Hi,
I have 2 cisco 7961 phones. I have been trying to get them to connect and register to my FreePBX server but they just are stuck in a registration loop and I don’t know why.
I’m using the latest version of FreePBX and can connect with my phone from another company. So now I want to connect my cisco phones.
I have found firmware and XML files online to get the phones setup. The firmware works and the phones don’t complain about the XML being invalid.
So I’m at a loss. Do cisco 7961 phones work with FreePBX? I saw some articles from years ago that said they got it to work so I assume it does and I’m missing a step.
For ref. here is my test XML config file (using simple test auth):
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>Password</sshPassword>
<ipAddressMode>0</ipAddressMode>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<olsonTimeZone>US/Eastern</olsonTimeZone>
<ntps>
<ntp>
<name>192.168.1.20</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.1.20</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<callForwardURI>x-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>true</natEnabled>
<phoneLabel>4</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>10000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Leann</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>4</name>
<displayName>Leann</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>4</authName>
<authPassword>04</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>4</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword>22222</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP41.9-2-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
</vendorConfig>
<versionStamp>002</versionStamp>
<networkLocale>United_Kingdom</networkLocale>
<networkLocaleInfo>
<name>United_Kingdom</name>
<uid>64</uid>
<version>1.0.0.0-4</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<servicesURL></servicesURL>
<directoryURL>http://192.168.1.20/directory.xml</directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<dialToneSetting>2</dialToneSetting>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<transportLayerProtocol>2</transportLayerProtocol>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
The log shows this line:sip:[email protected] :106 REGISTER::[email protected]
Yes, I have one setup and active. I am using it with chan_sip and the usecallmanager patch but that should not be an issue.
It is highly likely your XML file is wrong. One thing you must understand is that all of the Cisco phones will silently accept an XML file that is wrong and when the parser in the phone gets to whatever XML it does not like, the phone just stops processing the XML file, leaving the phone half-provisioned.
Note the following on the page SEPMAC.cnf.xml
"… transportLayerProtocol link
What protocol the phone will use to connect to Asterisk. Only use 1 (TCP) or 3 (TLS), as the phone causes SIP retransmit errors when using UDP…"
Besides that, go through that page, and verify all statements in your SIPMAC config file are correct.
miken32
March 24, 2025, 4:42pm
3
It’s been like 15 years since I’ve worked with one of these phones, but you need to have SIP-specific firmware loaded on them. Also that config file looks like it’s setting the phone up for a CCM server.
I got it to somewhat work.
it registers and works then goes offline. The FreePBX server shows it as connected but if I call it it goes straight to voicemail. Restarting the phone fixes it but then I have to constantly restart the phone.
It already has sip firmware.
Are you using chan_pjsip?
Are you using TCP or UDP transport?
Can you post the XML file you are using? (SEPMAC) I assume you modified it from the one you first posted.
TCP and no. I use PJSIP.
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<ipAddressMode>0</ipAddressMode>
<allowAutoConfig>true</allowAutoConfig>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>216.239.35.0</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.1.20</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<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>
<callHoldRingback>1</callHoldRingback>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>1</anonymousCallBlock>
<callerIdBlocking>0</callerIdBlocking>
<remoteCcEnable>true</remoteCcEnable>
<rfc2543Hold>false</rfc2543Hold>
<cnfJoinEnabled>true</cnfJoinEnabled>
<dndControl>0</dndControl>
<localCfwdEnable>true</localCfwdEnable>
<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>Phone</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>0</kpml>
<phoneLabel>Jeremy</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<startMediaPort>10000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<natEnabled>false</natEnabled>
<natReceivedProcessing>false</natReceivedProcessing>
<natAddress></natAddress>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>1</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>1</name>
<displayName>1</displayName>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
<autoAnswerMode>Auto Answer with Speakerphone</autoAnswerMode>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>1</authName>
<authPassword>01</authPassword>
<contact>1</contact>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>50</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
</sipLines>
<externalNumberMask></externalNumberMask>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<MissedCallLoggingOption>1</MissedCallLoggingOption>
<commonProfile>
<phonePassword>22222</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP41.9-2-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<enableMuteFeature>false</enableMuteFeature>
<enableGroupListen>true</enableGroupListen>
<g722CodecSupport>2</g722CodecSupport>
<handsetWidebandEnable>2</handsetWidebandEnable>
<headsetWidebandEnable>2</headsetWidebandEnable>
<headsetWidebandUIControl>1</headsetWidebandUIControl>
<handsetWidebandUIControl>1</handsetWidebandUIControl>
<settingsAccess>1</settingsAccess>
<videoCapability>0</videoCapability>
<webAccess>0</webAccess>
<webProtocol>0</webProtocol>
<sshAccess>0</sshAccess>
<displayRefreshRate>0</displayRefreshRate>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:00</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<recordingTone>0</recordingTone>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<moreKeyReversionTimer>5</moreKeyReversionTimer>
<autoSelectLineEnable>1</autoSelectLineEnable>
<autoCallSelect>1</autoCallSelect>
<minimumRingVolume>1</minimumRingVolume>
<peerFirmwareSharing>0</peerFirmwareSharing>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<rtcp>1</rtcp>
<garp>0</garp>
<ehookEnable>0</ehookEnable>
<pcPort>0</pcPort>
<spanToPCPort>1</spanToPCPort>
<voiceVlanAccess>0</voiceVlanAccess>
<enableCdpSwPort>1</enableCdpSwPort>
<enableCdpPcPort>0</enableCdpPcPort>
<enableLldpSwPort>1</enableLldpSwPort>
<enableLldpPcPort>0</enableLldpPcPort>
<eapAuthentication>1</eapAuthentication>
</vendorConfig>
<userLocale>
<name>English_United_States</name>
<uid></uid>
<langCode></langCode>
<version></version>
<winCharSet>utf-8</winCharSet>
</userLocale>
<networkLocale>English_United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<version>1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<!--<authenticationURL></authenticationURL>
<secureAuthenticationURL></secureAuthenticationURL>
<messagesURL></messagesURL>
<secureMessagesURL></secureMessagesURL>
<servicesURL></servicesURL>
<secureServicesURL></secureServicesURL>
<directoryURL></directoryURL>
<secureDirectoryURL></secureDirectoryURL>
<informationURL></informationURL>
<secureInformationURL></secureInformationURL>
<idleURL></idleURL>
<secureIdleURL></secureIdleURL>
<idleTimeout>0</idleTimeout>
<proxyServerURL></proxyServerURL>-->
<phoneServices useHTTPS="false">
<provisioning>2</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="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<transportLayerProtocol>1</transportLayerProtocol>
<phonePersonalization>1</phonePersonalization>
<autoCallPickupEnable>true</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<dndCallAlert>5</dndCallAlert>
<dndReminderTimer>50</dndReminderTimer>
<advertiseG722Codec>2</advertiseG722Codec>
<rollover>0</rollover>
<joinAcrossLines>0</joinAcrossLines>
<certHash></certHash>
<encrConfig>false</encrConfig>
<userId></userId>
<sshUserId>admin</sshUserId>
<sshPassword>22222</sshPassword>
</device>
Go to Asterisk SIP settings and chan_pjsip…did you switch this on?
1 Like
Yes. Without that turned on it will never connect.
You have a networking issue of some kind. Poor connectivity, bad ethernet switch, etc. How are you powering the phone, PoE or via adapter? These phones suck a lot of power, I have one of them and i power it from an external AC adapter. Maybe your switch is not happy with supplying power to it, or maybe your switch is not properly autodetecting the port, etc.
I have had the exact same problem in the past with the 79xx series. But in my case it was caused by too slow CPUs in a LAN2LAN vpn and, I think, due to problems with MTU Path Discovery. I extensively documented this in the DD-WRT forum years ago, I can dig it out if you are interested. The situation was that I have 2 sites about 100 miles apart, and 2 routers running an openVPN LAN2LAN VPN connecting the two, a FreePBX system at one site and phones at the other, including a 7961.
I had a lot of problems with VoIP phones losing registration and it took a number of years of fiddling with it before I got it dialed in. I had to increase CPU power in the VPN routers as well as make configuration changes in MSS and all kinds of other things including replacing the ethernet switch the phones were running off of.
Note that this problem also affected the Cisco UCM, see:
Solved: Cisco 7941 ip phone registers then after awhile it loses connectivity with the CUCM8.6 fallback to S… - Cisco Community
Now in fairness I also use chan_sip with the usecallmanager patch but I was having the same problem with Polycom phones losing registration over the VPN as well as a RCA voip phone (I’ll bet you never knew RCA actually manufactured VoIP desk telephones once) and those were going into chan_pjsip
Anyway, here’s my config if you want to review it but I don’t think the differences are significant:
$ cat SEP001F6C81328D.cnf.xml.CP7961-plus-7914-sidecar
you have mail
<?xml version="1.0" encoding="UTF-8"?>
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<ipAddressMode>0</ipAddressMode>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>172.16.1.1</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<name>172.16.1.16</name>
<description>FreePBX</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>172.16.1.16</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<commonProfile>
<phonePassword>cisco</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation model="IP Phone 7961">SIP41.9-2-1S</loadInformation>
<addOnModules>
<addOnModule idx="1">
<loadInformation model="Addon 7914">S00105000400</loadInformation>
</addOnModule>
</addOnModules>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<g722CodecSupport>2</g722CodecSupport>
<handsetWidebandEnable>1</handsetWidebandEnable>
<headsetWidebandEnable>0</headsetWidebandEnable>
<headsetWidebandUIControl>0</headsetWidebandUIControl>
<handsetWidebandUIControl>0</handsetWidebandUIControl>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<daysDisplayNotActive></daysDisplayNotActive>
<displayOnTime></displayOnTime>
<displayOnDuration></displayOnDuration>
<displayIdleTimeout>00:05</displayIdleTimeout>
<webAccess>0</webAccess>
<spanToPCPort>0</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<loadServer></loadServer>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
</vendorConfig>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>utf-8</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directory></directory>
<idleTimeout>0</idleTimeout>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>1</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></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>
</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>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>softkeys.xml</softKeyFile>
<phoneLabel>771</phoneLabel>
<natEnabled>true</natEnabled>
<natAddress></natAddress>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>771 Line 1</featureLabel>
<name>771</name>
<displayName>771</displayName>
<contact>771</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>771</authName>
<authPassword>abbfed</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*98</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>21</featureID>
<featureLabel>801 Office</featureLabel>
<speedDialNumber>801</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="3">
<featureID>21</featureID>
<featureLabel>802 Office Sec</featureLabel>
<speedDialNumber>802</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="4">
<featureID>21</featureID>
<featureLabel>803 Laptop</featureLabel>
<speedDialNumber>803</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="5">
<featureID>21</featureID>
<featureLabel>805 PCS</featureLabel>
<speedDialNumber>805</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="6">
<featureID>21</featureID>
<featureLabel>806 Sales</featureLabel>
<speedDialNumber>806</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="7">
<featureID>21</featureID>
<featureLabel>821 Master BR</featureLabel>
<speedDialNumber>821</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
</sipLines>
</sipProfile>
</device>
$
