Cisco 7962 SIP configuration, incoming calls problem

I have similar problem with 7962 as I had with 7960 earlier.

  • The phone is on the same subnet as FreePBX
  • I’m configuring the phone successfully via DHCP and TFTP
  • However, phone shows “Unavailable” when listing with “pjsip show endpoints” command.
  • The phone registers to the extent that I can make calls from the phone to other phones. My dialplan.xml is applied correctly.
  • But calls made to the phone fail with FreePBX reporting “service unavailable”.
  • Not using any patches such as call manager. FreePBX is vanilla install.
  • I have defined <natEnabled>true</natEnabled> because this was required with 7960.
  • Extension in FreePBX has Force rport setting disabled (“No”).

I would appreciate any hints or guidance. This seems to be quite close to working.

Here is my SEPMAC.cnf.xml

<device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>Jaakko</sshUserId>
    <sshPassword>redacted</sshPassword>
    <ipAddressMode>0</ipAddressMode>

    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D.M.Ya</dateTemplate>
            <timeZone>FLE Standard Time;GMT+02:00</timeZone>
            <olsonTimeZone>Europe/Helsinki</olsonTimeZone>
            <ntps>
                <ntp>
                    <name>185.251.115.30</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>

        <callManagerGroup>
            <members>
                <member priority="0">
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5060</sipPort>
                        </ports>
                        <processNodeName>192.168.10.63</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
    </devicePool>

    <sipProfile>
        <sipProxies>
            <registerWithProxy>true</registerWithProxy>
        </sipProxies>
        <sipCallFeatures>
            <cnfJoinEnabled>true</cnfJoinEnabled>
            <rfc2543Hold>false</rfc2543Hold>
            <callHoldRingback>2</callHoldRingback>
            <localCfwdEnable>true</localCfwdEnable>
            <semiAttendedTransfer>true</semiAttendedTransfer>
            <anonymousCallBlock>2</anonymousCallBlock>
            <callerIdBlocking>2</callerIdBlocking>
            <dndControl>0</dndControl>
            <remoteCcEnable>true</remoteCcEnable>
            <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>  
        </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>
        <phoneLabel>Jaakko</phoneLabel>
        <stutterMsgWaiting>0</stutterMsgWaiting>
        <callStats>false</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
        <startMediaPort>10000</startMediaPort>
        <stopMediaPort>20000</stopMediaPort>

        <sipLines>
            <line button="1">
                <featureID>9</featureID>
                <featureLabel>Jaakko</featureLabel>
                <proxy>USECALLMANAGER</proxy>
                <port>5060</port>
                <name>604</name>
                <displayName>604</displayName>
                <autoAnswer>
                    <autoAnswerEnabled>2</autoAnswerEnabled>
                </autoAnswer>
                <callWaiting>3</callWaiting>
                <authName>604</authName>
                <authPassword>redacted</authPassword>
                <sharedLine>false</sharedLine>
                <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
                <messagesNumber>*97</messagesNumber>
                <ringSettingIdle>4</ringSettingIdle>
                <ringSettingActive>5</ringSettingActive>
                <contact>604</contact>
                <forwardCallInfoDisplay>
                    <callerName>true</callerName>
                    <callerNumber>true</callerNumber>
                    <redirectedNumber>false</redirectedNumber>
                    <dialedNumber>true</dialedNumber>
                </forwardCallInfoDisplay>
            </line>

        </sipLines>

        <voipControlPort>5160</voipControlPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>dialplan.xml</dialTemplate>
    </sipProfile>

    <commonProfile>
        <phonePassword>foobar</phonePassword>
        <backgroundImageAccess>true</backgroundImageAccess>
        <callLogBlfEnabled>1</callLogBlfEnabled>
    </commonProfile>

    <loadInformation>SIP42.9-4-2SR3-1S</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>0</webAccess>
        <spanToPCPort>1</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <loadServer></loadServer>
        <sshAccess>0</sshAccess>
        <sshPort>22</sshPort>
    </vendorConfig>

    <versionStamp>002</versionStamp>
    <networkLocale>United_Kingdom</networkLocale>
    <networkLocaleInfo>
        <name>United_Kingdom</name>
        <uid>64</uid>
        <version>1.0.0.0-4</version> 
    </networkLocaleInfo>

    <deviceSecurityMode>1</deviceSecurityMode>
    <authenticationURL></authenticationURL>  
    <servicesURL></servicesURL> 
    <directoryURL>http://192.168.10.63/directory.xml</directoryURL>
    <idleURL></idleURL>  
    <informationURL></informationURL> 
    <messagesURL></messagesURL>  
    <proxyServerURL></proxyServerURL>  
    <dialToneSetting>2</dialToneSetting>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>  
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>  
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <capfAuthMode>0</capfAuthMode>  
    <capfList>  
      <capf>  
        <phonePort>3804</phonePort>  
      </capf>  
    </capfList> 
    <transportLayerProtocol>2</transportLayerProtocol>
    <certHash></certHash>  
    <encrConfig>false</encrConfig>  
</device>

Registration is not needed to make a call from a phone.

Look, I don’t want to start arguing about semantics and perhaps I’m not using the proper terminology, but this is what is shown in Asterisk for the phone in question:

This is why I say it has been “registered”.

Ok, looks like I’m also getting this in asterisk log:

[2024-12-01 17:26:31] WARNING[561351]: pjproject: <?>: sip_transport.c Dropping 1006 bytes packet from UDP 192.168.10.50:52578 : PJSIP syntax error exception when parsing 'Request Line' header on line 1 col 1

Ok that was related to UDP. It was hinted to me before that some people have had problems with UDP and switching to TCP has helped.

I tried this:

<transportLayerProtocol>1</transportLayerProtocol>

This changed the phone to use TCP and now I can make calls in both directions.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.