 | 1 | изначальная версия редактировать | |
в конфигурации циско телефона(которую он грузит по tftp) поставьте нат.с клавиатуры телефона нат в 1 вам никак не получиться поставить, только по tftp.
вот полный рабочий конфиг для 7940
SIPDefault.conf
# Image Version
image_version: "P0S3-8-12-00"
# Proxy Server
# Note: I put the proxy server information in the individual conf files
# for each machine, since each box has different configs. You could, of course,
# put all of them here in the Default file...
proxy1_address: "78.46.118.22"
preferred_codec:g711alaw
proxy1_port:"5060"
# Proxy Server Port (default - 5061)
# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: ""
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: ""
# change to different port for every phone
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "1"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "120"
# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "none"
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"
# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "1" ; 0-Disabled (default), 1-Enabled, 2-Privileged
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"
# 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 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "*98"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "17.254.0.49"
time_zone: "EEST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: "1"
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)
# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)
# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100
# XML file that specifies the dialplan desired
dial_template: "dialplan"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"
####### New Parameters added in Release 4.0 #######
# XML URLs
#services_url: "http://192.168.1.145/menu.pl" ; URL for external Phone Services
#services_url: "http://192.168.1.254/cgi-bin/cisco7960/menu.xml" ; test url
#directory_url: "http://192.168.7.101/voip_xml_utils/index.cfm?fuseaction=dspEmployeeMenu"
# URL for external Directory location
logo_url: "http://78.46.118.22/asterisk-tux.bmp"
# put your own logo in the logo_url location; I include the 10-20.com one for reference in building your own
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; 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: "192.168.3.1" ; restricted to dotted IP
# The dynamic tftp server should be set to whatever your TFTP server is. This way, it
# keeps the tftp server setting even though you might be using DHCP (default behavior
# is to use the DHCP server as a tftp server, which is rarely correct.)
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
SIP000XXXXXXXXX.conf(заменить на адрес телефона)
# Line 1 Settings
line1_name: "12" ; Line 1 Extension\User ID
line1_displayname: "12" ; Line 1 Display Name
line1_shortname: "12" ; Comment next to the button
line1_authname: "12" ; Line 1 Registration Authentication
line1_password: "111111" ; Line 1 Registration Password
| | 2 | No.2 Revision редактировать |
в конфигурации циско телефона(которую он грузит по tftp) поставьте нат.с клавиатуры телефона нат в 1 вам никак не получиться поставить, только по tftp.
вот полный рабочий конфиг для 7940
SIPDefault.confSIPDefault.cnf
# Image Version
image_version: "P0S3-8-12-00"
# Proxy Server
# Note: I put the proxy server information in the individual conf files
# for each machine, since each box has different configs. You could, of course,
# put all of them here in the Default file...
proxy1_address: "78.46.118.22"
preferred_codec:g711alaw
proxy1_port:"5060"
# Proxy Server Port (default - 5061)
# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: ""
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: ""
# change to different port for every phone
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "1"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "120"
# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "none"
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"
# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "1" ; 0-Disabled (default), 1-Enabled, 2-Privileged
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"
# 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 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "*98"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "17.254.0.49"
time_zone: "EEST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: "1"
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)
# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)
# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100
# XML file that specifies the dialplan desired
dial_template: "dialplan"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"
####### New Parameters added in Release 4.0 #######
# XML URLs
#services_url: "http://192.168.1.145/menu.pl" ; URL for external Phone Services
#services_url: "http://192.168.1.254/cgi-bin/cisco7960/menu.xml" ; test url
#directory_url: "http://192.168.7.101/voip_xml_utils/index.cfm?fuseaction=dspEmployeeMenu"
# URL for external Directory location
logo_url: "http://78.46.118.22/asterisk-tux.bmp"
# put your own logo in the logo_url location; I include the 10-20.com one for reference in building your own
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; 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: "192.168.3.1" ; restricted to dotted IP
# The dynamic tftp server should be set to whatever your TFTP server is. This way, it
# keeps the tftp server setting even though you might be using DHCP (default behavior
# is to use the DHCP server as a tftp server, which is rarely correct.)
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
SIP000XXXXXXXXX.conf(заменить SIP000XXXXXXXXX.cnf(заменить на адрес телефона)
# Line 1 Settings
line1_name: "12" ; Line 1 Extension\User ID
line1_displayname: "12" ; Line 1 Display Name
line1_shortname: "12" ; Comment next to the button
line1_authname: "12" ; Line 1 Registration Authentication
line1_password: "111111" ; Line 1 Registration Password
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.