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Conference call Issue in Cisco 7942G phone.

romilp
Level 1
Level 1

We connected Cisco 7942G phone with Matrix GE12S EPABX through SIP protocol by uploading xml file.

We are facing Conference call issue when we initiate conference.

In incoming call conference is working fine.

9 Replies 9

Manish Gogna
Cisco Employee
Cisco Employee

What exactly is the issue. Please describe in detail the exact steps you are doing to initiate conference and what happens when you try to initiate it ( fast busy , dead air ?? ). Where are the resources for conference configured , how are they being assigned to this phone?

Manish

- Do rate helpful posts -

Opearting Procedure :-

Place first call ( whether it is internal or external )

Press #2 ( which is flash code of Matrix system )-First party goes on hold.

Place Second call

Press #2 ( when Second party connect )

Pess *3  (Conference code of Matrix system )

Above Procedure is not worked.

If  Some one call ( incoming )  & press #2 then second call, press #2 *3 ( after connnectig second call ), Working filne.

We uploaded below mentioned XML file in Cisco 7942g Phone:-

<device>
   <deviceProtocol>SIP</deviceProtocol>
   <sshUserId>cisco</sshUserId>
   <sshPassword>cisco</sshPassword>
   <devicePool>
      <dateTimeSetting>
         <dateTemplate>D/M/Ya</dateTemplate>
         <timeZone>India Standard Time</timeZone>
         <ntps>
              <ntp>
                  <name>192.168.52.1</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>192.168.6.220</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>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>
      <dtmfInBand>avt</dtmfInBand>
      <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
      <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
      <kpml>3</kpml>
      <natEnabled>false</natEnabled>
      <natAddress></natAddress>
      <phoneLabel>Shilvant Prajapati</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>321</featureLabel>
            <proxy>USECALLMANAGER</proxy>
            <port>5060</port>
            <name>321</name>
            <displayName>321</displayName>
            <autoAnswer>
               <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>3</callWaiting>
            <authName>321</authName>
            <authPassword>321</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <contact>321</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>
   </sipProfile>
   <commonProfile>
      <phonePassword></phonePassword>
      <backgroundImageAccess>true</backgroundImageAccess>
      <callLogBlfEnabled>1</callLogBlfEnabled>
   </commonProfile>
   <loadInformation>SIP42.8-5-3S</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>1</webAccess>
      <spanToPCPort>1</spanToPCPort>
      <loggingDisplay>1</loggingDisplay>
      <loadServer></loadServer>
   </vendorConfig>
   <networkLocale>United_States</networkLocale>
   <networkLocaleInfo>
      <name>United_States</name>
      <version>5.0(2)</version>
   </networkLocaleInfo>
   <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>1</transportLayerProtocol>
   <capfAuthMode>0</capfAuthMode>
   <capfList>
      <capf>
         <phonePort>3804</phonePort>
      </capf>
   </capfList>
   <certHash></certHash>
   <encrConfig>false</encrConfig>
</device>

It seems the key presses for conference are not being registered on Matrix pbx. Can you check with their support if they can verify it through any traces or captures.

Manish

- Do rate helpful posts -

I think If key pressed issue then Conf is not working for incoming call.

Also I checked with Matrix & Changed DTMF type as suggested by Matrix but still issue persist.. Matrix tech team said that you need to check in Cisco Phone..

I think dialed digit is not reahed at Matrix while first call is connected.. Is there any way to send digit ?

Other Opearting way :

Place first call

Press soft key of conference ( first party goes on hold & get tone )

Place second call

Again Show Conference Soft key

While press Conf soft key, Phone Shows that Conf can not complete ..

Is there any way in XMl file for configuring soft key working ?

Waiting for resolution if any

Vivek Batra
VIP Alumni
VIP Alumni

Hi,

Do you mean if there is an incoming call and you create conference, it works. But if that calls were outgoing, then conference failed.

Please note that you might be using phones built in conference bridge because Matrix PBX won't be able to use its own conference bridge with third party phones (here Cisco is third party phone for Matrix PBX) unless you're using DTMF access codec to create conference.

If my assumption is right that you're facing issues with conference while calls were outbound, try to keep only G711 codec under respective SIP extension (in Matrix PBX) and ensure calls are made using G711 only. Then try to create conference and see if it makes any difference for you.

- Vivek

HI,

Yes , Conf is not working for outbound call..

Already we used G.711 coded in XML file & program G.711 is 1st preference in Matrix System but still issue persists..

Opearting Procedure :-

Place first call ( whether it is internal or external )

Press #2 ( which is flash code of Matrix system )-First party goes on hold.

Place Second call

Press #2 ( when Second party connect )

Pess *3  (Conference code of Matrix system )

Can you please help me to resolve the issue ?

Can you please help me if you have any other idea about solution of Conf issue ?

Hi,

Apologize for late reply.

Can you please share the PCAP traces of both working and non-working cases? You can simply take the PCAP traces from Matrix PBX.

In non-working case, if you can put the current call on hold using #2 and make second outgoing call, this means that there is no issue with DTMF and conference should work too. Please share the PCAP and will check further.

- Vivek

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