[asterisk-users] Trouble registering Cisco 7942
Warren Selby
wcselby at selbytech.com
Tue Nov 10 09:13:58 CST 2009
In your sip.conf file, be sure to specify nat=no for the phone, even
though the phone is behind a nat device. The cisco phones handle sip
packets differently than the way asterisk expects, so you have to do
this in order to make asterisk send the way the phone will accept.
Thanks,
--Warren Selby
On Nov 9, 2009, at 8:35 PM, Stephen Reese <rsreese at gmail.com> wrote:
> 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>
>
> _______________________________________________
> -- 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
More information about the asterisk-users
mailing list