Hallo,
mein Cisco 7960 meldet beim laden der SIPDefault.cnf den Fehler W310
(2 Errors Parsing SIPDefault.cnf)
Das Gerät hat den Softwarestand SIP 08-8 und läuft an einer Trixbox...
Vielleicht sehe ich ja den Wald vor lauter Bäumen nicht mehr...
# Image Version
image_version: "P0S3-08-8-00"
# Proxy Server
proxy1_address: "192.168.0.76"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:""
proxy3_port:""
proxy4_port:""
proxy5_port:""
proxy6_port:""
# Emergency Proxy info
proxy_emergency: "192.168.0.76"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "192.168.0.76"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: "192.168.0.76"
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"
# 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: "2" ; 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: "0"
# 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: "*97"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "unicast"
sntp_server: "192.168.0.76"
time_zone: "CET"
dst_offset: "0"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
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: "syncinfo"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "0"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"
# URL for external Phone Services
services_url: "http://192.168.0.76/xmlservices/index.php"
# URL for external Directory location
directory_url: "http://192.168.0.76/xmlservices/PhoneDirectory.php"
# URL for branding logo
logo_url: "http://192.168.0.76/cisco/bmp/asterisk-tux.bmp"
Gruß
Farnsy
mein Cisco 7960 meldet beim laden der SIPDefault.cnf den Fehler W310
(2 Errors Parsing SIPDefault.cnf)
Das Gerät hat den Softwarestand SIP 08-8 und läuft an einer Trixbox...
Vielleicht sehe ich ja den Wald vor lauter Bäumen nicht mehr...
# Image Version
image_version: "P0S3-08-8-00"
# Proxy Server
proxy1_address: "192.168.0.76"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:""
proxy3_port:""
proxy4_port:""
proxy5_port:""
proxy6_port:""
# Emergency Proxy info
proxy_emergency: "192.168.0.76"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "192.168.0.76"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: "192.168.0.76"
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"
# 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: "2" ; 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: "0"
# 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: "*97"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "unicast"
sntp_server: "192.168.0.76"
time_zone: "CET"
dst_offset: "0"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
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: "syncinfo"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "0"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"
# URL for external Phone Services
services_url: "http://192.168.0.76/xmlservices/index.php"
# URL for external Directory location
directory_url: "http://192.168.0.76/xmlservices/PhoneDirectory.php"
# URL for branding logo
logo_url: "http://192.168.0.76/cisco/bmp/asterisk-tux.bmp"
Gruß
Farnsy
