QUESTION Cisco 7975 and 7961 config in elastix 2.5

[Carlos Barillas]

Hi, hi have a cisco 7961 and 7975 phone, i want so use it with elastix 2.5, i had about two weeks trying to make it work. the firmware im using is "SIP41.9-4-2-1S" and "SIP75.9-4-2-1S" y used olders versions same result. i made hard reset on it.
The phone keeps on REGISTERING. Im using Tftpd64 and also capturing PC port using wireshark
in the wireshark capturing i see the phone stuck on ntp server. or not make any server contact. i tried to use "pool.ntp.org" in the <name>xxxxxxxx</name> no result, also create a windows ntp server from https://www.meinbergglobal.com/english/ ... ntp_stable no result again. also tried to use same addres of elastix server no result too.

Elastix sip line qualify: yes Nat: No

i have to install somenting else on elastix? or inly create the sip extencion its eought.

the SEP file is loaded well.
THE IP OF SERVER IS 192.2.240
THE IP OF THE NTP SERVER IS 192.168.2.103
i will post all the logs here. any help will be apreciate, thanks.
(no posting elastix log because i see nothing there. i thing the phone is not conctacting the server.)

 

[Carlos Barillas]

TFTPD64 LOG:
--------------------------------------------------------------------------------------------------
Connection received from 192.168.2.106 on port 50408 [22/09 22:09:55.999]
Read request for file <CTLSEP001E4A0C5375.tlv>. Mode octet [22/09 22:09:55.999]
File <CTLSEP001E4A0C5375.tlv> : error 2 in system call CreateFile The system cannot find the file specified. [22/09 22:09:55.999]
Connection received from 192.168.2.106 on port 50030 [22/09 22:09:56.168]
Read request for file <ITLSEP001E4A0C5375.tlv>. Mode octet [22/09 22:09:56.169]
File <ITLSEP001E4A0C5375.tlv> : error 2 in system call CreateFile The system cannot find the file specified. [22/09 22:09:56.170]
Connection received from 192.168.2.106 on port 53088 [22/09 22:09:56.388]
Read request for file <ITLFile.tlv>. Mode octet [22/09 22:09:56.388]
File <ITLFile.tlv> : error 2 in system call CreateFile The system cannot find the file specified. [22/09 22:09:56.388]
Connection received from 192.168.2.106 on port 51795 [22/09 22:09:56.733]
Read request for file <SEP001E4A0C5375.cnf.xml>. Mode octet [22/09 22:09:56.733]
Using local port 60022 [22/09 22:09:56.733]
<SEP001E4A0C5375.cnf.xml>: sent 22 blks, 10807 bytes in 0 s. 0 blk resent [22/09 22:09:56.799]
Connection received from 192.168.2.106 on port 51619 [22/09 22:10:12.017]
Read request for file <English_United_States/mk-sip.jar>. Mode octet [22/09 22:10:12.017]
File <English_United_States\mk-sip.jar> : error 3 in system call CreateFile The system cannot find the path specified. [22/09 22:10:12.017]
Connection received from 192.168.2.106 on port 51905 [22/09 22:10:13.374]
Read request for file <Venezuela/g3-tones.xml>. Mode octet [22/09 22:10:13.375]
File <Venezuela\g3-tones.xml> : error 3 in system call CreateFile The system cannot find the path specified. [22/09 22:10:13.375]
Connection received from 192.168.2.106 on port 49236 [22/09 22:10:14.937]
Read request for file <dialplan.xml>. Mode octet [22/09 22:10:14.937]
Using local port 60055 [22/09 22:10:14.938]
<dialplan.xml>: sent 1 blk, 249 bytes in 0 s. 0 blk resent [22/09 22:10:14.941]
Connection received from 192.168.2.106 on port 51633 [22/09 22:10:15.220]
Read request for file <softkeys.xml>. Mode octet [22/09 22:10:15.220]
Using local port 60056 [22/09 22:10:15.220]
<softkeys.xml>: sent 12 blks, 5824 bytes in 0 s. 0 blk resent [22/09 22:10:15.268]
--------------------------------------------------------------------------------------------------
WIRESHARK LOG:
--------------------------------------------------------------------------------------------------
1347 399.720545000 192.168.2.106 192.168.2.103 NTP 90 NTP Version 3, client
1348 399.728047000 192.168.2.103 192.168.2.106 NTP 90 NTP Version 3, server
THEESE LINES ARE REPEATING 4 TIMES.
--------------------------------------------------------------------------------------------------

 

[Carlos Barillas]

SEPMAC.CNF.XML FILE:
--------------------------------------------------------------------------------------------------

<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>192.168.2.103</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<name>192.168.2.240</name>
<description>elastix</description>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.2.240</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<sipProfile>
<sipProxies>
<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>
<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>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>103</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<natEnabled>false</natEnabled>
<natReceivedProcessing>false</natReceivedProcessing>
<natAddress></natAddress>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>103</featureLabel>
<proxy>192.168.2.240</proxy>
<port>5060</port>
<name>103</name>
<displayName>103</displayName>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>103</authName>
<authPassword>ext103aa</authPassword>
<contact>17</contact>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>5</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
</sipLines>
<externalNumberMask></externalNumberMask>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>softkeys.xml</softKeyFile>
</sipProfile>
<missedCallLoggingOption>1</missedCallLoggingOption>
<featurePolicyFile>featurepolicy.xml</featurePolicyFile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<vendorConfig>
<defaultWallpaperFile></defaultWallpaperFile>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<enableMuteFeature>false</enableMuteFeature>
<forwardingDelay>1</forwardingDelay>
<pcPort>0</pcPort>
<spanToPCPort>0</spanToPCPort>
<garp>0</garp>
<rtcp>1</rtcp>
<videoRtcp>1</videoRtcp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>1</videoCapability>
<hideVideoByDefault>0</hideVideoByDefault>
<separateMute>0</separateMute>
<ciscoCamera>1</ciscoCamera>
<usb1>1</usb1>
<usb2>1</usb2>
<usbClasses>0,1,2</usbClasses>
<sdio>1</sdio>
<wifi>1</wifi>
<bluetooth>1</bluetooth>
<bluetoothProfile>0,1</bluetoothProfile>
<btpbap>0</btpbap>
<bthfu>0</bthfu>
<ehookEnable>0</ehookEnable>
<autoSelectLineEnable>1</autoSelectLineEnable>
<autoCallSelect>1</autoCallSelect>
<incomingCallToastTimer>10</incomingCallToastTimer>
<dialToneFromReleaseKey>0</dialToneFromReleaseKey>
<joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
<minimumRingVolume></minimumRingVolume>
<simplifiedNewCall>0</simplifiedNewCall>
<showCallHistoryForSelectedLine>0</showCallHistoryForSelectedLine>
<kemOneColumn>0</kemOneColumn>
<allCallsOnPrimary>0</allCallsOnPrimary>
<softKeyControl>0</softKeyControl>
<settingsAccess>1</settingsAccess>
<webProtocol>0</webProtocol>
<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<g722CodecSupport>1</g722CodecSupport>
<handsetWidebandEnable>2</handsetWidebandEnable>
<headsetWidebandEnable>2</headsetWidebandEnable>
<headsetWidebandUIControl>1</headsetWidebandUIControl>
<handsetWidebandUIControl>1</handsetWidebandUIControl>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:00</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<daysBacklightNotActive>1,7</daysBacklightNotActive>
<backlightOnTime>08:00</backlightOnTime>
<backlightOnDuration>10:00</backlightOnDuration>
<backlightIdleTimeout>00:10</backlightIdleTimeout>
<backlightOnWhenIncomingCall>1</backlightOnWhenIncomingCall>
<recordingTone>0</recordingTone>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<moreKeyReversionTimer>5</moreKeyReversionTimer>
<peerFirmwareSharing>0</peerFirmwareSharing>
<enableCdpSwPort>1</enableCdpSwPort>
<enableCdpPcPort>0</enableCdpPcPort>
<enableLldpSwPort>1</enableLldpSwPort>
<enableLldpPcPort>0</enableLldpPcPort>
</vendorConfig>
<versionStamp>d902ed5a-c1e5-4233-b1d6-a960d53d1c3a</versionStamp>
<loadInformation>SIP41.9-4-2-1S</loadInformation>
<addOnModules>
<!-- <addOnModule idx="1">
<deviceType></deviceType>
<deviceLine></deviceLine>
<loadInformation></loadInformation>
</addOnModule> -->
</addOnModules>
<phoneServices>
<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>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode></langCode>
<version></version>
<winCharSet></winCharSet>
</userLocale>
<networkLocale>Venezuela</networkLocale>
<networkLocaleInfo>
<name>Venezuela</name>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>l</servicesURL>
<transportLayerProtocol>4</transportLayerProtocol>
<phonePersonalization>1</phonePersonalization>
<autoCallPickupEnable>true</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<dndCallAlert>1</dndCallAlert>
<dndReminderTimer>5</dndReminderTimer>
<advertiseG722Codec>1</advertiseG722Codec>
<rollover>0</rollover>
<joinAcrossLines>0</joinAcrossLines>
<capfAuthMode>0</capfAuthMode>
<capfList></capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sshUserId></sshUserId>
<sshPassword></sshPassword>
</device>

 

[cbunt1]

When I was fighting with my Cisco 7961 phone on various installs of IncrediblePBX 12, I found
If your elastix server is running core 12.0.39, you can add the following statement to the pjsip.endpoint_custom_post.conf: (assuming it's extension 103)

[103](+)
force_rport=no

I fought my 7961 for days with PJSIP before I came across the bugfix. It has to do with the way these phones handle their ports (Cisco calls it a "feature" and quotes security). I'm not sure it's limited to this series of phones though.

This is documented in a bug report/fix at http://issues.freepbx.org/browse/FREEPBX-9810. In short, the core release 12.0.39 specifically addresses this behavior and adds the custom_post.conf file behavior to the PJSIP configuration.

It made my 7961 come right up and stay up. I'm oversimplifying right now, but it should get you down the road. If you're not on 12.0.39, you can add the force_rport=no statement directly to your pjsip.endpoint.conf file, but understand that it'll get wiped out when the file is regenerated (like updating an extension) -- but it would provide a temporary fix and test opportunity.

If you're interested, I'll try to put together something a bit more detailed (although others described and solved the problem much better than I ever could), but hopefully this will save you some research and hassle--to me it was a pretty obscure glitch, and it drove me nuts--especially since I could make the phone work on SIP but not PJSIP.

I hope this helps. I know it's a little late (thread's about six weeks old), but I just saw it (I'm pretty new to this forum).