SOLVED Cisco 7970 registers then periodically unregisters then reregisters.

[pebley]

Running PBXIAF purple Asterisk 1.8

I've run into an issue where all is fine on asterisk 1.6, but 1.8 is causing issues.... Only on some phones.

Circa 10 x Cisco 7970 - working
4 x Cisco 7970 - resetting
1 x Cisco 7975 - resetting

They all have same firmware and same configuration files, except different extensions. (Have swapped extensions round - to sort it out, so it is definately specific phones that have the issue).

The phones are fine then all of a sudden they reset their ip address (they are on poe - but it is not power off/on) this causes the phones to unregister then reregister, this can happen during a call or when idle only takes 10 seconds, but means these 5 phones cannot be used in production environment. Sometimes it happens every 5 mins.

Asterisk cli is not giving me any clues, have seen a similar post on Tribox forum (2009) but no solution was offered.

Any clues anyone?

(they are all on DHCP, just thought to try on Static but i would prefer to remain on DHCP).

 

[pebley]

Made static IP addresses - sorted the issue

Obviously a network issue as opposed to Asterisk.

Making IP addresses static solved the issue... strange how only a few of the phones reset and others were fine.

 

[ohrass]

did you get the 7970 to work through the endpoint manager? I need some help getting mine going. I have 7940 and 6960 working as well. To push my luck further, how about the conference station 7935 or 7936?

 

[pebley]

Cisco 7970 config for endpoint

Hope below helps
The only change you'll have to make will be the firmware no: SIP70.8-5-4S (You may want to change timezones etc, it is all self explanatory.)

If you are firmware upgrading you'll have to upgrade to 70.8-5-2S first then upto -4

I did have an issue with the password field so I have removed {$pass} and put hardcode in for my working systems.

The Endpoint team have been in contact with me about below, I guess they'll release the 7970 as an option once they have sorted out the other phones they are working on.

The only other phones I have used are 7975, 7971, 7940, 7960. But I assume below will work on your others.

<!--
#################PROVISIONER.NET#################
# This Configuration file was generated from the Provisioner.net Library by {$provisioner_processor_info}
# Generated on: {$provisioner_generated_timestamp}
#
# Provisioner Information Follows:
# Brand Revision Timestamp: {$provisioner_brand_timestamp}
# Family Revision Timestamp: {$provisioner_family_timestamp}
#
##################################################
-->
<device xsi:type="axl:XIPPhone" ctiid="203849429" uuid="">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>user</sshUserId>
<sshPassword>pass</sshPassword>
<devicePool uuid="">
<name>Dallas 5.0 Beta</name>
<dateTimeSetting uuid="">
<name>CMLocal</name>
<dateTemplate>D/M/Y</dateTemplate> #Enter Date format
<timeZone>GMT Standard/Daylight Time</timeZone> #Timezone
</dateTimeSetting>
<callManagerGroup>
<name>5.0 Beta</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>{$server.ip.1}</name>
<description>CallManager 5.0 Beta Pub - 5.0.1.032</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>{$server.ip.1}</processNodeName> # ENTER PBX IP ADDRESS HERE
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid="">
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1>{$server.ip.1}</ipAddr1> # ENTER PBX IP ADDRESS HERE
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1>{$server.ip.1}</sipIpAddr1> # ENTER PBX IP ADDRESS HERE
<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>
<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>true</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>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>Suzuki-Sales</phoneLabel> #Phone name in top right corner max 12 characters inc spaces
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>{$ext.line.1} {$displayname.line.1}</featureLabel> # ENTER NAME OF THE BUTTON LABEL HERE
<proxy>{$server.ip.1}</proxy> # ENTER PBX IP ADDRESS HERE
<port>5060</port>
<name>{$ext.line.1}</name>
<displayName>Pebley</displayName> # ENTER A NAME HERE DOES NOT SHOW UP THOUGH
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>{$ext.line.1}</authName> # ENTER PBX USERNAME AS SETUP ON ASTERISK
<authPassword>{$[FONT='Verdana','sans-serif']pass[/font]}</authPassword> # ENTER PBX PASSWORD AS SETUP ON ASTERISK
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>7b452e87-4496-4762-e11f-b26751a1884b</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
</line>
<line button="3">
<featureID>2</featureID>
<featureLabel>Ciren</featureLabel>
<speedDialNumber>901285641800</speedDialNumber>
</line>
<line button="4">
</line>
<line button="5">
<featureID>2</featureID>
<featureLabel>Doms Office</featureLabel>
<speedDialNumber>900</speedDialNumber>
</line>
<line button="6">
<featureID>2</featureID>
<featureLabel>Park Pickup 71</featureLabel>
<speedDialNumber>71</speedDialNumber>
</line>
<line button="7">
<featureID>2</featureID>
<featureLabel>Park call (trsfr)</featureLabel>
<speedDialNumber>70</speedDialNumber>
</line>
<line button="8">
<featureID>2</featureID>
<featureLabel>General Call Pickup</featureLabel>
<speedDialNumber>*8</speedDialNumber>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml</softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-5-4S</loadInformation> # ENTER SIP FIRMWARE VERSION NUMBER HERE MUST BE IN THE TFTPBOOT ROOT
<vendorConfig>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<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>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:00</displayOnDuration>
<displayIdleTimeout>00:20</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>
<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>
<userLocale>
<name>English_United_Kingdom</name>
<uid>1</uid>
<langCode>en_UK</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_Kingdom</networkLocale>
<networkLocaleInfo>
<name>United_Kingdom</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://{$server.ip.1}/cisco/services/authenticate.php</authenticationURL> #ENTER URL FOR ALL BELOW
<directoryURL>http://{$server.ip.1}/cisco/services/index_cisco.php</directoryURL>
<idleURL></idleURL>
<informationURL>http://{$server.ip.1}/cisco/services/help.php</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://{$server.ip.1}/cisco/services/PhoneServices.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>{$server.ip.1}</processNodeName> # ENTER PBX IP ADDRESS HERE
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>

 

[ohrass]

Thanks, I'll try this later on tonight

 

[Adrian_Novice]

many thanks Pebley, had just upgrade my 7970g phone from 8.0.3s and nothing was working (stuck registering), its now registered with freepbx, happy days ;-)

 

[Adrian_Novice]

Hi Pebley, know it is some time ago, but can't get call pickup (*8 just says 'reorder', **200 works, both phones in same call group=1) to work, tried, the following dial plans, but neither work;
simple:
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>

more complex:
<DIALTEMPLATE>
<TEMPLATE MATCH="1.." TIMEOUT="1"/><!-- Internal extensions 100 to 199. Wait 1 second, then dial -->
<TEMPLATE MATCH="2.." TIMEOUT="1"/><!-- Internal extensions 200 to 299. Wait 1 second, then dial -->
<TEMPLATE MATCH="3.." TIMEOUT="1"/><!-- Internal extensions 300 to 399. Wait 1 second, then dial -->
<TEMPLATE MATCH="4.." TIMEOUT="1"/><!-- Internal extensions 400 to 499. Wait 1 second, then dial -->
<TEMPLATE MATCH="5.." TIMEOUT="1"/><!-- Internal extensions 500 to 599. Wait 1 second, then dial -->
<TEMPLATE MATCH="6.." TIMEOUT="1"/><!-- Internal extensions 600 to 699. Wait 1 second, then dial -->
<TEMPLATE MATCH="*8" TIMEOUT="0"/><!-- *8 (*8 for group pickup). Dial immediately -->
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>

Could you post an 'edited' version of your dialplan.xml file please?
thanks, Adrian