Warning: Undefined array key "kkmods_kkpostslider" in /home/clients/bb1b7c38f63c1b363aa0114be7911e68/web/collaboration/wp-content/plugins/kk-divi-mods/kk_divi_mods.php on line 67
CUCM | Cisco Collaboration

UCCX Check / Reminder

Here are some stuff regarding the UCCX .

First of all , please check that under the menu Subsystems>Cisco Media , you have well CTI port defined. It is depending on your license.

For reminder , the JTAPI User is the user which will control the CTI Route Point and the CTI Ports ( Everything must be populated via the appadmin).

Jtapi User

The RmCM user is the user which must control the agent extension and the supervisors. Don’t forget also to define under your agent/supervisor agent the primary line and the IPCC extension.

RmCm User

Define then the Telephony Call Control Group that you will use to define the amount of CTI ports that you will use for your script/applications. Don’t forget to click on the Show More button in order to configure correctly your ports

Call Control Group Definition

In your application script , the Maximum Number of Sessions has to match the CTI port that you have defined in your call control group. If it is not the case, then you can encounter the conditions where you will hit the default script message so the message “Sorry  we are experiencing ……”.

Script Application Definition

SNR – Side Effect

Hi ,

Take care to the fact that any change under Remote Destination Configuration will disable the Mobility. So in order to test it , don’t forget to enable it again if you need to test it .

CTI Route Points and Ports – Partition and CSS Approach

Pay attention that normally the order of the partitionins is always L1,L2,L3, D1,D2,D3 , where Lx are the partitions assosiated to the Line model and Dx are the partitions associated to the Device model.

So this order will give you the order/ priority in case of tie-break !

In the case of CTI Route Point or CTI ports , the order is changed by the following D1,D2,D3,L1,L2,L3 …. Please recall this if you are involved in a troubleshooting with these elements !

Call Manager Services

Here is the list with all the Call Manager services:

  • Call Manager : Provides call processing
  • TFTP : responsible for building and serving files for devices
  • Messaging Interface : used with voice mail systems which use SMDI
  • IP Voice Media Streaming Application: provides functionality underlying MOH, conferences, MTP, …
  • Telephony Call Dispatcher (TCD) : provides hunting group with Auto-Attendant console
  • CTL Provider : works with CTL Client to change security mode from nonsecure to secure
  • MOH Audio Translator : it converts audio files for MOH utilisation
  • RIS Data Collector : collects and distributes real-time information
  • Extension Mobility : login/logout features
  • CDR Insert : is responsible for CDR operations
  • IPMA : Manager/assistant service
  • Extended functions : call-back and QRT service
  • Serviceability Reporter : generates reports on logged information
  • Webdialer : allows web application to make call control
  • CAPF : Issues locally Significant Certificates (LSC)

Protecting Voice Media and Signaling Traffic

There is two types of security that you can enable with Call Manager

  • Mixed mode : In this mode, depending the security configured on each phone, you can have secure calls when both devices are security-enabled and when one of the phones is missing security, your call will be nonsecure.
  • Nonsecure mode : As all phones are not set up with security (default configuration), all calls are nonsecure.

When you device to put security on phones , they can support the three following levels:

  • Nonsecure mode : secure calls are not supported
  • Authenticated mode : the phone will be able to authenticate calls
  • Encrypted mode : the phone will be able to support encrypted calls

If you enable the authentication and the encrytion on your network , you are then able to secure the media traffic as well the voice signaling.

If you want to have security on the media flow, it is then mandatory to secure also the signaling as the keys which are used to secure the media traffic are exchanged during the signaling phase.

SCCP messages sent by IP Phones and Call Manager can be secured using TLS, it is the signaling part. Then for the protection of the media traffic so the RTP packets , you will use the Secure RTP which is providing a framework for encryption and authentications of your stream.

SRTP will be also use between your MGCP gateway and your IP Phone but you need to know that your SRTP keys are exchanged in cleartext session between the MGCP gateway and the Call Manager.