Voice Foundations for Cisco Collaboration (VFCC) is designed for engineers or administrators who are: new to voice but experienced with data or; experienced in voice but new to Cisco Voice and; need a fundamental knowledge of Cisco Voice architecture solutions used in typical Voice Collaboration environments.
In other words, if you only have 1 week of training to get up to speed on Voice Gateways and Cisco Unified Communications Manager, this class is for you.
The focus of this course is the implementation of the ’Dial Plan’ across the Voice Gateway and CUCM environment. Emphasis will be placed on Call Routing within the UC environment. This course is a lab-based course with lots of hands-on time spent working on the equipment; our goal is to make you a better administrator by making you a better engineer.
The first day will examine and configure the Unified Communications Manager. The next several days will be spent configuring the basic infrastructure of a telephony environment including Voice Gateways, Session Border Controllers, SIP Proxy Server, and Unified Communications Manager. By mid-week, students will have basic Inbound/Outbound dialing functionality configured for back-office users. The final days of this course will cover additional Dial-Plan related items including; Number translation (Translation Rules, Patterns), Telephony Class of Service (CSS/Partitions). Along the way, we’ll install and use some of the support utilities used within the environment including the Dialed Number Analyzer (DNA) and the Real-Time Monitoring Tool (RTMT). By the end of this one-week class, the student will be better prepared to support the basic telephony environment used in most Collaboration environments.
VFCC is also an excellent pre-requisite to attending an advanced Contact Center Course (UCCX, DUCCE, and AUCCE 1 & 2, etc.).
Students will be better prepared for advanced courses by taking VFCC and will learn more during their time in the advanced Contact Center courses if they first attend a VFCC course or have equivalent experience. This course will address the fundamentals of the Cisco Voice infrastructure including; Voice Gateways, Cisco Unified Communications Manager (CUCM) as well as the Call Signaling protocols and components used between these components (TDM, SIP, H.323, MGCP). The components and protocols discussed in this course are common to both the Contact Center Express and Enterprise environments. The goal of this course is to focus on the myriad of Trunk and Lineside connections that will be used in the Contact Center environment – Express or Enterprise.
- Working data and/or voice background. In this course, the assumption is that you have experience in either the ’data world’ or the ’telephony world’, and are now being asked to gain knowledge on ’Cisco Unified Communications’, which combines both worlds.
- ICND is a highly recommended prerequisite for this course if you are new to the data world
This course is intended for anyone supporting the dial-plan across any of the basic telephony components of the Cisco Collaboration environments including Voice Gateways, SIP Proxy, Gatekeeper and Unified Communications Manager. It is intended for engineers who are:
- New to Voice, but not new to Data or;
- Not new to Voice, but new to Cisco Voice and;
- Need a fundamental knowledge of basic Cisco Voice architecture solutions used in a Unified Communications environment, including anyone who will be working with Contact Center Express or Enterprise.
Upon completing this course, the learner will be able to meet these overall objectives:
- Effectively configure and utilize Unified CM Device Pools and all the accouterments which accompany them.
- Configure inbound/outbound Trunk functionality on Voice Gateway.
- Configure VoIP functionality on Ingress/Egress Gateways including SIP, H.323 and MGCP.
- Describe which Call Control Protocols are suitable for a given deployment and benefits of each.
- Configure basic SIP Proxy and/or Gatekeeper functionality.
- Configure corresponding Trunk type(s) in Unified CM including SIP, H.323 and MGCP.
- Configure digit manipulation on Voice Gateways and Unified CM.
- Implement telephony class-of-service using Calling Search Spaces and Partitions.
- Implement Media Resources (MOH, XCODE) and deploy using MRG’s and MRGL’s.
- Integrate Unified CM with LDAP.
Module 1: Basic Telephony Overview
- Cisco Contact Center Overview
- PBX functionality
- ACD functionality
- IVR/VRU functionality
Module 2: Unified Communications Manager
- What is a Cluster?
- Server Roles in the Cluster
- Enabling Server Services
- Deployment Models
- Redundancy Deployment
- Device Pools
Module 3: On-Net Calling
- IP vs. TDM Call Control Protocols overview
- SIP vs. H.323 vs. MGCP
- The Role of the IOS Voice Gateway
- CUBE Gateways
- DSP’s for the Gateway
- Voice Cards for the Gateway
- Traditional PSTN Gateways
- Inter-Cluster Trunks
Module 4: Off-Network (PSTN) Calling
- TDM Trunk Considerations
- The Importance of Binding
- Depth with Dial-Peers
- Configure SIP Trunk to GW
- Configure H.323 Trunk to GW
- Consolidation of Trunks using SIP Proxy
- Configure MGCP
Module 5: Advanced Dial Plan Considerations
- Call Control Protocol Considerations
- Modifying the numbering plan in Unified CM
- Modifying the numbering plan on the Gateway
- Voice Translation Rules on Gateway
- Route Groups/Route Lists
- Restrict inbound/outbound access using CSS/Partitions
Module 6: CUCM Features and Call Coverage
- Describing Basic Features
- Shared Lines
- Hunt Groups
Module 7: Media Resources
- Voice Signaling
- Audio Conference Streams
- MTP Signaling and Streams
- Annunciator Signaling
- Conference Bridge
- Ad Hoc and Meet-Me Conferencing
- Music on Hold
- Media Resource Groups, and Lists
- Lab 1-1: Explore the Lab Environment
- Lab 2-1: Initial Unified CM Configuration
- Lab 2-2: Device Pools and Initial Phone Registration
- Lab 3-1: Unified CM Trunking with Intercluster Trunks (ICT’s) and SIP Trunks
- Lab 3-2: Unified CM Trunking with SIP Proxy
- Lab 4-1: Configuring MGCP Gateways
- Lab 4-2: Configure Digital Voice Interfaces (ISDN PRI T-1) for H.323 and SIP
- Lab 4-3: Deploying H.323 Gateways
- Lab 4-4: Building SIP Trunks
- Lab 5-1: Configuring Calling Privileges and Restrictions
- Lab 6-1: Configuring User Features
- Lab 6-2: Configuring Hunt Groups and Call Coverage
- Lab 7-1: Configuring Media Resources