7940 + Freenet + Sipgate kann nicht rausrufen

 # SIP Default Generic Configuration File
########################################
# Image Version
image_version: P0S3-07-4-00 

# Proxy Server
proxy1_address: "freenet.de"      ; Can be dotted IP or FQDN
proxy2_address: "sipgate.de"      ; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: "5060"
proxy2_port: "5060"

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires:"60"

# 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: "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: "4"          ; 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: 

# 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: "130.149.17.21"         ; SNTP Server IP Address
sntp_mode: unicast                ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: GMT                ; 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: 1      ; 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


####### 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: "16390"         ; 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: 2         ; 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: ""      ; 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: 1      ; 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: 1      ; Default 0 (Call Hold Ringback feature is off)

####### New Parameters added in Release 6.0 #######

# Dialtone Stutter for MWI (Message Waiting Indicator)
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

# Telefonnummern automatisch vervollstaendigen (macht bei mir Probleme, also aus)
autocomplete: 0                 ; 0-Disabled, 1-Enabled (default)

 # SIP Configuration Generic File

 
######################################### Line 1 - Freenet

line1_name: "User"
line1_authname: "User"
line1_password: "Passwort"
line1_displayname: ""
line1_shortname: "Freenet"


######################################### Line 2 - 

line2_name: "SIP-ID"
line2_authname: "SIP-ID"
line2_password: "SIP-Passwort"
line2_displayname: ""
line2_shortname: "SipGate"


# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Sniper Phone"   ; Has no effect on SIP messaging

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SniperPhone"      ; 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"