As many people know, Cisco IP phones always use its own SCCP (Skinny Call Control Protocol) and SIP protocols with CUCM technology. In some different cases, engineers can be obliged to use other 3PCC servers in order to register their IP phones without CUCM.
Requirements:
- Cisco 7942 IP phone software (cmterm-7942_7962-sip.9-4-2SR3-1)
- Tftpd64 (https://archive.codeplex.com/?p=tftpd64)
- SIP Server
In this case, firstly we need to obtain new
cisco IP phone software that is supported SIP protocol from cisco’s official
webpage. https://software.cisco.com/download/home/281346593/type
After the download, SEPxxx.cnf.xml file should put in the
downloaded software file. I’ll share this xml file below of this page but it
must be prepared with your sip credential in order to register your sip server.
Configuring the SEPxxx.cnf.xml file;
·
Put IP phone’s MAC address on SEPxxx.cnf.xml
instead of xxx
·
Put SIP server IP address to : <processNodeName>111.111.111.111</processNodeName>
·
Put username to : <authName>egemenulus</authName>
·
Put password to : <authPassword>egemenulus</authPassword>
·
Put load file name to : <loadInformation>SIP42.9-4-2SR3-1S</loadInformation>
Note: loadInformation name must be
same with load file’s name that is in software file.
Upgrading new software to IP
phone;
1-
In
first, we have to erase ITL and CTL file on IP phone that remained from CUCM
server by manual.
- Setting
- **# (setting unlock)
- Security config
- Trust List
- ITL file
- Erase
2-
Second
step, you need to put your new software directory to Current Directory section
on tftpd64. Also, connect PC and IP phone with Ethernet cable.
3-
In
this step, we wait IP phone in order to obtain IP from DHCP server after
connected the test network. When you get IP, you can put Alternate TFTP IP as
seen on server interfaces by like this;
- Setting
- Network config
- Ipv4 Config
- Alternate TFTP
- Yes
- TFTP server 1
- Server Interfaces IP
Note: In same
cases like factory reset, IP phone may fall in boot loop and it does not let
you set tftp server ip. In order to fix the problem, you can figure out
solution with OPTION 150 to find DHCP server. Through the DHCP OPTION 150,
Cisco IP phones find the required TFTP server.
As a result,
after these steps, upgrading will starts automatically. In order to view
upgrading logs, you can press Log Viewer section on tftpd64. After the
successful finished, you will see Update Locale on the IP phone’s screen.
Now, you have
converted your Cisco IP phone’s protocol from SCCP to SIP...
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Ya</dateTemplate>
<timeZone>UTC+03:00 Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>x.x.x.x</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>x.x.x.x</processNodeName> Set Your SIP Server IP
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<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>1</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>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<phoneLabel>sip_username</phoneLabel> >>>>>>>> You can set your sip_username. This section will seen on phone's screen
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>sip_username</featureLabel> >>>>>> You can set your sip_username. This section will seen on phone's screen
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>sip_username</name> >>>>>>>> You can set your sip_username
<displayName>sip_username</displayName> >>>>>>>> You can set your sip_username
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>sip_username</authName> >>>>>>>> You must set your sip_username
<authPassword>sip_password</authPassword> >>>>>>>> You must set your sip_password
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>3501</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>sip_username</contact> >>>>>>>> You must set your sip_username
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>Dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.9-4-2SR3-1S</loadInformation> >>>>> you must set your load file name that was in downloaded software file
<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>
</vendorConfig>
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<networkLocale>US</networkLocale>
<networkLocaleInfo>
<name>US</name>
<version>5.0(2)</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>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
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