Hi
I've seen once your article :-) It is very interesting.
However, C 7931 phone behaves as described. I hope this is a configuration error. Without the correct value "realm" I can not use multitenant in FreeSWITCH. I send you the configuration.
In wireshark I see that TO and FROM fields contain the IP and not the realm (domain name).
No matter how I set to challenge-realm in FreeSWITCH (auto_from or auto_to). Anyway, this is understandable when the phone sends the wrong as I had expected. In FreeSWITCH console, I get the message:
[WARNING] sofia_reg.c: 2515 Can not find user [1000 @ abcd] from abcf
You must define a domain called 'ABCD' in your directory and add a user with the id = "1000" attribute and you must configure your device to use the proper domain in it's authentication credentials.
config:
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>xxx</sshUserId>
<sshPassword>xxx</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Y</dateTemplate>
<timeZone>your timezone</timeZone>
<olsonTimeZone>your timezone</olsonTimeZone>
<ntps>
<ntp>
<name>x.x.x.x</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>mysip.server.domain</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword/>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>0</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP31.9-3-1-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>1</pcPort>
<settingsAccess>2</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>10:00</displayOnTime>
<displayOnDuration>00:01</displayOnDuration>
<displayIdleTimeout>00:05</displayIdleTimeout>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer/>
</vendorConfig>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL/>
<directoryURL/>
<idleTimeout>10</idleTimeout>
<idleURL/>
<informationURL/>
<messagesURL/>
<proxyServerURL>mysip.server.domain</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/>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy/><emergencyProxyPort/>
<outboundProxy/>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>true</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>0</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>600</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>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g729</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>16399</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>LABEL</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>1000</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>1000</name>
<displayName>1000</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>1000</authName>
<authPassword>xxxx</authPassword>
<contact>1000</contact>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>121</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<softKeyFile>SoftKey.xml</softKeyFile>
</sipProfile>
</device>
Andy
W dniu 2013-06-11 15:28:14 użytkownik Cal Leeming [Simplicity Media Ltd] <cal.leeming@simplicitymedialtd.co.uk> napisał:
Hello,
The phone shouldn't be sending the IP address, it should be using the hostname you specified.
I actually did an article, albeit about a NAT issue, on the Cisco 7940 which goes into detail about the SIP packets going to and from the server [1]. As you can see from this article, the phone is sending the hostname in the To field.
Therefore, this could either be a dodgy firmware or some config option you have set wrong.
Can you please give us the following;
* FS SIP trace logs showing the Cisco phone during registration and/or call state
* Cisco phone config (this can be extracted using the telnet server if you are not using TFTP)
Hope this helps
Cal