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trinidad
Beginner

Configure Cisco DX650 with asterisk/Sip Provider

Hi Everyone,

 

I just picked up one of these hoping to replace my SPA303. I should have done my homework before purchasing it. *EDIT* Figured this out steps listed below for anyone else wondering. If you are knowledgeable to about getting daylight savings time working via the config file please let me know, it's the only other thing outstanding.

 

Thanks!

12 REPLIES 12
trinidad
Beginner

Quick update here, so I guess I figured out most of this out, I have it connected to my asterisk server. The only issue now is that some of the hard buttons don't work. For instance volume control on the speakerphone button will not work. Oddly the hold call button does work. So I can't adjust call or possibly ringer volume etc. Anyways making progress would appreciate help as always. *EDIT* the hard buttons were a hardware failure. Replaced all working now.

 

Thanks

Can you share, what you did to get it working please ? 

Sure np, I haven't seen any other posts out there on how to do this for this series phone just the 79xx and 88xx series etc. Here is my XML file I have marked where you need to replace your items, I'm not sure what some of those items do but over the last few nights I was able to parse out most of it also if you need to change the network time info/location feel free to modify. This site was super helpful in deciphering most of the items here: http://usecallmanager.nz/sip-conf.html

 

<?xml version="1.0" encoding="UTF-8"?>
<device  xsi:type="axl:XIPPhone" ctiid="62943" uuid="{e045c922-43ad-2320-24c9-be1f8abc3d0b}">
<fullConfig>true</fullConfig>
<portalDefaultServer></portalDefaultServer>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>YOUR USERNAME</sshUserId>
<sshPassword>YOUR PASSWORD</sshPassword>
<ipAddressMode>0</ipAddressMode>
<redirectEnable>false</redirectEnable>
<echoMultiEnable>false</echoMultiEnable>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<ipMediaAddressFamilyPreference>0</ipMediaAddressFamilyPreference>
<mlppDomainId>000000</mlppDomainId>
<mlppIndicationStatus>Off</mlppIndicationStatus>
<preemption></preemption>
<executiveOverridePreemptable></executiveOverridePreemptable>
<devicePool  uuid="{d0181915-1eac-910c-3a0f-f03c26afd832}">
<revertPriority>0</revertPriority>
<name>Phones - 1.5M Video EST EDT</name>
<dateTimeSetting  uuid="{daaf53f2-bb03-b274-953c-5090869fc211}">
<name>EST-5</name>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<olsonTimeZone>America/New_York</olsonTimeZone>
</dateTimeSetting>
<callManagerGroup>
<name>VOIP</name>
<tftpDefault>false</tftpDefault>
<members>
<member  priority="0">
<callManager>
<name>VOIP.ms</name>
<description></description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>INSERT YOUR ASTERISK OR VOIP ADDRESS</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo  uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1></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>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<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>
<URIDialingDisplayPreference>1</URIDialingDisplayPreference>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>0</anonymousCallBlock>
<callerIdBlocking>0</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>60</timerRegisterExpires>
<timerRegisterDelta>0</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<kpml>3</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<T302Timer>5000</T302Timer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<poundEndOfDial>false</poundEndOfDial>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<organizationTopLevelDomain>YOUR IP ADDRESS OF ASTERISK OR VOIP PROVIDER</organizationTopLevelDomain>
<sipLines>
<line  button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>Office</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>201</name>
<displayName>YOUR DISPLAY NAME</displayName>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>YOUR USERNAME</authName>
<authPassword>YOUR PASSWORD</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>YOUR VOICEMAIL NUMBER</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact></contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>10</maxNumCalls>
<busyTrigger>6</busyTrigger>
</line>
<line  button="3">
<featureID>1</featureID>
</line>
</sipLines>
<externalNumberMask>YOUR DID OR PHONE NUMBER</externalNumberMask>
<voipControlPort>5060</voipControlPort>
<ringSettingBusyStationPolicy>1</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>SKd7581e75-e2ff-277e-6fcc-2a7739543647.xml</softKeyFile>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
</sipProfile>
<MissedCallLoggingOption>10</MissedCallLoggingOption>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
<loadInformation>sipdx650.10-2-5-194</loadInformation>
<inactiveLoadInformation></inactiveLoadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<allowBTContactImport>1</allowBTContactImport>
<allowBTMobileHandsfree>1</allowBTMobileHandsfree>
<recordingTone>0</recordingTone>
<settingsAccess>1</settingsAccess>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<deviceUIProfile>0</deviceUIProfile>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<garp>1</garp>
<multiUser>0</multiUser>
</vendorConfig>
<commonConfig>
<ciscoCamera>1</ciscoCamera>
<videoCapability>1</videoCapability>
<webProtocol>0</webProtocol>
<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<sendKeyAction>1</sendKeyAction>
<RingLocale>0</RingLocale>
<appInstallFromAndroidMarket>true</appInstallFromAndroidMarket>
</commonConfig>

<versionStamp>1387322115-49d5fd49-52b6-4926-b708-11c02cb22c22</versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version></version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>Canada</networkLocale>
<networkLocaleInfo>
<name>Canada</name>
<uid>64</uid>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>

<transportLayerProtocol>1</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>YOUR ASTERISK OR VOIP ADDRESS</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<advertiseG722Codec>0</advertiseG722Codec>
<mobility>
<handoffdn>8888</handoffdn>
<dtmfdn>41200</dtmfdn>
<ivrdn>86547810</ivrdn>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<TLSResumptionTimer>0</TLSResumptionTimer>
<phoneServices  useHTTPS="true">
<provisioning>0</provisioning>
<phoneService  type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</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>
</phoneServices>
</device>

In the DX650 you will need to set the TFTP alternate ip address to yours where you have it setup.   Also of the  key components <loadInformation>sipdx650.10-2-5-194</loadInformation> under about phone check what your active load is just replace with yours. If you have firmware you can place that in there minus the extension.  Finally once you are finished with your settings save the XML as SEPXXXXXXXXXXXX.conf.xml replace the X's with the phones MAC address no ":" required just the 12 alphanumeric. 

 

If using Asterisk/FreePBX you will also need to set the extensions to use TCP as well under the advanced tab (assuming you are using freepbx.) Cisco seems to like using TCP vs UDP. Also you will have to change this in the general SIP settings as well:

 

**SETTINGS > ASTERISK SIP SETTINGS > SIP LEGACY chan sip"Advanced General Settings" **Enable TCP = YES     //  This will allow extensions to connect cisco seems to work best with TCP. I tried UDP no go. This assumes also you are using port 5060 which is what I am using in the config file in the link above.

 

Then just unplug the power and or POE and plug it back in. You should see the phone requesting some other files etc. I think that's really about it. Also when it loads it might ask you to set a PIN, I'm sure somewhere in this XML file it sets the requirement for that *EDIT* (NEW Config file below use that one). I tried this with both my SIP provider and my home asterisk server and both worked.

 

Lastly after it pulls down the conf file it takes a little while for you to see the line active where the phone icon shows your number and name etc, so just wait a little. The settings are persistent as well so if you have a power outage it will retain it's settings, it just takes a while after it tries to find the TFTP server if it's not running don't worry it will load the last config.

 

I have uploaded the dialplan.xml you can adjust as needed but that is a basic one be sure to place that in the TFTP as well in the root folder.

 

<DIALTEMPLATE>
    <TEMPLATE MATCH="999" Timeout="0"/> <!-- Emergency -->
    <TEMPLATE MATCH="112" Timeout="0"/> <!-- Emergency -->
    <TEMPLATE MATCH="0500......" Timeout="0"/> <!-- Apparently 0500 is always 10 digits -->
    <TEMPLATE MATCH="0800......" Timeout="0"/> <!-- Apparently 0800 is always 10 digits -->
    <TEMPLATE MATCH="00*" Timeout="5"/> <!-- International, 00 prefixed. No fixed length -->
	<TEMPLATE MATCH="1.........." Timeout="0"/> <!-- Local numbers -->
    <TEMPLATE MATCH="0.........." Timeout="0"/> <!-- UK 11 digit, 0 prefixed -->
    <TEMPLATE MATCH="26...." Timeout="0"/> <!-- My local STD numbers start 26 -->
    <TEMPLATE MATCH="\*.." Timeout="0"/> <!-- Asterisk *.. codes -->
    <TEMPLATE MATCH="\*97...." Timeout="0"/> <!-- Asterisk direct VM access *981234-->
    <TEMPLATE MATCH="1..." Timeout="0"/> <!-- Internal numbers -->
    <TEMPLATE MATCH="2..." Timeout="0"/>  <!-- Internal numbers -->
    <TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>

you will need to save that as dialplan.xml

 

Hope that helps. 

Regards

Good Job
Thanks for your participation , it works fine ?
I did 88xx & 78xx & it works fine but could not do the DX650 
I was looking for such file, thanks for this feedback
Will try to upload it
 

Hi there,

 

No problem at all, happy to help. My knowledge is very limited but I am glad this could be of assistance. So far it's working great have not had any issues, my only grip is that I could not get the NTP options working so if you can help there that would be great! Here is an updated config that also allows for web access to the device as well as no PIN and it's also removed a lot of the stuff that applies to other models as well as if you are using CUCM.


<?xml version="1.0" encoding="UTF-8"?>
<device  xsi:type="axl:XIPPhone">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>YOUR USERNAME</sshUserId>
<sshPassword>YOUR PASSWORD</sshPassword>
<ipAddressMode>0</ipAddressMode>

<echoMultiEnable>false</echoMultiEnable>
<ipPreferenceModeControl>0</ipPreferenceModeControl>

<devicePool>
<revertPriority>0</revertPriority>
        <dateTimeSetting>
            <dateTemplate>D/M/YA</dateTemplate>
            <timeZone>US Eastern Standard Time</timeZone>
            <ntps>
                <ntp>
                    <name>ca.pool.ntp.org</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
<callManagerGroup>
<members>
<member  priority="0">
<callManager>
<name>YOUR NAME</name>
<description>SIP phone Connection</description>
<ports>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>YOUR PROVIDER ADDRESS</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<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>1</callHoldRingback>
<URIDialingDisplayPreference>1</URIDialingDisplayPreference>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>0</anonymousCallBlock>
<callerIdBlocking>0</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
</sipCallFeatures>


<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>60</timerRegisterExpires>
<timerRegisterDelta>0</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>


<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>

<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>

<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<poundEndOfDial>false</poundEndOfDial>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<sipLines>
<line  button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>Office</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>YOUR EXTENSION NAME</name>
<displayName>Office</displayName>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>YOUR EXTENSION</authName>
<authPassword>YOUR PASSWORD</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact></contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>

<maxNumCalls>10</maxNumCalls>
<busyTrigger>6</busyTrigger>
</line>

</sipLines>
<externalNumberMask>YOUR NUMBER</externalNumberMask>
<voipControlPort>5060</voipControlPort>
<ringSettingBusyStationPolicy>1</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>

<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
</sipProfile>
<MissedCallLoggingOption>10</MissedCallLoggingOption>
<commonProfile>

<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>

<loadInformation></loadInformation>

<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<allowBTContactImport>1</allowBTContactImport>
<allowBTMobileHandsfree>1</allowBTMobileHandsfree>
<recordingTone>0</recordingTone>
<settingsAccess>1</settingsAccess>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<deviceUIProfile>0</deviceUIProfile>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<garp>1</garp>
<multiUser>0</multiUser>
<requireScreenLock>0</requireScreenLock>
</vendorConfig>
<commonConfig>
<ciscoCamera>1</ciscoCamera>
<videoCapability>1</videoCapability>
<webProtocol>0</webProtocol>
<webAccess>0</webAccess>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>09:00</displayOnDuration>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<displayIdleTimeout>00:10</displayIdleTimeout>
<sshAccess>0</sshAccess>
<sendKeyAction>1</sendKeyAction>
<RingLocale>0</RingLocale>
<appInstallFromAndroidMarket>true</appInstallFromAndroidMarket>
</commonConfig>


<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version></version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>Canada</networkLocale>
<networkLocaleInfo>
<name>Canada</name>
<uid>64</uid>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>

<transportLayerProtocol>1</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>


<encrConfig>false</encrConfig>
<advertiseG722Codec>0</advertiseG722Codec>
<mobility>
<handoffdn>8888</handoffdn>
<dtmfdn>41200</dtmfdn>
<ivrdn>86547810</ivrdn>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<TLSResumptionTimer>0</TLSResumptionTimer>
<phoneServices  useHTTPS="true">
<provisioning>0</provisioning>
<phoneService  type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</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>
</phoneServices>
</device>

 

Hello Trinidad 

Thanks for the feed back 
I actually got the phone working fine .. but could not do video call ( although Video codec is enabled)
is there any in the XML file to do in order to enable it ? 


Hi there,

 

Super glad it helped you out. Depending on which XML file you used in my post the earlier one had the video disabled:

 

<ciscoCamera>1</ciscoCamera>
<videoCapability>1</videoCapability>

Just make sure both of those are "1" enabled. And you should be good to go. The original XML I posted has the second tag as "0" 
Hope that helps.  I have edited the original post so no one else runs into that. If you have success with the NTP settings please let me know I was not able to get it to work, maybe I missed something simple. 

Well Trinidad

I really appreciate your cooperation here

I thought that this could be the issue because I know the 0 means disable, but just wanted to make sure 

About NTP, I am using standard Arabia time, but I guess since this phone is Android, it can take the time from the the device itself, not from the xml file 

So far, I haven't installed this phone for the customer, I am just doing this in my lab

Within few days I will, & will update you

 

Cheers 😁

 

 

Hello friend 

I appreciate your cooperation with this issue 

I have 1 issue I wonder if you faced 

There's USB port on this phone, when I plug my iPhone it says disabled by administrator 

I looked for in the XML file for a line to enable it but No 

Can you help with that pls ?

 

 

 

 

 

 

 

 

Sure try these options:

 

usb1
Enable USB port 1.

0 Disabled 1 Enabled
<usb1>1</usb1>

usb2
Enable USB port 2.

0 Disabled 1 Enabled
<usb2>1</usb2>


usbClasses
Comma separated list of supported USB device classes.

0 Mass Storage 1 Human Interface Device 2 Audio Class
<usbClasses>0,1,2</usbClasses>

 

apologies for the late reply as well. hope this helps

this post I am sure has helped a lot of people with this phone. Just wanted to ask if you have found a way to also use BLF / speed dials for the Cisco DX 650 without going through the Cisco UCM

Glad it can come in useful. These are such neat phones but no one seems to have something posted on how to use them in a non cisco environment. So I'm super happy if it helps anyone.  To be honest I have not tried to get those functions working yet, I think there is a way if you are proficient enough with programming. But I have not had the time to play around with it yet. 

 

based on https://usecallmanager.nz/extensions-conf.html


they have noted some options, might be worth trying some of them out. 

 

blfpickup
Busy Lamp Field Directed Call Pickup, ${EXTEN:10} is the number specified as the <speedDialNumber> in a BLF speed-dial line key definition, also needs to have a matching subscribe entry in sip.conf.

exten => _blfpickup-.,1,PickupChan(SIP/${EXTEN:10}) same => next,Hangup(normal_circuit_congestion)

 

Thanks

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