Cisco 7942 XML config help

#1
Hi Folks,

I am trying to get a Cisco 7942 phone working. These versions of the phones use an XML config file so is different to the 7940.

I have managed to get the phone working to the point it seems to register and I can get a dialtone but I cant dial any numbers.

Anyone got any experience of these?

Model: CP-7942G
Firmware: SIP42.8-5-3S

I can post a copy of the config if it would be of any use

Thanks

Re: Cisco 7942 XML config help

#3
This is what I have so far.

I have tried various settings with outbound proxy and NAT but nothing seems to work.

Thanks

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-Y</dateTemplate>
<timeZone>United Kingdom Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>192.168.1.1</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>sip.voipfone.net</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy>sip.voipfone.net</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
<registerWithProxy>false</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-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>60</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natReceivedProcessing>true</natReceivedProcessing>
<natEnabled>true</natEnabled>
<natAddress>MY IP Provide by ISP</natAddress>
<phoneLabel>VoipFone </phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Voipfone Number</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>Me</name>
<displayName>Voipfone-account-number</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>Voipfone-account-number</authName>
<authPassword>Voipfone-password</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<ringSettingIdle>4</ringSettingIdle>
<messagesNumber>1571</messagesNumber>
<ringSettingActive>5</ringSettingActive>
<contact>Voipfone-account-number</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.8-5-3S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>1</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp></versionStamp>
<userLocale>
<name>English_United_States</name>
<langCode>en_US</langCode>
</userLocale>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>

Re: Cisco 7942 XML config help

#4
You say above you can get it to register but unable to dial any numbers... What happens when you dial 152? Silence? Can you receive any inbound calls on the device?

Pretty sure this is a NAT issue...

I'm no experience with these phones myself but after reading up on them the first thing I read was this:
The Cisco 7942 phone expects generally to work with the CISCO callmanager software which is in the local network. Asterisk or other SIP Proxy can fake the CISCO callmanager. Make sure that the SIP proxy is not assuming NAT when connecting the phones as extensions. The CISCO phones expect SIP messages only on the preconfigured voipControlPort. If the SIP Proxy expects a NATted device it sends SIP answers back to the port where it received the SIP message from. This does not work here. Therefore out-of-the box NAT traversal mostly does not work with these phones.

Because the phones always expect SIP messages on the voipControlPort one can only use these phones behind NAT if one can configure the firewall.
If you were running your own Asterisk PBX just switching NAT to No and using the XML code below should be all that's needed.

Try resetting the phone and proceed to load this:

Code: Select all

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>United Kingdom Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>192.168.1.1</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>195.189.173.27</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<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>60</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>g711alaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>VoipFone</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Voipfone Number</featureLabel>
<port>5060</port>
<name>Me</name>
<displayName>Voipfone-account-number</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>Voipfone-account-number</authName>
<authPassword>Voipfone-password</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>1571</messagesNumber>  
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>Voipfone-account-number</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.8-5-3S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort> 
<settingsAccess>1</settingsAccess>
<garp>1</garp> 
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<sshAccess>1</sshAccess>
<sshPort>22</sshPort> 
<webAccess>0</webAccess>
<spanToPCPort>0</spanToPCPort>  
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp></versionStamp>
<userLocale>
<name></name> 
<uid>1</uid> 
<langCode></langCode> 
<version></version> 
<winCharSet></winCharSet> 
</userLocale>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
Don't forget to change some of the parameters such as your Voipfone UserID and Password. (I'm not expecting this to work though as Voipfone PBX would have NAT enabled.)

Post back with the results, we will try a different approach using NAT settings in the XML if above is a no go.
For everything VoIP
www.ukvoipforums.com

Re: Cisco 7942 XML config help

#5
Thanks, I will give it a go tomorrow.

The phone works perfectly with a test asterisk setup I have both on the same subnet and via NAT. This is only a internal system with no external connections. I was hoping to try and connect the phone directly to voipfone but I guess I may have to fire up a permanent asterisk solution. sorting out port forwarding etc is not an issue so might have a play with that as well.

I see in the config you have removed all the proxy references and the line that defines the voipfone server in the <sipLines> section. Will be interesting to see if it works.

The current config doesn't hang at registering like some have and gives a dial tone. Dialling a number just gives silence and ringing my number goes to voicemail.

Cheers

Re: Cisco 7942 XML config help

#6
I removed the following:

Code: Select all

<ethernetPhonePort>2000</ethernetPhonePort>
<securedSipPort>5061</securedSipPort>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy>sip.voipfone.net</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
<registerWithProxy>false</registerWithProxy>
</sipProxies>
<natReceivedProcessing>true</natReceivedProcessing>
<natEnabled>true</natEnabled>
<natAddress>MY IP Provide by ISP</natAddress>
<proxy>USECALLMANAGER</proxy>
Basically I want you to try it without any NAT settings and I removed the proxy settings as I think (wild shot in the dark here) that if none are set it will use the <processNodeName>195.189.173.27</processNodeName> as the default proxy IP.

195.189.173.27 = sip.voipfone.net

The XML guide I posted is what's recommended for use with an asterisk setup so should be good for Voipfone, failing that I would add back what I removed from my XML and try again.

I did notice that you had set <registerWithProxy>false</registerWithProxy> in your XML, that should be set to true. (if you revert back to your XML)
For everything VoIP
www.ukvoipforums.com

Re: Cisco 7942 XML config help

#7
Hi,

Your config has come up with "Unprovisioned"

The log is saying: Startup Module Loader|cip.cfg.h:? - No SIP Proxies Configured

I have added <proxy>USECALLMANAGER</proxy> and its back to the same state as before. Also tried the IP directly in the proxy tags and its the same.

The logs show:
JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - FULLY_REGISTERED - Resetting retry installer interval

So it may be registered????

When I try to make a call:

RR 20:12:23.507520 DSP: wcTrans*** Invalid response 5
ERR 20:12:23.508271 DSP: Tone*** Connect/disconnect fails: -1
ERR 20:12:23.585587 DSP: wcTrans*** Invalid response 5
ERR 20:12:23.586618 DSP: Tone*** Connect/disconnect fails: -1
ERR 20:12:23.706321 JVM: CFG Error: config_get_line_string(): line1_maxnumcalls - line 0 out of range
ERR 20:12:23.707134 JVM: CFG Error: config_get_line_string(): line1_busy_trigger - line 0 out of range
ERR 20:12:23.886614 DSP: wcTrans*** Invalid response 5
ERR 20:12:23.887356 DSP: Tone*** Connect/disconnect fails: -1
ERR 20:12:24.146653 DSP: wcTrans*** Invalid response 5
ERR 20:12:24.147391 DSP: Tone*** Connect/disconnect fails: -1
NOT 20:12:24.974134 DSP: CODEC[0] G.711 direction:2 cost:26 budget:100 available
NOT 20:12:24.974709 DSP: CODEC[1] G.729A or G.729AB direction:2 cost:41 budget:100 available
NOT 20:12:24.975245 DSP: CODEC[2] G.729 or G.729B direction:2 cost:41 budget:100 available
NOT 20:12:24.975761 DSP: CODEC[3] LINEAR 8 or 16kHz direction:2 cost:26 budget:100 available
NOT 20:12:24.976246 DSP: CODEC[4] G.722 direction:2 cost:32767 budget:100 NOT available
NOT 20:12:24.976725 DSP: CODEC[5] iLBC direction:2 cost:48 budget:100 available
NOT 20:12:24.977198 DSP: STREAM- GetCapableCodecList requestType:2 bitmap:0x2f
NOT 20:12:24.989575 DSP: ====== A phone call starts ....
NOT 20:12:24.990233 DSP: ETHSTAT- (unicast, broadcast, multicast) rx = 99, 732, 9; tx = 107, 5
NOT 20:12:24.990779 DSP: MIB2- ipInDelivers= 177, udpInDG= 46, udpOutDG= 37, udpNoPort= 95, udpInErr= 0, icmpInDestUnreach = 0, icmpOutDestUnreach = 0
NOT 20:12:24.991338 DSP: STREAM- OpenEgressChan- ChanType 1, local (multicast host 0, port 4000), MedType 2, Period 20, stream (3, 3) mix (0, 0)
NOT 20:12:24.991897 DSP: STREAM- OpenEgressChan --> local port x4000, reserved port x0, --> Chan 0
NOT 20:12:24.993254 DSP: Subtracted for CODEC[0] G.711 direction:0 cost:13 old budget:100
NOT 20:12:24.996275 DSP: TRSTREAM, isrEgressReqDiscardpreStreaming 0
NOT 20:12:26.158394 DSP: STREAM- CloseEgressChan- ChanType 1, stream (3, 3) --> Chan 0
ERR 20:12:26.158981 DSP: MT:***RTP- zero (0) RTP packets receveid in 2 + seconds
NOT 20:12:26.159972 DSP: ETHSTAT- (unicast, broadcast, multicast) rx = 102, 734, 9; tx = 109, 5
NOT 20:12:26.160667 DSP: MIB2- ipInDelivers= 180, udpInDG= 49, udpOutDG= 39, udpNoPort= 95, udpInErr= 0, icmpInDestUnreach = 0, icmpOutDestUnreach = 0
NOT 20:12:26.161240 DSP: select - total select returned = 4, last snapshot = 0

NOT 20:12:26.161732 DSP: IP 0, stm SSRC x0:0, age 0, stTime 0, MCst 0
NOT 20:12:26.162198 DSP: Packets discarded- Sequece number smaller than FIRST's 0, Clock Reset 0, Negative-Jitter packets 0, Dup Seq 0
NOT 20:12:26.165124 DSP: Added back for CODEC[0] G.711 direction:0 cost:13 old budget:87
NOT 20:12:26.166391 DSP: ETHSTAT- (unicast, broadcast, multicast) rx = 102, 734, 9; tx = 109, 5
NOT 20:12:26.167015 DSP: MIB2- ipInDelivers= 180, udpInDG= 49, udpOutDG= 39, udpNoPort= 95, udpInErr= 0, icmpInDestUnreach = 0, icmpOutDestUnreach = 0
WRN 20:12:26.406321 JVM: Startup Module Loader|cip.mmgr.dt:? - [MediaMgrSM]: Unhandled Event, State = StateOnHook Event = EventEndcall

I am guessing the "MT:***RTP- zero (0) RTP packets receveid" bit could be significant

Re: Cisco 7942 XML config help

#8
Interesting, do you get a dialtone with the above configuration? It does appear to think its registered but what i would do is confirm that with Voipfone. If your registereing the phone against an extension you can confirm that it is indeed registered via you voipfone control panel.

If your registered against the 30xxxxxx account number you will need to give them a call and they will tell you if your phone is registered.

If it is indeed registered the next step is to look at the NAT setup and firewall as you know zero RTP packets aint good. I Will do some more reaserch on this later and post back.
For everything VoIP
www.ukvoipforums.com

Re: Cisco 7942 XML config help

#10
Hi,

Voipfone support said that the phone wasn't registering. I tried ther NAT setting as both true and false but neither worked.

I have given up on the phone and its being relegated to work on an Asterisk system.

I have found a 7940 buried in the store room so I am giving that a go.

Thanks for your help

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