Unable to register more than two phones

used this script for the Cisco 7821 phone. Two phones registered successfully, but the third phone is not registering. Could the problem be in the script or in FreePBX?

The second issue is that I want to change the extension of a registered phone to another extension. However, no matter how many times I reset the phone, it no longer works with the old extension or the new one.
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SIP tester tester M/D/YY Afghanistan Standard Time pool.ntp.org unicast 5060 5061 172.30.xx.xxx true true x-serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 1 6 10 180 3600 5 120 120 5 500 4000 70 true Phone 1 false true false false 101 3 avt false false 3 MYPLABEL 0 false 10 false 9 HR USECALLMANAGER 5060 101 HR 0 3 101 abc123 false 1 *97 4 5 101 true false false true 5060 16348 20134 184 0 dialplan.xml true 2 sip78xx.14-2-1-0001-14 false false 0 1 0 0 0 0 1 1,2,3,4,5,6,7 00:00 00:00 00:10 1 1 en_US 1.0.0.0-1 iso-8859-1 1.0.0.0-1 1 96 0 96 4 0 3804 2 Missed Calls Application:Cisco/MissedCalls Received Calls Application:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Voicemail Application:Cisco/Voicemail false

Please encapsulate scripts or code of any kind in the </> highlight the test then click the button in the edit bar to do that. Otherwise what you have is unreadable.

Are you registering to chan_pjsip or are you using the Patching Asterisk usecallmager patch?

```
<device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>tester</sshUserId>
    <sshPassword>tester</sshPassword>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>M/D/YY</dateTemplate>
            <timeZone>Afghanistan Standard Time</timeZone>
            <ntps>
                <ntp>
                    <name>pool.ntp.org</name>
                    <ntpMode>unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <members>
                <member priority="0">
                    <callManager>
                        <ports>
                            <sipPort>5060</sipPort>
                            <securedSipPort>5061</securedSipPort>
                        </ports>
                        <processNodeName>172.30.xx.xxx</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
    </devicePool>
    <sipProfile>
        <sipProxies>
            <registerWithProxy>true</registerWithProxy>
        </sipProxies>
        <sipCallFeatures>
            <cnfJoinEnabled>true</cnfJoinEnabled>
            <callForwardURI>x-serviceuri-cfwdall</callForwardURI>
            <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
            <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
            <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
            <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
            <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
            <rfc2543Hold>false</rfc2543Hold>
            <callHoldRingback>2</callHoldRingback>
            <localCfwdEnable>true</localCfwdEnable>
            <semiAttendedTransfer>true</semiAttendedTransfer>
            <anonymousCallBlock>2</anonymousCallBlock>
            <callerIdBlocking>2</callerIdBlocking>
            <dndControl>0</dndControl>
            <remoteCcEnable>true</remoteCcEnable>
            <uriDialingDisplayPreference>1</uriDialingDisplayPreference>
        </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>false</transferOnhookEnabled>
        <enableVad>false</enableVad>
        <dtmfAvtPayload>101</dtmfAvtPayload>
        <dtmfDbLevel>3</dtmfDbLevel>
        <dtmfOutofBand>avt</dtmfOutofBand>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <kpml>3</kpml>
        <phoneLabel>MYPLABEL</phoneLabel>
        <stutterMsgWaiting>0</stutterMsgWaiting>
        <callStats>false</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
        <sipLines>
            <line button="1">
                <featureID>9</featureID>
                <featureLabel>HR</featureLabel>
                <proxy>USECALLMANAGER</proxy>
                <port>5060</port>
                <name>101</name>
                <displayName>HR</displayName>
                <autoAnswer>
                    <autoAnswerEnabled>0</autoAnswerEnabled>
                </autoAnswer>
                <callWaiting>3</callWaiting>
                <authName>101</authName>
                <authPassword>abc123</authPassword>
                <sharedLine>false</sharedLine>
                <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
                <messagesNumber>*97</messagesNumber>
                <ringSettingIdle>4</ringSettingIdle>
                <ringSettingActive>5</ringSettingActive>
                <contact>101</contact>
                <forwardCallInfoDisplay>
                    <callerName>true</callerName>
                    <callerNumber>false</callerNumber>
                    <redirectedNumber>false</redirectedNumber>
                    <dialedNumber>true</dialedNumber>
                </forwardCallInfoDisplay>
            </line>
        </sipLines>
        <voipControlPort>5060</voipControlPort>
        <startMediaPort>16348</startMediaPort>
        <stopMediaPort>20134</stopMediaPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>dialplan.xml</dialTemplate>
        <softKeyFile></softKeyFile>
    </sipProfile>
    <commonProfile>
        <phonePassword></phonePassword>
        <backgroundImageAccess>true</backgroundImageAccess>
        <callLogBlfEnabled>2</callLogBlfEnabled>
    </commonProfile>
    <loadInformation>sip78xx.14-2-1-0001-14</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>1</webAccess>
        <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
        <displayOnTime>00:00</displayOnTime>
        <displayOnDuration>00:00</displayOnDuration>
        <displayIdleTimeout>00:10</displayIdleTimeout>
        <spanToPCPort>1</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <loadServer></loadServer>
    </vendorConfig>
    <userLocale>
        <name></name>
        <uid></uid>
        <langCode>en_US</langCode>
        <version>1.0.0.0-1</version>
        <winCharSet>iso-8859-1</winCharSet>
    </userLocale>
    <networkLocale></networkLocale>
    <networkLocaleInfo>
        <name></name>
        <uid></uid>
        <version>1.0.0.0-1</version>
    </networkLocaleInfo>
    <deviceSecurityMode>1</deviceSecurityMode>
    <authenticationURL></authenticationURL>
    <directoryURL></directoryURL>
    <servicesURL></servicesURL>
    <idleURL></idleURL>
    <informationURL></informationURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <transportLayerProtocol>4</transportLayerProtocol>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
        <capf>
            <phonePort>3804</phonePort>
        </capf>
    </capfList>
    <phoneServices useHTTPS="false">
        <provisioning>2</provisioning>
        <phoneService type="1" category="0">
            <name>Missed Calls</name>
            <url>Application:Cisco/MissedCalls</url>
            <vendor />
            <version />
        </phoneService>
        <phoneService type="1" category="0">
            <name>Received Calls</name>
            <url>Application:Cisco/ReceivedCalls</url>
            <vendor />
            <version />
        </phoneService>
        <phoneService type="1" category="0">
            <name>Placed Calls</name>
            <url>Application:Cisco/PlacedCalls</url>
            <vendor />
            <version />
        </phoneService>
        <phoneService type="2" category="0">
            <name>Voicemail</name>
            <url>Application:Cisco/Voicemail</url>
            <vendor />
            <version />
        </phoneService>
    </phoneServices>
    <certHash></certHash>
    <encrConfig>false</encrConfig>
</device>
```

The config file for the 7821 isn’t in general going to be a problem - did he set the max extensions high enough - in chan_pjsip - if he’s using that…

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