here is how I got my Cisco IP Phone 7942 provisioned with Asterisk. this is the result of a whole weekend on searching, flashing, codeing, rebooting, … I believe it applies to the 7962 model as well.

anyhow, all the flashing of firmware, provisioning of configuration, localization, … is done via TFTP. furthermore a DHCP server announces the TFTP server. so in a first step you have to …

prepare the environment

create a host entry with tftp-server-name option in your dhcp server.

host cisco7942 {
  hardware ethernet 00:24:c4:ab:cd:ef;
  option tftp-server-name "_TFTPSERVER_";
}

replace _TFTPSERVER_ with the IP or FQDN of your TFTP server.

install the TFTP server.

aptitude install atftpd

if not running by inetd also add port number in default config!

USE_INETD=false
OPTIONS="--port 69 --tftpd-timeout 300 --retry-timeout 5 --mcast-port 1758 --mcast-addr 239.239.239.0-255 --mcast-ttl 1 --maxthread 100 --verbose=5 /srv/tftp"

start atftpd by

/etc/init.d/atftpd start

don’t forget to drill a hole into your firewall for TFTP on UDP port 69.

iptables -A INPUT -p udp -s _LAN_ -m multiport --dport 69 -m state --state NEW -j ACCEPT

replace _LAN_ with your networks address like 192.168.1.0/24.

get latest firmware

get latest Cisco IP Phone 7942 SIP Software (firmware) and unzip it into /srv/tftp, the tftp root directory. remember the filename in line 6.

apps42.9-3-1ES26.sbn
cnu42.9-3-1ES26.sbn
cvm42sip.9-3-1ES26.sbn
dsp42.9-3-1ES26.sbn
jar42sip.9-3-1ES26.sbn
SIP42.9-3-1SR4-1S.loads
term42.default.loads
term62.default.loads

if you like you can already factory reset your phone. power cycle it. as the phone reboots, you should hold down the # key until the line buttons flash. once they begin to flash, press 123456789*0#, which should make your phone reboot again and the firmware you put in the tftp root directory will start loading. allow this process to run on its own for about 10 to 15 minutes. in the meantime you can watch the firmware served to your phone by issuing following command.

tail -f /var/log/dameon.log | grep tftp

global configuration

the XMLDefault.cnf.xml is the base file for global settings for all of the Cisco IP Phones to be used.

<Default>
  <callManagerGroup>
    <members>
      <member priority="0">
        <callManager>
          <ports>
            <ethernetPhonePort>2000</ethernetPhonePort>
            <mgcpPorts>
              <listen>2427</listen>
              <keepAlive>2428</keepAlive>
            </mgcpPorts>
          </ports>
          <processNodeName>_ASTERISK_</processNodeName>
        </callManager>
      </member>
    </members>
  </callManagerGroup>
  <loadInformation434 model="Cisco 7942">SIP42.9-3-1SR4-1S</loadInformation434>
  <authenticationURL></authenticationURL>
  <directoryURL></directoryURL>
  <idleURL></idleURL>
  <informationURL></informationURL>
  <messagesURL></messagesURL>
  <servicesURL></servicesURL>
</Default>

replace _ASTERISK_ with the IP or FQDN of your Asterisk in line 13. the loadInformation tags in line 18 are used to determine which firmware each device will download and apply. the tags value is the name of the file SIP*.loads without the extension .loads in the firmware (you remember now line 6 of the listing of the files of the phone’s firmware?).

the dialplan.xml contains information on the timeouts for various different patterns of dialled numbers.

<DIALTEMPLATE>
  <TEMPLATE MATCH="2.." Timeout="0" User="Phone"/>
  <TEMPLATE MATCH="*" Timeout="3" User="Phone"/>
</DIALTEMPLATE>

all my extensions start with 2 and have 3 digits, so the phone dials immediatly after match.

device specific configuration

most brainpower needs the device configuration file SEP_MAC_.cnf.xml, unfortunately the format of this config files has never been well documented so Cisco’s latest 79×1 lineup serves as a reference guide to getting these phones working with Asterisk.

<device>
  <deviceProtocol>SIP</deviceProtocol>
  <sshUserId>cisco</sshUserId>
  <sshPassword>cisco</sshPassword>
  <devicePool>
    <dateTimeSetting>
      <dateTemplate>D.M.Y</dateTemplate>
      <timeZone>W. Europe Standard/Daylight Time</timeZone>
      <ntps>
        <ntp>
          <name>_NTPSERVER_</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_</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-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>g711alaw</preferredCodec>
    <dtmfAvtPayload>101</dtmfAvtPayload>
    <dtmfDbLevel>3</dtmfDbLevel>
    <dtmfOutofBand>avt</dtmfOutofBand>
    <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
    <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
    <kpml>3</kpml>
    <natEnabled>false</natEnabled>
    <natAddress></natAddress>
    <phoneLabel>_NAME_</phoneLabel>
    <stutterMsgWaiting>0</stutterMsgWaiting>
    <callStats>false</callStats>
    <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
    <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
    <startMediaPort>16384</startMediaPort>
    <stopMediaPort>32766</stopMediaPort>
    <sipLines>
      <line button="1">
        <featureID>9</featureID>
        <featureLabel>_USER_</featureLabel>
        <proxy>USECALLMANAGER</proxy>
        <port>5060</port>
        <name>_USER_</name>
        <displayName>_USER_</displayName>
        <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
        </autoAnswer>
        <callWaiting>3</callWaiting>
        <authName>_USER_</authName>
        <authPassword>_PASSWORD_</authPassword>
        <sharedLine>false</sharedLine>
        <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
        <messagesNumber>*97</messagesNumber>
        <ringSettingIdle>4</ringSettingIdle>
        <ringSettingActive>5</ringSettingActive>
        <contact>_USER_</contact>
        <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>true</callerNumber>
          <redirectedNumber>true</redirectedNumber>
          <dialedNumber>true</dialedNumber>
        </forwardCallInfoDisplay>
      </line>
    </sipLines>
    <voipControlPort>5060</voipControlPort>
    <dscpForAudio>184</dscpForAudio>
    <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
    <dialTemplate>dialplan.xml</dialTemplate>
  </sipProfile>
  <commonProfile>
    <phonePassword>1234</phonePassword>
    <backgroundImageAccess>true</backgroundImageAccess>
    <callLogBlfEnabled>2</callLogBlfEnabled>
  </commonProfile>
  <loadInformation>SIP42.9-3-1SR4-1S</loadInformation>
  <vendorConfig>
    <disableSpeaker>false</disableSpeaker>
    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
    <pcPort>1</pcPort>
    <settingsAccess>1</settingsAccess>
    <garp>0</garp>
    <voiceVlanAccess>0</voiceVlanAccess>
    <videoCapability>0</videoCapability>
    <autoSelectLineEnable>0</autoSelectLineEnable>
    <sshAccess>0</sshAccess>
    <sshPort>22</sshPort>
    <webAccess>0</webAccess>
    <spanToPCPort>1</spanToPCPort>
    <loggingDisplay>1</loggingDisplay>
    <loadServer></loadServer>
  </vendorConfig>
  <versionStamp></versionStamp>
  <userLocale>
    <name>English_United_Kingdom</name>
    <langCode>en</langCode>
  </userLocale>
  <networkLocale>Germany</networkLocale>
  <networkLocaleInfo>
    <name>Germany</name>
  </networkLocaleInfo>
  <deviceSecurityMode>1</deviceSecurityMode>
  <authenticationURL></authenticationURL>
  <directoryURL></directoryURL>
  <idleURL></idleURL>
  <informationURL></informationURL>
  <messagesURL></messagesURL>
  <proxyServerURL></proxyServerURL>
  <servicesURL></servicesURL>
  <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
  <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
  <dscpForCm2Dvce>96</dscpForCm2Dvce>
  <transportLayerProtocol>2</transportLayerProtocol>
  <capfAuthMode>0</capfAuthMode>
  <capfList>
    <capf>
      <phonePort>3804</phonePort>
    </capf>
  </capfList>
  <certHash></certHash>
  <encrConfig>false</encrConfig>
</device>

replace

• _NTPSERVER_ with NTP servers IP or FQDN,
• _ASTERISK_ with Asterisk IP or FQDN,
• _NAME_ with phones name,
• _USER_ with the username or extension in sip.conf,
• _PASSWORD_ with the password in sip.conf.

define your own timezone in line 8 referring to Time Zones.

the loadInformation tags in line 132 are used to determine which firmware this device will download and apply. the tags value is the name of the file SIP*.loads without the extension .loads in the firmware.

find appropriate values for the user and the network locale in lines 150 – 157 referring to System-Defined User and Network Locales.

Asterisk configuration

just for completeness I will also provide a sip.conf for your Asterisk.

[_USER_]
type=friend
username=_USER_
secret=_PASSWORD_
callerid=_NAME_ <_USER_>
context=from-internal
nat=no
canreinvite=no
host=dynamic

splash into the pool

if you have put all the above mentioned configuration files into the TFTP servers root directory a ls should show you:

apps42.9-3-1ES26.sbn
cnu42.9-3-1ES26.sbn
cvm42sip.9-3-1ES26.sbn
dsp42.9-3-1ES26.sbn
jar42sip.9-3-1ES26.sbn
SIP42.9-3-1SR4-1S.loads
term42.default.loads
term62.default.loads
dialplan.xml
SEP0024C4ABCDEF.cnf.xml
XMLDefault.cnf.xml

if you have not already factory reset your phone it’s time to do! power cycle it. as the phone reboots, you should hold down the # key until the line buttons flash. once they begin to flash, press 123456789*0#, which should make your phone reboot again and the firmware you put in the tftp root directory will start loading. allow this process to run on its own for about 10 to 15 minutes.

or reboot your phone by pressing **#** quickly from the settings menu and wait.

in the meantime you can watch the firmware served to your phone by issuing following command.

tail -f /var/log/daemon.log | grep tftp

you could also have a look if your phone is registering by connecting to a running Asterisk and obtaining a console interface for controlling it on a very high verbosity level.

asterisk -rvvvvvv

to display the current status of the phones additionally execute in Asterisks console interface.

sip show peers

if no entry appears in the list for this phone then review the username=_USER_ and secret=_PASSWORD_ in sip.conf to ensure they match the entries provisioned into the phone. also doublecheck in Asterisks console interface.

sip show users

good luck!

references

how-to

Setup Cisco 7941 or 7961 with Asterisk, en, 2009-10-22
Cisco IP Phones 79XX with Asterisk, en, 2011-11-25
Configure Cisco IP Phones with Asterisk using SIP, en, 2009-12-16
How to load SIP or SCCP on a Cisco 7940 7960 7941 7961 Ip Phone, en, 2011-02-16

SEP_MAC_.cnf.xml

Cisco’s latest 79×1 lineup, en, 2013-03-15
Cisco 7942 with local PBX, en, 2013-08-21
Cisco 7941 7942 7975 SIP Project with PFSense, en, 2012-03-18
Cisco 7942 configuration files, en, 2011-12-15

configuration

Complete Guide for configuring various Cisco 79XX IP Phones with Asterisk, en, 2014-03-14
How-To guide for Cisco 7945/7965/7970/7975/7941/7961, en, 2008-03-25

software/firmware

Cisco Unified IP Phone 7942G, en

lookup tables

System-Defined User and Network Locales, en
Time Zones, en