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Configuration for Cisco IP Phone 7942

Posted on 2011-03-16
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Medium Priority
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Last Modified: 2014-04-19
Does anyone has the configuration file SEP<MAC>.CNF.xml for Cisco IP PHone 7942 ?

I have difficulty to make it load succesfully and it return with "Error Verifying Config Info".

Tks
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Question by:AXISHK
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15 Comments
 
LVL 35

Expert Comment

by:Ernie Beek
ID: 35146476
Hi again. I posted it in your other question but I'll do it here as well:

 
<device>  
<Default>   
<callManagerGroup>   
<members>   
<member priority="0">   
<callManager>   
<ports>   
<ethernetPhonePort>2000</ethernetPhonePort>   
<sipPort>5060</sipPort>  
<securedSipPort>5061</securedSipPort>  
</ports>   
<processNodeName>!!!!! SIP SERVER !!!!!</processNodeName>   
</callManager>   
</member>   
</members>   
</callManagerGroup>  
<loadInformation434 model="Cisco 7942"></loadInformation434>  
</Default>   
<deviceProtocol>SIP</deviceProtocol>  
<sshUserId>admin</sshUserId>  
<sshPassword>admin</sshPassword>  
<devicePool>  
<dateTimeSetting>  
<dateTemplate>D/M/Y</dateTemplate> ; by adding a after the Y shows time in 12 hour mode i.e. D/M/Ya  
<timeZone>!!!!! TIME ZONE !!!!!</timeZone>  
<ntps>   
<ntp>   
<name>!!!!! NTP SERVER !!!!!</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>!!!!! SIP SERVER !!!!!</processNodeName>  
</callManager>  
</member>  
</members>  
</callManagerGroup>  
</devicePool>  
<sipProfile>  
<sipProxies>  
<backupProxy></backupProxy>   
<backupProxyPort></backupProxyPort>   
<emergencyProxy></emergencyProxy>   
<emergencyProxyPort></emergencyProxyPort>   
<outboundProxy></outboundProxy>   
<outboundProxyPort></outboundProxyPort>   
<registerWithProxy>true</registerWithProxy>   
</sipProxies>  
<enableVad>false</enableVad>  
<preferredCodec>g729a</preferredCodec>  
<natEnabled>!!!!! TRUE/FALSE !!!!!</natEnabled>   
<natAddress></natAddress>   
<phoneLabel>!!!!! LINE LABEL !!!!!</phoneLabel>  
<sipLines>  
&bull;   
<featureID></featureID>  
<featureLabel></featureLabel>  
<proxy>!!!!! SIP SERVER !!!!!</proxy>  
<port>5060</port>  
<name>09518096</name>  
<displayName>V-Spec</displayName>  
<autoAnswer>  
<autoAnswerEnabled>2</autoAnswerEnabled>  
</autoAnswer>  
<callWaiting>3</callWaiting>  
<authName>!!!!! USERNAME !!!!!</authName>  
<authPassword>!!!!! PASSWORD !!!!!</authPassword>  
<sharedLine>false</sharedLine>  
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>  
<messagesNumber>!!!!! SIP PHONE NUMBER!!!!!</messagesNumber>  
<ringSettingIdle>4</ringSettingIdle>  
<ringSettingActive>5</ringSettingActive>  
<contact>!!!!! SIP PHONE NUMBER !!!!!</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>   
</sipProfile>  
<loadInformation>SIP42.8-5-2S</loadInformation>  
<directoryURL></directoryURL>  
<messagesURL>!!!!! SIP SERVER !!!!!</messagesURL>  
<servicesURL></servicesURL>  
<networkLocale>!!!!! LOCALE !!!!!</networkLocale>   
<networkLocaleInfo>   
<name>!!!!! LOCALE !!!!!</name>   
</networkLocaleInfo>   
</device>

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LVL 35

Expert Comment

by:Ernie Beek
ID: 35146486
Take in to account though that the phones are very picky. One wrong character and they start to protest. Oh, it's also case sensitive.
0
 

Author Comment

by:AXISHK
ID: 35146815
No, it doesn't work.

Do I miss / overlook some steps beside the SEP<MAC>.cnf.xml ? Can u list out the steps that I need to check for any missing steps / configuration ?

Great Thanks
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LVL 35

Expert Comment

by:Ernie Beek
ID: 35146869
Just to ask the obvious, you did change the values in the config file of course (?)
0
 

Author Comment

by:AXISHK
ID: 35147072
Yes.

Tks.
0
 

Author Comment

by:AXISHK
ID: 35147076
BTW, the Cisco phone is connected to Asterisk server, not the Cisco one...

Tks
0
 
LVL 35

Expert Comment

by:Ernie Beek
ID: 35147136
I know, otherwise you'd be using sccm instead of sip.

Don't know if I asked this before, you have loaded sip firmware on the phone?
0
 

Author Comment

by:AXISHK
ID: 35153601
Below is my configuration and afterwards, I couldn't find the firmware SIP42.9-0-3S on the phone. Does it mean it doesn't load yet ?

Thanks

<device> 
<Default>   
<callManagerGroup>   
<members>   
<member priority="0">   
<callManager>   
<ports>   
<ethernetPhonePort>2000</ethernetPhonePort>   
<sipPort>5060</sipPort>  
<securedSipPort>5061</securedSipPort>  
</ports>   
<processNodeName>192.168.1.32</processNodeName>   
</callManager>   
</member>   
</members>   
</callManagerGroup>  
<loadInformation9 model="IP Phone 7942G"></loadInformation9>  
</Default>   
<deviceProtocol>SIP</deviceProtocol>  
<sshUserId>admin</sshUserId>  
<sshPassword>admin</sshPassword>  
<devicePool>  
<dateTimeSetting>  
<dateTemplate>D/M/Y</dateTemplate> 
<timeZone>China Standard/Daylight Time</timeZone>  
<ntps>   
<ntp>   
<name>hk.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>192.168.1.32</processNodeName>  
</callManager>  
</member>  
</members>  
</callManagerGroup>  
</devicePool>  
<sipProfile>  
<sipProxies>  
<backupProxy></backupProxy>   
<backupProxyPort></backupProxyPort>   
<emergencyProxy></emergencyProxy>   
<emergencyProxyPort></emergencyProxyPort>   
<outboundProxy></outboundProxy>   
<outboundProxyPort></outboundProxyPort>   
<registerWithProxy>true</registerWithProxy>   
</sipProxies>  
<enableVad>false</enableVad>  
<preferredCodec>g729a</preferredCodec>  
<natEnabled>FALSE</natEnabled>   
<natAddress></natAddress>   
<phoneLabel>ABC</phoneLabel>  
<sipLines> 
<featureID></featureID>  
<featureLabel></featureLabel>  
<proxy>192.168.1.32</proxy>  
<port>5060</port>  
<name>09518096</name>  
<displayName>V-Spec</displayName>  
<autoAnswer>  
<autoAnswerEnabled>2</autoAnswerEnabled>  
</autoAnswer>  
<callWaiting>3</callWaiting>  
<authName>2884</authName>  
<authPassword>8888</authPassword>  
<sharedLine>false</sharedLine>  
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>  
<messagesNumber>2884</messagesNumber>  
<ringSettingIdle>4</ringSettingIdle>  
<ringSettingActive>5</ringSettingActive>  
<contact>2884</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>   
</sipProfile>  
<loadInformation>SIP42.9-0-3S</loadInformation>  
<directoryURL></directoryURL>  
<messagesURL>192.168.1.32</messagesURL>  
<servicesURL></servicesURL>  
<networkLocale>United_States</networkLocale>   
<networkLocaleInfo>   
<name>United_States</name>   
</networkLocaleInfo>   
</device>

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LVL 6

Expert Comment

by:vikrantambhore
ID: 35153787
0
 
LVL 35

Expert Comment

by:Ernie Beek
ID: 35154604
Did you check: Settings – Status – Firmware Versions?

Also you should be able to connect to the phone through a web browser. You could find it there as well.
0
 

Author Comment

by:AXISHK
ID: 35154701
Load File : SIP42.9-0-3S
App Load ID : jar42sip.9-0-3TH1-22.sbn
JVM Load ID : cvm42sip.9-0-3TH1-22.sbn
OS Load ID : cnu42.9-0-3TH1-22.sbn
Boot Load ID : tnp42.8-3-1-21a.bin
DSP Load ID : dsp42.9-0-3TH1-22.sbn


The message logged in the phone is :
  Error Updating Locale
  No Trust List Installed


Thanks


0
 
LVL 35

Expert Comment

by:Ernie Beek
ID: 35154777
You can safely ignore those. These files are used for encryption and only work with cisco call manager.

I came across some posts from people also having issues with the 9.x firmware. Reverting to 8.x resolved their issues. Any change you could test that?
0
 

Accepted Solution

by:
AXISHK earned 0 total points
ID: 35155376
Which version are u currently using for connecting to Asterisk ?

Can I downgraded it ?

Tks

0
 

Author Comment

by:AXISHK
ID: 35163738
I have used another phone and try to load the P0S3-8-12-00. However, it return with "Load Authentication Failed P0S3-8-12-00" while downloading P0S3-8-12-00.

Afterwards, it keep "Registering" but the firmware hasn't been updated yet. Any idea ?

Here is the content of my files . Is it possible to upload these 3 files for my reference ?
Any advise for the probelm is appreciated.


Great thanks.
SEP<MAC>.cnf.xml
----------------
<device> 
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId> 
<sshPassword>admin</sshPassword> 
<devicePool> 
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>China Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>59.148.184.7</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>192.168.1.32</processNodeName> 
</callManager> 
</member> 
</members> 
</callManagerGroup> 
</devicePool> 
<sipProfile> 
<sipProxies> 
<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> 
<preferredCodec>g711ulaw</preferredCodec>
<natEnabled>false</natEnabled>
<phoneLabel>SIP</phoneLabel> 
<sipLines> 
<line button="1"> 
<featureID>9</featureID> 
<featureLabel>HK</featureLabel> 
<proxy>192.168.1.32</proxy> 
<port>5060</port>
<name>2893</name> 
<displayName>2893</displayName> 
<autoAnswer> 
<autoAnswerEnabled>2</autoAnswerEnabled> 
</autoAnswer>
<callWaiting>3</callWaiting> 
<authName>2893</authName> 
<authPassword>8888</authPassword>
<sharedLine>false</sharedLine>
<contact>2893</contact> 
<forwardCallInfoDisplay> 
<callerName>true</callerName> 
<callerNumber>false</callerNumber> 
<redirectedNumber>false</redirectedNumber> 
<dialedNumber>true</dialedNumber> 
</forwardCallInfoDisplay> 
</line> 
</sipLines> 
<dialTemplate>dialplan.xml</dialTemplate> 
</sipProfile> 
<loadInformation>P0S3-8-12-00</loadInformation> 
</device>

SIP<MAC>.cnf
------------
image_version : "P0S3-8-12-00"
directory_url : ""
services_url : ""
logo_url : ""
http_proxy_addr : ""
http_proxy_port : ""
transfer_onhook_enabled : "1"
dscpForAudio : 184
date_format : "M/D/Y"
time_format_24hr : 0
dial_template : "dialplan"
sntp_server : "59.148.184.7"
sntp_mode : "Unicast"
time_zone : "EST"
dst_auto_adjust : 1
dst_offset : 01/00
dst_start_day : 0
dst_start_day_of_week : Sunday
dst_start_month : 3
dst_start_week_of_month : 2
dst_start_time : "02/00"
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_month : 11
dst_stop_week_of_month : 1
dst_stop_time : "02/00"
proxy1_address : "192.168.1.32"
proxy_backup : ""
proxy_emergency : ""
connection_monitor_duration : 120
line1_shortname : "2893"
line1_name : "2893"
line1_displayname : "2893"
line1_authname : "2893"
line1_password : "8888"
messages_uri : "8560"
line2_shortname : ""
line2_name : ""
line2_displayname : ""
line2_authname : ""
line2_password : ""
phone_label : "CISCO"
cnf_join_enable : "1"
rfc_2543_hold : "0"
call_hold_ringback : "2"
semi_attended_transfer : "1"
anonymous_call_block : "2"
callerid_blocking : "2"
dnd_control : "0"
sip_invite_retx : "6"
sip_retx : "10"
timer_invite_expires : "180"
timer_register_expires : "3600"
timer_register_delta : "5"
timer_keepalive_expires : "120"
timer_t1 : "500"
timer_t2 : "4000"
sip_max_forwards : "70"
enable_vad : "0"
dtmf_avt_payload : "101"
dtmf_db_level : "3"
user_info : "None"
stutter_msg_waiting : "2"
call_stats : "1"
start_media_port : "16384"
end_media_port : "32766"
phone_password : "cisco"
voip_control_port : "5060"
proxy_emergency_port : "5060"
outbound_proxy : ""
outbound_proxy_port : "5060"
proxy_register : "1"
dtmf_outofband : "avt"
autocomplete : "2"
network_media_type : "Auto"
local_cfwd_enable : "1"
call_waiting : "1"
preferred_codec : "none"
remote_party_id : "1"
telnet_level : 2

SIPdefault.cnf
-------------
# Image Version
image_version: "P0S3-8-12-00"

# Proxy Server
proxy1_address: "192.168.1.32"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""

# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:""
proxy3_port:""
proxy4_port:""
proxy5_port:""
proxy6_port:""

# Emergency Proxy info
proxy_emergency: "192.168.1.32"
proxy_emergency_port: "192.168.1.32"

# Backup Proxy info
proxy_backup: "192.168.1.32"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: "192.168.1.32"
outbound_proxy_port: "5060"


# NAT/Firewall Traversal
nat_enable: ""
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"

# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "g729"

# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"

# Enable VAD (0-disable (default), 1-enable)
enable_vad: "1"

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "1" ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"

# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: "1"

# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: "3"

# SIP Timers
timer_t1: "500" ; Default 500 msec
timer_t2: "4000" ; Default 4 sec
sip_retx: "11" ; Default 11
sip_invite_retx: "7" ; Default 7
timer_invite_expires: "180" ; Default 180 sec

# Setting for Message speeddial to UOne box
messages_uri: "*97"

#********* Release 2 new config parameters **********

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"

# Time Server
sntp_mode: "unicast"
sntp_server: "59.148.184.7"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: "1"

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)

# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)

# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)

# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100

# XML file that specifies the dialplan desired
dial_template: "dialplan"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"

#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"

#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"

# URL for external Phone Services
;services_url: "http://59.188.52.42/cisco/services/index_cisco.php"

# URL for external Directory location
;directory_url: "http://59.188.52.42/cisco/services/PhoneDirectory.php"

# URL for branding logo
;logo_url: "http://59.188.52.42/cisco/bmp/trixbox.bmp"

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Author Closing Comment

by:AXISHK
ID: 35239204
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