Cisco 7945g can't register

Asterisk Asterisk SIP
1

Hello, I am using FreePBX 14.0.1.24, Running Asterisk Asterisk 13.18.4
I get these logs

[2018-04-26 21:56:09] DEBUG[1972]: res_pjsip/pjsip_distributor.c:383 find_dialog: Could not find matching transaction for Request msg REGISTER/cseq=114 (rdata0x26714f8)
[2018-04-26 21:56:09] DEBUG[1972]: res_pjsip/pjsip_distributor.c:461 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-0000002d to use for Request msg REGISTER/cseq=114 (rdata0x26714f8)
[2018-04-26 21:56:09] DEBUG[8078]: res_pjsip_endpoint_identifier_ip.c:197 ip_identify_match_check: Source address 172.16.30.33:49165 does not match identify '501-identify'
[2018-04-26 21:56:09] DEBUG[8078]: res_pjsip_endpoint_identifier_ip.c:222 ip_identify: Identify checks by IP address failed to find match: '172.16.30.33:49165' did not match any identify section rules
[2018-04-26 21:56:09] DEBUG[8078]: res_pjsip_endpoint_identifier_user.c:133 username_identify: Attempting identify by From username '501' domain '192.168.4.64'
[2018-04-26 21:56:09] DEBUG[8078]: res_pjsip_endpoint_identifier_user.c:145 username_identify: Identified by From username '501' domain '192.168.4.64'
[2018-04-26 21:56:09] DEBUG[8078]: res_pjsip_authenticator_digest.c:454 digest_check_auth: Using default realm 'asterisk' on incoming auth '501-auth'.
[2018-04-26 21:56:10] DEBUG[10012]: manager.c:6383 process_message: Running action 'Login'
[2018-04-26 21:56:11] DEBUG[10019]: manager.c:6383 process_message: Running action 'Login'
[2018-04-26 21:56:12] DEBUG[10026]: manager.c:6383 process_message: Running action 'Login'
[2018-04-26 21:56:13] DEBUG[10033]: manager.c:6383 process_message: Running action 'Login'

Over and over :frowning:

I am using SIP45.8-4-2S right now but I’ve also tried using SIP45.8-5-4S and SIP45.9-4-2SR3-1S with no luck.

Pasting my config files just in case.

<Default> 
<callManagerGroup> 
    <members>  
       <member priority="0">  
          <callManager>  
             <ports>  
                <ethernetPhonePort>2000</ethernetPhonePort>  
                <mgcpPorts>  
                   <listen>2427</listen>  
                   <keepAlive>2428</keepAlive>  
                </mgcpPorts>  
             </ports>  
             <processNodeName></processNodeName>  
          </callManager>  
       </member>  
    </members>  
 </callManagerGroup>  
<loadInformation8 model="IP Phone 7940">P0S3-8-12-00</loadInformation8>  
<loadInformation7 model="IP Phone 7960">P0S3-8-12-00</loadInformation7> 
<loadInformation435 model="Cisco 7945">SIP45.8-4-2S</loadInformation435>
<loadInformation436 model="Cisco 7965">SIP45.8-4-2S</loadInformation436> 
<loadInformation30006 model="IP Phone 7970">SIP70.8-0-3S</loadInformation30006> 
<authenticationURL></authenticationURL>  
<directoryURL></directoryURL>  
<idleURL></idleURL>  
<informationURL></informationURL>  
<messagesURL></messagesURL>  
<servicesURL></servicesURL>  
</Default>

<device> 
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId> 
<sshPassword>password</sshPassword> 
<devicePool> 
   <dateTimeSetting> 
      <dateTemplate>D/M/YA</dateTemplate> 
      <timeZone>Central Standard/Daylight Time</timeZone> 
      <ntps> 
         <ntp> 
            <name>192.168.4.64</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>192.168.4.64</processNodeName> 
            </callManager> 
         </member> 
      </members> 
   </callManagerGroup> 
</devicePool> 
<sipProfile> 
   <sipProxies> 
      <backupProxy></backupProxy> 
      <backupProxyPort>5060</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> 
   <dtmfAvtPayload>101</dtmfAvtPayload> 
   <dtmfDbLevel>3</dtmfDbLevel> 
   <dtmfOutofBand>avt</dtmfOutofBand> 
   <alwaysUsePrimeLine>false</alwaysUsePrimeLine> 
   <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> 
   <kpml>3</kpml> 
   <phoneLabel>Aynitech</phoneLabel> 
   <stutterMsgWaiting>1</stutterMsgWaiting> 
   <callStats>false</callStats> 
   <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
   <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> 
   <sipLines> 
      <line button="1"> 
         <featureID>9</featureID> 
         <featureLabel>Username 501</featureLabel> 
         <proxy>192.168.4.64</proxy> 
         <port>5060</port> 
         <name>501</name> 
         <displayName>501</displayName> 
         <autoAnswer> 
            <autoAnswerEnabled>2</autoAnswerEnabled> 
         </autoAnswer> 
         <callWaiting>3</callWaiting> 
         <authName>501-auth</authName> 
         <authPassword></authPassword> 
         <sharedLine>false</sharedLine> 
         <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
         <messagesNumber>*99</messagesNumber> 
         <ringSettingIdle>4</ringSettingIdle> 
         <ringSettingActive>5</ringSettingActive> 
         <contact>501</contact> 
         <forwardCallInfoDisplay> 
            <callerName>true</callerName> 
            <callerNumber>false</callerNumber> 
            <redirectedNumber>false</redirectedNumber> 
            <dialedNumber>true</dialedNumber> 
         </forwardCallInfoDisplay> 
      </line> 
      <line button="2"> 
         <featureID>20</featureID> 
         <featureLabel>Menu</featureLabel> 
         <serviceURI>http://example.domain.ext/services/menu.xml</serviceURI> 
      </line> 
   </sipLines> 
   <voipControlPort>5060</voipControlPort> 
   <startMediaPort>16348</startMediaPort> 
   <stopMediaPort>20134</stopMediaPort> 
   <dscpForAudio>184</dscpForAudio> 
   <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> 
   <dialTemplate>dialplan.xml</dialTemplate> 
   <softKeyFile></softKeyFile> 
</sipProfile> 
<commonProfile> 
   <phonePassword></phonePassword> 
   <backgroundImageAccess>true</backgroundImageAccess> 
   <callLogBlfEnabled>2</callLogBlfEnabled> 
</commonProfile> 
<loadInformation>SIP45.8-4-2S</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> 
   <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive> 
   <displayOnTime>00:00</displayOnTime> 
   <displayOnDuration>00:00</displayOnDuration> 
   <displayIdleTimeout>00:00</displayIdleTimeout> 
   <spanToPCPort>1</spanToPCPort> 
   <loggingDisplay>1</loggingDisplay> 
   <loadServer></loadServer> 
</vendorConfig> 
<userLocale> 
   <name></name> 
   <uid></uid> 
   <langCode>en_US</langCode> 
   <version>1.0.0.0-1</version> 
   <winCharSet>iso-8859-1</winCharSet> 
</userLocale> 
<networkLocale></networkLocale> 
<networkLocaleInfo> 
   <name></name> 
   <uid></uid> 
   <version>1.0.0.0-1</version> 
</networkLocaleInfo>    
<deviceSecurityMode>1</deviceSecurityMode> 
<authenticationURL>http://example.domain.ext/services/authenticate.php</authenticationURL> 
<directoryURL>http://example.domain.ext/services/directory.php</directoryURL> 
<servicesURL>http://example.domain.ext/services/menu.xml</servicesURL> 
<idleURL></idleURL> 
<informationURL></informationURL> 
<messagesURL></messagesURL> 
<proxyServerURL></proxyServerURL> 
<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> 
</device>

Please advice

2

The logs you’ve posted are debug messages and are harmless, nothing is out of the ordinary in them. What you likely need to do is ensure “force_rport” is set to “no”. How that is done in FreePBX I do not know.

1 Like
3

Sorry forgot to mention that the phone keeps Trying to register without success,

<--- Received SIP request (680 bytes) from UDP:172.16.30.33:49166 --->
REGISTER sip:192.168.4.64 SIP/2.0
Via: SIP/2.0/UDP 172.16.30.33:5060;branch=z9hG4bK7704f20b
From: <sip:501@192.168.4.64>;tag=00083030a7910004ded00557-31f8a2fb
To: <sip:501@192.168.4.64>
Call-ID: 00083030-a7910002-9cffa498-4154d030@172.16.30.33
Max-Forwards: 70
Date: Thu, 13 Nov 2008 22:42:27 GMT
CSeq: 103 REGISTER
User-Agent: Cisco-CP7945G/8.5.3
Contact: <sip:501@172.16.30.33:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00083030a791>";+u                                                                    .sip!model.ccm.cisco.com="435"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP00083030A791 Load=SIP45.8-5-4S Last=initialized"
Expires: 3600


[2018-04-26 22:29:53] DEBUG[1972]: res_pjsip/pjsip_distributor.c:383 find_dialog: Could not find matching transaction                                                                     for Request msg REGISTER/cseq=103 (rdata0x26714f8)
[2018-04-26 22:29:53] DEBUG[1972]: res_pjsip/pjsip_distributor.c:461 ast_sip_get_distributor_serializer: Calculated s                                                                    erializer pjsip/distributor-0000003c to use for Request msg REGISTER/cseq=103 (rdata0x26714f8)
[2018-04-26 22:29:53] DEBUG[25915]: res_pjsip_endpoint_identifier_ip.c:197 ip_identify_match_check: Source address 17                                                                    2.16.30.33:49166 does not match identify '501-identify'
[2018-04-26 22:29:53] DEBUG[25915]: res_pjsip_endpoint_identifier_ip.c:222 ip_identify: Identify checks by IP address                                                                     failed to find match: '172.16.30.33:49166' did not match any identify section rules
[2018-04-26 22:29:53] DEBUG[25915]: res_pjsip_endpoint_identifier_user.c:133 username_identify: Attempting identify b                                                                    y From username '501' domain '192.168.4.64'
[2018-04-26 22:29:53] DEBUG[25915]: res_pjsip_endpoint_identifier_user.c:145 username_identify: Identified by From us                                                                    ername '501' domain '192.168.4.64'
[2018-04-26 22:29:53] DEBUG[25915]: res_pjsip_authenticator_digest.c:454 digest_check_auth: Using default realm 'aste                                                                    risk' on incoming auth '501-auth'.
<--- Transmitting SIP response (516 bytes) to UDP:172.16.30.33:49166 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.30.33:5060;rport=49166;received=172.16.30.33;branch=z9hG4bK7704f20b
Call-ID: 00083030-a7910002-9cffa498-4154d030@172.16.30.33
From: <sip:501@192.168.4.64>;tag=00083030a7910004ded00557-31f8a2fb
To: <sip:501@192.168.4.64>;tag=z9hG4bK7704f20b
CSeq: 103 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1524781793/09c7360efac8cca6110592f171de7bf3",opaque="2137925b3a7e98                                                                    af",algorithm=md5,qop="auth"
Server: FPBX-14.0.1.24(13.18.4)
Content-Length:  0

I added pjsip set logger on just in case,
EDIT: I just changed it to SIP45.8-5-4S

4

You need to have “force_rport” set to “no” as I stated.

1 Like
5

I found a way to include it adding the below to pjsip.endpoint_custom

[233](+) 
force_rport=no

But didnt work

However I also tried adding it directly in pjsip.endpoint.conf but didnt work. keeps registering

Any other advice or direction to go?

I appreciate your help

Btw I cant see logs now they are not showing up after issuing asterisk -rvvvv or rgggg

There is ping between asterisk and the phone

6

Nvm I got this


<--- Received SIP request (681 bytes) from UDP:172.16.30.33:49352 --->
REGISTER sip:192.168.4.64 SIP/2.0
Via: SIP/2.0/UDP 172.16.30.33:5060;branch=z9hG4bK4b87c5be
From: <sip:501@192.168.4.64>;tag=00083030a791000d814e7b95-7868b668
To: <sip:501@192.168.4.64>
Call-ID: 00083030-a7910002-5b59bfe0-5f27cc9e@172.16.30.33
Max-Forwards: 70
Date: Thu, 13 Nov 2008 23:32:05 GMT
CSeq: 112 REGISTER
User-Agent: Cisco-CP7945G/8.5.3
Contact: <sip:501@172.16.30.33:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00083030a791>";+u.sip!model.ccm.cisco.com="435"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00083030A791 Load=SIP45.8-5-4S Last=phone-keypad"
Expires: 3600


<--- Transmitting SIP response (516 bytes) to UDP:172.16.30.33:49352 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.30.33:5060;rport=49352;received=172.16.30.33;branch=z9hG4bK4b87c5be
Call-ID: 00083030-a7910002-5b59bfe0-5f27cc9e@172.16.30.33
From: <sip:501@192.168.4.64>;tag=00083030a791000d814e7b95-7868b668
To: <sip:501@192.168.4.64>;tag=z9hG4bK4b87c5be
CSeq: 112 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1524784772/6f225fb9fbee58e78b6024e3adf2eeee",opaque="36461e262cbfc3c8",algorithm=md5,qop="auth"
Server: FPBX-14.0.1.24(13.19.1)
Content-Length:  0

Still Registering…

7

The change did not work. If it had then the response would have gone to port 5060.

1 Like
8

This fixed the issue, Thank you very much!

9

Cisco phones will only register if you enable TCP signalling. Your welcome :slight_smile:

10

Thanks! Any clue how to do it from Asterisk or Freepbx? I am digging around but I cant find anything in google

11