Introduction
This document describes how to configure, basic routing rules to allow incoming and outgoing calls on Cisco Meeting Server (CMS) version 3.0.
Contributed by Jefferson Madriz and Octavio Miralrio, Cisco TAC Engineers.
Prerequisites
Requirements
Cisco recommends that you have knowledge of these topics:
- CMS core services
- Session Initiation Protocol (SIP) basis
Components Used
The information in this document is based on CMS version 3.0.
The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, ensure that you understand the potential impact of any command.
Configuration
Basic call settings
- Log in to the Webadmin interface with a browser.
- Navigate to Configuration > Call settings.
- Select the appropriate SIP media encryption setting (allowed, required or disabled).
- On the same page, you can optionally:
- Choose to enable SIP call participant labels if you want participant names to display overlaid on video images.
- Customize the maximum bandwidth per call to use for the different call types.
- Select Submit.

Incoming call rules
The Incoming calls page determines how the Meeting Server handles incoming SIP calls. Any call routed to the Meeting Server contains the called alias, checked against the rules in the Call matching table to determine where Meeting Server looks for potential matches.
Note: Rules with a higher priority value are matched first. In cases where multiple rules have the same priority, match occurs based on alphabetical order of the domain.
- Log in to the Webadmin interface with a browser.
- Navigate to Configuration > Incoming calls.
- Configure the highest priority incoming rule to be the SIP domain you use for spaces.
- Use the empty row to add a rule with the next values:
Domain name: <your SIP domain for Meeting Server>(for example, mxc.lab)
Priority: 100
Target spaces, users, IVRs: set to yes
Select Add New to save the changes.
- To ensure compatibility with different trunk configurations, add a rule for the Fully Qualified Domain Name (FQDN) of your Meeting Server.
Domain name: (for example, meetingserver.mxc.lab)
Priority: 90
Target spaces, users, IVRs: set to yes
Click Add New to save the changes.
- To ensure compatibility with different trunk configurations, add a rule for the IP address of your Meeting Server.
Domain name: <IP address of interface of where Call Bridge is listening>
Priority: 90
Target spaces, users, IVRs: set to yes
Click Add New to save the changes.

Outgoing call rules
To make calls out from Meeting Server, calls must be directed via the Outbound calls rules to a destination, such as Unified Call Manager (UCM) or Expressway. Rules are processed from highest priority to lowest, and if matched, Meeting Server attempts to send the call to the SIP proxy defined.
In order to route all outbound calls to a singular call control (Unified CM or Expressway) follow the next steps:
- Log in to the Webadmin Interface with a browser.
- Navigate to Configuration > Outbound calls.
- Create a new outbound rule with the next values:
Domain name: [Leave blank. Note that this is a special use that allows to match all domains]
SIP Proxy to use: Enter the FQDN of your Unified CM or Expressway call control node (IP Address can be used, but FQDN is recommended)
Local contact domain: [Leave blank]
Local from domain: Enter your SIP domain for Meeting Server (for example: meet.mxc.lab)
Trunk type: Standard SIP
Behavior: Continue
Priority: 1
Encryption: Auto
- Select Add New to save the changes.

By this point Expressway must already have a Zone created, and proper search rule to route call to Space on CMS. More details about Expressway configuration can be found on link below, from page 20:
Configuring Call Control to route to the Meeting Server
Verify
Create a test space.
- Log in to the Web Admin Interface with a browser.
- Navigate to Configuration > Spaces.
- Use an empty row to create a new space.
- Enter the next values:
Name: Test Meeting
URI user part: test
Call ID: 5001
- Select Add New to save the new values.

- Start a call from an Enpoint registered on a SIP server that is able to route calls to CMS
- Validate the call is connected correctly