Re: [Freeswitch-users] Cisco phone registration
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 [1] http://blog.simplicitymedialtd.co.uk/476/analysis-of-cisco-7940-sip-alg-and-... On Tue, Jun 11, 2013 at 9:30 AM, andpe <andpe@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@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Check if you're running the latest firmware - older Cisco sip implementations did contain bugs I've seen similar behaviour in counterpart eyebeam btw On Wednesday, June 12, 2013, andpe 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@simplicitymedialtd.co.uk <javascript:_e({}, 'cvml', '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
[1] http://blog.simplicitymedialtd.co.uk/476/analysis-of-cisco-7940-sip-alg-and-...
On Tue, Jun 11, 2013 at 9:30 AM, andpe <andpe@poczta.onet.pl<javascript:_e({}, 'cvml', 'andpe@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@freeswitch.org <javascript:_e({}, 'cvml', 'consulting@freeswitch.org');> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e({}, 'cvml', 'FreeSWITCH-users@lists.freeswitch.org');> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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-...
On Tue, Jun 11, 2013 at 9:30 AM, andpe <andpe@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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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@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@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@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-...
On Tue, Jun 11, 2013 at 9:30 AM, andpe <andpe@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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participants (4)
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andpe
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Andrew Cassidy
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Cal Leeming [Simplicity Media Ltd]
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Steven Ayre