CISCO7960 und T-Online: Raustelefonieren geht nicht

[Luckman]

Hallo zusammen,

bis jetzt habe ich hier im Forum nur fleißig mitgelesen, habe jetzt allerdings ein Problem, bei dem ich seit zwei Wochen nicht mehr weiterkomme:

Ich habe eine neues CISCO-Phone (7960 mit SIP) und zwei VoIP-Providern: T-Online und Sipgate. Sipgate läuft astrein, T-Online hat ein Problem:
Anrufe kommen rein, werden signalisiert und ich kann wunderbar telefonieren; Raustelefonieren funktioniert allerdings garnicht: ich bekomme kurz nachdem ich die Rautetaste gedrückt habe einen "Besetzt-Ton"! Ich finde es äußerst komisch!

Vielleicht habt Ihr eine Idee.

Hier meine SIPdefault:

# SIP Default Generic Configuration File 
######################################## 
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
;image_version: P0S3-08-2-00

# Proxy Server
proxy1_address: "tel.t-online.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: 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: ""			; 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: 5061      	; UDP port used for SIP messages (default - 5060)
start_media_port: 7082	 	; Start RTP range for media (default - 16384)
end_media_port: 7086	   	; 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.voip-info.de/feeds/cisco7960/index.xml"; 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: 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)

und hier meine MAC.cnf:

# SIP Configuration Generic File 

phone_label: "CISCO 7960" 
 
######################################### Line 1 - T-Online


# Line 1 appearance
line1_name: "03222767xxxx"   ; ohne Leerzeichen/Sonderzeichen!
# Line 1 Registration Authentication 
line1_authname: "[email protected]"  ; Kleinbuchstaben z.B. "[email protected]"!
# Line 1 Registration Password
line1_password: "xxxx"
# Line 1 Display Name (Name, der beim Angerufenen angezeigt werden soll)
line1_displayname: "03222767xxxx"
# Line 1 Short Name (Name, der fuer die Line-Taste benutzt werden soll)
line1_shortname: "T-Online"

######################################### Line 1 - T-Online


# Line 2 appearance
line2_name: "800xxxx"   ; ohne Leerzeichen/Sonderzeichen!
# Line 2 Registration Authentication 
line2_authname: "800xxxx"  ; Kleinbuchstaben z.B. "[email protected]"!
# Line 2 Registration Password
line2_password: "xxxx"
# Line 2 Display Name (Name, der beim Angerufenen angezeigt werden soll)
line2_displayname: "xxxx"
# Line 2 Short Name (Name, der fuer die Line-Taste benutzt werden soll)
line2_shortname: "Sipgate"



####### 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: "Cisco 7960 "	; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "User ID"





####### 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

Folgendes zu meinem Aufbau:

FritzboxFon(LAN:192.168.178.1)---(WAN:192.168.178.2)Linksys(LAN:192.168.1.1)---CISCO7960(192.168.1.150)

Die Ports 5060 und 7078-7081 werden von der Fritzbox benötigt (dort hängen analoge Telefone) und ich kann sie auch nicht weiterleiten, deswegen nutze ich die folgenden Ports für das CISCO-Phone:

voip_control_port: 5061
start_media_port: 7082
end_media_port: 7086

Die Ports sind in der Fritzbox auf meinen Linksys weitergeleitet und von dort zum Telefon.

Meiner Meinung nach müsste es funktionieren - tut es aber nicht :noidea:

Brauche Hilfe. Werde langsam wahnsinnig!

Gruß und danke

Luckman