Cisco 7942 - Missed Calls Appear on Line1

Asterisk Asterisk General
1

So, I hope I have the correct forum…

Anyway, I have an Asterisk implementation with I’ve been using Cisco 7940 phones which works just great. I have recently started deploying 7942s and they connect, make and recieve calls and everythign appears to work.

However, if you missed a call it does not appear in the missed calls menu but on LIne1 (ie the extension number). This just seems to be pain, but, what it means is that until it is cleared you are unable to pickup inbound calls. The only way to clear it is do a redial on the phone. This will then clear it. It gets worse, if you have missed several calls you have to repeat the redial for each missed calls.

The firmware I’m using is SIP42.8-5-3SR1S. Below is the xml conf file

  • SIP admin XXXXX
  • D/M/Y GMT Standard/Daylight Time
  • 59.148.184.7 Unicast
  • 2000 5060 5061 192.168.105.88
  • true
  • true x--serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true
  • 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false g729a false 8041
  • 9 bazd 192.168.105.88 5060 bazd1 bazd1
  • 2 3 bazd1 XXXXXX false 1 222 4 5 bazd1
  • true false false true
  • 21 Call_Pickup *8 dialplan.xml SIP42.8-5-3SR1S
2

Hi,

I am having the exact same problem. I just wanted to know if you have found a fix for this issue?

I am using SIP42.8-5-3S and here is my sample SEPXXXXXXXXX.cnf.xml:

SIP

cisco

cisco

  <dateTimeSetting>

     <dateTemplate>M/D/Ya</dateTemplate>

     <timeZone>China Standard/Daylight Time</timeZone>

     <ntps>

          <ntp>

              <name>192.168.1.190</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>192.168.1.190</processNodeName>

           </callManager>

        </member>

     </members>

  </callManagerGroup>

  <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>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>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>

  <natEnabled>false</natEnabled>

  <natAddress></natAddress>

  <phoneLabel>107</phoneLabel>

  <stutterMsgWaiting>0</stutterMsgWaiting>

  <callStats>false</callStats>

  <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>

  <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

  <startMediaPort>16384</startMediaPort>

  <stopMediaPort>32766</stopMediaPort>

  <sipLines>

     <line button="1">

        <featureID>9</featureID>

        <featureLabel>Reza Samimi</featureLabel>

        <proxy>192.168.1.190</proxy>

        <port>5060</port>

        <name>107</name>

        <displayName>107</displayName>

        <autoAnswer>

           <autoAnswerEnabled>2</autoAnswerEnabled>

        </autoAnswer>

        <callWaiting>3</callWaiting>

        <authName>107</authName>

        <authPassword>123456</authPassword>

        <sharedLine>false</sharedLine>

        <messageWaitingLampPolicy>1</messageWaitingLampPolicy>

        <messagesNumber>3501</messagesNumber>

        <ringSettingIdle>4</ringSettingIdle>

        <ringSettingActive>5</ringSettingActive>

        <contact>107</contact>

        <forwardCallInfoDisplay>

           <callerName>true</callerName>

           <callerNumber>true</callerNumber>

           <redirectedNumber>false</redirectedNumber>

           <dialedNumber>true</dialedNumber>

        </forwardCallInfoDisplay>

     </line>   

  </sipLines>

  <voipControlPort>5060</voipControlPort>

  <dscpForAudio>184</dscpForAudio>

  <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>

  <dialTemplate>dialplan.xml</dialTemplate>

  <phonePassword></phonePassword>

  <backgroundImageAccess>true</backgroundImageAccess>

  <callLogBlfEnabled>1</callLogBlfEnabled>

SIP42.8-5-3S

  <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>

  <spanToPCPort>1</spanToPCPort>

  <loggingDisplay>1</loggingDisplay>

  <loadServer></loadServer>

1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37

US

  <name>US</name>

  <version>5.0(2)</version>

1

96

0

96

2

0

  <capf>

     <phonePort>3804</phonePort>

  </capf>

false

Any help is highly appreciated

3

Unfortunately not, will be looking at it again in the next couple of weeks. Will keep you posted. If you find anything out, please let me know. Thanks Matt.