EX90 - presentation-sharing ends of appeal

Similar Questions

• Presentation app ended up in appworld

  Well, my questions are.

  Everything I submit is the. COD file correct?

  In addition, the .txt & .jpg files that are needed to run the program are that they included in the .cod file? or I have to send a zip file instead?

  I noticed when I clicked on Add selected files to COD and a .cod, he ended by adding 3 different .cod files... This means that it is running correctly?

  SirSaula wrote:

  Everything I submit is the. COD file correct?

  Yes

  In addition, the .txt & .jpg files that are needed to run the program are that they included in the .cod file? or I have to send a zip file instead?

  If these resources were included in your project, they will be included in the file of cod so

  I noticed when I clicked on Add selected files to COD and a .cod, he ended by adding 3 different .cod files... This means that it is running correctly?

  If the cod file exceeds 64 KB it is turned in archive containing smaller COD files. If your cod file is unpacked by the backend BB App World and is well served to the end user. No worries

• Automatic presentation sharing

  Hello

  is there a way to share the presentation automatically without the push of a button on c-codec? With CTS we´ve learned, this would facilitate users life.

  Thank you

  Tino

  Ah ok. Hmm, like the Touch of Cisco does not have a customization that not help you here

  (in addition to put a sticker on it "Please press present" ;-)

  Like I said there could be more likely to do it with a C60/C90, I don't have one here.

  There is a script that the c-series runs on the boot time, there may be something to

  call connect and disconnect as well.

  Didn't test if the xstatus shows a change if a cable / resolution is connected and if you could

  get it with xfeedback, ASA, it could also be done via a control system.

  In general, has filed a feature request? seems reasonable to have a "xconf presentation autostart: on" feature.

  I would like TI + 1.

  Otherwise, I guess it would be a great arduino or raspberry pi projcet :-)

• C40 how to disable local photo content as the default on the introduction screen when there is no presentation

  We have a conference room with a C40 facility.

  We have several projectors that we can control the content of the codec appears which projector.

  When this piece is in a direct appeal of the VC and the East end of talk/active with presentation - all is well main projector a Live, other 2 projectors have presentation flow. This is when the content of the presentation the end stops, & the person speak always active - the local 2 projectors who did not have the presentation now show the picture of local Codec on the screens. Would prefer that they will just black. As the local camera or not can someone have in there.

  Any ideas would be appreciated.

  Thank you

  Do you care about at the second output to display selfview will your presentation projectors?  Try changing the output mode 2 display only presentaiton.  I guess that it is set to dual right now?

• Cisco MX200/MX300 & screen tactile EX90 and Crestron

  Does anyone know if they can control these units using a Crestron touchscreen?   I know the other c40/c60 c90, codecs, can be controlled via a screen touch crestron, but not sure about these.

  I ask because we put 8 of these units in conference rooms that already have a Crestron in them and want to use only one of the panels rather than having the two sat on the table.

  Kind regards

  CJ

  Hello

  First of all, I want to mention that even if you try to use the Touch panels, it wouldn't work. The reason being that when Cisco Touch Panel is connected, it takes control of the OSD and it is not possible to control using other interfaces. I recommend that you use only the Crestron.

  Come to the second part. In Ex90, there is a maintenance RJ45 port at the bottom that can be used to combat it. If you can get the cable for RJ45 connected on Ex90 to one end of the cable and the other end to Crestron, I think you should be able to achieve the desired results.

  For the MX series, there is a serial port, allowing you to control the device, so I think you can plugin the Crestron using the serial port and you should be able to send commands to it and control it. However, keep in mind that if the Cisco button panel is connected, you could face questions by using the Crestron. For your reference, you will find the parameters of the serial port on the Mx series below:

  SerialPort Mode

  Set the COM 1 serial port to be enabled/disabled.

  Requires the user role: ADMIN

  Value space:

  On: Enable the serial port COM 1.

  OFF: Disables the serial port COM 1.

  Example: SerialPort Mode: on

  SerialPort BaudRate

  Specify the speed of transmission (data transmission rate, bits per second) for the COM 1 port on the codec.

  The default is 38400.

  Connection of the COM port settings: data bits: 8; Parity: None; Stop bits: 1; Flow control:

  None.

  Requires the user role: ADMIN

  Value space:<9600 9200/38400/57600/115200="">

  Range: Select a baud rate listed baud rate (bps).

  Example: SerialPort BaudRate: 38400

  Kind regards

  Saint André

• Presentations of edition is more correctly

  Presentations published by correctly two weeks ago. Source files look correct. For example a slide with a first line of text that does not build, followed by four bullets that build, all look normal with proper audio etc. mouse clicks. But now the opening of line built even if she didn't any what animation mark and all points of reference are disabled. The difficulty was to penetrate all elements of separate text boxes, then they will build correctly. However, I have about 30 presentations already build that are close to be definitive and rebuild with such a hack is simply not an option.
  
  Someone at - it ideas? Could he have an upgrade of windows or a recent update of the virtual Java machine that caused such a problem?
  
  I'm under PPT 2002 SP3 and Breeze Presenter 5.1.248
  
  I get the same results publishing locally or publish on our hosted account breeze/acrobat.
  
  Thank you!

  It seems that Adobe as applied some updates to their hosted servers and that the old version of Breeze no longer works with updates. The difficulty seems to be simple. Update your PPT file by logging in to your account on admin.acrobat.com and down load the latest version of Acrobat Communicator. ( http://www.adobe.com/support/breeze/updates.html) We also complete some windows updates, but also an update of java in the context of trying to solve this problem. I don't know if they are a necessary part of the solution, but I recommend them.

  A warning. The new Communicator Acrobat seems to treat audio very differently - rather then the events of compression process early in the process of creation, it seems now occur when presentations are published. The process can also lead to changes in the applied compression. We had worked with 16 bit files mp3 with good results. After completing these updates our 16-bit mp3s were cleaner. Maybe they have been compressed twice given the calendar of our updates, but to complete the presentations, we ended up going back to our audio source files and work with files of higher quality source. The results are better - however, the ppc files are now huge! Once they are published presentations are the same size they were before, but the local files ppc are considerably higher. (if you work with 32-bit mono... they might be 100 MB for 10 minutes.

  Should first update all our versions right back to where they were.

  If you work with the old version - a temporary work around that, I discovered in the process was brake up blocks of text so that the items that were supposed to be static were not part of the animation of text blocks and that seems to fix the problem.

• Random Jabber disconnections of MCU calls

  Good afternoon

  Hope you can help me. I've seen a few posts with these error messages in but they are not quite what is happening here

  My customer is having a problem, where a number of their Jabber users (for windows) is randomly calls MCU is disconnected on their server of telepresence. The disconnection occurs in less than a minute.

  None of the presentations are shared when this happens.

  The errors are:

  Operation BYE failed because the network error

  End of appeal because INVITE it transaction timeout

  Call due to the failure of closing invitations

  GST 320 - software version: 4.0 (2.8)

  Made - known Jabber clients versions 4.4 and 4.7

  Currently waiting for the client to update me on MSDS, blade of conductor and versions of software VCSE and VCSC

  event_log_screenshot.PNG

  

  telepresence_server_status.PNG

  

  I hope you can help

  Thank you very much

  Luke

  In general, this is because of a network problem, where SIP messages are not sent correctly between Jabber and reliable TP server, in this case, I'm guessing the OK/acknowledgement of receipt are not sent to the head of the Orchestra/TP server and the server TP is disconnecting the call. Is there a firewall in the mixture, do you SIP ALG turned on? If I look at the firewall and disable SIP algo

• Bandwidth configuration Expressway

  Hello

  Can someone let me know to change the bandwidth on Hwy C and E server configuration.

  This will affect the quality of B2B video calling and Conference I endpoints MX800, DX 80, EX80 and C60.

  I also have the TMS, is it also requires change bandwidth on MSDS.

  The conductor, I configured HD for the teleconference scheduled.

  Bandwidth restrictions: the speed of maximum Session for video calls on the default region on Cisco Unified Communications Manager is 384 kbit/s by default. The bandwidth by default call on Highway-C is also 384 kbit/s by default. These settings may be too low to provide the quality expected video for the DX series.

  http://www.Cisco.com/c/dam/en/us/TD/docs/voice_ip_comm/Expressway/Admin _...

  Kind regards

  RACLOT

  Hi r.,.

  Cucm point of view, it is a function of the parameters of the region. As explained with examples in the SRND

  When a video call is established between two video end points, audio and video streams are established and the bandwidth is deducted for the negotiated rate. Unified CM uses regions to determine the maximum bit rate for the call. For example, with a Cisco TelePresence EX90 system to higher detail of 1080 p at 30 frames per second (fps), the rate negotiated between regions would have set at 6.5 Mbps. EX90s used in this scenario would be on average about 6.1 MB/s for the session. When endpoints start the presentation during the sharing session, BFCP is negotiated between the two ends and a video stream is activated at 5 fps or 30 fps, depending on the endpoint configuration. In this case, endpoints are butterfly to the bottom of their main video stream to include the video presentation so that the entire session use more than the allocated 6.5 Mbps throughput. Thus, the average bandwidth consumption remains the same with or without presentation sharing.

  Immersive Telepresence and desktop settings such as the Cisco TelePresence system 500, 1000, 3000, and TX9000 Series, who are negotiating a call between them work a little differently in the sense that the video of presentation of sharing is additional bandwidth beyond what is allocated for the main video session, and so it is not deducted from the Enhanced location CAC.

  More details are given here

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/CUCM/srnd/collab09/CL...

  In addition, as stated in the previous post "bandwidth restrictions put in place by the call control infrastructure will be replace this parameter.", it means that for the endpoints registered with cucm cucm parameter will prevail.

  Manish

• TCS Auto Dial with MCU

  I have 4 points of C20 endpoint that will call in a standing conference that is configured on the MCU 4501.  The customer has also a TCS as we would like to record calls with.

  If I place the TCS to call the microcontroller on a permanent conference it never ends.  All my clients can end it calls but how am I supposed to get the TCS to put an end to his call?

  For now, I tried to settign the Permanent Conference call TCS, but then the TCS records when nothing happens and never ends his appeal.

  Is there a better way to implement?  I don't want to use Adhoc calls and it must be easy for the customer.

  Most of the time the CHT requires a VCS.

  You can try to do only one, the "President" of your Conference permanent.

  Set the Conference to terminate when only participants are present.

  Configure the alias for the TCS as an endpoint.

  Configure a specific permanent conference to call for the specific TCS as a preconfigured participant alias.

  Join the call of the President.

  Join the call as the participant.

  MCU call TCS.

  Chair leaves the meeting.

  Meeting move out.

  While I don't have this Setup myself, we are essentially doing the same thing with an outgoing PSTN, so I don't see why it would be different.

  The TCS is registered to the GA on the MCU as well?

  Why no VCS?

• Integration: Jabber with VCSC

  Hi experts,

  I'm new in the world of telepresence and I will install into a new customer a solution with VCSC, TMS and MCU.

  The customer already has a CUCM with Jabber and they want to deploy the Jabber with VCSC, they want that, for example, a Manager actually a video conference with other offices through your Ipad with Jabber

  I have not found a document with this configuration, can someone help me with the documentation of this integration?

  With their experience of telepresence, including the complexity of this integration? Can give me of highlights and key points of this integration?

  Best regards

  In addition to what Alok suggested, there are other things, you should consider:

  • The version of the recommended software is CallManager 8.6.2 and later (you can try 8.5.1, but there is no support BFCP)
  • It must integrate CallManager and VCS by following this document:

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/VCs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_8_9_and_X7-2.PDF

  • To enable content sharing between registered in CUCM Jabber and the end point registered in VCS, make sure that your Jabber in CUCM devices have a SIP profile with BFCP protocol enabled, as well as the setting "Allow presentation sharing via BFCP" set to allow.
  • If you want to schedule conferences in MSDS for the Jabber in CUCM, you can add them to TMS as a 'room '. By using this option, you will be able to plan the Conference and the calendar control (it's better than using the Participant model)
  • About the phone book, you might have some problems. If your devices registered in VCS use device with an alphanumeric URI address, for example [email protected] / * /, you will not be able to add this contact to Jabber clients registered with CUCM, alphanumeric once is not supported. A solution would use only digital device address for all devices. About Jabber for Telepresence, you can define directories using TMS, in this case, you can add contacts without problem, either using a digital or non-digital address.
  • If you have endpoints another enrolled in CUCM who will call the end points registered in VCS, make sure your devices are running compatible software versions, mainly the videophones. It is strongly recommended to use the latest version of the software.
  • Be aware that Jabber for Telepresence doesn't support Instant Messaging, as the common Jabber support with presence server

  I hope this helps.

  Paulo Souza

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

• Map storage IBM to Esxi 5.1 CIFS host

  Hello

  I am a beginner with vmware, I just want to know is it supported for mapping a CIFS storage ESXi 5.1 IBM host, then use it as a data store for place of the virtual computer.

  Thanks and respect

  J.P. Raynaud

  Welcome to the community - no. it is not taken in charge - if you have the storage NAS/NFS, you will be able to access to I for your ESXi host - or if you want to present the end to VM you can access VM shared the CIF

• Attack of the clones

  (My package head in tape)

  The most progressive thinking of the solution for a system that works on 'Widgets' by interacting with each cell of 'Widget' through 'Gadgets '.  The path, I think to create 'A cell Widget' and drop in sub-panels 1-n.  Easy right? just operator.vi Widget"to reenterant (pre-urechi clones) and the GUI is running all the widgets it is configured to run.  the problem is the resource "Gizmos" will have to be re-enterant too (except Clones) so that each widget provider, associated with a module resources unique gizmo and the gizmos operating in their own space memory sharing between generations appeals to the resources module.

  It's what happens to the clones of gadgets?  Preallocating space for operator of widget creates also a clone of gizmo or all the vi.s of gizmo will share the space unless a larger pool of clone is necessary due to collisions?  That would be BAD.  Or is the right approach as prealocate gizmo.vi but, through any means forcing all calls Gizmo of operator of widget to share the same copy clone?

  I was looking for these references. I thank nathand - I knew that it HAD to be a way to do

  Gizmo.Vit works well as a demo of shell code.

• Telepresence server, control assets drops Conference after 30 seconds

  Much of community support.

  I have the following Setup:

  10.5 (1) version CUCM

  Conductor of telepresence Version XC2.3

  Telepresence Virtual Server Version: 4.0 (2.8)

  EX60 TC 7.1.4

  Cisco 9971, Firmware 1.0000 SR1

  Some other phones (7962, 6941)

  All the end points are saved to CUCM.

  When I do a conference appointment without active active control, everything works fine.

  When I activate the active control (according to ActiveControlDeploymentGuideR3.pdf) the Conference ends after 30 seconds.

  the Conference ends even if I have only one participant at the Conference and any of the phones that I use.

  In CUCM, I configured the relevant parameter in the SIP profile (according to ActiveControlDeploymentGuideR3.pdf)

  on the EX60 I use Active Control, so I see the list of participants and I am able to change the layout of the Conference within these 30 seconds.

  any ideas why this happens?

  the journal of the conductor shows nothing statement

  the telepresence server log looks like this:

  327	14:17:09.170	APP	Info	Conference "8080.MeetMe_mtg' created
  328	14:17:09.264	SIP	Info	Incoming call on 192.168.33.xxx:37124
  329	14:17:09.264	APP	Info	"' Call 24: incoming new SIP call of '[email protected] / * /"
  330	14:17:09.379	APP	Info	Call 24: 'Buero_EX60_Test' has now joined Conference "8080.MeetMe_mtg".
  331	14:17:09.386	IX	Info	Call 24: no encrypted connection to 192.168.xx.xx:31135 for protocols: ping, xccp
  332	14:17:09.814	CMGR	Info	Call 24: ActiveControl negotiated
  333	14:17:46.736	SIP	Error	Call 24: end of appeal because INVITE it transaction timeout
  334	14:17:46.736	CMGR	Info	Call 24: disconnection, '[email protected] / * /"-timeout.
  335	14:17:46.736	APP	Info	Call 24: shoot call them "Buero_EX60_Test" - destroy at request end. timeout
  336	14:17:46.736	SIP	Error	Call 24: BYE transaction failed because the network error
  337	14:17:46.740	SIP	Warning	Cannot determine the destination for the incoming message transaction
  338	14:17:46.741	SIP	Warning	Request received from out-of-call
  339	14:17:48.989	APP	Info	Conference "8080.MeetMe_mtg": deleted via API (not of participants)

  Journal of the conductor, it seems that CUCM closes the connection to RE-INVITE during the transition between the lobby.

  There are several possible reasons, but the best way to determine "why" would be to open a track on CUCM.  More precisely a SDI/SDL (are later versions of CUCM SDL only) track on the CUCM 10.0.2.52 node.  Search for RE-INVITE him entering the IP phone and examine the lines of nearby log for some reason why.

  Alternatively, you can try a few other things:

  #1: Look in Service CUCM settings in the editor, then Cisco Call MAnager to the advanced parameter 'Maximum incoming message size' (there is an Advanced button, you must click above to expose this setting).  If you upgraded your CUCM from an earlier version, this value may need door to 11000 to manage RE-INVITE him to the conductor.

  #2: try disabling the Protocol iX on the SIP, conductor profile and the alias of the meeting of the conductor.

  MP.

• CTS 1.7: Audio snap (call RTC) not heard on all sites

  After the upgrade of our endpoints of 1.6 to 1.7, we met multipoint where we added a PSTN user.

  The audio of the RTC user has not been heard at all the endpoints.

  As soon as we shared an audio presentation was OK on all sites.

  I have a nice SR with that.

  Someone else runs into the same problem?

  See you soon,.

  Martin

  Hi Martin

  BU and DALE were able to reproduce the problem.

  This is incorrect behavior:

  CSCtk61376    CFD: Endpoint MP call RTC does not sure audio component software snap-participant

  Conditions: CTS and CTMS running 1.7.0 software.  Secure Audio component software plug-in point of endpoint is added in secure multipoint call.

  Possible workarounds:

  1 - presentation-sharing is started from endpoint that has made audio snap. Or, Hold/recovery on the endpoint, which makes the audio complement will restore audio to the audio complement in all the endpoints.

  2. - do not use secure audio audio component software plug-in, if possible.

  Default is integrated in version 1.7.1

  Concerning
  Fernando

• BlackBerry smartphone Bold 9700 browser freeze problem

  The browser seems to have frozen when the menu is open. It won't let me use all the options menu and the back button does not work either, the only way to get it off the screen is to press the end key appeal, but as its operation in the background, the battery drains faster. I went to the phone and it's atill even. Any help would be appreciated I searched some topics before posting and couldn't see a similar problem. Thank you.

  Have you tried removing the battery?