@rsarceno wrote:
Asterisk 13.9.1 Freepbx 13.0.190.7
Chan_SIP ext 5001, I can register using Cisco 7960 but when I tried the same extension using Cisco 7965, I'm getting a warning message[2016-12-28 21:45:35] WARNING[26939] pjproject: sip_transactio Unable to register REGISTER transaction (key exists)
Firmware was download from Cisco. I also tried SIP45.9-4-2SR2-2S with the same result.
I also tried setting NAT = Never on the extension.I appreciate if someone can verify the XML and see if I miss anything.
Thanks
XMLDefault.cnf.xml
<Default> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <!--<analogAccessPort>2002</analogAccessPort> <digitalAccessPort>2001</digitalAccessPort> --> <ethernetPhonePort>2000</ethernetPhonePort> <mgcpPorts> <listen>2427</listen> <keepAlive>2428</keepAlive> </mgcpPorts> </ports> <!-- IP Address of the PBX Server --> <processNodeName></processNodeName> <!-- <processNodeName>162.255.22.118</processNodeName> --> </callManager> </member> </members> </callManagerGroup> <!-- <loadInformation431 model="Cisco 7937"></loadInformation431> --> <!-- <loadInformation436 model="Cisco 7965"></loadInformation436> --> <loadInformation436 model="Cisco 7965">SIP45.8-5-4S</loadInformation436> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <idleURL></idleURL> <informationURL></informationURL> <messagesURL></messagesURL> <servicesURL></servicesURL> </Default>
SEPmacaddress.cnf.xml
<device xsi:type="axl:XIPPhone" ctiid="9166475392"> <!-- FIXME: Set your cti ID --> <deviceProtocol>SIP</deviceProtocol> <sshUserId>default</sshUserId> <sshPassword>user</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>M/D/YA</dateTemplate> <!-- FIXME: Set your preferred date format and timezone here --> <timeZone>Pacific Standard/Daylight Time</timeZone> <!-- <timeZone>Central Standard/Daylight Time</timeZone> --> <ntps> <ntp> <name> 209.118.204.201</name> <ntpMode>unicast</ntpMode> </ntp> <ntp> <name>66.228.59.187</name> <ntpMode>unicast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <!-- FIXME: PBX IP Address --> <processNodeName>162.255.22.118</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> <!-- Force short registration timeout to keep NAT connection alive --> <timerRegisterExpires>3600</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <!-- from false to true - modified 12-28 9pm --> <remotePartyID>true</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>false</natEnabled> <natAddress></natAddress> <!-- FIXME: Text to display top right conner--> <phoneLabel>Test 7965</phoneLabel> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <sipLines> <line button="1"> <featureID>9</featureID> <!-- FIXME: Text to display next to line button #1 --> <featureLabel>5001</featureLabel> <!-- FIXME: FQDN or IP of your SIP proxy --> <proxy>162.255.22.118</proxy> <port>5060</port> <!-- FIXME: SIP username --> <name>5001</name> <!-- FIXME: Name to display on outbound caller ID --> <displayName>5001</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <!-- FIXME: SIP authorization name (often matches username) --> <authName>5001</authName> <!-- FIXME: SIP authorization password --> <authPassword>PasswordTest7965</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber></messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <!-- FIXME: Ext Number --> <contact>5001</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> <line button="2"> <featureID>9</featureID> <featureLabel></featureLabel> <speedDialNumber></speedDialNumber> </line> </sipLines> <voipControlPort>5060</voipControlPort> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> </sipProfile> <commonProfile> <phonePassword></phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile> <loadInformation>SIP45.8-5-4S</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>1</webAccess> <daysDisplayNotActive>1,7</daysDisplayNotActive> <displayOnTime>08:00</displayOnTime> <displayOnDuration>10:30</displayOnDuration> <displayIdleTimeout>01:00</displayIdleTimeout> <spanToPCPort>1</spanToPCPort> </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>
Posts: 1
Participants: 1