Topic 'Cisco 7970 NAT' in forum 'Cisco' - dslreports.com http://www.dslreports.com/forum/Cisco-7970-NAT-27610157 en Thu, 20 Jun 2013 06:23:36 EDT Thu, 20 Jun 2013 06:23:36 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27921479 said by st1nga:

I have 7940 that works fine behind NAT. Have not tried my 7970.
NAT works fine for me and others.

This is a dumb post. In my post I said that NAT is broken for "these phones" implying 797x series..

All others seem to be unaffected by this bug.]]> http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27921479 Thu, 17 Jan 2013 14:11:25 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27738486 NAT works fine for me and others.]]> http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27738486 Sun, 18 Nov 2012 05:46:59 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27629806 said by unixwolf:

You will never get this to work. NAT is broken on these phones...

Guess I best throw it into the technology grinder damn useless phone.

But in reality I'll probably just buy a VPN and create a site to site VPN.
--
Edrick Smith
Independent Film & Broadcast Producer
»edricksmith.com]]> http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27629806 Tue, 16 Oct 2012 21:21:37 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27628970 http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27628970 Tue, 16 Oct 2012 17:30:03 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27617087 http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27617087 Fri, 12 Oct 2012 16:41:08 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27611012 http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27611012 Wed, 10 Oct 2012 22:58:45 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27610894
It sounds like the issue is the two NATs... 7970 internal to internet, and internet to internal call manager. I've seen this before and quickly removed all NAT from the equation -- which includes telling the phones/pbx to stop trying to be "smart": don't fuck with any addresses. Depending on your setup, you may need to do that as well. Remote SIP phone behind a residential NAT router connecting to an internet visible pbx, if the routers doing NAT are SIP aware, then neither the phone nor pbx need to touch anything -- alternately, tell the natters to stop screwing with SIP. (not always possible)]]> http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27610894 Wed, 10 Oct 2012 22:14:08 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27610828 http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27610828 Wed, 10 Oct 2012 21:47:52 EDT Re: Cisco 7970 NAT http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27610808 through NAT? Nothing good will come of this. :-)

(I have the good sense to use VPN tunnels.)]]> http://www.dslreports.com/forum/Re-Cisco-7970-NAT-27610808 Wed, 10 Oct 2012 21:42:40 EDT Cisco 7970 NAT http://www.dslreports.com/forum/Cisco-7970-NAT-27610157
I can't seem to modify the SIP settings from the phone menu even with it unlocked as such I'm trying to use the SEP.cnf.xml file to have it pull its settings. I know it did at one point as it pulled the extension info and updated the firmware. From the status screen it looks like it pulls the config file.

Here's the config file,

<?xml version="1.0" encoding="UTF-8"?><device>   <deviceProtocol>SIP</deviceProtocol>   <sshUserId>admin</sshUserId>  <sshPassword>cisco</sshPassword>   <devicePool>                <dateTimeSetting>                        <dateTemplate>M/D/YA</dateTemplate>                        <timeZone>Eastern Standard/Daylight Time</timeZone>                        <ntps>                                <ntp>                                        <name>192.168.0.1</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><TRIXREMOTEIP></processNodeName>              </callManager>           </member>        </members>     </callManagerGroup>  </devicePool>   <commonProfile>     <phonePassword></phonePassword>     <backgroundImageAccess>true</backgroundImageAccess>     <callLogBlfEnabled>2</callLogBlfEnabled>  </commonProfile>   <loadInformation>SIP70.9-3-1-1S</loadInformation>   <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>0</webAccess>     <spanToPCPort>1</spanToPCPort>     <loggingDisplay>1</loggingDisplay>     <loadServer></loadServer>     <daysDisplayNotActive></daysDisplayNotActive>     <displayOnTime>07:00</displayOnTime>     <displayOnDuration>17:00</displayOnDuration>     <displayIdleTimeout>1:00</displayIdleTimeout>  </vendorConfig>   <deviceSecurityMode>1</deviceSecurityMode>   <authenticationURL>http://TRIXREMOTEIP/cisco/services/authentication.php</authenticationURL>  <directoryURL>http://TRIXREMOTEIP/xmlservices/PhoneDirectory.php</directoryURL>  <idleURL>http://TRIXREMOTEIP/xmlservices/index.php</idleURL>  <informationURL></informationURL>   <messagesURL></messagesURL>  <proxyServerURL></proxyServerURL>  <servicesURL>http://phone-xml.berbee.com/menu.xml</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>false</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>true</natEnabled>     <natAddress>TRIXREMOTEIP</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>201</phoneLabel>     <sipLines>        <line button="1">           <featureID>9</featureID>           <featureLabel>201</featureLabel>                   <name>201</name>                   <displayName>201</displayName>                   <contact>201</contact>            <proxy>TRIXREMOTEIP</proxy>           <port>5060</port>           <autoAnswer>              <autoAnswerEnabled>2</autoAnswerEnabled>           </autoAnswer>           <callWaiting>3</callWaiting>            <authName>201</authName>           <authPassword>PASSWORD</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>

Again I'm able to use soft phones without an issue, so I'm not sure if it's still possibly something at the remote end on Trixbox or in the config. I have the SIP Ports and Voice Ports open on the remote end.

Also here's my sip_nat.cnf

externhost = EXTERNIPlocalnet = 10.0.1.0/255.255.255.0

I've attached a dump from terminal showing the sip debug info, [officeip] is where the trixbox is and [remoteip] is where the phone is.
--
Edrick Smith
Independent Film & Broadcast Producer
»edricksmith.com
Log.txt 21,438 bytes
]]> http://www.dslreports.com/forum/Cisco-7970-NAT-27610157 Wed, 10 Oct 2012 18:24:54 EDT