[Freeswitch-users] Trouble registering Cisco 7942

Mon Jul 30 17:33:46 MSD 2012

Hi.

I am using firmware 8.5(4) SIP. I see the register message on the FreeSwitch machine using ngrep.

Everything seems ok, but I get unauthorized. Then I thought to myseld that this might be NAT issue. So I tried setting up a Asterisk on the same subnet as the telephone is sitting on. Same result.

I have taken some info out of security reasons. Theese are marked in red text. The ngrep log below is from the FreeSwitch machine.

I have also read from zero to hero here.
http://www.freeswitch.org/node/401

The two first link give description of how to setup a TFTP server and soforth. I already have this. Important to mention that the Cisco 7942 is a newer phone than in theese first two links so to get the relevant config look at the third link. Below is the third link from this page.

Have used lots of info from this page.
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP

A little snippet from the console log on the phone.
977: ERR 09:43:30.785766 JVM: dns_gethostbysrv 3 h_errno
 978: ERR 09:43:30.786378 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 979: ERR 09:43:30.803646 JVM: dns_gethostbysrv 3 h_errno
 980: ERR 09:43:30.804270 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 981: ERR 09:43:30.821237 JVM: dns_gethostbysrv 3 h_errno
 982: ERR 09:43:30.821830 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 983: ERR 09:43:30.838442 JVM: dns_gethostbysrv 3 h_errno
 984: ERR 09:43:30.839066 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 985: ERR 09:43:30.856124 JVM: dns_gethostbysrv 3 h_errno
 986: ERR 09:43:30.856715 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 987: ERR 09:43:30.874024 JVM: dns_gethostbysrv 3 h_errno
 988: ERR 09:43:30.874615 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 989: ERR 09:43:30.891639 JVM: dns_gethostbysrv 3 h_errno
 990: ERR 09:43:30.892265 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 991: ERR 09:43:30.908952 JVM: dns_gethostbysrv 3 h_errno
 992: ERR 09:43:30.909538 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error.
 993: NOT 09:43:30.985443 JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - propertyChanged - device.callagent.messages.0 value=0
 994: NOT 09:43:30.988054 JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - propertyChanged - device.callagent.messages.0 value=0
 995: ERR 09:43:31.002746 JVM: dns_gethostbysrv 3 h_errno

Best regards

Andre

U public_ip_office_network:3197 -> ip_of_the_pbx_public_ip:5060
REGISTER sip:dns_name_of_pbx SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed.
From: <sip:3000 at dns_name_of_pbx>;tag=58bfea206ae00003efc5fd41-f09664b5.
To: <sip:3000 at dns_name_of_pbx>.
Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240.
Max-Forwards: 70.
Date: Wed, 16 Dec 2009 08:44:00 GMT.
CSeq: 102 REGISTER.
User-Agent: Cisco-CP7942G/8.5.3.
Contact: <sip:3000 at 192.168.1.240:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-58bfea206ae0>";+u.sip!model.ccm.cisco.com="434".
Supported: (null),X-cisco-xsi-7.0.1.
Content-Length: 0.
Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP[mac address of phone] Load=term42.default Last=cm-reset-tcp".
Expires: 3600.
.

U ip_of_the_public_ip:5060 -> public_ip_office_network:38398
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed;received=public_ip_of_office_network;rport=38398.
From: <sip:3000 at dns_name_of_pbx>;tag=58bfea206ae00003efc5fd41-f09664b5.
To: <sip:3000 at dns_name_of_pbx>;tag=02KZmrmjpDyHD.
Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240.
CSeq: 102 REGISTER.
User-Agent: XXXX internal.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
WWW-Authenticate: Digest realm="centrex.iplink.no", nonce="b91fe311-c1d7-e111-bd36-001e0bc2ead2", algorithm=MD5, qop="auth".
Content-Length: 0.

Then it repeats itself.

Config file phone SEP[macaddress].cnf.xml this is downloaded to phone via TFTP.

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>username</secret>
<sshPassword>password</secret>
<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-Y</dateTemplate>
<timeZone>Central Europe Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>dns_name_of_pbx</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>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
g711a
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress>external IP address</natAddress>
<phoneLabel>Bilutleie</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>3000</featureLabel>
                <name>3000</name> 
                <displayName>3000</displayName>
                <contact>3000</contact>
                <proxy>dns_name_pbx</proxy>
                <port>5060</port>
                <autoAnswer>
                    <autoAnswerEnabled>2</autoAnswerEnabled>
                </autoAnswer>
                <callWaiting>3</callWaiting>
                <authName>3000</authName>
                <authPassword>secret</authPassword>
                <sharedLine>false</sharedLine>
                <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
                <messagesNumber>*97</messagesNumber>
                <ringSettingIdle>4</ringSettingIdle>
                <ringSettingActive>5</ringSettingActive>
                <forwardCallInfoDisplay>
                    <callerName>true</callerName>
                    <callerNumber>false</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>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.8-5-4S</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>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>00:00</displayOnDuration>
<displayIdleTimeout>00:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
</device>

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