Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

arriben
Joined: Tue 17 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Thu 02 of Sep, 2010 (09:54 UTC)
d_turburville, did you see my log files posted on here?
Thanks..
arriben
Joined: Tue 17 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Mon 30 of Aug, 2010 (06:36 UTC)
Here are the log files from tft32.

I assinged an IP of 192.168.1.1 to my LT and the DHCP side in the laptop. You can clearly see that the LT is talking to the phone by picking up the handsets MAC address, but that is about as far as it goes.

The IP the phone is being assinged is 192.168.1.4 from the range of 5 that I setup on the DHCP spool. So in a nutshell, my LT is 192.168.1.2, router 192.168.1.1 and the handset 192.168.1.4 all of which can be pinged wiithout a problem.

I am stating to go insance over this as I just cannot get tftd32 to see the actual handset as shown on the log files below..

Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:28:45.564]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:29:01.439]
Rcvd DHCP Discover Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:29:33.221]
Suppress pingable address 192.168.1.1 [30/08 08:29:33.221]
Suppress pingable address 192.168.1.2 [30/08 08:29:36.396]
Suppress pingable address 192.168.1.3 [30/08 08:29:39.556]
Suppress pingable address 192.168.1.4 [30/08 08:29:42.715]
Suppress pingable address 192.168.1.5 [30/08 08:29:45.891]
no more address or address previously allocated by another server [30/08 08:29:45.891]
Rcvd DHCP Discover Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:29:45.953]
no more address or address previously allocated by another server [30/08 08:29:45.953]
Rcvd DHCP Discover Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:29:46.015]
no more address or address previously allocated by another server [30/08 08:29:46.015]
Rcvd DHCP Discover Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:29:49.097]
no more address or address previously allocated by another server [30/08 08:29:49.097]
Rcvd DHCP Discover Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:30:04.988]
no more address or address previously allocated by another server [30/08 08:30:04.988]
Rcvd DHCP Discover Msg for IP 0.0.0.0, Mac 00:1B:0C:95:DC:73 [30/08 08:30:08.953]
no more address or address previously allocated by another server [30/08 08:30:08.953]

d_turburville
Joined: Wed 18 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Fri 27 of Aug, 2010 (14:02 UTC)
If you are using TFTPD32 for DHCP you should be able to see in the Log Viewer tab what the phone is doing. Could you copy it here for me pls.
arriben
Joined: Tue 17 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Fri 27 of Aug, 2010 (10:28 UTC)
Hi, I am beginning to lose hope with all of this.

I tried your system tftpd32, the DHCP etc, but no joy. The phone is simply dead. I have tried both with POE, a switch and the phones physical power supply.

One of the things I did was to setup an standard DSL router with 4 ethernet ports and enables DHCP. So I had my PC plugged into it as well as the cisco phone. This allowed me to have an isolated network. it appears that the phone is getting an IP from the DHCP side as I am able to ping it.

After that did not work, I simply plugged the AC supply into the phone, with an ethernet cable directly to my PC, and ran the DHCP side from TFTP32. I can confirm that I do not need an cross over cable as I am able to ping the handset from my LT.

Is it possible that the handset itself has been hardcoded to work only in a set range of IPs, even though I managed to wipe the unit?

It just doesnt seem to be working and it is not the handset because I have another doing the exact same thing. The NIC card on the phone is working as I can see the address allocated to it through log files of Tftpd32 and can also ping.

So is there anything else that I can look for if you dont mind advising me again?

Thanks..
d_turburville
Joined: Wed 18 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Fri 27 of Aug, 2010 (10:26 UTC)
Unfortunately I dont think so. If you have 2 spare ports on your PoE switch you could put them both in a different vlan to the other ports and use them one for the laptop and one for the phone. That should keep the traffic logically separate from the rest of your network.
arriben
Joined: Tue 17 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Thu 26 of Aug, 2010 (14:28 UTC)
Thanks!!!! will try that.

One more question, the only POE switch that I have is in use at my office for other stuff, so I cannot use it to power the phone. I also do not have any of the cisco power supplies for the handset. If I were to use a spare ether net port on our POE switch to the handset, and then an ethernet cable from my LT to the phone, would that work? (IN OTHER WORDS JUST USE THAT POE SWITCH TO GENERATE POWER TO THE PHONE)

Or is it advisable to either get a standalone POE switch or find a power supply for the handset?

ArriB
d_turburville
Joined: Wed 18 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Thu 26 of Aug, 2010 (13:40 UTC)
I do not have an IM account, sorry.
Sounds like the phone has been wiped. I am doing this from memory but what should work for you is the following.
Setup the phone in a local subnet with only itself and a PC (i.e. set the PC to IP address 192.168.1.1)
Run a DHCP server on the PC (you can use TFTPD32 for this)
Configure all the options for the DHCP within the subnet that your PC is in and make sure Additional options is set to port 66 (if this doesnt work try 150) and the IP address of the PC which is also the DHCP and TFTP server
Create a file in the TFTP directory (which should also contain your unzipped firmware files) called xmlDefault.CNF.XML
This file should contain the following

<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation307 model="Cisco 7911">SIP11.8-5-2S</loadInformation307>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>


Now reboot your phone and when the message red light comes on hold # for about 10 seconds
then key in
123456789*0#

Hopefully your phone should then pick up a DHCP address, scan the xmlDefault file, find the firmware name and download it.

Good luck!
arriben
Joined: Tue 17 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Thu 26 of Aug, 2010 (12:22 UTC)
Do you have msn or something that I can contact you on?
I am including a link to a you tube video that I found demonstrating the exact same problem that I have with the physical handsets not doing anything.
I send the guy a message but unfortunately he has not replied. If you do have MSN, or some for of contact and do not mind giving me a hand I would appreicate it.

The problem seems to come in when the actual hanset is involved. The screen is totally dead and I know that their is power going to the phone because when you first plug it into a POE switch (I cannot use a normal AC adapter because I do not have any.) But when you plug it into the POE switch the message light comes on and then the 2 other round buttons by the up/down arrow flash every few minutes. That is as much information as I get from the phone.

You can view the you tube video that I posted earlier from somebody in the world who had the same problem, but never replied to me when I asked him if he ever found a fix.

Or you can get hold me on arrib@integr8it.com (MSN)
or arrib@kf.co.za (mail).

The link for the video is:

http://www.youtube.com/user/janusgod#p/a/u/0/bCuy22oF_Io

Please see what you can do for me?

Thanks..
d_turburville
Joined: Wed 18 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Thu 26 of Aug, 2010 (11:56 UTC)
You will need network connectivity between a PC, the Asterisk server and the phone. For this example I will assume the Asterisk Server is 10.10.10.2, your PC is 10.10.10.10, your default gateway is 10.10.10.1 and your phone is 10.10.10.20
Download TFTPD32 (or another tftp server of your choice) and install on the PC
Extract all the files in the cmterm-7911_7906-sip.8-5-2.zip file to a directory of your choice and point the TFTP server to that directory
Make sure TFTP security is set to None and that the TFTP server is bound to the IP of your PC (i.e. 10.10.10.10). Also make sure the TFTP port is set to 69 (if this does not work try 150)
You will need a SEPmacaddressofphone.cnf.xml file also in that directory (case sensitive) i.e. SEP001D453C78ED.cnf.xml where 001D453C78ED is the MAC address of the phone
That file should contain the following


<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>abc123</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/YA</dateTemplate>
<timeZone>GMT Standard Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>ASTERISK IP ADDRESS</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

<commonProfile>
<phonePassword>abc123</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>

<loadInformation>SIP11.8-5-2S</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>
</vendorConfig>

<networkLocale></networkLocale>

<networkLocaleInfo>
<name></name>
<uid></uid>
<version></version>
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL></authenticationURL>
<directoryURL>http://yourwebserver/dir</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>

<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>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>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
none
<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>

<phoneLabel>PHONE EXTN</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<name>PHONE EXTN</name>
<displayName></displayName>
<contact></contact>
<proxy>ASTERISK IP ADDRESS</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>PHONE EXTN</authName>
<authPassword>PASSWORD</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>
</sipLines>
</sipProfile>
</device>


Change the <processNodeName> and <proxy> settings to your Asterisk server IP, i.e. 10.10.10.2
Make sure <loadInformation> is set to your surrent SIP image, in this case SIP11.8-5-2S
Choose an extension number for you phone i.e. 50000 and set <phoneLabel>, <name> and <authName> to this.
Choose a password for the phone to authenticate to the Asterisk server and set <authPassword> to this. I would recommend something like Abcd123
Leave the other settings and save the file
Create a file called dialplan.xml in the same directory
That file should contain the following


<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="3"/> <!-- Anything else -->
</DIALTEMPLATE>


Now goto the phone
Goto Settings, Network Configuration, IPv4 Configuration
Press **# on the keypad to unlock the settings
Change the settings as follows
1 DHCP - Disabled
2 IP Address - Set a valid IP address for your phone, i.e. 10.10.10.20
3 Subnet Mask - I am assuming this will be 255.255.255.0 for your network
4 Default Router 1 - Set this to your default gateway IP address, i.e. 10.10.10.1
9 DNS Server 1 - Set this to any DNS server you may use, otherwise leave blank
10 DNS Server 2 - Same as DNS Server 1
17 TFTP Server 1 - Set this to the IP of your PC running TFTPD32 i.e. 10.10.10.10
Press the Save softkey
Now goto the web admin page for your Asterisk Server
I am using Elastix so I would then goto PBX, PBX Configuration, Extensions, choose a new Generic SIP device and click Submit
Add the settings as follows

User Extension - The extn number you chose earlier, i.e. 50000
Display Name - Any value, i.e. Smith
secret - The authentication password you chose earlier, i.e. Abcd123

click Submit
Now Smith <50000> should appear in the list of extensions on the right of the page, click it.

change nat (under Device Options) to never rather than yes

Click Submit
Click the Apply Configuration Changes Here banner across the top of the screen
Reboot the phone
When it boots up you should see it requesting all the files from the TFTP server both on the phone screen and in the Log Viewer tab of TFTPD32
It will go through the upgrade process and will eventually come back up, show Registering briefly and voila you should then have dialtone.
Repeat that process with another phone and you should be able to dial between them.

Good luck!
arriben
Joined: Tue 17 of Aug, 2010

Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!!

Posted:Tue 24 of Aug, 2010 (09:07 UTC)
Believe it or not, I found the firmware sitting on one of my server, but under a different folder. Can anyone explain to me how I take this firmware upgrade and push it to a 7911 handset? Do I need to copy it to my Asterisk box, and if so, what then?