
time_zone: CET; Time Zone Phone is in
dst_offset: 1; Offset from Phone's time when DST is in effect
dst_start_month: April; Month in which DST starts
dst_start_day: ""; Day of month in which DST starts
dst_start_day_of_week: Sun; Day of week in which DST starts
dst_start_week_of_month: 1; Week of month in which DST starts
dst_start_time: 02; Time of day in which DST starts
dst_stop_month: Oct; Month in which DST stops
dst_stop_day: ""; Day of month in which DST stops
dst_stop_day_of_week: Sunday; Day of week in which DST stops
dst_stop_week_of_month: 8; Week of month in which DST stops 8=last week of month
dst_stop_time: 2; Time of day in which DST stops
dst_auto_adjust: 1; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
# Do Not Disturb Control (0-off (default), 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 2
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 0; (Default is 0 - disabled and sending all calls as anonymous)
# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0; (Default is 0 - disabled and blocking of anonymous calls)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101; Default 101
# Sync value of the phone used for remote reset
# Value against which to compare the value in the syncinfo.xml file before a remote reboot is performed
sync: 1; Default 1
####### New Parameters added in Release 2.1 #######
# Backup Proxy Support
proxy_backup: 0.0.0.0; Dotted IP of Backup Proxy
proxy_backup_port: ""; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: 0.0.0.0; Dotted IP of Emergency Proxy
proxy_emergency_port: ""; Emergency Proxy port (default is 5060)
# Configurable VAD option (voice activation detection)
enable_vad: 0; VAD setting 0-disable (Default), 1-enable
####### New Parameters added in Release 2.2 ######
# NAT/Firewall Traversal
nat_enable: 1; 0-Disabled (default), 1-Enabled
nat_address: ""; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060; UDP port used for SIP messages (default - 5060)
start_media_port: 16384; Start RTP range for media (default - 16384)
end_media_port: 32766; End RTP range for media (default - 32766)
nat_received_processing: 1; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: ""; restricted to dotted IP or DNS A record only
outbound_proxy_port: ""; default is 5060
####### New Parameter added in Release 3.0 #######
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: 1; 0-Disabled, 1-Enabled (default)
####### New Parameters added in Release 3.1 #######
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to Telnet into the phone)
telnet_level: 2; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: ""; URL for external Phone Services
directory_url: ""; URL for external Directory location
logo_url: ""; URL for branding logo to be used on phone display
# HTTP Proxy Support
http_proxy_addr: ""; Address of HTTP Proxy server
http_proxy_port: ""; Port of HTTP Proxy Server (80-default)
# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""; restricted to dotted IP
dyn_dns_addr_2: ""; restricted to dotted IP
dyn_tftp_addr: ""; restricted to dotted IP
# Remote Party ID
remote_party_id: 0; 0-Disabled (default), 1-Enabled
####### New Parameters added in Release 4.4 #######
# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0; Default 0 (Call Hold Ringback feature is off)
####### New Parameters added in Release 6.0 #######
# Dialtone Stutter for MWI
stutter_msg_waiting: 1; 0-Disabled (default), 1-Enabled
# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 1; 0-Disabled (default), 1-Enabled
####### Other parameters #######
date_format: D/M/Y; Default is M/D/Y
dscpForAudio: 184; Differentiated Services Code Point (DSCP) specifies the class of service for each audio packet
garp_enable: 0; Gratuitous ARP
language: english; English is the only value that is currently supported
local_cfwd_enable: 1; Whether the phone can do local call forwarding
messages_uri: ""; Configures the voice-mail number that is dialed when the messages button is pressed
network_media_type: Auto; Auto (default), Full100, Half100, Full10, Half10
network_port2_type: Hub/Switch; Hub/Switch (default), PC (Specifying the PC option and then connecting port 2 to a switch results in spanning-tree loops and network confusion)
rfc_2543_hold: 0; 0 (default)
sip_max_forwards: 70; The phone uses the value specified in this parameter in the Max-Forwards header of the SIP requests that it generates
timer_register_delta: 5; Configures the time interval at which reregistration will occur
####### Phone specific ######
autocomplete: 1; Configures automatic completion of numbers
call_waiting: 1;
phone_label: ""; Text desired to be displayed in upper right corner
#phone_password: cisco; Password to be used for console or telnet login (default: cisco)
phone_prompt: "SIP Phone"; The prompt that will be displayed on console and telnet (default: "SIP Phone")
user_info: none; User classification used when Registering [none (default), phone, ip]
line1_authname: ""
line2_authname: ""
line3_authname: ""
line4_authname: ""
line5_authname: ""
line6_authname: ""
line1_contact: ""
line2_contact: ""
line3_contact: ""
line4_contact: ""
line5_contact: ""
line6_contact: ""
line1_displayname: ""
line2_displayname: ""
line3_displayname: ""
line4_displayname: ""
line5_displayname: ""
line6_displayname: ""
line1_name: ""
line2_name: ""
line3_name: ""
line4_name: ""
line5_name: ""
line6_name: ""
line1_password: ""
line2_password: ""
line3_password: ""
line4_password: ""
line5_password: ""
line6_password: ""
line1_shortname: ""
line2_shortname: ""
line3_shortname: ""
line4_shortname: ""
line5_shortname: ""
line6_shortname: ""
call_manager1_addr: ""
call_manager2_addr: ""
call_manager3_addr: ""
call_manager4_addr: ""
call_manager5_addr: ""
call_manager1_sip_port: ""
call_manager2_sip_port: ""
call_manager3_sip_port: ""
call_manager4_sip_port: ""
call_manager5_sip_port: ""