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• Cisco 7911 Sip Firmware Upgrade
• Cisco 7911 Sip Firmware
• Cisco Cp-7911 Sip Firmware

Cisco IP Phone 7940 / 7960 VoIP settings & configuration

Cisco has stated that they will not continue to develop SIP software for the Cisco line of phones, so our recommendation is not to invest in a Cisco phone for use with VoIPVoIP.

We are providing these instructions as a courtesy for customers who have already purchased these phones, but we cannot give more technical support.

Before beginning, the Cisco phone must have the SIP firmware installed. Proof of this is the SIP logo in the upper right corner of the phone's screen. Cisco has the SIP phone image available to the public at ftp.cisco.com. Also make sure your device firmware version is 8.2 or better. The Cisco IP Unified Phone Model 7911G and 7906G Firmware version 9.0 (3) is also supplied in a zip file, as an alternative in case a Cisco Unified CallManager is not available to run the executable installer program. The name of this file is cmterm-79117906-sccp.9-0-3.zip. This video will will show you how to get your phone off the boot loop nightmare. We provide support for the cisco ip phone and convert them to sip. These are the latest tested Cisco 79XX files firmware version P003-8-12-00. Extract the.zip package to a folder on your computer. Step 2: Prepare a TFTP Server to deploy the firmware files to the phone. Download any TFTP. server software. The one that was used in this specific SIP upgrade is called Pumpkin, but you can use any TFTP software.

Before beginning, the Cisco phone must have the SIP firmware installed. Proof of this is the SIP logo in the upper right corner of the phone's screen.

Cisco has the SIP phone image available to the public at ftp.cisco.com

Also make sure your device firmware version is 8.2 or better. You can find the firmware version your phone is currently running by using the 'Settings' button on your phone and navigating to Status -> Firmware Versions. If you are running a version prior to 8.2, please upgrade your phone's software before proceeding.

The following instructions assume your phone has successfully booted and completed network configuration including obtaining valid IP addresses for itself, an IP gateway and DNS servers either via DCHP, manual or other means, that your phone is running with factory defaults with firmware version 8.2 or better and that the SIP image has been loaded.

STEP 1

Unlock the configuration protection by pressing '**#'

STEP 2

Press 'settings' button

Configure IP connectivity in 'Network Configuration Menu' (e.g., DHCP support)

STEP 3

- Select 'SIP Configuration'

- Select 'Line 1'

- Enter the account number assigned to you when signed up for VoIPVoIP service, in the fields 'name' and 'authentication name'

- Enter the same password used to login to your VoIPVoIP account

- Enter 'sip3.voipvoip.com' as proxy address

- Set 'g729a' as Preferred Codec

- Press 'back' and 'save' buttons to leave the configuration mode

STEP 4

Make Calls!

NAT/Firewall Issues

If the phone fails to login, please take the time to double check your configuration as above. If everything appears to be correct, the problem may be your firewall.

I have seen a few people experiment with VoIP phones running over a HSMM-MESHÔ link and thought this could be an interesting experiment. I have been thinking of experimenting with VoIP for a couple of years but never found a good reason to as I use my cellphone for just about everything. I figured I would
One day searching through eBay a couple of weeks ago, I found an auction from a local electronics warehouse that was selling 16 Cisco 7911G phones for a price I couldn't pass up. Cursory look on-line seemed that the configuration files were adequately documented so I started off on my journey.
I was looking for a entry-level Asterisk server program that I could load on an old laptop for testing. Searching through the Internet, I found Trixbox - which is a self-contained Linux distro with Asterisk and FreePBX already installed. Configuration is done through a web page interface - perfect for a Linux Padawan like myself.
Setting Trixbox to talk to my Polycom IP501 phone was extremely easy: went through the extension and phone setup pages and had everything configured in less than an hour thanks to YouTube videos and online tutorials. Same thing with a softphone application on my Ubuntu desktop - Twinkle. The Cisco phones, however, were not as easy....
Upgrading the phones to SIP firmware
Cisco has two different firmwares that you can load into the phone which will change it's behavior: SIP (Session Initiated Protocol) and SCCP (Skinny Call Control Protocol). SCCP is a proprietary protocol created and defined by Cisco, where SIP is a more open protocol and generally universal from what I've seen. Since I don't know where this project will go in the long term, I'd rather not limit myself to one brand of phone or equipment so I decided to go with SIP. Unfortunately the phones I bought were pulled from a business environment so it had SCCP firmware installed and needed to be changed.
Changing the firmware is a relatively easy task. It involves having a tftp server on the network that serves the firmware files to the phone upon bootup. After registering for a free account on the Cisco website, I was able to download the SIP9-3-1-1S firmware files and followed the instructions at this link to upgrade 7911 phones to SIP firmware:
http://lodge.glasgownet.com/tech/upgrade-a-cisco-7911-to-sip-firmware/
The configuration file
The Cisco 7911G phones use a XML file that gets loaded during bootup for all of it's settings and configuration. This file sets things like the time and date format, the IP address of the VoIP server, the username and password for debugging and registration, etc. This is where it started to get hairy.

My hope was that with Trixbox, I would just go through the same motions that I did with the Polycom phone and it would be a simple matter to get it working. Boy, was I wrong. Trixbox apparently has limited support for Cisco phones and does not have a default configuration for the 7911G phones. Off to the Internet I went, hoping I could find a valid configuration file that someone had posted but came up empty handed. I found a lot of forum posts that were similar to my situation but none of the configuration files worked.
After a week of trying different configuration files, settings, searching online and just as I was about to write the phones off and re-sell them on eBay, I finally found a configuration file that worked!
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>user</sshUserId>
<sshPassword>pass</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-YA</dateTemplate>
<timeZone>New Zealand Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>xx.xx.xx.xx</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>xx.xx.xx.xx</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</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--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>true</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>

Cisco 7911 Sip Firmware Upgrade

<natAddress></natAddress>
<phoneLabel>Version 3</phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button='1'>
<featureID>9</featureID>
<featureLabel>6003</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>cisco1</name>
<displayName>Cisco Phone 1</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>user</authName>
<authPassword>pass</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>1111</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>cisco1</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button='2'>
<featureID>21</featureID>
<featureLabel>Dial 6004</featureLabel>
<speedDialNumber>6004</speedDialNumber>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>DRdialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP11.9-2-1S</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>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<networkLocale>New_Zealand</networkLocale>
<networkLocaleInfo>
<name>New_Zealand</name>
<version>5.0(2)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL>http://mako/authenticate.php</authenticationURL>
<directoryURL>http://mako/directory.xml</directoryURL>
<idleURL></idleURL>
<informationURL>http://mako/GetTelecasterHelpText.jsp</informationURL>

Cisco 7911 Sip Firmware

<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://mako/services.xml</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>

Cisco Cp-7911 Sip Firmware

<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
From this base, make sure you change the appropriate values for your setup:

• Change the timezone information to whatever you prefer
• Change the <name> tag in the <sipLines> block. It needs to be the same as the username
• Change xx.xx.xx.xx to the IP address of the Trixbox server
• Check all the URLs to make sure they align with the files hosted on the server
• make a authentication script for the <authenticationURL> field
• Change the <loadInformation> tag to match the firmware you have loaded in the phone

• Working on the NAT firewall and forwarding so that phones behind a HSMM-MESH nodes will be able to talk to the Trixbox server
• Coming up with a directory scheme for users
• connecting multiple Asterisk servers together for redundancy/load sharing?
• What is the difference between Trixbox SEP.cnf file and one that works?