Cisco 7962G IP Telefon lässt sich nicht mit FritzBox 7530 verbinden

<Default>
<autoRegistrationName>AUTO-REG</autoRegistrationName>
<autoRegistration>disabled</autoRegistration>
<selfProvisioningSecureMode>true</selfProvisioningSecureMode>
<adminProvisionMode>false</adminProvisionMode>
<ipAddressMode>0</ipAddressMode>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<ipMediaAddressFamilyPreference>0</ipMediaAddressFamilyPreference>
<callManagerGroup>
<name>Default</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0"> 
<callManager>
<name>192.168.178.1</name>
<description>FritzBox</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>192.168.178.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<TVS>
<members>
<member priority="0"> 
<port>2445</port>
<address>192.168.178.1</address>
</member>
</members>
</TVS>
<loadInformation  model="Cisco 7962">sip42.9-4-2SR3-1S</loadInformation> 

</Default> 

# SIP Default Generic Configuration File 
 
# Image Version
image_version: P0S3-07-5-00

# Proxy Server
proxy1_address: "Fritz.Box"		; 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: 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: "192.168.178.1"			; SNTP Server IP Address (ntp0.fau.de)  
sntp_mode: unicast	; 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: March		; 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: 3	; 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: 3		; 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-disabled, 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: 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: ""	 	; 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: ""		; URL for external Phone Services  
directory_url: ""		; URL for external Directory location  
logo_url: "http://192.168.1.1/meinlogo.bmp"			; 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

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0			; 0-Disabled (default), 1-Enabled

<device>
  <deviceProtocol>SIP</deviceProtocol>
  <sshUserId>Schreibtisch</sshUserId>
  <sshPassword>(mein Password)</sshPassword>
  <devicePool>
                <dateTimeSetting>
                        <dateTemplate>D.M.YY</dateTemplate>
                        <timeZone>Central Europe Standard/Daylight Time</timeZone>
                        <ntps>
                                <ntp>
                                        <name>fritz.box</name>
                                        <ntpMode>Unicast</ntpMode>
                                </ntp>
                        </ntps>
                </dateTimeSetting>
     <callManagerGroup>
        <members>
           <member priority="0">  
              <callManager>
                 <ports>
                    <ethernetPhonePort>2000</ethernetPhonePort>
                    <sipPort>5060</sipPort>
                    <securedSipPort>5061</securedSipPort>
                 </ports>
                 <processNodeName>fritz.box</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
  </devicePool>
  <commonProfile>
     <phonePassword></phonePassword>
     <backgroundImageAccess>true</backgroundImageAccess>
     <callLogBlfEnabled>2</callLogBlfEnabled>
  </commonProfile>
  <loadInformation>SIP42.9-4-2SR3-1S</loadInformation>
  <vendorConfig>
     <disableSpeaker>false</disableSpeaker>
     <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
     <pcPort>0</pcPort>
     <settingsAccess>1</settingsAccess>
     <garp>0</garp>
     <voiceVlanAccess>0</voiceVlanAccess>
     <videoCapability>0</videoCapability>
     <autoSelectLineEnable>0</autoSelectLineEnable>
     <sshAccess>0</sshAccess>
     <sshPort>22</sshPort>
     <webAccess>0</webAccess>
     <spanToPCPort>1</spanToPCPort>
     <loggingDisplay>1</loggingDisplay>
     <loadServer></loadServer>
     <daysDisplayNotActive></daysDisplayNotActive>
     <displayOnTime>07:00</displayOnTime>
     <displayOnDuration>00:03</displayOnDuration>
     <displayIdleTimeout>00:05</displayIdleTimeout>
     <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
  </vendorConfig>
  <userLocale>
        <name>German_Germany</name>
        <uid>1</uid>
        <langCode>de_DE</langCode>
        <version></version>
        <winCharSet>iso-8859-1</winCharSet>
	</userLocale>
  <networkLocale>Germany</networkLocale>
  <networkLocaleInfo>
      <name>Germany</name>
      <version></version>
  </networkLocaleInfo>
  <deviceSecurityMode>1</deviceSecurityMode>
  <authenticationURL></authenticationURL>
  <directoryURL></directoryURL>
  <idleURL></idleURL>
  <informationURL></informationURL>

  <messagesURL></messagesURL>
  <proxyServerURL></proxyServerURL>
  <servicesURL></servicesURL>
  <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
  <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
  <dscpForCm2Dvce>96</dscpForCm2Dvce>

  <transportLayerProtocol>1</transportLayerProtocol>

  <capfAuthMode>0</capfAuthMode>
  <capfList>
     <capf>
        <phonePort>3804</phonePort>
     </capf>
  </capfList>

  <certHash></certHash>
  <encrConfig>false</encrConfig>

   <sipProfile>
     <sipProxies>
        <backupProxy></backupProxy>
        <backupProxyPort></backupProxyPort>
        <emergencyProxy></emergencyProxy>
        <emergencyProxyPort></emergencyProxyPort>
        <outboundProxy></outboundProxy>
        <outboundProxyPort></outboundProxyPort>
        <registerWithProxy>true</registerWithProxy>
     </sipProxies>

     <sipCallFeatures>
        <cnfJoinEnabled>true</cnfJoinEnabled>
        <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
        <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
        <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
        <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
        <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
        <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
        <rfc2543Hold>false</rfc2543Hold>
        <callHoldRingback>2</callHoldRingback>
        <localCfwdEnable>true</localCfwdEnable>
        <semiAttendedTransfer>true</semiAttendedTransfer>
        <anonymousCallBlock>0</anonymousCallBlock>
        <callerIdBlocking>0</callerIdBlocking>
        <dndControl>0</dndControl>
        <remoteCcEnable>true</remoteCcEnable>
     </sipCallFeatures>

     <sipStack>
        <sipInviteRetx>6</sipInviteRetx>
        <sipRetx>10</sipRetx>
        <timerInviteExpires>180</timerInviteExpires>
        <timerRegisterExpires>3600</timerRegisterExpires>
        <timerRegisterDelta>5</timerRegisterDelta>
        <timerKeepAliveExpires>120</timerKeepAliveExpires>
        <timerSubscribeExpires>120</timerSubscribeExpires>
        <timerSubscribeDelta>5</timerSubscribeDelta>
        <timerT1>500</timerT1>
        <timerT2>4000</timerT2>
        <maxRedirects>70</maxRedirects>
        <remotePartyID>true</remotePartyID>
        <userInfo>Phone</userInfo>
     </sipStack>

     <autoAnswerTimer>1</autoAnswerTimer>
     <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
     <autoAnswerOverride>true</autoAnswerOverride>
     <transferOnhookEnabled>false</transferOnhookEnabled>
     <enableVad>false</enableVad>
     <preferredCodec>g711alaw</preferredCodec>
     <dtmfAvtPayload>101</dtmfAvtPayload>
     <dtmfDbLevel>3</dtmfDbLevel>
     <dtmfOutofBand>avt</dtmfOutofBand>
     <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>

     <natEnabled>false</natEnabled>
     <natAddress></natAddress>

     <stutterMsgWaiting>0</stutterMsgWaiting>

     <callStats>true</callStats>

     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

     <startMediaPort>16384</startMediaPort>
     <stopMediaPort>32766</stopMediaPort>

     <voipControlPort>5060</voipControlPort>
     <dscpForAudio>184</dscpForAudio>
     <ringSettingBusyStationPolicy>0</ringSeettingBusyStationPolicy>
     <dialTemplate>dialplan.xml</dialTemplate>

     <phoneLabel>Buero</phoneLabel>
     <sipLines>
        <line button="1">  
           <featureID>9</featureID>
           <featureLabel>620</featureLabel>
                   <name>Schreibtisch</name>
                   <displayName>620</displayName>
                   <contact>620</contact>

           <proxy>USECALLMANAGER</proxy>
           <port>5060</port>
           <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>

           <authName>Schreibtisch</authName>
           <authPassword>(mein Password)</authPassword>

           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
           <messagesNumber>**600</messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>

           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>true</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
        </line>
        <line button="2">  
           <featureID>2</featureID>
           <featureLabel>Mobil-1</featureLabel>
           <speedDialNumber>017112345678</speedDialNumber>
        </line>
	<line button="3">  
           <featureID>2</featureID>
           <featureLabel>Mobil-2</featureLabel>
           <speedDialNumber>015112345678</speedDialNumber>
        </line>
	<line button="4">  
           <featureID>2</featureID>
           <featureLabel>Nummer-3</featureLabel>
           <speedDialNumber>08911234567</speedDialNumber>
        </line>
       <line button="5">  
           <featureID>2</featureID>
           <featureLabel>VoiceBox</featureLabel>
           <speedDialNumber>**600</speedDialNumber>
        </line>
     </sipLines>
  </sipProfile>
</device>

<loadInformation36217 model="Cisco 7962">sip42.9-4-2SR3-1S</loadInformation36217>  
<loadInformation683 model="Cisco 7962">sip42.9-4-2SR3-1S</loadInformation683>  
<loadInformation684 model="Cisco 7962">sip42.9-4-2SR3-1S</loadInformation684>  
<loadInformation685 model="Cisco 7962">sip42.9-4-2SR3-1S</loadInformation685>