ABOUT CISCO IP PHONE 7945G & 7940 to SIP

[bayula1977]

I have Cisco 7945g and 7940 Phone to upgrade to SIP.I have downloaded the Firmware from cisco site but the issue im having is SEPMAC config and DialPlan with Default config to use. I have tried several samples found online but creating more issue. my asterisk PBX IP is 10.10.10.1. my account Sip account is 101 while the password is LL101 as well. Please i need your assistance on how to go about this.I will be very greatful if i can get a configure sample that will work asterisk box so that i can modify. 

[Leo Laohoo]

Post or attach the SEPmacaddress files.

[bayula1977]

<device> 
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId> 
<sshPassword>password</sshPassword> 
<devicePool> 
   <dateTimeSetting> 
      <dateTemplate>D/M/YA</dateTemplate> 
      <timeZone>Central Standard/Daylight Time</timeZone> 
      <ntps> 
         <ntp> 
            <name>172.100.101.229</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>172.100.101.229</processNodeName> 
            </callManager> 
         </member> 
      </members> 
   </callManagerGroup> 
</devicePool> 
<sipProfile> 
   <sipProxies> 
      <backupProxy></backupProxy> 
      <backupProxyPort>5060</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>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> 
   <dtmfAvtPayload>101</dtmfAvtPayload> 
   <dtmfDbLevel>3</dtmfDbLevel> 
   <dtmfOutofBand>avt</dtmfOutofBand> 
   <alwaysUsePrimeLine>false</alwaysUsePrimeLine> 
   <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> 
   <kpml>3</kpml> 
   <phoneLabel>PHONE TITLE</phoneLabel> 
   <stutterMsgWaiting>1</stutterMsgWaiting> 
   <callStats>false</callStats> 
   <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
   <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> 
   <sipLines> 
      <line button="1"> 
         <featureID>9</featureID> 
         <featureLabel>Bunmi</featureLabel> 
         <proxy>172.100.101.229</proxy> 
         <port>5060</port> 
         <name>101</name> 
         <displayName>101</displayName> 
         <autoAnswer> 
            <autoAnswerEnabled>2</autoAnswerEnabled> 
         </autoAnswer> 
         <callWaiting>3</callWaiting> 
         <authName>101</authName> 
         <authPassword>ln101</authPassword> 
         <sharedLine>false</sharedLine> 
         <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
         <messagesNumber>*99</messagesNumber> 
         <ringSettingIdle>4</ringSettingIdle> 
         <ringSettingActive>5</ringSettingActive> 
         <contact>101</contact> 
         <forwardCallInfoDisplay> 
            <callerName>true</callerName> 
            <callerNumber>false</callerNumber> 
            <redirectedNumber>false</redirectedNumber> 
            <dialedNumber>true</dialedNumber> 
         </forwardCallInfoDisplay> 
      </line> 
      <line button="2"> 
         <featureID>20</featureID> 
         <featureLabel>Menu</featureLabel> 
         <serviceURI>http://example.domain.ext/services/menu.xml</serviceURI> 
      </line> 
   </sipLines> 
   <voipControlPort>5060</voipControlPort> 
   <startMediaPort>16348</startMediaPort> 
   <stopMediaPort>20134</stopMediaPort> 
   <dscpForAudio>184</dscpForAudio> 
   <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> 
   <dialTemplate>dialplan.xml</dialTemplate> 
   <softKeyFile></softKeyFile> 
</sipProfile> 
<commonProfile> 
   <phonePassword></phonePassword> 
   <backgroundImageAccess>true</backgroundImageAccess> 
   <callLogBlfEnabled>2</callLogBlfEnabled> 
</commonProfile> 
<loadInformation>SIP45.8-5-4S</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>1,2,3,4,5,6,7</daysDisplayNotActive> 
   <displayOnTime>00:00</displayOnTime> 
   <displayOnDuration>00:00</displayOnDuration> 
   <displayIdleTimeout>00:00</displayIdleTimeout> 
   <spanToPCPort>1</spanToPCPort> 
   <loggingDisplay>1</loggingDisplay> 
   <loadServer></loadServer> 
</vendorConfig> 
<userLocale> 
   <name></name> 
   <uid></uid> 
   <langCode>en_US</langCode> 
   <version>1.0.0.0-1</version> 
   <winCharSet>iso-8859-1</winCharSet> 
</userLocale> 
<networkLocale></networkLocale> 
<networkLocaleInfo> 
   <name></name> 
   <uid></uid> 
   <version>1.0.0.0-1</version> 
</networkLocaleInfo>    
<deviceSecurityMode>1</deviceSecurityMode> 
<authenticationURL>http://example.domain.ext/services/authenticate.php</authenticationURL> 
<directoryURL>http://example.domain.ext/services/directory.php</directoryURL> 
<servicesURL>http://example.domain.ext/services/menu.xml</servicesURL> 
<idleURL></idleURL> 
<informationURL></informationURL> 
<messagesURL></messagesURL> 
<proxyServerURL></proxyServerURL> 
<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>

[bayula1977]

Leo That is my SEP:MAC

[bayula1977]

LEO Attached is also my XMLDEFAULT and DIAL PLAN below

<DIALTEMPLATE>
    <TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>

[Leo Laohoo]

<securedSipPort>5061</securedSipPort>

Change this to 5062

<backupProxy></backupProxy> 

Stick your Asterisk IP address here.

<timerRegisterExpires>3600</timerRegisterExpires>

240

<preferredCodec>BLANK</preferredCodec>

What codec are you planning to use?  This tag is missing from your config.

<authName>101</authName> 
<authPassword>ln101</authPassword> 

Under Asterisk Extensions, make sure these information is correct. 

<transportLayerProtocol>4</transportLayerProtocol> 

Under Asterisk Extensions you should have "UDP only".

[jonallport · Accepted solution]

Below is my working template. 

For SIP firmware >v9 then TCP must be used; firmware no longer supports SIP/udp.  This needs to be reflected in the Asterisk extension conf.  Also, in the config, take note of the use of 'USECALLMANAGER' in the line proxy setup.  This took a while to find out.

In my case I have line 2 (7945G) dials *8 & the 'messages' button dials *97 - asterisk feature codes for group pickup & VM.

<device>
   <deviceProtocol>SIP</deviceProtocol>
   <sshUserId>cisco</sshUserId>
   <sshPassword>cisco</sshPassword>
   <devicePool>
      <dateTimeSetting>
         <dateTemplate>D/M/YA</dateTemplate>
         <timeZone>GMT Standard/Daylight Time</timeZone>
         <ntps>
            <ntp>
               <name>{{{NTP server name or IP}}}</name>
               <ntpMode>Unicast</ntpMode>
            </ntp>         
         </ntps>
      </dateTimeSetting>
      <callManagerGroup>
         <members>
            <member priority="0">
               <callManager>
                  <ports>
                     <ethernetPhonePort>2000</ethernetPhonePort>
                     <sipPort>5060</sipPort>
                     <securedSipPort>5061</securedSipPort>
                  </ports>
                  <processNodeName>{{{Asterisk server name or IP}}}</processNodeName>
               </callManager>
            </member>
         </members>
      </callManagerGroup>
   </devicePool>
   <sipProfile>
      <sipProxies>
         <backupProxy></backupProxy>
         <backupProxyPort>5060</backupProxyPort>
         <emergencyProxy></emergencyProxy>
         <emergencyProxyPort></emergencyProxyPort>
         <outboundProxy></outboundProxy>
         <outboundProxyPort></outboundProxyPort>
         <registerWithProxy>true</registerWithProxy>
      </sipProxies>
      <sipCallFeatures>
         <cnfJoinEnabled>false</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>false</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>true</transferOnhookEnabled>
      <enableVad>false</enableVad>
      <dtmfAvtPayload>101</dtmfAvtPayload>
      <dtmfDbLevel>3</dtmfDbLevel>
      <dtmfOutofBand>avt</dtmfOutofBand>
      <alwaysUsePrimeLine>true</alwaysUsePrimeLine>
      <alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
      <kpml>3</kpml>
      <phoneLabel>{{{Phone label}}}</phoneLabel>
      <stutterMsgWaiting>1</stutterMsgWaiting>
      <callStats>false</callStats>
      <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
      <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
      <sipLines>
         <line button="1">
            <featureID>9</featureID>
            <featureLabel>{{{Line button label}}}</featureLabel>
            <proxy>USECALLMANAGER</proxy>
            <port>5060</port>
            <name>{{{Extension num?}}}</name>
            <displayName>{{{Extension num?}}}</displayName>
            <autoAnswer>
               <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>3</callWaiting>
            <authName>{{{SIP username}}}</authName>
            <authPassword>{{{SIP password}}}</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <contact>{{{Extension num?}}}</contact>
            <forwardCallInfoDisplay>
               <callerName>true</callerName>
               <callerNumber>false</callerNumber>
               <redirectedNumber>false</redirectedNumber>
               <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
         </line>
         <line button="2">
            <featureID>21</featureID>
            <featureLabel>Pickup</featureLabel>
        <speedDialNumber>*8</speedDialNumber>
         </line>
      </sipLines>  
      <voipControlPort>5060</voipControlPort>
      <startMediaPort>16348</startMediaPort>
      <stopMediaPort>20134</stopMediaPort>
      <dscpForAudio>184</dscpForAudio>
      <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
      <dialTemplate>dialplan.xml</dialTemplate>
      <softKeyFile></softKeyFile>
   </sipProfile>
   <commonProfile>
      <phonePassword></phonePassword>
      <backgroundImageAccess>true</backgroundImageAccess>
      <callLogBlfEnabled>2</callLogBlfEnabled>
   </commonProfile>
   <loadInformation>SIP45.9-4-2-1S</loadInformation>
   <vendorConfig>
      <sshAccess>0</sshAccess>
      <sshPort>22</sshPort>
      <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>1,2,3,4,5,6,7</daysDisplayNotActive>
      <displayOnTime>00:00</displayOnTime>
      <displayOnDuration>00:00</displayOnDuration>
      <displayIdleTimeout>00:00</displayIdleTimeout>
      <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
      <spanToPCPort>1</spanToPCPort>
      <loggingDisplay>1</loggingDisplay>
      <loadServer></loadServer>
   </vendorConfig>
   <userLocale>
      <name></name>
      <uid></uid>
      <langCode>en_US</langCode>
      <version>1.0.0.0-1</version>
      <winCharSet>iso-8859-1</winCharSet>
   </userLocale>
   <networkLocale></networkLocale>
   <networkLocaleInfo>
      <name></name>
      <uid></uid>
      <version>1.0.0.0-1</version>
   </networkLocaleInfo>    
   <deviceSecurityMode>1</deviceSecurityMode>
   <authenticationURL></authenticationURL>
   <directoryURL>{{{URL for directory}}}</directoryURL>
   <idleURL></idleURL>
   <informationURL></informationURL>
   <messagesURL></messagesURL>
   <servicesURL></servicesURL>
   <proxyServerURL></proxyServerURL>
   <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>

[Leo Laohoo]

For SIP firmware >v9 then TCP must be used; firmware no longer supports SIP/udp.  This needs to be reflected in the Asterisk extension conf. 

I got a 7945 working on version 9.4(2)SR1 but my Asterisk extension settings for "Transport" is still unchanged at "UDP only", however, the <transportLayerProtocol> has been changed to a value of "2".

Also, in the config, take note of the use of 'USECALLMANAGER' in the line proxy setup.

Thanks for this information.

[johnymar]

Hi,

I am configuring a Cisco 7945 in asterisk and i followed your recommendations but so far my handset is alway in "Registering Status"

Could you please guide me based on your experience?

Kind Regards