On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby <wcse...@selbytech.com> wrote: > I think your featureLabel definition is wrong. > > On the login issue, ssh to the ip of the phone and login first with > the user/pass you defined in the file (admin/123), then at the second > login prompt use log/log. That should get you the log files which will > show you your error.
Thanks for the insight. After you mentioned that the syntax of the XML file may be wrong I looked around and found a more complete configuration I could find since mine was a copy and paste special. Using the new configuration the phone comes up but is unable register I *think* it may be an issue with NAT. When the phone fires up for the first time it tries to register for a while and the log didn't help much so I took a peak at the asterisk logging. It seems like packets are not getting back to the phone. I've enabled NAT in the configuration similar to how the other phones are configured but no dice. Note that the Asterisk device is not NATed but the phones are behind a NAT device. I get multiple of the following message in the phone: ERR 16:40:16.273722 JVM: %REG send failure: REGISTER On the asterisk server I keep getting NAT retries: Retransmitting #4 (NAT) to 71.226.175.137:1026: OPTIONS sip:1...@ip of NAT device:1027;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP ASTERISK IP:5060;branch=z9hG4bK53121c03;rport From: "asterisk" <sip:aster...@209.251.157.91>;tag=as5b0b32f5 To: <sip:1...@ip of NAT:1027;user=phone;transport=udp> Contact: <sip:aster...@209.251.157.91> Call-ID: 090e1e583f29f9f000dd30ff5719f...@209.251.157.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Nov 2009 02:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 Below is the full XML config for the phone: <device xsi:type="axl:XIPPhone" ctiid="9044468655"> <deviceProtocol>SIP</deviceProtocol> <sshUserId>admin</sshUserId> <sshPassword>123</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>M/D/Ya</dateTemplate> <timeZone>Eastern Standard/Daylight Time</timeZone> <ntps> <ntp> <name>192.43.244.18</name> <ntpMode>directedbroadcast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>Asterisk IP</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <sipProfile> <sipProxies> <backupProxy></backupProxy> <backupProxyPort></backupProxyPort> <emergencyProxy></emergencyProxy> <emergencyProxyPort></emergencyProxyPort> <outboundProxy>Asterisk IP</outboundProxy> <outboundProxyPort>5060</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> <preferredCodec>g711ulaw</preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>true</natEnabled> <natAddress>IP outside of NAT Device</natAddress> <phoneLabel>ATLAS</phoneLabel> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>true</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>Line 102</featureLabel> <proxy>Asterisk IP</proxy> <port>5060</port> <name>102</name> <displayName>ATLAS</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>102</authName> <authPassword>pass</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber>*97</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>102</contact> <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> <softKeyFile>softkeys.xml</softKeyFile> </sipProfile> <commonProfile> <phonePassword>1111</phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>1</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>1</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <webAccess>1</webAccess> <spanToPCPort>0</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer></loadServer> </vendorConfig> <versionStamp></versionStamp> <userLocale> <name>English_United_States</name> <uid>1</uid> <langCode>en_US</langCode> <version>1.0.0.0-1</version> <winCharSet>iso-8859-1</winCharSet> </userLocale> <networkLocale>United_States</networkLocale> <networkLocaleInfo> <name>United_States</name> <uid>64</uid> <version>1.0.0.0-1</version> </networkLocaleInfo> <deviceSecurityMode>1</deviceSecurityMode> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <idleURL></idleURL> <informationURL></informationURL> <messagesURL></messagesURL> <proxyServerURL></proxyServerURL> <servicesURL></servicesURL> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>4</transportLayerProtocol> <capfAuthMode>0</capfAuthMode> <capfList> <capf> <phonePort>3804</phonePort> </capf> </capfList> <certHash></certHash> <encrConfig>false</encrConfig> </device> _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users