- Cisco Ip Phone 303 Downloading Xmldefault Cnf Xml Download
- Cisco Ip Phone 303 Downloading Xmldefault Cnf Xml Code
Feb 23, 2018. Cisco ip phone 303 downloading xmldefault cnf xml. A separate VLAN and its separate IP subnet to carry the voice traffic allow a large number of new devices, such as IP phones, to be introduced into the network without extensive modifications to the existing IP address scheme.
Note The use of dual boot file OS79XX.TXT is deprecated in favor of individual XML configuration files for the phone using their SEP.cnf.xml style names. This allows Cisco CallManager and SIP-based configurations to share a common TFTP server, as the XML configuration is phone-specific and allows indivbidual phones to be switched between SIP, SCCP, or MGCP images. Feb 16, 2011. Inderpal that is the EXACT answer I have been looking for to figure out all the model types for XMLDefault.cnf.xml. So instead of the individual file I have been using the XMLDefault.cnf.xml, but websites only have a portion of the phones that are available,.
SIP Configurations for 79×1 & 79×5 phones
Applies to Cisco CP-7941G, CP-7945G, CP-7961G, CP-7965G, CP-7970G, CP-7971G-GE, CP-7975G handsets.
The second wave of Cisco VoIP phones work very well with SIP platforms, although are simplest to implement in a ‘non-NAT’ environment, where the phones and PBX can be connected without any intervening NAT router.
Some of the settings for these handsets can be configured directly from the phone keypad (go to settings, then press **# and the padlock symbol in the upper-right corner will unlock).
TFTP (trivial file transfer protocol)
For the best results, the phones should be configured via a configuration file, supplied by a TFTP server.
If you have installed any of the Linux-based Asterisk/FreePBX (or similar) distributions, you will be able to use your PBX server as a TFTP server. Use your local package manager (yum, apt, etc) to install & activate.
There is also great open-source TFTP server software available for Windows. I recommend tftpd32
Creating the necessary configuration files
In the ideal world, you have Cisco CallManager running, which will allow you to configure your phones via a swanky GUI, and will generate the appropriate config files for you.
In the open-source VoIP world, you have two ways to create configuration files:
- Make use of tools supplied by your PBX software
- Manually create the files with a text editor
FreePBX Tools
FreePBX has two endpoint configuration tools available. The various versions allow the user to generate configuration files for handsets from a number of manufacturers. Both options mentioned here will create files for the older (79×0) Cisco phones, as well as the newer (79×1, 79×5) handsets.
First there is the latest commercial solution, which I expect will give you the smoothest route to generating Cisco config files. See your FreePBX installation for details on downloading and installing.
Secondly there is an open source community supported version of the endpoint manager known as the OSS Endpoint Manager which is detailed here. There are YouTube videos that cover the installation and use. I will add something more detailed here soon.
Manually Creating the Necessary Files
The files required are text based and can be created by copy/pasting suitable text into properly named files. These files should be located in the tftp server ‘home’ directory/folder.
Gigabyte motherboard g41mts2 driver for mac. The files needed are:
XMLDefault.cnf.xml – default values – will be required when starting the phone
SEPmac.cnf.xml – Handset specific values (‘mac’ should be replaced with the handset-specific MAC address found on a label at the rear of the phone, and in the on-screen settings pages).
Examples of the contents for these files are listed at the end of this article.
Telling the phone where to find the files
Once your TFTP server is running and you have created the necessary configuration files – you have to ensure that the phone will be able to retrieve the files. Once again, there are two ways to achieve this.
Cisco Ip Phone 303 Downloading Xmldefault Cnf Xml Download
Manually setting the TFTP server IP address via the phone keypad.
To do this, you must be able to unlock the configuration settings via the phone keypad.
- Press the ‘settings’ button
- Key **# to unlock
- Select the ‘Network Configuration’ option (2)
- Scroll down to the ‘Altername TFTP’ option (24 on the phone I am looking at now) – and change to YES.
- Scroll back up to the ‘TFTP Server 1’ option (8 on my phone) and press yje Edit softkey. Now you can key in the IP address of your TFTP server – using * for the ‘dot’ parts.
- Press ‘validate’ to check that the entry is syntactically acceptable.
- Press ‘Save’ to save the change.
- Press ‘Exit’ – at which point the phone will reboot and pick up the configuration from the TFTP server (if all goes well).
Automatically supplying the TFTP server IP address via DHCP.
To use this mechanism, the DHCP server on your network must be capable of supplying DHCP options data. Most home network routers do not implement these features. Microsoft office 2014 mac free full version.
If you do not have a router capable of supplying DHCP option data, then you can either use a Linux server to temporarily replace your usual DHCP server – or the Windows tftpd32 software (mentioned above) does provide a suitable DHCP server.
Using a suitable DHCP server, set the option 150 data to be the IP of your TFTP server. Then when your phone boots, it will get it’s IP address details from DHCP, AND the IP address of your TFTP server – so you can avoid the step of having to manually set the data in the phone’s settings pages.
Example Configuration Files
XMLDefault.cnf.xml
Cisco Ip Phone 303 Downloading Xmldefault Cnf Xml Code
[xml]
<Default>
<callManagerGroup>
<members>
<member priority='0'>
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model='IP Phone 7940'>P003-08-4-00</loadInformation8>
<loadInformation7 model='IP Phone 7960'>P003-08-4-00</loadInformation7>
<loadInformation6 model='IP Phone 7970'>SIP70.8-0-3S</loadInformation6> *** identifies the filename to LOAD (SIP70.8-0-3S.loads)
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
[/xml]
<Default>
<callManagerGroup>
<members>
<member priority='0'>
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model='IP Phone 7940'>P003-08-4-00</loadInformation8>
<loadInformation7 model='IP Phone 7960'>P003-08-4-00</loadInformation7>
<loadInformation6 model='IP Phone 7970'>SIP70.8-0-3S</loadInformation6> *** identifies the filename to LOAD (SIP70.8-0-3S.loads)
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
[/xml]
SEPmac.cnf.xml
- The file should be renamed, using device MAC address – e.g. SEP001BE264C906.cnf.xml
- References to xxx.xxx.xxx.xxx should be changed to reflect the IP address of your PBX server.
- References to SipUser and SipPass in the <SipLines> section should be changed to reflect the SIP Username and associated SIP Password that you are using to register the handset.
[xml]
<device>
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<dateTimeSetting>
<dateTemplate>D-M-Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</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>xxx.xxx.xxx.xxx</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<members>
<member priority='0'>
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>xxx.xxx.xxx.xxx</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<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>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL>http://xxx.xxx.xxx.xxx/cisco/services/authentication.php</authenticationURL>
<directoryURL>http://xxx.xxx.xxx.xxx/xmlservices/PhoneDirectory.php</directoryURL>
<idleURL>http://xxx.xxx.xxx.xxx/xmlservices/index.php</idleURL>
<informationURL></informationURL>
<directoryURL>http://xxx.xxx.xxx.xxx/xmlservices/PhoneDirectory.php</directoryURL>
<idleURL>http://xxx.xxx.xxx.xxx/xmlservices/index.php</idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://xxx.xxx.xxx.xxx/xmlservices/index.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<proxyServerURL></proxyServerURL>
<servicesURL>http://xxx.xxx.xxx.xxx/xmlservices/index.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<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>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</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>
<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>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<natAddress></natAddress>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<stopMediaPort>32766</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>YourPhoneLabel</phoneLabel>
<sipLines>
<line button='1'>
<featureID>9</featureID>
<featureLabel>SipUser</featureLabel>
<name>SipUser</name>
<displayName>SipUser</displayName>
<contact>SipUser</contact>
<sipLines>
<line button='1'>
<featureID>9</featureID>
<featureLabel>SipUser</featureLabel>
<name>SipUser</name>
<displayName>SipUser</displayName>
<contact>SipUser</contact>
<proxy>xxx.xxx.xxx.xxx</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>SipUser</authName>
<authPassword>SipPass</authPassword>
<authPassword>SipPass</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
</device>
[/xml]
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
</device>
[/xml]
dialplan.xml
This file tells the phone when to actually place a call to the server. Without it the phones will not successfully place any calls – they will try to send each digit dialled directly to the PBX, which will not work.
Below is the standard default file. Which will accept all key presses and only send to the server after 5 seconds of keypad inactivity. Usually, the user will type the digits and then press the ‘dial’ soft key.
More complex dialplan.xml examples can be found on the www, and will recognise local dialling formats and place a call immediately after the appropriate number of digits have been received.
The file should be placed in the same tftp root folder as the other configuration files
Hindi typing keyboard software. [xml]
<DIALTEMPLATE>
<TEMPLATE MATCH='*' Timeout='5'/> <!– Anything else –>
</DIALTEMPLATE>
[/xml]
<DIALTEMPLATE>
<TEMPLATE MATCH='*' Timeout='5'/> <!– Anything else –>
</DIALTEMPLATE>
[/xml]
Home > Articles > Cisco
␡- Understanding the Theory of Voice VLANs
Page 1 of 9Next >
This chapter describes the typical boot process for Cisco IP phones and indicates the best practices related to configuring the network to support Voice over IP (VoIP).
This chapter is from the book CCNA Voice Exam Cram
This chapter is from the book
This chapter is from the book
To deploy Cisco Unified Communications Manager Express, network services need to be established and properly configured on all routers and switches for Cisco IP phones to function. This chapter describes the typical boot process for Cisco IP phones and indicates the best practices related to configuring the network to support Voice over IP (VoIP).
Understanding the Theory of Voice VLANs
A Cisco IP phone can act as a three-port switch. Just like a switch, the phone can support 802.1Q frames between itself and another switch, and more than one VLAN can be supported between the Cisco Unified IP phone and an access switch. Figure 7.1 illustrates how the Cisco IP phone acts as a three-port switch.
Figure 7.1 Cisco IP phones include a three-port switch.
The following are the three ports of the Cisco IP phone:
- The external port that connects to a 10/100/1000 Ethernet switch
- The external 10/100/1000 Ethernet port for PC connections
- An internal 10/100/1000 Ethernet port for VoIP traffic
The benefits of this type of configuration include the following:
- Cisco IP phones can be deployed on the network without IP address scalability problems. IP subnets usually have more than 50 percent—and often more than 80 percent—of their IP addresses allocated. A separate VLAN and its separate IP subnet to carry the voice traffic allow a large number of new devices, such as IP phones, to be introduced into the network without extensive modifications to the existing IP address scheme.
- Voice and data VLANs allow the logical separation of data and voice traffic due to different characteristics. This separation allows you to handle each traffic type individually, applying different quality of service (QoS) policies to each VLAN for monitoring and managing them separately.
The following are IP addressing recommendations when adding Cisco IP phones to an existing data network:
- Continue to use existing addressing for data devices (PCs, workstations, and so on).
- Add Cisco IP phones and use DHCP to provision IP addresses and operating parameters.
- Use new subnets for Cisco IP phones if they are available in the existing address space, or use private addressing such as the 10.0.0.0 network (see RFC 1918 for details) if subnets are not available in the existing address space.
With IP phones residing in a separate VLAN—a voice VLAN—it is easier for you to automate the process of deploying IP phones. The IP phone communicates with the switch, using the Cisco Discovery Protocol (CDP), to request the voice VLAN if present. The switch CDP response provides the phone with the appropriate 802.1Q VLAN ID, known as the voice VLAN ID (VVID). The PC traffic travels across the same connection without the 802.1Q tag inserted.
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