NO JOY That Darned Cisco 7970

Nallchan

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Im pulling my hair out with this 7970 Cisco Phone and about to just toss it in the trash and buy a Yealink Sip phone... ive spent in excess of 30+ Hours Fudging with this thing and this is my last try before I destroy this sucker out of frustration =(

I would really...REALLY appreciate some help with this..
Ill Offer up $30 via paypal to the person finally gets this phone working with my PBX.
Can Teamviewer/skype if needed.


I dont have call manager or an active Cisco Contract

1st - ive looked at the following sites a few times now... (im missing something!)
http://nerdvittles.com/?p=149 - Introducing the Cisco 7970 WonderPhone
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP

Here are the Config files ive compiled

**Note: 10.1.10.200 = PBX
10.1.10.25 = DHCP / BootFTP

phonescreen.jpg


phonelog.jpg


I dont have TD-SIP.JAR or ITLSEP & CTLSEP.TLV (are they Required?)
ftplog.jpg



XMLdefault.cnf.xml


Code:
<Default>
<callManagerGroup>
    <members>
       <member priority="0">
          <callManager>
             <ports>
                <ethernetPhonePort>2000</ethernetPhonePort>
                <mgcpPorts>
                   <listen>2427</listen>
                   <keepAlive>2428</keepAlive>
                </mgcpPorts>
             </ports>
             <processNodeName>10.1.10.200</processNodeName>
          </callManager>
       </member>
    </members>
</callManagerGroup>
<loadInformation30007 model="CP-7970G">CP7912080000SIP060111A</loadInformation30007>
<loadInformation115 model="CP-7970G">SIP70.9-4-2SR1-1S</loadInformation115>
<loadInformation6 model="CP-7970">SIP70.9-4-2SR1-1S</loadInformation6>
<loadInformation30006 model="CP-7970">SIP70.9-4-2SR1-1S</loadInformation30006>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
 
Last edited:

Nallchan

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SEP#MacAddressOfPhone#.CNF.XML
Code:
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId> 
<sshPassword>MYPHONEPASSWORD</sshPassword>
<devicePool> 
<dateTimeSetting> 
    <dateTemplate>M/D/Ya</dateTemplate> 
    <timeZone>Eastern Standard/Daylight Time</timeZone>
    <ntps> 
         <ntp> 
             <name>169.229.70.183</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>10.1.10.200</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>1200</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>none</preferredCodec> 
<dtmfAvtPayload>101</dtmfAvtPayload> 
<dtmfDbLevel>3</dtmfDbLevel> 
<dtmfOutofBand>avt</dtmfOutofBand> 
<alwaysUsePrimeLine>false</alwaysUsePrimeLine> 
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> 
<kpml>3</kpml>
<natReceivedProcessing>true</natReceivedProcessing> 
<natEnabled>false</natEnabled> 
<natAddress>10.1.10.200</natAddress> 
<phoneLabel></phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats> 
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort> 
<stopMediaPort>32766</stopMediaPort>
<sipLines>
  <line button="1">
       <featureID>9</featureID> 
       <featureLabel>1337</featureLabel> 
       <proxy>10.1.10.200</proxy> 
       <port>5060</port>
       <name>1337</name>
       <displayName>1337</displayName> 
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer> 
       <callWaiting>3</callWaiting>
       <authName>1337</authName> 
       <authPassword>OTHERPASSWORDFROMPBX</authPassword> 
       <sharedLine>false</sharedLine>
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
       <messagesNumber>*97</messagesNumber> 
       <ringSettingIdle>4</ringSettingIdle> 
       <ringSettingActive>5</ringSettingActive>
       <contact>1337</contact> 
       <forwardCallInfoDisplay> 
          <callerName>true</callerName> 
          <callerNumber>false</callerNumber> 
          <redirectedNumber>false</redirectedNumber> 
          <dialedNumber>true</dialedNumber> 
       </forwardCallInfoDisplay> 
    </line>
<line button="2">
       <featureID>9</featureID> 
       <featureLabel>1338</featureLabel> 
       <proxy>10.1.10.200</proxy> 
       <port>5060</port>
       <name>1338</name>
       <displayName>1338</displayName> 
       <autoAnswer> 
          <autoAnswerEnabled>2</autoAnswerEnabled> 
       </autoAnswer> 
       <callWaiting>3</callWaiting>
       <authName>1338</authName> 
       <authPassword>PASSWORDTHATSLISTEDINPBX</authPassword> 
       <sharedLine>false</sharedLine>
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
       <messagesNumber>1338</messagesNumber> 
       <ringSettingIdle>4</ringSettingIdle> 
       <ringSettingActive>5</ringSettingActive>
       <contact></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>dialplan.xml</dialTemplate> 
</sipProfile>
<commonProfile> 
<phonePassword>PHONEPASSWORD</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess> 
<callLogBlfEnabled>2</callLogBlfEnabled> 
</commonProfile>
<loadInformation>SIP70.9-4-2SR1-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>1</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability> 
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess> 
<spanToPCPort>0</spanToPCPort> 
<loggingDisplay>1</loggingDisplay> 
<loadServer></loadServer> 
</vendorConfig>
<versionStamp></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>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL>http://10.1.10.200/cisco/services/authentication.php</authenticationURL> 
<directoryURL>http://10.1.10.200/xmlservices/PhoneDirectory.php</directoryURL> 
<idleURL>http://10.1.10.200/xmlservices/index.php</idleURL> 
<informationURL></informationURL>
<messagesURL></messagesURL> 
<proxyServerURL></proxyServerURL> 
<servicesURL></servicesURL>
<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> 
</device>

SIPDefault.cnf
Code:
image_version: "SIP70.9-4-2SR1-1S"
# proxy1_address: "10.1.10.200"
# proxy2_address: "10.1.10.200"
# proxy3_address: "10.1.10.200"
# proxy4_address: "10.1.10.200"

# Proxy Server Port
# proxy1_port:"5060"
# proxy2_port:"5060"
# proxy3_port:"5060"
# proxy4_port:"5060"
proxy_emergency: ""
proxy_emergency_port: "5060"
proxy_backup: ""
proxy_backup_port: "5060"
outbound_proxy: ""
outbound_proxy_port: "5060"

nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "1"
dyn_dns_addr_1: ""
dyn_dns_addr_2: ""
dyn_tftp_addr: "10.1.10.25”
tftp_cfg_dir: "./"

proxy_register: "1"
timer_register_expires: "120"
preferred_codec: "none"
tos_media: "5"
enable_vad: "0"
dial_template: "dialplan"
network_media_type: "auto"
autocomplete: "1"
telnet_level: "2”

cnf_join_enable: "1"
semi_attended_transfer: "0"
call_waiting: "1"
anonymous_call_block: "0"
callerid_blocking: "0"
dnd_control: "0"
dtmf_inband: "1"
dtmf_outofband: "avt"
dtmf_db_level: "3"
dtmf_avt_payload: "101”
timer_t1: "500"
timer_t2: "4000"
sip_retx: "10"
sip_invite_retx: "6"
timer_invite_expires: "180"

sntp_mode: "directedbroadcast"
sntp_server: "us.pool.ntp.org"
time_zone: "CST"
time_format_24hr: "1"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Mon"
dst_start_week_of_month: "1"
dst_start_time: "2"
dst_stop_month: "Nov"
dst_stop_day: "1"
Firmware to use
Asterisk server address(es)
and port(s)
Advanced and failover
server locations
Firewall, NAT
Traversal, DNS and
t*f*t*p settings
Asterisk server communication
and registration settings
Calling features enabling
Quality of Service (QoS) and
SIP protocol settings
Time and DST settings
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: ""
dst_stop_time: "2"
dst_auto_adjust: "1"

messages_uri: "*99"
services_url: "”
directory_url: ""
logo_url: "t*f*t*p:1.bmp”
http_proxy_addr: ""
http_proxy_port: 80
remote_party_id: 0
 
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People have wasted lifetimes on this...Unless you deal with them all the time it will be one thing after another....

Get the Yealink t46g or t48g you;ll be happier
 

Bill Coghill

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I had a couple of 7965s working under Piaf-Green, but since migrating to the new Incredible PBX 13 I've not been able to get them to register. What version are you using?

I tired as chan_sip devices, but they wouldn't connect on port 5061 correctly (they ignored the port setting in the xml file). I actually considered rolling back to an earlier version as the phones were rock solid. defining them as PJSIP using 5060 didn't work for some reason. I have spent many hours on these and its been an exercise in frustration !

Looking at the logs from the phones it looks like its missing the language files. Let me fire up my laptop and check the t*f*t*p file tree I have for the 7965s. I'll also post the working XML config files I used under PIAF-Green in case they help. Unfortunately they don't work under Incredible PBX 13 as the phones won't register.

Bill.
 

rjaiswal

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Hmmm. Interesting. My only concern is that it might force the SPA504g phones I have in to sccp mode rather than sip where they just work.

Time to experiment !

Bill.
There is a setting on the 504g, somewhere in the advanced area, or network area, where you can tell the phone to ignore any sccp requests.
 

Nallchan

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are you suggesting using SCCP firmware instead of the SIP? (I went out of my way to Flash them to SIP reading the tutorials
Thoughts?
 

rjaiswal

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are you suggesting using SCCP firmware instead of the SIP? (I went out of my way to Flash them to SIP reading the tutorials
Thoughts?
Yup. It doesn't have as many bugs, and you can configure shared call/line appearances.
 

Adrian_Novice

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got 2 x 7970G, 1 x 7971G (SIP F/W ) and 7921 (SCCP) running fine on my Raspberry Pi, using incrediblePBX image. Using DNSMASQ & opentftp on my QNAP NAS to serve up files via t*f*t*p. You'll need the jar files. extract from t*f*t*p log:
[17-Feb-16 09:53:36] Client 172.16.32.199:49152 /share/MD0_DATA/Public/7970G/CTLSEP002155D5FxxD.tlv, No Such File/No Access
[17-Feb-16 09:53:36] Client 172.16.32.199:49153 /share/MD0_DATA/Public/7970G/SEP002155D5FxxD.cnf.xml, 17 Blocks Served
[17-Feb-16 09:53:48] Client 172.16.32.199:49154 /share/MD0_DATA/Public/7970G/English_United_Kingdom/td-sip.jar, 139 Blocks Served
[17-Feb-16 09:54:05] Client 172.16.32.199:49155 /share/MD0_DATA/Public/7970G/dialplan.xml, 1 Blocks Served
[17-Feb-16 09:54:05] Client 172.16.32.199:49156 /share/MD0_DATA/Public/7970G/SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml, No Such File/No Access
[I've masked some of the MAC with xx]

if you're building from scratch suggest 11.20 freepbx;
http://pbxinaflash.com/community/resources/cisco-7900-series-presence-patch.11/

well documented here;
http://docs.acsdata.co.nz/asterisk-cisco/document-overview.shtml

HTH

Adrian
 

Nallchan

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Got a "Yealink T48G" - works PERFECTLY... still trying to get the Ciscos to work... (will try the links above this week) - havent had time in the past few days
 
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Having spent WEEKS on the 7941 handsets a few years back I wrote an extensive guide about them here:

http://pbxinaflash.com/community/resources/cisco-7900-series-tutorial-v1-1.25/

The file you want to be concentrating on is the sepxxx.cnf.xml - one single line out and the handsets won't work, they won't tell you why either. Go through my guide and compare your sep file with mine linked, particularly pay attention to transport settings. Even try copying my sep file and adding your details to log on where it matters.

Once they are set up these phones are fantastic. Setting them up is a real PITA but if you persevere you will do it, speaking as someone in your shoes a few years back I know exactly what you're going through.
 

Nallchan

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I've skimmed the guide - Been super busy - havent forgotten - will attempt this very soon and report back.
 

siliconvalley785

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Code:
<sipLines>
  <line button="1">
       <featureID>9</featureID>
       <featureLabel>1337</featureLabel>
       <proxy>USECALLMANAGER</proxy> ** change this to USECALLMANAGER for all SIP Lines
       <port>5060</port>
       <name>1337</name>
       <displayName>1337</displayName>
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>
       <callWaiting>3</callWaiting>
       <authName>1337</authName>
       <authPassword>OTHERPASSWORDFROMPBX</authPassword>
       <sharedLine>false</sharedLine>
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
       <messagesNumber>*97</messagesNumber>
       <ringSettingIdle>4</ringSettingIdle>
       <ringSettingActive>5</ringSettingActive>
       <contact>1337</contact>
       <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>false</callerNumber>
          <redirectedNumber>false</redirectedNumber>
          <dialedNumber>true</dialedNumber>
       </forwardCallInfoDisplay>
    </line>

Try this and let me know. This is what worked for me when i was messing around with these phones a while back.

Pritesh
 

Nallchan

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Code:
<sipLines>
  <line button="1">
       <featureID>9</featureID>
       <featureLabel>1337</featureLabel>
       <proxy>USECALLMANAGER</proxy> ** change this to USECALLMANAGER for all SIP Lines
       <port>5060</port>
       <name>1337</name>
       <displayName>1337</displayName>
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>
       <callWaiting>3</callWaiting>
       <authName>1337</authName>
       <authPassword>OTHERPASSWORDFROMPBX</authPassword>
       <sharedLine>false</sharedLine>
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
       <messagesNumber>*97</messagesNumber>
       <ringSettingIdle>4</ringSettingIdle>
       <ringSettingActive>5</ringSettingActive>
       <contact>1337</contact>
       <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>false</callerNumber>
          <redirectedNumber>false</redirectedNumber>
          <dialedNumber>true</dialedNumber>
       </forwardCallInfoDisplay>
    </line>

Try this and let me know. This is what worked for me when i was messing around with these phones a while back.

Pritesh

Made the "USECALLMANAGER" change (which I had tried before)

no change - Sits on Registering for 15 - 20 min.. then Reboots and starts the process all over again - see below pictures.
cisco1.jpg



cisco2.jpg
 

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