Problem mit 7960 (SIP)

[Torsten-]

Hi,

ich hab mein 7960 (SIP) das letzte mal August-Oktober eingesetzt und ende November an einem lokalen Asterisk - nun habe ich die Konfiguration von August-Oktober wieder eingespielt, aber es funktioniert nicht mehr - hat sich zwischendurch was geändert? Seht ihr einen Fehler in der Konfiguration?
Direkt nach dem Booten kann ich evtl kurz telefonieren, aber was mich stark wundert ist, dass ich nach einiger Zeit nichtmal mehr nach draußen Telefonieren kann .. Hab schon 2 Router ausprobiert (auch den von August-Oktober) - raus müsste ja eigentlich immer gehen ..

Bin bei Sipgate.

SIPDefault.cnf

# SIP Default Generic Configuration File
########################################
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
#image_version: P0S3-08-6-00

# 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: "none"

# TOS bits in media stream [0-5] (Default - 5)
#tos_media: 5

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 0

# 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

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101		# Default 101

# 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

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# 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: "ntp.sipgate.net"	# SNTP Server IP Address freenet
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: 1			# 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: 31			# 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 - 24Hrr)/Disable(0 - 12Hr)
date_format: D/M/Y		# 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)

# Sync value of the phone used for remote reset
sync: 1				# Default 1

# Configurable VAD option
enable_vad: 0			# VAD setting 0-disable (Default), 1-enable

# 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)

# 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 (CLIP/CLIR Rufnummeranzeige)
remote_party_id: 1		# 0-Disabled (default), 1-Enabled

# 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)

# 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: 0                   # 0-Disabled (default), 1-Enabled

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


# Backup Proxy Support
proxy_backup: "217.10.79.9"             # Dotted IP of Backup Proxy (sipgate.de)
proxy_backup_port: 5061         # Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "217.10.79.9"  # Dotted IP of Emergency Proxy (sipgate.de)
proxy_emergency_port: 5061      # Emergency Proxy port (default is 5060)

SIP<mac>.cnf

# SIP Configuration Generic File
phone_label: "Eddie "

#### Lines ####

line1_name: "line1" 
line1_displayname: "Torsten" 
line1_authname: "123xxxxe0" 
line1_password: "xxxx" 
line1_shortname: "0251-0"
proxy1_address: "sipgate.de"
proxy1_port: 5061

line2_name: "line2" 
line2_displayname: "Torsten"
line2_authname: "123xxxxe1" 
line2_password: "xxxx" 
line2_shortname: "0251-1"
proxy2_address: "sipgate.de"
proxy2_port: 5061

line3_name: "line3" 
line3_displayname: "Torsten" 
line3_authname: "123xxxxe2" 
line3_password: "xxxx" 
line3_shortname: "0251-2"
proxy3_address: "sipgate.de"
proxy3_port: 5061

line4_name: "" 
line4_displayname: "" 
line4_authname: "" 
line4_password: "" 
line4_shortname: ""
proxy4_address: ""
proxy4_port: 5061

line5_name: "" 
line5_displayname: "" 
line5_authname: "" 
line5_password: "" 
line5_shortname: ""
proxy5_address: ""
proxy5_port: 5061

line6_name: "" 
line6_displayname: "" 
line6_authname: "" 
line6_password: ""
line6_shortname: ""
proxy6_address: ""
proxy6_port: 5061

# Backup Proxy Support
proxy_backup: "217.10.79.9"		# Dotted IP of Backup Proxy (sipgate.de)
proxy_backup_port: 5061		# Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "217.10.79.9"	# Dotted IP of Emergency Proxy (sipgate.de)
proxy_emergency_port: 5061	# Emergency Proxy port (default is 5060)

# Outbound Proxy info
outbound_proxy: sipgate.de
outbound_proxy_port: 5061

# NAT/Firewall Traversal
nat_enable: 1         		          # 0-Disabled (default), 1-Enabled
nat_received_processing: 1		  # 0-Disabled (default), 1-Enabled
nat_address: "stun.sipgate.net:10000"	  # WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5061      		  # UDP port used for SIP messages (default - 5060)
start_media_port: 16384	 		  # Start RTP range for media (default - 16384)
end_media_port: 16389	   		  # End RTP range for media (default - 32766)

# Phone prompt/password for telnet/console session
phone_prompt: "Eddie" # Telnet/Console Prompt
phone_password: "xxxx" # Telnet/Console Password

# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 0			# 0-Disabled (default), 1-Enabled, 2-Privileged

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none

# Auto-generated by marvin.amshove.net
services_url: "http://xxxx.xml"
directory_url: "http://xxxx.xml"
logo_url: "http://xxxx.bmp"