Cisco 7942 with local PBX

VoIP Hardware Solutions
Provider Solution Details
VoIP Hardware Zycoo UC Solutions
  • Modular Design IP PBX for SMB
  • Remote office Centralized Management solution
  • 3rd party app integration, Enterprise Billing, Android & iOS client
Details
Yeastar Communications Solutions
  • Cost-effective IP-PBX Solution for SMB
  • FXS, FXO, GSM, BRI and PRI VoIP Gateways
  • Rich features and reliable performance
Details
The Cisco 7942 phone expects generally to work with the CISCO callmanager software which is in the local network. Asterisk or other SIP Proxy can fake the CISCO callmanager. Make sure that the SIP proxy is not assuming NAT when connecting the phones as extensions. The CISCO phones expect SIP messages only on the preconfigured voipControlPort. If the SIP Proxy expects a NATted device it sends SIP answers back to the port where it received the SIP message from. This does not work here. Therefore out-of-the box NAT traversal mostly does not work with these phones.

Because the phones always expect SIP messages on the voipControlPort one can only use these phones behind NAT if one can configure the firewall.

Example of 7942 xml configuration file for a CISCO 7942 attached to an Asterisk box (delete inline comments starting with ------------------ before using this configuration

SEPF0257278C861.cnf.xml (filename must contain MAC address of device)

<device>
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId>  
<sshPassword>admin</sshPassword> 
<devicePool>  
 <dateTimeSetting>  
    <dateTemplate>D/M/Y</dateTemplate>  
    <timeZone>E. European Standard/Daylight Time</timeZone>
    <ntps>  
         <ntp>  
             <name>80.81.32.231</name>  
             <ntpMode>Unicast</ntpMode>  
         </ntp>  
    </ntps>  
 </dateTimeSetting>  
 <callManagerGroup>  
    <members>  
       <member priority="0">  
          <callManager>  
             <ports>  
                 <sipPort>5060</sipPort>  
             </ports>  
             <processNodeName>10.17.1.2</processNodeName>    --------------- IP of Asterisk box 
          </callManager>  
       </member>  
    </members>  
 </callManagerGroup>  
</devicePool>  
<sipProfile> 
  <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>  
 <dtmfAvtPayload>101</dtmfAvtPayload>  
 <dtmfDbLevel>3</dtmfDbLevel>  
 <dtmfOutofBand>avt</dtmfOutofBand>  
 <alwaysUsePrimeLine>false</alwaysUsePrimeLine>  
 <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>  
 <kpml>3</kpml>
 <phoneLabel>6344443</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>102</featureLabel>  
       <port>5060</port> 
       <name>102</name> 
       <displayName>102</displayName>  
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>  
       <callWaiting>3</callWaiting> 
       <authName>102</authName>  
       <authPassword>8dffrffr54</authPassword>  
       <sharedLine>false</sharedLine> 
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy>  
       <messagesNumber>102</messagesNumber>  
       <ringSettingIdle>4</ringSettingIdle>  
       <ringSettingActive>5</ringSettingActive> 
       <contact>102</contact>  
       <forwardCallInfoDisplay>  
          <callerName>true</callerName>  
          <callerNumber>false</callerNumber>  
          <redirectedNumber>false</redirectedNumber>  
          <dialedNumber>true</dialedNumber>  
       </forwardCallInfoDisplay>  
    </line>
  <line  
     button="2"> 
       <featureID>9</featureID>  
       <featureLabel>103</featureLabel>  
       <port>5060</port> 
       <name>103</name> 
       <displayName>103</displayName>  
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>  
       <callWaiting>3</callWaiting> 
       <authName>103</authName>  
       <authPassword>fd34rvf</authPassword>  
       <sharedLine>false</sharedLine> 
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy>  
       <messagesNumber>103</messagesNumber>  
       <ringSettingIdle>4</ringSettingIdle>  
       <ringSettingActive>5</ringSettingActive> 
       <contact>103</contact>  
       <forwardCallInfoDisplay>  
          <callerName>true</callerName>  
          <callerNumber>false</callerNumber>  
          <redirectedNumber>false</redirectedNumber>  
          <dialedNumber>true</dialedNumber>  
       </forwardCallInfoDisplay>  
    </line>
</sipLines>
 <voipControlPort>5061</voipControlPort>  --------------------------- The phones always expect SIP messages on this port
 <dscpForAudio>184</dscpForAudio>  
 <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>  
 <dialTemplate>dialplan.xml</dialTemplate>  
</sipProfile> 
<commonProfile>  
 <phonePassword></phonePassword>
 <backgroundImageAccess>true</backgroundImageAccess>  
 <callLogBlfEnabled>2</callLogBlfEnabled>  
</commonProfile> 
<loadInformation>SIP42.9-3-1-1S</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> 
<sshAccess>1</sshAccess>
<sshPort>22</sshPort> 
<webAccess>0</webAccess>  
 <spanToPCPort>0</spanToPCPort>  
 <loggingDisplay>1</loggingDisplay>  
 <loadServer></loadServer>  
</vendorConfig> 
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp> 
<userLocale> 
 <name></name> 
<uid>1</uid> 
 <langCode></langCode> 
<version></version> 
 <winCharSet></winCharSet> 
</userLocale> 
<deviceSecurityMode>1</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>2</transportLayerProtocol>

<capfAuthMode>0</capfAuthMode>  
<capfList>  
 <capf>  
    <phonePort>3804</phonePort>  
 </capf>  
</capfList> 
<certHash></certHash>  
<encrConfig>false</encrConfig>  
</device>

The Cisco 7942 phone expects generally to work with the CISCO callmanager software which is in the local network. Asterisk or other SIP Proxy can fake the CISCO callmanager. Make sure that the SIP proxy is not assuming NAT when connecting the phones as extensions. The CISCO phones expect SIP messages only on the preconfigured voipControlPort. If the SIP Proxy expects a NATted device it sends SIP answers back to the port where it received the SIP message from. This does not work here. Therefore out-of-the box NAT traversal mostly does not work with these phones.

Because the phones always expect SIP messages on the voipControlPort one can only use these phones behind NAT if one can configure the firewall.

Example of 7942 xml configuration file for a CISCO 7942 attached to an Asterisk box (delete inline comments starting with ------------------ before using this configuration

SEPF0257278C861.cnf.xml (filename must contain MAC address of device)

<device>
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId>  
<sshPassword>admin</sshPassword> 
<devicePool>  
 <dateTimeSetting>  
    <dateTemplate>D/M/Y</dateTemplate>  
    <timeZone>E. European Standard/Daylight Time</timeZone>
    <ntps>  
         <ntp>  
             <name>80.81.32.231</name>  
             <ntpMode>Unicast</ntpMode>  
         </ntp>  
    </ntps>  
 </dateTimeSetting>  
 <callManagerGroup>  
    <members>  
       <member priority="0">  
          <callManager>  
             <ports>  
                 <sipPort>5060</sipPort>  
             </ports>  
             <processNodeName>10.17.1.2</processNodeName>    --------------- IP of Asterisk box 
          </callManager>  
       </member>  
    </members>  
 </callManagerGroup>  
</devicePool>  
<sipProfile> 
  <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>  
 <dtmfAvtPayload>101</dtmfAvtPayload>  
 <dtmfDbLevel>3</dtmfDbLevel>  
 <dtmfOutofBand>avt</dtmfOutofBand>  
 <alwaysUsePrimeLine>false</alwaysUsePrimeLine>  
 <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>  
 <kpml>3</kpml>
 <phoneLabel>6344443</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>102</featureLabel>  
       <port>5060</port> 
       <name>102</name> 
       <displayName>102</displayName>  
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>  
       <callWaiting>3</callWaiting> 
       <authName>102</authName>  
       <authPassword>8dffrffr54</authPassword>  
       <sharedLine>false</sharedLine> 
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy>  
       <messagesNumber>102</messagesNumber>  
       <ringSettingIdle>4</ringSettingIdle>  
       <ringSettingActive>5</ringSettingActive> 
       <contact>102</contact>  
       <forwardCallInfoDisplay>  
          <callerName>true</callerName>  
          <callerNumber>false</callerNumber>  
          <redirectedNumber>false</redirectedNumber>  
          <dialedNumber>true</dialedNumber>  
       </forwardCallInfoDisplay>  
    </line>
  <line  
     button="2"> 
       <featureID>9</featureID>  
       <featureLabel>103</featureLabel>  
       <port>5060</port> 
       <name>103</name> 
       <displayName>103</displayName>  
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>  
       <callWaiting>3</callWaiting> 
       <authName>103</authName>  
       <authPassword>fd34rvf</authPassword>  
       <sharedLine>false</sharedLine> 
       <messageWaitingLampPolicy>1</messageWaitingLampPolicy>  
       <messagesNumber>103</messagesNumber>  
       <ringSettingIdle>4</ringSettingIdle>  
       <ringSettingActive>5</ringSettingActive> 
       <contact>103</contact>  
       <forwardCallInfoDisplay>  
          <callerName>true</callerName>  
          <callerNumber>false</callerNumber>  
          <redirectedNumber>false</redirectedNumber>  
          <dialedNumber>true</dialedNumber>  
       </forwardCallInfoDisplay>  
    </line>
</sipLines>
 <voipControlPort>5061</voipControlPort>  --------------------------- The phones always expect SIP messages on this port
 <dscpForAudio>184</dscpForAudio>  
 <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>  
 <dialTemplate>dialplan.xml</dialTemplate>  
</sipProfile> 
<commonProfile>  
 <phonePassword></phonePassword>
 <backgroundImageAccess>true</backgroundImageAccess>  
 <callLogBlfEnabled>2</callLogBlfEnabled>  
</commonProfile> 
<loadInformation>SIP42.9-3-1-1S</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> 
<sshAccess>1</sshAccess>
<sshPort>22</sshPort> 
<webAccess>0</webAccess>  
 <spanToPCPort>0</spanToPCPort>  
 <loggingDisplay>1</loggingDisplay>  
 <loadServer></loadServer>  
</vendorConfig> 
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp> 
<userLocale> 
 <name></name> 
<uid>1</uid> 
 <langCode></langCode> 
<version></version> 
 <winCharSet></winCharSet> 
</userLocale> 
<deviceSecurityMode>1</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>2</transportLayerProtocol>

<capfAuthMode>0</capfAuthMode>  
<capfList>  
 <capf>  
    <phonePort>3804</phonePort>  
 </capf>  
</capfList> 
<certHash></certHash>  
<encrConfig>false</encrConfig>  
</device>

Created by: geejay101, Last modification: Wed 21 of Aug, 2013 (15:53 UTC) by GeeJay
Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+