NO JOY Cisco cp8945 SIP Driver

thewball

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Joined
Dec 6, 2021
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Hello!
I am having issues with my Cisco CP-8945 registering with IPBX.

The phone is 192.168.1.54
The IPBX is 192.168.1.55

The phone shows that it is registered within the system, it shows the extension, and I can call outbound from the phone. The phone can not receive any calls though.

I see this in the asterisk console:
Code:
vmware*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
0 SIP registrations.

And I believe it is in communication with the phone properly:
Code:
Reliably Transmitting (no NAT) to 192.168.1.54:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK5bbb2b95
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as2521c157
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: IncrediblePBX-15.0.17.34.1(18.2.1)
Date: Tue, 28 Dec 2021 21:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.54:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK5bbb2b95
From: "Unknown" <sip:[email protected]>;tag=as2521c157
To: <sip:[email protected]:5060;transport=udp>;tag=c4143cb00e97009c216c8693-1c0454ab
Call-ID: [email protected]:5060
Date: Tue, 28 Dec 2021 21:07:07 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP8945/9.4.2
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0
Content-Length: 337
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27152 0 IN IP4 192.168.1.54
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 102 9 0 8 116 18 101
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 15 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

Here is my configuration for the phone.
XML:
<device>
   <deviceProtocol>SIP</deviceProtocol>
   <sshUserId>admin</sshUserId>
   <sshPassword>password</sshPassword>
   <devicePool>
      <dateTimeSetting>
         <dateTemplate>M/D/YA</dateTemplate>
         <timeZone>Pacific Standard/Daylight Time</timeZone>
         <ntps>
            <ntp>
               <name>pool.ntp.org</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>192.168.1.55</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>
      <natEnabled>false</natEnabled>
      <natReceivedProcessing>false</natReceivedProcessing>
      <natAddress>192.168.1.54</natAddress>
      <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>Matt Cisco 8945</phoneLabel>
      <stutterMsgWaiting>1</stutterMsgWaiting>
      <callStats>true</callStats>
      <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
      <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
      <sipLines>
         <line button="1">
            <featureID>9</featureID>
            <featureLabel>1234</featureLabel>
            <proxy>USECALLMANAGER</proxy>
            <port>5060</port>
            <name>1234</name>
            <displayName>1234</displayName>
            <autoAnswer>
               <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>3</callWaiting>
            <authName>1234</authName>
            <authPassword>12341234</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <contact>cisco8945</contact>
            <forwardCallInfoDisplay>
               <callerName>true</callerName>
               <callerNumber>false</callerNumber>
               <redirectedNumber>false</redirectedNumber>
               <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
         </line>
      </sipLines>
      <voipControlPort>5060</voipControlPort>
      <startMediaPort>15000</startMediaPort>
      <stopMediaPort>59999</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>SIP894x.9-4-2SR3-1</loadInformation>
   <vendorConfig>
      <disableSpeaker>false</disableSpeaker>
      <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
      <pcPort>1</pcPort>
      <settingsAccess>1</settingsAccess>
      <garp>0</garp>
      <voiceVlanAccess>0</voiceVlanAccess>
      <videoCapability>0</videoCapability>
      <autoSelectLineEnable>0</autoSelectLineEnable>
      <webAccess>1</webAccess>
      <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
      <displayOnTime>00:00</displayOnTime>
      <displayOnDuration>00:50</displayOnDuration>
      <displayIdleTimeout>00:20</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></authenticationURL>
   <directoryURL></directoryURL>
   <servicesURL></servicesURL>
   <idleURL></idleURL>
   <informationURL></informationURL>
   <messagesURL></messagesURL>
   <proxyServerURL></proxyServerURL>
   <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>
 
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