Cisco 9971 and TFTP

Ok as an update. I have managed to get the phone firmware updated. There was a block size limit on the TFTP server on the trixbox, which was choking on the rootfs9971.9-0-2.sebn file (34MB) from being loaded on the phone, so I used a TFTP server my PC instead. The active firmware is now successfully showing as sip9971.9-0-2.

I found a good Asterisk and Cisco phone configuration article here: http://minded.ca/2009-12-16/configure-cisco-ip-phones-with-asteri...

I have used the configuration files zipped up in this article as starting off point. I know that the SEP081FF3622FB1.cnf.xml file is being processed, as I have altered the user and network locales and the phone is now asking for g4-tones.xml and gd-sip.jar. I will need to try and dig these up somewhere from Cisco, but I don't think they are critical for operation - happy to be corrected.

The configurations files I have as follows:

SEP081FF3622FB1.cnf.xml
XMLDefault.cnf.xml
dialplan.xml
ringlist.xml
List.xml (optional really)

I am ignoring all other non XML configuration files including SIPDefault.cnf as the phone does not appear to use them.

The phone however is still not registering with trixbox, so any further hints or tips would be greatly appreciated. I think the key to making it register is still somewhere in the SEP081FF3622FB1.cnf.xml file but I'm not sure what to try next.

Cheers
Sean

Hi,

Afraid I never did get this phone working with Trixbox. I spent about a week or more looking for a solution, but it was fruitless. Perhaps with more debugging it could be possible, but not for the fainthearted!

I think it's a Call Manager only solution. Although I'd be happy to be proven wrong!

Good luck

Sean

seansimpson: working on this a bit more. Got into the debug mode of the phone.
Not sure if you turned on the web interface or not, but it seems the only way to get at the debug logs is via the web interface.
just ssh to the phone, then use username: debug pass: debug and set some debugs.
Make sure you have the web interface on. Log onto the web interface then pull the debug logs.

From what i can see sofar i get a "CVM-ccsip_register_send_msg: Error: cc_cfg_table is null." error. Looking these up online it seems a few people with the 79xx series phone are starting to have that issue with the 9.x firmwares.

What firmware are you running?

I am going to try to test a few firmwares out right now to see if it works.

FYI: you need to use 0 for this, not 1..

Update: Interesting discovery, the phone does work with Asterisk no problem, however it has a issue registering with it.
Try this. In your SEPxxxxxxx.cnf.xml file change the from true to false. Then the phone fully boots and allows for calls to be made out.. I can even get the KEM sidecar to work. However without registration no inbound connections will work. Maybe someone can figure out a work around.

The problem seems to be with the registration to proxy issue.

Got it to register!!! it works .. So no it is not just a Callmanager phone, it works with trix.

The trick is to use TCP for registration and all the sip traffic, thus one has to enable tcp and add transport=tcp in the sip user extension.

Got to work out some of the bugs of the phone, as "forward all", redial, etc soft keys aren't playing nice, but all in all it works.

Sean - Did you enable TCP for SIP on your trixbox as the noted earlier in this thread? You must be running 2.8 to support this.

Personally I would install chan_sccp-b and use the phone in SCCP mode. Many more features.

The guys over at the chan-sscp-b project are doing some amazing work.

this message means:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

I can not register the phone.
failure to do something

Cayo.

......I have outgoing calls, but I have no incoming calls...
help

change XXX.XXX.XXX.XXX to your servers IP address
change 200 to your extension number as well as password and contact

<?xml version="1.0" encoding="UTF-8"?>
<device  xsi:type="axl:XIPPhone" ctiid="7" uuid="{89b84111-68bf-4498-8233-d9935482e0f2}">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>
<ipAddressMode>0</ipAddressMode>
<allowAutoConfig>true</allowAutoConfig>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<tzdata>
<tzolsonversion>2009p</tzolsonversion>
<tzupdater>tzupdater.jar</tzupdater>
</tzdata>
<featurePolicyFile>DefaultFP0000000000-c7a6c673-7479-46b0-839e-014d3d093963.xml</featurePolicyFile>
<devicePool  uuid="{1b1b9eb6-7803-11d3-bdf0-00108302ead1}">
<revertPriority>0</revertPriority>
<name>Default</name>
<dateTimeSetting  uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}">
<name>CMLocal</name>
<dateTemplate>M/D/Y</dateTemplate>
<timeZone>Greenwich Standard Time</timeZone>
<olsonTimeZone>Etc/GMT</olsonTimeZone>
</dateTimeSetting>
<callManagerGroup>
<name>Default</name>
<tftpDefault>true</tftpDefault>
<members>
<member  priority="0">
<callManager>
<name>CUCM</name>
<description>CUCM</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>XXX.XXX.XXX.XXX</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo  uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
<name>Enable</name>
<srstOption>Enable</srstOption>
<userModifiable>true</userModifiable>
<ipAddr1>XXX.XXX.XXX.XXX</ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1></sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
<preemption>Default</preemption>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<TVS>
<members>
<member  priority="0">
<port>2445</port>
<address>XXX.XXX.XXX.XXX</address>
</member>
</members>
</TVS>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy>USECALLMANAGER</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
<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>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
</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>0</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>
<kpml>3</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<T302Timer>15000</T302Timer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<poundEndOfDial>false</poundEndOfDial>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line  button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>299</name>
<displayName></displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>200</authName>
<authPassword>200</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>200</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>4</maxNumCalls>
<busyTrigger>2</busyTrigger>
</line>
<line  button="7">
<featureID>21</featureID>
<featureLabel>User1</featureLabel>
<speedDialNumber>202</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
<line  button="8">
<featureID>21</featureID>
<featureLabel>User2</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
</sipLines>
<externalNumberMask></externalNumberMask>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
<dscpForTelepresence>128</dscpForTelepresence>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>Softkey.xml</softKeyFile>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
</sipProfile>
<MissedCallLoggingOption>1000</MissedCallLoggingOption>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>sip9971.9-0-3</loadInformation>

<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<sdio>1</sdio>
<garp>1</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>0</loggingDisplay>
<recordingTone>1</recordingTone>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<logServer></logServer>
<g722CodecSupport>0</g722CodecSupport>
<headsetWidebandUIControl>0</headsetWidebandUIControl>
<headsetWidebandEnable>0</headsetWidebandEnable>
<lldpAssetId></lldpAssetId>
<powerPriority>0</powerPriority>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
</vendorConfig>

<commonConfig>
</commonConfig>
<enterpriseConfig>
</enterpriseConfig>
<versionStamp>1280870288-91f38f2d-7fff-4a42-87cd-76f9b4a802fc</versionStamp>
<addOnModules>
<addOnModule  uuid="{c42eaca3-d387-58e7-4f3f-46dfea8b7593}" idx="1">
<deviceType>CKEM</deviceType>
<deviceLine>36</deviceLine>
<loadInformation></loadInformation>
<phoneTemplateId>a6a6664c-b468-c012-ef23-9b8e09a2b5f0</phoneTemplateId>
</addOnModule>
</addOnModules>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>8.0.0.1(4)</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>8.0.0.1(4)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://CUCM:8080/ccmcip/authenticate.jsp</authenticationURL>
<directoryURL>http://CUCM:8080/ccmcip/xmldirectory.jsp</directoryURL>
<idleURL></idleURL>
<informationURL>http://CUCM:8080/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://CUCM:8080/ccmcip/getservicesmenu.jsp</servicesURL>
<secureAuthenticationURL>https://CUCM:8443/ccmcip/authenticate.jsp</secureAuthenticationURL>
<secureDirectoryURL>https://CUCM:8443/ccmcip/xmldirectory.jsp</secureDirectoryURL>
<secureIdleURL></secureIdleURL>
<secureInformationURL>https://CUCM:8443/ccmcip/GetTelecasterHelpText.jsp</secureInformationURL>
<secureMessagesURL></secureMessagesURL>
<secureServicesURL>https://CUCM:8443/ccmcip/getservicesmenu.jsp</secureServicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<phonePersonalization>1</phonePersonalization>
<rollover>0</rollover>
<singleButtonBarge>0</singleButtonBarge>
<joinAcrossLines>0</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>CUCM</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<advertiseG722Codec>1</advertiseG722Codec>
<mobility>
<handoffdn></handoffdn>
<dtmfdn></dtmfdn>
<ivrdn></ivrdn>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<userId>299</userId>
<phoneServices  useHTTPS="true">
<provisioning>2</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>Personal Directory</name>
<url>Application:Cisco/PersonalDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService  type="1" category="0">
<name>Corporate Directory</name>
<url>Application:Cisco/CorporateDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>

The entry should go in sip_custom.post.conf

See this link for instructions:

http://www.freepbx.org/configuration_files

vi /var/www/html/admin/modules/core/functions.inc.php

Line 3769

array($account,'port',$db->escapeSimple((isset($_REQUEST['port']))?$_REQUEST['port']:'5060'),$flag++),
array($account,'transport',$db->escapeSimple((isset($_REQUEST['transport']))?$_REQUEST['transport']:'udp'),$flag++),
array($account,'qualify',$db->escapeSimple((isset($_REQUEST['qualify']))?$_REQUEST['qualify']:'yes'),$flag++),

Line 5968

$tmparr['nat'] = array('value' => 'yes', 'level' => 1);
$tmparr['port'] = array('value' => '5060', 'level' => 1);
$tmparr['transport'] = array('value' => 'udp', 'level' => 1);
$tmparr['qualify'] = array('value' => 'yes', 'level' => 1);
$tmparr['callgroup'] = array('value' => '', 'level' => 1);

for freePBX 2.5
http://www.asterisk-peru.com/node/1501

image

you are scared to modify the code?

sip_additional.conf

Hi,
solstice: What do you mean with CUCM, USECALLMANAGER. is the address of the Asterisk server?

DefaultFP0000000000-c7a6c673-7479-46b0-839e-014d3d093963.xml

is this file needed?

regards.
Cayo

hi,
the sip.conf configurations on asterisk 1.6 is:
[general]
allowguest=no
tcpenable=yes
tcpbindaddr=0.0.0.0
callevents=yes
context=internos
domain=172.20.1.3
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
useragent=X-Lite
videosupport=yes

[101]
transport=tcp
type=friend
accountcode=101
language=es
secret=101
qualify=no
nat=no
host=dynamic
dtmfmode=rfc2833
context=internos
canreinvite=yes
callerid=Prueba
disallow=all
allow=ulaw
allow=alaw

Some corrections?

Cayo.

For the "sample" config above change the

<featurePolicyFile>DefaultFP0000000000-c7a6c673-7479-46b0-839e-014d3d093963.xml</featurePolicyFile >

to:

<featurePolicyFile>DefaultFP.xml</featurePolicyFile>

This way you can have one generic policy file for multiple phones that isn't specific to one config.

Also a default policy file will look like:

<featurePolicy  name="Default Policy">
<versionStamp>0000000000</versionStamp>
<featureDef  name="Forward All">
<id>1</id>
<enable>true</enable>
</featureDef>
<featureDef  name="Park">
<id>2</id>
<enable>false</enable>
</featureDef>
<featureDef  name="Divert (Alerting)">
<id>3</id>
<enable>false</enable>
</featureDef>
<featureDef  name="Conference List">
<id>4</id>
<enable>true</enable>
</featureDef>
<featureDef  name="Speed Dial">
<id>5</id>
<enable>true</enable>
</featureDef>
<featureDef  name="Call Back">
<id>6</id>
<enable>true</enable>
</featureDef>
<featureDef  name="Redial">
<id>7</id>
<enable>true</enable>
</featureDef>
<featureDef  name="Barge">
<id>8</id>
<enable>true</enable>
</featureDef>
<featureDef  name="Divert (Connected)">
<id>9</id>
<enable>false</enable>
</featureDef>
</featurePolicy>

Hope this helps! The phone now works great for the most part here, with the exception of CallforwardAll and BLF but that shall be solved too.

Also a reminder, the sample config i posted above is for a 9971 with a side car. If you don't have a side car, you will need to disable those options.

Also, because we all are getting a bit annoyed by people who are just toooo darn lazy to do their own debugs, please post proper debugs if you want help, by SSH'ing into the phone. The username/password is defined in your config, in the sample it is admin/config. After entering you will get a prompt to enter another login, which you use debug/debug. From there you can run any debug command required. The info is logged into a data messages file which can be pulled via web interface. One will discover ones problems most of the time on their own.

hi,

such as enabling video calls and conference key.

Cayo.

video is a interesting issue, as of Cisco callmanager 8.1 it doesn't even support it either.. there are some custom firmwares and patches on CM that are supposed to enable it.. but if you can make it happen there, its awesome!

federicom.. its fairly unstable .. It seems to take down the whole system enabling it with the 9971.

solstice...

Thanks sooo much for your postings, it has helped me solved this issue connecting a 9951, that i have worked on for weeks.

I had a lot of problems getting it to connect with sip tcp, so i started looking into trying to connect it with udp.

I think i might actually have solved this :)

I changed your config line:

transportLayerProtocol set for 4

to

transportLayerProtocol set for 2

and then it just connected and working.

Hope someone else will benefit from this.

Happy holidays to everyone :)

Danscout

Could you please share the XMLDefault.cnf.xml and SEP081ff3622FB1.cnf.Xml for Cisco 9951 or cisco 9971 sample files

Thanks in Advance

Jay

Thanks for XMLdefault. But as per the tftp log it is picking the sip firmware.

Mar 12 16:09:45 trixbox1 atftpd[24686]: File /tftpboot/CTLSEP1C17D3408358.tlv not found
Mar 12 16:09:45 trixbox1 atftpd[24686]: Server thread exiting
Mar 12 16:09:45 trixbox1 atftpd[24686]: Serving SEP1C17D3408358.cnf.xml to 10.0.0.140:49152
Mar 12 16:09:45 trixbox1 atftpd[24686]: File /tftpboot/SEP1C17D3408358.cnf.xml not found
Mar 12 16:09:45 trixbox1 atftpd[24686]: Server thread exiting
Mar 12 16:09:46 trixbox1 atftpd[24686]: Serving XMLDefault.cnf.xml to 10.0.0.140:49152
Mar 12 16:09:46 trixbox1 atftpd[24686]: Server thread exiting
Mar 12 16:10:16 trixbox1 atftpd[24686]: Serving CTLSEP1C17D3408358.tlv to 10.0.0.140:49152
Mar 12 16:10:16 trixbox1 atftpd[24686]: File /tftpboot/CTLSEP1C17D3408358.tlv not found
Mar 12 16:10:16 trixbox1 atftpd[24686]: Server thread exiting
Mar 12 16:10:16 trixbox1 atftpd[24686]: Serving SEP1C17D3408358.cnf.xml to 10.0.0.140:49152
Mar 12 16:10:16 trixbox1 atftpd[24686]: File /tftpboot/SEP1C17D3408358.cnf.xml not found
Mar 12 16:10:16 trixbox1 atftpd[24686]: Server thread exiting
Mar 12 16:10:16 trixbox1 atftpd[24686]: Serving XMLDefault.cnf.xml to 10.0.0.140:49152
Mar 12 16:10:16 trixbox1 atftpd[24686]: Server thread exiting

Please let me know how to proceed futher

Ok it worked. and go regsitered too,

malu, if you check the cisco forums, video doesn't properly work on most call managers even. I have tried with full blown call managers here and it didn't play nice. The only way i was able to make it work was with a "Unreleased" firmware. Have never tried using this phone against trix, due to multiple factors (licenses, some compatibility issues with that firmware)
However you could check that all the video settings are correct in trix first and test it against a couple xlite clients first. Once it works with those, intercept the video stream from the phone.
It may be that it requires a special resolution setting for the stream which isn't documented.

I think I had a few incorrect parameters set in SEP081FF3622FB1.cnf.xml and also some other places. So it helped to just run through everything again after I had posted my latest missive.

The phone is now registered, showing "299" as an extension and has dial tone...and the wrong time set! but its still major progress as far as I'm concerned!!

Cheers

Sean

Help me. I've done all that is written in this forum, but my 9951 is not registered on asterisk
here's what I did... in sip_general_custom.conf
tcpenable=yes
tcpbindaddr=0.0.0.0
allowsubscribe=yes
notifyringing = yes
notifyhold = yes
notifycid = yes
callcounter = yes

in sip_custom_post.conf
[XXX](+)
transport=tcp,udp
But my cisco still not registered
Please HELP!!!

Hi,

When I was trying to get my camera working a little while ago, I posted some findings in the Cisco Support Forums. See: https://supportforums.cisco.com/message/3417096#3417096

If you update the "vendorConfig" section of the phone XML configuration file to correctly set the "powerNegotiation", "ciscoCamera" and "videoCapability" XML tags then your camera will start working. You may need to be a little patient after you change the settings and reboot it as the camera take a little time to get enabled the first time.

Very briefly what works (not an exhaustive list) on my phone is as follows:
1. All basic phone functions
2. Message Waiting Lamp / Presence Indication / Red Message Envelope on screen next to extension (although I had to patch chan_sip.c)
3. The camera (although video calls presently do not, see note below)
4. Wired headsets into the headset port

What doesn't work (not an exhaustive list):
1. Voicemail button (in lieu I have assigned an extension soft key to call my VM)
2. Directory button
3. Video calls, although I strongly suspect this is an Asterisk H.264 signalling issue as I'm still using TB v2.8.0.4 (Asterisk 1.6.0.26)

I'm about to kick of the process of migrating to PBX in a Flash 2.0 with Asterisk 10 as it has been pretty clear for a while now that TB CE is at a dead end developmentally.

Good luck!

This post has help me greatly!!! Thank you for all the info!!! I still have a couple of minor issues I am trying to work out, if someone can help me that would be great!
1) Intercom/Paging not working: I have added seperate extension on freepbx and turn on intercom/paging. I have also added

<autoAnswer>
<autoAnswerEnabled>3</autoAnswerEnabled>
<autoAnswerMode>Auto Answer with Speakerphone</autoAnswerMode>
<autoAnswer>

2) I can't get the phone to display any other time zone other than GMT. I would like for it to display Eastern Standard Time.

3)I can't the feature plan to work either. I have added a feature plan but it does not seem to read it. Any help would be great!!!

Thanks

<?xml version="1.0" encoding="UTF-8"?>
<device  xsi:type="axl:XIPPhone" ctiid="7" uuid="{89b84111-68bf-4498-8233-d9935482e0f2}">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>
<ipAddressMode>0</ipAddressMode>
<allowAutoConfig>true</allowAutoConfig>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<featurePolicyFile>DefaultFP.xml</featurePolicyFile>
<devicePool  uuid="{1b1b9eb6-7803-11d3-bdf0-00108302ead1}">
<revertPriority>0</revertPriority>
<name>Default</name>
<dateTimeSetting  uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}">
<name>CMLocal</name>
<dateTemplate>M/D/Y</dateTemplate>
<timeZone>Eastern Standard Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<name>Default</name>
<tftpDefault>true</tftpDefault>
<members>
<member  priority="0">
<callManager>
<name>CUCM</name>
<description>CUCM</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>10.0.1.253</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo  uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
<name>Enable</name>
<srstOption>Enable</srstOption>
<userModifiable>true</userModifiable>
<ipAddr1>10.0.1.253</ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1></sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
<preemption>Default</preemption>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<TVS>
<members>
<member  priority="0">
<port>2445</port>
<address>10.0.1.253</address>
</member>
</members>
</TVS>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy>USECALLMANAGER</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
<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>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
</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>2</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>fasle</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<kpml>3</kpml>
<phoneLabel>Eric Webb</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<T302Timer>15000</T302Timer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line  button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>Line 1</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>103</name>
<displayName>103</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
<autoAnswer>
<callWaiting>3</callWaiting>
<authName>103</authName>
<authPassword>password12</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>200</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>4</maxNumCalls>
<busyTrigger>2</busyTrigger>
</line>
<line  button="2" lineIndex="1">
<featureID>9</featureID>
<featureLabel>intercom</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>203</name>
<displayName>203</displayName>
<autoAnswer>
<autoAnswerEnabled>3</autoAnswerEnabled>
<autoAnswerMode>Auto Answer with Speakerphone</autoAnswerMode>
<autoAnswer>
<callWaiting>3</callWaiting>
<authName>203</authName>
<authPassword>password12</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>5</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>200</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>4</maxNumCalls>
<busyTrigger>2</busyTrigger>
</line>
<line  button="3">
<featureID>21</featureID>
<featureLabel>User2</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
 <line button="4">  
         <featureID>20</featureID>  
         <featureLabel>Menu</featureLabel>  
         <serviceURI>http://phone.icnventures.com/directory.xml</serviceURI>  
      </line>
<line  button="5">
<featureID>21</featureID>
<featureLabel>User5</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
<line  button="6">
<featureID>21</featureID>
<featureLabel>User6</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
<line  button="20">
<featureID>21</featureID>
<featureLabel>User20</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
<line  button="25">
<featureID>21</featureID>
<featureLabel>User25</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
<line  button="30">
<featureID>21</featureID>
<featureLabel>User40</featureLabel>
<speedDialNumber>204</speedDialNumber>
<featureOptionMask>0</featureOptionMask>
</line>
</sipLines>
<externalNumberMask></externalNumberMask>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
<dscpForTelepresence>128</dscpForTelepresence>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>softkey.xml</softKeyFile>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
</sipProfile>
<MissedCallLoggingOption>1000</MissedCallLoggingOption>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>sip9971.9-2-3-27</loadInformation>

<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<sdio>1</sdio>
<garp>1</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>0</loggingDisplay>
<recordingTone>0</recordingTone>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<logServer></logServer>
<g722CodecSupport>0</g722CodecSupport>
<headsetWidebandUIControl>0</headsetWidebandUIControl>
<headsetWidebandEnable>0</headsetWidebandEnable>
<lldpAssetId></lldpAssetId>
<powerPriority>0</powerPriority>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
</vendorConfig>

<commonConfig>
</commonConfig>
<enterpriseConfig>
</enterpriseConfig>
<versionStamp>1280870288-91f38f2d-7fff-4a42-87cd-76f9b4a802fc</versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>8.0.0.1(4)</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>8.0.0.1(4)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://CUCM:8080/ccmcip/authenticate.jsp</authenticationURL>
<directoryURL>http://CUCM:8080/ccmcip/xmldirectory.jsp</directoryURL>
<idleURL>http://phone.icnventures.com/logo.bmp</idleURL>
<logoURL>http://phone.icnventures.com/logo.jpg</logoURL>
<informationURL>http://CUCM:8080/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://CUCM:8080/ccmcip/getservicesmenu.jsp</servicesURL>
<secureAuthenticationURL>https://CUCM:8443/ccmcip/authenticate.jsp</secureAuthenticationURL>
<secureDirectoryURL>https://CUCM:8443/ccmcip/xmldirectory.jsp</secureDirectoryURL>
<secureIdleURL></secureIdleURL>
<secureInformationURL>https://CUCM:8443/ccmcip/GetTelecasterHelpText.jsp</secureInformationURL>
<secureMessagesURL></secureMessagesURL>
<secureServicesURL>https://CUCM:8443/ccmcip/getservicesmenu.jsp</secureServicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<phonePersonalization>1</phonePersonalization>
<rollover>0</rollover>
<singleButtonBarge>0</singleButtonBarge>
<joinAcrossLines>0</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>CUCM</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<advertiseG722Codec>1</advertiseG722Codec>
<mobility>
<handoffdn></handoffdn>
<dtmfdn></dtmfdn>
<ivrdn></ivrdn>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<userId>299</userId>
<phoneServices  useHTTPS="true">
<provisioning>2</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>Personal Directory</name>
<url>Application:Cisco/PersonalDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService  type="1" category="0">
<name>Corporate Directory</name>
<url>http://phone.icnventures.com/directory.xml</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService  type="1" category="0">
<name>Personal Directory 2</name>
<url>http://phone.icnventures.com/phone.xml</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>

....Well just about all working. That's almost 2 years now since I started this thread! Hahaha

Further to my last post, I now have working (using Asterisk 1.8.7.0 (patched) and FreePBX 2.9.0.10) the following major phone features:

- 2-way Video (Cisco to Cisco and Cisco to 3rd party H.264 enabled endpoints)
- MWI
- Call Forward All soft-key
- Directories under the Directory feature button
- Voicemail access (including visual voicemail) from the Messages feature button
- Call History lists (missed, placed, received calls) with BLF
- BLF/Presence enabled on programmable feature keys for speed-dials and also call lists
- DND toggle programmed on line key with 'Cisco' DND animation
- SIP NOTIFY for reset and restart of Cisco phones

Thanks again to all those folks here originally who helped me so much when I was using TB!

Over at the FPBX forums, SkykingOH kindly suggested that I share my Cisco XML phone config file implementation on my FreePBX setup. So here is my (somewhat anonymised) Cisco 9971 phone configuration file, together with some configuration tips. I have a lot of people from all over the net to thank for getting me this far. There are still some phone features which don't work (Conference key for example), but by and large the main phone functions are working very well.

My setup consists of

1. Cisco 9971 phones running 9.2.3-27 firmware, with attached CUVC camera
2. Soft-phones (a mixture of Vippie Video with H.264 support on iPhone and X-Lite 4 on OS X)
3. FreePBX 2.9.0.10 with Asterisk 1.8.7.0 with
   
   1. Asterisk patch Presence Subscription on Cisco Phones as listed on JIRA
   2. Asterisk patch video format 'negotiation' as published in the Asterisk-Video mailing list from Artem Makhutov
4. Linksys SPA3102-UK analogue voice gateway for terminating PSTN circuit (FXO) and analogue extension (FXS)
5. Dialectic v1.84 dialler integration on OS X

Please note the following (hopefully I'm not telling you how to 'suck eggs' too much here):

1. Some Basic Stuff
   1. Timezone and NTP values. My timezone is set to "GMT Standard/Daylight Time" and my date format "D/M/Ya" set accordingly to my regional preference. The values entered here must conform to the Cisco preferred values for timezones. Change the NTP server IP or DNS name to your preferred.
   2. My asterisk server IP address is 192.168.0.70 - change this to reflect your server IP address.
   3. <loadInformation> tag is set to reflect the firmware versions on my phone, alter yours to suit.
   4. <featurePolicyFile> should be set to reflect the name of the Feature Policy file as installed in your /tftpboot directory. The 89xx and 99xx series of phones apparently no longer uses the soft keys.xml configuration method (although I actually have one listed/installed as a hangover from a previous setup)
   5. Labels for the phone and lines labels, secrets for SIP lines etc to be changed from my dummy values.
2. Phone Display Activation
   See <daysDisplayNotActive> I've set the phone display to be blacked out on all days, until the phone is used, then remains on for 1 hour; therefore all days Sunday (1) through Saturday (7) are listed here. Note this overrides the settings in <displayOnTime> and <displayOnDuration>.
3. BLF and Presence Functionality
   
   1. To enable BLF/presence to function correctly you will need to apply Asterisk patch Presence Subscription on Cisco Phones as listed on JIRA.
   2. Update the cisco phone extension entries in sip_custom_post.conf to include SIP functionality in the presence patch and video patch (more on that below). e.g. for extension 100 include an entry such as:
      

      [100](+)
      ciscounified=yes
      dndbusy=yes
      video_fmtp=profile-level-id=42801E\;packetization-mode=0\;level-asymmetry-allowed=1
      video_btias=1000000
      video_imageattr=recv [x=640,y=480,q=0.50]
      

      (Please Note: If you have a mixed estate of Cisco and non-Cisco devices it is much better to include these additional SIP settings only for the Cisco endpoints (in the manner above) rather than for the entire SIP estate. I had previously written here that the additional SIP setting can be added via the FreePBX 2.9 GUI under Asterisk SIP settings -> Other SIP settings. This would constitute a global SIP configuration change and is fine if your endpoints are 100% Cisco but if you have a mixture of different endpoints then you will likely get strange error messages in your Asterisk CLI/logs or other strange consequences)

   3. <timerRegisterExpires> If you have particularly busy/intensive phone users I've heard that setting this value to 900 will help from keep the phone from losing the ability to display BLF status after a few weeks. I've not personally had reason to change mine from the default. See Asterisk patch Presence Subscription on Cisco Phones.
4. SIP Lines
   See section <sipLines>
   1. Ordinary Lines - this configuration file reflects a phone setup with 2 ordinary lines, per <featureID>9</featureID>. The first line is a "normal" line and the second line is an "intercom" line which is set to auto answer.
   2. Speed Dials - the remaining 4 line keys are configured as speed dials, per <featureID>21</featureID>. Where presence indication is desired set <featureOptionMask>1</featureOptionMask> for the speed dial entry.
   3. Setting preferred line - as I wanted the phone to default to the first "normal' line when the phone was taken off hook I set <alwaysUsePrimeLine>true</alwaysUsePrimeLine> and <alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
5. Voicemail
   My VM number is defined within the lines section under <messagesNumber>1000</messagesNumber>. I have a customised dial plan using '1000' to dial into VoiceMailMain, but of course you can use the usual *97 or *98 here or whatever you prefer to dial your voicemail.
6. Video and Camera Capability
   
   1. My 9971 phones have cameras installed hence you will need the tags <ciscoCamera> and <videoCapability> as set out variously below.
   2. If you are using your phone with Power over Ethernet supplied by a non-Cisco switch then you will also need to set <powerNegotiation>0</powerNegotiation> so that the phone can extract the full per port PoE power from the ethernet port on the non-cisco switch, otherwise the camera will fail to initialise due to lack of power. This could also be overcome by using a Cisco power brick to power the phone.
   3. To enable two-way H.264 based video calls you will also need to apply the Asterisk patch as published ion the Asterisk-Video mailing list from Artem Makhutov.
7. Call Lists Application (native)
   To enable the built in Cisco application for missed/received/placed calls refer to the code under tags <phoneServices>. Once you have the presence patch applied BLF for call lists can be applied by <callLogBlfEnabled>3</callLogBlfEnabled> under the commonProfile section
8. Other Applications
   I have added apps in the <phoneServices> section for some existing external Cisco XML services like Berbee, some XML directory demos as well as my own custom PHP-based voicemail application (which I'm not allowed to call by its more commonly know name) residing on my asterisk server.
   1. Application Menu. I have set the external Berbee and 'Australian Services' XML apps in the "Application" menu of the phone, accessed by the dedicated Application feature button and set by tag <phoneService type="0" category="0">
   2. Contacts Menu. To set up apps accessed by the dedicated "Contacts" button set the appropriate phone service using <phoneService type="1" category="0">. I've currently got placeholder (external) XML directory apps there
   3. Messages Menu. Finally apps included in the "Messages" menu accessed by corresponding dedicated button are set using <phoneService type="2" category="0">. The native voicemail app is here (just need to set your <messagesNumber> accordingly for one-touch access to voicemail. I've also added my custom voicemail app here so when I press the "Messages" button I get a small menu allowing me either (1) dial voicemail or (2) access my custom voicemail app
9. Call Forward All
   This can be enabled by setting the following in extension_custom.conf. Note the presence patch noted above must be applied.

   [from-internal-custom]
   ; Strip the x-cisco-serviceuri- prefix
   exten => _[x]-cisco-serviceuri-.,1,Goto(${EXTEN:19},1)
   
   ; Enable forwarding
   exten => _cfwdall-.,1,Answer
   same => next,Set(SIPPEER(${CHANNEL(peername)},callforward)=${EXTEN:8})
   same => next,Hangup(normal_clearing)
   
   ; Disable forwarding
   exten => cfwdall,1,Answer
   same => next,Set(SIPPEER(${CHANNEL(peername)},callforward)=)
   same => next,Hangup(normal_clearing)
   

10. Do Not Disturb
    The following will enable a customised DND toggle feature, along the lines of the default *76 feature code, but updated to use the 'DONOTDISTURB' device state available once the Presence Subscription on Cisco Phones patch has been applied. The benefit is that the Cisco DND animations will be displayed for other subscribers to the extension hint, also the line button assigned for DND on the phone will glow red and also show the DND animation.
    1. firstly enable "Enable Custom Device States" under "Advanced Settings" in FPBX
    2. In extensions_custom.conf create a customised dnd-toggle app. In the example below I've used 760 as my assigned extension to call app-dnd-toggle-custom.
       

       [app-dnd-toggle-custom]
       ; Custom DND Toggle using Cisco DoNotDisturb state
       exten => 760,1,Answer
       exten => 760,n,Wait(1)
       exten => 760,n,Macro(user-callerid,)
       exten => 760,n,GotoIf($["${DB(DND/${AMPUSER})}" = ""]?activate:deactivate)
       exten => 760,n(activate),Set(DB(DND/${AMPUSER})=YES)
       exten => 760,n,Set(STATE=DONOTDISTURB)
       exten => 760,n,Gosub(app-dnd-toggle-custom,sstate,1)
       exten => 760,n(hook_on),Playback(do-not-disturb&activated)
       exten => 760,n,Macro(hangupcall,)
       exten => 760,n(deactivate),Noop(Deleting: DND/${AMPUSER} ${DB_DELETE(DND/${AMPUSER})})
       exten => 760,n,Set(STATE=NOT_INUSE)
       exten => 760,n,Gosub(app-dnd-toggle-custom,sstate,1)
       exten => 760,n(hook_off),Playback(do-not-disturb&de-activated)
       exten => 760,n,Macro(hangupcall,)
       exten => sstate,1,Set(DEVICE_STATE(Custom:DND${AMPUSER})=${STATE})
       exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)})
       exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
       exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
       exten => sstate,n,Set(ITER=1)
       exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVDND${CUT(DEVICES,&,${ITER})})=${STATE})
       exten => sstate,n,Set(ITER=$[${ITER} + 1])
       exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
       exten => sstate,n(return),Return()
       

    3. In the dialplan create extensions (in this example 760100 and 760120 for extensions 100 and 120 respectively) which are to be assigned to the line keys on the relevant phones.
       

       [ext-dnd-hints-custom]
       exten => 760100,1,Goto(app-dnd-toggle-custom,760,1)
       exten => 760100,hint,Custom:DND100
       exten => 760120,1,Goto(app-dnd-toggle-custom,760,1)
       exten => 760120,hint,Custom:DND120
       

    4. On the phone assign set up a speed dial on the desired 'DND' line key (for the first phone this would be 760100) which calls app-dnd-toggle-custom. This also enables the line key to subscribes to the relevant DND hint for extension 100. As such when DND is activated by pressing the line key the usual audible confirmation is heard, the line key then lights up and the DND animation is displayed. Ditto when the key is pressed again to toggle DND off.
    5. Finally don't forget to set <featureOptionMask>1</featureOptionMask> against the relevant line key otherwise it will not subscribe to the DND hint.
11. SIP NOTIFY for reset and restart of Cisco phones
    Although it is possible to write a php or perl script (see my recent post) to remotely restart or reset Cisco 89xx and 99xx phones, by far the simplest and quickest method is to use the SIP NOTIFY command. By applying the patch, it is possible to use the following commands from the Asterisk CLI to restart (reloads line keys, dial-plan and soft keys) or reset phones:

    CLI> sip notify cisco-restart ${PEERNAME}
    CLI> sip notify cisco-reset ${PEERNAME}

    Also I've just noticed (although it was probably there all along) that the MWI indicator can be manually cleared using the following command:

    CLI> sip notify clear-mwi ${PEERNAME}

    The following variables and functions should be included in sip_notify_custom.conf settings:

    [clear-mwi]
    Event=>message-summary
    Content-type=>application/simple-message-summary
    Content=>Messages-Waiting: no
    Content=>Message-Account: sip:asterisk@127.0.0.1
    Content=>Voice-Message: 0/0 (0/0)
    Content=>
    
    ; Cisco
    
    [cisco-restart]
    Event=>service-control
    Subscription-State=>active
    Content-Type=>text/plain
    Content=>action=restart
    Content=>RegisterCallId={${SIPPEER(${PEERNAME},regcallid)}}
    Content=>ConfigVersionStamp={0000000000000000}
    Content=>DialplanVersionStamp={0000000000000000}
    Content=>SoftkeyVersionStamp={0000000000000000}
    
    [cisco-reset]
    Event=>service-control
    Subscription-State=>active
    Content-Type=>text/plain
    Content=>action=reset
    Content=>RegisterCallId={${SIPPEER(${PEERNAME},regcallid)}}
    Content=>ConfigVersionStamp={0000000000000000}
    Content=>DialplanVersionStamp={0000000000000000}
    Content=>SoftkeyVersionStamp={0000000000000000}
    

Extract from SEPmac_address.cnf.xml

<?xml version="1.0" encoding="UTF-8"?>
<device>
<deviceProtocol>SIP</deviceProtocol>
 
  <sshUserId>admin</sshUserId>
  <sshPassword>cisco</sshPassword>
 
  <devicePool>
		<dateTimeSetting> 
			<dateTemplate>D/M/Ya</dateTemplate> 
			<timeZone>GMT Standard/Daylight Time</timeZone> 
			<ntps> 
				<ntp>
					<name>130.88.203.12</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>192.168.0.70</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
  </devicePool>
 
  <commonProfile>
     <phonePassword></phonePassword>
     <backgroundImageAccess>true</backgroundImageAccess>
     <callLogBlfEnabled>3</callLogBlfEnabled>
  </commonProfile>
 
  <loadInformation>sip9971.9-2-3-27</loadInformation>
  <featurePolicyFile>DefaultFP.xml</featurePolicyFile>
 
  <vendorConfig>
     <disableSpeaker>false</disableSpeaker>
     <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
     <pcPort>0</pcPort>
     <settingsAccess>1</settingsAccess>
     <garp>0</garp>
     <voiceVlanAccess>0</voiceVlanAccess>
     <ciscoCamera>1</ciscoCamera>
     <videoCapability>1</videoCapability>
     <usbClasses>0,1,2</usbClasses>
     <sdio>1</sdio>
     <wifi>0</wifi>
     <bluetoothProfile>0,1</bluetoothProfile>
     <powerNegotiation>0</powerNegotiation>
     <autoSelectLineEnable>0</autoSelectLineEnable> 
     <webAccess>0</webAccess>
     <sshAccess>0</sshAccess>
     <g722CodecSupport></g722CodecSupport>
     <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
     <displayOnTime>08:30</displayOnTime>
     <displayOnDuration>09:30</displayOnDuration>
     <displayIdleTimeout>01:00</displayIdleTimeout>
     <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
     <spanToPCPort>1</spanToPCPort>
     <loggingDisplay>1</loggingDisplay>
     <loadServer></loadServer>
  </vendorConfig>

  <commonConfig>
     <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>

  <enterpriseConfig>
     <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>
     <videoCapability>1</videoCapability>
     <webAccess>0</webAccess>
     <eapAuthentication>2</eapAuthentication>
     <webProtocol>0</webProtocol>
  </enterpriseConfig>
 
  <advertiseG722Codec></advertiseG722Codec>

  <networkLocale>United_Kingdom</networkLocale> 
 
	<networkLocaleInfo> 
		<name>English_United_Kingdom</name> 
		<uid>64</uid> 
		<version>1.0.0.0-1</version> 
	</networkLocaleInfo> 
 
  <deviceSecurityMode>1</deviceSecurityMode>
 
  <idleTimeout>0</idleTimeout>
  <authenticationURL></authenticationURL>
  <directoryURL></directoryURL>
  <idleURL></idleURL>
  <informationURL></informationURL>
  <messagesNumber></messagesNumber>  
  <messagesURL></messagesURL>
  <proxyServerURL></proxyServerURL>
  <servicesURL></servicesURL>
  <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
  <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
  <dscpForCm2Dvce>96</dscpForCm2Dvce>
 
  <transportLayerProtocol>2</transportLayerProtocol>
  <dndCallAlert>5</dndCallAlert>
  <phonePersonalization>1</phonePersonalization>
  <rollover>0</rollover>
  <singleButtonBarge>0</singleButtonBarge>
  <joinAcrossLines>1</joinAcrossLines>
  <autoCallPickupEnable>false</autoCallPickupEnable>
  <blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
  <blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
 
  <capfAuthMode>0</capfAuthMode>
  <capfList>
     <capf>
        <phonePort>3804</phonePort>
     </capf>
  </capfList>
 
  <certHash></certHash>
  <encrConfig>false</encrConfig>
  <sipProfile>
     <sipProxies>
        <backupProxy>USECALLMANAGER</backupProxy>
        <backupProxyPort>5060</backupProxyPort>
        <emergencyProxy>USECALLMANAGER</emergencyProxy>
        <emergencyProxyPort>5060</emergencyProxyPort>
        <outboundProxy></outboundProxy>
        <outboundProxyPort></outboundProxyPort>
        <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>2</callHoldRingback>
        <localCfwdEnable>true</localCfwdEnable>
        <semiAttendedTransfer>true</semiAttendedTransfer>
        <anonymousCallBlock>2</anonymousCallBlock>
        <callerIdBlocking>2</callerIdBlocking>
        <dndControl>0</dndControl>
        <remoteCcEnable>true</remoteCcEnable>
	<retainForwardInformation>true</retainForwardInformation>
     </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>0</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>true</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>
     <natEnabled>false</natEnabled>
     <natAddress>192.168.0.70</natAddress>
 
     <stutterMsgWaiting>2</stutterMsgWaiting>
 
     <callStats>false</callStats>
     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
 
 
     <startMediaPort>10000</startMediaPort>
     <stopMediaPort>20000</stopMediaPort>
     <voipControlPort>5060</voipControlPort>
     <dscpForAudio>184</dscpForAudio>
	 <dscpVideo>136</dscpVideo>
	 <dscpForTelepresence>128</dscpForTelepresence>
     <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
	 <softKeyFile>softkey.xml</softKeyFile>
     <dialTemplate>dialplan.xml</dialTemplate>
     <phoneLabel>Your_Label_Here</phoneLabel>
     <sipLines>
        <line button="1" lineIndex="1">
           <featureID>9</featureID>
           <featureLabel>Sean 100</featureLabel>
           <name>100</name>
           <displayName>Study</displayName>
           <contact>100</contact>
           <proxy>USECALLMANAGER</proxy>
           <port>5060</port>
           <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>
 
           <authName>100</authName>
           <authPassword>your_secret_here</authPassword>
 
           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
	   <messageWaitingAMWI>1</messageWaitingAMWI>
           <messagesNumber>1000</messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>
 
           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
			<maxNumCalls>4</maxNumCalls>
			<busyTrigger>2</busyTrigger>
        </line>
        <line button="2" lineIndex="2">
           <featureID>9</featureID>
           <featureLabel>Intercom</featureLabel>
           <name>101</name>
           <displayName></displayName>
           <contact>101</contact>
           <proxy>USECALLMANAGER</proxy>
           <port>5060</port>
           <autoAnswer>
              <autoAnswerEnabled>3</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>

           <authName>101</authName>
           <authPassword>your_secret_here</authPassword>

           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
           <messagesNumber></messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>

           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
                        <maxNumCalls>4</maxNumCalls>
                        <busyTrigger>2</busyTrigger>
        </line>
		<line  button="3">
			<featureID>21</featureID>
			<featureLabel>Annette</featureLabel>
			<speedDialNumber>110</speedDialNumber>
			<featureOptionMask>1</featureOptionMask>
		</line>
                <line  button="4">
                        <featureID>21</featureID>
                        <featureLabel>Bedroom</featureLabel>
                        <speedDialNumber>120</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
                </line>
		<line  button="5">
                        <featureID>21</featureID>
                        <featureLabel>DECTphone</featureLabel>
                        <speedDialNumber>130</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
                </line>
		<line  button="6">
                        <featureID>21</featureID>
                        <featureLabel>DND</featureLabel>
                        <speedDialNumber>760100</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
                </line>
	</sipLines>
  </sipProfile>

  <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>Personal Directory</name>
		<url>Application:Cisco/PersonalDirectory</url>
		<vendor></vendor>
		<version></version>
	</phoneService>
	<phoneService  type="1" category="0">
		<name>Corporate Directory (Demo)</name>
		<url>http://directory.ciscoxmlservices.com/demo1/demo1/</url>
		<vendor></vendor>
		<version></version>
	</phoneService>
	<phoneService  type="0" category="0">
                <name>Australian Services</name>
                <url>http://cisco.internect.net/</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
	<phoneService  type="0" category="0">
                <name>Berbee</name>
                <url>http://phone-xml.berbee.com/menu.xml</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
	<phoneService  type="2" category="0">
                <name>Visual Voice Mail</name>
                <url>http://192.168.0.70/cisco/voicemail.php?user=100</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
  </phoneServices>
</device>