[Freeswitch-users] Cisco phone registration

Andrew Cassidy andrew at cassidywebservices.co.uk
Wed Jun 12 16:12:38 MSD 2013


Not sure how legal this site is, but a whilre ago I found Cisco firmware
that isn't freely available here:
http://radiotwenterand.nl/~graver/cisco/SIP-7960/


On 12 June 2013 11:54, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> I think your next steps would be;
>
> 1) Make sure you have the latest firmware (as mentioned by Steven in other
> reply)
> 2) Manually check each one of those variables in the Cisco documentation,
> and see if any of them have impact on the real/rhost. This is a very
> tedious job :)
> 3) If the above doesn't impact, then revert to factory default config, and
> change ONLY the variables you need to make a valid register, see if the
> problem goes away, then re-enable each option and re-test
>
> Sadly these things are quite often trial and error, be prepared to sink a
> whole day into this.. just be sure to write an article up so others can
> learn from it!
>
> Cal
>
> On Wed, Jun 12, 2013 at 7:20 AM, andpe <andpe at poczta.onet.pl> wrote:
>
>> 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
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
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>>>
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>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
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