Motorway E to call MCU

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• 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

• TMS can generate voice alerts when a scheduled conference ends?

  The Cisco / Codian MCU will provide prompts to the end of the Conference at specific times before the conference end time.

  Unfortunately TMS appears not to propose on the warnings of the screen at the end of the Conference and to schedule the call MCU intentionally to make sure that the MCU cannot generate the voice warnings (conference on the MCU is longer than the time TMS on demand - TMS tears down the Conference at the correct end time).

  Is it possible to configure TMS to generate both the screen and sound alerts - or to generate "real" length, allowing conferences Codian warning messages to be heard?

  Thank you

  No, not for the moment. This should be a feature request.

• SIP spam attack and MCU and vcs - e call

  as far as I know sip call spam attacks is done against the videoconference, connected with a public ip address, I disabled the sip but im not sure if my mcu and vcs - e with sound are vulnerable to them? they pose no threat to security for them? and if so, how? and what can we do about it?

  It is a well known problem and it affects H.323 and SIP, take a look at the below threads:

  https://supportforums.Cisco.com/discussion/12340591/nuisance-h323-calls-SX20

  https://supportforums.Cisco.com/discussion/12336591/sourceh323idcisco-incomingcalls

  https://supportforums.Cisco.com/discussion/12508641/Cisco-source-spam-calls-stepped-complexity

  https://supportforums.Cisco.com/discussion/12613681/attack-vcse

  There are many more discussions on this issue, the above, this is just a small selection. :)

  You do not need to disable SIP on the VCS-E, all you need to do is turn SIP UDP unless you need it for voice services.

  You can protect yourself by using a CPL on the VCS-E who will avoid calls to go through your MCU, or anything else you have sitting behind the VCS-E. This is assuming that you are using a combo of VCS-C/VCS-E, with the VCS - C behind a firewall and the VCS-E outside the firewall, for example in the demilitarized zone.

  Having just trouble ask points of termination or MCU sitting in nature with public IP addresses.

  These scans, moreover, mainly looking for systems that will allow them to make free international calls.

  /Jens

  Please evaluate the answers and makr as 'answered' questions as appropriate.

• BFCP of MCU for encrypted calls

  Hello

  I know that it has been discussed before on the CUCM especially earlier this year:

  https://supportforums.Cisco.com/message/4001153#4001153

  We have a series using VCS (7.2), conductor (2.2) and MCU (4.4) where we need to content SIP on encrypted calls for dual display using BFCP systems, because the driver does NOT.  It is not acceptable to disable encryption on the MCU for all types of calls as a workaround.

  The 4.4 release notes:

  Binary control protocol floor on encrypted calls:

  The transmission of content SIP of the MCU by using the binary protocol (BFCP) ground control is not supported on encrypted calls. To allow content to pass on SIP calls in a separate main video channel, you must disable encryption on the MCU or target endpoint.

  Is anyone aware of a fix for this?

  Thank you very much

  Trevyn

  I think that the objective of this implementation will be 4.5.  The date from which this version will be public is (interim) August, but the SW not EFT, so you may register and test if you wish.

  VR

  Patrick

• Is it possible to join/merge two conference calls in MCU for one call in the same MCU

  Situation - endpoint is part of two meetings of MCU (with one call on hold).  I want to merge these two meetings to the MCU.  Is there a way.  MCU is 45xx series and this is a deployment of VCS - MCU.  Thanks in advance.

  No way to merge that I know, I've had similar scenarios and there is no automatic way to do it, the Conference, the participants will have to unplug one and join the other.  Multiway will not work because that is intended to the unique endpoints being transmitted to a MCU, no MCU is not lift themselves.

• External Audio calls to Cisco MCU

  I have a requirement for an external audio channel to access the shared meeting room that is hosted on the MCU 5320 Cisco.

  This can be achieved directly through the MCU or is it a requirment for another audio bridge

  Well, you can integrated MCU with VCS using SIP Trunk, even if the common deployment is to have registered in VCS, MCU or by using SIP, H323, or both. You can also integrate the MCU with CUCM, but in this case, MCU will be a multimedia resource of CUCM, thats why it is only recommended to the ad hoc conferences. You can also integrate the MCU with the driver of telepresence using the API, but it is often only when you have several MCU pools to manage.

  To integrate the Cisco MCU with VCS, you can consult this document:

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/MCU/Install_Guide/Cisco_TelePresence_MCU_Deployment_guide_1-31.PDF

  To learn more about how to work with auxiliary self on Cisco MCU, take a look at this guide:

  Page 88 - http://www.cisco.com/en/US/docs/telepresence/infrastructure/mcu/admin_guide/Cisco_Telepresence_MCU_5300_Series_4-4_Product_administration_guide.pdf

  If you want to integrate MCU with control systems third call, I can't say that it will work, because I still have my suggestions on basis Cisco official documentation, and there is little in the literature on integration of MCU with control systems third call.

  If you think that your question has been resolved, please mark the right answer.  =)

  Concerning

  Paulo Souza

  Please note the answers and mark it as "answered" as appropriate.

• CUCM mcu siptrunk video call bindwith

  Hi team

  I have a question about how can I config the bindwith for the siptrunk?

  I used a tunk sip to leave the cucm forward the video to the mcu conference call (5310), but no matter how I set up the bindwith video of location in cucm profile, call rates transmit only 384 k, but the receive call rate is 4000 k, but the hit rate and resolution are well when the mcu call codec (C60) directly by using H.323.

  I think I missed something in cucm configuration.

  Hi Jason,

  Could you give more information about the configuration of CUCM? How your regions are configured? Do you have the regions applied to SIP Trunk and the phone? The relationship between the regions allow enough bandwidth for the video?

  If I'm not mistaken, the default bandwidth for the video within the region setting is 384 k. Did you change this value to your regions?

  I also suggest to examine the configuration of your location and check if there is any restrictions applied.

  Concerning

  Paulo Souza

  Please note the answers and mark it as "answered" as appropriate.

• VCS-E calls directly to Codian MCU Auto standard

  How do I create rules of research on my E - VCS to route direct calls to its IP address or hostname to go in-house auto attendant Codian MCU--> my autoatendant has a 31 conference ID and we have 99 registered in my inner - VCS - C so 9931).  We had so many difficulties with external companies, connection using several addressing methods (VCS - eIP ##MCUmeetingID, or [email protected] / * /, whatever).  It is much easier to have external users, type an IP address and be dumped directly in the auto attendant and we can move everywhere where from there.  I don't know everything that the Alias Search and regex and replace the criteria and do not want to cause problems with external courses and Movi SIP calls that we use.  Need help.

  Hi, Frank.  My apologies.  X5.1.1, its under VCS Configuration > search rules > Configuration.  Should be the help alias section here.  Let me know how it turns out.

  VR

  Patrick

• A button to push causes endpoint to call twice the MCU

  Hi, I have a problem with the TMS conferences that use the type of button to push a connection. When the ends are the meeting quickly reach, I select join and end points, then dial conference number twice at the same time. This starts a video call and an audio call at the same time, causing an echo. If you hang up the audio call the Conference continues normally.

  Can someone shed light on how the meeting gets setup of TMS or advise what opens a session for further investigation.

  I have v8.6.2 CUCM, VCS - c v7.2.2, TMS v14.3.2 and MCU 5320 v4.4. All the end points are saved to CUCM. Manual and automatic calls all work well.

  Thanks in advance

  Hi Ian,

  Above problem is a known, documented bug in CSCul80930.

  This has been resolved in version 7.1 of TC. You can go to TC 7.1.1 version on your codec.

  http://software.Cisco.com/download/release.html?mdfid=283644997&flowid=2...

  Kind regards

  Amit

• Allowing endpoints make multi-point on Cisco 4505 MCU calls

  Hello

  I have a Cisco 4505 MCU and several endpoints.  Is there a way to allow that my end point called multipoint adhoc using MCU 4505?  For example, I would say that dial site1 using mcu, then dial site2 help mcu or call a group (site 1 and site1...) using mcu and it would know that it is supposed to use the 4505.

  Thank you

  Ravi

  Of course, you can configure (a few 100) static, meeting rooms, auto create conferences on the fly, based on a model or have the means put in place and there is even an on screen menu allows users to create or join conferences.

  Check out the example (page 37ff) Administrator's guide

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/MCU/admin_guide/Cisco_TelePresence_MCU_4-4_Product_administration_guide.PDF

• Motorway E v8.5.1 / / URI Source of change for a call out

  I'm changing the source URI for the calls from our internal network to the internet.  We have an internal area that is not routable on the internet and I would like to change for the routable domain.

  Example:

  Internal domain = int.company.com

  External domain = societe.com

  I tried to use a CPL script on highway E (Version 8.5.1) to achieve, but it does not work for some reason any.  Here's the last CPL I am using:

  
  
  "xmlns:TAA ="http://www.tandberg.net/cpl-extensions"
  "" xmlns: xsi = "http://www.w3.org/2001/XMLSchema-instance"
  xsi: schemaLocation = "urn: ietf:params:xml:ns:cpl cpl.xsd" >
        
          
            
               "[email protected] / * / ' >
                
              
            
          
        
  

  Someone has any ideas why this wouldn't work?

  Thanks in advance.

  Brandon

  Take a look at the example in this thread for reference, ask-expert-cisco-telepresence-company, and CPL reference material in the Guide Admin Expressway.

  You must change your origin depending on your internal domain, rather than match any incoming/outgoing call, see below.

  [email protected]/*  */" destination=".*">

  Everything else is fine, I tested it with a codec here using your example domains and the called endpoint has shown the correct source address. You have a cpl in place to change the external domain in-house for incoming calls from outside your organization to be able to find the domain that is used for your registered endpoints? Joint a CPL example is based on one that you posted where it will change the outgoing calls to use the external domain and use the internal domain, incoming calls. In addition, for use later, just say "it doesn't work" provides no information of what's going on. Sometimes you can get lucky and see the issue, but other times that we have more detailed information or see what is happening in order to help better understand where is the problem.

• Can we call a point of endpoint not registered by IP without MCU

  Hello

  We have several items endpoint with CUCM 9.5. We also have a legacy Tandberg, probably aged 7 years located on one of our sites WAN (connected via leased line) that we want to call and receive calls through IP. We do not have any microcontroller or other products of voice, only CUCM.

  I've been looking for examples, but since I am brand new to this subject I'd like help experienced forum members. From what I understood, to call an end point recorded since that not saved, we have to deal the [email protected] / * /, is - that correct? What is dial the IP of the endpoint not inherited of the registered endpoint?

  Any help is appreciated.

  Concerning

  MOE Shea

  You need Expressway-C/E to call endpoint.  CUCM does not support the IP numbering, but using a channel Express-C/E, you can see the Dial IP addresses for endpoints registered to CUCM VCS / example Configuration of the highway.

• 2 MCU and VCS Expressway, routing problem

  Hi all

  We have a design with a group of control VCS (2 members), cluster VCS Expressway (2 members), and a couple of microcontrollers (registred H.323 on VCS control cluster with the same prefix: 90).

  Each highway has a public IP address and incoming calls from outside can only be routed to microcontrollers:

  [email protected]/ * / for MCU_1 and [email protected]/ * / for MCU_2 (we don't have external DNS resolution).

  I put a conversion into motorway of VCS to change [email protected]/ * / to [email protected] / * / and [email protected]/ * / to [email protected] / * /.

  The problem is when someone calls [email protected]/ * / sometimes (randomly) the call is routed to MCU_2 (instead of MCU_1) and if the appellant see the auto attendant.

  The occcurs even then of the appeal [email protected]/ * / (MCU_2), sometimes the call is routed to MCU_1.

  Any idea what can cause this device or a way to make it work well?

  Thanks for help.

  José

  I think still that separate prefix would work, but here are a few ideas:

  Are incoming calls which do not possibly using SIP that is being interoperability H323 or incoming calls all certainly the H323?

  If you want to keep pure H323, you could perhaps just have a search rule/turn on your VCS-E who changed [email protected] to an E164 e.g. 90... and had a search on your VCS - C rule that says 90... stop at the local area.

  I also noticed that you direct calls to [number]@MCU-IP - have you tried to direct all calls to [number]@VCS-C IP instead?  If the VCS is the holder of a registration for a number, it should be able to deliver accordingly.

• [Issue] it is possbile to use MCU 4501 on network 1:1 NAT?

  Hi, I have a very simple question.

  I'll put MCU 4501 model on a private network to connect to public IP codec devices.

  Without devices INCAPACITANTS, Turn, or VCS, it works standalone with H323 and its private IP address is set to 1:1 public IP address.

  In this position, is that it is possible another codec public IP can join the session of MCU 4501?

  In my view, would be well accepted, but given media cannot be accepted because the IP address in the payload would be signaling missmatching.

  No one knows about it? or can I use some free public STUN, ENABLE Server H323 Direct mode? or do it by myself?

  I have no senior engineer and I have to learn about it, everybody ask me when they stuck on the frame.

  This site is the last hope for me, and I'm sorry for imconvient read, it is not my mother tongue.

  Thank you.

  For H.323, IP address of the port media negotiate during the call setup is not using the IP header.

  If your compatible Firewall H.323 AGL, handful of firewall NAT IP and H.323 signal header conversion.

  However for MCU, you can consider the firewall on MCU option if you are looking for private/public double network connection.