CCNA Chapters List

Learning Guide for
Cisco Voice Gateways
This document is the engineer‟s guide to learn and understand Cisco Voice Gateways
Modification History
Rev. Date Originator Comment
1 24-Nov-2003 Gokul Krishnan Initial Draft
2 10-Apr-2012 Gokul Krishnan Updated gateway physical configurations,
initialization and information on gatekeeper,
IME, SRST, CAC etc.
Get Started with Cisco Voice Gateways Page 1

Table of Contents
1. INTRODUCTION .................................................................................................................. 4
1.1 Overview ................................................................................................................................ 4
2. UNDERSTANDING GATEWAY HARDWARE ............................................................. 4
2.1 Router rear-view .................................................................................................................... 4
2.2 Analog Interfaces................................................................................................................... 5
2.3 Digital Interfaces ................................................................................................................... 7
2.4 Port Numbering ..................................................................................................................... 8
3. BRINGING UP A ROUTER ................................................................................................ 8
3.1 Router Initialization/Erasing the Configuration .................................................................. 8
3.2 Assigning an IP Address ....................................................................................................... 9
3.3 Transferring the Gateway Load/Image/Firmware ............................................................. 10
3.4 Assigning card type (T1/E1) ............................................................................................... 11
3.5 Gobal Configurations .......................................................................................................... 12
3.6 T1/E1 Card/Controller Configuration ................................................................................ 13
4. CALL ROUTING BASICS ................................................................................................ 13
4.1 Dial-peer Configuration ...................................................................................................... 13
4.2 Route Pattern Configuration in Unified CM...................................................................... 14
5. CONFIGURING AN H.323 GATEWAY ......................................................................... 16
5.1 Unified CM Admin page configurations ........................................................................... 16
5.2 Gateway Console Sample Configuration ........................................................................... 16
5.3 Call Example........................................................................................................................ 19
6. CONFIGURING A SIP GATEWAY ................................................................................ 20
6.3 Call Example........................................................................................................................ 23
7. CONFIGURING A MGCP GATEWAY .......................................................................... 24
7.3 Call Example........................................................................................................................ 29
8. CONFIGURING CAT6K FOR PRI ................................................................................. 29
8.1 Gateway Configurations ..................................................................................................... 29
8.3 Additional (sample) CM Configurations ........................................................................... 33
8.4 Call Example........................................................................................................................ 34
9. CONFIGURING CAT6K FOR CAS ................................................................................ 34
9.1 Gateway Configurations ..................................................................................................... 34

9.3 Additional (sample) Unified CM Configurations.............................................................. 36
9.4 Call Example........................................................................................................................ 37
10. CONFIGURING CAT6K FOR 24-PORT FXS .............................................................. 37
10.1 Gateway Configurations.................................................................................................. 37
10.2 Unified CM Admin page configurations........................................................................ 37
10.3 Additional (sample) Unified CM Configurations .......................................................... 39
10.4 Call Example .................................................................................................................... 39
11. AS5300 AS DIGITAL H.323 GATEWAY ....................................................................... 40
11.1 CallManager Admin page configurations ...................................................................... 40
11.2 Additional (sample) CallManager Configurations ........................................................ 40
11.3 Gateway Console configurations .................................................................................... 40
11.4 Additional Gateway Configurations ............................................................................... 41
11.5 Call Example .................................................................................................................... 41
12. CALL ADMISSION CONTROL ...................................................................................... 42
13. GATEKEEPER CONFIGURATION ............................................................................... 42
13.1 Unified CM Admin page configurations........................................................................ 43
13.2 Gatekeeper Console configurations ................................................................................ 44
14. TRUNK CONFIGURATION ............................................................................................. 44
14.1 Unified CM Configuration .............................................................................................. 45
14.2 SIP Transparency and Normalization............................................................................. 45
15. UNDERSTANDING Q.SIG ................................................................................................ 46
15.1 About QSIG ..................................................................................................................... 46
15.2 QSIG Specifications ........................................................................................................ 46
15.3 Configuring Unified CM for QSIG ................................................................................ 46
15.4 QSIG Path Replace Feature ............................................................................................ 48
15.5 Trombone PR Scenario ................................................................................................... 48
15.6 Non-Trombone PR Scenario ........................................................................................... 48
16. IME FEATURE OVERVIEW ........................................................................................... 49
17. SURVIVABLE REMOTE SITE TELEPHONY (SRST) .............................................. 49

1. Introduction
1.1 Overview
This document is created to help anyone new to Cisco IP Telephony, understand the physical
configuration and provisioning of/for Cisco Voice Gateways.
2. Understanding Gateway Hardware
2.1 Router rear-view
One of the most important parts of a Router is the LAN interface (its way to connect to the
packet-switching world). It could be Fast Ethernet ports as shown in 2811 (2&3) or Gigabit
Ethernet ports as shown in 3845 (8).
Routers also have HWIC (High-Speed WAN Interface Card) and NME (Network Module
Enhanced) slots to host voice cards to interface with circuit-switched world. Routers also have
onboard slots for Packet Voice DSP (Digital Signal Processor) Modules (PVDMs); these enable
Cisco Integrated Services Routers to provide high-density voice connectivity, conferencing, and
transcoding capabilities in Cisco IP Communications solutions.
An NME slot (8 in 2811 and 10-13 in 3845) can hold a digital voice card such as NM-HDV2-
2T1E1. Similarly HWIC slots (4-7 in 2811 and 3-5 & 7 in 3845) can hold digital voice cards
such as VWIC2-2MFT-T1E1 or analog voice cards such as VIC-4FXS/DID and/or VIC2-4FXO.

2.2 Analog Interfaces
FXS (foreign-exchange-station) and FXO (foreign-exchange-office) are the name of ports that
allow you to connect analog phones to a VOIP Phone System or traditional PBXs to a VOIP
service provider or to each other via the Internet. An FXS port in a router is used to connect a
regular analog telephone to the VoIP system. An FXS port is a device that, from the point of
view of a telephone, seems to be a telephone exchange but connects to a VoIP service instead.
An FXO port can be used to connect VoIP systems such as an IP-PBX in a LAN environment, to
regular analog telephone lines (POTS PSTN lines). An FXO port from the point of view of a
telephone exchange seems to be a regular telephone. As such, it is able to accept ring signals, go
on-hook and off-hook, and send and receive voice signals. From the VoIP system it appears to be
an external line.
References:
http://www.nch.com.au/kb/10050.html
http://www.3cx.com/PBX/FXS-FXO.html
http://www.digium.com/en/docs/misc/fxs_fxo_desc.php
The high-density Cisco 4-Port FXS/DID VIC can support up to four FXS ports for directly
connecting phones or fax machines, or it can be used to connect up to four direct-inward-dial
(DID) analog trunks, providing customers the flexibility they need for their unique business
environment. Each port on the Cisco 4-Port FXS/DID VIC is selectable for use in either FXS or
DID mode. Cisco 4-Port FXS/DID VIC is the same size as the existing 2-port voice/WAN
interface cards (VWICs) to which it can easily slide in.

Cisco also provides 4-port FXO voice/fax interface card to connect to PBX or key system, or to
provide off-premise connections to PSTN. It supports battery reversal detection and caller ID.
These VICs can be software configured to work in all countries.
VIC-4FXS/DID VIC2-4FXO
A sample network diagram, showing how to use the FXS and FXO ports of the router is shown
below. Phone A1 is a normal telephone extension (like the Infosys extension in IODC) from a
traditional PBX and Phone P1 is an IP phone in the new VoIP based IP-PBX. Take a connection
from the PBX (wall jack), and instead of connecting a normal analog phone, put that to the FXO
port of the router – with some configurations, now P1 will be able to make a call to any phones
(like A1) connected to the PBX. Phone A2 is an extension of phones for the IP-PBX. P1 will be
able to directly call A1 – this is useful, when an enterprise migrate to VoIP, but still want to
retain some low-cost analog phones.

2.3 Digital Interfaces
E1 and T1 links are serial links provided by telephone companies. These are TDM links that are
frequently divided into time slots - 24 time slots of 64kbs on a T1 and 32 time slots on an E1.
The NM-HDV2 IP Communications High-Density Digital Voice or Fax Network Modules

combines WAN Interface Card (WIC) and Voice Interface Card (VIC) functionality to provide
unparalleled flexibility and power. An NM-HDV2-2T1E1 card has two onboard T1/E1 slots. It is
“T1/ E1” means, the card can be configured as either T1 or E1. It can host an additional VWIC2 -
2MFT-T1E1 thus increasing port numbers to 4. Once the main module is configured as T1, the
VWIC2 module will also be T1 and vice versa.
VWIC2-2MFT-T1E1
NM-HDV2-2T1E1
The NM-HDV2 card usually comes with PVDM2 cards. A PVDM2-n supports a maximum of n
Channels in G.711 (example, PVDM2-8). The second-generation 1- and 2-port T1/E1 multiflex
trunk (MFT) voice/WAN interface cards (multiflex VWICs), support data and voice applications
in Cisco multiservice routers. The multiflex VWIC also combines WAN interface card (WIC)
and voice interface card (VIC) functionality.
The number of T1/E1 channels that can be configured, will depend on the PVDM availability.
The E1/T1 cards are usually used in loopback mode in labs, using special crossover cables.

2.4 Port Numbering
Consider a 2811 router loaded with cards as shown below. The HWIC slots 0 has 3 has one
VWIC2-2MFT-T1E1 card each. The HWIC slots 1 and 2 has one VIC-4FXS/DID each. The
NME module/slot has a NM-HDV2-2T1E1, hosted in which is another VWIC2-2MFT-T1E1.
The HWIC slots 0-3 are considered to be in Module 0; NME 1 is Module 1. The port/slot
numbering always starts from zero and are counted from right-to-left and then from bottom-to-
top. Hence the third port in the FXS card placed in HWIC slot 2 (which is Module – 0) is
represented as 0/2/2 (Module – 0, Slot – 2, port – 2, not 3 as port numbers start from 0).
0/3/1
1/1 1/0 0/0/1 0/0/0
0/1/2
1/0/1 1/0/0 0/1/0 0/2/2
Also, note the difference in naming convention for the onboard ports in NM-HDV2-2T1E1. In a
3845 router, the HWIC slots 0-3 are considered to be in Module 0. The NME 1 is Module 1.
Rest of the naming convention is similar to that of 2811.
3. Bringing up a Router
3.1 Router Initialization/Erasing the Configuration
Connect a console cable from the Router to any Windows machine with Hyperterminal. Access
Start->Programs->Accessories->Communications->HyperTerminal. Keep pressing Ctrl-Break
until ROMMON is initialized. Execute commands as shown below:

rommon 1 >
rommon 1 > confreg
Configuration Summary
(Virtual Configuration Register: 0x2102)
enabled are:
load rom after netboot fails
console baud: 9600
boot: image specified by the boot system commands
or default to: cisco2-c2811
do you wish to change the configuration? y/n [n]: y

enable "diagnostic mode"? y/n [n]:
enable "use net in IP bcast address"? y/n [n]:
disable "load rom after netboot fails"? y/n [n]:
enable "use all zero broadcast"? y/n [n]:
enable "break/abort has effect"? y/n [n]:
disable "ignore system config info"? y/n [n]: y
change console baud rate? y/n [n]:
change the boot characteristics? y/n [n]:
Configuration Summary
(Virtual Configuration Register: 0x2142)
enabled are:
load rom after netboot fails
ignore system config info
console baud: 9600
boot: image specified by the boot system commands
or default to: cisco2-c2811
do you wish to change the configuration? y/n [n]: n

You must reset or power cycle for new config to take effect
rommon 1 >
rommon 1 > reset
c2811 platform with 786432 Kbytes of main memory

Main memory is configured to 64 bit mode with ECC enabled
Readonly ROMMON initialized
program load complete, entry point: 0x8000f000, size: 0xcb80
program load complete, entry point: 0x8000f000, size: 0xcb80
Once the system boots up, enter "no" for System Configuration Dialog.
Would you like to enter the initial configuration dialog? [yes/no]: no
For more details on xmodem usage, refer Cisco site.
3.2 Assigning an IP Address
Through HyperTerminal, copy-paste the following text (the ones in bold would need to be
changed). Please note that you have to send “enable (en)” command, followed by a “configure
terminal (conf t)” to get into the configuration/provisioning mode.

en
conf t
config-register 0x2102
hostname infya-mgcpgw7
enable password lab
clock timezone IST 0
ip route 0.0.0.0 0.0.0.0 10.77.44.1
ip domain name vtgtest.com
ip name-server 10.77.35.200
interface GigabitEthernet0/0
#depending on the interface; it can be FE also
#interface FastEthernet0/0
ip address 10.77.44.77 255.255.255.0
duplex auto
speed auto
no shut
exit
line vty 0 4
password lab
login
transport preferred none
exit
exit
wr mem
copy run start
3.3 Transferring the Gateway Load/Image/Firmware
Access the Cisco site (old page) for an image for your system. Copy the load into any TFTP
server (in the below case, 10.77.35.200). Telnet into the Router and do the following:
Password: lab
infya-mgcpgw7>en
Password: lab
infya-mgcpgw7#copy tftp: flash:
Address or name of remote host []? 10.77.35.200
Source filename []? image.bin
Destination filename [image.bin]?
Accessing tftp://10.77.35.200/image.bin...
Once the image is transferred, make this as the boot image for the router and restart the router.

infya-mgcpgw7#conf t
Enter configuration commands, one per line. End with CNTL/Z.
infya-mgcpgw7(config)#boot system flash:image.bin
infya-mgcpgw7(config)#exit
infya-mgcpgw7#wr
Building configuration...
[OK]
infya-mgcpgw7#copy run start
Destination filename [startup-config]?
[OK]
infya-mgcpgw7#reload
Proceed with reload? [confirm]
Before restarting, confirm whether the load is properly marked as the boot image.
infya-mgcpgw7#sh run
...
!
boot-start-marker
boot system flash:image.bin
boot-end-marker
!
...
Once the router boots up, ensure that the image is correctly loaded.
infya-mgcpgw7#sh ver
Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9-M), Version 12.4(17a)
, RELEASE SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2007 by Cisco Systems, Inc.
Compiled Wed 07-Nov-07 13:17 by prod_rel_team
ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

infya-mgcpgw7 uptime is 1 day, 18 hours, 12 minutes
System returned to ROM by reload at 09:13:30 IST Tue Oct 28 2008
System image file is "flash:image.bin"
...
3.4 Assigning card type (T1/E1)
The E1/T1 cards in the system would need to be configured, as mentioned before. When a "show
run" is executed, the user might see instructions as given:
!
! card type command needed for slot/vwic-slot 0/0
! card type command needed for slot/vwic-slot 0/2
!
Execute the "config terminal" command and type the following:

infya-mgcpgw7(config)#card type t1 0 0
infya-mgcpgw7(config)#card type t1 0 2
After that, a "show run" would show lines as follows:
card type t1 0 0
card type t1 0 2
This can be changed later; it would need a “reset” for the changes to take effect.
3.5 Gobal Configurations
When using T1/E1 cards following global configuration would be required.
infya-mgcpgw7(config)# isdn switch-type primary-ni
When working with a SIP gateway following need to be configured.
!
voice service voip
sip
!
When working with an MGCP gateway following configurations are required – in the sample
configuration, the IP address mentioned is that of the IP-PBX.
!
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.77.35.204
ccm-manager config
!
mgcp
mgcp call-agent 10.77.35.204 2427 service-type mgcp version 0.1
mgcp sdp simple
!

3.6 T1/E1 Card/Controller Configuration
The T1/E1 card needs special configuration before they can be used.
router(config)# controller T1 0/0

router(config-controller)# clock source internal
router(config-controller)# linecode ami
router(config-controller)# pri-group timeslots 1-24
The timeslots command configures voice channels; the maximum value in case of a E1 will be
30, instead of 24. In case the PVDM modules are limited, then, timeslots number will also have
to be reduced. Let‟s say, command will be pri-group timeslots 1-10. As the T1/E1 is usually used
in loopback mode, the configurations done on both ends must be similar/complementary.
The controller configuration creates a interface configuration and this needs to be updated.
router(config)# interface Serial0/0/0:23

router(config-if)# encapsulation hdlc
router(config-if)# isdn switch-type primary-ni
router(config-if)# isdn protocol-emulate network
router(config-if)# isdn incoming-voice voice
When connecting ports back-to-back (loopback), one needs to be configured as network and
other as user. The port configured as network must have the line isdn protocol-emulate network;
the other must not.
4. Call Routing Basics
Routing a call to-or-from a voice gateway needs two configurations – Dial-Peer configuration
done on the router CLI and Route Pattern configuration done via Unified CM Administration
pages.
4.1 Dial-peer Configuration
Dial peers are used to identify call source and destination endpoints and to define the
characteristics applied to each call leg in the call connection. Dial peers are used for both
inbound and outbound call legs. For inbound calls from the packet network, the router matches a
“pots” dial peer to terminate the call. Similarly, for inbound calls from a POTS interface, the
router matches a “voip” dial peer to terminate the call. Use "dial-peer voice" to enter into dial-
peer configuration mode. This is followed by a unique number.
The router selects a dial peer for a call leg by matching the string that is defined by using the
“destination-pattern" command in the dial peer configuration. If the dialed string matches the

destination pattern, the call is routed according to the voice port in POTS dial peers, or the
"session target" (network address of the remote router) in voice-network dial peers.
When a “pots” dial-peer is selected, the access server or router strips off the left-justified digits
that match the destination pattern. If you have configured a prefix, the prefix is added to the front
of the remaining digits, creating a dial string, which the router then dials. If all numbers in the
destination pattern are stripped out, the user receives a dial tone. The "no digit-strip" command
disables the automatic digit-stripping function.
When working with MGCP gateways, the dial-peers will not have pattern details. As MGCP
works in master-slave mode, the MGC (or the IP-PBX) takes care of the routing. Hence the dial-
peer configuration will just have a line service mgcpapp.
Like controller configuration, even the dial peer configurations would need to be duplicated.
4.2 Route Pattern Configuration in Unified CM
A route pattern comprises a string of digits (an address) and a set of associated digit
manipulations that route calls to a route list or a gateway. Route patterns provide flexibility in
network design. The simplest route pattern specifies a set of one or more digits. For example, the
number 8912 specifies a route pattern. Gateways and Cisco IP Phones can also use more
complex route patterns that can contain wildcards. A wildcard represents a range of numbers; for
example, X represents any digit 0 through 9.
Call Routing > Route/Hunt > Route Pattern
Field Description
Route Pattern Enter the route pattern, including numbers and wildcards (do not use
spaces); for example, for NANP, enter 9.@ for typical local access or
8XXX for a typical private network numbering plan. Valid characters
include the uppercase characters A, B, C, and D and \+, which represents
the international escape character +.
Gateway/Route Choose the gateway or route list for which you are adding a route
List pattern.
Note: If the gateway is included in a Route Group, this drop-down list
box does not display the gateway. When a gateway is chosen in the drop-
down list box, Cisco Unified Communications Manager uses all the ports
in the gateway to route/block this route pattern. This action does not
apply for MGCP gateways.
Discard Digits From the Discard Digits drop-down list box, choose the discard digits
instructions that you want to associate with this route pattern. For
example, if a Route Patter, 9.XXXX is created, with “PreDot” as the
selected option, when a user dials 91234, 9 gets discarded and the rest of
the digits are passed into the gateway.

Table: Fields of interest to start with
To classify a call as OnNet or OffNet, administrators can set the Call Classification field to
OnNet or OffNet, respectively, on the Route Pattern Configuration window. Administrators can
override the route pattern setting and use the trunk or gateway setting by checking the Allow
Device Override check box on the Route Pattern Configuration window.
Route Patterns work in conjunction with route filters and route lists to direct calls to specific
devices. Route groups contain one or more devices, and route lists contain one or more route
groups.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03rp.html

5. Configuring an H.323 Gateway
5.1 Unified CM Admin page configurations
Device -> Gateway -> H.323 Gateway
Field Description
Device Name Enter a unique name that Cisco CallManager uses to identify the device.
Use either the IP address or the host name as the device name
Device Pool From the drop-down list box, choose the appropriate device pool
Calling Party Any outbound call on a gateway can send directory number information.
Selection This field determines which directory number is sent - Originator/First
Redirect Number/Last Redirect Number
Calling Party Choose whether the Cisco CallManager transmits or blocks caller ID
Presentation
5.2 Gateway Console Sample Configuration
dod-h323gw1#sh run
Current configuration : 3282 bytes

!
! Last configuration change at 08:13:01 IST Mon Jan 30 2012
!
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname dod-h323gw1
!
boot-start-marker
boot system flash:c2800nm-ipvoicek9-mz.151-3.T.bin
boot-end-marker
!
!
card type t1 0 0
card type e1 0 2
enable password lab
!
no aaa new-model
clock timezone IST 0 0
network-clock-participate wic 0

!
dot11 syslog
ip source-route
!
!
ip cef
!
!
no ipv6 cef
multilink bundle-name authenticated
!
!
isdn switch-type primary-ni
!
!
voice service voip
h323
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
license udi pid CISCO2811 sn FHK1304F16N
archive
log config
hidekeys
!
!
controller T1 0/0/0
clock source internal
linecode ami
cablelength long 0db
pri-group timeslots 1-24
!
controller T1 0/0/1
linecode ami
pri-group timeslots 1-10,24
!
controller E1 0/2/0
clock source line primary
linecode ami
!
controller E1 0/2/1
shutdown
linecode ami
!
!
interface FastEthernet0/0
ip address 10.77.46.183 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip id cluster_1 ipaddr 10.77.35.204 1720
!

no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.77.46.1
!
!
control-plane
!
!
voice-port 0/0/0:23
!
voice-port 0/2/0:15
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/0/1:23
!
voice-port 0/2/1:15
!

voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
mgcp profile default
!
!
dial-peer voice 183001 pots
description "H323 - FXS Phone 1"
destination-pattern 46523
port 0/1/0
!
dial-peer voice 183003 voip
destination-pattern 27..
session target ipv4:10.77.35.204
!
destination-pattern 2...
direct-inward-dial
port 0/0/0:23
!
destination-pattern ....
direct-inward-dial
port 0/0/1:23
!
!
line con 0
line aux 0
line vty 0 4
password lab
login
transport input all
!
scheduler allocate 20000 1000
end
5.3 Call Example
IP Phones registered to the Unified CM can dial 46523 and reach a phone connected to FXS port
0/1/0 – this uses dial-peer 183001. Using 1830003, an FXS phone can call IP Phone DNs of 4 -
digits starting with 27. The session target is pointing to the IP-PBX.
With a route pattern configured as 83.XXXX dialing to 4-digit DNs will also work – like an IP
Phone call another IP Phone with DN 2704 by dialing Dialing 832704. The call will get routed
via the T1 loopback and reach back the Unified CM. Dial-peer configurations 183004 and
183005 ensures parity between the two back-to-back connected ports.

6. Configuring a SIP Gateway
Device -> Trunk -> SIP Trunk (Trunk Type)
Field Description
Device Name Enter a unique name that Cisco CallManager uses to identify the device.
Destination IP Address of the gateway device
Address
dod-sipgw1#sh run

!
!
version 15.1
!
hostname dod-sipgw1
!
boot-start-marker
boot-end-marker
!
!
card type t1 0 0
card type e1 0 2
enable password lab
!
no aaa new-model
!
dot11 syslog
ip source-route
!
!

ip cef
!
!
ip multicast-routing
no ipv6 cef
!
!
!
!
voice service voip
sip
!
!
voice-card 0
!
!
!
license udi pid CISCO2811 sn FHK1304F16M
archive
log config
hidekeys
!
!
controller T1 0/0/0
!
controller T1 0/0/1
!
controller E1 0/2/0
!
controller E1 0/2/1
!
!
ip address 10.77.46.182 255.255.255.0
duplex auto
speed auto
!
no ip address
shutdown
duplex auto
speed auto
!
no ip address
encapsulation hdlc

no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
!
no ip http server
!
ip route 0.0.0.0 0.0.0.0 10.77.46.1
ip route 0.0.0.0 0.0.0.0 10.77.44.1
ip route 0.0.0.0 0.0.0.0 10.77.35.1
!
!
tftp-server flash:c2800nm-ipvoicek9-mz.151-3.T.bin
!
control-plane
!
voice-port 0/0/0:23
!
voice-port 0/2/0:15
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/0/1:23
!
voice-port 0/2/1:15
!

voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
!
!
description "SIP - FXS Phone 1"
port 0/1/0
!
destination-pattern 27..
session protocol sipv2
codec g711ulaw
!
destination-pattern 0....
direct-inward-dial
port 0/0/0:23
!
direct-inward-dial
port 0/0/1:23
!
!
line con 0
line aux 0
line vty 0 4
password lab
login
transport input all
!
end
6.3 Call Example
IP Phones registered to the Unified CM can dial 46521 and reach a phone connected to FXS port
0/1/0 – this uses dial-peer 182001. Using 1820003, an FXS phone can call IP Phone DNs of 4-
digits starting with 27. The session target is pointing to the IP-PBX.

With a route pattern configured as 9.0XXXX dialing to 4-digit DNs will also work – like an IP
Phone call another IP Phone with DN 2704 by dialing Dialing 902704. The call will get routed
via the T1 loopback and reach back the Unified CM. Dial-peer configurations 182004 and
182005 ensures parity between the two back-to-back connected ports.
7. Configuring a MGCP Gateway
Device -> Gateway -> Model (say, 2811) -> Protocol (MGCP)
Once basic cards are selected, Save the configuration.
Once the page is refreshed, the subunit informations will be editable. Assuming a card
configuration as in the Port Numbering section, fill in the details – note the numbering. Click on
each port to configure them further.
Use the port type as POTS for FXS ports and Ground Start for FXO ports. FXS port will have
the line configuration similar to that of an IP Phone. When configuring FXO, Attendant DN is a
mandatory information, which will be the DN registered with Unified CM that will take any
incoming calls from the PSTN side. Any PSTN user will be dialing out the extension number of
the cable going to FXO – this will land on the Phone that has the Attendant DN configured.

Protocol Side is an important configuration while configuring E1 or Digital Access PRI (on T1).
This should be according to the T1/E1 controller configuration.
dod-mgcpgw1#sh run

!
!
version 15.1
!
hostname dod-mgcpgw1
!
boot-start-marker
boot-end-marker
!
!
card type t1 0 0
card type e1 0 2
enable password lab
!
no aaa new-model
!
dot11 syslog

ip source-route
!
!
ip cef
!
!
!
no ipv6 cef
!
!
!
!
!
voice-card 0
!
!
!
license udi pid CISCO2811 sn FHK1304F16Q
archive
log config
hidekeys
!
!
controller T1 0/0/0
linecode ami
pri-group timeslots 1-24 service mgcp
!
controller T1 0/0/1
linecode ami
pri-group timeslots 1-13,24 service mgcp
!
controller E1 0/2/0
linecode ami
!
controller E1 0/2/1
linecode ami
!
!
ip address 10.77.46.184 255.255.255.0
duplex auto
speed auto
!
no ip address

shutdown
duplex auto
speed auto
!
no ip address
encapsulation hdlc
isdn bind-l3 ccm-manager
no cdp enable
!
no ip address
encapsulation hdlc
isdn bind-l3 ccm-manager
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
no ip address
encapsulation hdlc
no cdp enable
!
ip default-gateway 10.77.46.1
!
ip http server
!
ip route 0.0.0.0 0.0.0.0 10.77.46.1
!
!
control-plane
!
!
voice-port 0/2/0:15
!
voice-port 0/0/0:23
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!

voice-port 0/1/3
!
voice-port 0/2/1:15
!
voice-port 0/0/1:23
!
voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.77.35.204
ccm-manager config
!
mgcp
mgcp call-agent 10.77.35.204 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726r16 static
!
!
!
service mgcpapp
destination-pattern 2...
port 0/0/0:23
!
service mgcpapp
port 0/0/1:23
!
codec g711ulaw
!
!
line con 0
line aux 0
line vty 0 4
password lab
login

transport input all
!
end
7.3 Call Example
With the above gateway configuration, with a Route Pattern 84.XXXX created in Unified CM,
which is associated with 0/0/1 (Protocol Type – User), a call can be routed back to CM via a
loopback.
8. Configuring Cat6K for PRI
8.1 Gateway Configurations
PRI can be configured on either Digital E1 or T1.
C:\>telnet 10.77.31.158
Cisco Systems Console
Hi.. How are you
Enter password: infosys

CAT6K> en
Enter password: infosys
Cat6K gateway has different modules as shown below:
CAT6K> (enable) sh mod

Mod Slot Ports Module-Type Model Sub
Status
--- ---- ----- ------------------------- ------------------- --- -----
1 1 2 1000BaseX Supervisor WS-X6K-SUP1A-2GE yes ok
2 2 48 10/100BaseTX Ethernet WS-X6348-RJ-45 yes ok
3 3 24 FXS WS-X6624-FXS no ok
4 4 8 T1 WS-X6608-T1 no ok
5 5 8 E1 WS-X6608-E1 no ok
....
A E1/T1 module has 8 ports each. Each port has a unique preassigned IP address. When
configuring a port the user just need to issue the following command:

CAT6K> (enable) set port voice interface 5/2 dhcp disable 10.77.31.168
255.255.255.0 tftp 10.77.31.141 gateway 10.77.31.1
Port 5/2 DHCP disabled.
System DNS configurations used.
Using the command shown, we are configuring module 5 (E1) – port 2 (that is, 5/2) which has IP
address 10.77.31.168, to Call Manager 10.77.31.141. You can verify the gateway setting as
follows:
CAT6K> (enable) sh port 5/2

Port Name Status Vlan Duplex Speed Type
----- -------------------- ---------- ---------- ------ ----- --------
5/2 notconnect 1 full 2.048 E1
Port DHCP MAC-Address IP-Address Subnet-Mask

-------- ------- ----------------- --------------- ---------------
5/2 disable 00-01-64-11-dd-f5 10.77.31.168 255.255.255.0
Port Call-Manager(s) DHCP-Server TFTP-Server Gateway

-------- ----------------- --------------- --------------- -----------
5/2 10.77.31.141 - 10.77.31.141 10.77.31.1
Port DNS-Server(s) Domain

-------- ----------------- -------------------------------------------
5/2 192.135.240.5 -
Port CallManagerState DSP-Type

-------- ---------------- --------
5/2 registered C549
Port NoiseRegen NonLinearProcessing

----- ---------- -------------------
5/2 enabled enabled
Port Trap IfIndex

----- -------- -------
5/2 disabled 43

Call Manager supports PRI on both Digital E1 and T1.
Field Description
MAC Address Enter MAC address of the Cat6K port. The MAC address
uniquely identifies the hardware device. (use “show port”
command at Cat6K console to retrieve the MAC/IP address of a
port)
PRI Protocol Type Choose the communications protocol for the span. If you include
this gateway in a route group, you cannot switch between QSIG
and non-QSIG protocol types until you remove the gateway
from the route group. PRI spans have several options, depending
on the carrier or switch.

Protocol Side Choose the appropriate protocol side. This setting specifies
whether the gateway connects to a Central Office/Network
device or to a User device. Make sure that the two ends of the
PRI connection use opposite settings.

Note: In the IODC lab, we will be looping back two E1-PRI ports or two T1-PRI ports. Admin
page configurations will be needed separately for each port.
8.3 Additional (sample) CM Configurations
IPAddress: 10.77.31.141
RoutePattern1: 1581.XXXXXX
RoutePattern2: 1583.XXXXXX
(Both RoutePatterns use PreDot as the “Discard Digits” option in the “Called Party
Transformations”)
RoutePattern1 points to one of the T1 ports.
RoutePattern2 points to one of the E1 ports.
IPPhones: 133203 and 133204

8.4 Call Example
133203 dials 1581133204 and reaches another IP phone 133204. (The call uses RoutePattern1:
1581.XXXXXX)
133203 dials 1583133204 and reaches another IP phone 133204. (The call uses RoutePattern2:
1583.XXXXXX)
9. Configuring Cat6K for CAS
Similar as above. CAS is usually configured on Digital T1.
Call Manager supports CAS on only Digital T1.

Now click Add New Port. A port configuration dialog box opens. From the Port Type drop-
down list box, choose a port type.

Note: In the IODC lab, we will be looping back two T1-CAS ports. Admin page configurations
will be needed separately for each port.
9.3 Additional (sample) Unified CM Configurations
IPAddress: 10.77.31.141
RoutePattern: 1582.XXXXXX
(PreDot is used as the “Discard Digits” option in the “Called Party Transformations”)
RoutePattern points to one of the T1 ports.

9.4 Call Example
133203 dials 1582133204 and reaches another IP phone 133204. (The call uses RoutePattern:
1581.XXXXXX)
10. Configuring Cat6K for 24-port FXS
In the IODC lab, module 3 supports FXS ports – 24 ports that share one MAC address and IP
address. When configuring FXS module use the following command:
CAT6K> (enable) set port voice interface 3/1-24 dhcp disable

10.77.31.199/255.255.255.0 vlan 1 tftp 10.77.31.141 gateway 10.77.31.1
Port 5/2 DHCP disabled.
System DNS configurations used.
Using the command shown, we are configuring module 3 (FXS) – ports 1 to 24 (that is, 3/1-24)
which has IP address 10.77.31.199, to Call Manager 10.77.31.141.

Now click Add New Port. A port configuration dialog box opens. From the Port Type drop-
down list box, choose a port type.

Note: In the IODC lab, an extra piece of hardware is connected to WS-X6624-FXS, which
supports 24 FXS ports. Insert a POTS phone into one of the ports and assign a DN (say, 14101).
10.3 Additional (sample) Unified CM Configurations
IPPhones: 133203
10.4 Call Example
133203 dials 14101 and reaches POTS phone.

11. AS5300 as Digital H.323 Gateway
11.1 CallManager Admin page configurations
Same as in “Configuring H.323 Gateway”.
11.2 Additional (sample) CallManager Configurations
IPAddress: 10.77.31.141
RoutePattern: 155.8XXXXXX
(PreDot is used as the “Discard Digits” option in the “Called Party Transformations”)
This RoutePattern point to AS5300 gateway.
11.3 Gateway Console configurations
C:\>telnet 10.77.31.155
User Access Verification
Password:cisco
Cisco2600>en
Password:cisco
!
controller T1 0
framing esf
fdl ansi
linecode b8zs
!
controller T1 1
framing esf
fdl ansi
clock source line secondary 1
linecode b8zs
!
!
interface Serial0:23
no ip address
isdn incoming-voice modem
no isdn T309-enable

isdn T310 30000
no cdp enable
!
interface Serial1:23
no ip address
isdn incoming-voice modem
no cdp enable
!
!
voice-port 0:D
!
voice-port 1:D
!
!
destination-pattern 8......
direct-inward-dial
port 0:D
!
destination-pattern 1.....
codec g711ulaw
!
direct-inward-dial
port 1:D
!
Note: It may seem that we require only a voip dial-peer for port-1. But for a succesful call setup
we need to duplicate port-0 settings in port-1 (the ports that are looped back) as well. This
accounts for the extra 111 dial-peer.
11.4 Additional Gateway Configurations
IPAddress: 10.77.31.155
Ports 0-1 are loopbacked
11.5 Call Example
133203 dials 21110 and reaches another IP phone 133205. (The call uses RoutePattern:
155.8XXXXXX and dial-peers:100 and 101)

12. Call Admission Control
Call admission control enables you to control the audio quality and video quality of calls over a
wide-area (IP WAN) link by limiting the number of calls that are allowed on that link at the same
time. For example, you can use call admission control to regulate the voice quality on a 56-kbps
frame relay line that connects your main campus and a remote site.
Audio and video quality can begin to degrade when too many active calls exist on a link and the
amount of bandwidth is oversubscribed. Call admission control regulates audio and video quality
by limiting the number of calls that can be active on a particular link at the same time. Call
admission control does not guarantee a particular level of audio or video quality on the link, but
it does allow you to regulate the amount of bandwidth that active calls on the link consume.
Call admission control operates by rejecting a call for bandwidth and policy reasons. When a call
gets rejected due to call admission control, the phone of the called party does not ring, and the
caller receives a busy tone. The caller also receives a message on their phone, such as "Not
enough bandwidth."
Without call admission control, customers may perceive that IP voice is low in quality and
unreliable. With call admission control, customers experience situations similar to the time-
division multiplexing (TDM) processing and realize that they need more bandwidth for peak
hours.
For details, refer the presentation on CAC.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_1_1/ccmsys/a02cac.html
13. Gatekeeper Configuration
A Gatekeeper device, also known as a Cisco Multimedia Conference Manager (MCM), supports
the H.225 Registration, Admission, and Status Protocol (RAS) message set that is used for call
admission control, bandwidth allocation, and dial pattern resolution (call routing). The
gatekeeper provides these services for communications between the Cisco Unified
Communications Manager server and H.323 networks, in a distributed call-processing
environment. You can configure multiple gatekeeper devices per Cisco Unified Communications
Manager server. You can configure alternate gatekeepers for redundancy.
Route patterns or route groups can route the calls to and from the gatekeeper. In a distributed
call-processing environment, the gatekeeper uses the E.164 address (phone number) and
determines the appropriate IP address for the destination of each call, and the local Cisco Unified
Communications Manager uses that IP address to complete the call.

Following menu option is used to add a Gatekeeper:
Device -> Gatekeeper
Field Description
Host Name/IP Enter the IP address or host name of the gatekeeper in this required field.
Address
Enable Device This check box allows you to register this gatekeeper with Cisco Unified
Communications Manager. By default, this check box remains checked.
To unregister the gatekeeper from Cisco Unified Communications
Manager gracefully, uncheck this check box. The gatekeeper unregisters
within approximately 1 minute of updating this field.
Registration Request Time to Live (default of 60 seconds) and Registration Retry Timeout
(default of 300 seconds) are usually not changed. The Registration Request Time to Live field
indicates the time that the gatekeeper considers a registration request (RRQ) valid. The system
must send a keepalive RRQ to the gatekeeper before the RRQ Time to Live expires. Cisco
Unified Communications Manager sends an RRQ to the gatekeeper to register and subsequently
to maintain a connection with the gatekeeper. The gatekeeper may confirm (RCF) or deny (RRJ)
the request. The Registration Retry Timeout field indicates the time that Cisco Unified
Communications Manager waits before retrying gatekeeper registration after a failed registration
attempt.
After that, add H.225 Gatekeeper controlled Trunk (from Device -> Trunk). Under Gatekeeper
section, add the details (which will have to be in sync with the console configuration as well).

13.2 Gatekeeper Console configurations
gatekeeper
zone local SJGK1 cisco.com
zone prefix SJGK1 408*
gw-type-prefix 1#* default-technology
no shutdown
!--- The Cisco CallManager trunks register and appear as VoIP-GW.
3640-1#show gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION

================================
CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
--------------- ----- --------------- ----- --------- ---- -----
171.69.85.31 1720 171.69.85.31 4724 SJGK1 TERM
E164-ID: 3166188111
171.69.85.171 4613 171.69.85.171 1160 SJGK1 VOIP-GW
H323-ID: TrunkDevice1GK_1
Total number of active registrations = 2
In order to specify the directory number range for a particular Cisco CallManager, use the zone
prefix command to configure the range on the gatekeeper. For example, this command specifies
the DN for zone SJGK1 that starts from 408-527.
zone prefix SJGK1 408527*
Use the bandwidth command on the gatekeeper in order to specify the available bandwidth. For
example, this command allocates 512 kbps to the SJGK1 zone.
bandwidth total zone SJGK1 512
Reference:
http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00801177
68.shtml
14. Trunk Configuration
In a distributed call-processing environment, Cisco Unified Communications Manager

communicates not only with the public switched telephone network (PSTN), but with other IP-
PBXes (could be another Unified CM cluster) as well. Inter-cluster Trunks (ICTs) are used to
communicate to other IP-PBXes. Trunks can be Gatekeeper controlled or Non-Gatekeeper
controlled. For large distributed networks where many Cisco Unified Communications Manager
clusters exist, you can avoid configuring individual intercluster trunks between each cluster by
using gatekeepers.

You can configure these types of trunk devices in Cisco Unified Communications Manager
Administration:
o H.225 Trunk (Gatekeeper Controlled)

o Intercluster Trunk (Gatekeeper Controlled)
o Intercluster Trunk (Non-Gatekeeper Controlled)
o SIP Trunk
14.1 Unified CM Configuration
Following menu option is used to add a trunk. Like gateways, trunks are also associated with
route patterns to make a call. Trunks must be created on both initiating and destination clusters,
pointing to each other or else, the call will be rejected.
Device -> Trunk
Field Description
Device Name Enter a unique identifier for the trunk.
Server IP Address/Host Name Enter the IP address or host name of the remote Cisco
or Destination Address Unified Communications Manager that this trunk accesses,
in the case of non-gatekeeper-controlled intercluster trunks.
Gatekeeper Information Choose the gatekeeper that controls this trunk and other
related information; required for gatekeeper-controlled
H.225 trunks and intercluster trunks
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/6_0_1/ccmcfg/b06trunk.html
14.2 SIP Transparency and Normalization
SIP trunks can connect to a variety of endpoints, including PBXs, gateways, and service
providers. Each endpoint may implement the SIP protocol differently, which can cause a unique
set of interoperability issues. SIP transparency and normalization allow Cisco Unified
Communications Manager to interoperate seamlessly with a variety of PBXs and service
providers. Normalization allows you to modify incoming and outgoing SIP messages at a
protocol level on their way through Cisco Unified Communications Manager. Transparency
allows Cisco Unified Communications Manager to pass headers, parameters, and content bodies
from one call leg to another.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmsys/a08sip.html

15. Understanding Q.SIG
This section is prepared based on inputs from TC (Chatchanop Tanapipatkulchai). For more
information refer TC‟s Wiki.
15.1 About QSIG
Refer word defintion, wikimapia and QSIG home page.
Q Signaling (abbreviated QSIG) is an ISDN based signaling protocol for signaling between
private branch exchanges (PBXs). It makes use of the connection-level Q.931 protocol and the
application-level ROSE (Remote Operations Service Element, defined in ASN.1) protocol.
QSIG has two layers, called BC (basic call) and generic function (GF). QSIG BC ensures that
signaling is transparent among nodes from multiple vendors. GF provides additional
functions for large-scale corporate, educational, and government networks, such as line
identification, call intrusion, call diversion, and support for multiple applications. QSIG
transports the feature related information via APDU (Application Protocol Data Unit).
15.2 QSIG Specifications
Refer Cisco Router pages.
QSIG is a variant of ISDN D-channel signaling. The protocol was originally specified by ECMA
(European Computer Manufacturers Association), then was adopted by European
Telecommunications Standards Institute (ETSI) and the ISO. With QSIG, which is becoming the
standard for PBX interoperability in Europe and North America, Cisco networks emulate the
functionality of the public-switched telephone network (PSTN) and allows easy migration for
enterpries which already had normal PBXes.
15.3 Configuring Unified CM for QSIG
QSIG implementation on CUCM is based on ECMA V1 and ISO standards. ECMA support was
introduced in Skate Release (CIA-178). An administrator can configure which standard to use by
setting the “QSIG Variant” service parameter (under „Advanced‟).
In Seaview, QSIG was supported only on PRI Gateways; usually ECMA variant is used over E1
and ISO is used over T1 trunks. The gateway does Layer1 and Layer2 (Q.921) functionality
while CUCM takes care of the Layer3 (Q.931) functionality.

“QSIG over ICT/H.323” (H.323 Annex M1 support) was introduced in Skate. Then H.323
Annex M1 support was extended to support in H.225 trunk and H.323 gateways as part of CIA-
334. Initially the support was only on ISO and from Rally the support is extended for ECMA V1
(CSCsk52682). In the ICT page there is a field “Tunneling” which takes values as QSIG or
None. If “QSIG” is chosen, then the trunk can be used to send APDUs across CUCMs. Also note
that there is a checkbox for enabling “Path Replacement” in the Trunk/Gateway page.
QSIG tunneled calls were never supported in H323 GW (CSCsc38287). The requirement to
support ECMA V1 (Protocol 91) APDUs over Annex M.1 came up as part of the CSCsk52682
during Charge Release.
In the diagram shown, originating CUCM encodes a “Calling Name APDU” from SIP INVITE
message and sends the APDU via QSIG gateway in a “Q.931 Setup” message. The CCM
Backhaul (PriQsigSetup) uses a proprietary messaging for sending this Signal. Destination
PBX/CUCM receives the Setup message and decodes the Calling Name APDU. Later on, when
B-Channels would need to be setup for media CUCM instructs the gateways using MGCP.
Let us see how QSIG works over ICT by taking the example of CallBack. Let us say Phone-A in
CUCM-1 calls Phone-B in CUCM-2. An initial H225 Setup will be send across the ICT. In
initial H225 Setup, tunnel signaling will contain APDU such as Calling Name APDU. When
Phone-A initiates CallBack, CUCM-1 prepares an APDU with CallBack information and sends
across to CUCM-2 inside another Setup message. If the Trunk was not configured for QSIG
Tunneling, the Callback request would have failed.
Also it can be noted that. even before Rally, even if the QSIG variant was configured for ECMA
(which was not supported over ICT before Rally) in the service parameter the Callback may
succeed by using ISO ! This is the same assumption for QSIG PRI also when the service

parameter is set to ISO but CUCM receives a protocol profile as ECMA V1 (not ISO). The call
and QSIG APDUs may succeed.
15.4 QSIG Path Replace Feature
Path Replacement (PR) is a features that can be configured to be triggered by Call Transfer
(Blind, Consult) or Forward Switching (CFA, CFNA etc). There is a Service Parameter to
control Forward Switching; by default it supports PR – but if configured for “Forward by
Reroute”, then it does not triggger PR.
APDUs for Transfer are ctComplete and ctActive; while Forward

uses DivertingLegInformation1 APDU. PR gets triggered only on active calls (after the call is
connected/answered) and the feature is completed.
Test Scenario: Phones A1 and A2 registered to CUCM-1. Phone B1 registered to CUCM-2 and
C1 registered to CUCM-2. PR established within one CUCM is Trombone PR and across
CUCMs is Non-Trombone PR.
15.5 Trombone PR Scenario
A1 calls B1 and B1 hits transfer and dials A2. A2 answers. Two B-Channels are used at this time
– one for call between A1 & B1 and another for call between B1 and A2.
Once transfer is completed, A1 and A2 will be connected. A PR timer (configured on service

parameter) will be started now (in case of Blind transfer A2 has to answer for timer to start). On
the timer expiry, the initial B-channels will be released and A1 will be connected to A2 as an
internal call.
15.6 Non-Trombone PR Scenario
A1 calls B1 and B1 is CFNA-ed to C1. C1 answers. Two B-Channels are used at this time – one
for call between A1 & B1 (it is used/blocked though there is no active call between A1 and B1)
and another for call between B1 and C1.
Let us assume that PR Timer for CUCM-1 is 5 seconds and that for CUCM-3 is 10 seconds.
After 5 seconds, PRPropse APDU will be send from CUCM-1 (known as “Requesting PINX”)
as a Facility message. CUCM-2 (known as “Transit PINX”) will pass this APDU to CUCM-3.
Then CUCM-3 (known as “Cooperating PINX”) will accept this message and initiate a PRSetup
message to CUCM-1. This will be send only if there is Route Pattern configured for call from
CUCM-3 to CUCM-1. On receiving the PRSetup, CUCM-1 will send a Connect message. At
this time, the initial B-channels between CUCM-1 & CUCM-2 and CUCM-2 & CUCM-3 will be
released and A1 will be connected to C1 through a direct B-Channel from CUCM-1 to CUCM-3.
After PR kicks in, both users should not see any changes such as name display and number
display.

16. IME Feature Overview
Cisco Intercompany Media Engine (Cisco IME) provides a technique for establishing direct IP
connectivity between enterprises by combining peer-to-peer technologies with the existing public
switched telephone network (PSTN) infrastructure.
Cisco IME allows companies that have deployed Cisco Unified Communications Manager to
communicate securely over the Internet rather than the PSTN by creating dynamic Session
Initiation Protocol (SIP) trunks between the enterprises. By enabling traffic outside of the
enterprise to travel over the Internet, Cisco IME extends features and functionality to external
calls that have previously worked exclusively within the enterprise, such as video enabled calls,
wideband audio support, rich caller ID, presence, and others.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/ime/8_0_2/config/ime_802.html
17. Survivable Remote Site Telephony (SRST)
As the enterprise extends its IP telephony deployments from central sites to remote branch
offices and teleworkers, a critical factor in achieving a successful deployment is the capability to
support backup call control at these remote locations. Survivable Remote Site Telephony (SRST)
gets used at sites that depend on a centralized Cisco Unified Communications Manager cluster
that is accessible via a WAN connection. SRST provides telephony service to IP phones at the
remote site in the event of a WAN outage. An SRST-enabled router has features that allow calls
between IP phones at the remote site to call each other, allow calls from the PSTN to reach the IP
phones, and allow calls from the IP phones to reach the external world through the PSTN.
Intelligence in the SRST router that can accept registrations from the IP phones and route calls
based on the directory numbers that are registered, and based on the routing that is configured for
the PSTN link, accomplishes that.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_1/ccmcfg/b02srst.html
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-
570481.html

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Learning Guide for

Cisco Voice Gateways

Get Started with Cisco Voice Gateways Page 1

Get Started with Cisco Voice Gateways Page 2

Get Started with Cisco Voice Gateways Page 3

2. Understanding Gateway Hardware

2.1 Router rear-view

Get Started with Cisco Voice Gateways Page 4

Get Started with Cisco Voice Gateways Page 5

Get Started with Cisco Voice Gateways Page 6

The NM-HDV2 IP Communications High-Density Digital Voice or Fax Network Modules

Get Started with Cisco Voice Gateways Page 7

1/0/1 1/0/0 0/1/0 0/2/2

3.1 Router Initialization/Erasing the Configuration

Get Started with Cisco Voice Gateways Page 8

do you wish to change the configuration? y/n [n]: y

do you wish to change the configuration? y/n [n]: n

c2811 platform with 786432 Kbytes of main memory

Would you like to enter the initial configuration dialog? [yes/no]: no

For more details on xmodem usage, refer Cisco site.

3.2 Assigning an IP Address

Get Started with Cisco Voice Gateways Page 9

3.3 Transferring the Gateway Load/Image/Firmware

Get Started with Cisco Voice Gateways Page 10

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

3.4 Assigning card type (T1/E1)

Execute the "config terminal" command and type the following:

Get Started with Cisco Voice Gateways Page 11

After that, a "show run" would show lines as follows:

3.5 Gobal Configurations

When using T1/E1 cards following global configuration would be required.

infya-mgcpgw7(config)# isdn switch-type primary-ni

When working with a SIP gateway following need to be configured.

Get Started with Cisco Voice Gateways Page 12

router(config)# controller T1 0/0

router(config)# interface Serial0/0/0:23

4. Call Routing Basics

4.1 Dial-peer Configuration

Get Started with Cisco Voice Gateways Page 13

4.2 Route Pattern Configuration in Unified CM

Call Routing > Route/Hunt > Route Pattern

Get Started with Cisco Voice Gateways Page 14

Get Started with Cisco Voice Gateways Page 15

5.1 Unified CM Admin page configurations

Device -> Gateway -> H.323 Gateway

Table: Fields of interest to start with

5.2 Gateway Console Sample Configuration

Current configuration : 3282 bytes

Get Started with Cisco Voice Gateways Page 16

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Get Started with Cisco Voice Gateways Page 18

5.3 Call Example

Get Started with Cisco Voice Gateways Page 19

6.1 Unified CM Admin page configurations

Device -> Trunk -> SIP Trunk (Trunk Type)

Table: Fields of interest to start with

6.2 Gateway Console Sample Configuration

Current configuration : 3263 bytes

Get Started with Cisco Voice Gateways Page 20

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Get Started with Cisco Voice Gateways Page 22

6.3 Call Example

Get Started with Cisco Voice Gateways Page 23

7. Configuring a MGCP Gateway