[Freeswitch-users] Cisco phone registration
andpe
andpe at poczta.onet.pl
Wed Jun 12 10:20:23 MSD 2013
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 at 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
[1] http://blog.simplicitymedialtd.co.uk/476/analysis-of-cisco-7940-sip-alg-and-nat-traversal-problems
On Tue, Jun 11, 2013 at 9:30 AM, andpe <andpe at poczta.onet.pl> wrote:
Hi
I have a problem with the registration of the Cisco 79xx phones. The SIP Message Header (MH) which sends the phone, in the fields "TO" and "FROM", places the IP address instead of the domain name. The configuration of the phone is set to the domain name (xx.yy.com.) Is there any way to set the phone to send the domain name in the field instead of an IP address? Is there a firmware version that sends MH correctly?
Andy
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