CUCM Basics Training

CUCM Basics

Preparing the Voice Gateways

Emulating the Telco

So, now when you have your CUCM installed it is time to add some more ingredients to our lab environment. The Cisco Unified Communications Manager is a great software but no one is not using it as a standalone technology. In fact it is a must that you will need to add some external connections to the Telco via trunks or voice gateways in order to have access to the outside world. To simulate such external connection we will use two virtual Cisco 3725 routers with voice enabled IOS images. Both routers will have direct connection to the CUCM and in the same time they will have a loop between them so that in the end we should be able to route some calls through them.

The basic idea is to use H.323 protocol between the Voice Gateways and the CUCM, and SIP signalling in the loop between the routers. Our goal is to play with the digits' manipulation in the routers by using the translation rules and to simulate a calls between our phones by dialing their DID (Direct Inward Dial) numbers. Check the following image for a reference. It is our GNS3 topology where the cloud represents the VMnet8 interface, where is connected our CUCM node as well.

GNS3 Emulating the Voice Gateways

As this guide is regarding CUCM only, we are not going to explain in details how and why we should configure the routers. However, we are providing you the running configurations of both routers (or voice gateways in our case) which are used for the purpose of the virtual lab.

Configure the routers (Voice Gateways)

There are few major things which should be configured in our routers before switching to the voice. Of course the routing is the first one. Both routers should have interface directly connected to the CUCM network. These interfaces will be used to configure the H.323 connections. The second interfaces on the routers are directly connected and they are utilizing a point-to-point network. This is our loop which will be simulating the Telco operations. Through this connection both routers will be using SIP signalling with which our environment is becoming pretty close to the real world experience.

Now we need to prepare the routers for our voice lab environment. Of course this is not enough to make a test calls through the voice gateways as our CUCM is brand new without any configuration. Here are the initial configurations that you can use:

R1#sh run
...
!
hostname R1
!
!
voice service voip
 allow-connections h323 to sip
 allow-connections sip to h323
!
!
voice translation-rule 1
 rule 1 /^9\(.*$\)/ /\1/
!
!
voice translation-profile OUT
 translate called 1
!
!
interface FastEthernet0/0
 description ### CUCM LAN ###
 ip address 10.1.1.101 255.255.255.0
 duplex auto
 speed auto
 no cdp enable
 h323-gateway voip bind srcaddr 10.1.1.101
!
interface FastEthernet0/1
 description ### Telco LOOP ###
 ip address 10.1.2.1 255.255.255.252
 duplex auto
 speed auto
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.1.1.1
!
!
dial-peer voice 14085552 voip
 destination-pattern 14085552...
 session target ipv4:10.1.1.10
 codec g711ulaw
!
dial-peer voice 9 voip
 translation-profile outgoing OUT
 destination-pattern 9.T
 session protocol sipv2
 session target ipv4:10.1.2.2
 codec g711ulaw
!
...

R2#sh run
...
!
voice service voip
 allow-connections h323 to sip
 allow-connections sip to h323
!
!
voice translation-rule 1
 rule 1 /^9\(.*$\)/ /\1/
!
!
voice translation-profile OUT
 translate called 1
!
!
interface FastEthernet0/0
 description ### CUCM LAN ###
 ip address 10.1.1.102 255.255.255.0
 duplex auto
 speed auto
 no cdp enable
 h323-gateway voip bind srcaddr 10.1.1.102
!
interface FastEthernet0/1
 description ### Telco LOOP ###
 ip address 10.1.2.2 255.255.255.252
 duplex auto
 speed auto
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.1.1.1
!
!
dial-peer voice 9 voip
 translation-profile outgoing OUT
 destination-pattern 9.T
 session protocol sipv2
 session target ipv4:10.1.2.1
 codec g711ulaw
!
dial-peer voice 13124443 voip
 destination-pattern 13124443...
 session target ipv4:10.1.1.10
 codec g711ulaw
!
...

Keep in mind that these configurations are not complete and they are containing only the initial setup of the voice features. As you can see in the configuration used for R1, there is an outgoing translation profile, based on the numbering plan from our lab topology. Only H.323-to-SIP calls are enabled on R1, while only SIP-to-H.323 calls are enabled on R2. There are dial-peers configured on both routers which are used in the "loop" connection between them. It is a SIP communication channel and while the VGW (R1) is configured to modify the internal numbers and to mask them as DID numbers, the Telco router (R2) is configured completely opposite - to strip the prefix again. Later in the course we will see how to configure the connections to the CUCM.