ITL Issue with CUCM 8.6(2a)SU3

Hi, We have a cluster of 6 nodes pub, 2 standalone ccm, 2 (ccm+tftp) and 1MoH server.
when we move a phone from one device pool to another, the phone displays "Registration Rejected:Security Error"
I tried deleting the ITL files on the phones but still no luck. with 'show itl' command, we see that the 'System Administrator Security Token' and the 'TFTP' function ITL Record parameters twice with two different certificates in  all the nodes.
On another working cluster, the ITL Records of System Administrator Security Token' and the 'TFTP' function can be seen only once.
any idea why this difference? The change in CUCM happened before this issue was that we changed the DSCP enterprise parameter and reset all the phones and restarted the CCM service on all the 4 nodes.

I would start by restarting the TVS and TFTP services on the servers within the cluster.
Thanks,
Tony
Please rate helpful posts!

Similar Messages

  • Compatibility Uccx 9.0(2)SU2 with CUCM 9.1(2)SU3

    Have seen the recent discussion on UCCX 9.0(2)SU2 and CUCM 9.1(2)SU2 here:
    https://supportforums.cisco.com/discussion/12482146/compatibility-uccx-902su2-cucm-912su2
    But with the release of CUCM 9.1(2)SU3 looking at the same compatibility matrix:
    http://docwiki.cisco.com/wiki/Unified_CCX_Software_Compatibility_Matrix_for_9.0(2)_SU2
    I do not see that UCCX 9.0(2)SU2 and CUCM 9.1(2)SU3 are compatible.  Is this an error in the document or do we need to upgrade UCCX if we upgrade CUCM from 9.1(1) to 9.1(2)SU3?
    Thanks,
    Ryan

    Hi Ryan-
    Take a look the posts by Arundeep Nagaraj in this thread:  https://supportforums.cisco.com/discussion/12439306/ucm-uccx-compatibiity
    Thanks,
    DJ

  • Cisco Unified Communications Manager Attendant Console issue with cucm ver 9.1.1

    Dear support,
    As we used before Cisco Unified Communications Manager Attendant Console with CUCM version 7.x as now we upgrade to version 9.1.1 and we could not be able to use this service .due to attendant console is not available with cucm versio 9.1.1.
    before upgrade cucm ver 7.x  from PC user maintain all contact details store with Attendant console software on his desktop PC.  right now its give error unable to connect to the server. due to attendant console is not available with cucm version 9.1.1.
    So how i can retrieve the user store contact details from attendant console software from his PC desktop .
    so please provide you feedback highly appreciate.
    Regards
    syed

    Dear support,
    As i check this on user  pc c:\program files\cisco\Unified Communications Manager Attendant Console\data\GlobalSettings.xml
    I couldn't  be able to retrieve the store contact data .most of the user are get effected by this service. 
    As we do upgrade to cucm versioun 9.1.1 to use of more services but not user are to be get effect by the attendant  services like we can't use Attendant console software on CUCM 9.1.1 after upgrade......
    so please advice ........... atleast we need to retrieve the user store contact data ...
    Highly appreciate for your resposne .....
    regards
    syed

  • Voice mail issue with CUCM or CUCME

    Hi,
    I want to know a basic information regarding voice mail configuration in CUCM or CUCME.
    With the very basic configuration of CUCM or CUCME without any unity or unity connection, is it possible to have basic voice mail features with the system. Say, I have configured call manager server with voice gateway connected to PSTN. Now is it possible to have voice mail for the system without cisco unity or unity connection configured.
    Same goes for CUCME. We have CME bundled router configured. Can we configure voice mail with unity express without having separate license?
    Please help ,e to get the answer for this query.
    Regards,
    Sagar

    Hi Amer,
    Thanks for your help. Is this same for both CUCM & CUCME?
    For welcome greetings, I have configured its-CISCO.2.0.2.0.tcl in the VG. Can you please guide me about basicAA.tcl script, i want to give it a shot.
    Regards,
    Sagar

  • Error message when syncing CUACEE 9.0.1.2 with CUCM 9.1.1

    Hi all,
    When I set up the CTI template, I chose owner userid as "none" but then when syncing the AC, I get the message on the sync report for every device of "Error -206 The specified table (owneruserid) is not in the database."
    If I set the template to use a specific owner userid and resync, then it works correctly, but I don't want these CTI ports associated with a userid.
    Is there any reason why this would be happening, as I have been able to do it in the past with "none" on 8.6

    Sean,
    Yes, this is a known defect:  CSCue69477 - CTI Route Port Synch Issue with CUCM.  As per the bug: 
    "This issue will occur if the "SUBSCRIBE Calling Search Space" under Protocol Specific Information in the Template for the CTI Route Ports is configured. To work around this, the "Owner User ID" under Device Information for the Template needs to be configured."
    Regards,
    Jason

  • Polycom phones with CUCM registration issues

    I have Polycom IP5000 phones running UC software 4.1.1 which do not stay registered with CUCM 9.1 for more than couple of seconds, I created digest user with username matching the DN of the phone, the phone registers for few seconds but then unregisters and stays unregistered for long period of time.  Does anyone have a good guide or tips on how to make these phones work?
    Polycom logs are not very helpful as all I see is this:
    000118.316|so   |*|03|Network initialized. Starting network tasks.
    000118.316|ice  |5|03|Network ICE stack failed to initialize in 5000 ms
    000118.316|log  |*|03|Install file upload callback for 'so'
    000118.346|sip  |*|03|Sip Register Usr:01026002 Dsp:01026002 Auth:'01026002' Inx:0
    000118.354|sip  |*|03|Fast Boot Measurement Point: Ready for Call, uptime: 78.354 sec.
    000118.356|app1 |4|03|[AppHybridC::procCfgParamChange] unexpected line index=(-1)
    000118.440|app1 |*|03|Ctx [0] Registered [true]
    000119.820|cfg  |4|03|Prov|Download of master configuration file failed
    000119.820|cfg  |4|03|Prov|Trying to boot from existing configuration
    000120.798|cfg  |*|03|Prov|Finished updating configuration
    0630094436|cfg  |4|03|Prov|Download of master configuration file failed
    0630094436|cfg  |4|03|Prov|Trying to boot from existing configuration
    0630094436|cfg  |*|03|Prov|Finished updating configuration
    0630094436|log  |4|03|UtilLogC::uploadFifoLog: upload error. protocol 0 result = -1
    0630094436|log  |4|03|UtilLogC::uploadFifoLog: upload error. protocol 0 result = -1
    0630094436|log  |4|03|UtilLogC: Upload succeeded. (ConsecutiveFailures 1 Total 1 MsgSendErr 0)
    0630094438|utilm|4|03|uBLFCompressed: File /ffs0/local/local-directory_xml.zzz does not exist or is empty
    0630094441|utilm|4|03|uBLFCompressed: File /ffs0/local/local-directory_xml.zzz does not exist or is empty
    Chris

    000119.820|cfg  |4|03|Prov|Download of master configuration file failed
    it looks like you do not have configuration file for this phone
    check is it in default device profiles?

  • Unity Connection 8.5.1 MWI issue with French Locale

    Hi There,
    I am experiencing issues with MWI while using french locale on CUC 8.5.1. While, it works perfectly fine with English (US).
    CUCM version is 8.0.3. Any ideas will be much appreciated.
    Regards,
    Sami

    Hi Armin,
    You could be seeing this behavior due to this bug;
    Incorrect locale version cause VM outage
    CSCtq97240
    Symptom:
    Unity connection 8.5.1 installed with locales may experience the following issues
    1.When users try to leave message ,they get the fail safe message
    2.Sometimes they are able to leave message , message doesnt get delivered and no MWI
    Conditions:
    When Locale version ES 24 is installed on UC 8.5(1) [ ES 16] or less
    Workaround:
    please downgrade the locale version to ES 16  if you have 8.5(1)SU1 or use US English language.
    Though the Release notes of 8.5(1)
    http://www.cisco.com/web/software/282074314/49311/851su1cucrm.pdf
    clearly  mentions about the compatibility , Many customers think its good for  them to install the latest locale ES and hence end with this condition.
    This is the compatibility matrix for locales
    ========================================
    a) No locales are available for Connection 8.5(1) base version only.
    b) Locale SU1 is supported with Connection 8.5(1)SU1.
    c) Locale ES 24 is supported with Connection 8.5(1)ES 17 to ES 36 only.
    d) Locale ES 24 is not tested for Connection 8.5(1)ES 37 to ES46.
    e) Locale 8.5(1)SU2 is supported with Connection 8.5(1)SU2 only. Not all locales available as of now
    f)  Locale 8.5(1)SU3 is supported with Connection 8.5(1)SU3 only. Only Japanese is available as of now.
    g)  Locale 8.5(1)ES72 is supported with Connection 8.5(1)ES72 only. Only Brazilian-Portuguese is available as of now.
    Cheers!
    Rob
    "Your life is worth much more than gold." 
    - Bob Marley

  • Issue with instant ringback when using sip trunk to SP

    Hi all,
    We use CUCM 8.0.2.
    We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
    Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
    c2900-universalk9-mz.SPA.150-1.M3.bin
    Cisco CISCO2911/K9 (revision 1.0)
    Technology Package License Information for Module:'c2900'
    Technology Technology-package
                      Current       Type
    ipbase        ipbasek9      Permanent
    security      securityk9    Permanent
    uc              uck9            Permanent
    data           None            None
    We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
    We use 7945 and CIPC for our phones.
    We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
    Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
    Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
    Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
    Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
    Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
    Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
    Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
    Any ideas why this happens and how to stop it?
    I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
    Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
    voice service voip
    address-hiding
    mode border-element
    allow-connections sip to sip
    sip
      bind control source-interface GigabitEthernet0/0
      bind media source-interface GigabitEthernet0/0
      header-passing error-passthru
      early-offer forced
      midcall-signaling passthru
    interface GigabitEthernet0/0
    ip address x.x.x.x 255.255.255.252
    ip access-group acl.SIP-IN in
    no ip redirects
    no ip unreachables
    ip verify unicast reverse-path
    ip virtual-reassembly
    duplex full
    speed 100
    no cdp enable
    gateway
    timer receive-rtp 1200
    sip-ua
    connection-reuse
    gatekeeper
    shutdown
    dial-peer voice 1 voip
    description *** INBOUND CALLS FROM CARRIER ***
    translation-profile incoming SIPTRUNK-INCOMING
    session protocol sipv2
    incoming called-number #blah blah#
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 61 voip
    description **** WA, SA AND NT NUMBERS ****
    destination-pattern 0[8]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[8]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 81 voip
    description **** MOBILE NUMBERS ****
    destination-pattern 0[4]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[4]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 500 voip
    description *** INBOUND SIP TRUNK TO CUCM PUB ***
    translation-profile outgoing SIPTRUNK-CALLING-ADD-0
    preference 1
    destination-pattern 5[12]..
    session protocol sipv2
    session target ipv4:<OUR CUCM PUBLISHER IP>
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    Any help or a point in the right direction would be greatly appreciated.
    Cheers,
    Brett

    I ended up resolving this issue as follows:
    In CUCM, under Device > Device Settings > SIP Profile.
    I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
    Now, I get the expected delay before hearing ringback.
    Solved!

  • Problem with versioning on iPod, backwards compatibility issue with iTunes

    i apologize in advance because i'm sure this has been asked time and time again, but i did attempt several searches.
    the search function is very muddled and does not narrow down the results very well at all
    here's the situation.
    from all accounts i've heard lots of people having issues with version 7 of iTunes.
    i'm a simple kinda guy who definitely believes in the age old phrase "if it ain't broke, don't fix it."
    so i've stuck with v6.
    i recently got a new laptop along with my PC. so i downloaded and installed the latest iTunes without thinking twice.
    i had completely forgotten about my concerns from hear say about v7 issues.
    i plugged in my iPod and it proceeded to completely erase all of the media on it and then load all media that was on my laptop in place of it.
    "well that's a cool new feature", i thought.
    after my initial frustration wore off, i immediately uninstalled v7, deleted the .itl file and installed v6.
    everything was all fine and dandy. v6 works perfectly on the laptop with no hangups.
    here comes the issue...
    i plug in my iPod to begin rebuilding my media onto it after the completely unauthorized deletion, creditted to iTunes v7.
    as it autoplays and iTunes v6 opens i'm prompted with this error:
    "The iPod [my ipod] cannot be used because it requires iTunes version 7.0 or later.
    Go to www.itunes.com to download the latest version of iTunes."
    i've attempted everything i can think of to completely erase my iPod of any "versioning".
    - i reset it using the "select/menu" button method
    - i reformatted it through XP disk manager
    - i reformatted it through context menu in my computer explorer window
    - i completely just deleted the "iPod control" directory in the iPod explorer window when clicking through the my computer directory
    i'm at a loss.
    i cannot figure out a way to get my iPod to function with iTunes v6.
    and again, i'm really hoping to avoid going to v7.
    iPod information:
    2nd generation iPod nano 4GB
    thank you in advance for any and all information/advice

    I am guessing that you used your iPod enough with iTunes v. 7.x to update the firmware to the latest version. That being the case, reformatting your iPod won't "undo" the firmware update, so your only choice now is to update your iTunes software to v. 7.x.
    FWIW, iTunes 7 is a vast improvement over previous versions of iTunes. I use it on both Windows and OS X, and without bugs in both environments.

  • Jabber 9.6 no voicemail tab for CUC with CUCM 9 and CUC 8.6

    Hi guys,
    I have Jabber 9.2 and 9.6 clients with CUCM 9.1.2, CUC 8.5.1 and Cisco IM + P 9.1.1.
    We've recently updated the CUCM and CUPS to Cisco IM+P.  Also we have migrated from CUPC Clients to Jabber clients.  The CUC server has remained as is.  I was able to get the CUPC clients to get voicemail/visual voicemail but the Jabber client doesn't even display an ico for voicemail. 
    The service profile for all users is set correctly to point to the CUC server and there's no errors when I go to the jabber client "show connction status".  
    I haven't been able to find much about this issue.  I have noticed that if I set the mail store the voicemail tab appears but we are not using a mail store as the voicemail repositary.  Is this now required to check voicemail?  If so can someone point me to the configuration guide?  We have exchange 2012.  Thanks very much!
    -Akin

    Hi every body,
    I  have some problems but not with all the users, is only working with 3  users and the rest of them don't work, I attached the errors that I get  when I try to log in Telephony Accounts in the Jabber, I used the Jabber  of one User, and from here I introduce my credentials (user and  password) and I can get the voice mails (only from de 3 users that is  woriking fine), but when I introduce the credentials of the other user I  get the error and Can't get the voicemails.
    The 127.23.0.7 is the IP address of the Cisco Unity Connection.
    Please Help Me I'm stuck with this!

  • Calling issue with Cisco 7937 conference station

    Hi Friends,
    I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
    When making calls from once remote site to another using Cisco 6921 phones calls working fine
    When making calls from once remote site to another using Cisco 7937 conference station to make call  any phone at remote office, calls are getting disconneted, remote phone rings when calls,  but its gets fast busy tone when other party picks up the phone and  not able to talk.
    I suspect the issue with Codec but we have configured transcoders  in VG and registered with CUCM
    Please help me if any one experience such issue earlier.
    Regards
    Siva

    hi Basant,
    1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider. 
    Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
    Show Run
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
    sh run
    Building configuration...
    Current configuration : 12139 bytes
    ! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
    ! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
    ! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname eucamvgw01
    boot-start-marker
    boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
    boot-end-marker
    card type e1 0 0
    logging buffered 51200 warnings
    no logging console
    no aaa new-model
    no network-clock-participate wic 0
    no ipv6 cef
    ip source-route
    ip traffic-export profile cuecapture mode capture
    bidirectional
    ip cef
    ip multicast-routing
    ip domain name drreddys.eu
    ip name-server 10.197.20.1
    ip name-server 10.197.20.2
    multilink bundle-name authenticated
    stcapp ccm-group 2
    stcapp
    stcapp feature access-code
    stcapp feature speed-dial
    stcapp supplementary-services
    port 0/1/0
    fallback-dn 5428025
    port 0/1/1
    fallback-dn 5428008
    port 0/1/2
    fallback-dn 5421462
    port 0/1/3
    fallback-dn 5421463
    isdn switch-type primary-net5
    crypto pki token default removal timeout 0
    voice-card 0
    dsp services dspfarm
    voice call send-alert
    voice call disc-pi-off
    voice call convert-discpi-to-prog
    voice rtp send-recv
    voice service voip
    ip address trusted list
    ipv4 10.198.0.0 255.255.255.0
    ipv4 152.63.1.0 255.255.255.0
    address-hiding
    allow-connections sip to sip
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    fax-relay ans-disable
    sip
    rel1xx supported "track"
    privacy pstn
    no update-callerid
    early-offer forced
    call-route p-called-party-id
    voice class uri 100 sip
    host 41.206.187.71
    voice class codec 10
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 ilbc
    codec preference 4 g729r8
    codec preference 5 g729br8
    voice class codec 20
    codec preference 1 g729br8
    codec preference 2 g729r8
    voice moh-group 1
    moh flash:moh/Panjo.alaw.wav
    description MOH G711 alaw
    multicast moh 239.1.1.2 port 16384 route 10.198.2.9
    voice translation-rule 1
    rule 1 /^012237280\(..\)/ /54280\1/
    rule 2 /^012236514\(..\)/ /54214\1/
    rule 3 /^01223651081/ /5428010/
    rule 4 /^01223506701/ /5428010/
    voice translation-rule 2
    rule 1 /^00\(.+\)/ /+\1/
    rule 2 /^0\(.+\)/ /+44\1/
    rule 3 /^\([0-9].+\)/ /+\1/
    voice translation-rule 3
    rule 1 /^9\(.+\)/ /\1/
    rule 2 /^\+44\(.+\)/ /0\1/
    rule 3 /^\+\(.+\)/ /00\1/
    voice translation-rule 4
    rule 1 /^54280\(..\)/ /12237280\1/
    rule 2 /^54214\(..\)/ /12236514\1/
    rule 3 /^\+44\(.+\)/ /\1/
    rule 4 /^.54280\(..\)/ /12237280\1/
    rule 5 /^.54214\(..\)/ /12236514\1/
    voice translation-rule 9
    rule 1 /^\(....\)/ /542\1/
    voice translation-rule 10
    voice translation-rule 11
    rule 1 /^\+44122372\(....\)/ /542\1/
    rule 2 /^\+44122365\(....\)/ /542\1/
    voice translation-rule 12
    voice translation-rule 13
    rule 1 /^\([18]...\)/ /542\1/
    voice translation-rule 14
    voice translation-profile MPLS-incoming
    translate calling 10
    translate called 9
    voice translation-profile MPLS-outgoing
    translate calling 11
    translate called 12
    voice translation-profile PSTN-incoming
    translate calling 2
    translate called 1
    voice translation-profile PSTN-outgoing
    translate calling 4
    translate called 3
    voice translation-profile SRST-incoming
    translate calling 14
    translate called 13
    license udi pid CISCO2921/K9 sn FGL145110RE
    hw-module ism 0
    hw-module pvdm 0/0
    username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
    redundancy
    controller E1 0/0/0
    ip tcp path-mtu-discovery
    ip scp server enable
    interface Embedded-Service-Engine0/0
    no ip address
    shutdown
    interface GigabitEthernet0/0
    description internal LAN
    ip address 10.198.2.9 255.255.255.0
    duplex auto
    speed auto
    interface ISM0/0
    ip unnumbered GigabitEthernet0/0
    service-module ip address 10.198.2.8 255.255.255.0
    !Application: CUE Running on ISM
    service-module ip default-gateway 10.198.2.9
    interface GigabitEthernet0/1
    description to TATA NGN
    ip address 115.114.225.122 255.255.255.252
    duplex auto
    speed auto
    interface GigabitEthernet0/2
    description SIP Trunks external
    ip address 79.121.254.83 255.255.255.248
    ip access-group SIP-InBound in
    ip traffic-export apply cuecapture size 8000000
    duplex auto
    speed auto
    interface ISM0/1
    description Internal switch interface connected to Internal Service Module
    no ip address
    shutdown
    interface Vlan1
    no ip address
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 10.198.2.1
    ip route 10.198.2.8 255.255.255.255 ISM0/0
    ip route 41.206.187.0 255.255.255.0 115.114.225.121
    ip route 77.37.25.46 255.255.255.255 79.121.254.81
    ip route 83.245.6.81 255.255.255.255 79.121.254.81
    ip route 83.245.6.82 255.255.255.255 79.121.254.81
    ip route 95.223.1.107 255.255.255.255 79.121.254.81
    ip route 192.54.47.0 255.255.255.0 79.121.254.81
    ip access-list extended SIP-InBound
    permit ip host 77.37.25.46 any
    permit ip host 83.245.6.81 any
    permit ip host 83.245.6.82 any
    permit ip 192.54.47.0 0.0.0.255 any
    permit icmp any any
    permit ip host 95.223.1.107 any
    deny ip any any log
    control-plane
    voice-port 0/1/0
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/1
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/2
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/3
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    no ccm-manager fax protocol cisco
    ccm-manager music-on-hold bind GigabitEthernet0/0
    ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
    ccm-manager sccp local GigabitEthernet0/0
    ccm-manager sccp
    mgcp profile default
    sccp local GigabitEthernet0/0
    sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
    sccp ccm 152.63.1.19 identifier 4 version 7.0
    sccp ccm 152.63.1.100 identifier 5 version 7.0
    sccp ccm 172.27.210.5 identifier 6 version 7.0
    sccp
    sccp ccm group 2
    bind interface GigabitEthernet0/0
    associate ccm 4 priority 1
    associate ccm 5 priority 2
    associate ccm 6 priority 3
    associate ccm 3 priority 4
    associate profile 1002 register CFB_UK_CAM_02
    associate profile 1001 register XCODE_UK_CAM_02
    associate profile 1000 register MTP_UK_CAM_02
    dspfarm profile 1001 transcode
    codec ilbc
    codec g722-64
    codec g729br8
    codec g729r8
    codec gsmamr-nb
    codec pass-through
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    maximum sessions 18
    associate application SCCP
    dspfarm profile 1002 conference
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 2
    associate application SCCP
    dspfarm profile 1000 mtp
    codec g711alaw
    maximum sessions software 200
    associate application SCCP
    dial-peer cor custom
    name SRSTMode
    dial-peer cor list SRST
    member SRSTMode
    dial-peer voice 100 voip
    description *** Inbound CUCM ***
    translation-profile incoming PSTN-incoming
    incoming called-number .
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 500 voip
    description *** Inbound TATA MPLS ***
    translation-profile incoming MPLS-incoming
    session protocol sipv2
    session target sip-server
    incoming called-number ....
    incoming uri from 100
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 510 voip
    description *** Outbound TATA MPLS ***
    translation-profile outgoing MPLS-outgoing
    destination-pattern 54[013-9]....
    session protocol sipv2
    session target ipv4:41.206.187.71
    session transport udp
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 520 voip
    description *** Outbound TATA MPLS ***
    translation-profile outgoing MPLS-outgoing
    destination-pattern 5[0-35-9].....
    session protocol sipv2
    session target ipv4:41.206.187.71
    session transport udp
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 200 voip
    description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 0122365....
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 201 voip
    description *** Inbound M12 *** 012237280XX
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 012237280..
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 202 voip
    description *** Inbound M12 *** 01223506701
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 01223506701
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 210 voip
    description *** Outbound M12 ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern +...T
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 211 voip
    description *** Outbound ISDN for SRST and emergency ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern 9.T
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 212 voip
    description *** Outbound ISDN for emergency ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern 11[02]
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 2000 voip
    description *** Outbound to CUCM Primary ***
    preference 1
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:152.63.1.19
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 2001 voip
    description *** Outbound to CUCM Secondary ***
    preference 2
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:152.63.1.100
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 2002 voip
    description *** Outbound to CUCM Teritiary ***
    preference 3
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:172.27.210.5
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 999010 pots
    service stcapp
    port 0/1/0
    dial-peer voice 999011 pots
    service stcapp
    port 0/1/1
    dial-peer voice 999012 pots
    service stcapp
    port 0/1/2
    dial-peer voice 999013 pots
    service stcapp
    port 0/1/3
    sip-ua
    no remote-party-id
    gatekeeper
    shutdown
    call-manager-fallback
    secondary-dialtone 9
    max-conferences 4 gain -6
    transfer-system full-consult
    ip source-address 10.198.2.9 port 2000
    max-ephones 110
    max-dn 400 dual-line no-reg
    translation-profile incoming SRST-incoming
    moh flash:/moh/Panjo.ulaw.wav
    multicast moh 239.1.1.1 port 16384 route 10.198.2.9
    time-zone 22
    time-format 24
    date-format dd-mm-yy
    line con 0
    login local
    line aux 0
    line 2
    no activation-character
    no exec
    transport preferred none
    transport input all
    transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
    stopbits 1
    line 131
    no activation-character
    no exec
    transport preferred none
    transport input all
    transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
    stopbits 1
    line vty 0 4
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    line vty 5 15
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    scheduler allocate 20000 1000
    ntp server 10.1.30.1
    end
    eucamvgw01#
    Sh SCCP
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
    SCCP Admin State: UP
    Gateway Local Interface: GigabitEthernet0/0
    IPv4 Address: 10.198.2.9
    Port Number: 2000
    IP Precedence: 5
    User Masked Codec list: None
    Call Manager: 10.198.2.9, Port Number: 2000
    Priority: 3, Version: 7.0, Identifier: 3
    Call Manager: 152.63.1.19, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 4
    Trustpoint: N/A
    Call Manager: 152.63.1.100, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 5
    Trustpoint: N/A
    Call Manager: 172.27.210.5, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 6
    Trustpoint: N/A
    MTP Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1000
    Reported Max Streams: 400, Reported Max OOS Streams: 0
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    TLS : ENABLED
    Transcoding Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1001
    Reported Max Streams: 36, Reported Max OOS Streams: 0
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Supported Codec: g722r64, Maximum Packetization Period: 30
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
    Supported Codec: pass-thru, Maximum Packetization Period: N/A
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    Conferencing Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1002
    Reported Max Streams: 16, Reported Max OOS Streams: 0
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    TLS : ENABLED
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    eucamvgw01#

  • Directory Caching issue with Cisco Jabber client for Windows

    Hi ,
    I am facing cache issue with Cisco Jabber client for Windows. If I do any change related to modification or deletion of contacts in Active Directory/ Callmanager, it does not reflect in the Jabber. Because jabber takes the contacts from the locally stored cache file in the Windows system.
    Every time I have to remove the cache file to overcome this issue, practically it's not possible to do the same with all the Widows users. As, if any employee leaves the company and still I can see his contact appears in the "Cisco Jabber client". I have not seen this issue with Android/Apple iOS.
    Is there any automated way to remove the cache file? 
    Here is the detail of CUCM,Presence and Jabber.
    CUCM version: 9.1.x
    Presence          : 9.1.X
    Jabber              : 10.5 and 10.6

    Hello
    On our environment we had to install a dedicated Microsoft Certificate Authority "just for Cisco Jabber usage" to house the
    Network Device Enrollment Service.
    Our certificate for the CUPS were generated on this Certification Authority too.
    I discussed this certificate matter with my colleagues this afternoon and nobody seems to remember how these certificates were deployed into the
    Enterprise Trust store for the users.
    But I think they asked all 400 users to accept the 3 certificates by answering "yes" to the popup instead of using a script deployed by GPO...
    I wish you success with that deployment and really hope you have a technical partner that *Knows* this subject.
    Our partner left us alone with that unfortunately.
    Florent
    EDIT: If the "Certutil script method" works, please let me know. This could be useful in our own deployment.

  • DRS issues with Unity Connection 7.1.5

    Hi all,
    We're having some Issues with the DRS Backup of a CUCM BE 7.1.5 System.
    System Version is 7.1.5.10000-12.
    When selecting the CUC Feature in DRS the backup fails,
    The Error I get is the following:
    Failed to initiate backup. Server [ZURCM01], component [CLM] requires component [PLATFORM] to be registered.
    When deactivating the CUC Feature the Backup runs normally.
    Any ideas how to fix this?
    Thanks for all your Help
    BR
    Alex

    Hi Alex,
    You may want to try the steps in the below defect and see if it helps you any:
    http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCsw64366
    Hope that helps,
    Brad

  • Cisco Jabber for Mac 9.2.1: slow to register with CUCM

    Hi,
    We have this specific problem on our Jabber for Mac 9.2.1 client only: the client takes 60 to 70 seconds to connect to CUCM (running 8.6.2) when it starts.
    After Jabber registers with CUCM, there is no issue, it's just very slow to register.
    Anybody getting the same behaviour? We have no such issue on previous versions of Jabber for Mac version 8 or Jabber for Windows version 9.
    Thanks,
    Fabrice

    For those interested, I have the answer to my own question: if one of your configured DNS servers is not responding, Jabber 9.2.1 will delay registration to the phone services. This behaviour is seen only in this specifc version of Jabber, hopefully it will be fixed in the next release. Watch out if one of your DNS server goes offilne.
    Fabrice

  • Issue with transferring calls to VM for the correct DN Unity Connection 9.1.2

    Hi all
    I have been facing an issue with a Unity Connection Server v9.1.2. Every time an internal extension (assigned to VM profile and to Unity as a user) which is configured to be transferred to VM after 20 sec or so NoAN,is called , Unity treats the call as the extension of the calling party and not the called party. Furthermore , I dont know if this has any relation with the problem I am facing but when I check the voicemail port status in RTMT its seems like that regardless it is a direct call to the Unity from an extension (dial the pilot number or press the messages button on the IP phone) or a redirected call from an extension to Unity due to NoAN configuration, the Reason is Direct and the caller party number is always the extension initiated the call and not the one redirected to the Unity (second case).
    I have changed the Use Last (Rather than First) Redirecting Number for Routing Incoming Call  Unity parameter in the
    System Settings > Advanced >Conversations -> checked   and
    Redirecting Diversion Header Delivery - Outbound CUCM parameter ->checked
    in the SIP Trunk configuration used for the integraton of the CUCM with Unity but none of those seem to address this issue. Is there any guidelines you can give me to overcome this issue?
    The servers I am using are
    CUCM v9.1.2 BE
    CUC    v9.1.2 BE
    Thank You in Advance

    Hello again,
    The Voice Mail Box Mask was blank before. However I tried XXXX (I use 4-digit extension for the VM pilot) but this did not seem to fix anything to the system....Same situation as before. Any more suggestions?
    Thank you

Maybe you are looking for