Cisco7940G kann nicht angerufen werden.

[Patrix2911]

Hallo. Nachdem ich nun endlich ein (ja ich weiss die Dinger sind alt), 7940G mit SIP Firmware an meiner FreePBX angemeldet bekommen habe, kann ich mit diesem zwar nach außen anrufen, aber wenn ich versuche nach drinnen auf dem Telefon anzurufen, bleibt dieses stumm.

Ich habe mal das Telefon ausgeschaltet und PhonerLite mit den Zugangsdaten des Telefons an der FreePBX angemeldet, und das hat auch sofort reagiert als ich angerufen habe. Es liegt wohl also am Telefon und nicht an der FreePBX. Ich füge mal meine SIPDefault.cnf und die Konfiguration des Telefons an, vielleicht sieht ja jemand von euch woran es liegt. Vielen Dank vorab für die Mühe und Hilfe....

# SIP Configuration Generic File

# Call Manager
call_manager1_addr: "192.168.1.117"
call_manager1_sip_port: 5060
call_manager2_addr: "192.168.1.117"
call_manager2_sip_port: 5060

# Line 1
proxy1_address: "USECALLMANAGER"
proxy1_port: 5060
line1_name: "80"
line1_authname: "80"
line1_password: "Passwort"
line1_displayname: "Schreibtisch"
line1_shortname: "St1"

# Line 2
proxy2_address: "USECALLMANAGER"
proxy2_port: 5060
line2_name: "80"
line2_authname: "80"
line2_password: "Passwort"
line2_displayname: "St--"
line2_shortname: "St2"

phone_label: "Schreibtisch"

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

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco"

;SIPDefault.cnf 

# SIP Default Generic Configuration File 

# Image Version 
image_version: P0S3-8-12-00 

# Proxy Server 
proxy1_address: "192.168.1.117" ; 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) 
proxy1_port: 5060 
proxy2_port: 5060 
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: 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: "192.53.103.104" ; SNTP Server IP Address 
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) 
time_zone: EAST ; 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: 0 ; Enable(1-Default)/Disable(0) DST automatic adjustment 
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) 
date_format : D/M/Y 

# 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: 2 ; 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: 32766 ; End RTP range for media (default - 32766) 
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled 

# Outbound Proxy Support 
outbound_proxy: "192.168.1.117" ; 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 ####### 
; 0-Disabled (default), 1-Enabled 

# XML URLs 
services_url: "" ; 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 
stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled 

#Voice Mail extention 
messages_uri: 8500 

# RTP Call Statistics (SIP BYE/200 OK message exchange) 
call_stats: 0 

#Transfer by hanging up the phone 
transfer_onhook_enabled:1