Issue of outside calling on Cisco MX200G2

Similar Questions

• How to set up a '0' prior to call for outside calls (CCM 6.2.1)

  Community salvation.

  I think that this issue could be very easy to be decided by the experts of the calls voice cisco. What are the steps required to configure all the internal components of the phones that they must dial '0' to get an outside/international appeal of prior work? Is it a matter of templates for itinerary?

  Thanks in advance.

  Kind regards.

  Hello

  Yes, it is a RP "template routing." I hope to share with us your topology what your CUCM & gateway type (MGCP & H323). In any case regarding steps e, please find the below: -.

  1-configuration of your gateway that will be responsible for tree calls.on incoming and outgoing CUCM gateway device - select type.

  RG 2-group to configure or channel that will include your gateway recorded. CUCM call routing-road/hunting-lines group and assign your gateway listed

  3 - RL set which includes your RG that configured in step 2. Routing-road/hunting-other calls list, then assign the RG.

  3-model RP to configure or channel. for example

  0.161xxxxxxx for mobile

  0.00!                 for international

  At the partiton award

  B - RL that configured in step 3

  C - on the bottom, discard numbers "predot."

  Test; user dials a number that fits the RPS following 0.161xxxxxxx & 0.00!  the calls will go.

  Thank you

  Please note all relevant information

• Cannot receive an outside call

  Hi all!

  We have such a problem. We have an infrastructure of control VCS, VCS Expressway and MCU. VCS Expressway has dual NIC option enabled; 1 LAN has an internal IP, LAN 2 - public IP address (without NAT).

  Two VCS are configured according to manual)

  ( http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Basic_Configuration_Control_with_Expressway_Deployment_Guide_X7-2.pdf), with the exception of the settings NAT and firewall - we do not have.

  When we call for our network internal (for example [email protected] / * /) to the Organization of foreigners ([email protected] / * /)-everything is ok. But we cannot receive calls from outside.

  An external DNS is also configured according to manual, domain name of VCE Expressway is external can be solved.

  You have any ideas what may be wrong? Thanks in advance.

  What exactly do you see in the newspaper of the network? Can paste you it here? (Be sure to hide sensitive information such as addresses IP, etc.).

  It does not appear that you have created the SRV records required, such as

  _h323ls._udp.

  _h323cs._tcp.

  _sip._tcp.

  _sips._tcp.

  at least I'm not able to look up one of these SRV from here.

  -Andreas

• Randomly call in Cisco TelePresence SX80.

  I have a recurring problem of many unknown incoming calls in a Cisco SX80 recently deployed. An example of the address that we get these calls from the East [email protected]/ * / I have no idea what is the part of the address in 1986, but the 129.59 address is the IP address of the codec. I am somewhat a novice in the world of VTC, so I am at a loss on how to stop the calls. Automatic response has been disabled pending just so that the device can be used.

  I assume that the system uses an accessible public IP address?  If it uses a public IP address, you should put behind your firewall with NAT, or allow only traffic from known IP addresses.  If you don't use a SIP, you can enable turn off completely, however if you are, you can disable SIP ListenPort and activate outgoing SIP so to avoid these calls, see bug CSCue55239.  There is a form any of these calls which will appear as coming from "cisco", the only way to prevent these is to put endpoint behind your firewall and use NAT or an ACL.

  You can also watch here for a bit more information, anyone-know-why-edge-95-mxp-will-call-itself, as well as further discussions in the forums associated with these types of calls.

• Transfer calls on Cisco's Finesse

  I am to evaluate the web client of Cisco's Finesse.

  Need help how to transfer a call to another agent that I can't not know where and how.

  I have a button to view and set up an address book using Finesse administrator it works. See attachment.

  But how do I do a Direct transfer (blind) to another agent? Is it possible to transfer the call to another agent without having to add entries in the address book for this?

  Also, is there anyway to see the call history (who called and when)? I can only see changes of States agent, which adds a lot of value for the agent.

  Best regards
  Björn

  Unfortunately, it is not available with fine out of the box. Check below:

  http://www.Cisco.com/c/en/us/products/collateral/customer-collaboration/...

  Also, is there anyway to see the call history (who called and when)? I can only see changes of States agent, which adds a lot of value for the agent.

  Yes, it is possible. Basically, you'll need create a copy of one of the necessary historical reports such as detailed call by call CCDR report etc. and then view it inside the finesse. See Add custom Live Data Gadgets to Desktop Layout section of document on how to do this:

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/cust_contact/contact _...

  Concerning

  Deepak

• Unit outside call connecting you transfer?

  I have a problem with the implementation of the Appeals Unit to dial an outside number. In this case, these calls are malicious and I need to know where they are from. Which record turn on to see what subscriber or handler call these calls come from?

  Thank you

  You can also use the "Call for transfer of billing" report that you will find under the Subscriber on the SA reports - you can run it against all subscribers and then again on all call them managers - if you know the number, you can find it here.

  Also, if you know the number, you can find it in the database with a simple query - Audiotext Manager (a tool found in the depot of tools) allows you to find a transfer station in the system that matches all or part of the dial string flow down what Manager/Subscriber is originally from the call.

• 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.

• Call TCS Cisco fails unexpectedly

  Hello

  I am facing a problem with the TCS, just make the call to TCS (point to point and call for the application of the MCU) after a specific time. Once that the appeal has not I can't see any videos stored in TCS for this problematic call as well.

  If it is an MCU at the request of appeal, only the TCS call leg will get fail other calls will remain as even. If I make a direct call to TCS alias after a point in time, it will fail.

  I use TC V5.1

  One of you faced a problem like this with TCS.

  Thank you best regards &,.

  Nikhil Jayan

  It could be that the engine of the content is blocked so it is impossible to treat the incoming stream of RTP. Hat happens then in a call of the CHT will claim that his record for 1 minute, then the call disconnects (no support received). This is normal on the TCS to disconnect the call if no support is received in a minute. Maybe a reboot of TCS services may solve the problem.

  / Magnus

  Sent by Cisco Support technique iPhone App

• issue of the call control feature

  I am running an application with an unidentified Developer (LuxRender), triggered by another application by a developer identified (mixer)

  Both applications are produced executable 'stand alone '.

  If I perform an action in a mixer called LuxRender there is no problem with the launcher application and do this is special, and in certain situations, I need this app run stand-alone.

  If I call LuxRender from the finder, it gives me an error that it is an unidentified developer. From the doorman ( > System Preferences > Security & private/General life) by clicking Open anyway I can't open the application, unless I drop my safety for the 'anywhere' level.

  The fact that I can get around this with another application, what is not somehow try to outsmart security is relative to me. The question I have is: is this behavior of design?

  

  

  GateKeeper settings apply only when you start an application from the Finder * first *.

  If an application does not start because it is not signed, then instead of double-click you can press the control key and choose open - then a similar message but instead of just an OK button, you will have two buttons, one that says "Open". Click on it to launch the application. After that you can start normally.

• Issue of phone call from PBX

  Hi all

  I want to implement is the request for phone call PBX.

  I don't have this idea because my company has no PBX system.

  I just googled and got an idea.

  I need to install MVS on my handheld blackbery. and

  Then, create the application to call a phone call.

  Now, the question is:

  is - this simple RIM call Invoke. invokeApplication (APP_TYPE_PHONE, new PhoneArguments(ARG_CALL, 'Phone PBX' + 'extension'));

  to connect with a coporation PBX phone system?

  Any information would be great interacted.

  Thanks in advance.

  It is possible to write extensions PBX with the BlackBerry Mobile Voice System (MVS).

  When creating a call from a third party the application, the call is made from the phone line to the default user (MVS/PBX line or cell).  This is configurable by the user in the phone app.  Currently has no means for an application to specify the telephone line to use when you start a call.

• Issue of Posture 1.3 Cisco ISE.

  Hi all

  Small issue: assessment of posture for ISE customer how you configure ISE to use several AV vendors?

  Example: I have configured ISE posture rating for Sophos AV definitions that works well, if I have introduced another provider of antivirus for the posture of the client assessment does not pass compliance because she's trying the two AV vendors? How do you ensure ISE verifies and marks as consistent customer for each suite AV?

  Clients use AnyConnect V4.

  Thanks in advance for any response.

  Actually, after checking I don't think you can use two terms consisting of the VA, in another State, so you will need to create it using both your requirements in the same requirements, and then select "any condition to succeed", only problem is that you can have as a repair action, which can be a problem.

• Interface issues Netgear Smartswitch to the Cisco 881 LAN port

  Hi, we have 100 routers Cisco 881 in our network and they work all fine for the Linksys, 3Com, switches etc.  The problem we encountered is interfacing to switches from Netgear.  Netgear switches use automatic detection on their ports and it doesn't seem to be compatible with auto MDIX detection on the LAN Cisco 881 4 serial ports on the router 881 hub.  Someone has encountered this problem before?  A cross over cable solve the problem?  Since both executed MDIX autosensing they never synch - so probably a cross on would not make much.  I see this with all Netgear smartswitches.  If you put a small switch between the Netgear switch and router Cisco 881 everything works well except to pass traffic to port 9000.  Any ideas would be appreciated.

  See you soon,.

  Len

  Hello

  There should be no problem using crossover cable. You can try disabling autoMDI/MDIX (not auto mdix) on the cisco device and keep a straight cable but if it fails, use a crossover cable.

  Concerning

  Alain

  Remember messages useful rate.

• Cisco SX 20 past question call composition

  Hello team,

  Hello team,

  We use Cisco SX20 and configured the public Ip address,.

  We can receive the call for outside call but not in outgoing call, I tried several public IP address for the test, but none of them does not.

  There is no guardian, address Public Ip directly to the configuration system.

  Please suggest the same

  Thanks in advance

  Are Set - up mode set to 'Direct' call and you specify H.323 call Protocol?

  /Jens

  Please note the answers and score the questions as "answered" as appropriate.

• RMS Expressway Cisco licenses for B2B calls

  Hi Experts,

  We have Cisco Expressway Setup for B2B calls. We have 6 licenses RMS across Highway C & E, making 3 on EXP E and 3 on EXP C.

  Now, I'm able to do 3 Conference of the party including the host. Myself and 2 other B2B calls for a video conference. Where that we watched for 3 with the exception of the host Party Conference.

  We were under the impression that if we buy 3 licenses RMS, we have 4 video party conference. It's 3 outside calls excluding the host.

  Can someone explain how the RMS license.

  Our IPT put in place:

  CUCM 10.5

  SX20 - recorded at CUCM

  Motorway C - containing 3 licenses RMS

  Highway E - containing 3 licenses RMS

  Concerning

  Benyoucef

  Hi, Renaud,.

  Licenses RMS are concurrent, so in your case both licenses RMS (one in each Expressway) per call are necessary. You can check the following positions related to similar queries

  https://supportforums.Cisco.com/discussion/12444331/Jabber-guest-require...

  https://supportforums.Cisco.com/discussion/12522961/RMS-license-Cisco-ex...

  https://supportforums.Cisco.com/discussion/12240916/Expressway-LIC-exp-r...

  Manish

  -Does the rate of useful messages-

• I have a nettalk duo and I can not calls to the outside through an arris 16db modom

  I have a nettalk duo and I can't just outside calls incoming calls only y at - it a problrm

  try to support Net Talk Duo

  http://www.NetTalk.com/en/support