Cisco 7970 and 3cx - It works :)

Discussion in '3CX Phone System - General' started by riceboi, Apr 11, 2007.

  1. riceboi

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    Hi Guys,

    I'd like to know if anyone here has had any luck with using Cisco 7970 Handsets to work with 3cx

    The the configuration files which the 7970 uses are not quite the same as the 7940 as the config file which the 7970 uses is a sep[mac address].cnf.xml

    Could anyone share some ideas or help ?

    Cheers..

    Dion
     
  2. riceboi

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    Update:

    I've had a stab at trying to get this working.

    So far I have configured 3 7970s which have all authenticated with 3cx.
    My next hurdle is that when i attempt to dial either extensions configured on the 3 handsets, i am denied with a busy tone.

    I am unsure as to why this is the case though here is a copy of my 3cx service status log.

    00:00:12.541 CallConf::eek:nIncoming Can not resolve target for call from "Dion Ignacio"<sip:100@192.168.0.9>;tag=001243d1defa00031e594670-e327c1fe to <sip:1@192.168.0.9;user=phone>
    00:00:12.541 CallConf::Rejected Call (C:14) is rejected: Target is not resolved
    00:00:12.491 CallConf::eek:nIncoming Incoming call from Ext.100 to sip:1@192.168.0.9
    23:59:47.746 ServRegs::eek:nAdd Registered: Ext.102
    23:59:33.796 ServRegs::eek:nAdd Registered: Ext.100
    23:59:20.026 ServRegs::eek:nAdd Registered: Ext.101

    Can anyone shed some light on the issue ?

    Cheers,

    Dion
     
  3. riceboi

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    Update:

    I again had to check my config files for each phone and found out that the time register expires was set the default setting of 3600 and not 60.
    i found out after this was corrected that all 3 phones can now call each other on there extensions :lol:

    now that the hard part is out of the way, im going to do a little more tweaking with the settings.

    i was surprised that these 7970s actually support g711alaw aswell when they usually a different compression on CCM

    I am taking delivery of a spa-3102 this week. I hope to integrate this piece of hardware once its recieved and test its capabilities over a pstn line and a sip provider which comes as a free trial :lol:

    will keep you posted later through the week.

    Cheers,

    Dion
     
  4. riceboi

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    Update:

    Hi Again guys,

    Ive installed the spa-3102 and have been able authenticate between it and 3cx.

    Although i am having problems with attempting to dial outbound via my pstn line and calls coming into my pstn line.

    here is a copy of my error shown in 3cx

    14:17:20.535 StratLink::eek:nHangUp Call(C:71): got Hang-Up from Ln:10001@Telstra PSTN; reason: BYE
    14:16:48.549 CallLegImpl::eek:nConnected Established media channel for Ln:10001@Telstra PSTN: remote=127.0.0.1:16478; local=:23730
    14:16:48.539 StratInOut::eek:nConnected Call from Ext.100 to Ln:10001@Telstra PSTN is established

    As you can see 3cx does talk with my spa3102 but the spa3102 does not initiate a call.

    Any ideas on this one ? Im starting to pull my hair out :p

    Firmware has been upgraded on the spa-3102 and can make internal calls with the 7970 handsets.

    Urgent help would be much appreciated.

    Cheers,

    Dion
     
  5. Anonymous

    Anonymous Guest

    http://www.dayboro.info/3cx/Take2/NetworkView.htm


    start there, you are in AU so it should work.

    If not get back on the blower to me :).
     
  6. riceboi

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    I've followed and checked all settings with my spa-3102 and 3cx as shown on the website you gave, (its very helpful by the way :))but its still not letting me recieve or dial outbound via my pstn line.

    it appears im still getting the same errors.

    00:35:11.170 StratLink::eek:nHangUp Call(C:81): got Hang-Up from Ln:10001@Telstra PSTN; reason: BYE
    00:34:39.194 CallLegImpl::eek:nConnected Established media channel for Ln:10001@Telstra PSTN: remote=127.0.0.1:16426; local=:20086
    00:34:39.184 StratInOut::eek:nConnected Call from Ext.100 to Ln:10001@Telstra PSTN is established

    I dont think my spa3102 is initiating the dial once 3cx requests it to after it authenticates its request to dial.

    a little lost on what the problem could be.

    going back to basics, do existing analog phones need to be disconnected and only the spa3102 to be connected to the pstn line ?
    will that affect how it works ?

    and yes you are right, i am from the land of oz.. so this should work..
     
  7. Anonymous

    Anonymous Guest

    No you can have analogue phones on the same line, does not hurt to have them removed during testing.

    Is your ADSL filter between the socket and the SPA still ok?

    Other question, does the phone you call actually ring?

    Ok some minor alterations:
    To connect straight to the PSTN (this if you do not want to use your VSP for outbound calls):

    In the Line 1 Tab (enable the line)
    -In your dial plan enter something like <#9:><:mad:gw0>, #9 being what --you dial to get through to the PSTN.
    - Codec: G711U
    - Use Pref Codec Only: No
    - Dialplan:
    (000S0<:mad:gw0>|106S0<:mad:gw0>|1[38]xxxxxxxxS0<:mad:gw0>|13[1-9]xxxS0<:mad:gw0>|1[9]xxxxxxxxS0<:mad:gw0>|0[23478]xxxxxxxxS0|[2-9]xxxxxxxS0|001xxxx.S5|xxx.<:mad:gw0>|<*,:*>xx.|<#,:><:mad:gw0>)


    In the PSTN Tab
    - Leave dial plan 1 empty
    - make sure VoIP-To-PSTN Gateway Enable is set to yes
    - Line 1 VoIP Caller DP: 1
    - VoIP Caller Auth Method:pIN
    - Codec: G711U
    - Use Pref Codec Only: No

    Dialplan:
    (000S0<:mad:gw0>|106S0<:mad:gw0>|1[38]xxxxxxxxS0<:mad:gw0>|13[1-9]xxxS0<:mad:gw0>|1[9]xxxxxxxxS0<:mad:gw0>|0[23478]xxxxxxxxS0|[2-9]xxxxxxxS0|001xxxx.S5|xxx.<:mad:gw0>|<*,:*>xx.|<#,:><:mad:gw0>)
     
  8. riceboi

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    thanks for you help on this one.. its much appreciated.

    Im not running adsl over.
    Im using Telstra cable actually.
    How i have it set up here is my c3x server is running as a dhcp and tftp.
    My internet g/w is my router address and spa has been allocated a static ip address on the ethernet interface.

    i've put in the settings which you've mentioned above but still no avail.

    When i initiate a call to an outbound no. the phone does not give a dial signal but connects for 30secs (as there is a timer on the screen) and then disconnects. 3cx then gives the same errors as i've previously mentioned above.

    do you think its got to do with my dial plans on spa or spa is not able to pick up or unhook the line ? i dont hear the click from spa when i make an outbound external call
     
  9. Anonymous

    Anonymous Guest

    Ok so it is not your ADSL filter :).

    Cable (bfrffffrrrr).

    Anyway, disconnects occur for several reasons, in general it is because the ATA does not sees any traffic and thinks the call is finished so hangs up.

    If you have set the call progress tones exactly like the examples it should work.

    Questions;
    What firmware are you using
    It might be something Cisco specific that does not cater for the call progress tones required.

    Ill have a look arround.
     
  10. riceboi

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    wel i've set the call progress tones as shown and rechecked them, they all seem to be fine at the moment.

    firmware on the spa is the latest one available from the linksys site.
    unsure about the call progress... but thanks for your help... :)
     
  11. Anonymous

    Anonymous Guest

    Call is 30 seconds, this is about the time frame for SIP regestering.

    What might happen (and this is a cable issue) that the sip loses the registration and drops the call (hence the BYE).

    Try to decrease the time taken for SIP registration in your ATA, I hope it you pole more frequent the cable connection stays up.

    (Hmm i have to write something about this SIP 30 and 35 seconds thingy.)
     
  12. riceboi

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    I've done a bit of research here and am thinking if this is going to cause an issue with the phone.

    taken from: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP


    Specifying the networkLocale gives me NZ tones, which makes the phone almost feel like a normal phone for the rest of the house. These networkLocale files are on CCO - look under CallManager sections for these. If you don't specify this the phone will run with the standard US default.

    Setting the locale does not seem to reset the dialtone to anything local, but it does customise the ringing tone and other PSTN tones.

    <networkLocale>New_Zealand</networkLocale>

    <networkLocaleInfo>
    <name>New_Zealand</name>
    <version>5.0(2)</version>
    </networkLocaleInfo>

    <deviceSecurityMode>1</deviceSecurityMode>


    I am wondering if this is causing any issues. 7970s require to tftp a config file for them to work (this file aswell contains auth details, proxy, etc)
    im going to give this a try tonight and get this set correctly. Cant do it now as im at work.

    Another way to test this is install a softphone on one of my pcs and check to see if it can dial out via the pstn, probably help rule out the ip phones being the culprit :)
     
  13. Anonymous

    Anonymous Guest

    Don't tell me you have not tried a soft phone alread <<shaking my "wise" old head>> (LOL).

    Give that a go, but you might be onto somthing. If it does not work I will ask one of the guys here at work (CCIE types) how call manager and sip do things.

    Never liked Cisco in that respect but than again SPA's are Cisco now :).
     
  14. 3CXsupport

    3CXsupport New Member
    3CX Staff

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    Hi,

    Thanks for pointing out the Locale issue for cisco phones. Defenitly Do try a softphone and see what happens, please keep us posted.
     
  15. Anonymous

    Anonymous Guest

    If you are really desperate i got some "old" Cisco VoIP books you can look into.

    But I only have one copy (due to copyrights etc) so drop me a PM if you (riceboi) want them.
     
  16. riceboi

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    It Works !!!!

    Well its been a long week since i've started this project i've finally got it working.. well outbound is :p

    havent tested inbound yet.. its a little too late to test it as people are sleeping already (1am in the morning)

    Anyways i didnt make any changes to the 7970 configuration, nor did i make a changes on 3cx other than what itfarmer advises on his website link (very helpful :) )

    it appears that there were some issues with the spa3102 where the configurations needed to be a little different to work with the 7970s. With this i've had to simplify my dial plans for the time being.

    Will give further information on how i believe i got it to works and info on how to get it working if anyone requests it.

    but for now, IT WORKS... and im going to get some sleep, work is a mere 6 hours sleep away :)
     
  17. Anonymous

    Anonymous Guest

    WHAT.... you get to work at 7am what's the go there, half day's is it :) LOL.

    Good to see you got it working, and yeah I would like to have a copy/description of your config wouldbe good.

    I think 3cx is trying to collect as many possible configurations and phones as possible so they can test the different platforms before they endorse it :).
     
  18. riceboi

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    i hope to get a config written up during the week and possibly for 3cx if they are interested as you've noted.

    I want to save the current config i have at the moment and start over again just to ensure i know what is going on and see if that is the way it can only work.

    but yes, it works now, i just need to push it through its paces and see how reliable it is, plus im still yet to sign up for a sip provider to test that side of things.

    im really only half done at the moment when i think about :)

    thanks for you help and input aswell, it was greatly appreciated :)
     
  19. riceboi

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    7970 Config & 3cx

    Hi Guys,

    My apologies to those who were waiting for this information, Time was a cut a bit short for me to provide a reply as I went overseas to Japan for a holiday.

    Anyways, to cut the crap, here is the information needed to get a Cisco 7970 SIP firmware loaded IP Phone to work with 3cx.

    Some helpful reading i would recommend would be the following links before attempting this to familiarise yourself with how the phone works. The online video link below will give a clear understanding of this.

    Understanding the config xml file for Cisco IP Phones & SIP:
    http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP

    Understanding the Cisco 7970:
    http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP

    Video on configuring a Cisco 7970 - using Asterisk but very similar to 3cx when registering
    http://asterisktutorials.com/showproduct.php?ProductID=10

    As mentioned in the helpful reading links provided above, after installing 3cx on your windows based pc/server, you will need to install a tftp server and a dhcp server on either the same pc with 3cx or another pc.

    If you are running 3cx on a win2k3, you can use the DHCP server with option 66 to point to the tftp server.

    Once you've installed and configured your DHCP server, ensure that you point option 66 'tftp server ip address' to the IP address which you wish to install your tftp server is installed on. This could be the same ip address as your 3cx. I have it configured this way.

    Install your tftp server on the pc/server which you've pointed the TFTP server ip address on your dhcp server. Once installed you will need to drag your firmware SIP files for the Cisco IP Phone (7970) into the tftp root.

    The Next Step is to then create the XMLDefault.cnf.xml text file
    This file i required by the IP Phone to load the correct firmware and load files for the phone to work.
    This is what the phone looks for first before refering to the SEP(mac address).cnf.xml file

    The Contents of the file which need to be pasted in are:

    <Default>
    <callManagerGroup>
    <members>
    <member priority="0">
    <callManager>
    <ports>
    <ethernetPhonePort>2000</ethernetPhonePort>
    <mgcpPorts>
    <listen>2427</listen>
    <keepAlive>2428</keepAlive>
    </mgcpPorts>
    </ports>
    <processNodeName></processNodeName> ** 3cx server name or ip address **
    </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), you will need to change this accordingly to the load file which you have in your tftp root. Careful, its case sensitive **
    <authenticationURL></authenticationURL>
    <directoryURL></directoryURL>
    <idleURL></idleURL>
    <informationURL></informationURL>
    <messagesURL></messagesURL>
    <servicesURL></servicesURL>
    </Default>

    Once updated with the correct load file, save the file and load it into your tftp root


    In 3cx you will then need to create your extension and keep in mind the Firstname/Lastname, sip authentication user/pass, Outbound Caller ID and extension number as you will need to statically set this for the 7970s config file you wish to register.

    Next you will have to create an xml filename in notepad or in a text editor in the following format:

    SEP<Mac Address>.cnf.xml

    This file needs to match the MAC Address of the IP Phone you wish to register with 3cx

    eg. SEP1234ABCDEFG0.cnf.xml

    Once created you will need to paste the following config information & edit where required to register the phone with 3cx

    <device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>admin</sshUserId> ** username of 7970 ip phone when logging into ip phone via SSH **
    <sshPassword>testbed</sshPassword> ** password of 7970 ip phone when logging into ip phone via SSH **
    <devicePool>
    <dateTimeSetting>
    <dateTemplate>D-M-YA</dateTemplate> ** Date format **
    <timeZone>Central Standard/Daylight Time</timeZone> ** Select your timezone. See this link for more info: http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cme34/cme_cr/cme_t1ht.htm#wp1071293 **
    <ntps>
    <ntp>
    <name>A.B.C.D</name> ** Enter the server name or IP of your NTP Server **
    <ntpMode>Unicast</ntpMode>
    </ntp>
    </ntps>
    </dateTimeSetting>
    <callManagerGroup>
    <members>
    <member priority="0">
    <callManager>
    <ports>
    <ethernetPhonePort>2000</ethernetPhonePort>
    <sipPort>5060</sipPort>
    <securedSipPort>5061</securedSipPort>
    </ports>
    <processNodeName>A.B.C.D</processNodeName> ** Enter the name/IP of your 3CX Server **
    </callManager>
    </member>
    </members>
    </callManagerGroup>
    </devicePool>
    <sipProfile>
    <sipProxies> ** Self explanatory, you can put the 3cx server ip addresses here on port 5060 (optional) **
    <backupProxy></backupProxy>
    <backupProxyPort></backupProxyPort>
    <emergencyProxy></emergencyProxy>
    <emergencyProxyPort></emergencyProxyPort>
    <outboundProxy></outboundProxy>
    <outboundProxyPort>5060</outboundProxyPort>
    <registerWithProxy>true</registerWithProxy>
    </sipProxies>
    <sipCallFeatures> ** Do not modify any of these settings **
    <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> ** Do not modify any of these settings **
    <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> ** Select your preferred codec - G711ulaw, G711alaw, or G729a.
    NOTE: All 3 codecs will be sent, but the preferred one will be listed first. But for 3cx put in G711alaw **
    <dtmfAvtPayload>101</dtmfAvtPayload>
    <dtmfDbLevel>3</dtmfDbLevel>
    <dtmfOutofBand>avt</dtmfOutofBand>
    <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
    <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
    <kpml>3</kpml>
    <natEnabled>0</natEnabled>
    <natAddress></natAddress>
    <phoneLabel>0296862222</phoneLabel> ** Name that will display in upper right hand corner of the phone, its usually best to put your external PSTN or DID number. **
    <stutterMsgWaiting>2</stutterMsgWaiting>
    <callStats>false</callStats>
    <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
    <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
    <startMediaPort>16384</startMediaPort>
    <stopMediaPort>32766</stopMediaPort>
    <sipLines>
    <line button="1">
    <featureID>9</featureID> ** This must be 9 (this is not for acceessing an outside line) for a regular SIP phone extension.
    Use 21 here to create a speed dial **
    <featureLabel>6755</featureLabel> ** Text that displays next to line button, usually best to put the extension no. of the IP Phone here. **
    <proxy>A.B.C.D</proxy> ** ip address or server name of your 3cx server**
    <port>5060</port> ** SIP Port 3cx communicates on **
    <name>6755</name> ** SIP registration name - this is the firstname and last name you create the extension with in 3cx **
    <displayName>Steve</displayName> ** SIP Display name **
    <autoAnswer>
    <autoAnswerEnabled>2</autoAnswerEnabled>
    </autoAnswer>
    <callWaiting>3</callWaiting>
    <authName></authName> ** sip authentication username you set in 3cx **
    <authPassword></authPassword> ** sip authenication password you set in 3cx **
    <sharedLine>false</sharedLine>
    <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
    <messagesNumber>505</messagesNumber>
    <ringSettingIdle>4</ringSettingIdle>
    <ringSettingActive>5</ringSettingActive>
    <contact></contact>
    <forwardCallInfoDisplay>
    <callerName>true</callerName>
    <callerNumber>false</callerNumber>
    <redirectedNumber>false</redirectedNumber>
    <dialedNumber>true</dialedNumber>
    </forwardCallInfoDisplay>
    </line> ** you can use the extra buttons along the top right hand side of phone as speed dials by using the following below as an example
    <line button="3">
    <featureID>21</featureID> ** the feature id needs to be set as 21 if its to be used as a speed dial **
    <featureLabel>Lounge</featureLabel> ** Label on phone next to speed dial button **
    <speedDialNumber>212</speedDialNumber> ** the ext or phone number speed dial will dial/ring **
    </line>
    <line button="4">
    <featureID>21</featureID>
    <featureLabel>Sunroom</featureLabel>
    <speedDialNumber>213</speedDialNumber>
    </line>
    <line button="5">
    <featureID>21</featureID>
    <featureLabel>Cordless</featureLabel>
    <speedDialNumber>214</speedDialNumber>
    </line>
    <line button="7"> ** you can configure the speed dial button to dial an external number aswell **
    <featureID>21</featureID>
    <featureLabel>Mum</featureLabel>
    <speedDialNumber>92860002</speedDialNumber>
    </line>
    <line button="8">
    <featureID>21</featureID>
    <featureLabel>Dad</featureLabel>
    <speedDialNumber>92860001</speedDialNumber>
    </line>
    </sipLines>

    <voipControlPort>5060</voipControlPort>
    <dscpForAudio>184</dscpForAudio>
    <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
    <dialTemplate>dialplan.xml</dialTemplate> ** dial plan set for the phone locally not for 3cx **
    </sipProfile>
    <commonProfile>
    <phonePassword></phonePassword>
    <backgroundImageAccess>true</backgroundImageAccess>
    <callLogBlfEnabled>2</callLogBlfEnabled>
    </commonProfile>

    <loadInformation>SIP41.8-0-2SR1S</loadInformation> ** SIP load file name needs to be inserted here - refer firware files loaded in tftp for the cisco phone. Do not add ".load"**
    <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>
    <spanToPCPort>1</spanToPCPort>
    <loggingDisplay>1</loggingDisplay>
    <loadServer></loadServer>
    </vendorConfig>
    <versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
    <networkLocale></networkLocale>
    <networkLocaleInfo>
    <name></name>
    <version>5.0(2)</version>
    </networkLocaleInfo>
    <deviceSecurityMode>0</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>4</transportLayerProtocol>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
    <capf>
    <phonePort>3804</phonePort>
    </capf>
    </capfList>
    <certHash></certHash>
    <encrConfig>false</encrConfig>
    </device>

    Note: where i have added comments in the config file "** blah blah blah **" please remove these.

    Once completed, Save the file and copy it to the TFTP root directory.

    Lastly you will need to create the dialplan.xml text file

    you will then need to paste the following information.

    <DIALTEMPLATE>
    <TEMPLATE MATCH="#..." Timeout="5" User="Phone" />
    <TEMPLATE MATCH="*" Timeout="5" User="Phone" />
    </DIALTEMPLATE>

    This basically means by default '*' means anything and that it will times out after 5 seconds.
    Users must push 'Dial' or '#' to connect if they don't want to wait 5 seconds

    Once you have completed all of the follow, you should be able to boot your Cisco 7970 up.
    It should then recognise the config details and register.

    If it has not already been loaded with the SIP firmware you will need to push the phone to obtain the correct sip firmware.
    To do this you will need to hold the '#' button down and turn the phone on. Remain holding the '#' button until orange lights cycle down where the speed dial buttons are. Once its begins to cycle release the '#' button and press each of the following button after each other: 1 2 3 4 5 6 7 8 9 0 * #, this should then turn the orange speed dial lights to red. This will then begin the firmware reload on the phone which pick up the files off the TFTP server.

    Once it has completed the firmware upgrade, it will then register the phone using the SEP(mac address).cnf.xml file which you configured and saved in the tftp root.

    You should see the phone registered in 3cx under the Line Status link in 3cx and be able to dial your IP phones extensions from each other.

    Please be aware that the message wait indicator (MWI) does not work on some of the firmware releases. Im currently using the latest firmware release and where the MWI works to no avail when messages are left.

    As for configuring the SPA-3102 to use these phones with 3cx, i'll have to post that information tomorrow as i've run out of time while at work :)


    Any questions, feel free to ask

    To 3cx : do i get a free copy of business edition for providing this information :)

    Cheers,

    Dion
     
  20. mfaster

    Joined:
    Jun 19, 2007
    Messages:
    21
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    7970 DTMF does not work

    I am running Firmware 8-0-3 for the 7970. I cannot get DTMF tones to work? Anybody have any suggestions?
     

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