QUESTION 7970 "401 Unauthorized" - Phone not registering

Yoav Zuri

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Hi,

First of all I'd like to thank this community, a lot of useful information.
I know that 7970 is considerably problematic phone w/ PIAF, but I got one and I really want to get it working.

It took some time to get it to work. Tried 20 XML files combination till I managed to make one that the phone agrees to load.

Cisco 7970 is accessing the PIAF server over internet. NAT (Fortigate 60D, SIP ALG disabled).

Cisco 7970 LAN IP: 10.123.123.111 / WAN: 212.xx.xx.106
PIAF server: 192.xx.xx.249 (WAN) - No NAT / FW
Firmware: 8.4.2S

Asterisk 13.8.2


Asterisk sip debug output:
Code:
<--- SIP read from UDP:212.xx.xx.106:50362 --->
REGISTER sip:192.xx.xx.249 SIP/2.0
Via: SIP/2.0/UDP 212.xx.xx.106:5060;branch=z9hG4bKc170c255
From: <sip:[email protected]>;tag=0013190237b10016524fa649-874887dd
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 14 Nov 2008 00:02:45 GMT
CSeq: 122 REGISTER
User-Agent: Cisco-CP7970G/8.4.0
Contact: <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0013190237b1>";+u.sip!model.ccm.cisco.com="30006"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP0013190237B1 Load=SIP70.8-4-2S Last=phone-keypad"
Expires: 3600

<------------->
--- (14 headers 0 lines) ---


Sending to 212.xx.xx.106:5060 (no NAT)

<--- Transmitting (no NAT) to 212.xx.xx.106:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 212.xx.xx.106:5060;branch=z9hG4bKc170c255;received=212.xx.xx.106
From: <sip:[email protected]>;tag=0013190237b10016524fa649-874887dd
To: <sip:[email protected]>;tag=as618eaa1b
Call-ID: [email protected]
CSeq: 122 REGISTER
Server: FPBX-12.0.70(13.8.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="294a4edb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
Extension:
Code:
[202]
deny=0.0.0.0/0.0.0.0
secret=xxxxxxxxxxxxxxxxx
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/202
permit=0.0.0.0/0.0.0.0
callerid=Cisco 7970 <202>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
Any help will be highly appreciated.

Thanks!
 

Yoav Zuri

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Joined
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SEP0013190237B1.cnf.xml:
Code:
<device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>admin</sshUserId>
    <sshPassword>password</sshPassword>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D/M/YA</dateTemplate>
            <timeZone>Eastern Standard/Daylight Time</timeZone>
            <ntps>
                <ntp>
                    <name>213.251.128.249</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <members>
                <member priority="0">
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5060</sipPort>
                        </ports>
                        <processNodeName>192.xx.xx.249</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
    </devicePool>
    <sipProfile>
        <natEnabled>true</natEnabled>
        <natAddress>212.xx.xx.106</natAddress>
        <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>Black IT LTD</phoneLabel>
        <stutterMsgWaiting>1</stutterMsgWaiting>
        <callStats>false</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
        <sipLines>
            <line button="1">
                <featureID>9</featureID>
                <featureLabel>Yoav</featureLabel>
                <proxy>192.xx.xx.249</proxy>
                <port>5060</port>
                <name>202</name>
                <displayName>Yoav</displayName>
                <autoAnswer>
                    <autoAnswerEnabled>2</autoAnswerEnabled>
                </autoAnswer>
                <callWaiting>3</callWaiting>
                <authName>202</authName>
                <authPassword>xxxxxxxxxxxxxxxxxx</authPassword>
                <sharedLine>false</sharedLine>
                <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
                <messagesNumber>*99</messagesNumber>
                <ringSettingIdle>4</ringSettingIdle>
                <ringSettingActive>5</ringSettingActive>
                <contact>202</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://192.xx.xx.249/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>SIP70.8-4-2S</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://192.xx.xx.249/services/authenticate.php</authenticationURL>
    <directoryURL>http://192.xx.xx.249/services/directory.php</directoryURL>
    <servicesURL>http://192.xx.xx.249/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>
 

Yoav Zuri

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Update: when using the same installation w/ LAN installation - phone registers immediately.
Not problem solved yet, but at least I know my configuration is somewhat correct.
 

Neil Cudmore

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Jan 9, 2015
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Just a thought, while trying to get some Cisco 7960's working, I found that if you stick with the default random password, it appears to be too long for the Cisco phones. I trimmed last 6 characters from the system randomly created password, and suddenly I found phones connecting.

On the other hand, for a couple Cisco phones I found that doing a copy and paste from one template which was working, still didn't work, and I ended up deleting the user and creating again from scratch... No idea of why this then worked but I was doing laods of phones in a short time and just went with it, and didn't have time to work out the whys and wherefores.....
 

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