schön wärs ja gewesen
im Ernst, ich kann keine Nr. wählen, ohne entweder den Hörer abzunehmen bzw. das Freisprechen zu aktivieren.
Ich poste hier mal die beiden Konf. Inhalte, vielleicht ist da ja auch noch ein Fehler den ich nicht finden konnte.
SIPDefault:
# SIP Default Generic Configuration File
# Registrar FRITZ!Box 6360
# Image Version
#image_version: P0S3-8-12-00 Auskommentiert da SIP Firmware schon geladen ist
dial_template: dailplan.xml
# Proxy Server Nichtbenötigte auskommentiert
proxy1_address: "192.168.178.1" ; Can be dotted IP or FQDN
# proxy2_address: "" ; 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) Nichtbenötigte auskommentiert
proxy1_port: 5060
# proxy2_port: 5061
# 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: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: "g711alaw"
# 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: "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 10
sip_invite_retx: 6 ; Default 6
timer_invite_expires: 180 ; Default 180 sec
####### New Parameters added in Release 2.0 #######
# template (.xml format file relative to the TFTP root directory)
# dial_template: dialplan Auskommentiert, da keine dialplan.xml da ist
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./" ; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.168.178.1" ; 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)
# 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: "192.168.178.1" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "192.168.178.1" ; 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: 0 ; 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: "192.168.178.1" ; 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: 0 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "http://webserver/cisco7960/cisco7960services.xml" ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
#logo_url: "http://www.gbergemann.de/logo.cip" ; URL for image
# 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: "192.168.178.1" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "<IP des TFTP-Servers>" ; 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
# Setting for Message speeddial to UOne box
# Belegung mit der FritzBox Voice Mail Nummer
#messages_uri: "**600"
SIP<MAC>
# SIP Configuration Generic File
# Line 1
proxy1_address: "192.168.178.1"
proxy1_port: 5061
line1_name: "620"
line1_authname: "620"
line1_password: "1111"
line1_displayname: "620"
line1_shortname: "620"
# Line 2 appearance
#line2_name: "Privat"
# Line 2 Registration Authentication
#line2_authname: "621"
# Line 2 Registration Password
#line2_password: "1111"
# Line 2
#proxy2_address: "192.168.178.1"
#proxy2_port: 5061
#line2_name: "621"
#line2_authname: "621"
#line2_password: "1111"
#line2_displayname: "621"
#line2_shortname: "621"
####### New Parameters added in Release 2.0 #######
# All user_parameters have been removed
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: ""
# Line 2 Display Name (Display name to use for SIP messaging)
#line2_displayname: ""
####### New Parameters added in Release 3.0 ######
# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SIP Phone" ; 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: none
