Time based dial plan configuration

Hello experts!
We're trying to maximize security on our VOIP Gateway to avoid being victimized by long distance/international toll fraud. In efforts to address this concern, we're looking to somehow deploy a time based dial plan on our gateway (2821) which based on that it automatically shutdown outbound international dial peer for any calls made during off work hours (i.e 7pm- 7am including weekends):
dial-peer voice 119 pots
destination-pattern 01T
port 0/0/0:23
forward-digits 16
I understand this can be done inside Callmanager (we're running Callmanager 4.1(3)) as well but for extra security pre-caution we'd like to have it on our gateway preferably. Is there any way to accomplish this task on a 2821 VOIP Gateway?
Thanks,

You can use kron command.
This is an example:
kron occurrence NIGHT at 20:00 recurring
policy-list SHUT_DIALPEER
kron occurrence DAY at 8:00 recurring
policy-list NOSHUT_DIALPEER
kron policy-list SHUT_DIALPEER
cli dial-peer voice 119 pots
cli shutdown
kron policy-list NOSHUT_DIALPEER
cli dial-peer voice 119 pots
cli no shutdown
Regards.

Similar Messages

  • Dial plan: Can we change the redirect number (sip-sip)?

    Hello Scott Page/Alexei/Dirk Anyone.
    I have a question on dial plan:
    Call flow: A---cucm----pgw----voicemail
                  B-----|
    Its a sip to sip call, what i need to do is change the redirect number or rather add a digit to the redirect number at the PGW side.
    For instance:
    A number: 902228990
    B number: 902228996
    Redirect number: 92228996  --->Here i want to change it to 902228996
    pgw patch: 9.8.2    "Patch:"CSCOgs014/CSCOnn014""
    bash-3.00$ uname -a
    SunOS tcs-tza1 5.10 Generic_127127-11 sun4u sparc SUNW,Netra-440
    I checked from the cisco website but i think its only for isup sip calls only but not sure.
    numan-add:fullnumbertrans:svcname="Service Name",numbtype="Number Type", digstring="Original Digits",
    translatednum="Translated Digits"
    Where numtype can be choose 3 for redirect number.
    numtype—Identifier for the number type (1-5), it is one of the following values: 1—called party number 2—calling party number 3—redirecting number 4—calling party number and redirecting number 5—original called number
    Can some help me out with the dial plan configuration?.
    Regards,
    Aby

    Guys,
    This configuration would work?, please let me know
    1. mml> numan-add:service:custgrpid="DP00",name="BATMANredirect"
    2. mml> numan-add:fullnumbertrans:svcname="BATMANredirect",digstring="92228996", translatednum="902228996",numtype="3"
    numtype=3-->redirect number
    3. Adding a result type of NUM_TRANS:
    mml>   numan-add:resulttable:custgrpid="DP00",name="results",resulttype="NUM_TRANS",  dw1="BATMANredirect",dw2="3",dw3="3",setname="setname3"
    dw2--->number type hence i put it 3
    dw3--->NOA--->hence i put it as national ie 3,
    dw4--->Dialplan?-- can i avoid this?
    4. mml> numan-add:resulttable:custgrpid="DP00",name="noar",resulttype="R_NUMBER_TYPE", dw1="4",setname="setname3"
    dw1===>NOA again, i put it as dw1="4" as national.
    Can any of you help me if this would do the trick or anything else needs to be added?.
    Thanks and appreciate your time.
    Regards,
    Aby

  • Configuring Dial-Plan in CME

    Have to configure the dial-plan in CME version 10
    Is ther any kind of timer setting as in Call Manager to wait for all the
    digits and to route the call properly
    If user dial 0 to reception
    if user dial 024 to internal extension
    if user dial 02433 to other site going through the voip-dialpeer
    Can We do this please help

    For sccp phones under Telephony Service for sip phone I guess under voice register global
    This is the max time to wait for another digit 
    timeouts interdigit (telephony-service)
    To set the interdigit timeout value for all Cisco IP phones in a Cisco Unified CME system, use the timeouts interdigit command in telephony-service configuration mode. To return to the default value, use the no form of this command.
    timeouts interdigit seconds
    no timeouts interdigit
    Hope it helps

  • MRP - Auto creation of del schedule lines based on planned delivery time

    Hi,
    We have activated MRP (type PD) where Purchase requiesition is auto created by system for requirement quantity. We require to optimize delivery schedule in such a way that entire PR quantity is broken into various delivery schedule based on planned delivery time and requirement.  Scenario can be further explained with following example.
    Material Requirement for a month is say 1,25,000 units
    Closing Stock say 25,000 units
    PR Generated by system for 1,00,000 units. The entire quantity is schdulled with only one delivery schedule line as per planned delivery time. 
    The requirement is to generate multiple delivery schedule lines automatically in Purchase requisition based on planned delivery time so that Purchase orders can be placed with system generated delivery schedule lines.
    How can it be achieved ?
    Regards,
    Nirav Kinkhabwala

    Nirav,
    This subject has been discussed repeatedly in this and other forums.  I must assume that you overlooked the rules of engagement, which state that you should first search the forums and other public sites, before posting questions here.
    Standard SAP MRP cannot be made to generate multiple items in a Purchase requisition.  The functionality you seek is usually achieved when converting purchase reqs to Purchase orders, where many single purchase reqs can be adopted into a single Purchase order.
    You also might want to investigate use of Vendor Scheduling agreements.
    Best Regards,
    DB49

  • How do I configure Dial-plan redundancy - Priority

    Hello
    I am setting up Dial Plan to single Destination PREFIX to 2 seperate gateway (not under my control).
    1st Gateway as 8 PSTN line
    2nd Gateway has 4 PSTN line
    how do I configure my cisco 3620 gateway that will allow me to route calls to 1st Gateway as long as all 8 PSTN lines are available.
    I know if you configure preference under dial plan like the one below
    dial-peer voice 88021 voip
    destination-pattern 8802T
    session target ivp4:x.x.x.x
    preference 10
    dial-peer voice 88022 voip
    destination-patter 8802T
    session target ipv4:y.y.y.y
    preference 9
    but this does not work as long as IP network is fine. I need as solution that will allow me to route calls to 2nd gateway if PSTN interfaces are all full/busy
    These 2 destination gateways are not under my control therefore, I can't request them to use my gatekeeper.
    Any other solution that I can try in my gateway
    Faisal

    This will work. The clal will be routed to the first gateway with prefernce 9 and then if the call fails, the call will be routed to the second gateway with prefernce 10.
    The only trick is the call has to fail with an error message which will tell the gateway to hunt. For example user busy will not cause the gateway to hunt. No cirucit or channle, no resource, no route to destination etc will all make the gateway hunt.
    You can on the other hand use advanced busyout on the remote gateways to tell the originating when the interfaces are busy.
    Taimoor

  • New Dial plan & Voice policies not taking effect with Polycom CX 600 Desktop Phone in production deployment, Worked fine in Testing

    Hi,
    We are in the process of Migrating Cisco CUCM & Voice Gateway (From another vendor to Cisco).
    The requirement is all internal calls between Cisco IP Phones & Lync to be flown through CUCM. Means internal extension to extension. Remaining all calls like Mobile, National, International, Toll Free, Emergency, Shared numbers calling to be routed
    to Cisco Voice Gateway.
    We created the test dial plan, Voice policies, Route and assigned it to couple of user from Lync (2 extensions) and from Cisco side we have taken 2 IP Phones which is pointed to new CUCM. We tested all below scenarios,everything was working fine.
    Lync to Lync Call using internal Extension number – Routed through Cisco new CUCM
    Lync to Cisco Call using internal Extension number – Routed through Cisco new CUCM
    Cisco to Lync Call using internal Extension number – Routed through Cisco new CUCM
    Lync to Hotline Numbers (66XX, 68XX Numbers) – Routed through Cisco Gateway
    Lync to Shared Numbers starting with 600 (Verified the number 600535353) - Routed through Cisco Gateway
    Lync to Emergency numbers & Toll Free Numbers (Not verified the emergency Number as we decided to do it at end) - Routed through Cisco Gateway
    Lync to Landline Numbers – Any 7 digit numbers - Routed through Cisco Gateway
    Lync to National Numbers – Starting with 3,4,6,7,8 followed by 7 digits - Routed through Cisco Gateway
    Lync to Mobile Phones – Starting with 05 contains exactly 10 digits - Routed through Cisco Gateway
    Lync to International Numbers – Starting 00 contains at least 11 digits - Routed through Cisco Gateway
    All Incoming calls – From Landline, Mobiles, International Numbers - Routed through Cisco Gateway
    Call Transfer – To another Lync Extension, Cisco Extension, Landline, Mobiles, International Number
    Conference – with another Lync Extension, Cisco Extension, Landline, Mobiles, International Number
    Call Forwarding – To another Number, Voice mail
    Response Groups
    Click to call – As if user try to place a call by directly click the number from Outlook, Websites will be in E.164 format
    Dial in meeting – Conference calls are works fine
    But when we roll out to the production we are facing issues listed below
    1) The phones we used during testing are working which is using same dial plan, Voice policy, Route, PSTN Usage. But from production most of the phones are not working (using the same dial plan, voice policy, Route). Also Problem is only with external calls
    as the internal calls are working fine between Cisco & Lync even in production (Routed through CUCM) NOTE: All incoming calls are working fine (From international, local, national, extension)
    2) How long its going to take for Lync to push the new voice policies, Dial plans to the Phones?
    3) Is there a way to forcefully update the policies, dial plans to the Phone?
    4) Also the environment is using over 100 dial plans, so I just copied and pasted the Normalization rules that we tested and working fine.  Most of the dial plans are assigned to individual users as every dial plan contains a normalization rule for
    international calling with Unique Prefix (Example: User John international Normalization rules says #1234#00#CountrycodePhonenumber, means if John has to place the international call he need to dial #1234# followed by 00 and then country code, then actual
    phone number). In this case how long its take for the users / phones to get updated with new dial plans? 
    6) Is it recommended to use multiple dial plans ? What are the best practices?
    5) Also calls are working fine one & failing on subsequent tries. Means when I dial first 1 or 2 times. Call fails, but when I try 3rd time and subsequently it works. After some again there will be failure during 1 or 2 attempts. Why is it so?
    6) After updating the dial policies, voice Route, Voice policies If i reboot all the phones from Switch, Will the changes take effect immediately?
    7) Also when some one calling from mobile or external number to Lync extensions they cant here any Dial tones or caller tunes? Its working fine when they call Cisco Extensions. Also to Lync its working if we dial in E.164 Format, if we dial like 023XXXXX
    format its not working. Any guess about this issue?
    Waiting for some one to help, 
    Best regards
    Krishna
    Thanks & Regards Krishnakumar B

    Hi,
    1.  As all incoming call worked normally, please double check outgoing ports for Lync FE Server and Mediation Server.
    You can refer to the link of “Ports and protocols for internal servers in Lync Server 2013” below:
    http://technet.microsoft.com/en-us/library/gg398833.aspx
    2.  When an administrator makes a change to Lync Server (for example, when an administrator creates a new voice policy or changes the Address Book server configuration settings) that change is recorded in the Central Management store.
    In turn, the change must then be replicated to all the computers running Lync Server services or server roles.
    So it may not replication completely immediately.
    3.  You can run the following cmdlet with Lync Server Management Shell on FE server to
    forcibly replicate information to a computer: Invoke-CsManagementStoreReplication
    4.  As you used over 100 dial plans, it may be the issue of multiple dial plans. Would you please tell us why you created different dial plan for individual user with unique prefix?
    5.  Multiple dial plans and undue normalization rules may cause call fail. You can double check the normalization rule.
    Best Regards,
    Eason Huang
    Eason Huang
    TechNet Community Support

  • Cisco WS-C6513 taking long time to save the configuration

    HI,
    cisco WS-C6513 taking long time to save the configuration.
    Any ideas?
    Thank you 

    Hello,
    do you have correct dial plan ? It very depends on the country you live and you have your VoIP operator.
    Try to finish your dialing with # - this may speed it up.

  • Time based mode runs 4 hours before time set

    time based modes are running 4 hours early...
    The mode was set for 6:00 pm,  ran at 2:00 pm.
    changed it to 8:00 pm it then ran at 4:00 pm
    when creating the mode it said that EST after the begin time, so it knows the correct time zone that I am in
    Bill T

    Paul,
    Can you elaborate on where you applied the configuration. I have a scenario where Rightfax sends the call to CUCM via an H323 trunk, then CUCM sends the call to the PSTN Gateway and then out to the PSTN. I am experiencing the same issue where the call disconnects before the destination fax answers if it is set to ring 3 or 4 times before answering.
    I'm not sure that I understand what you mean by splitting the incoming and outgoing dialpeer. I have the following dialpeer to match and route my outbound number to the PSTN.
    dial-peer voice 92 pots
      translation-profile incoming PREFIX-9
    translation-profile outgoing CALLERID
    destination-pattern 91[2-9]..[2-9]......
    progress_ind setup enable 3
    progress_ind progress enable 8
    port 0/0/0:23
    forward-digits 11
    I appreciate any help you can provide.

  • AA unable to transfer to Lync 2013, but only on User dial plans

    The current environment has two Lync 2013 standard edition servers.  Lync 2010 is still in existence, and both pools are connected to an Exchange 2010 SP2 UM server.
    UserA is part of the default Global dial plan in Lync and part of a UM dial plan named Global.  UserA has been moved to Lync 2013 pool.
    UserB is part of a user dial plan in Lync named UserDialPlan and a UM dial plan named UserDialPlan.  UserB has been moved to Lync 2013 pool.
    Both dial plans have their own route/pstn usage tied to their own sip trunk from Intelepeer.  Both UM dial plans have an AA identically configured with key mapping #1 tied to dial the 11 digit phone number of the user.  Both users can successfully
    be dialed directly via PSTN.
    If I call Global’s AA and press 1, the call is successfully connected and rings through to UserA.
    If I call UserDialPlan’s AA and press 1, I get “Sorry I couldn’t transfer you to the extension” and the UM server logs shows errors 1079 and 1136:
    1079 The VoIP platform encountered an exception Microsoft.Rtc.Signaling.OperationFailureException: Failed to transfer, successful refer notification not received
    1136 An error occurred while transferring a call to "15552735555". Additional information: The call transfer type is "Blind.", the transfer target is "phone number", and the caller
    ID is: "0ed8114d-a068-41be-9790-9342d0a02d7b".
    If I switch the ip address of our sip trunk back to the Lync 2010 mediation server and alter the Lync topology to connect that trunk back to the 2010 server, then these transfers start working again.  This goes for other
    user dial plans as well.  What I don’t understand is why the Global dial plan would work on 2013 and the user dial plans will not.  Refer is disabled on the trunk config (has been for
    over 2 years).  I assume if it was a general setting like that, the Global plan wouldn’t work either, but what is it that’s special about the Global plan vs. user plans?
    Any thoughts would be great, thanks!

    I have the same issue. I can actually transfer to extensions when it is speech enabled and I say the person's name. However if I try a key mapping to transfer to the same user using the Extention I get the same "Sorry I couldn't transfer you to the extension"
    An error occurred while transferring a call to "151". Additional information: The call transfer type is "Blind.", the transfer target is "phone number", and the caller ID is: "4813feae-2a10-4d33-9612-ee4bccf7c0f9". 
    The VoIP platform encountered an exception Microsoft.Rtc.Signaling.OperationFailureException: Failed to transfer, successful refer notification not received
       at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
       at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
       at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result, String operationId)
       at Microsoft.Rtc.Collaboration.Call.EndTransferCore(IAsyncResult result)
       at Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoCall.EndTransfer(IAsyncResult result)
       at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.BlindTransferSessionState.Call_TransferCompleted(IAsyncResult r)
       at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.SubscriptionHelper.<>c__DisplayClass5f`1.<>c__DisplayClass62.<WrapCallback>b__5e()
       at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.<>c__DisplayClassd.<CatchAndFireOnError>b__9()
       Detected at System.Environment.get_StackTrace()
       at Microsoft.Rtc.Signaling.OperationFailureException..ctor(String message)
       at Microsoft.Rtc.Collaboration.Call.CallTransferAsyncResult.Refer_StateChanged(Object sender, ReferStateChangedEventArgs e)
       at Microsoft.Rtc.Signaling.ReferStateChangedEventArgs.Microsoft.Rtc.Signaling.IWorkitem.Process()
       at Microsoft.Rtc.Signaling.WorkitemQueue.ProcessItems()
       at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()
       at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Object state)
       at Microsoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallback method, Object state)
       at System.Threading.ExecutionContext.Run(ExecutionContext executionContext, ContextCallback callback, Object state)
       at System.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallback tpWaitCallBack)
       at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)
    FailureReason = 0 during the call with ID "4813feae-2a10-4d33-9612-ee4bccf7c0f9". This exception occurred at the Microsoft Exchange Speech Engine VoIP platform during an event-based asynchronous operation submitted by the Unified Messaging server. The Unified
    Messaging server will attempt to recover from this exception. If this warning occurs frequently, contact Microsoft Product Support.

  • Need UC320 Region Pack or dial plan modification for Egypt

    We have bought a UC320W device from a Cisco local distributer in  Egypt, but we are facing a problem with the configuration as the device  doesn't have a Region Pack for Egypt so we used some other country's  Region Pack, the problem with this configuration is that it only  supports dialing external land line as it doesn't support dialing mobile  phone or International numbers, is there a Region Pack for Egypt? Or a  way to modify a dial plan?
    Thanks.

    Hi Friedrich,
    To answer your question about the authorized local Cisco  Distributor, we have bought the UC320W from Metra Computer, I've  configured everything correctly using the Brasil Region Pack except  for  the time-zone, each time I change the time on the phone-set (SPA504G) it  reverts  after a little while to the time set by the UC320W because it's  being  provisioned by it, I have found the ChangeNTPserver.pmf among my   exploration, so I guess it's possible to provide a .pmf file that can   add the GMT+2 time-zone to the Brasil Region Pack or disable the   time-zone provisioning, can you help me  with that? How can I request   such support task adequately?
    Thanks and regards.

  • Problem in Time Based Publishing Content

    Hi every1,
      Im working with Time based publishing.
    Using xml form builder i created 3 contents means 3 xmls.
    Then i created one iView for reading the contents (KM Navigation iView) and i setup the properties like
    layout set, layout set mode, root folder of contents.
    After creation of iView i checked in the preview all 3 contents visible in my iView.
      Now i want to show time based content in that iView.
    Contents displayed as per time based
    for that i enabled time based publishing and life time of particular content(xml)by using the given way
    Time dependant publishing in Km. I clicked on the right hand side of the name of my folder-> go to details -> Settings -> Lifetime. there you have to enable the time dependant publishing. Then i opened the folder and click on the rt hand side of the document-> properties -> lifetime, here give the time span of the document.
    After life time setup , again i seen in the iView for reading the contents (previous created) in the preview
    again all 3 contents displayed including life time expired content also.
       Please give me solution for this, or any more configurations required.
    Note :
    I required to display the contents in between time applicable only ( from time and until time).
    Thanks in advaince
    Vasu

    I have waited more than 3 hours for settings to apply.
    But i couldn't find any changes.
    any other solution?
    Thanks
    Vasu

  • Best practice to Change Dial plan?

    Hi,
    Customer has made plenty of misdialed 911 calls so they want to change the dial plan. They have CUCM, CUC and UCCX .. I will try to suggest putting a delay for 3 sec or so and blocking 911! or 911!# translation pattern .. but in case if they do want to change their dial out number.. what's the best practice for this? I tried looking for a suggestion or document but couldn't find it... at this point I can only think of copying existing RP's and change the dial out number to 8 and if required if they will have an H.323 gateway then might require configuration on dial peers... Any suggestion on this is appreciated? Thanks

    Hi Vishal,
    If 911 is being dialed accidentally, you can try configuring a 9.911 or 911# route pattern. 9.911 will  require you to change the destination pattern and forward digit settings on dial-peer or you can actually strip it on cucm itself and will not require a change on dial-peers. Other than that you need to check your dial-plan and see if there are any router patterns that are overlapping with 911 you can try editing them as well by changing the first digit for those route patterns to something other than 9.
    HTH
    Manish

  • SPA3000 F/W 3.1.20(GW) last digit repetition within the dial plan is not working for gateway 0-4

    H/W: SPA3000
    F/W: 3.1.20(GW)
    Problem: The last digit repetition within the dial plan is not working for gateway 0-4.
    Line 1 Dial Plan: (****|<#,xx.<:@gw1>|1747xxxxxxxx.<:@gw1>)
    Dial "17474743246#" which works for gw1.
    Dial "#17474743246" which also works for gw1.
    Dial "17474743246" which doesn't work and actually goes through VoIP 1 but not gw1.
    It results in I have to specify a fixed number of digits for gw0-gw4 in the dial plan or it won't works. This could be very difficult for international calls.
    Is there any way to specify Interdigit Short Timer with <:@gw0>?
    BTW, it's strange that I have to include (****) in the dial plan to enter IVR. Without it I can't even hear any sound coming from gw1 (SIPphone). Maybe the SPA needs a factory reset. The test dial plan might look like,
    S:4,(****|*xxx.|[2-9]|xxxxx|*0<:@gw1>|1747xxxxxxx<:@gw1>|1408xxxxxxx<:@gw0>|xxxxxxxxxxxxxxxxxxxx.!)
    The last barring sequence "xxxxxxxxxxxxxxxxxxxx!" is to make sure the Interdigit Short Timer works after the last digit dialed otherwise the number will be transmitted immediately which is not preferred.

    IMHO , if there's a need for you to include *** on the dial plan to enter the IVR menu there might be a need to reset the unit.
    I believe that you can set the interdigit timer by including that on the dial plan ie: xxxxS0 just like on a hotline, i just never had a chance to try including that syntax when invoking gw0 on the dial plan , but i guess you may give it a try.

  • Repository Services - Time based publishing missing

    Hi,
    We are running NW07, and want to configure time based publishing.
    I can't find the Repository Services for this it is suppose to be under
    System administratoin -> Content Management -> Repository Services
    But it is not,
    can anyone help?

    After that, you have to define the real lifetime
    http://help.sap.com/saphelp_nw70/helpdata/en/e8/a9a76828b8dc469969ff450ec81ced/frameset.htm
    An keep in mind that only users with not more than read permissions will see the document only during its lifetime. Users with write additional permissions can always see it
    Kind regards
    Karin

  • Time-based publishing stopped working

    Hi,
    We currently have a problem with time-based publishing in KM. Since a few days ago, documents stopped becoming visible once they reach their "valid from" date. We have not been able to publish documents with TBP since then on that system.
    These errors keep appearing in the knowledgemanagement.#.log files, which seem related to this issue :
    #1.5#C000AC10005900130000012D00000B3400040F325EE040B8#1142608921379#com.sapportals.wcm.WcmException#irj#com.sapportals.wcm.WcmException.WcmException(62)#System#0#####ThreadPool.Worker1##0#0#Error##Plain###application property service not found com.sapportals.wcm.repository.service.timebasedpublish.wcm.TimebasedPublishException: application property service not found
         at com.sapportals.wcm.repository.service.timebasedpublish.wcm.TimebasedPublishServiceManager.getApplicationPropertyService(TimebasedPublishServiceManager.java:589)
         at com.sapportals.wcm.repository.service.timebasedpublish.wcm.TimebasedPublishServiceManager.setValidEventSent(TimebasedPublishServiceManager.java:540)
         at com.sapportals.wcm.repository.service.timebasedpublish.wcm.TimebasedPublishServiceManager.handleVisibleResources(TimebasedPublishServiceManager.java:327)
         at com.sapportals.wcm.repository.service.timebasedpublish.wcm.CheckValidFromSchedulerTask.run(CheckValidFromSchedulerTask.java:65)
         at com.sapportals.wcm.service.scheduler.SchedulerEntry.run(SchedulerEntry.java:128)
         at com.sapportals.wcm.service.scheduler.crt.PoolWorker.run(PoolWorker.java:107)
         at java.lang.Thread.run(Thread.java:479)
    The KMC version is SP2 with Patch level 29 hotfix 1, and is running on Windows Server 2003 with an Oracle database. We have opened an OSS message but while we are waiting I thought I would post this here in case anyone ever experienced this.
    Best regards,
    Olivier

    Hi,
    1.  Have you checked that tbl service continue assigned to your repository ?
    2. If you create a new repository and assign these service, does it work ?
    Enables users to define a time frame during which documents are published (visible).
    Note that the time-dependent publishing service requires the application property service.
    This service cannot be configured.
    Patricio.

Maybe you are looking for

  • How to create a report with dynamic no of columns

    Hi All, we have a report with 6 columns. and its been access by 10-20 people. the user dont want to have a fix report with 6 columns rather they want to have a flexibility to select the columns from the report they want to see. i.e. user one one time

  • Alpha keys are locked on wireless keyboard?

    The iMac wireless keyboard will not type alpha characters but is paired and batteries are good. Help? Isctheirca lock command that I should use to reverse an accidental lock?

  • Internal Table To Excel

    Hello everyone, I am downloading an internal table using the FM SAP_CONVERT_TO_XLS_FORMAT. Is there another FM which would allow me to specify a range of columns?? There are currently 300 columns in my table and because of the limitation of excel i.e

  • How do i make an rollback?

    Hi, how do i make an rollback? I have the following code but it's not making the rollback correctly. First query runs fine and second one results in error but it not rollback the first query, the update, as i want. Any help? Thanks. java.sql.Statemen

  • Quicktime in firefox browser

    If i want to download a movie i place the url in cocoawget or what seems to happen in firefox browser is that the quicktime symbol appears in the broswer and the movie downloads then plays in the browswer. Once i close the broswer where is the movie