Cisco 7962 an Fritzbox 7590 betreiben

[gismotro]

Hallo Fachwelt,

Ich habe mir nun entlich ein CISCO 7962 besorgt und habe die aktuelle Fw wohl geflascht bekommen, aber leider wird nach einem Neustart des Telefons immer nur : Unprovisioned

Folgende Dateien habe ich erzeugt:

SEPxxxxxxxxxxx.cnf.XML ( xxx natürlich mit Richtigen Werten der MAC):

<?xml version="2.0" encoding="UTF-8"?>
<device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>admin</sshUserId> <!-- Benutzer-->
    <sshPassword>Geheim</sshPassword> <!-- Passwort-->
    <devicePool>
        <dateTimeSetting>
            <name>W.Europe</name>
            <dateTemplate>D.M.Y</dateTemplate>
            <timeZone>W. Europe Standard/Daylight Time</timeZone>
            <ntps>
                <ntp>
                    <name>192.168.178.1</name>                 <!-- NTP-Server-->
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <members>
                <member priority="0">
                    <callManager>
                        <processNodeName>192.168.178.1</processNodeName> <!-- IP Adresse Fritzbox-->
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>                       
                            <sipPort>5060</sipPort>
                            <securedSipPort>5061</securedSipPort>
                        </ports>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
    </devicePool>
    <sipProfile>
        <sipProxies>
            <registerWithProxy>true</registerWithProxy>
        </sipProxies>
        <transportLayerProtocol>4</transportLayerProtocol>
        <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>2</anonymousCallBlock>
         <callerIdBlocking>2</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>false</remotePartyID>
         <userInfo>None</userInfo>
      </sipStack>
      <autoAnswerTimer>1</autoAnswerTimer>
      <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
      <autoAnswerOverride>true</autoAnswerOverride>
      <transferOnhookEnabled>false</transferOnhookEnabled>
      <enableVad>false</enableVad>
      <preferredCodec>g729a</preferredCodec>
      <dtmfAvtPayload>101</dtmfAvtPayload>
      <dtmfDbLevel>3</dtmfDbLevel>
      <dtmfOutofBand>avt</dtmfOutofBand>
      <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
      <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
      <kpml>3</kpml>
    <phoneLabel>OG</phoneLabel>
    <stutterMsgWaiting>2</stutterMsgWaiting>
    <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
      <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
      <startMediaPort>16384</startMediaPort>
      <stopMediaPort>32766</stopMediaPort>
        <sipLines>
            <line button="1" lineIndex="1">                 <!-- SIP Account -->
                <featureID>9</featureID>
                   <featureLabel>Homeoffice</featureLabel> <!-- Name der oben im Display steht-->
                   <proxy>USECALLMANAGER</proxy>                 
                <port>5060</port>

        <!-- ### WICHTIG: der <name> UND <authName> muessen dem username in der Fritzbox entsprechen ### -->
          
        <name>cisco7962</name>          <!-- Name der im Telefon angezeigt wird, auf der Taste-->
        <authName>cisco7962</authName> <!-- erstelle Telefonnummer der Box-->
                <authPassword>Geheim</authPassword>  
          
        <messageWaitingLampPolicy>4</messageWaitingLampPolicy>    <!-- FB AB auf Brieftaste -->
                <messagesNumber>**600</messagesNumber>       
        <ringSettingIdle>4</ringSettingIdle>             <!-- Wichtig fuer Anklopfen -->
        <ringSettingActive>5</ringSettingActive>
        </line>
        <line button="2">                         <!-- Kurzwahl 2-->
        <featureID>21</featureID>
        <featureLabel>Mustermann</featureLabel>
        <speedDialNumber>123456789</speedDialNumber>
        </line>  
        <line button="3">                         <!-- Kurzwahl 3-->
        <featureID>21</featureID>
        <featureLabel>Büro Ferkel</featureLabel>
        <speedDialNumber>**611</speedDialNumber>
        </line>         
        <line  button="4">                           <!-- Kurzwahl 4 Nicht stören-->
        <featureID>130</featureID>
        <featureLabel>Nicht stören</featureLabel>
        <helpID>369</helpID>
        </line>
        </sipLines>
        <dialTemplate>dialplan.xml</dialTemplate>             <!-- Dialplan, einfach mal googeln -->
    </sipProfile>
<userLocale>
    <name>en_US</name>
    <langCode>en</langCode>
</userLocale>
    <networkLocale>Germany</networkLocale>
<networkLocaleInfo>
    <name>Germany</name>
</networkLocaleInfo>
    <vendorConfig>
        <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
        <displayIdleTimeout>00:45</displayIdleTimeout>
        <daysDisplayNotActive> ,</daysDisplayNotActive>                <!-- Display immer an -->
        <displayOnTime>00:00</displayOnTime>
        <displayOnDuration>00:00</displayOnDuration>
        <settingsAccess>1</settingsAccess>
        <webAccess>0</webAccess>                     <!-- Web Zugriff erlaubt -->
    </vendorConfig>
    <phoneServices>
        <provisioning>2</provisioning>
        <phoneService  type="1" category="0">                 <!-- verpasster Anrufe -->
            <name>Missed Calls</name>
            <url>Application:Cisco/MissedCalls</url>
            <vendor></vendor>
            <version></version>
        </phoneService>
        <phoneService  type="2" category="0">                 <!-- zum nutzen der Brieftaste -->
        <name>Voicemail</name>
        <url>Application:Cisco/Voicemail</url>
        <vendor></vendor>
        <version></version>
    </phoneService>
    <phoneService  type="1" category="0">                 <!-- empfangener Anrufe -->
            <name>Received Calls</name>
            <url>Application:Cisco/ReceivedCalls</url>
            <vendor></vendor>
            <version></version>
        </phoneService>
    <phoneService  type="1" category="0">                 <!-- getaetigte Anrufe -->
            <name>Placed Calls</name>
            <url>Application:Cisco/PlacedCalls</url>
            <vendor></vendor>
            <version></version>
        </phoneService>

    </phoneServices>
    <addOnModules>
        <addOnModule  uuid="" idx="1">
            <deviceType>CKEM</deviceType>
            <deviceLine>36</deviceLine>
            <loadInformation></loadInformation>
            <phoneTemplateId></phoneTemplateId>
        </addOnModule>
    </addOnModules>
    <commonConfig>                             <!-- USB, Bluetooth, WLAN(9971) -->
        <usb1>1</usb1>
        <usb2>1</usb2>
        <ciscoCamera>1</ciscoCamera>
        <usbClasses>0,1,2</usbClasses>
        <sdio>1</sdio>
        <bluetooth>1</bluetooth>
        <wifi>1</wifi>
        <bluetoothProfile>0,1</bluetoothProfile>
        <joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
    </commonConfig>
    <featurePolicyFile>DefaultFP.xml</featurePolicyFile>         <!-- steuert Funktionen, zB Wahlwiederholung -->
    <loadInformation>SIP894x.9-4-2SR3-1</loadInformation>              <!-- FW Version, durch die für das jeweilige Grät zu erstezen -->
</device>

SIPDefault.cnf.xml

# SIP Default Generic Configuration File

# Image Version
image_version: 9-4-2ES26

# Proxy Server
proxy1_address: "192.168.178.1"        ; 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: "192.53.103.104"            ; SNTP Server IP Address
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: 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)

# 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: "217.10.79.9"        ; Dotted IP of Backup Proxy
proxy_backup_port: 5061        ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "sipgate.de"         ; 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: "hostname.dyndns.org"                ; 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: 32766       ; End RTP range for media (default - 32766)
nat_received_processing: 1    ; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: "192.168.178.1"         ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5061       ; 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: ""            ; 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

Telefon ist per Web erreichbar


Kann mir jemand sagen wie ich die Konfig ändern muß das das Telefon online geht, also Telefonieren kann ?

Quelle der Informationen :

1.) https://administrator.de/tutorial/cisco-telefone-ip-anschluss-fritzbox-andere-voip-anlagen-fit-machen-297337.html

2.) https://ctx4tom.wordpress.com/2014/03/16/cisco-ip-phone-an-fritzbox-teil-1-sip-firmware-aufspielen/

 

[derJackl]

Servus,

deine Vorlage für die SEPMACXXX.cnf.xml kommt mir bekannt vor. Ich hatte anfangs das selbe Problem (unprovisioned, keine Verbindung nach draußen...) und hab eine andere Vorlage ausprobiert.
Keine Ahnung ob es daran lag, oder woran überhaupt...

Alles trial and error bei mir

Kannst ja mal ausprobieren. Verschiedenes wie die display-on-Einstellungen funktionieren zwar nicht, aber telefonieren kann ich.

<?xml version="1.0" encoding="UTF-8"?>
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>admin</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.YY</dateTemplate>
<timeZone>Central Europe Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>192.168.178.1</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<processNodeName>192.168.178.1</processNodeName>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<advertiseG722Codec>1</advertiseG722Codec>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<preferredCodec>g722</preferredCodec>
<phoneLabel>XXX</phoneLabel> <!-- edit -->
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>XXX</featureLabel> <!-- edit -->
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<authName>XXX</authName>
<name>XXX</name> <!-- edit -->
<authPassword>XXX</authPassword> <!-- edit -->
<messageWaitingLampPolicy>0</messageWaitingLampPolicy>
<messagesNumber>**600</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>2</ringSettingActive>
</line>
<line button="2">
<featureID>2</featureID>
<featureLabel>Int-JacklFon</featureLabel> <!-- edit -->
<speedDialNumber>**610</speedDialNumber> <!-- edit -->
</line>
</sipLines>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>Softkeys.xml</softKeyFile>
</sipProfile>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<version></version>
</networkLocaleInfo>
<vendorConfig>
<g722CodecSupport>2</g722CodecSupport>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<displayIdleTimeout>00:30</displayIdleTimeout>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>07:00</displayOnTime>
<displayOnDuration>14:00</displayOnDuration>
<settingsAccess>1</settingsAccess>
<webAccess>1</webAccess>
</vendorConfig>
<phoneServices useHTTPS="true">
<provisioning>0</provisioning>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="0" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="0" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="0" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Contacts</name>
<url>Application:Cisco/Contacts</url>
<vendor/>
<version/>
</phoneService>
<phoneService type="0" category="0">
<name>Wetter</name>
<url>http://www.wetter.com/</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<featurePolicyFile>DefaultFP.xml</featurePolicyFile>
<commonConfig>
<ehookEnable>1</ehookEnable>
</commonConfig>
<authenticationURL></authenticationURL>
<loadInformation>sip42.9-4-2SR3-1S</loadInformation> <!-- edit -->
</device>

 

[derJackl]

Nochn Tipp, guckst du hier: https://usecallmanager.nz/document-overview.html

Hab ich durch durch Zufall gefunden. Ich hab selbst bei Null angefangen, und die Erklärungen hier haben mir den ganzen Salat um einiges verständlicher gemacht..

 

[gismotro]

Danke für deine Config, aber die funktioniert auch nicht.

Ich werde mal weiter spielen.

Gut das mein zweites Telefon( Grandstream GXP1625 HD ) nicht so kompliziert war

 

[derJackl]

Sorry, ich kann dir leider nicht sagen, warum meins funktioniert. Hast du mal den Full Reset versucht?

Hab das mal copy paste geklaut

1. Power off the phone (if using inline power, just remove the ethernet).
2. Hold the # key down.
3. Power on the phone while still holding down the # key.
4. When the line buttons scroll down orange, you can take your finger off the hash button
5. Press the following buttons: 3491672850*#
6. The line buttons should continue to scroll, but now they should be red
7. After about 60sec, this will change to green, and the phone should the boot up correctly

 

[gismotro]

Ja, hat auch nix gebracht.

 

[seelenreiter]

Das ist aber komisch. Entweder, du sprichst von einer XML Datei oder von einer CNF

SIPDefault.cnf.xml

# SIP Default Generic Configuration File

# Image Version
image_version: 9-4-2ES26

# Proxy Server
proxy1_address: "192.168.178.1" ; 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
[...]

Das sind keine XML Zeilen.

Telefone, die eine SEPMACXXX.cnf.xml lesen, brauchen keine .CNF bzw. imho können eine CNF nicht verwerten. Da lasse ich mich aber gern berichtigen.

Wenn das 7962G die .XML liest, kommst du mit den folgenden Dateien auf dem TFTP Server aus.

SEPMACXXX.cnf.xml
dialplan.xml
softkeys.xml
DefaultFP.xml

ggf. deutsche Töne, deutsche Sprachfiles, Klingeltöne und je nach Model Hintergrundbilder.

Ansonsten temporär halt Firmware Files.

Full Reset
Ja, hat auch nix gebracht.

Mir hat der Cisco Mitarbeiter gesagt, dass das nicht immer funktioniert. Manchmal muss man den Reset mehrfach wiederholen. Manchmal ginge das auch nicht und man solle das Telefon zurückschicken. :-(

Sorry, ich kann dir leider nicht sagen, warum meins funktioniert. Hast du mal den Full Reset versucht?

Hab das mal copy paste geklaut

1. Power off the phone (if using inline power, just remove the ethernet).
2. Hold the # key down.
3. Power on the phone while still holding down the # key.
4. When the line buttons scroll down orange, you can take your finger off the hash button
5. Press the following buttons: 3491672850*#
6. The line buttons should continue to scroll, but now they should be red
7. After about 60sec, this will change to green, and the phone should the boot up correctly

Mir ist nur das 123456789*0# bekannt. Mehrere Quellen im Web.