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IP Phone 7970 not registering with Asterisk

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Andres Sukanec
Level 1
Level 1

‎07-02-2020 04:20 PM

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.

 

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Andres Sukanec
Level 1
Level 1
In response to Leo Laohoo

‎07-04-2020 03:32 PM

I couldn't find SIP firmware v9.4(2) anywhere, but I did tried with v8.4.2 which is listed as Asterisk compatible. Sadly it still does the same thing when debuged with Asterisk. I'll try checking the registration process using Wireshark.

Do you know where can I download last SIP firmware version?

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Andres Sukanec
Level 1
Level 1
In response to Andres Sukanec

‎07-05-2020 12:52 AM

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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joeyangx
Level 1
Level 1
In response to Andres Sukanec

‎10-09-2022 02:18 AM

Hi, Could you tell me how to do it ? modify sip.conf ? could you share what you did before ? Thanks

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Andres Sukanec
Level 1
Level 1
In response to joeyangx

‎10-11-2022 09:04 AM

In my case I was using FreePBX, so my solution was to disable NAT in the SIP extension configuration.

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joeyangx
Level 1
Level 1
In response to Andres Sukanec

‎10-12-2022 04:45 AM - edited ‎10-12-2022 04:57 AM

Thanks for update, BTW, is your timezone correct on left corner ? mine is wrong even I have modified SEP[MAC] as listed below, any other comment would be appreciated

 

 

    <dateTemplate>M/D/Ya</dateTemplate>
            <timeZone>China Standard/Daylight Time</timeZone>
            <ntps>
               <ntp>
                  <name>140.137.11.50</name>

 

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Andres Sukanec
Level 1
Level 1
In response to joeyangx

‎10-12-2022 05:53 AM

If you are seeing a 1 hour mismatch with the current time it may be refered to the summer time. During certain months of the year the phones enter this mode and add 1 hour to the time retrieved by the NTP server. 

I couldn't find a way to disabled this function so I just let it be. 

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