Hallo hab folgendes Prob mit dem 7960:
folegende Dateien sind da:
OS79XX.txt,POS3-08-6-00-loads,POS3-08-6-00.sb2,P003-08-6-00.bin,P003-08-6-00.sbn,dialplan.xml,Ringlist.dat
SIPMAC.cnf
meine SIPDefault.cnf
wieso zeigt mir das 7960 alle Lines an obwohl nur 2 eingetragen sind und der Rest auskommentiert ist.Egal von welchem TFTP Server ich versuche die Daten zu laden es bootet immer vom DSP.
folegende Dateien sind da:
OS79XX.txt,POS3-08-6-00-loads,POS3-08-6-00.sb2,P003-08-6-00.bin,P003-08-6-00.sbn,dialplan.xml,Ringlist.dat
SIPMAC.cnf
Code:
P Configuration Generic File
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Arbeitszimmer" ; Has no effect on SIP messaging
#### Line 1 - Sipgate
# Proxy Server
proxy1_address: "sipgate.de" ; Can be dotted IP or FQDN
# Proxy Server Port (default - 5060)
proxy1_port: 5060
# Line 1 appearance
line1_name: "xxxxxx"
# Line 1 Registration Authentication
line1_authname: "xxxxxxx"
# Line 1 Registration Password
line1_password: "xxxxxx"
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "xxxxxx"
# Line 1 Short Name (Name to be displayed on own display)
line1_shortname: "Sipgate"
#### Line 2 - Dus.net 2
proxy2_address: "voip.dus.net" ; Can be dotted IP or FQDN
proxy2_port: 3478
line2_name: "xxxxxxxxxxx"
line2_authname: "xxxxxxxxxxx"
line2_password: "xxxxxxxxxxx"
line2_displayname: "xxxxxxxxxx"
line2_shortname: "Dus.net"
#### Line 3 - Nikotel
#proxy3_address: "calamar0.nikotel.com"
#proxy3_port: 5060
#line3_name: "<NikotelUserID>"
#line3_authname: "<NikotelUserID>"
#line3_password: "........"
#line3_displayname: "R. Bisges"
#line3_shortname: "Nikotel"
#### Line 4 - pbx-network
#proxy4_address: "pbx-network.de"
#proxy4_port: 5060
#line4_name: "<pbxUserID>"
#line4_authname: "<pbxUserID>"
#line4_password: "........"
#line4_displayname: "R. Bisges"
#line4_shortname: "pbx-network"
#### Line 5 - Gmx
#proxy5_address: "sip-gmx.net"
#proxy5_port: 5060
#line5_name: "<FestnetzNr(49...)>"
#line5_authname: "<FestnetzNr(49...)>"
#line5_password: "........"
#line5_displayname: "R. Bisges"
#line5_shortname: "gmx"
#### Line 6 - PURtel
#proxy6_address: "deu1.purtel.com"
#proxy6_port: 5060
#line6_name: "<PURtelNr(5stellig)>"
#line6_authname: "<PURtelNr(5stellig)>"
#line6_password: "........"
#line6_displayname: "R. Bisges"
#line6_shortname: "PURtel"
###################################################
########## BeispielKonfigurationen ################
###################################################
# Line 5 - Free World DialUp
#proxy5_address: "fwd.pulver.com"
#proxy5_port: 5060
#line5_name: "<FWDNr(6stellig)>"
#line5_authname: "<FWDNr(6stellig)>"
#line5_password: "........"
#line5_displayname: "R. Bisges"
#line5_shortname: "FWD"
# Line 4 - blueSIP
#proxy4_address: "bluesip.net"
#proxy4_port: 5060
#line4_name: "<bluesipUserName>"
#line4_authname: "<bluesipAuthName(bluesip/xxxx)>"
#line4_password: "........"
#line4_displayname: "R. Bisges"
#line4_shortname: "blueSIP"
#### Line 4 - sipsnip
#
#proxy4_address: "sipsnip.com"
#proxy4_port: 5060
#line4_name: "<sipsnipUserID>"
#line4_authname: "<sipsnipUserID>"
#line4_password: "........"
#line4_displayname: "R. Bisges"
#line4_shortname: "sipsnip"
####### New Parameters added in Release 2.2 ######
# NAT/Firewall Traversal
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: 16388 ; End RTP range for media (default - 32766)
####### New Parameters added in Release 3.0 ######
# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "Cisco7960" ; Limited to 15 characters (Default - SIP Phone)
# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: phone
# Automatische Vervollstaendigung der getippten Rufnummer
# Gibt bei mir nur Chaos...
autocomplete: 0 ; 0-disable, 1-enable (default)
# URI fuer die Taste "messages"
messages_uri: 50000 ; Fuer Sipgatemeine SIPDefault.cnf
Code:
P Default Generic Configuration File
########################################
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
image_version: P003-08-6-00
# Proxy Server
proxy1_address: "sipgate.de" ; Can be dotted IP or FQDN
proxy2_address: "voip.dus.net " ; Can be dotted IP or FQDN
proxy3_address: "" ; Can be dotted IP or FQDN
proxy4_address: "" ; Can be dotted IP or FQDN
proxy5_address: "" ; Can be dotted IP or FQDN
proxy6_address: "" ; Can be dotted IP or FQDN
# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 3478
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 500
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5
# 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 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: "/opt/tftpboot/" ; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "ntp.sipgate.net" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
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)
date_format: D-M-YY ; Dateformat Day, month, year
# 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-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0 ; Default 0 (Disable 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
sync: 1 ; Default 1
####### New Parameters added in Release 2.1 #######
# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)
# Configurable VAD option
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: 16388 ; 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: 5060 ; 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: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
#services_url: "http://www.fo-pa.de/cgi-bin/rss2cisco.pl" ; URL for external Phone Services
#directory_url: "//niels/tftp-root/directory.xml" ; 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: 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: "" ; 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 (Message Waiting Indicator)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
stutter_msg_waiting: 1 ; 0-Disabled (default), 1-Enabled
# RTP Call Statistics (SIP BYE/200 OK message exchange)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
call_stats: 1 ; 0-Disabled (default), 1-Enabled
# Telefonnummern automatisch vervollstaendigen (macht bei mir Probleme, also aus)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
autocomplete: 0 ; 0-Disabled, 1-Enabled (default)wieso zeigt mir das 7960 alle Lines an obwohl nur 2 eingetragen sind und der Rest auskommentiert ist.Egal von welchem TFTP Server ich versuche die Daten zu laden es bootet immer vom DSP.
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