@jeez
Yes, it was the case, I noticed it right after I copied the log file.
Ownership is whatever the default sets it as.
Ok, upgrade completed.
It says unprovisioned in the bottom left and it is looking for a file that I don’t recall creating so I need to go back and see how I missed it.
Mar 3 22:21:50 localhost in.tftpd[2050]: RRQ from 192.168.1.129 filename CTLSEPMACADDRESS.tlv Mar 3 22:21:50 localhost in.tftpd[2050]: sending NAK (1, File not found) to 192.168.1.129 Mar 3 22:21:50 localhost in.tftpd[2051]: RRQ from 192.168.1.129 filename SEPMACADDRESS.cnf.xml
I don’t recall a .tlv file
The only thing regarding version in that file is this:
English_United_States 1 en_US 1.0.0.0-1 iso-8859-1 United_States United_States 64 1.0.0.0-1
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
I am going to try that tomorrow.
The tlv file is not needed, it is a security file for some type of encrypted communications.
There is no such thing as default ownership of files, you need to be aware of what user owns a file and who has read and write privs or you will beat your head against a wall trying to troubleshoot.
You are getting close, do you have NAT set to yes in your extension (a Cisco oddity)? Is the phone trying to register with the server? Go into Asteisk and do a SIP debug.
What does the SIP LINE section of your config look like?
The forum was having an issue this morning, I don’t see this posted in the thread (maybe they recovered from a backup?), but I did see this in my email:
| New comment: | Author: SkykingOH | Title: The tlv file is not needed,The tlv file is not needed, it is a security file for some type of encrypted
communications.There is no such thing as default ownership of files, you need to be aware of
what user owns a file and who has read and write privs or you will beat your
head against a wall trying to troubleshoot.You are getting close, do you have NAT set to yes in your extension (a Cisco
oddity)? Is the phone trying to register with the server? Go into Asteisk
and do a SIP debug.What does the SIP LINE section of your config look like?
I am sure some of the options in my config are not right, here is some more info from the phone log (IP of the phone in my browser).
I will check the config when I have some extra time, a little tied up right now.
5:53:09a SEPMACADDRESS.cnf.xml 5:53:09a Error Verifying Config Info 5:53:40a File Not Found : CTLFile.tlv 5:53:40a No CTL installed 5:53:40a SEPMACADDRESS.cnf.xml 5:53:40a Error Verifying Config Info 5:54:11a File Not Found : CTLFile.tlv 5:54:11a No CTL installed 5:54:11a SEPMACADDRESS.cnf.xml 5:54:11a Error Verifying Config Info
As far as the permissions, what I was getting at was…whatever the defaults are. If the default is for root to have access to everything, that is where I am at, I haven’t modified any permission.
You still don’t get permissions. Each file has it’s own set of permissions. You set them with chmod
You can see them with the long option of ls -l
I understand what you are asking. All I am saying is that (as of now) I haven’t done anything with permissions.
It appears to be off (in the config file).
false
And it was off in the FreePBX extension page, but I turned it on and applied the changes.
I can’t find anything that says SIP Line, I will have to take a closer look later on.
I tried to follow this, to do a SIP Debug, and I kept getting an error, I will also have to try that again, later.
The config file has a lot of options that I am not sure about, I copied it from this page (and changed a few things): http://www.pbxinaflash.com/community/index.php?threads/cisco-7970-7960-configuration-working-4-2012.12865/
here are the permissions of all the files in /tftpboot.
[root@localhost ~]# cd /tftpboot [root@localhost tftpboot]# ls -l total 8180 -rw-r--r-- 1 root root 2082 Jan 16 2012 ahh.pcm -rw-r--r-- 1 root root 2486603 May 22 2007 apps70.8-3-0-50.sbn -rw-r--r-- 1 root root 5040 Jan 16 2012 AreYouThereF.raw -rw-r--r-- 1 root root 5280 Jan 16 2012 AreYouThere.raw -rw-r--r-- 1 root root 12090 Jan 16 2012 asleep.raw -rw-r--r-- 1 root root 7407 Jan 16 2012 caramba.raw -rw-r--r-- 1 root root 10800 Jan 16 2012 ClockShop.raw -rw-r--r-- 1 root root 518495 May 22 2007 cnu70.8-3-0-50.sbn -rw-r--r-- 1 root root 16080 Jan 16 2012 CTU24raw.raw -rw-r--r-- 1 root root 12175 Jan 16 2012 Curley.raw -rw-r--r-- 1 root root 2711232 May 22 2007 cvm70sip.8-3-0-50.sbn -rw-r--r-- 1 root root 70 Mar 3 20:09 dialplan.xml -rw-r--r-- 1 root root 3446 Mar 3 20:08 distinctiveringlist.xml -rw-r--r-- 1 root root 2034 Jan 16 2012 doh.pcm -rw-r--r-- 1 root root 9600 Jan 16 2012 Drums1.raw -rw-r--r-- 1 root root 13440 Jan 16 2012 Drums2.raw -rw-r--r-- 1 root root 530417 May 22 2007 dsp70.8-3-0-50.sbn -rw-r--r-- 1 root root 15840 Jan 16 2012 FilmScore.raw -rw-r--r-- 1 root root 15840 Jan 16 2012 FlintPhone.raw -rw-r--r-- 1 root root 16080 Jan 16 2012 HarpSynth.raw -rw-r--r-- 1 root root 8160 Jan 16 2012 Jamaica.raw -rw-r--r-- 1 root root 540281 May 22 2007 Jar70sip.8-3-0-50.sbn -rw-r--r-- 1 root root 15360 Jan 16 2012 Klaxons.raw -rw-r--r-- 1 root root 16080 Jan 16 2012 KotoEffect.raw -rw-r--r-- 1 root root 114 Mar 3 20:08 List.xml -rw-r--r-- 1 root root 11909 Jan 16 2012 mayihelp.raw -rw-r--r-- 1 root root 9570 Jan 16 2012 merlin2.pcm -rw-r--r-- 1 root root 9604 Jan 16 2012 merlin3.pcm -rw-r--r-- 1 root root 9553 Jan 16 2012 merlin4.pcm -rw-r--r-- 1 root root 9627 Jan 16 2012 merlin5.pcm -rw-r--r-- 1 root root 9604 Jan 16 2012 merlin6.pcm -rw-r--r-- 1 root root 9605 Jan 16 2012 merlin7.pcm -rw-r--r-- 1 root root 12720 Jan 16 2012 MusicBox.raw -rw-r--r-- 1 root root 16080 Jan 16 2012 Neuro.raw -rw-r--r-- 1 root root 13421 Jan 16 2012 Ohno.raw -rw-r--r-- 1 root root 12 Mar 3 20:06 OS7970.TXT -rw-r--r-- 1 asterisk asterisk 14 Mar 3 20:13 OS79XX.TXT -rw-r--r-- 1 asterisk asterisk 130552 Mar 3 20:13 P003-8-12-00.bin -rw-r--r-- 1 asterisk asterisk 130956 Mar 3 20:13 P003-8-12-00.sbn -rw-r--r-- 1 asterisk asterisk 458 Mar 3 20:13 P0S3-8-12-00.loads -rw-r--r-- 1 asterisk asterisk 756524 Mar 3 20:13 P0S3-8-12-00.sb2 -rw-r--r-- 1 root root 8160 Jan 16 2012 Piano1.raw -rw-r--r-- 1 root root 15360 Jan 16 2012 Piano2.raw -rw-r--r-- 1 root root 9360 Jan 16 2012 Pop.raw -rw-r--r-- 1 root root 7200 Jan 16 2012 Pulse1.raw -rw-r--r-- 1 root root 16080 Jan 16 2012 ringer1.pcm -rw-r--r-- 1 root root 8160 Jan 16 2012 ringer2.pcm -rw-r--r-- 1 root root 8160 Jan 16 2012 ringer3.pcm -rw-r--r-- 1 root root 16040 Jan 16 2012 ringer4.pcm -rw-r--r-- 1 root root 8160 Jan 16 2012 ringer6.pcm -rw-r--r-- 1 root root 196 Mar 3 20:05 RINGLIST.dat -rw-r--r-- 1 root root 3446 Mar 3 20:05 ringlist.xml -rw-r--r-- 1 root root 10800 Jan 16 2012 Sax1.raw -rw-r--r-- 1 root root 14160 Jan 16 2012 Sax2.raw -rw-r--r-- 1 root root 7762 Mar 4 17:34 SEP00192FEEA5E0.cnf.xml -rw-r--r-- 1 root root 638 May 22 2007 SIP70.8-3-1S.loads -rw-r--r-- 1 root root 642 May 22 2007 term70.default.loads -rw-r--r-- 1 root root 642 May 22 2007 term71.default.loads -rw-r--r-- 1 root root 1108 Mar 3 20:09 XMLDefault.cnf.xml [root@localhost tftpboot]#
(forgot to blank out the MAD address above, not a big deal I suppose)
I can’t find any information on the SIP LINE in my config.
Also, I am not sure how to do a SIP debug in asterisk.
The MAC address, that’s not an issue. If you don’t know how do something go poking around. Hit sip ? in Asterisk
I will tell you that ‘core set verbose 0’ turns off the dialplan debug so you can see the sip debug.
When you say ‘hit sip ? in asterisk’ are you saying this is something I type while connected via SSH?
‘command not found’
here are my settings in freepbx for the cisco phone, ext 1000
Here is the SEP cml conf file i am using. I only changed what I thought needed to be changed after reading what values I needed to change. I was going to clean it up with extensions/labels that I didn’t need after a successful registration. According to him, this file worked.
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>64.90.182.55</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>192.168.1.15</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>0</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-3-1S</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>
<daysDisplayNotActive></daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>23:59</displayOnDuration>
<displayIdleTimeout>00:05</displayIdleTimeout>
<webAccess>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL>http:// 192.168.1.15/cisco/xml/authentication.php</authenticationURL>
<directoryURL>http:// 192.168.1.15/cisco/xml/PhoneDirectory.php</directoryURL>
<idleTimeout>0</idleTimeout>
<idleURL></idleURL>
<informationURL>http:// 192.168.1.15/cisco/xml/index.php</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http:// 192.168.1.15/cisco/xml/index.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<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--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>true</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>0</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>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g729a</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>100000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>1000</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>1000</featureLabel>
<name>1000</name>
<displayName>1000</displayName>
<contact>1000</contact>
<proxy>192.168.1.15</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>1000</authName>
<authPassword>ciscophone77</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</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>9</featureID>
<featureLabel>Ross Desk Intercom ext. 0215</featureLabel>
<name>0215</name>
<displayName>Ross Desk Intercom ext. 0215</displayName>
<contact>0215</contact>
<proxy>192.168.1.15</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>1</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>0215</authName>
<authPassword>ciscophone77</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="5">
<featureID>2</featureID>
<featureLabel>Buddy Cell</featureLabel>
<speedDialNumber>843xxxxxxx</speedDialNumber>
</line>
<line button="6">
<featureID>2</featureID>
<featureLabel>Marie Cell</featureLabel>
<speedDialNumber>843xxxxxxx/speedDialNumber>
</line>
<line button="7">
<featureID>2</featureID>
<featureLabel>Hilton AT&T NOC</featureLabel>
<speedDialNumber>xxxxxxxxxx</speedDialNumber>
</line>
<line button="8">
<featureID>2</featureID>
<featureLabel>Pickup</featureLabel>
<speedDialNumber>*8</speedDialNumber>
</line>
</sipLines>
</sipProfile>
</device>
</blockquote>- sep.cnf.xml
I just realized my config did not show up properly.
Yeah, the new forum seems to be broke on code tags, highly annoying. Another thing on the to do list.
There is the link to the conf file in the /tftpboot directory.
(good for 1 month)
So where are you with this? After reading the post I am confused as hell and don’t get what the real problem was or is. Since I had read the whole thing, I am very curious.