I used the 7940/7960 pages as a starter for this. This page still needs some love. please help out.

By default most Cisco VoIP phones come configured for Call Manager, which uses the 'Skinny' protocol - SCCP.
This wiki will focus on changing from the default SCCP to the newly support SIP v8.0.2 for the 7970 ONLY


With the right Cable, 7970G series phones can use standard POE injectors, they also work out of the box with Aironet power injectors.
(N.B., the wrong cable may damage your phone!)

Cisco 7970 SIP Phone Software Image
Cisco's SIP phone software images currently include versions v8.0.2 and 8.0.3, both work with Asterisk. While most users had to implement SCCP (http://www.voip-info.org/wiki/view/chan_sccp2) to use this phone, Cisco has finally created SIP for non-call manager implementations. I think they finally wised up to the proliferation of things like Asterisk.

Latest release dates:
v8.0.4SR2 released 2007-01-17
v8.0.4 released 2006-08-29
V8.2.1 released 2006-12-08

For a list of Resolved Problems and Known Problems with this firmware version, you can obtain the Firmware Release Notes in English by clicking the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7970/relnote/index.htm

For a list of Release Notes for other Firmware versions, click the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7970/relnote/index.htm

8.0.4 software has implemented:
-

8.0.4 software has this problems:
- unable to register with Asterisk 1.2.5
-

The v8.0.2 software has implemented: EDIT

Alert-Info (play internal ring tones based on Alert-Info within the SIP header)
Auto Answer (2-way paging conversation without picking up handset)
DHCP Option 66
Directory Enhancements (user can add/change/delete entries in Personal Directory)
DSP (new digital signal processor)
DSP Alarms, Debugging Aids, and Logging (help diagnose problems)
Enhanced Tone and Ring Support (support for more complex tones and ringing patterns)
Hot Line / Speeddials (each line button can be programmed to act as a speeddial button)
Local Call Forwarding (redirects incoming calls to another extension/URL)
Message Waiting Stutter Tone
Multiple Call Appearance (receptionist style, all lines have the same extension)
Outbound Proxy Redesign (improves use of outbound proxy based on multiple DNS records)
SIP Call Statistics (call statistics sent in BYE / 200 OK messages)
Resolved Caveats (several previously documented problems have been resolved)

This will evolve into a step by step guide to upgrading Cisco 7970G to sip version 8.x from the default SCCP load.

v8.0 and up:

First time that SIP is implemented for the 7970G


Cisco SIP IP Phone Administrator Guide, Versions 8.x
http://cisco.com/en/US/products/sw/voicesw/add link
http://www.cisco.com/en/US/products/hw/phones/ps379/products_user_guide_list.html

Cisco SIP IP Telephone 7970G Software (NOTE: This page is only available to registered Cisco.com users with a Cisco Service Agreement)

http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser

Latest Version: v8.0.3 is now released. The file from cisco is designed for the cisco call manager software and is a ".cop" file. This file is just a GZIP compressed TAR file. Just ungzip and untar the file to extract the the new files for the phone. It installs just like the version 7 software with a loader and an application file.

The following is Cisco's pdf on the 7970.

http://www.cisco.com/application/pdf/en/us/guest/products/ps5946/c1629/ccmigration_09186a0080644003.pdf

Unfortunately Cisco have NOT released a detailed breakdown of the workings of the 7970 configuration (SEP<mac>.cnf.xml) file. This is because Cisco now generates the SEP files from within Call Manager (CCM). As a rule Cisco now tell configurators how to make configuration changes from with the CCM application (which then generates the SEP<mac>.cnf.xml). Because of this it's more difficult to 'hand craft' the config files - If anyone has an annotated SEP<mac>.cnf.xml file they could post it would be very useful.

For upgrading to SIP version 8.0.3 the following files come in the .cop file.

For both 7970/7971:
apps70.1-1-2-26.sbn
cnu70.3-1-2-26.sbn
cvm70sip.8-0-2-25.sbn
dsp70.1-1-2-26.sbn
jar70sip.8-0-2-25.sbn
SIP70.sbn

For 7970:
load300006.txt
term70.default.loads

For 7971:
load119.txt
term71.defaults.loads

There are also two files included in the .cop file that should be copied to the TFTBOOT directory:-

SIP70.8-0-3S.loads
copstart.sh

Please note the above file names are SPECIFIC to Cisco SIP 8.0.3 for the 7970/7971, naming conventions for different versions may be slightly different (I don't have 8.0.2, perhaps if someone does and they could post the filenames here that would be useful).

XMLDefault.cnf.xml (NOTE - CASE SENSITIVE - check TFTP or Status logs on phone to confirm case required )
This file contains the default settings that are common to all phones. It is similar to the SIPDefault.cnf file used by the 7940/7960 phones. The 7940/7960 also use this file to determine which version of the firmware to load.
<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> 
<loadInformation8 model="IP Phone 7940">P003-07-4-00</loadInformation8> 
<loadInformation7 model="IP Phone 7960">P003-07-4-00</loadInformation7>
<loadInformation6 model="IP Phone 7970">SIP70.8-0-3S</loadInformation6>   *** identifies the filename to LOAD (SIP70.8-0-3S.loads)
<authenticationURL></authenticationURL> 
<directoryURL></directoryURL> 
<idleURL></idleURL> 
<informationURL></informationURL> 
<messagesURL></messagesURL> 
<servicesURL></servicesURL> 
</Default>


SEP<mac>.cnf.xml example SEP123456789ABC.cnf.xml the <mac> refers to the MAC address of the phone.

Now for individual phone settings, you need this file.

One thing to note: Anyone know how to set individual password for each sip line so it can be authenticated?

There may also be a few tags that aren't recognized/required by the phone - this will need to be updated!

<device xsi:type="axl:XIPPhone" ctiid="203849429" uuid="{96f8508b-10ef-f98c-d20d-0471777ec725}">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId></sshUserId>
<sshPassword></sshPassword>
<devicePool uuid="{a755aa55-089c-2b47-9603-c7d51b9ca4b5}">
<dateTimeSetting uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}">
<dateTemplate>M/D/Y</dateTemplate> ; by adding a after the Y shows time in 12 hour mode i.e. D/M/Ya
<timeZone>Greenwich Standard Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>ccm-beta-5-1</name>
<description>CallManager 5.0 Beta Pub - 5.0.1.032</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>ccm-beta-5-1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1></ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1></sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
Default
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy>USECALLMANAGER</outboundProxy>
<outboundProxyPort>5060</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>
none
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3302</name>
<displayName>3302</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName></authName>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>7b452e87-4496-4762-e11f-b26751a1884b</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate></dialTemplate>
<softKeyFile>SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml</softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-0-0-38S</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><daysDisplayNotActive>1,7</daysDisplayNotActive><displayOnTime>07:30</displayOnTime><displayOnDuration>10:30</displayOnDuration><displayIdleTimeout>01:00</displayIdleTimeout><spanToPCPort>1</spanToPCPort></vendorConfig>
<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>
<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>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://ccm-beta-5-1:8080/ccmcip/authenticate.jsp</authenticationURL>
<directoryURL>http://ccm-beta-5-1:8080/ccmcip/xmldirectory.jsp</directoryURL>
<idleURL></idleURL>
<informationURL>http://ccm-beta-5-1:8080/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://10.86.5.102/CiscoServices/index.xml</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>ccm-beta-5-1</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<line button="3">
<featureID>2</featureID>
<featureLabel>2000</featureLabel>
<speedDialNumber>2000</speedDialNumber>
</line>
<natReceivedProcessing>true</natReceivedProcessing>
<natEnabled>true</natEnabled>
<natAddress></natAddress>

<dialTemplate>dialplan.xml</dialTemplate>

</device>



I now have a fully working system and will post annotated versions of these files later tonight


NOTE from twisted: To enable auto-answering for use with paging/intercom on a line, do the following:

In the line config for the line button you want to be an intercom, change this:

<autoAnswer>
  <autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>

to this:

<autoAnswer>
  <autoAnswerEnabled>3</autoAnswerEnabled>
  <autoAnswerMode>Auto Answer with Speakerphone</autoAnswerMode>
<autoAnswer>



The previous information is below so that it may be merged/delated when someone has time.








load30006.txt (7970) (The content of this file is solely the software image filename stripped of the .cop, i.e. SIP70.8-0-2-0S)(Contains the universal application loader image in 8.x)

cvm70sip.8-0-1-16.sbn (Secure universal application loader for upgrades from images 5.x or later.)
?????????.loads (File that contains the universal application loader and application image, where "a" represents the protocol of the application image loads file 0-SCCP, S-SIP, M-MGCP.)
?????????.sb2 (Application firmware image, where "a" represents the application firmware image.)
SIPDefault.cnf (Contains generic parameters for all Cisco phones at your location)
SEP00036BAAD139.cnf.xml (Where the last 12 hex digits is the MAC address of your Cisco phone) Sample php script to create the cnf file.

In addition, the following optional files may also be present in the TFTP directory: (this was for the 7960 - not sure on impact)

dialplan.xml (contains entries like "9,1...." that cause the phone to automatically dial after a match)
ringlist.xml (a list of ringing tones to be downloaded, like ringer1.pcm)
distinctiveringlist.xml
ringer1.pcm (a ringing tone to be downloaded to the phone)

See also: John Todd's examples (not sure if this applies to the 7970G - Please update if you knwo the answer)

Edit*** Simplify Updates (Auto-Loader Support) ***
This will save you alot of wasted time trying to update newer firmware! For easiest, direct firmware updates from say factory installed SCCP images direct to latest SIP firmware (e.g. SCCP v3.1 to SIP v7.4) — add the following files to your TFTP directory to assist the SCCP based generic Auto-Loader added as of v5.x to Cisco's SIP/SCCP images:

XMLDefault.cnf.xml
xmlDefault.CNF.XML

Due to inconsistent coding by Cisco, different firmware may look for different case-sensitive versions of the same file, thus the need for at least the two variations above to cover new phones and a good portion of older one's. Additionally, here is a usable example of the XML content that should be inserted into the files (be sure and update with the firmware version you wish to load, and match the SIPDefault.cnf and SEPxxxxxxx.cnf.xml file's "image_version=" entries to match!):

<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> 
 <loadInformation8 model="IP Phone 7940">P003-07-4-00</loadInformation8> 
 <loadInformation7 model="IP Phone 7960">P003-07-4-00</loadInformation7>
 <loadInformation6 model="IP Phone 7970">SIP70.8-0-2-0S</loadInformation6>
 <authenticationURL></authenticationURL> 
 <directoryURL></directoryURL> 
 <idleURL></idleURL> 
 <informationURL></informationURL> 
 <messagesURL></messagesURL> 
 <servicesURL></servicesURL> 
</Default>

Misc

See reboot.pl. A perl script to handle remote rebooting of the 79xx class phones (useful for multiple-phone upgrades). Requires Net::Telnet.

NOTE: For Reference, reboot.pl has been moved to http://www.nmedia.net/~mklein/reboot.pl . Still Requires Net::Telnet.

---

Time Zone Codes

The time zone codes must be input EXACTLY as written below including caps, spaces (do not use the_underscore_for spaces) & punctuation. If not, the time zone displayed on the screen will revert to UTC.

Dateline Standard Time
Samoa Standard Time
Hawaiian Standard Time
Alaskan Standard/Daylight Time
Pacific Standard/Daylight Time
Mountain Standard/Daylight Time
US Mountain Standard Time
Central Standard/Daylight Time
Mexico Standard/Daylight Time
Canada Central Standard Time
SA Pacific Standard Time
Eastern Standard/Daylight Time
US Eastern Standard Time
Atlantic Standard/Daylight Time
SA Western Standard Time
Newfoundland Standard/Daylight Time
South America Standard/Daylight Time
SA Eastern Standard Time
Mid-Atlantic Standard/Daylight Time
Azores Standard/Daylight Time
GMT Standard/Daylight Time
Greenwich Standard Time
W. Europe Standard/Daylight Time
GTB Standard/Daylight Time
Egypt Standard/Daylight Time
E. Europe Standard/Daylight Time
Romance Standard/Daylight Time
Central Europe Standard/Daylight Time
South Africa Standard Time
Jerusalem Standard/Daylight Time
Saudi Arabia Standard Time
Russian Standard/Daylight Time
Iran Standard/Daylight Time
Caucasus Standard/Daylight Time
Arabian Standard Time
Afghanistan Standard Time
West Asia Standard Time
Ekaterinburg Standard Time
India Standard Time
Central Asia Standard Time
SE Asia Standard Time
China Standard/Daylight Time
Taipei Standard Time
Tokyo Standard Time
Cen. Australia Standard/Daylight Time
AUS Central Standard Time
E. Australia Standard Time
AUS Eastern Standard/Daylight Time
West Pacific Standard Time
Tasmania Standard/Daylight Time
Central Pacific Standard Time
Fiji Standard Time
New Zealand Standard/Daylight Time

---

Logo Displayed on 7970 Screen

A non-Cisco logo can be displayed on the 7970 screen.
Cisco's documentation states the logo be a .PNG bitmap format.

Please create the following directory in your TFTPBOOT directory (case sensitive)

/Desktops/320x212x12

In this directory store your PNG files, each file can be up to 4096 colours and 320x212 pixels.

For each file you need a Fullsize PNG file (320x212x12) and a Thumbnail PNG (80x53x12)

Also generate a List.xml file in this directory

The format of this file is:-
<CiscoIPPhoneImageList>

         <ImageItem Image="TFTP:Desktops/320x212x12/thumbnail.png"  URL="TFTP:Desktops/320x212x12/Fullsize.png"/>

</CiscoIPPhoneImageList>

Where thumbnail.png is the name of the thumbnail file and fullsize.png is the name of the corresponding fullsize file.

You can have multiple listings in this directory and they are then accessed via the phone from Menu-->User Preferences-->Background Images

---

Company Telephone Directory

The 7970G phone has four panel keys labeled as Messages, Services, Directories, and Settings. The Directory key can be programmed to view your company's telephone directory by displaying Names and Telephone Numbers that are stored on any web site available to you.
Modify the SIPDefault.cnf file entry to point to the web site:
directory_url: "http://www.mywebserver.com/asterisk/directory.xml"
The phone must be rebooted in order to read the above file.
The web site file /asterisk/directory.xml should include xml entries like:


<CiscoIPPhoneDirectory>        
  <Title>IP Telephony Directory</Title> 
  <Prompt>People reachable via VoIP</Prompt>                     
  <DirectoryEntry>                                               
    <Name>Rich</Name>          
    <Telephone>3000</Telephone> 
  </DirectoryEntry>                                       
  <DirectoryEntry>              
    <Name>Todd</Name>        
    <Telephone>3001</Telephone>                                             
  </DirectoryEntry>       
</CiscoIPPhoneDirectory> 



Note: Each time a user presses the Directory key and accesses the External Directory option from the menu, the phone will access the contents of this html file and display whatever text entries included in it. Therefore, changes to the html file do not require any futher rebooting of the Cisco phone. Cisco has published Cisco IP Phone Services Application Development Notes (CMXML_App_Guide.pdf ) that further explains options and file contents.

QUESTION:
- Phone directory can display 32 entrys per page. How to display second, third... page?

Asterisk Cisco 7970 XML Services

Services Button

The 7970 phones have four panel keys labeled as Messages, Services, Directories, and Settings. The Services key can be programmed to execute CGI scripts that are stored on any web site available to you. The CGI scripts can perform any action that you are capable of programming. None are provided by Cisco.
Modify the SIPDefault.cnf file entry to point to the web site:
services_url: "http://www.mywebserver.com/asterisk/myscriptpage.html"
The phone must be rebooted in order to read the above file.
The web site file /asterisk/myscriptpage.html should include entries for each service that you plan to make available to your phone users. The exact content and syntax is also documented in the Cisco IP Phone Services Application Development Notes (CMXML_App_Guide.pdf ) noted above.

Messages Button

When the Messages button is pressed, you can cause the phone to directly dial an extension in your asterisk dialplan. Just configure the phone as:
      <messagesNumber>extension</messagesNumber>
where "extension" is what you wish the phone to dial when the Messages button is pressed. You can then catch the call in either a standard VoiceMailMain() invocation a la
      exten => _42,1,VoiceMailMain() 
or, be cute and bypass entry of mailbox number and password a la
      exten => _42,1,VoiceMailMain(s<mbox num>) 
To make the Messages button work for any extension (assuming your extensions are numbered appropriately), use:
      exten => _42,1,VoiceMailMain(s${CALLERIDNUM}) 

Asterisk Cisco 79XX XML Services

Ringtones

http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080087511.html#1042487
The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By
default, your ring type options will be those two choices. However,
using the RINGLIST.DAT file, you can customize the ring types that are
available to the Cisco SIP IP phone users.

Step 1
Create a pulse code modulation (PCM) file of the desired ring
types and store the PCM files in the root directory of your TFTP server.
PCM files must contain no header information and comply with the
following format guidelines:

8000 Hz sampling rate
8 bits per sample
ulaw compression
240 - 16080 samples long ( 0.03 sec - 2.01 sec )

For example, to use sox to generate the tones, use


sox -t wav in.wav -t raw -r 8000 -U -b -c 1 out.raw resample -ql



Step 2
Using a ASCII editor, open the RINGLIST.DAT file and for each
of the ring types you are adding, specify the name as you want it to
display on the Ring Type menu, press Tab, and then specify the filename
of the ring type. For example, the format of a pointer in your
RINGLIST.DAT file should appear similar to the following:

Ring Type 1 ringer1.pcm

Step 3
After defining pointers for each of the ring types you are
adding, save your modifications and close the RINGLIST.DAT file.


Caveat:
If you have configured a secondary tftp-server(ie. dyn_tftp_addr : 192.168.1.10) in SIPDefault.cnf, or SEP<MACADDR>.cnf.xml which cannot be reached then the phone will not attempt to download the RINGLIST.DAT file.


EditControlling ring tones from Asterisk
By setting the Asterisk variable ALERT_INFO before you call Dial, Asterisk will add ringer tone info to the SIP invite that is sent to the phone.

exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer 
exten => 3010,2,Dial(SIP/3010,15) 

Note: In SIP_HEAD or v1+ you wil need to do the following:
exten => 3010,1,SetVar(_ALERT_INFO=something) 


Available ring tones by default
Bellcore-BusyVerify  
Bellcore-Stutter  
Bellcore-MsgWaiting  
Bellcore-dr1   
Bellcore-dr2  
Bellcore-dr3  
Bellcore-dr4  
Bellcore-dr5 

---

Ringtones for Cisco 7970

Setting up your ringlist on a 7970 is very easy but aside from Cisco's documentation it was difficult to find a good example.

For you beginners out there you should note that *everything* is case-sensitive.

ringlist.xml is required in /tftpboot.

File format is as follows:
<CiscoIPPhoneRingList>
    <Ring>
         <DisplayName/>
         <FileName/>
    </Ring>
</CiscoIPphoneRingList>

Sample ringlist.xml:
<CiscoIPPhoneRingList>
    <Ring>
         <DisplayName>Merlin 2</DisplayName>
         <FileName>merlin2.pcm</FileName>
    </Ring>
    <Ring>
         <DisplayName>Merlin 3</DisplayName>
         <FileName>merlin3.pcm</FileName>
    </Ring>
</CiscoIPPhoneRingList>

---

DialPlan Notes (DIALPLAN.XML)

The DIALPLAN.XML file controls the phone's matching of digits. By default "*" matches anything and times out after 5 seconds. Users must push 'Dial' or '#' to connect if they don't want to wait 5 seconds. For a variety of reasons, not least of which being that most phone users are not accustomed to pressing 'Dial' on their offices, it may be desirable to configure a dial plan for your organization.

If you wish to use the hash (by default it will immediately dial the number entered) include it explicitly as part of a pattern in DIALPLAN.XML
  <DIALTEMPLATE> 
      <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> 
      <TEMPLATE MATCH="*" Timeout="5" User="Phone" /> 
  </DIALTEMPLATE> 

In another example, we immediately match 9+10digits or 9+1+10digits, and match 5+2digits as internal extensions
 <DIALTEMPLATE> 
    <TEMPLATE MATCH="5.." TIMEOUT="0"/> 
    <TEMPLATE MATCH="9,1.........." TIMEOUT="0" Tone="Bellcore-Alerting"/> 
    <TEMPLATE MATCH="9,.........." TIMEOUT="0"/> 
 </DIALTEMPLATE> 

In the above example, a secondary dial tone is invoked by the comma character. If the Tone attribute is left blank, the default will be used. Or you can specify one of the following:


Bellcore-Alerting
Bellcore-Busy
Bellcore-BusyVerify
Bellcore-CallWaiting
Bellcore-Confirmation
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5
Bellcore-dr6
Bellcore-Hold
Bellcore-Inside
Bellcore-None
Bellcore-Outside (default)
Bellcore-Permanent
Bellcore-Reminder
Bellcore-Reorder
Bellcore-Stutter


CallWaiting-2
CallWaiting-3
CallWaiting-4


Cisco-BeepBonk
Cisco-Zip
Cisco-ZipZip

Notes: This file is case sensitive in some firmware versions; all elements and attributes should be uppercase (except Tone) or the entries may be ignored. According to Cisco, the phone will always match the LONGEST expression.

---

Call Waiting Feature

The 79XX series phones have a good way of handling SIP registrations provided the Call Waiting feature isn't turned off. Most other SIP phones require an individual SIP username and password for each line appearance. Instead, the 79XX will automatically roll-over to the next available line. So, for example, on a 7960 you can have all six lines programmed to the same SIP username/password and the phone will automatically handle the call waiting function. Note: If you use any other method of ringing multiple lines on the phone (i.e. dialing SIP/123&SIP/456) your phone will show a confusingly high number of missed calls.




For example:


SEPXXXXX.cnf.xml: (THIS NEEDS TO BE UPDATED FOR THE 7970 AS THE CONFIG IS DIFFERENT)
??????


In your sip.conf: (THIS NEEDS TO BE UPDATED FOR THE 7970 AS THE CONFIG IS DIFFERENT)
510 
type=friend 
username=510 
secret=test 
host=dynamic 
dtmfmode=rfc2833 
context=whatever 
canreinvite=no 
nat=no 
mailbox=510@default 
callerid=<510> 


In your extensions.conf: (THIS NEEDS TO BE UPDATED FOR THE 7970 AS THE CONFIG IS DIFFERENT)
exten => 510,1,Dial(SIP/510,20,mTt) 
exten => 510,2,Voicemail(u510@default) 
exten => 510,3,Hangup 
exten => 510,102,Voicemail(b510@default) 
exten => 510,103,Hangup 

---

Asterisk Configuration File Examples

sip.conf (THIS NEEDS TO BE UPDATED FOR THE 7970 AS THE CONFIG IS DIFFERENT)
d4b74cfc198b6adbd3037271e258036f 
type=friend ; This device takes and makes calls 
host=dynamic ; This host is not on the same IP addr every time 
username=d4b74cfc198b6adbd3037271e258036f ; Username programmed into Cisco phone 
secret=mypassword ; Password for device 
context=from-sip        ; Inbound calls from this phone go to this context 
nat=yes ; nat=yes if this phone is behind a NAT box or firewall 
callgroup=2 ; the group to which this phone belongs for *8 phone ringing pickup 
pickupgroup=2 ; the pickup group allowed from this phone when *8 is dialed 
mailbox=d4b74cfc198b6adbd3037271e258036f             ; Activate the message waiting light if this voicemailbox has messages in it 


extensions.conf (THIS NEEDS TO BE UPDATED FOR THE 7970 AS THE CONFIG IS DIFFERENT)
exten => d4b74cfc198b6adbd3037271e258036f,1,Dial(SIP/d4b74cfc198b6adbd3037271e258036f,15,t) ; see "show application dial" for options and formats 
exten => d4b74cfc198b6adbd3037271e258036f,2,Voicemail2(ud4b74cfc198b6adbd3037271e258036f) ; go to Voicemail2 if phone is "U"nanswered 
exten => d4b74cfc198b6adbd3037271e258036f,102,Voicemail2(bd4b74cfc198b6adbd3037271e258036f)  ; go to Voicemail2 if phone is "B"usy 
exten => d4b74cfc198b6adbd3037271e258036f,103,Hangup ; and then hangup. 


voicemail.conf (THIS NEEDS TO BE UPDATED FOR THE 7970 AS THE CONFIG IS DIFFERENT)
format=gsm  
servermail=mail.myserver.com 

attach=no 
maxmessage=120  
maxsilence=10 
pbxskip=yes 
fromstring=NPI VM 
emailbody=\nVM for x ${VM_MAILBOX} from ${VM_CALLERID} dur: ${VM_DUR} \n 
default 
; Note: following sends a text message to a cell phone telling me someone left a voicemail 
d4b74cfc198b6adbd3037271e258036f => d4b74cfc198b6adbd3037271e258036f,FirstName LastName,4015719329@vtext.com  

---

Troubleshooting Phone Registration

From the asterisk command line, execute "sip show peers" and "sip show users" to display the current status of the Cisco phone. If no entries appear in the list for this phone, then review the "username=d4b74cfc198b6adbd3037271e258036f" and "secret=mypassword" in sip.conf to ensure they match the entries programmed into the Cisco phone.
*CLI> sip show peers 
Name/username    Host                 Mask             Port     Status     

d4b74cfc198b6adbd3037271e258036f/d4b74cfc198b6adbd3037271e258036f        67.11.89.61     (D)  255.255.255.255  5060     Unmonitored 
*CLI> sip show users 
Username         Secret           Authen           Def.Context      A/C   
d4b74cfc198b6adbd3037271e258036f             mypassword          md5,plaintext    from-sip         No    


Troubleshooting Cisco Phone
The Cisco 79XX phones support telnet. To diagnose problems with the Company Directory function noted above (as an example), telnet to the phone's IP address using the login password provided in the SIP00036BAAD139.cnf file noted above. For example, to diagnose a possible http problem, do the following:


SIP Phone> debug http 
Enabling bug logging on this terminal - use 'tty mon 0' to disable 
debugs: http timestamp 
SIP Phone> 11:39:39 Connect2WWWIPPort called IpAddr0, port80, hostNamewww.mydomain.com 
11:39:39 Connect2WWWIPPort Sending Request to IpAddr207.212.93.75, port80 
11:39:39 HTTP RECV (ACK CMD) 
11:39:39 HTTP RECV (OPEN CMD) 
11:39:39 
HTTP Send 178 Bytes of Data 
Data Packet is: 
=============== 
GET /asterisk/directory.html?name=SIP00036BC38B88 HTTP/1.1 
User-Agent: Allegro-Software-WebClient/3.10b1 
Host: www.mydomain.com 
Connection: Close 


(Note: You can also reset the 7970 by pressing the # after cycling the power. After the 10 line buttons have completed flashing enter 1-2-3-4-5-6-7-8-9-*-0-#.)

---

Another SEP<mac>.xml.cnf example

<device xsi:type="axl:XIPPhone" ctiid="203849429" uuid="{96f8508b-10ef-f98c-d20d-0471777ec725}">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>user</sshUserId>
<sshPassword>pass</sshPassword>
<devicePool uuid="{a755aa55-089c-2b47-9603-c7d51b9ca4b5}">
<name>Dallas 5.0 Beta</name>
<dateTimeSetting uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}">
<name>CMLocal</name>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<name>5.0 Beta</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>ccm-beta-5-1</name>
<description>CallManager 5.0 Beta Pub - 5.0.1.032</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>ccm-beta-5-1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1>10.0.0.26</ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1>10.0.0.26</sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
Default
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy>10.0.0.26</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>10.0.0.26</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy>10.0.0.26</outboundProxy>
<outboundProxyPort>5060</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>
none
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>Lama d.o.o.</phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>495-131</featureLabel>
<proxy>10.0.0.26</proxy>
<port>5060</port>
<name>Name</name>
<displayName>My Name</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>user</authName>
<authPassword>pass</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>7b452e87-4496-4762-e11f-b26751a1884b</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>

<line button="2">
<featureID>2</featureID>
<featureLabel>My friend</featureLabel>
<speedDialNumber>123456</speedDialNumber>
</line>
</sipLines>

<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml</softKeyFile>
</sipProfile>

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

<loadInformation>SIP70.8-0-2SR1S</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>1</webAccess>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:30</displayOnDuration>
<displayIdleTimeout>01:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>

<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>

<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>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://ccm-beta-5-1:8080/ccmcip/authenticate.jsp</authenticationURL>
<directoryURL>http://10.0.0.20/cisco_voip/PhoneDirectory.xml</directoryURL>
<idleURL></idleURL>
<informationURL>http://ccm-beta-5-1:8080/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL>10.0.0.26</proxyServerURL>
<servicesURL>http://10.0.0.20/cisco_voip/services.xml</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>

<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>ccm-beta-5-1</processNodeName>
</capf>
</capfList>

<certHash></certHash>
<encrConfig>false</encrConfig>
</device>

dialplan.xml

To make it work you'll need dialplan as well, here is example that works well in Croatia. dialplan.xml needs to be in root of your tftp directory.

<DIALTEMPLATE>
<!-- Lokalni pozivi -->
<TEMPLATE MATCH="1.." Timeout="0"/>

<!-- Zupanijski pozivi -->
<TEMPLATE MATCH="01....." Timeout="0"/>
<TEMPLATE MATCH="02....." Timeout="0"/>
<TEMPLATE MATCH="03....." Timeout="0"/>
<TEMPLATE MATCH="04....." Timeout="0"/>
<TEMPLATE MATCH="05....." Timeout="0"/>
<TEMPLATE MATCH="06....." Timeout="0"/>
<TEMPLATE MATCH="07....." Timeout="0"/>
<TEMPLATE MATCH="08....." Timeout="0"/>
<TEMPLATE MATCH="09....." Timeout="0"/>

<!-- Meduzupanijski pozivi -->
<TEMPLATE MATCH="001......." Timeout="0"/> <!-- Zagrebacka i grad Zagreb -->
<TEMPLATE MATCH="0049......" Timeout="0"/> <!-- Krapinsko - zagorska -->
<TEMPLATE MATCH="0044......" Timeout="0"/> <!-- Sisacko - moslavacka -->
<TEMPLATE MATCH="0047......" Timeout="0"/> <!-- Karlovacka -->
<TEMPLATE MATCH="0042......" Timeout="0"/> <!-- Varazdinska -->
<TEMPLATE MATCH="0048......" Timeout="0"/> <!-- Koprivnicko - krizevacka -->
<TEMPLATE MATCH="0043......" Timeout="0"/> <!-- Bjelovarsko - gilogorska -->
<TEMPLATE MATCH="0051......" Timeout="0"/> <!-- Primorsko - goranska -->
<TEMPLATE MATCH="0053......" Timeout="0"/> <!-- Licko - senjska -->
<TEMPLATE MATCH="0033......" Timeout="0"/> <!-- VIroviticko - podravska -->
<TEMPLATE MATCH="0034......" Timeout="0"/> <!-- Pozesko - slavonska -->
<TEMPLATE MATCH="0035......" Timeout="0"/> <!-- Brodsko - posavska-->
<TEMPLATE MATCH="0023......" Timeout="0"/> <!-- Zadarska -->
<TEMPLATE MATCH="0031......" Timeout="0"/> <!-- Osjecko - baranjska -->
<TEMPLATE MATCH="0022......" Timeout="0"/> <!-- Sibensko - kninska -->
<TEMPLATE MATCH="0032......" Timeout="0"/> <!-- Vukovarsko - srijemska -->
<TEMPLATE MATCH="0021......" Timeout="0"/> <!-- Splitsko - dalmatinska -->
<TEMPLATE MATCH="0052......" Timeout="0"/> <!-- Istarska -->
<TEMPLATE MATCH="0020......" Timeout="0"/> <!-- Dubrovacko - neretvanska -->
<TEMPLATE MATCH="0040......" Timeout="0"/> <!-- Medimurska-->

<!-- Mobiteli -->
<TEMPLATE MATCH="0099......." Timeout="0"/>
<TEMPLATE MATCH="0098......." Timeout="0"/>
<TEMPLATE MATCH="0095......." Timeout="0"/>
<TEMPLATE MATCH="0091......." Timeout="0"/>

<!-- Cisco - default -->
<TEMPLATE MATCH="9,59....." Timeout="0"/>
<TEMPLATE MATCH="9,29....." Timeout="0"/>
<TEMPLATE MATCH="9,832......." Timeout="0"/>
<TEMPLATE MATCH="9,713......." Timeout="0"/>
<TEMPLATE MATCH="9,281......." Timeout="0"/>
<TEMPLATE MATCH="9,903......." Timeout="0"/>
<TEMPLATE MATCH="\*500" Timeout="0"/>
<TEMPLATE MATCH="\*54" Timeout="0"/>
<TEMPLATE MATCH="\*55" Timeout="0"/>
<TEMPLATE MATCH="\*69" Timeout="0"/>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>

Cisco 7970 IP Phone - restart

To restart the 7970 with the sip software press **#** while in the settings menu.

Cisco 7970 IP Phone - SSH login

- Find out your IP address (Settings => Network Configuration => IP Address=
- Use some ssh client to login on your IP address and port 22
- for loging in use user/pass that you have defined in SEP<mac>.xml.cnf file (sshUserId/sshPassword)
- when prompted again log in as user debug and password debug

Still need to configure:

If sombody knows how to configure following please edit this page or e-mail me at tparcina@lama.hr

- How to setup hinting (Multiple Call Appearance) - does it work with SIP firmware?
- How to make Java application for this phone?
- How to remouve gray lines from display (when line is defined)?

Thank you for your help!