Cisco 7841 phone , no inbound calls

I am following the post and the xml config file from that post https://community.freepbx.org/t/cisco-cp7841-help-me-register-a-cisco-only-phone-to-asterisk/93748?u=brxx and the phone is kind of working but I cannot register it to the system. I bet it is something with the configuration but I modified/tried already too many fields that it is becoming counterproductive.

I used the config from that post and modified it with my settings. I don’t see the phone system responding. The phone sends SIP REFER to the phone system and the phone system is responding with 501 not implemented. I can call outbound with no issues. but inbound is not working. The asterisk info shows:

Endpoint: 135/135 Not in use 0 of inf
OutAuth: 135-auth/135
InAuth: 135-auth/135
Aor: 135 2
Contact: 135/sip:135@:65477;x-ast-orig- d6e3f364c3 Avail 17.002

Any suggestions how to force that phone to send Register and not use Refer method or perhaps “implement” the Refer method on my PBXact ?

Marked up better:

Endpoint:  135/135                                              Not in use    0 of inf
OutAuth:  135-auth/135
InAuth:  135-auth/135
Aor:  135                                                2
Contact:  135/sip:135@<public ip address>:65477;x-ast-orig- d6e3f364c3 Avail        17.002

That shows something has registered (or no registration is needed). I’m not sure where the x-ast-orig parameter has come from and it is not clear whose public IP address.

Outbound auth should not normally be set for a phone, but is probably harmless, as it won’t be used.

We need the actual configuration (screen shot for FreePBX), and the full log, with “pjsip set logger on” in effect.

It seems that the problem is with the “To:” field in the SIP invite packet. My topology is: FreePBX in the cloud on Public ip address and the phones are in the remote location.

I just did some testing putting the FreePBX a 10.0.200.28/30 subnet (ip address of FrePBX 10.0.200.30 with gateway being 10.0.200.29) while the phones were on 10.0.100.x/24 (cisco phone on 10.0.100.121 ip) subnet. Everything works perfectly for inbound calls (between extensions) until I enable NAT which simulates the public ip address. The SIP “To:” field shows the NATed ip address in the INVITE “To:sip:[email protected];user=phone” and the phone returns 404 not found. Without NAT that field shows simply the ip address of the Cisco phone “To:sip:[email protected];user=phone”

This happens only to that Cisco phone, I have Sangoma P330 phone with identical network/NAT setup on that system and it simply works for incoming calls from the Cisco phone.

It seems that I am missing a setting that would enable that phone to understand that it is behind NAT. There is a setting in the configuration file natEnable and natAddress but it does not seem to do anything.

Here is the phone config:

<?xml version="1.0" encoding="UTF-8"?>
<device>
<deviceProtocol>SIP</deviceProtocol>
  <sshUserId>cisco</sshUserId>
  <sshPassword>cisco</sshPassword>
	<tzdata>
		<tzolsonversion>2015a</tzolsonversion>
		<tzupdater>tzupdater.jar</tzupdater>
	</tzdata>
  <devicePool>
		<dateTimeSetting> 
		<name>CMLocal</name>
			<dateTemplate>D/M/YYa</dateTemplate> 
			<timeZone>US Eastern Standard/Daylight Time</timeZone>
			<olsonTimeZone>US/New York</olsonTimeZone>
			<ntps> 
				<ntp>
					<name>10.0.100.1</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>weha.cci.tel</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
  </devicePool>
  <commonProfile>
     <phonePassword></phonePassword>
     <backgroundImageAccess>true</backgroundImageAccess>
     <callLogBlfEnabled>2</callLogBlfEnabled>
  </commonProfile>
  <loadInformation>sip9971.9-4-2SR1-2</loadInformation>
  <featurePolicyFile>DefaultFP.xml</featurePolicyFile>
  <vendorConfig>
     <disableSpeaker>false</disableSpeaker>
     <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
     <pcPort>0</pcPort>
     <settingsAccess>1</settingsAccess>
     <garp>0</garp>
     <voiceVlanAccess>0</voiceVlanAccess>
     <ciscoCamera>1</ciscoCamera>
     <videoCapability>1</videoCapability>
     <usbClasses>0,1,2</usbClasses>
     <sdio>1</sdio>
     <wifi>0</wifi>
     <bluetoothProfile>0,1</bluetoothProfile>
     <powerNegotiation>0</powerNegotiation>
     <autoSelectLineEnable>0</autoSelectLineEnable> 
     <webAccess>0</webAccess>
     <sshAccess>0</sshAccess>
	 <sshPort>22</sshPort>
     <g722CodecSupport>2</g722CodecSupport>
     <daysDisplayNotActive>1,7</daysDisplayNotActive> 
     <displayOnTime>07:00</displayOnTime> 
     <displayOnDuration>12:00</displayOnDuration> 
     <displayIdleTimeout>00:15</displayIdleTimeout> 
	 <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
     <spanToPCPort>1</spanToPCPort>
     <loggingDisplay>1</loggingDisplay>
     <loadServer></loadServer>
  </vendorConfig>
  <commonConfig>
     <usb1>1</usb1>
     <usb2>1</usb2>
     <ciscoCamera>1</ciscoCamera>
     <usbClasses>0,1,2</usbClasses>
     <sdio>1</sdio>
     <bluetooth>1</bluetooth>
     <wifi>0</wifi>
     <bluetoothProfile>0,1</bluetoothProfile>
     <joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
  </commonConfig>
  <enterpriseConfig>
     <usb1>1</usb1>
     <usb2>1</usb2>
     <ciscoCamera>1</ciscoCamera>
     <usbClasses>0,1,2</usbClasses>
     <sdio>1</sdio>
     <bluetooth>1</bluetooth>
     <wifi>0</wifi>
     <bluetoothProfile>0,1</bluetoothProfile>
     <joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
     <videoCapability>0</videoCapability>
     <webAccess>1</webAccess>
     <eapAuthentication>2</eapAuthentication>
     <webProtocol>1</webProtocol>
  </enterpriseConfig>
  <advertiseG722Codec>1</advertiseG722Codec>
  <networkLocale>United_States</networkLocale>
	<networkLocaleInfo> 
		<name>United_States</name> 
		<uid>64</uid> 
		<version>1.0.0.0-1</version> 
	</networkLocaleInfo>
  <deviceSecurityMode>1</deviceSecurityMode>
  <idleTimeout>0</idleTimeout>
  <authenticationURL></authenticationURL>
  <directoryURL></directoryURL>
  <idleURL></idleURL>
  <informationURL></informationURL>
  <messagesNumber></messagesNumber>  
  <messagesURL></messagesURL>
  <proxyServerURL></proxyServerURL>
  <servicesURL>http://cisco.internect.net/</servicesURL>
  <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
  <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
  <dscpForCm2Dvce>96</dscpForCm2Dvce>
  <transportLayerProtocol>2</transportLayerProtocol>
  <dndCallAlert>5</dndCallAlert>
  <phonePersonalization>1</phonePersonalization>
  <rollover>0</rollover>
  <singleButtonBarge>0</singleButtonBarge>
  <joinAcrossLines>1</joinAcrossLines>
  <autoCallPickupEnable>false</autoCallPickupEnable>
  <blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
  <blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
  <capfAuthMode>0</capfAuthMode>
  <capfList>
     <capf>
        <phonePort>3804</phonePort>
     </capf>
  </capfList>
  <certHash></certHash>
  <encrConfig>false</encrConfig>
  <sipProfile>
     <sipProxies>
        <backupProxy>USECALLMANAGER</backupProxy>
        <backupProxyPort>5060</backupProxyPort>
        <emergencyProxy>USECALLMANAGER</emergencyProxy>
        <emergencyProxyPort>5060</emergencyProxyPort>
        <outboundProxy></outboundProxy>
        <outboundProxyPort></outboundProxyPort>
        <registerWithProxy>true</registerWithProxy>
     </sipProxies>
     <sipCallFeatures>
        <cnfJoinEnabled>true</cnfJoinEnabled>
        <callForwardURI>x-cisco-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>1</dndControl>
        <remoteCcEnable>true</remoteCcEnable>
	<retainForwardInformation>true</retainForwardInformation>
     </sipCallFeatures>
     <sipStack>
        <sipInviteRetx>6</sipInviteRetx>
        <sipRetx>10</sipRetx>
        <timerInviteExpires>180</timerInviteExpires>
        <timerRegisterExpires>120</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>phone</userInfo>
     </sipStack>
     <autoAnswerTimer>1</autoAnswerTimer>
     <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
     <autoAnswerOverride>true</autoAnswerOverride>
     <transferOnhookEnabled>false</transferOnhookEnabled>
     <enableVad>false</enableVad>
     <preferredCodec>none</preferredCodec>
     <dtmfAvtPayload>101</dtmfAvtPayload>
     <dtmfDbLevel>3</dtmfDbLevel>
     <dtmfOutofBand>avt</dtmfOutofBand>
     <alwaysUsePrimeLine>true</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>
     <natEnabled>true</natEnabled>
     <natAddress>10.0.200.29</natAddress>
     <stutterMsgWaiting>2</stutterMsgWaiting>
     <callStats>true</callStats>
     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
     <startMediaPort>10000</startMediaPort>
     <stopMediaPort>20000</stopMediaPort>
     <voipControlPort>5060</voipControlPort>
     <dscpForAudio>184</dscpForAudio>
	 <dscpVideo>136</dscpVideo>
	 <dscpForTelepresence>128</dscpForTelepresence>
     <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
	 <softKeyFile>softKey9971.xml</softKeyFile>
     <dialTemplate>dialplan.xml</dialTemplate>
     <phoneLabel>135 Cisco</phoneLabel>
     <sipLines>
        <line button="1" lineIndex="1">
           <featureID>9</featureID>
           <featureLabel>135</featureLabel>
           <name>135</name>
           <displayName>135</displayName>
           <contact>135</contact>
           <proxy>USECALLMANAGER</proxy>
           <port>5060</port>
           <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>
           <authName>135</authName>
           <authPassword>passwrod for the phone</authPassword>
           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
		   <messageWaitingAMWI>1</messageWaitingAMWI>
           <messagesNumber>*98</messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>
           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
			<maxNumCalls>4</maxNumCalls>
			<busyTrigger>2</busyTrigger>
        </line>
		<line  button="2">
                        <featureID>21</featureID>
                        <featureLabel>106</featureLabel>
                        <speedDialNumber>106</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
        </line>
        <line  button="3">
                        <featureID>21</featureID>
                        <featureLabel>104</featureLabel>
                        <speedDialNumber>104</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
        </line>
		<line  button="4">
                        <featureID>21</featureID>
                        <featureLabel>Cell</featureLabel>
                        <speedDialNumber>8444444444</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
        </line>
		<line  button="5">
                        <featureID>21</featureID>
                        <featureLabel>LABEL BUTTON 5</featureLabel>
                        <speedDialNumber>NUMBER</speedDialNumber>
                        <featureOptionMask>1</featureOptionMask>
        </line>
		<line  button="6">
                        <featureID>21</featureID>
                        <featureLabel>LABEL BUTTON 6</featureLabel>
                        <speedDialNumber>NUMBER</speedDialNumber>
        </line>
	</sipLines>
  </sipProfile>
	<phoneServices>
     <provisioning>0</provisioning>
     	<phoneService  type="1" category="0">
     		<name>Missed Calls</name>
     		<url>Application:Cisco/MissedCalls</url>
        	<vendor></vendor>
     		<version></version>
     	</phoneService>
	<phoneService  type="2" category="0">
		<name>Voicemail</name>
		<url>Application:Cisco/Voicemail</url>
		<vendor></vendor>
		<version></version>
	</phoneService>
	<phoneService  type="1" category="0">
		<name>Received Calls</name>
		<url>Application:Cisco/ReceivedCalls</url>
		<vendor></vendor>
		<version></version>
	</phoneService>
	<phoneService  type="1" category="0">
		<name>Placed Calls</name>
		<url>Application:Cisco/PlacedCalls</url>
		<vendor></vendor>
		<version></version>
		</phoneService>
			<phoneService  type="0" category="0">
				<name>Australian Services</name>
                <url>http://cisco.internect.net/</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
	</phoneServices>
</device>

I still cannot make inbound call to the phone when the phone is behind NAT.

Per the first post. I was mistaken , the phone successfully registers, seeing it in the pcap, “200OK (register) 1binding” back from the FreePBX.

Thank you

Anybody on this knowing Cisco 7800 series phones ? I made the phone working over the VPN to remote FreePBX with no NAT in between.

The moment I enable NAT and send traffic over public internet when the FreePBX sends INVITE the phone responds with 404 not found. As far as I can tell comparing packets from the pcap trace the only difference in the packet in “To:” field where on local or VPN network is “To:[email protected]” and over the public internet or NAT in general “To:135@<NATed-ip-address” which is a public ip address usually.

Thank you

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