Register TANDBERG MXP 6000 over SIP

Hi, i have MXP6000 with 9.1 software. Cant make it register with SIP. No single packet comes from MXP to server.
Has anyone been able to make it register with SIP server?
Config is quite simple:
xConfiguration Conference SIP URI: "[email protected]"
*c xConfiguration SIP Mode: On
*c xConfiguration SIP Server Discovery: Manual
*c xConfiguration SIP Server Address: "10.96.37.10"
*c xConfiguration SIP Server Type: Auto
*c xConfiguration SIP Authentication UserName: "6000"
*c xConfiguration SIP Transport Default: UDP
*c xConfiguration SIP TLS Verify: Off
*c xConfiguration SIP ICE Mode: Off
*c xConfiguration SIP MNS Mode: Off
*c xConfiguration SIP ForceTurn Mode: Off
*c xConfiguration SIP DefaultCandidate Type: Host
*c xConfiguration SIP Legacy Mask: ""
*c xConfiguration SIP ReplyTo URI: ""

Does it matter? NO REGISTER packets arrived to server, i was sniffing traffic.
Problem solved just after i entered valid DNS server address in IP parameters. Why would it need DNS if i'm using direct IP addresses...
Anyways, my SIP server (Asterisk) does not support duo-video and because there is two video streams in SDP message, it choses wrong RTP port and streams other's side video to presentation channel.

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Does it matter? NO REGISTER packets arrived to server, i was sniffing traffic.
Problem solved just after i entered valid DNS server address in IP parameters. Why would it need DNS if i'm using direct IP addresses...
Anyways, my SIP server (Asterisk) does not support duo-video and because there is two video streams in SDP message, it choses wrong RTP port and streams other's side video to presentation channel.

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You can download it from either of the two locations:
Cisco website
TANDBERG FTP
Also, if you're upgrading from F8 or below, you can contact TAC and point them to this security advisory, with it, you can get the MXP F9.3.1 software for free without the need of a service contract.
You will need a release key if you're going from one major version to another, ex: F8 or below to F9. Minor versions don't require a release key, ex: F9.1 to F9.3.1.
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I have to 3 Question
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3. and do I need a license Key for it.

You can download it from either of the two locations:
Cisco website
TANDBERG FTP
Also, if you're upgrading from F8 or below, you can contact TAC and point them to this security advisory, with it, you can get the MXP F9.3.1 software for free without the need of a service contract.
You will need a release key if you're going from one major version to another, ex: F8 or below to F9. Minor versions don't require a release key, ex: F9.1 to F9.3.1.
Instructions on how to upgrade an MXP codec can be found in the Admin Guide, on pg 141.

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Hi All,
Does anybody have any pointers on getting a CUCM 7.1 cluster talking to a Tandberg MXP 3000 via SIP.
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you can test it before you remove the existing PSTN lines ( recommended )
good luck

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CM Register over SIP Trunk

Hi guys,
would it be possible to allow sip users to register over the sip trunk on the Call Manager? or is this method not allowed?
Thanks.
Best regards

Hi Manish,
between sip client and webrtc gw -> ws and between webrtc gw and CM -> sip.
here are the sip messages.
both phones are registered, 9000 is a 7912 and 8080 is sip.
192.168.15.2 - CM
192.168.15.202 - webrtc
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>
Contact: <sip:[email protected]:10060;ws-src-ip=192.168.251.105;ws-src-port=50731;ws-src-proto=ws;transport=udp>
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Content-Type: application/sdp
Content-Length: 978
Max-Forwards: 70
Authorization: Digest username="8080",realm="ccmsipline",nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6",uri="sip:[email protected]",response="352cb2e17e36b32ee4e0d52443d0a106",algorithm=MD5
User-Agent: webrtc2sip Media Server 2.6.0
v=0
o=doubango 1983 678901 IN IP4 192.168.15.202
s=-
c=IN IP4 192.168.15.202
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 58690 RTP/AVP 8 0 101
c=IN IP4 192.168.15.202
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1YfBfgbhIdMB6YVtyZgJqc77QPHwm9o42aEPbkHD
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:fujGVOi70hQnKkeUimcFUw2bH3ajZ2iW0xKy5Nrw
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:4034073057 cname:c08c56217e96dbc1e8234373eb5d2fcc
a=ssrc:4034073057 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:4034073057 label:doubango@audio
a=ice-ufrag:uaektHZ6KFVn1fw
a=ice-pwd:HAj21nuOrDmIKl3ANXTc3K
a=candidate:tWR5PLw1x 1 udp 2130706431 192.168.15.202 58690 typ host
a=candidate:tWR5PLw1x 2 udp 2130706430 192.168.15.202 58691 typ host
RECV:SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>
Date: Wed, 19 Mar 2014 13:26:05 GMT
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Allow-Events: presence
Content-Length: 0
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>;tag=856401750
Date: Wed, 19 Mar 2014 13:26:05 GMT
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Allow-Events: presence
WWW-Authenticate: Digest realm="ccmsipline", nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6", algorithm=MD5
Content-Length: 0
SEND: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>;tag=856401750
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 ACK
Content-Length: 0
Max-Forwards: 70
Receiving SIP o/ WebSocket message: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKZEk81zTwfVde8oImts6ZHiTzchfBWh1N;rport
From: "8080"<sip:[email protected]>;tag=XFKqC4zu0S9QfzzMzQ4u
To: <sip:[email protected]>;tag=1464334432
Call-ID: ecc84fa2-3de3-d953-527f-5e7515cabca3
CSeq: 29519 ACK
Content-Length: 0
Route: <sip:192.168.15.2:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Thanks.

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Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//Per

Thanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
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Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per.

MXP's and SIP Registration Failure Using a FQDN

A majority of our MXP's will not register to our SIP Server using a FQDN. They will register using the SIP Server IP Address. Why do they not like the FQDN? Our C Series, EX, and SX systems do not have this problem.

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Cisco 7962G phone is not gettting register or send any traffic sip to register !!

Hi Guys i have cisco 7962 G phone that is not going to get register !
i have the conf working on asteriks when asterisk in the same subnet.
but with this phone its not registering to destination outside the network !
im sure no firewall and it can reach 8.8.8.8 and dns okay and flashed correctly with sip firmware .
=========
so i will paste here the config files :
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>xxx.0.220</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation434 model="Cisco 7942">SIP42.9-2-1S</loadInformation434>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
and here is sepmac file :
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.Y</dateTemplate>
<timeZone>W. Europe Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>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>192.168.0.220</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-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>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<phoneLabel>103</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>103</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>103</name>
<displayName>103</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>103</authName>
<authPassword>abc123</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>103</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword>1234</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.9-2-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>1</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp></versionStamp>
<userLocale>
<name>English_United_Kingdom</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>Germany</networkLocale>
<networkLocaleInfo>
<name>Germany</name>
</networkLocaleInfo>
<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>
</device>
as u see the ip is private
if i set it to external pbx , the phone dont register
how can u help me with enable loggin to debug on dat phone ??
cheers

phone has public ip and didnt work !
we use softphone and get register but the cisco .....not working !!
about my problem
u can see here :
http://www.voip-info.org/boards/view/topic/38915
it has no solution so far .
i do debug and tcpdump on the pbx and i cant see any incoming resgteration trials.
cheers

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Hello,
I have an TANDBERG MXP550 configured as an H.323 client in call manager 4.2 without a gatekeeper.
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Call manager provides the called party with the extension or directory number of the calling party on a display. You can use the Calling Line ID Presentation field in the Gateway Configuration window to control whether the CLID displays for all outgoing calls on the gateway.Refer URL
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008070e48b.html#wp1051056

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I am a consultant doing an install of CUCM BE 6K for a company with the help of a CCNP
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Register TANDBERG MXP 6000 over SIP

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