Mailing List Archive

Problem with changing 7975g phone to SIP
Hello,

I am trying to convert a 7975g phone to SIP and have it register to my
PBX (Firebrick FB2700  latest firmware).

I have done a full reset (3491672850*#) and have successfully updated
the bootloader and firmware to SIP75.9-4-2-1S.

However I am having trouble provisioning the phone and getting it to
register with my PBX. I can get as far as the phone saying it is
registering, but I do not see any SIP traffic from the phone. I am using
a passive lan tap on the rj45 cable from the phone.

I have tried a number of variations of the XMDefault.cnf.xml file. This
is the current version I am trying.

<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>10.151.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation437 model="Cisco IP Phone 7975"></loadInformation437>
</Default>

Similarly with the SEP<mac>.cnf.xml file.

[xml]
<device>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>

<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>

<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>10.151.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>

<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>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>

<networkLocale>United_States</networkLocale>

<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL>http://10.151.0.1/cisco/services/authentication.php</authenticationURL>
<directoryURL>http://10.151.0.1/xmlservices/PhoneDirectory.php</directoryURL>
<idleURL>http://10.151.0.1/xmlservices/index.php</idleURL>
<informationURL></informationURL>

<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://10.151.0.1/xmlservices/index.php</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>

<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>
<timerRegisterExpires>3600</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>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>

<natEnabled>false</natEnabled>
<natAddress></natAddress>

<stutterMsgWaiting>0</stutterMsgWaiting>

<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>

<phoneLabel>Roger</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>SipUser</featureLabel>
<name>SipUser</name>
<displayName>SipUser</displayName>
<contact>SipUser</contact>

<proxy>10.151.0.1</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>SipUser</authName>
<authPassword>SipPass</authPassword>

<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>

<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
</device>
[/xml]

This combination gets the phone into registering state. But no sip
traffic goes out on the LAN. In common with most attempts it also
results in the loss of the web server access to the phone.

$ nmap 10.151.0.129
Starting Nmap 7.80 (https://nmap.org ) at 2023-02-03 19:21 GMT
Nmap scan report for 10.151.0.129
Host is up (0.0011s latency).
All 1000 scanned ports on 10.151.0.129 are closed

Nmap done: 1 IP address (1 host up) scanned in 1.54 seconds

So I have something wrong somewhere, but I cannot figure out what.

Anyone got any ideas?

Thanks.

Roger
Re: Problem with changing 7975g phone to SIP [ In reply to ]
I can only refer you to some resources available on voip-info.org from years ago:

https://www.voip-info.org/cisco-7942-with-local-pbx/

https://www.voip-info.org/standalone-cisco-79457965/

one of those two might provide some guidance.

It's been a long time since I did this, and I recall it becoming much more difficult after 8.4? or 8.5? think firmware.
Going from very foggy scraps of memory, if the configuration file isn't the way it wants, it will do as you describe. It will say registering but do nothing. I e-wasted mine after enough wrangling with them and Asterisk or trying directly to a SIP provider. The older 7940/60 will function as a basic SIP phone without NAT but other than for some fun during ham radio events, lacking security or traversal capabilities makes them sort of rough nowadays.

Hope any of that is helpful.

Best,

Adam Pawlowski
SUNYAB

From: cisco-voip <cisco-voip-bounces@puck.nether.net> On Behalf Of roger
Sent: Friday, February 3, 2023 2:29 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Problem with changing 7975g phone to SIP

Hello,

I am trying to convert a 7975g phone to SIP and have it register to my PBX (Firebrick FB2700 latest firmware).

I have done a full reset (3491672850*#) and have successfully updated the bootloader and firmware to SIP75.9-4-2-1S.

However I am having trouble provisioning the phone and getting it to register with my PBX. I can get as far as the phone saying it is registering, but I do not see any SIP traffic from the phone. I am using a passive lan tap on the rj45 cable from the phone.

I have tried a number of variations of the XMDefault.cnf.xml file. This is the current version I am trying.

<Default>

<callManagerGroup>

<members>

<member priority="0">

<callManager>

<ports>

<ethernetPhonePort>2000</ethernetPhonePort>

</ports>

<processNodeName>10.151.0.1</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

<loadInformation437 model="Cisco IP Phone 7975"></loadInformation437>

</Default>
Similarly with the SEP<mac>.cnf.xml file.

[xml]

<device>



<deviceProtocol>SIP</deviceProtocol>



<sshUserId>admin</sshUserId>

<sshPassword>cisco</sshPassword>



<devicePool>

<dateTimeSetting>

<dateTemplate>D-M-Y</dateTemplate>

<timeZone>GMT Standard/Daylight Time</timeZone>

<ntps>

<ntp>

<name>pool.ntp.org</name>

<ntpMode>Unicast</ntpMode>

</ntp>

</ntps>

</dateTimeSetting>



<callManagerGroup>

<members>

<member priority="0">

<callManager>

<ports>

<ethernetPhonePort>2000</ethernetPhonePort>

<sipPort>5060</sipPort>

<securedSipPort>5061</securedSipPort>

</ports>

<processNodeName>10.151.0.1</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

</devicePool>



<commonProfile>

<phonePassword></phonePassword>

<backgroundImageAccess>true</backgroundImageAccess>

<callLogBlfEnabled>2</callLogBlfEnabled>

</commonProfile>



<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>

<spanToPCPort>1</spanToPCPort>

<loggingDisplay>1</loggingDisplay>

<loadServer></loadServer>

</vendorConfig>



<networkLocale>United_States</networkLocale>



<networkLocaleInfo>

<name>United_States</name>

<uid>64</uid>

<version>1.0.0.0-1</version>

</networkLocaleInfo>



<deviceSecurityMode>1</deviceSecurityMode>



<authenticationURL>http://10.151.0.1/cisco/services/authentication.php<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2F10.151.0.1%2Fcisco%2Fservices%2Fauthentication.php&data=05%7C01%7Cajp26%40buffalo.edu%7Cce26ed1e5f124d98c5a908db061ceb64%7C96464a8af8ed40b199e25f6b50a20250%7C0%7C0%7C638110493857277653%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C2000%7C%7C%7C&sdata=sGooV8b7j2iBi1CIx1GzM1MQgkCqONagg0385hjyDzo%3D&reserved=0></authenticationURL>

<directoryURL>http://10.151.0.1/xmlservices/PhoneDirectory.php<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2F10.151.0.1%2Fxmlservices%2FPhoneDirectory.php&data=05%7C01%7Cajp26%40buffalo.edu%7Cce26ed1e5f124d98c5a908db061ceb64%7C96464a8af8ed40b199e25f6b50a20250%7C0%7C0%7C638110493857277653%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C2000%7C%7C%7C&sdata=fLNzQGPeAU84QWS66oif1rWPIVAu8axNubh3J9E9O%2BE%3D&reserved=0></directoryURL>

<idleURL>http://10.151.0.1/xmlservices/index.php<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2F10.151.0.1%2Fxmlservices%2Findex.php&data=05%7C01%7Cajp26%40buffalo.edu%7Cce26ed1e5f124d98c5a908db061ceb64%7C96464a8af8ed40b199e25f6b50a20250%7C0%7C0%7C638110493857277653%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C2000%7C%7C%7C&sdata=Ii6yKiCriyQu3UtkyytPrIixiMHp697jESvvf3K4GuE%3D&reserved=0></idleURL>

<informationURL></informationURL>



<messagesURL></messagesURL>

<proxyServerURL></proxyServerURL>

<servicesURL>http://10.151.0.1/xmlservices/index.php<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2F10.151.0.1%2Fxmlservices%2Findex.php&data=05%7C01%7Cajp26%40buffalo.edu%7Cce26ed1e5f124d98c5a908db061ceb64%7C96464a8af8ed40b199e25f6b50a20250%7C0%7C0%7C638110493857277653%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C2000%7C%7C%7C&sdata=Ii6yKiCriyQu3UtkyytPrIixiMHp697jESvvf3K4GuE%3D&reserved=0></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>



<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>

<timerRegisterExpires>3600</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>false</transferOnhookEnabled>

<enableVad>false</enableVad>

<preferredCodec>none</preferredCodec>

<dtmfAvtPayload>101</dtmfAvtPayload>

<dtmfDbLevel>3</dtmfDbLevel>

<dtmfOutofBand>avt</dtmfOutofBand>

<alwaysUsePrimeLine>false</alwaysUsePrimeLine>

<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>

<kpml>3</kpml>



<natEnabled>false</natEnabled>

<natAddress></natAddress>



<stutterMsgWaiting>0</stutterMsgWaiting>



<callStats>false</callStats>

<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>

<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>



<startMediaPort>16384</startMediaPort>

<stopMediaPort>32766</stopMediaPort>



<voipControlPort>5060</voipControlPort>

<dscpForAudio>184</dscpForAudio>

<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>

<dialTemplate>dialplan.xml</dialTemplate>



<phoneLabel>Roger</phoneLabel>

<sipLines>

<line button="1">

<featureID>9</featureID>

<featureLabel>SipUser</featureLabel>

<name>SipUser</name>

<displayName>SipUser</displayName>

<contact>SipUser</contact>



<proxy>10.151.0.1</proxy>

<port>5060</port>

<autoAnswer>

<autoAnswerEnabled>2</autoAnswerEnabled>

</autoAnswer>

<callWaiting>3</callWaiting>



<authName>SipUser</authName>

<authPassword>SipPass</authPassword>



<sharedLine>false</sharedLine>

<messageWaitingLampPolicy>1</messageWaitingLampPolicy>

<messagesNumber>*97</messagesNumber>

<ringSettingIdle>4</ringSettingIdle>

<ringSettingActive>5</ringSettingActive>



<forwardCallInfoDisplay>

<callerName>true</callerName>

<callerNumber>false</callerNumber>

<redirectedNumber>false</redirectedNumber>

<dialedNumber>true</dialedNumber>

</forwardCallInfoDisplay>

</line>

</sipLines>

</sipProfile>

</device>

[/xml]
This combination gets the phone into registering state. But no sip traffic goes out on the LAN. In common with most attempts it also results in the loss of the web server access to the phone.

$ nmap 10.151.0.129

Starting Nmap 7.80 ( https://nmap.org<https://nam12.safelinks.protection.outlook.com/?url=https%3A%2F%2Fnmap.org%2F&data=05%7C01%7Cajp26%40buffalo.edu%7Cce26ed1e5f124d98c5a908db061ceb64%7C96464a8af8ed40b199e25f6b50a20250%7C0%7C0%7C638110493857277653%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C2000%7C%7C%7C&sdata=uRNQqRYgVBwGgld0%2B49I6uPdMIsldqz6rmv1lLAwITM%3D&reserved=0> ) at 2023-02-03 19:21 GMT

Nmap scan report for 10.151.0.129

Host is up (0.0011s latency).

All 1000 scanned ports on 10.151.0.129 are closed



Nmap done: 1 IP address (1 host up) scanned in 1.54 seconds
So I have something wrong somewhere, but I cannot figure out what.

Anyone got any ideas?

Thanks.

Roger
Re: Problem with changing 7975g phone to SIP [ In reply to ]
webAccess set to 1 is actually disabled.

Good reference here- https://usecallmanager.nz/sepmac-cnf-xml.html

Hopefully you can get in and view the console logs then via the webpage.
It's probably having issues parsing something.

Brian Meade

On Fri, Feb 3, 2023, 2:29 PM roger <roger@beardandsandals.co.uk> wrote:

> Hello,
>
> I am trying to convert a 7975g phone to SIP and have it register to my PBX
> (Firebrick FB2700 latest firmware).
>
> I have done a full reset (3491672850*#) and have successfully updated the
> bootloader and firmware to SIP75.9-4-2-1S.
>
> However I am having trouble provisioning the phone and getting it to
> register with my PBX. I can get as far as the phone saying it is
> registering, but I do not see any SIP traffic from the phone. I am using a
> passive lan tap on the rj45 cable from the phone.
>
> I have tried a number of variations of the XMDefault.cnf.xml file. This is
> the current version I am trying.
>
> <Default>
> <callManagerGroup>
> <members>
> <member priority="0">
> <callManager>
> <ports>
> <ethernetPhonePort>2000</ethernetPhonePort>
> </ports>
> <processNodeName>10.151.0.1</processNodeName>
> </callManager>
> </member>
> </members>
> </callManagerGroup>
> <loadInformation437 model="Cisco IP Phone 7975"></loadInformation437>
> </Default>
>
> Similarly with the SEP<mac>.cnf.xml file.
>
> [xml]
> <device>
>
> <deviceProtocol>SIP</deviceProtocol>
>
> <sshUserId>admin</sshUserId>
> <sshPassword>cisco</sshPassword>
>
> <devicePool>
> <dateTimeSetting>
> <dateTemplate>D-M-Y</dateTemplate>
> <timeZone>GMT Standard/Daylight Time</timeZone>
> <ntps>
> <ntp>
> <name>pool.ntp.org</name>
> <ntpMode>Unicast</ntpMode>
> </ntp>
> </ntps>
> </dateTimeSetting>
>
> <callManagerGroup>
> <members>
> <member priority="0">
> <callManager>
> <ports>
> <ethernetPhonePort>2000</ethernetPhonePort>
> <sipPort>5060</sipPort>
> <securedSipPort>5061</securedSipPort>
> </ports>
> <processNodeName>10.151.0.1</processNodeName>
> </callManager>
> </member>
> </members>
> </callManagerGroup>
> </devicePool>
>
> <commonProfile>
> <phonePassword></phonePassword>
> <backgroundImageAccess>true</backgroundImageAccess>
> <callLogBlfEnabled>2</callLogBlfEnabled>
> </commonProfile>
>
> <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>
> <spanToPCPort>1</spanToPCPort>
> <loggingDisplay>1</loggingDisplay>
> <loadServer></loadServer>
> </vendorConfig>
>
> <networkLocale>United_States</networkLocale>
>
> <networkLocaleInfo>
> <name>United_States</name>
> <uid>64</uid>
> <version>1.0.0.0-1</version>
> </networkLocaleInfo>
>
> <deviceSecurityMode>1</deviceSecurityMode>
>
> <authenticationURL>http://10.151.0.1/cisco/services/authentication.php</authenticationURL>
> <directoryURL>http://10.151.0.1/xmlservices/PhoneDirectory.php</directoryURL>
> <idleURL>http://10.151.0.1/xmlservices/index.php</idleURL>
> <informationURL></informationURL>
>
> <messagesURL></messagesURL>
> <proxyServerURL></proxyServerURL>
> <servicesURL>http://10.151.0.1/xmlservices/index.php</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>
>
> <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>
> <timerRegisterExpires>3600</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>false</transferOnhookEnabled>
> <enableVad>false</enableVad>
> <preferredCodec>none</preferredCodec>
> <dtmfAvtPayload>101</dtmfAvtPayload>
> <dtmfDbLevel>3</dtmfDbLevel>
> <dtmfOutofBand>avt</dtmfOutofBand>
> <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
> <kpml>3</kpml>
>
> <natEnabled>false</natEnabled>
> <natAddress></natAddress>
>
> <stutterMsgWaiting>0</stutterMsgWaiting>
>
> <callStats>false</callStats>
> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
>
> <startMediaPort>16384</startMediaPort>
> <stopMediaPort>32766</stopMediaPort>
>
> <voipControlPort>5060</voipControlPort>
> <dscpForAudio>184</dscpForAudio>
> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
> <dialTemplate>dialplan.xml</dialTemplate>
>
> <phoneLabel>Roger</phoneLabel>
> <sipLines>
> <line button="1">
> <featureID>9</featureID>
> <featureLabel>SipUser</featureLabel>
> <name>SipUser</name>
> <displayName>SipUser</displayName>
> <contact>SipUser</contact>
>
> <proxy>10.151.0.1</proxy>
> <port>5060</port>
> <autoAnswer>
> <autoAnswerEnabled>2</autoAnswerEnabled>
> </autoAnswer>
> <callWaiting>3</callWaiting>
>
> <authName>SipUser</authName>
> <authPassword>SipPass</authPassword>
>
> <sharedLine>false</sharedLine>
> <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
> <messagesNumber>*97</messagesNumber>
> <ringSettingIdle>4</ringSettingIdle>
> <ringSettingActive>5</ringSettingActive>
>
> <forwardCallInfoDisplay>
> <callerName>true</callerName>
> <callerNumber>false</callerNumber>
> <redirectedNumber>false</redirectedNumber>
> <dialedNumber>true</dialedNumber>
> </forwardCallInfoDisplay>
> </line>
> </sipLines>
> </sipProfile>
> </device>
> [/xml]
>
> This combination gets the phone into registering state. But no sip traffic
> goes out on the LAN. In common with most attempts it also results in the
> loss of the web server access to the phone.
>
> $ nmap 10.151.0.129
> Starting Nmap 7.80 ( https://nmap.org ) at 2023-02-03 19:21 GMT
> Nmap scan report for 10.151.0.129
> Host is up (0.0011s latency).
> All 1000 scanned ports on 10.151.0.129 are closed
>
> Nmap done: 1 IP address (1 host up) scanned in 1.54 seconds
>
> So I have something wrong somewhere, but I cannot figure out what.
>
> Anyone got any ideas?
>
> Thanks.
>
> Roger
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>