[asterisk-users] Trouble registering Cisco 7942
Stephen Reese
rsreese at gmail.com
Mon Nov 9 20:35:16 CST 2009
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby <wcselby at 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:102 at 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:asterisk at 209.251.157.91>;tag=as5b0b32f5
To: <sip:102 at IP of NAT:1027;user=phone;transport=udp>
Contact: <sip:asterisk at 209.251.157.91>
Call-ID: 090e1e583f29f9f000dd30ff5719f371 at 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>
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