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+configurati... 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@dns_name_of_pbx>;tag=58bfea206ae00003efc5fd41-f09664b5. To: <sip:3000@dns_name_of_pbx>. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d@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@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@dns_name_of_pbx>;tag=58bfea206ae00003efc5fd41-f09664b5. To: <sip:3000@dns_name_of_pbx>;tag=02KZmrmjpDyHD. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d@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>