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Andres Sukanec
Beginner

IP Phone 7970 not registering with Asterisk

Hello I'm sorry if this is a recurring subject, but I'm having problems registering my SIP configured 7970 phone with my Asterisk server.

The phone is using SIP 70.9-2-1S firmware, and my SEPmac.conf.xml file looks like this:

 

<?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>192.168.1.201</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>192.168.1.201</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
  </devicePool>

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

  <loadInformation>SIP70.9-2-1S</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></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>

   <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>g711ualaw</preferredCodec>
     <dtmfAvtPayload>101</dtmfAvtPayload>
     <dtmfDbLevel>3</dtmfDbLevel>
     <dtmfOutofBand>avt</dtmfOutofBand>
     <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>

     <natEnabled>true</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>MyPhoneLabel</phoneLabel>
     <sipLines>
        <line button="1">
           <featureID>9</featureID>
           <featureLabel>202</featureLabel>
                   <name>202</name>
                   <displayName>202</displayName>
                   <contact>202</contact>

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

           <authName>202</authName>
           <authPassword>astrum.2o16</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>

I've tried multiple config files but this is the only one that seems to be able to send the REGISTER request to the server (the others simply hung up displaying registering).

 

By debuging the Asterisk server (located at 192.168.1.201) with "asterisk -rvvvvvvvvvvvvvvvvvvv" and "sip set debug ip 192.168.1.139" (correspondig this to the phones IP) I get the following:

 

[Jul  1 15:49:50] <--- SIP read from UDP:192.168.1.139:51322 --->
[Jul  1 15:49:50] REFER sip:192.168.1.201 SIP/2.0
[Jul  1 15:49:50] Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK04a9cc08
[Jul  1 15:49:50] From: <sip:001956a8d862@192.168.1.139>;tag=001956a8d8620002c4ee52a8-c4ed4788
[Jul  1 15:49:50] To: <sip:192.168.1.201>
[Jul  1 15:49:50] Call-ID: 001956a8-d8620002-81cb1b68-c875d648@192.168.1.139
[Jul  1 15:49:50] Date: Wed, 11 May 2011 13:14:35 GMT
[Jul  1 15:49:50] CSeq: 1000 REFER
[Jul  1 15:49:50] User-Agent: Cisco-CP7970G/9.2.1
[Jul  1 15:49:50] Expires: 10
[Jul  1 15:49:50] Max-Forwards: 70
[Jul  1 15:49:50] Contact: <sip:001956a8d862@192.168.1.139:5060>
[Jul  1 15:49:50] Require: norefersub
[Jul  1 15:49:50] Referred-By: <sip:001956a8d862@192.168.1.139>
[Jul  1 15:49:50] Refer-To: cid:4a4b3828@192.168.1.139
[Jul  1 15:49:50] Content-Id: <4a4b3828@192.168.1.139>
[Jul  1 15:49:50] Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
[Jul  1 15:49:50] Content-Length: 1369
[Jul  1 15:49:50] Content-Type: application/x-cisco-alarm+xml
[Jul  1 15:49:50] Content-Disposition: session;handling=required
[Jul  1 15:49:50]
[Jul  1 15:49:50] <?xml version="1.0" encoding="UTF-8"?>
[Jul  1 15:49:50] <x-cisco-alarm>
[Jul  1 15:49:50] <Alarm Name="LastOutOfServiceInformation">
[Jul  1 15:49:50] <ParameterList>
[Jul  1 15:49:50] <String name="DeviceName">SEP001956A8D862</String>
[Jul  1 15:49:50] <String name="DeviceIPv4Address">192.168.0.116/24</String>
[Jul  1 15:49:50] <String name="IPv4DefaultGateway">192.168.0.201</String>
[Jul  1 15:49:50] <String name="DeviceIPv6Address"></String>
[Jul  1 15:49:50] <String name="IPv6DefaultGateway">fe80::238:dfff:fe9b:4419</String>
[Jul  1 15:49:50] <String name="ModelNumber">CP-7970G</String>
[Jul  1 15:49:50] <String name="NeighborIPv4Address">192.168.0.100</String>
[Jul  1 15:49:50] <String name="NeighborIPv6Address"></String>
[Jul  1 15:49:50] <String name="NeighborDeviceID">SEP001956BD29B5</String>
[Jul  1 15:49:50] <String name="NeighborPortID">Port 1</String>
[Jul  1 15:49:50] <Enum name="DHCPv4Status">3</Enum>
[Jul  1 15:49:50] <Enum name="DHCPv6Status">0</Enum>
[Jul  1 15:49:50] <Enum name="TFTPCfgStatus">1</Enum>
[Jul  1 15:49:50] <Enum name="DNSStatusUnifiedCM1">2</Enum>
[Jul  1 15:49:50] <Enum name="DNSStatusUnifiedCM2">0</Enum>
[Jul  1 15:49:50] <Enum name="DNSStatusUnifiedCM3">0</Enum>
[Jul  1 15:49:50] <String name="VoiceVLAN">4095</String>
[Jul  1 15:49:50] <String name="UnifiedCMIPAddress">192.168.1.201</String>
[Jul  1 15:49:50] <String name="LocalPort">-1</String>
[Jul  1 15:49:50] <String name="TimeStamp">13908815094331390883577596</String>
[Jul  1 15:49:50] <Enum name="ReasonForOutOfService">14</Enum>
[Jul  1 15:49:50] <String name="LastProtocolEventSent">Sent:REGISTER sip:192.168.1.201 SIP/2.0 Cseq:102 REGISTER CallId:001956a8-d8620002-29fcbe28-f4337110@192.168.1.139</String>
[Jul  1 15:49:50] <String name="LastProtocolEventReceived"></String>
[Jul  1 15:49:50] </ParameterList>
[Jul  1 15:49:50] </Alarm>
[Jul  1 15:49:50] </x-cisco-alarm>
[Jul  1 15:49:50] <------------->
[Jul  1 15:49:50] --- (19 headers 30 lines) ---
[Jul  1 15:49:50] Sending to 192.168.1.139:5060 (no NAT)
[Jul  1 15:49:50] Call 001956a8-d8620002-81cb1b68-c875d648@192.168.1.139 got a SIP call transfer from caller: (REFER)!
[Jul  1 15:49:50]
[Jul  1 15:49:50] <--- Transmitting (no NAT) to 192.168.1.139:5060 --->
[Jul  1 15:49:50] SIP/2.0 603 Declined (No dialog)
[Jul  1 15:49:50] Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK04a9cc08;received=192.168.1.139
[Jul  1 15:49:50] From: <sip:001956a8d862@192.168.1.139>;tag=001956a8d8620002c4ee52a8-c4ed4788
[Jul  1 15:49:50] To: <sip:192.168.1.201>;tag=as328fc18d
[Jul  1 15:49:50] Call-ID: 001956a8-d8620002-81cb1b68-c875d648@192.168.1.139
[Jul  1 15:49:50] CSeq: 1000 REFER
[Jul  1 15:49:50] Server: Sistema-Astrum(Astrum)
[Jul  1 15:49:50] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul  1 15:49:50] Supported: replaces, timer
[Jul  1 15:49:50] Content-Length: 0
[Jul  1 15:49:50]
[Jul  1 15:49:50]
[Jul  1 15:49:50] <------------>
[Jul  1 15:49:50]
[Jul  1 15:49:50] <--- SIP read from UDP:192.168.1.139:49210 --->
[Jul  1 15:49:50] REGISTER sip:192.168.1.201 SIP/2.0
[Jul  1 15:49:50] Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK906d3f48
[Jul  1 15:49:50] From: <sip:202@192.168.1.201>;tag=001956a8d8620003f37975a8-dbccb888
[Jul  1 15:49:50] To: <sip:202@192.168.1.201>
[Jul  1 15:49:50] Call-ID: 001956a8-d8620002-afb02ce8-e0fe0ec8@192.168.1.139
[Jul  1 15:49:50] Max-Forwards: 70
[Jul  1 15:49:50] Date: Wed, 11 May 2011 13:14:35 GMT
[Jul  1 15:49:50] CSeq: 101 REGISTER
[Jul  1 15:49:50] User-Agent: Cisco-CP7970G/9.2.1
[Jul  1 15:49:50] Contact: <sip:202@192.168.1.139:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001956a8d862>";+u.sip!devicename.ccm.cisco.com="SEP001956A8D862";+u.sip!model.ccm.cisco.com="30006"
[Jul  1 15:49:50] Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
[Jul  1 15:49:50] Content-Length: 0
[Jul  1 15:49:50] Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001956A8D862 Load=SIP70.9-2-1S Last=cm-closed-tcp"
[Jul  1 15:49:50] Expires: 3600
[Jul  1 15:49:50]
[Jul  1 15:49:50] <------------->
[Jul  1 15:49:50] --- (14 headers 0 lines) ---
[Jul  1 15:49:50] Sending to 192.168.1.139:5060 (no NAT)
[Jul  1 15:49:50] Sending to 192.168.1.139:5060 (no NAT)
[Jul  1 15:49:50]
[Jul  1 15:49:50] <--- Transmitting (NAT) to 192.168.1.139:49210 --->
[Jul  1 15:49:50] SIP/2.0 401 Unauthorized
[Jul  1 15:49:50] Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK906d3f48;received=192.168.1.139;rport=49210
[Jul  1 15:49:50] From: <sip:202@192.168.1.201>;tag=001956a8d8620003f37975a8-dbccb888
[Jul  1 15:49:50] To: <sip:202@192.168.1.201>;tag=as192b7152
[Jul  1 15:49:50] Call-ID: 001956a8-d8620002-afb02ce8-e0fe0ec8@192.168.1.139
[Jul  1 15:49:50] CSeq: 101 REGISTER
[Jul  1 15:49:50] Server: Sistema-Astrum(Astrum)
[Jul  1 15:49:50] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul  1 15:49:50] Supported: replaces, timer
[Jul  1 15:49:50] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7fdbdcdf"
[Jul  1 15:49:50] Content-Length: 0
[Jul  1 15:49:50]
[Jul  1 15:49:50]
[Jul  1 15:49:50] <------------>
[Jul  1 15:49:50] Scheduling destruction of SIP dialog '001956a8-d8620002-afb02ce8-e0fe0ec8@192.168.1.139' in 32000 ms (Method: REGISTER)
[Jul  1 15:49:50] Really destroying SIP dialog '001956a8-d8620002-81cb1b68-c875d648@192.168.1.139' Method: REFER

And then it just keeps trying to register.

 

The phone's status log doesn't show anything, it just hangs displaying "registering".

 

I'm not really sure, but by looking at the Asterisk's debug output I think its trying to register using a different port than the 5060 and also doesn't send the Authorization parameters in the header. I'm saying this after looking at the debug output of a 7960 phone that does connect to the server succesfully leaving this output:

 

[Jul  2 11:44:21] <--- SIP read from UDP:192.168.1.210:5060 --->
[Jul  2 11:44:21] REGISTER sip:192.168.1.201 SIP/2.0
[Jul  2 11:44:21] Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK5b0f497b
[Jul  2 11:44:21] From: <sip:202@192.168.1.201>;tag=001563ee344300064fb22939-6b22006e
[Jul  2 11:44:21] To: <sip:202@192.168.1.201>
[Jul  2 11:44:21] Call-ID: 001563ee-34430002-039ed7b4-2645ec8b@192.168.1.141
[Jul  2 11:44:21] Max-Forwards: 70
[Jul  2 11:44:21] CSeq: 103 REGISTER
[Jul  2 11:44:21] User-Agent: Cisco-CP7960G/8.0
[Jul  2 11:44:21] Contact: <sip:202@192.168.1.210:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001563ee3443>";+u.sip!model.ccm.cisco.com="7"
[Jul  2 11:44:21] Content-Length: 0
[Jul  2 11:44:21] Expires: 3600
[Jul  2 11:44:21]
[Jul  2 11:44:21] <------------->
[Jul  2 11:44:21] --- (11 headers 0 lines) ---
[Jul  2 11:44:21] Sending to 192.168.1.210:5060 (no NAT)
[Jul  2 11:44:21] Sending to 192.168.1.210:5060 (no NAT)
[Jul  2 11:44:21]
[Jul  2 11:44:21] <--- Transmitting (NAT) to 192.168.1.210:5060 --->
[Jul  2 11:44:21] SIP/2.0 401 Unauthorized
[Jul  2 11:44:21] Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK5b0f497b;received=192.168.1.210;rport=5060
[Jul  2 11:44:21] From: <sip:202@192.168.1.201>;tag=001563ee344300064fb22939-6b22006e
[Jul  2 11:44:21] To: <sip:202@192.168.1.201>;tag=as5ff3c6d0
[Jul  2 11:44:21] Call-ID: 001563ee-34430002-039ed7b4-2645ec8b@192.168.1.141
[Jul  2 11:44:21] CSeq: 103 REGISTER
[Jul  2 11:44:21] Server: Sistema-Astrum(Astrum)
[Jul  2 11:44:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul  2 11:44:21] Supported: replaces, timer
[Jul  2 11:44:21] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75100321"
[Jul  2 11:44:21] Content-Length: 0
[Jul  2 11:44:21]
[Jul  2 11:44:21]
[Jul  2 11:44:21] <--- SIP read from UDP:192.168.1.210:5060 --->
[Jul  2 11:44:21] REGISTER sip:192.168.1.201 SIP/2.0
[Jul  2 11:44:21] Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK2b1ded48
[Jul  2 11:44:21] From: <sip:202@192.168.1.201>;tag=001563ee344300064fb22939-6b22006e
[Jul  2 11:44:21] To: <sip:202@192.168.1.201>
[Jul  2 11:44:21] Call-ID: 001563ee-34430002-039ed7b4-2645ec8b@192.168.1.141
[Jul  2 11:44:21] Max-Forwards: 70
[Jul  2 11:44:21] CSeq: 104 REGISTER
[Jul  2 11:44:21] User-Agent: Cisco-CP7960G/8.0
[Jul  2 11:44:21] Contact: <sip:202@192.168.1.210:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001563ee3443>";+u.sip!model.ccm.cisco.com="7"
[Jul  2 11:44:21] Authorization: Digest username="202",realm="asterisk",uri="sip:192.168.1.201",response="fb9fb581f3a426395565539583afc71d",nonce="75100321",algorithm=MD5
[Jul  2 11:44:21] Content-Length: 0
[Jul  2 11:44:21] Expires: 3600
[Jul  2 11:44:21]
[Jul  2 11:44:21] <------------->
[Jul  2 11:44:21] --- (12 headers 0 lines) ---
[Jul  2 11:44:21] Sending to 192.168.1.210:5060 (no NAT)
[Jul  2 11:44:21]     -- Registered SIP '202' at 192.168.1.210:5060
[Jul  2 11:44:21] Reliably Transmitting (NAT) to 192.168.1.210:5060:
[Jul  2 11:44:21] OPTIONS sip:202@192.168.1.210:5060;transport=udp SIP/2.0
[Jul  2 11:44:21] Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK7c33b421;rport
[Jul  2 11:44:21] Max-Forwards: 70
[Jul  2 11:44:21] From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as3b81ed24
[Jul  2 11:44:21] To: <sip:202@192.168.1.210:5060;transport=udp>
[Jul  2 11:44:21] Contact: <sip:asterisk@192.168.1.201:5060>
[Jul  2 11:44:21] Call-ID: 55e660f01ce458bf11c08c8a634914c6@192.168.1.201:5060
[Jul  2 11:44:21] CSeq: 102 OPTIONS
[Jul  2 11:44:21] User-Agent: Sistema-Astrum(Astrum)
[Jul  2 11:44:21] Date: Thu, 02 Jul 2020 14:44:21 GMT
[Jul  2 11:44:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul  2 11:44:21] Supported: replaces, timer
[Jul  2 11:44:21] Content-Length: 0
[Jul  2 11:44:21]
[Jul  2 11:44:21]
[Jul  2 11:44:21] ---
[Jul  2 11:44:21]        > Saved useragent "Cisco-CP7960G/8.0" for peer 202

 

Any help would be appretiated.

 

1 ACCEPTED SOLUTION

Accepted Solutions

I managed to track the error down. As you pointed out, the problem solved by disabling the nat configuration in the Asterisk configuration file.

Thanks for all the help.

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Leo Laohoo
VIP Community Legend

<natEnabled>true</natEnabled>

Change that to "false" and try again.

I've tried that, but nothing seems to change.


@Andres Sukanec wrote:

I've tried that, but nothing seems to change.


Did you factory-default the phone (so the phone can download the changed config)?

Yes, the phones came with the SCCP firmware. I've upgraded and tried this configuration on two of those phones with this results.


@Andres Sukanec wrote:
<transportLayerProtocol>1</transportLayerProtocol>

Try the option above. 

Next, after the phone has rebooted.  Look at the status message of the phone.  What does it say?

By changing the transportLayerProtocol to 1 no registration log is captured by the Asterisk server.

Also the status message screen shows the following:

Untitled.png

2nd line (from the top) means the config was downloaded by the phone and was "accepted" with no issues.
Last line states the dialplan is missing.

Yes I know that I'm not providing any dialplan configuration, but from what I recall from other cisco phone models that shouldn't interfere with the phone line registration.

Is NAT and ALG enabled on the network?
NOTE:  I've had 7970 registered to Asterisk but I've never used 9.2 as a firmware.  I used whatever latest firmware that was available.

NAT is enabled but I'm not really sure about the ALG. The phones' network SIP gateway is different from the internet one (this one beeing the main router), where should I look for this configuration?

I don't see a SIP ALG option in neather of them, but I could replace one if needed. My main router is a "Linksys WRT54G2", while the one that functions only as SIP gateway is a "Nisuta NS-WIR150N".

Talk to the provider if they support NAT &/or ALG. I suspect they don't so these two functions need to be disabled on the router as well as on the Asterisk, Extension settings.

The NAT function is supported and needed for the phones to work (the 7960 models only work with NAT enabled).

In the case of the ALG as I understand (and please correct me if I'm wrong) messes up the connection with the public network. In my case the problem is within the internal private network. The phones don't get to register with the internal Asterisk server as it is seen in the Asterisk's debug output.

I'd bet it has something to do with the phone's network configuration. Or does the registration request in the 7970 models always looks like this, totally different from the 7960's one?

7940/7960 phones are easy to integrate with Asterisk because the last known working firmware version is very stable.
The firmware for the 7970 is, however, not. I know of two version that is and 9.2 is not one of them.

Attached is known working SEPmacaddress.cnf.xml config from a working 7970. 

Take note the config is based on 9.4(2) firmware.  

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