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Cisco Unified Communications Manager (CallManager)

Unified CallManager Interactive Voice Network Configuration and Troubleshooting Case Study

Document ID: 82014



Introduction

This series of interactive documents uses a typical head office and branch office in order to demonstrate some common voice network configurations. This interactive diagram allows you to view network configurations:

Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting.
Click to view configurations and troubleshooting. Click to view configurations and troubleshooting. Click to view configurations and troubleshooting.
This image is interactive!
Click a component to view configuration
and troubleshooting information.

See Requirements for requirements assumed by this series of interactive documents.

Components Used

This series of documents is based on these software and hardware versions:

  • Cisco Unified CallManager 4.1(3)sr3a
  • Cisco 3725 Router that runs Cisco IOS 12.4.(3a)
  • Cisco Unity 4.0(5)

Note: In addition to these components, the information in this series of documents can be used with other Cisco routers, such as Cisco 2800 and Cisco 3800 Series Routers.

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, make sure that you understand the potential impact of any command.

Requirements

This series of interactive documents assumes the requirements described in this table.

Resource
Description
Dial Plan The dial plan consists of Direct-Inward-Dial (DID) (408) 501-5000 at the head office and (919) 501-4000 at the branch office. At each office, some phones include restricted dialing capabilities, such as internal and 911 calls only, and other phone include unlimited dialing capabilities, such as international calls.

For more information, see Introduction or refer to Voice Dial Plan Interactive Voice Network Configuration Example.
Device Pools This network uses two device pools: the default device pool and an RTP device pool that was created manually.

For more information, see Introduction or refer to Device Pools Interactive Voice Network Configuration Example.
Regions This network uses two regions: the default region and a remote region that was created manually.

For more information, see Introduction or refer to Regions Interactive Voice Network Configuration Example.
Phones

For information about phone configurations, see Introduction or refer to these documents:

Extension Mobility All users are allowed to log into any phone with Extension Mobility. When a user from the San Jose office logs into a phone in RTP, the device profile must be configured to allow local gateway access in RTP.

For more information, see Introduction or refer to Unified CallManager Extension Mobility Interactive Voice Network Configuration Example.
IP Manager Assistant

Users in San Jose have the IPMA with proxy line feature enabled on their phones.

For more information, see Introduction or refer to IPMA with Proxy Line Support Interactive Voice Network Configuration Example.

Media Resources

Users in San Jose and RTP use media resources that are configured locally. Multicast Music-On-Hold is used across the WAN.

For more information, see Introduction or refer to Media Resources Interactive Voice Network Configuration Example.

SRST

RTP configures SRST on the voice gateway to ensure RTP users can access the PSTN and users in San Jose in case of a frame relay failure.

For more information, see Introduction or refer to Unified SRST Interactive Voice Network Configuration Example.

Location-Based Call Admission Control

Location-based CAC is configured between the San Jose office and the RTP branch, which allows two G.729 calls between the two offices.

For more information, see Introduction or refer to Call Admission Control Interactive Voice Network Configuration Example.

QoS

Quality of service is implemented on the network in order to ensure that voice traffic is not dropped during periods of high congestion.

For more information, see Introduction or refer to Quality of Service Interactive Voice Network Configuration Example.

PSTN Access Gateways

MGCP gateways are used at the San Jose and RTP sites. In RTP, SRST failover is implemented.

For more information, see Introduction or refer to these documents:

Voice Mail All users have voice mail access. Voice mail must be available during automated alternate routing (AAR) and survivable remote site telephony (SRST).

For more information, see Introduction or refer to these documents:



Updated: Apr 16, 2007Document ID: 82014