Call SIP with Tandberg C40

Similar Questions

• The call history on Tandberg C40

  Can someone tell me where on a Tandberg C40 call history? I get frequent calls and crash and I can't tell where they come from.

  Thank you

  -mike

  Hello

  You can access C40 CLI (using SSH) and use the command xhistory.

  Concerning

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• Mounting ears for a Cisco/Tandberg C40

  I've been going around in circles now for weeks.  Anyone where I can retrieve a set of ears for a Cisco/Tandberg C40 system rack?

  I have a lot here that will not be used, which you can have for free. PM me with details.

  

  /Jens

• How can I put shortcuts to websites in a bar (?) tool bar/browser... dunno what it's called. With IE, I can just drag the tab of the site to the area I want.

  How can I put shortcuts to websites in a bar (?) tool bar/browser... dunno what it's called. With IE, I can just drag the tab of the site to the area I want

  Near the right end of the address bar is a White Star which can be hard to see, but look closely. When you want to add a shortcut to the toolbar, double-click the star. A window opens with the name of the Web page. Under this is the area of folder which will probably read "Unsorted Bookmarks". Click on the drop down arrow and change it to "Bookmarks Toolbar", then click done.

  I have several folders on my toolbar such as music, Mail, etc. in which I add several Web sites. If you want to add a folder, click on the second drop-down arrow next to "Bookmark Toolbar" and click on "New folder" and give it a name.

  To add a bookmark to a folder, once again double click on the star, make sure you're in the bookmarks bar and click one of the fall of two arrows to find the name of the folder.

  For other bookmarks use Menu bookmark. What is added here are accessible through bookmarks in the menu bar. If your Menu bar with bookmark, file, display, etc. is not visible, click the ALT key and they will become visible. To keep the menu bar, click view, toolbars and check the menu bar.

• When I play a game on facebook called words with friends, my computer freezes the game after a few movements.

  original title: computer freezes

  When I play a game on facebook called words with friends, my computer freezes the game after a few movements.  Why?

  Hi ileendunman,

  1. have you done any hardware changes or software on your computer before this problem?

  2. what browser do you use?

  If you use Internet Explorer, you can follow this link & check if the problem persists:

  The problems of games online using Internet Explorer

  Hope the helps of information.

• The service URL used in datacontrol call failed with code of State 404 not found

  Hello

  Using JDeveloper 12.1.3.0

  I'm creating a rest using EJB WebService.

  When I test my service it works fine but when I added it to the customer, what finish loading it return me this error

  

  The method that I created in the service requires two arguments, BigDecimal and String

      @GET
      @Produces("application/xml")
      @Path("/{key}/{name}")
      public UsersList checkUser(@PathParam("key") BigDecimal key, @PathParam("name") String name) {
          UsersList listOfUsers = new UsersList();
          listOfUsers.setUsersList(SessionBean.getUsersCheck(key, name));
          return listOfUsers;
      }
  

  but in my DataControl, it appears in the customer like that

  

  I created all on again once the Client and it always seems that way.

  Is the link that I defined in the service will not / {keys} / {name} or in the datacontrol /##key #/ ##name #?

  Thank you

  I discovered what I was doing wrong, sort of wheel, I tried to create a search query by using the arguments of the functions like @PathParams instead of using @QueryParams, so when I tested it I was inserting the values into the path because there was no form of insert values.

  Like this

      @GET
      @Produces("application/xml")
      @Path("/getname/")
      public CusttypList getByName(@QueryParam("name") String name) {
          CusttypList listOfCusttyp = new CusttypList();
          listOfCusttyp.setCusttypList(SessionBean.getCusttypFindByName(name));
          return listOfCusttyp;
      }
  

  With regard to the release I had to create the schema of an object by hand, using the portion of the schema that is generated by jdeveloper when I called count with all function.

  Thank you for your help

• Calling functions with and without parentheses

  Hello

  I was wondering why it is that I can create a function and call it with or without brackets (for example "select double f" OR 'Select double f()'), but I can call only SYSDATE, CURRENT_DATE etc. without parentheses.  If I do this "select SYSDATE() from dual", I get an error.

  SYSDATE is a function, no?

  Does anyone know?  I'm just curious.

  Thank you

  Jason

  It's funny how incompatible Oracle is. SQL SYS_GUID function must be specified with the bracket, while his counterpart from PL/SQL it takes two ways:

  SQL > set serveroutput on
  SQL > exec dbms_output.put_line (sys_guid);
  EDC6F517330B4F1281739CB7184CD3F0

  PL/SQL procedure successfully completed.

  SQL > exec dbms_output.put_line (sys_guid ());
  2FDB6ECD6F4042CA91068E0CC9EF4599

  PL/SQL procedure successfully completed.

  SQL >

  Reason is - after all the "single engine" about Oracle SQL and PL/SQL are always two separate motors. PL/SQL counterpart functions are declared in SYS. The STANDARD package, so if we want to use the parentheses, we could refer to SYS. SQL STANDARD:

  SQL > select sys.standard.sys_guid from double;

  SYS_GUID
  ---------------------------------------------
  3452B1206ABE484187A09928C1960F93

  SQL > select sys.standard.current_date (double);

  SYS. STAND
  ---------
  JUNE 10, 13

  SQL >

  However, works such as SYSDATE, USER and a few others are not is the standard package and we have not choice but do not use parentheses.

  SY.

• How to avoid the ASSERT message "PMString::SetKey - call SetKey with a string that is not a translation?

  Getting below to ASSERT the message when you try to store a value in the PMString object.

  PMString::SetKey - call SetKey with a string that is not a translation

  No idea how can I avoid this message? Y at - it any changes required in the .fr file?

  Try using a Widestring instead, because you don't need translation. If you do not, try SetTranslated()

• Exception by calling 'SearchDatastoreSubFolders_Task' with '2' or the arguments

  We are trying to create a script that will search our LUN and save my VM to an ESX host. I have a few FC LUNS and LUN iSCSI. The problem is none I only get an error message when he discovered the Local VMFS volume.

  The error is the followinng.

  Exception calling "SearchDatastoreSubFolders_Task" with "2" or the arguments: "path of the invalid data store ' [http://bru03srvesx:storage1 |]» [http://bru03srvesx:storage1] "". " "

  To: line: 142 tank: 62

  + $TaskMoRef = $DatastoreBrowser.SearchDatastoreSubFolders_Task < < < < ($DatastorePath, $SearchSpec)

  Does anyone have an idea?

  I think the line should say $FolderID instead of $folder.

  ...
                 $FolderID.RegisterVM_Task($VMx,$matches[1],$FALSE,$ResourcePool.MoRef,$null)
  ...
  

• Call doSubmit() with custom value does not update process

  I have a page with a standard form that has a "Save" button, but also a button that I have manually created via an anonymous PL/SQL block in the source of the element of a field on the form. (I did because I was already using a PL/SQL block to display the value of the item, but wanted to update the button appear immediately next to him, rather than somewhere on the page template.)
  
  Originally, this button was just a simple redirect to a separate detail page, but I found that if the user has made changes to the form, then click on this button, these changes would be lost. So I changed it to a call doSubmit() instead. But I had to distinguish between the user pressing the "Save" button and the custom button, the button call doSubmit ('CUSTOM'). I then created a branch that has verified the value of demand and went to the appropriate page that was pressed.
  
  What I found was that the process to update the underlying table for the form had not shot, even if it is unconditional, and therefore updates the original form have been lost.
  
  I eventually worked around this by changing the button so that instead of calling doSubmit with a custom value, it calls a JavaScript function that: 1) define a hidden field to '1' and 2) called doSubmit ('SAVE'). Then, I changed the direction to look at the value of this hidden field, rather than the value of the request. That has worked well.
  
  So this got me thinking: why my original implementation did not work?
  
  I suspect that doSubmit() only works to the custom of the values if you have defined a real button in the designer of the ApEx and configured with a name for this custom value. The HTML code for my button seems indistinguishable from a button created by ApEx, I think that there is something that happens behind the scenes to make doSubmit() work simply with a random value.
  
  Can anyone confirm or deny this hypothesis?

  Hello

  Could this be your problem:
  If you have a treatment of automatic line (DML), which only fires from below, if I understand correctly

  Valid Update Request Values: SAVE, APPLY CHANGES, UPDATE, UPDATE ROW, CHANGE, APPLY, APPLY%CHANGES%, GET_NEXT%, GET_PREV%
  Valid Insert Request Values: INSERT, CREATE, CREATE_AGAIN, CREATEAGAIN
  Valid Delete Request Values:DELETE, REMOVE, DELETE ROW, DROP
  

  At least on Apex 3.2

  BR, Jari

• Activate a call forward with a third party SIP device or an analog device

  Hello

  In a CUCMv9, how do I activate a call forward (all, not busy, no answer...) with third party SIP device or an analog device, connected to a fxs?

  I want to activate call forward like an Alcatel or Aastra PBX with a code.

  For example, I pick up the phone, with the code * 95 followed by the destination number and hang up the phone. And use the #95 to desactivate the call forward.

  Is it possible?

  Thank you.

  No codes for 3rd not part of SIP phones, no way to do it. Or for that matter, not even for Cisco phones, other than the CFA.

  Something other than CFA must be done via CCMadmin or CCMuser to any type of phone.

  For FXS which is not feasible if you run SCCP

  http://www.Cisco.com/en/us/partner/docs/iOS/voice/FXS/configuration/guide/fxssccpsplmft.html

  HTH

  Java

  If it helps, please note

  www.Cisco.com/go/pdihelpdesk

• Strange call of loop on C40

  I've seen a few times on a specific C40 systematically, it appears after select Connect conferences, I see it several times again and again in the xhistory.  Our MCU makes all the calls, the codec in the room does nothing, but auto answers the incoming call on the bridge.  The Protocol default call used by the gateway in H.323, the codec is not configured for SIP at this time.  The composition and call back numbers are those of the codec itself.

  *h xHistory CallLogs Call 3153 CallId: 296
  *h xHistory CallLogs Call 3153 Protocol: "Sip"
  *h xHistory CallLogs Call 3153 Direction: Incoming
  *h xHistory CallLogs Call 3153 CallType: Audio
  *h xHistory CallLogs Call 3153 RemoteNumber: "sip:[email protected]/* */"
  *h xHistory CallLogs Call 3153 CallbackNumber: "sip:[email protected]/* */"
  *h xHistory CallLogs Call 3153 DisplayName: "100"
  *h xHistory CallLogs Call 3153 CallRate: 768
  *h xHistory CallLogs Call 3153 DisconnectCauseValue: 1
  *h xHistory CallLogs Call 3153 DisconnectCause: ""
  *h xHistory CallLogs Call 3153 DisconnectCauseType: LocalDisconnect
  *h xHistory CallLogs Call 3153 DisconnectCauseCode: 0
  *h xHistory CallLogs Call 3153 DisconnectCauseOrigin: SIP
  *h xHistory CallLogs Call 3153 StartTime: "2013/09/24 12:07:02"
  *h xHistory CallLogs Call 3153 Duration: 0
  *h xHistory CallLogs Call 3153 Encryption: "Aes-128"
  *h xHistory CallLogs Call 3153 BookingId: ""
  

  All our conferences are scheduled through TMS 14.2.2 with a VCS X7.2.2.  C40 is running TC5.6, waiting on TC6.3.

  Patrick,

  My guess is that your endpoint is sitting with a public ip address and the sip scanning applications such as SipVicious are used to detect the possibilities to exploit the PSTN circuits. This may cause your end point to the research as he calls himself.

  Of the endpoint, you must define outgoing SIP listening off port and Sip it so that the VCS can still reach the end point.

  xConfiguration SIP ListenPort: Off

  xConfiguration SIP profile 1 out: on

  Deactivation of ListenPort stops to the endpoint to listen on port 5060/5061. Enabling means leaving all incoming and outgoing calls will reuse the connection open for the endpoint of the VCS in the original SIP Register message.

  There are a few changes in TC6.2 to solve this problem. Before TC6.2, if you had the ListenPort on stop and coming out on, this combination of configuration did not always work as expected.

  I would like to make these changes on the endpoint and the upgrade to TC6.2 If you can.

  This is the id of the bug you can watch for that matter: CSCue55239

• SX20 call SIP

  Hello

  I would like to know, is it possible to call the SX20 of another endpoint with the address ([email protected] / * /) without registering to VCS

  Appeal of sense as a standalone device at the other end.

  If so, could you guide me.

  Best regards

  Titebiket Mohamed Yatim

  Why do you do this? If its between cisco/tandberg/3 rd party

  equipment and its able to use h323, I would have preferred that.

  Stand alone sip very likely work properly behind the NAT.

  To do this, I highly recommend using a VCS-E

  Not to mention to factory default, it would be just to write what you wrote.

  Maybe you want to set the call by default sip Protocol and also play

  with the DefaultTransport framework for sip, but you can simply dial the ip address.

  Please remember useful frequency responses and identify useful or correct answers.

• Dialing SIP calls SIP Trunk (Avaya SM)

  Hello. I work with the telco team to send sip of our endpoints VTC voice calls to an Avaya Session Manager.

  I think I have all the areas and rules of research on the side of VCS in order to transfer the call to the SM.  I am able to confirm with the team of the phone company that the call reached the SM, but they receive the error "cannot determine realm."

  On the side of the VCS, I see the output below.  Is the problem on the VCS or SM side?  I need to create a trust with the SM? THX. - F

  • searchRule (2)
  • • Name: SIP Audio out rule2
  • • Zone (1)
    • • Name: TEST Session Mgr
    • • Type: neighbor
    • • Protocol: SIP
    • • Found: false
    • • Reason: Internal server error (cannot determine the domain)
    • • StartTime: 2015-03-09 13:21:58
    • • Time: 0.05
    • • Access controller (1)
      • • Address: 172.x.x.x:5061
      • • Alias (1)
        • • Type: Url
        • • Origin: unknown
        • • Value: [email protected] / * /

  Another option is to see if you get up with 5060/udp or 5060/tcp, which could make it easier to trace. But yes, I prefer the encryption as well.

  Often you have set a kind of trust for a specific connection, which could include the field src or dst, ip, port,...

  good luck ;-)

• Conference MCU SIP with VCS registration question

  Hi all

  We have a cluster of MSE 8510 MCU (software version 4.5 (1.72), Build 6.19 (1.72)) with 86 active confeneces. All of them are configured in the same way, to use H.323 and SIP registrar calls control.

  The thing is, I can see all conferences recorded under H323 and some of them just registered as SIP on the VCS cluster (X.8.2.2).

  What is the problem? Why can't all register using SIP?

  See you soon,.

  Petar

  Can you check to see how many conferences are recorded using SIP?

  If it's 50, this seems to be the maximum that can be recorded, see enhancement request CSCum48651.