Cisco Cisco Unified MeetingPlace Audio Server Quick Setup Guide

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Configuring Call Control for Cisco Unified MeetingPlace
How to Configure Call Control for Voice Conferencing
4
 
Step 8
Configure all other required fields appropriately for your current deployment.
If you configured a Calling Search Space to block unwanted dial-out calls, then apply the Calling Search 
Space accordingly to the SIP trunk.
Tip
For field descriptions, select Help > This Page.
Step 9
Select Save.
Table 1
Fields for Adding a SIP Trunk in Cisco Unified Communications Manager 6.x or a Later Release 
Field
Action
Device Name
Enter a unique identifier for this trunk, such as the name or IP address of the 
Cisco Unified MeetingPlace server.
Device Pool
AAR Group
The device pool must use a codec that is compatible with the conferencing gateway 
(or bridge).
For security and toll fraud prevention, use a device pool and an automatic alternate 
routing (AAR) group that will block any undesired phone numbers from being 
dialed out.
Media Resource Group List
(Optional) If Cisco Unified MeetingPlace–supported endpoints are registered to 
this Cisco Unified Communications Manager, then we recommend that you 
choose one of the following to prevent conference calls from being disrupted by 
music whenever a user places a call on hold:
  •
Default value of <None>.
  •
A Media Resource Group List that does not contain music on hold resources. 
Note
See 
 in the 
 module.
Media Termination Point Required
Uncheck this check box.
Destination Address
The DNS hostname or IP address of the Cisco Unified MeetingPlace Application 
Server.
In an 
 deployment, make sure you enter the shared 
hostname and IP address of eth0. 
Destination Port
Keep the default value of 5060.
SIP Trunk Security Profile
Select the SIP trunk security profile that you created specifically for Cisco Unified 
MeetingPlace. 
If you did not create a SIP trunk security profile, then select the default Non Secure 
SIP Trunk Profile
.
Rerouting Calling Search Space
Make sure you set this value appropriately to ensure that call transfers (out to 
attendant or other systems) are successful. Consult your Communications Manager 
administrator for the appropriate CSS to use.
DTMF Signaling Method
Select No Preference.