This is the instruction I am referring to.
“[then search for “patch12” (note the change in case) and add this exact line directly below it:]
%patch13 -p1”
Perhaps something changed in the source rpms. Here’s what I see around the 600 line:
do you find anything resembling this?
Well I found the patch12 line, my text search function wasn’t searching the whole document.
How do I confirm that the patch has been installed? I gather that all configuration is done within the SEPMAC cofig document and not through EPM? Also, is the TFTP boot folder an add-on module? I had to point my phone to the TFTP server on my laptop in order to get the file uploaded.
Thanks for your help!
A guess a quick way to check the patch is to try to restart a phone from the console. If you have added the lines to sip_notify_custom.conf (do a ‘core reload’ in asterisk afterward) then you should be able to restart a phone with ‘sip notify cisco-restart 101’ or whatever extension number. Plenty of other ways but that’s the first that came to mind.
As for TFTP, FreePBX runs it’s own tftp server and places the files in /tftpboot this is where the files that EPM generates are placed. The phone gets told where to look by a 150 response in the DHCP response. Hope that makes sense.
Thanks for the info. The config file isn’t validating at the endpoint. I’m going to have to play around with the file and see if I can get it applied.
I tried your test with the ‘sip notify cisco-restart 5007’ and got the phone to restart successfully.
As for the config file, I have it listed below.
<!-- SEP1234567890.cnf.xml -->
<?xml version="1.0" encoding="utf-8"?>
<device>
<versionStamp>c1b05fde-ef64-11e5-9ce9-5e5517507c66</versionStamp>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<transportLayerProtocol>1</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<featurePolicyFile>featurepolicy.xml</featurePolicyFile>
<missedCallLoggingOption>0</missedCallLoggingOption>
<commonProfile>
<backgroundImageAccess>true</backgroundImageAccess>
<phonePassword></phonePassword>
<callLogBlfEnabled>0</callLogBlfEnabled>
</commonProfile>
<advertiseG722Codec>1</advertiseG722Codec>
<authenticationURL></authenticationURL>
<messagesURL></messagesURL>
<informationURL></informationURL>
<servicesURL></servicesURL>
<directoryURL></directoryURL>
<!-- ** Firmware file -->
<loadInformation>sip88xx.11-0-1-11</loadInformation>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/YYa</dateTemplate>
<timeZone>Central Standard/Daylight Time</timeZone>
<olsonTimeZone>America/Chicago</olsonTimeZone>
<ntps>
<ntp>
<name>time.nist.gov</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member>
<callManager>
<processNodeName>MyServerIP</processNodeName>
<ports>
<sipPort>5060</sipPort>
</ports>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<alwaysUsePrimeLine>true</alwaysUsePrimeLine>
<phoneLabel>5007</phoneLabel>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-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>0</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>Phone</userInfo>
</sipStack>
<transferOnhookEnabled>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<startMediaPort>10000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<natEnabled>false</natEnabled>
<natReceivedProcessing>false</natReceivedProcessing>
<natAddress></natAddress>
<kpml>0</kpml>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialtemplate.xml</dialTemplate>
<softKeyFile>softkeys.xml</softKeyFile>
<sipLines>
<!-- Main line, register with asterisk using your credentials as in FreePBX -->
<line button="1" lineIndex="1">
<!-- the 'lineIndex' here is ESSENTIAL, without it, the phone will 'lock' missed calls on the display, causing the phone to be basically broken. Do not put a line index on the other keys, that will also break things. -->
<featureID>9</featureID>
<featureLabel>Bryan</featureLabel>
<callWaiting>3</callWaiting>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>5007</name>
<authName>5007</authName>
<authPassword>MyPW</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>2</ringSettingActive>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>3</maxNumCalls>
<busyTrigger>3</busyTrigger>
</line>
<!-- I like a clear view of the phones Do Not Disturb state, this linekey provides that -->
<line button="2">
<featureID>130</featureID>
<featureLabel>DND</featureLabel>
</line>
<!-- BLF KEY, set to the extension number of a station you want this phone to monitor and be able to pick up calls for etc -->
<line button="3">
<featureID>21</featureID>
<featureLabel>SecondPhone</featureLabel>
<speedDialNumber>5005</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="5">
<featureID>159</featureID>
<featureLabel>Record</featureLabel>
</line>
</sipLines>
</sipProfile>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>1</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>1</videoCapability>
<ciscoCamera>1</ciscoCamera>
<autoSelectLineEnable>1</autoSelectLineEnable>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<!-- the next four lines: At 8am, activate phone and keep the screen on for an hour
gives the office a good 'awake' look. Then after 9am, turn the display on if there is an
incoming call. Turn off after 10 minutes of inactivity. -->
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>01:00</displayOnDuration>
<displayIdleTimeout>00:30</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<simplifiedNewCall>0</simplifiedNewCall>
<phoneBookWebAccess>1</phoneBookWebAccess>
<handsetWidebandEnable>2</handsetWidebandEnable>
<headsetWidebandEnable>2</headsetWidebandEnable>
<headsetWidebandUIControl>1</headsetWidebandUIControl>
<handsetWidebandUIControl>1</handsetWidebandUIControl>
<separateMute>0</separateMute>
<sshAccess>1</sshAccess>
<sshPort>22</sshPort>
<autoCallSelect>1</autoCallSelect>
<incomingCallToastTimer>5</incomingCallToastTimer>
<joinAndDirectTransferPolicy>1</joinAndDirectTransferPolicy>
<bluetooth>1</bluetooth>
<bluetoothProfile>0,1</bluetoothProfile>
<btpbap>1</btpbap>
<usb1>1</usb1>
<usb2>1</usb2>
<usbClasses>0,1,2</usbClasses>
<g722CodecSupport>2</g722CodecSupport>
<!-- set to 1 to use the softkey.xml file, set to 0 to use the featurePolicy.xml file -->
<softkeyControl>1</softkeyControl>
</vendorConfig>
<!--userLocale>
<name>English_United_Kingdom</name -->
<!-- the name above is the exact name of the folder inside /tftpboot where the locale files are stored -->
<!--uid>1</uid>
<langCode>en_GB</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale -->
<!--networkLocaleInfo>
<name>English_United_Kingdom</name>
<uid>1</uid>
<langCode>en_GB</langCode>
<version>1.0.0.0-1</version>
</networkLocaleInfo-->
<phoneServices>
<provisioning>0</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<!--phoneService type="1" category="0">
<name>Contacts</name -->
<!-- set the URL below to match your server DNS name, and edit the extension to match the correct extension for this file -->
<!--url>http://freepbx.server.internal/cisco/directory.php?xtn=101</url>
<vendor></vendor>
<version></version>
</phoneService -->
<!--phoneService type="0" category="0">
<name>Visual Voicemail</name>
<url>**url of the visual voicemail application**</url>
<vendor></vendor>
<version></version-->
</phoneService>
</phoneServices>
</device>The firmware file is the wrong one for the 8845 (the two video phones use different firmware from the rest of the range, it seems) so I’ve changed that in the wiki. That’s not what was causing the file not to validate though.
Remove the comment from the first line, turns out these phones are super fussy over the XML and this seems to annoy it. Even with that change I couldn’t get your file to load onto the phone.
The problem is the sections you have commented out. Switch back to the sample file, making only the obvious changes to the serverIP, username etc and you’ll find that the config will load.
Hi …
Any luck getting 9971 or the 88xx series to have 2 lines, both can receive and make calls?
on the 88xx I managed to get a second line but I can’t make calls from it and on the 9971 I can’t recieve call on second line … strange.
Please advise.
Thanks