This week i changed from Trixbox to FreePBX Distro because of the asterisk 1.8 support. I have tried to migrate all settings to the FreePBX installation and much of it is working.
However i cannot get the Cisco 7960/7961 phones to register. I have copied the tftpboot files from Trixbox to FreePBX so the should have worked but the phone wil nog register. After this i have tried to make new configurations using PBX End Point Manager 2.9.1.2 by Andrew Nagy but the problem still occurs
When i make a tcpdump i get the following information
tcpdump -i eth1 -n -s 0 port 5060 -v and host 192.168.110.118
16:25:00.508166 IP (tos 0x60, ttl 62, id 37873, offset 0, flags [none], proto: UDP (17), length: 726) 192.168.110.118.50700 > 192.168.120.222.sip: SIP, length: 698
REGISTER sip:192.168.120.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.118:5060;branch=z9hG4bK51a07556
From: <sip:[email protected]>;tag=001ebe9059fc003252f4157d-5b8238a0
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 17 Dec 2009 11:22:30 GMT
CSeq: 149 REGISTER
User-Agent: Cisco-CP7961G/8.5.3
Contact: <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001ebe9059fc>";+u.sip!model.ccm.cisco.com="30018"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP001EBE9059FC Load=SIP41.8-5-4S Last=phone-keypad"
Expires: 3600
16:25:00.508491 IP (tos 0x60, ttl 64, id 24300, offset 0, flags [none], proto: UDP (17), length: 478) 192.168.120.222.sip > 192.168.110.118.50700: SIP, length: 450
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.120.222:5060;branch=z9hG4bK51a07556;received=192.168.110.118
From: <sip:[email protected]>;tag=001ebe9059fc003252f4157d-5b8238a0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 149 REGISTER
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
16:25:00.508643 IP (tos 0x60, ttl 64, id 24301, offset 0, flags [none], proto: UDP (17), length: 575) 192.168.120.222.sip > 192.168.110.118.50700: SIP, length: 547
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.120.222:5060;branch=z9hG4bK51a07556;received=192.168.110.118
From: <sip:[email protected]>;tag=001ebe9059fc003252f4157d-5b8238a0
To: <sip:[email protected]>;tag=as05cf9a01
Call-ID: [email protected]
CSeq: 149 REGISTER
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a31d0d9"
Content-Length: 0
sip_additional.conf
[89]
deny=0.0.0.0/0.0.0.0
secret=45fgsdaas
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=never
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/89
mailbox=89@device
permit=0.0.0.0/0.0.0.0
callerid=device <89>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
/tftpboot/SEP
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Central Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>0.pool.ntp.org</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>192.168.120.222</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP41.8-5-4S</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>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>0</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>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>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>true</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>false</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>x89</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>89</featureLabel>
<name>89</name>
<displayName>Cisco7961</displayName>
<contact>89</contact>
<proxy>192.168.120.222</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>89</authName>
<authPassword>45fgsdaas</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>
