Great
Just be cautious of any unwanted tag-a-longs from any unknown source
JN
Ionel Chila wrote:
Your Firmware here —>
http://www.firewall.cx/downloads/cisco-tools-a-applications/cisco-ip-phone-a-ata-firmware-downloads/133-7942-a-7962-ip-phone-sccp-a-sip.html
Load the latest SIP firmware and you’re good to go. I have the 7945 version
which is perfect. I have 4 of these in my home. How to below:
Recently I needed to change the firmware on some Cisco 7965 phones from SCCP to
SIP. By far the simplest method is loading the COP file on UCM and letting the
phone upgrade on its own. In my case, this upgrade was being done without
using UCM. The Cisco read-me doc for the SIP firmware covers the COP upgrade
procedure only. It tells you that you may unzip the files on a TFTP server but
there is no procedure which explains what else you must do to load the SIP
firmware.
In this example I am upgrading Cisco 7965 phones to SIP firmware 8.5. Once you
have downloaded the zipped version of the SIP firmware from CCO place the
unzipped files in your TFTP servers root directory. Modify your
XMLDefaults.cnf.xml file so the load information matches your firmware.
<loadInformation8 model=”Cisco 7965″>SIP45.8-5-3TH1</loadInformation8>
You should connect your IP phone to LAN where DHCP provides the IP, subnet, and
TFTP server IP. Make sure your phone has DHCP enabled = YES. Your DHCP server
needs to support DHCP Options. TFTP option 66 is required for Cisco phones
running SIP. Option 66 can be used to provide an IP address (recommended) but
can also support a DNS names (assuming you are also providing at least one DNS
server IP via DHCP). Option 150 only supports IP addresses and is required for
SCCP firmware. You can safely configure your DHCP to issue both TFTP options.
Next pull the power from your phone and plug it back in. Hold down # until the
line keys start to blink and press 123456789*0# and your phone should reset. Your
phone should display “Upgrading” on the screen. If you are using a Unix based
tftp server you can execute tcpdump port 69 and you should see your phone
requesting the files. Your phone should display the progress of the SIP firmware
upgrade and eventually reboot. After it reboots you can press Settings > Model
Information and scroll down until you see the Call Control Protocol = SIP.
If you performed a factory reset and did not have DHCP enabled then your phone
is most likely stuck at the Upgrading screen. Pressing keys on the phone will
not change the status. At this point you should pull the power, plug it back
in, hold # and then enter the keys 3491672850*# to factory reset the phone.
This allows the phone to clear its flash and still download new firmware. Your
screen is going to be totally black and it will appear as if your phone is not
functional, but the phone is really sending a DHCP request and waiting for an
IP, subnet, and TFTP IP assignment before proceeding to download the firmware.
All of this is happening while the phone’s screen is black. If you want to read
the official word on this, Cisco has a field notice on their web site.
Monitoring tcpdump on the TFTP server is useful in this case because you know
the phone is doing something. Also, you can view the DHCP bindings to verify
your phone successfully acquired an IP address.
Then setup the XML config file with your Mac address in /tftpboot in the
Astlinux box and make sure your DHCP server know where to pass the tftp IP.
After SEP put your Cisco phone MAC address. I have SEP00235EB65698.cnf.xml
HOME-PBX tftpboot # cat SEP00235EB65698.cnf.xml
<!-- FIXME: Change to your own phone number (or another unique ID) -->
<device xsi:type="axl:XIPPhone" ctiid="7134542795">
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
<devicePool>
<dateTimeSetting>
<!-- FIXME: Set your preferred date format and timezone here -->
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Central Standard/Daylight Time</timeZone>
<ntps>
<!-- NTP might not actually work, but the phone can set the
date/time from the SIP response headers -->
<ntp>
<name>pool.ntp.org <http://pool.ntp.org></name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<!-- This section probably does not do anything useful. -->
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>192.168.0.15</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<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>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<!-- Force short registration timeout to keep NAT connection alive -->
<timerRegisterExpires>180</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>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress>1</natAddress>
<!-- FIXME: This will appear in the upper right corner of the display -->
<phoneLabel>Yenos Budros</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>16391</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<!-- FIXME: Text to display next to line button #1 -->
<featureLabel>Home Office Line</featureLabel>
<!-- FIXME: FQDN or IP of your SIP proxy -->
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<!-- FIXME: SIP username -->
<name>200</name>
<!-- FIXME: Name to display on outbound caller ID -->
<displayName>Yenos Budros</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<!-- FIXME: SIP authorization name (often matches username) -->
<authName>200</authName>
<!-- FIXME: SIP authorization password -->
<authPassword>ciscocisco</authPassword>
<sharedLine>true</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<!-- FIXME: "Messages" key will autodial this number -->
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact></contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>2</featureID>
<featureLabel>CVX Conference</featureLabel>
<speedDialNumber>18773444239</speedDialNumber>
</line>
<line button="3">
<featureID>2</featureID>
<featureLabel>Teresa Cell</featureLabel>
<speedDialNumber>18333547560</speedDialNumber>
</line>
<line button="4">
<featureID>2</featureID>
<featureLabel>Yenos Cell</featureLabel>
<speedDialNumber>17134222795</speedDialNumber>
</line>
<line button="6">
<featureID>2</featureID>
<featureLabel>KIDS EMERGENCY 911</featureLabel>
<speedDialNumber>911</speedDialNumber>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<!-- FIXME: Change this to upgrade the firmware -->
<loadInformation>SIP45.9-3-1SR3-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>1</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<!-- For Sunday (1) and Saturday (7):
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
Current default is to enable the display 24/7.
-->
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>06:30</displayOnTime>
<displayOnDuration>12:00</displayOnDuration>
<displayIdleTimeout>00:01</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp></versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>0</deviceSecurityMode>
<!--
<authenticationURL>http://yourwebserver/authenticate.php</authenticationURL>
<directoryURL>http://yourwebserver/directory.xml</directoryURL>
-->
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<!--
<servicesURL>http://phone-xml.berbee.com/menu.xml</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
On Feb 12, 2021, at 4:10 PM, John Novack SCII_U <jnov...@comcast.net
<mailto:jnov...@comcast.net>> wrote:
Does the phone have the SIP firmware load?
I believe the 7940/7960 series can have more than one type of firmware, Sip,
one for the Cisco Call Manager, and possibly one other
You would need the SIP firmware, and if it was used with Call Manager, probably
doesn't have it
Not sure you can obtain it if you aren't hooked up with Cisco.
Aren't these phones now EOL, or nearly so?
John Novack
Jerry Gartner wrote:
I want to use some Cisco 7942's from a retired Cisco system. Being of newb,
what are best practices for this? Are there provision templates available for
these phones? Should they be factory reset? Etcetera, etcetera.
--
Kind Regards,
Jerry Gartner
--
Dog is my Co-Pilot
_______________________________________________
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
<mailto:Astlinux-users@lists.sourceforge.net>
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
--
Dog is my Co-Pilot
_______________________________________________
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.