Voice Port Config

Americas Headquarters
Cisco Systems, Inc.

170 West Tasman Drive
San Jose, CA 95134-1706
USA
http://www.cisco.com
Tel: 408 526-4000
800 553-NETS (6387)
Fax: 408 527-0883
Cisco IOS Voice Port Configuration Guide
Release 12.4T

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL
STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT
WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.
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OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.
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NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED AS IS WITH
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WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO
OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
CCDE, CCENT, Cisco Eos, Cisco Lumin, Cisco Nexus, Cisco StadiumVision, Cisco TelePresence, the Cisco logo, DCE, and Welcome to the Human Network are
trademarks; Changing the Way We Work, Live, Play, and Learn and Cisco Store are service marks; and Access Registrar, Aironet, AsyncOS, Bringing the Meeting To You,
Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems,
Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Collaboration Without Limitation, EtherFast, EtherSwitch, Event Center, Fast Step, Follow Me Browsing,
FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQ Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, IronPort, the IronPort logo,
LightStream, Linksys, MediaTone, MeetingPlace, MeetingPlace Chime Sound, MGX, Networkers, Networking Academy, Network Registrar, PCNow, PIX, PowerPanels,
ProConnect, ScriptShare, SenderBase, SMARTnet, Spectrum Expert, StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and the
WebEx logo are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a partnership relationship
between Cisco and any other company. (0807R)
Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the
document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.
2008 Cisco Systems, Inc. All rights reserved.

i
About Cisco IOS and Cisco IOS XE Software
Documentation
Last updated: August 6, 2008
This document describes the objectives, audience, conventions, and organization used in Cisco IOS and
Cisco IOS XE software documentation, collectively referred to in this document as Cisco IOS
documentation. Also included are resources for obtaining technical assistance, additional
documentation, and other information from Cisco. This document is organized into the following
sections:
Documentation Objectives, page i
Audience, page i
Documentation Conventions, page ii
Documentation Organization, page iii
Additional Resources and Documentation Feedback, page xi
Documentation Objectives
Cisco IOS documentation describes the tasks and commands available to configure and maintain Cisco
networking devices.
Audience
The Cisco IOS documentation set is i ntended for users who configure and maintain Cisco networking
devices (such as routers and switches) but who may not be familiar with the configuration and
maintenance tasks, the relationship among tasks, or the Cisco IOS commands necessary to perform
particular tasks. The Cisco IOS documentation set is also intended for those users experienced with
Cisco IOS who need to know about new features, new configuration options, and new software
characteristics in the current Cisco IOS release.

About Cisco IOS and Cisco IOS XE Software Documentation
Documentation Conventions
ii

Documentation Conventions
In Cisco IOS documentation, the term router may be used to refer to various Cisco products; for example,
routers, access servers, and switches. These and other networking devices that support Cisco IOS
software are shown interchangeably in examples and are used only for illustrative purposes. An example
that shows one product does not necessarily mean that other products are not supported.
This section includes the following topics:
Typographic Conventions, page ii
Command Syntax Conventions, page ii
Software Conventions, page iii
Reader Alert Conventions, page iii
Typographic Conventions
Cisco IOS documentation uses the following typographic conventions:
Command Syntax Conventions
Cisco IOS documentation uses the following command syntax conventions:
Convention Description
^ or Ctrl Both the ^ symbol and Ctrl represent the Control (Ctrl) key on a keyboard. For
example, the key combination ^D or Ctrl-D means that you hold down the
Control key while you press the D key. (Keys are indicated in capital letters but
are not case sensitive.)
string A string is a nonquoted set of characters shown in italics. For example, when
setting a Simple Network Management Protocol (SNMP) community string to
public, do not use quotation marks around the string; otherwise, the string will
include the quotation marks.
bold Bold text indicates commands and keywords that you enter as shown.
italic Italic text indicates arguments for which you supply values.
[x] Square brackets enclose an optional keyword or argument.
| A vertical line, called a pipe, indicates a choice within a set of keywords
or arguments.
[x | y] Square brackets enclosing keywords or arguments separated by a pipe indicate an
optional choice.
{x | y} Braces enclosing keywords or arguments separated by a pipe indicate a
required choice.
[x {y | z}] Braces and a pipe within square brackets indicate a required choice within an
optional element.

Documentation Organization
iii
Software Conventions
Cisco IOS uses the following program code conventions:
Reader Alert Conventions
The Cisco IOS documentation set uses the following conventions for reader alerts:
Caution Means reader be careful. In this situation, you might do something that could result in equipment
damage or loss of data.
Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the
manual.
Timesaver Means the described action saves time. You can save time by performing the action described in the
paragraph.
This section describes the Cisco IOS documentation set, how it is organized, and how to access it on
Cisco.com. Included are lists of configuration guides, command references, and supplementary
references and resources that make up the documentation set. The following topics are included:
Cisco IOS Documentation Set, page iv
Cisco IOS Documentation on Cisco.com, page iv
Configuration Guides, Command References, and Supplementary Resources, page v
Courier font
Courier font is used for information that is displayed on a PC or terminal screen.
Bold Courier font
Bold Courier font indicates text that the user must enter.
< > Angle brackets enclose text that is not displayed, such as a password. Angle
brackets also are used in contexts in which the italic font style is not supported;
for example, ASCII text.
! An exclamation point at the beginning of a line indicates that the text that follows
is a comment, not a line of code. An exclamation point is also displayed by
Cisco IOS software for certain processes.
[ ] Square brackets enclose default responses to system prompts.

iv

Cisco IOS Documentation Set
Cisco IOS documentation consists of the following:
Release notes and caveats provide information about platform, technology, and feature support for
a release and describe severity 1 (catastrophic), severity 2 (severe), and severity 3 (moderate) defects
in released Cisco IOS code. Review release notes before other documents to learn whether or not
updates have been made to a feature.
Sets of configuration guides and command references organized by technology and published for
each standard Cisco IOS release.
Configuration guidesCompilations of documents that provide informational and
task-oriented descriptions of Cisco IOS features.
Command referencesCompilations of command pages that provide detailed information
about the commands used in the Cisco IOS features and processes that make up the related
configuration guides. For each technology, there is a single command reference that covers all
Cisco IOS releases and that is updated at each standard release.
Lists of all the commands in a specific release and all commands that are new, modified, removed,
or replaced in the release.
Command reference book for debug commands. Command pages are listed in alphabetical order.
Reference book for system messages for all Cisco IOS releases.
Cisco IOS Documentation on Cisco.com
The following sections describe the documentation organization and how to access various document
types.
Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS
software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An
account on Cisco.com is not required.
New Features List
The New Features List for each release provides a list of all features in the release with hyperlinks to the
feature guides in which they are documented.
Feature Guides
Cisco IOS features are documented in feature guides. Feature guides describe one feature or a group of
related features that are supported on many different software releases and platforms. Your Cisco IOS
software release or platform may not support all the features documented in a feature guide. See the
Feature Information table at the end of the feature guide for information about which features in that
guide are supported in your software release.
Configuration Guides
Configuration guides are provided by technology and release and comprise a set of individual feature
guides relevant to the release and technology.

v
Command References
Command reference books describe Cisco IOS commands that are supported in many different software
releases and on many different platforms. The books are provided by technology. For information about
all Cisco IOS commands, use the Command Lookup Tool at http://tools.cisco.com/Support/CLILookup
or the Cisco IOS Master Command List, All Releases, at
http://www.cisco.com/en/US/docs/ios/mcl/all_release/all_mcl.html.
Cisco IOS Supplementary Documents and Resources
Supplementary documents and resources are listed in Table 2 on page xi.
Configuration Guides, Command References, and Supplementary Resources
Table 1 lists, in alphabetical order, Cisco IOS and Cisco IOS XE software configuration guides and
command references, including brief descriptions of the contents of the documents. The Cisco IOS
command references are comprehensive, meaning that they include commands for both Cisco IOS
software and Cisco IOS XE software, for all releases. The configuration guides and command references
support many different software releases and platforms. Your Cisco IOS software release or platform
may not support all these technologies.
For additional information about configuring and operating specific networking devices, go to the
Product Support area of Cisco.com at http://www.cisco.com/web/psa/products/index.html.
Table 2 lists documents and resources that supplement the Cisco IOS software configuration guides and
command references. These supplementary resources include release notes and caveats; master
command lists; new, modified, removed, and replaced command lists; system messages; and the debug
command reference.
Table 1 Cisco IOS and Cisco IOS XE Configuration Guides and Command References
Configuration Guide and Command Reference Titles Features/Protocols/Technologies
Cisco IOS AppleTalk Configuration Guide
Cisco IOS XE AppleTalk Configuration Guide
Cisco IOS AppleTalk Command Reference
AppleTalk protocol.
Cisco IOS Asynchronous Transfer Mode
Configuration Guide
Cisco IOS Asynchronous Transfer Mode
Command Reference
LAN ATM, multiprotocol over ATM (MPoA), and WAN ATM.

vi

Cisco IOS Bridging and IBM Networking
Configuration Guide
Cisco IOS Bridging Command Reference
Cisco IOS IBM Networking Command Reference
Transparent and source-route transparent (SRT) bridging,
source-route bridging (SRB), Token Ring Inter-Switch Link
(TRISL), and token ring route switch module (TRRSM).
Data-link switching plus (DLSw+), serial tunnel (STUN),
block serial tunnel (BSTUN); logical link control, type 2
(LLC2), synchronous data link control (SDLC); IBM
Network Media Translation, including Synchronous Data
Logical Link Control (SDLLC) and qualified LLC (QLLC);
downstream physical unit (DSPU), Systems Network
Architecture (SNA) service point, SNA frame relay access,
advanced peer-to-peer networking (APPN), native client
interface architecture (NCIA) client/server topologies, and
IBM Channel Attach.
Cisco IOS Broadband and DSL Configuration Guide
Cisco IOS XE Broadband and DSL Configuration Guide
Cisco IOS Broadband and DSL Command Reference
Point-to-Point Protocol (PPP) over ATM (PPPoA) and PPP over
Ethernet (PPPoE).
Cisco IOS Carrier Ethernet Configuration Guide
Cisco IOS Carrier Ethernet Command Reference
Connectivity fault management (CFM), Ethernet Local
Management Interface (ELMI), IEEE 802.3ad link bundling,
Link Layer Discovery Protocol (LLDP), media endpoint
discovery (MED), and operations, administration, and
maintenance (OAM).
Cisco IOS Configuration Fundamentals
Configuration Guide
Cisco IOS XE Configuration Fundamentals
Configuration Guide
Cisco IOS Configuration Fundamentals
Command Reference
Autoinstall, Setup, Cisco IOS command-line interface (CLI),
Cisco IOS file system (IFS), Cisco IOS web browser user
interface (UI), basic file transfer services, and file management.
Cisco IOS DECnet Configuration Guide
Cisco IOS XE DECnet Configuration Guide
Cisco IOS DECnet Command Reference
DECnet protocol.
Cisco IOS Dial Technologies Configuration Guide
Cisco IOS XE Dial Technologies Configuration Guide
Cisco IOS Dial Technologies Command Reference
Asynchronous communications, dial backup, dialer technology,
dial-in terminal services and AppleTalk remote access (ARA),
large scale dialout, dial-on-demand routing, dialout, modem and
resource pooling, ISDN, multilink PPP (MLP), PPP, virtual
private dialup network (VPDN).
Cisco IOS Flexible NetFlow Configuration Guide
Cisco IOS Flexible NetFlow Command Reference
Flexible NetFlow.
Table 1 Cisco IOS and Cisco IOS XE Configuration Guides and Command References (continued)

vii
Cisco IOS H.323 Configuration Guide Gatekeeper enhancements for managed voice services,
Gatekeeper Transaction Message Protocol, gateway codec order
preservation and shutdown control, H.323 dual tone
multifrequency relay, H.323 version 2 enhancements, Network
Address Translation (NAT) support of H.323 v2 Registration,
Admission, and Status (RAS) protocol, tokenless call
authorization, and VoIP gateway trunk and
carrier-based routing.
Cisco IOS High Availability Configuration Guide
Cisco IOS XE High Availability Configuration Guide
Cisco IOS High Availability Command Reference
A variety of High Availability (HA) features and technologies
that are available for different network segments (from
enterprise access to service provider core) to facilitate creation
of end-to-end highly available networks. Cisco IOS HA features
and technologies can be categorized in three key areas:
system-level resiliency, network-level resiliency, and embedded
management for resiliency.
Cisco IOS Integrated Session Border Controller
Command Reference
A VoIP-enabled device that is deployed at the edge of networks.
An SBC is a toolkit of functions, such as signaling interworking,
network hiding, security, and quality of service (QoS).
Cisco IOS Intelligent Service Gateway
Configuration Guide
Cisco IOS Intelligent Service Gateway
Command Reference
Subscriber identification, service and policy determination,
session creation, session policy enforcement, session life-cycle
management, accounting for access and service usage, session
state monitoring.
Cisco IOS Interface and Hardware Component
Configuration Guide
Cisco IOS XE Interface and Hardware Component
Configuration Guide
Cisco IOS Interface and Hardware Component
Command Reference
LAN interfaces, logical interfaces, serial interfaces, virtual
interfaces, and interface configuration.
Cisco IOS IP Addressing Services Configuration Guide
Cisco IOS XE Addressing Services Configuration Guide
Cisco IOS IP Addressing Services Command Reference
Address Resolution Protocol (ARP), Network Address
Translation (NAT), Domain Name System (DNS), Dynamic
Host Configuration Protocol (DHCP), and Next Hop Address
Resolution Protocol (NHRP).
Cisco IOS IP Application Services Configuration Guide
Cisco IOS XE IP Application Services Configuration
Guide
Cisco IOS IP Application Services Command Reference
Enhanced Object Tracking (EOT), Gateway Load Balancing
Protocol (GLBP), Hot Standby Router Protocol (HSRP), IP
Services, Server Load Balancing (SLB), Stream Control
Transmission Protocol (SCTP), TCP, Web Cache
Communication Protocol (WCCP), User Datagram Protocol
(UDP), and Virtual Router Redundancy Protocol (VRRP).
Cisco IOS IP Mobility Configuration Guide
Cisco IOS IP Mobility Command Reference
Mobile ad hoc networks (MANet) and Cisco mobile networks.
Cisco IOS IP Multicast Configuration Guide
Cisco IOS XE IP Multicast Configuration Guide
Cisco IOS IP Multicast Command Reference
Protocol Independent Multicast (PIM) sparse mode (PIM-SM),
bidirectional PIM (bidir-PIM), Source Specific Multicast
(SSM), Multicast Source Discovery Protocol (MSDP), Internet
Group Management Protocol (IGMP), and Multicast VPN
(MVPN).

viii

Cisco IOS IP Routing Protocols Configuration Guide
Cisco IOS XE IP Routing Protocols Configuration Guide
Cisco IOS IP Routing Protocols Command Reference
Border Gateway Protocol (BGP), multiprotocol BGP,
multiprotocol BGP extensions for IP multicast, bidirectional
forwarding detection (BFD), Enhanced Interior Gateway
Routing Protocol (EIGRP), Interior Gateway Routing Protocol
(IGRP), Intermediate System-to-Intermediate System (IS-IS),
on-demand routing (ODR), Open Shortest Path First (OSPF),
and Routing Information Protocol (RIP).
Cisco IOS IP SLAs Configuration Guide
Cisco IOS XE IP SLAs Configuration Guide
Cisco IOS IP SLAs Command Reference
Cisco IOS IP Service Level Agreements (IP SLAs).
Cisco IOS IP Switching Configuration Guide
Cisco IOS XE IP Switching Configuration Guide
Cisco IOS IP Switching Command Reference
Cisco Express Forwarding, fast switching, and Multicast
Distributed Switching (MDS).
Cisco IOS IPv6 Configuration Guide
Cisco IOS XE IPv6 Configuration Guide
Cisco IOS IPv6 Command Reference
For IPv6 features, protocols, and technologies, go to the IPv6
Start Here document at the following URL:
http://www.cisco.com/en/US/docs/ios/ipv6/configuration/
guide/ip6-roadmap.html
Cisco IOS ISO CLNS Configuration Guide
Cisco IOS XE ISO CLNS Configuration Guide
Cisco IOS ISO CLNS Command Reference
ISO connectionless network service (CLNS).
Cisco IOS LAN Switching Configuration Guide
Cisco IOS XE LAN Switching Configuration Guide
Cisco IOS LAN Switching Command Reference
VLANs, Inter-Switch Link (ISL) encapsulation, IEEE 802.10
encapsulation, IEEE 802.1Q encapsulation, and multilayer
switching (MLS).
Cisco IOS Mobile Wireless Gateway GPRS Support Node
Configuration Guide
Cisco IOS Mobile Wireless Gateway GPRS Support Node
Command Reference
Cisco IOS Gateway GPRS Support Node (GGSN) in a
2.5-generation general packet radio service (GPRS) and
3-generation universal mobile telecommunication system (UMTS)
network.
Cisco IOS Mobile Wireless Home Agent
Configuration Guide
Cisco IOS Mobile Wireless Home Agent
Command Reference
Cisco Mobile Wireless Home Agent, an anchor point for mobile
terminals for which mobile IP or proxy mobile IP services are
provided.
Cisco IOS Mobile Wireless Packet Data Serving Node
Configuration Guide
Cisco IOS Mobile Wireless Packet Data Serving Node
Command Reference
Cisco Packet Data Serving Node (PDSN), a wireless gateway that
is between the mobile infrastructure and standard IP networks and
that enables packet data services in a code division multiple access
(CDMA) environment.
Cisco IOS Mobile Wireless Radio Access Networking
Configuration Guide
Cisco IOS Mobile Wireless Radio Access Networking
Command Reference
Cisco IOS radio access network products.

ix
Cisco IOS Multiprotocol Label Switching
Configuration Guide
Cisco IOS XE Multiprotocol Label Switching
Configuration Guide
Cisco IOS Multiprotocol Label Switching
Command Reference
MPLS Label Distribution Protocol (LDP), MPLS Layer 2 VPNs,
MPLS Layer 3 VPNs, MPLS Traffic Engineering (TE), and
MPLS Embedded Management (EM) and MIBs.
Cisco IOS Multi-Topology Routing Configuration Guide
Cisco IOS Multi-Topology Routing Command Reference
Unicast and multicast topology configurations, traffic
classification, routing protocol support, and network
management support.
Cisco IOS NetFlow Configuration Guide
Cisco IOS XE NetFlow Configuration Guide
Cisco IOS NetFlow Command Reference
Network traffic data analysis, aggregation caches, export
features.
Cisco IOS Network Management Configuration Guide
Cisco IOS XE Network Management Configuration Guide
Cisco IOS Network Management Command Reference
Basic system management; system monitoring and logging;
troubleshooting, logging, and fault management;
Cisco Discovery Protocol; Cisco IOS Scripting with Tool
Control Language (Tcl); Cisco networking services (CNS);
DistributedDirector; Embedded Event Manager (EEM);
Embedded Resource Manager (ERM); Embedded Syslog
Manager (ESM); HTTP; Remote Monitoring (RMON); SNMP;
and VPN Device Manager Client for Cisco IOS Software
(XSM Configuration).
Cisco IOS Novell IPX Configuration Guide
Cisco IOS XE Novell IPX Configuration Guide
Cisco IOS Novell IPX Command Reference
Novell Internetwork Packet Exchange (IPX) protocol.
Cisco IOS Optimized Edge Routing Configuration Guide
Cisco IOS Optimized Edge Routing Command Reference
Optimized edge routing (OER) monitoring, policy
configuration, routing control, logging and reporting, and
VPN IPsec/generic routing encapsulation (GRE) tunnel
interface optimization.
Cisco IOS Quality of Service Solutions
Configuration Guide
Cisco IOS XE Quality of Service Solutions
Configuration Guide
Cisco IOS Quality of Service Solutions
Command Reference
Class-based weighted fair queuing (CBWFQ), custom queuing,
distributed traffic shaping (DTS), generic traffic shaping (GTS),
IP- to-ATM class of service (CoS), low latency queuing (LLQ),
modular QoS CLI (MQC), Network-Based Application
Recognition (NBAR), priority queuing, Security Device
Manager (SDM), Multilink PPP (MLPPP) for QoS, header
compression, AutoQoS, QoS features for voice, Resource
Reservation Protocol (RSVP), weighted fair queuing (WFQ),
and weighted random early detection (WRED).
Cisco IOS Security Configuration Guide
Cisco IOS XE Security Configuration Guide
Cisco IOS Security Command Reference
Access control lists (ACLs), authentication, authorization, and
accounting (AAA), firewalls, IP security and encryption,
neighbor router authentication, network access security, network
data encryption with router authentication, public key
infrastructure (PKI), RADIUS, TACACS+, terminal access
security, and traffic filters.

x

Cisco IOS Service Selection Gateway Configuration Guide
Cisco IOS Service Selection Gateway Command Reference
Subscriber authentication, service access, and accounting.
Cisco IOS Software Activation Configuration Guide
Cisco IOS Software Activation Command Reference
An orchestrated collection of processes and components to
activate Cisco IOS software feature sets by obtaining and
validating Cisco software licenses.
Cisco IOS Software Modularity Installation and
Configuration Guide
Cisco IOS Software Modularity Command Reference
Installation and basic configuration of software modularity
images, including installations on single and dual route
processors, installation rollbacks, software modularity binding,
software modularity processes and patches.
Cisco IOS Terminal Services Configuration Guide
Cisco IOS Terminal Services Command Reference
Cisco IOS XE Terminal Services Command Reference
DEC, local-area transport (LAT), and X.25 packet
assembler/disassembler (PAD).
Cisco IOS Virtual Switch Command Reference Virtual switch redundancy, high availability, and packet handling;
converting between standalone and virtual switch modes; virtual
switch link (VSL); Virtual Switch Link Protocol (VSLP).
Note For information about virtual switch configuration, refer
to the product-specific software configuration
information for the Cisco Catalyst 6500 series switch or
for the Metro Ethernet 6500 series switch.
Cisco IOS Voice Configuration Library
Cisco IOS Voice Command Reference
Cisco IOS support for voice call control protocols, interoperability,
physical and virtual interface management, and troubleshooting.
The library includes documentation for IP telephony applications.
Cisco IOS VPDN Configuration Guide
Cisco IOS XE VPDN Configuration Guide
Cisco IOS VPDN Command Reference
Layer 2 Tunneling Protocol (L2TP) dial-out load balancing and
redundancy, L2TP extended failover, L2TP security VPDN,
multihop by Dialed Number Identification Service (DNIS),
timer and retry enhancements for L2TP and Layer 2 Forwarding
(L2F), RADIUS Attribute 82: tunnel assignment ID, shell-based
authentication of VPDN users, tunnel authentication via
RADIUS on tunnel terminator.
Cisco IOS Wide-Area Networking Configuration Guide
Cisco IOS XE Wide-Area Networking Configuration Guide
Cisco IOS Wide-Area Networking Command Reference
Frame Relay, Layer 2 Tunneling Protocol Version 3 (L2TPv3),
Link Access Procedure, Balanced (LAPB), Switched
Multimegabit Data Service (SMDS), and X.25.
Cisco IOS Wireless LAN Configuration Guide
Cisco IOS Wireless LAN Command Reference
Broadcast key rotation, IEEE 802.11x support, IEEE 802.1x
authenticator, IEEE 802.1x local authentication service for
Extensible Authentication Protocol-Flexible Authentication via
Secure Tunneling (EAP-FAST), Multiple Basic Service Set ID
(BSSID), Wi-Fi Multimedia (WMM) required elements, and
Wi-Fi Protected Access (WPA).

Additional Resources and Documentation Feedback
xi
Whats New in Cisco Product Documentation is published monthly and describes all new and revised
Cisco technical documentation. The Whats New in Cisco Product Documentation publication also
provides information about obtaining the following resources:
Technical documentation
Cisco product security overview
Product alerts and field notices
Technical assistance
Cisco IOS technical documentation includes embedded feedback forms where you can rate documents
and provide suggestions for improvement. Your feedback helps us improve our documentation.
Table 2 Cisco IOS Supplementary Documents and Resources
Document Title Description
Cisco IOS Master Command List, All Releases Alphabetical list of all the commands documented in all
Cisco IOS releases.
Cisco IOS New, Modified, Removed, and
Replaced Commands
List of all the new, modified, removed, and replaced commands
for a Cisco IOS release.
Cisco IOS Software System Messages List of Cisco IOS system messages and descriptions. System
messages may indicate problems with your system; be
informational only; or may help diagnose problems with
communications lines, internal hardware, or the
system software.
Cisco IOS Debug Command Reference Alphabetical list of debug commands including brief
descriptions of use, command syntax, and usage guidelines.
Release Notes and Caveats Information about new and changed features, system
requirements, and other useful information about specific
software releases; information about defects in specific
Cisco IOS software releases.
MIBs Files used for network monitoring. To locate and download
MIBs for selected platforms, Cisco IOS releases, and feature
sets, use Cisco MIB Locator at the following URL:
http://www.cisco.com/go/mibs
RFCs Standards documents maintained by the Internet Engineering
Task Force (IETF) that Cisco IOS documentation references
where applicable. The full text of referenced RFCs may be
obtained at the following URL:
http://www.rfc-editor.org/

xii

CCDE, CCENT, Cisco Eos, Cisco Lumin, Cisco Nexus, Cisco StadiumVision, Cisco TelePresence, the Cisco logo, DCE, and Welcome to the
Human Network are trademarks; Changing the Way We Work, Live, Play, and Learn and Cisco Store are service marks; and Access Registrar,
Aironet, AsyncOS, Bringing the Meeting To You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified
Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Collaboration
Without Limitation, EtherFast, EtherSwitch, Event Center, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient,
IOS, iPhone, iQ Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, IronPort, the IronPort logo, LightStream, Linksys, MediaTone,
MeetingPlace, MeetingPlace Chime Sound, MGX, Networkers, Networking Academy, Network Registrar, PCNow, PIX, PowerPanels, ProConnect,
ScriptShare, SenderBase, SMARTnet, Spectrum Expert, StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and
the WebEx logo are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply
a partnership relationship between Cisco and any other company. (0807R)

Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and
figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and
coincidental.


i
Using the Command-Line Interface in Cisco IOS
and Cisco IOS XE Software
Last updated: August 6, 2008
This document provides basic information about the command-line interface (CLI) in Cisco IOS and
Cisco IOS XE software and how you can use some of the CLI features. This document contains the
following sections:
Initially Configuring a Device, page i
Using the CLI, page ii
Saving Changes to a Configuration, page xii
Additional Information, page xii
For more information about using the CLI, see the Using the Cisco IOS Command-Line Interface
section of the Cisco IOS Configuration Fundamentals Configuration Guide.
For information about the software documentation set, see the About Cisco IOS and Cisco IOS XE
Software Documentation document.
Initially Configuring a Device
Initially configuring a device varies by platform. For information about performing an initial
configuration, see the hardware installation documentation that is provided with the original packaging
of the product or go to the Product Support area of Cisco.com at
http://www.cisco.com/web/psa/products/index.html.
After you have performed the initial configuration and connected the device to your network, you can
configure the device by using the console port or a remote access method, such as Telnet or Secure Shell
(SSH), to access the CLI or by using the configuration method provided on the device, such as Security
Device Manager.

Using the Command-Line Interface in Cisco IOS and Cisco IOS XE Software
Using the CLI
ii

Changing the Default Settings for a Console or AUX Port
There are only two changes that you can make to a console port and an AUX port:
Change the port speed with the config-register 0x command. Changing the port speed is not
recommended. The well-known default speed is 9600.
Change the behavior of the port; for example, by adding a password or changing the timeout value.
Note The AUX port on the Route Processor (RP) installed in a Cisco ASR1000 series router does not serve
any useful customer purpose and should be accessed only under the advisement of a customer support
representative.
Using the CLI
This section describes the following topics:
Understanding Command Modes, page ii
Using the Interactive Help Feature, page v
Understanding Command Syntax, page vi
Understanding Enable and Enable Secret Passwords, page viii
Using the Command History Feature, page viii
Abbreviating Commands, page ix
Using Aliases for CLI Commands, page ix
Using the no and default Forms of Commands, page x
Using the debug Command, page x
Filtering Output Using Output Modifiers, page x
Understanding CLI Error Messages, page xi
Understanding Command Modes
The CLI command mode structure is hierarchical, and each mode supports a set of specific commands.
This section describes the most common of the many modes that exist.
Table 1 lists common command modes with associated CLI prompts, access and exit methods, and a
brief description of how each mode is used.

Using the CLI
iii
Table 1 CLI Command Modes
Command
Mode
Access Method Prompt Exit Method Mode Usage
User EXEC Log in.
Router>
Issue the logout or exit
command.
Change terminal
settings.
Perform basic tests.
Display device status.
Privileged
EXEC
From user EXEC mode,
issue the enable
command.
Router#
Issue the disable
command or the exit
command to return to
user EXEC mode.
Issue show and debug
commands.
Copy images to the
device.
Reload the device.
Manage device
configuration files.
Manage device file
systems.
Global
configuration
From privileged EXEC
mode, issue the
configure terminal
command.
Router(config)#
Issue the exit command
or the end command to
return to privileged
EXEC mode.
Configure the device.
Interface
configuration
From global
configuration mode,
issue the interface
command.
Router(config-if)#
to return to global
configuration mode or
the end command to
EXEC mode.
Configure individual
interfaces.
Line
configuration
From global
configuration mode,
issue the line vty or line
console command.
Router(config-line)#
to return to global
configuration mode or
the end command to
EXEC mode.
Configure individual
terminal lines.

Using the CLI
iv

ROM monitor From privileged EXEC
mode, issue the reload
command. Press the
Break key during the
first 60 seconds while
the system is booting.
rommon # >
The # symbol
represents the line
number and increments
at each prompt.
Issue the continue
command.
Run as the default
operating mode when a
valid image cannot be
loaded.
Access the fall-back
procedure for loading an
image when the device
lacks a valid image and
cannot be booted.
Perform password
recovery when a
CTRL-Break sequence is
issued within 60 seconds
of a power-on or reload
event.
Diagnostic
(available only
on the Cisco
ASR1000
series router)
The router boots or
enters diagnostic mode
in the following
scenarios. When a
Cisco IOS process or
processes fail, in most
scenarios the router will
reload.
A user-configured
access policy was
configured using
the transport-map
command, which
directed the user
into diagnostic
mode.
The router was
accessed using an
RP auxiliary port.
A break signal
(Ctrl-C,
Ctrl-Shift-6, or the
send break
command) was
entered, and the
router was
configured to enter
diagnostic mode
when the break
signal was received.
Router(diag)#
If a Cisco IOS process
failure is the reason for
entering diagnostic
mode, the failure must
be resolved and the
router must be rebooted
to exit diagnostic mode.
If the router is in
diagnostic mode
because of a
transport-map
configuration, access
the router through
another port or using a
method that is
configured to connect to
the Cisco IOS CLI.
If the RP auxiliary port
was used to access the
router, use another port
for access. Accessing
the router through the
auxiliary port is not
useful for customer
purposes.
Inspect various states on
the router, including the
Cisco IOS state.
Replace or roll back the
configuration.
Provide methods of
restarting the Cisco IOS
software or other
processes.
Reboot hardware, such
as the entire router, an
RP, an ESP, a SIP, a SPA,
or possibly other
hardware components.
Transfer files into or off
of the router using
remote access methods
such as FTP, TFTP, and
SCP.
Table 1 CLI Command Modes (continued)
Command
Mode
Access Method Prompt Exit Method Mode Usage

Using the CLI
v
EXEC commands are not saved when the software reboots. Commands that you issue in a configuration
mode can be saved to the startup configuration. If you save the running configuration to the startup
configuration, these commands will execute when the software is rebooted. Global configuration mode
is the highest level of configuration mode. From global configuration mode, you can enter a variety of
other configuration modes, including protocol-specific modes.
ROM monitor mode is a separate mode that is used when the software cannot load properly. If a valid
software image is not found when the software boots or if the configuration file is corrupted at startup,
the software might enter ROM monitor mode. Use the question symbol (?) to view the commands that
you can use while the device is in ROM monitor mode.
rommon 1 > ?
alias set and display aliases command
boot boot up an external process
confreg configuration register utility
cont continue executing a downloaded image
context display the context of a loaded image
cookie display contents of cookie PROM in hex
.
.
.
rommon 2 >
The following example shows how the command prompt changes to indicate a different command mode:
Router> enable
Router# configure terminal
Router(config)# interface ethernet 1/1
Router(config-if)# ethernet
Router(config-line)# exit
Router(config)# end
Router#
Note A keyboard alternative to the end command is Ctrl-Z.
Using the Interactive Help Feature
The CLI includes an interactive Help feature. Table 2 describes how to use the Help feature.

Table 2 CLI Interactive Help Commands
Command Purpose
help Provides a brief description of the help feature in any command mode.
? Lists all commands available for a particular command mode.
partial command? Provides a list of commands that begin with the character string (no
space between the command and the question mark).
partial command<Tab> Completes a partial command name (no space between the command
and <Tab>).
command ? Lists the keywords, arguments, or both associated with the command
(space between the command and the question mark).
command keyword ? Lists the arguments that are associated with the keyword (space between
the keyword and the question mark).

Using the CLI
vi

The following examples show how to use the help commands:
help
Router> help
Help may be requested at any point in a command by entering a question mark '?'. If
nothing matches, the help list will be empty and you must backup until entering a '?'
shows the available options.
Two styles of help are provided:
1. Full help is available when you are ready to enter a command argument (e.g. 'show ?')
and describes each possible argument.
2. Partial help is provided when an abbreviated argument is entered and you want to know
what arguments match the input (e.g. 'show pr?'.)
?
Router# ?
Exec commands:
access-enable Create a temporary access-List entry
access-profile Apply user-profile to interface
access-template Create a temporary access-List entry
alps ALPS exec commands
archive manage archive files
<snip>
partial command?
Router(config)# zo?
zone zone-pair
partial command<Tab>
Router(config)# we<Tab> webvpn
command ?
Router(config-if)# pppoe ?
enable Enable pppoe
max-sessions Maximum PPPOE sessions
command keyword ?
Router(config-if)# pppoe enable ?
group attach a BBA group
<cr>
Understanding Command Syntax
Command syntax is the format in which a command should be entered in the CLI. Commands include
the name of the command, keywords, and arguments. Keywords are alphanumeric strings that are used
literally. Arguments are placeholders for values that a user must supply. Keywords and arguments may
be required or optional.
Specific conventions convey information about syntax and command elements. Table 3 describes these
conventions.

Using the CLI
vii
The following examples show syntax conventions:
Router(config)# ethernet cfm domain ?
WORD domain name
Router(config)# ethernet cfm domain dname ?
level
Router(config)# ethernet cfm domain dname level ?
<0-7> maintenance level number
Router(config)# ethernet cfm domain dname level 7 ?
<cr>
Router(config)# snmp-server file-transfer access-group 10 ?
protocol protocol options
<cr>
Router(config)# logging host ?
Hostname or A.B.C.D IP address of the syslog server
ipv6 Configure IPv6 syslog server
Router(config)# snmp-server file-transfer access-group 10 ?
protocol protocol options
<cr>
Table 3 CLI Syntax Conventions
Symbol/Text Function Notes
< > (angle brackets) Indicate that the option is an
argument.
Sometimes arguments are displayed
without angle brackets.
A.B.C.D. Indicates that you must enter a
dotted decimal IP address.
Angle brackets (< >) are not always
used to indicate that an IP address is
an argument.
WORD (all capital letters) Indicates that you must enter
one word.
used to indicate that a WORD is an
argument.
LINE (all capital letters) Indicates that you must enter
more than one word.
used to indicate that a LINE is an
argument.
<cr> (carriage return) Indicates the end of the list of
available keywords and argu-
ments, and also indicates when
keywords and arguments are
optional. When <cr> is the only
option, you have reached the
end of the branch or the end of
the command if the command
has only one branch.

Using the CLI
viii

Understanding Enable and Enable Secret Passwords
Some privileged EXEC commands are used for actions that impact the system, and it is recommended
that you set a password for these commands to prevent unauthorized use. Two types of passwords, enable
(not encrypted) and enable secret (encrypted), can be set. The following commands set these passwords
and are issued in global configuration mode:
enable password
enable secret password
Using an enable secret password is recommended because it is encrypted and more secure than the
enable password. When you use an enable secret password, text is encrypted (unreadable) before it is
written to the config.text file. When you use an enable password, the text is written as entered (readable)
to the config.text file.
Each type of password is case sensitive, can contain from 1 to 25 uppercase and lowercase alphanumeric
characters, and can start with a number. Spaces are also valid password characters; for example,
two words is a valid password. Leading spaces are ignored, but trailing spaces are recognized.
Note Both password commands have numeric keywords that are single integer values. If you choose a number
for the first character of your password followed by a space, the system will read the number as if it were
the numeric keyword and not as part of your password.
When both passwords are set, the enable secret password takes precedence over the enable password.
To remove a password, use the no form of the commands: no enable password or
no enable secret password.
For more information about password recovery procedures for Cisco products, see
http://www.cisco.com/en/US/products/sw/iosswrel/ps1831/
products_tech_note09186a00801746e6.shtml.
Using the Command History Feature
The CLI command history feature saves the commands you enter during a session in a command history
buffer. The default number of commands saved is 10, but the number is configurable within the range of
0 to 256. This command history feature is particularly useful for recalling long or complex commands.
To change the number of commands saved in the history buffer for a terminal session, issue the
terminal history size command:
Router# terminal history size num
A command history buffer is also available in line configuration mode with the same default and
configuration options. To set the command history buffer size for a terminal session in line configuration
mode, issue the history command:
Router(config-line)# history [size num]
To recall commands from the history buffer, use the following methods:
Press Ctrl-P or the up arrow keyRecalls commands beginning with the most recent command.
Repeat the key sequence to recall successively older commands.

Using the CLI
ix
Press Ctrl-N or the down arrow keyRecalls the most recent commands in the history buffer after
they have been recalled using Ctrl-P or the up arrow key. Repeat the key sequence to recall
successively more recent commands.
Note The arrow keys function only on ANSI-compatible terminals such as the VT100.
Issue the show history command in user EXEC or privileged EXEC modeLists the most recent
commands that you entered. The number of commands that are displayed is determined by the
setting of the terminal history size and history commands.
The CLI command history feature is enabled by default. To disable this feature for a terminal
session, issue the terminal no history command in user EXEC or privileged EXEC mode or the
no history command in line configuration mode.
Abbreviating Commands
Typing a complete command name is not always required for the command to execute. The CLI
recognizes an abbreviated command when the abbreviation contains enough characters to uniquely
identify the command. For example, the show version command can be abbreviated as sh ver. It cannot
be abbreviated as s ver because s could mean show, set, or systat. The sh v abbreviation also is not valid
because the show command has vrrp as a keyword in addition to version. (Command and keyword
examples from Cisco IOS Release 12.4(13)T.)
Using Aliases for CLI Commands
To save time and the repetition of entering the same command multiple times, you can use a command
alias. An alias can be configured to do anything that can be done at the command line, but an alias cannot
move between modes, type in passwords, or perform any interactive functions.
Table 4 shows the default command aliases.
To create a command alias, issue the alias command in global configuration mode. The syntax of the
command is alias mode command-alias original-command. Following are some examples:
Router(config)# alias exec prt partitionprivileged EXEC mode
Router(config)# alias configure sb source-bridgeglobal configuration mode
Router(config)# alias interface rl rate-limitinterface configuration mode
Table 4 Default Command Aliases
Command Alias Original Command
h help
lo logout
p ping
s show
u or un undebug
w where

Using the CLI
x

To view both default and user-created aliases, issue the show alias command.
For more information about the alias command, see
http://www.cisco.com/en/US/docs/ios/fundamentals/command/reference/cf_book.html.
Using the no and default Forms of Commands
Most configuration commands have a no form that is used to reset a command to its default value or
disable a feature or function. For example, the ip routing command is enabled by default. To disable this
command, you would issue the no ip routing command. To re-enable IP routing, you would issue the
ip routing command.
Configuration commands may also have a default form, which returns the command settings to their
default values. For commands that are disabled by default, using the default form has the same effect as
using the no form of the command. For commands that are enabled by default and have default settings,
the default form enables the command and returns the settings to their default values.
The no and default forms of commands are described in the command pages of command references.
Using the debug Command
A debug command produces extensive output that helps you troubleshoot problems in your network.
These commands are available for many features and functions within Cisco IOS and Cisco IOS XE
software. Some debug commands are debug all, debug aaa accounting, and debug mpls packets. To
use debug commands during a Telnet session with a device, you must first enter the terminal monitor
command. To turn off debugging completely, you must enter the undebug all command.
For more information about debug commands, see the Cisco IOS Debug Command Reference at
http://www.cisco.com/en/US/docs/ios/debug/command/reference/db_book.html.
Caution Debugging is a high priority and high CPU utilization process that can render your device unusable. Use
debug commands only to troubleshoot specific problems. The best times to run debugging are during
periods of low network traffic and when few users are interacting with the network. Debugging during
these periods decreases the likelihood that the debug command processing overhead will affect network
performance or user access or response times.
Filtering Output Using Output Modifiers
Many commands produce lengthy output that may use several screens to display. Using output modifiers,
you can filter this output to show only the information that you want to see.
Three output modifiers are available and are described as follows:
begin regular expressionDisplays the first line in which a match of the regular expression is found
and all lines that follow.
include regular expressionDisplays all lines in which a match of the regular expression is found.
exclude regular expressionDisplays all lines except those in which a match of the regular
expression is found.

Using the CLI
xi
To use one of these output modifiers, type the command followed by the pipe symbol (|), the modifier,
and the regular expression that you want to search for or filter. A regular expression is a case-sensitive
alphanumeric pattern. It can be a single character or number, a phrase, or a more complex string.
The following example illustrates how to filter output of the show interface command to display only
lines that include the expression protocol.
Router# show interface | include protocol
FastEthernet0/0 is up, line protocol is up
Serial4/0 is up, line protocol is up
Serial4/1 is up, line protocol is up
Serial4/2 is administratively down, line protocol is down
Serial4/3 is administratively down, line protocol is down
Understanding CLI Error Messages
You may encounter some error messages while using the CLI. Table 5 shows the common CLI error
messages.

For more system error messages, see the following documents:
Cisco IOS Release 12.2SR System Message Guide
Cisco IOS System Messages, Volume 1 of 2 (Cisco IOS Release 12.4)
Cisco IOS System Messages, Volume 2 of 2 (Cisco IOS Release 12.4)
Table 5 Common CLI Error Messages
Error Message Meaning How to Get Help
% Ambiguous command:
show con
You did not enter enough
characters for the command to
be recognized.
Reenter the command followed by a
space and a question mark (?). The
keywords that you are allowed to
enter for the command appear.
% Incomplete command. You did not enter all the
keywords or values required
by the command.
Reenter the command followed by a
space and a question mark (?). The
keywords that you are allowed to
enter for the command appear.
% Invalid input detected at ^
marker.
You entered the command in-
correctly. The caret (^) marks
the point of the error.
Enter a question mark (?) to display
all the commands that are available in
this command mode. The keywords
that you are allowed to enter for the
command appear.

Saving Changes to a Configuration
xii

Saving Changes to a Configuration
To save changes that you made to the configuration of a device, you must issue the copy running-config
startup-config command or the copy system:running-config nvram:startup-config command. When
you issue these commands, the configuration changes that you made are saved to the startup
configuration and saved when the software reloads or power to the device is turned off or interrupted.
The following example shows the syntax of the copy running-config startup-config command:
Router# copy running-config startup-config
Destination filename [startup-config]?
You press Enter to accept the startup-config filename (the default), or type a new filename and then press
Enter to accept that name. The following output is displayed indicating that the configuration was saved:
Building configuration...
[OK]
Router#
On most platforms, the configuration is saved to NVRAM. On platforms with a Class A flash file system,
the configuration is saved to the location specified by the CONFIG_FILE environment variable. The
CONFIG_FILE variable defaults to NVRAM.
Additional Information
Using the Cisco IOS Command-Line Interface section of the
Cisco IOS Configuration Fundamentals Configuration Guide:
http://www.cisco.com/en/US/docs/ios/fundamentals/configuration/guide/cf_cli-basics.html
or
Using Cisco IOS XE Software chapter of the Cisco ASR1000 Series Aggregation Services Routers
Software Configuration Guide:
http://www.cisco.com/en/US/docs/routers/asr1000/configuration/guide/chassis/using_cli.html
Cisco Product Support Resources
http://www.cisco.com/web/psa/products/index.html
Support area on Cisco.com (also search for documentation by task or product)
http://www.cisco.com/en/US/support/index.html
White Paper: Cisco IOS Reference Guide
http://www.cisco.com/en/US/products/sw/iosswrel/ps1828/products_white_paper09186a00801830
5e.shtml
Software Download Center (downloads; tools; licensing, registration, advisory, and general
information) (requires Cisco.com User ID and password)
http://www.cisco.com/kobayashi/sw-center/
Error Message Decoder, a tool to help you research and resolve error messages for
Cisco IOS software
http://www.cisco.com/pcgi-bin/Support/Errordecoder/index.cgi

xiii
Command Lookup Tool, a tool to help you find detailed descriptions of Cisco IOS commands
(requires Cisco.com user ID and password)
http://tools.cisco.com/Support/CLILookup
Output Interpreter, a troubleshooting tool that analyzes command output of supported
show commands
https://www.cisco.com/pcgi-bin/Support/OutputInterpreter/home.pl\
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Human Network are trademarks; Changing the Way We Work, Live, Play, and Learn and Cisco Store are service marks; and Access Registrar,
Aironet, AsyncOS, Bringing the Meeting To You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified
Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Collaboration
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the WebEx logo are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.

coincidental.


xiv

Americas Headquarters:
Cisco Systems, Inc., 170 West Tasman Drive, San Jose, CA 95134-1706 USA

Cisco IOS Voice Port Feature Roadmap
This roadmap provides information about Cisco IOS voice features involving voice port configuration.
It contains the following:
Platforms and Cisco IOS Software Images, page 7
Cisco IOS Voice Port Feature List, page 7
Note This chapter describes how to access Cisco Feature Navigator. It also lists and describes, by Cisco IOS
release, voice port features for that release.
For information about the full set of Cisco IOS voice features, see the entire Cisco IOS Voice
Configuration Libraryincluding library preface, glossary, and other documentsat
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.
html.
Platforms and Cisco IOS Software Images
Cisco IOS Voice Port Feature List
Table 1 lists voice port features by Cisco IOS release. Features that are introduced in a particular release
are available in that and subsequent releases.

8
Table 1 Voice Port Features by Cisco IOS Release
Release Feature Name Feature Description Where Documented
12.4(20)T Software-Configurable 128-ms Echo
Cancellation
Beginning in Release 12.4(20)T,
software-configurable echo
cancellation coverage was expanded
to include 80, 96, 112, and 128
milliseconds. The default is changed
from 64 ms to 128 ms.
Configuring Echo Cancellation
and Configuring Hardware
Echo Cancellation on T1/E1
Multiflex Voice/WAN
Interface Cards and
NextPort-Based Voice Tuning
and Echo Cancellation
12.4(15)XZ Enhancements to the Cisco 880 Series
Routers
Beginning in Release 12.4(15)XZ, the
Cisco 880 broadband series routers
integrate access technologies such as
the VDSL2, 3G; adds voice for
service provider-managed services
and for enterprise teleworkers. In
addition, the broadband series routers
offer UTM features such as URL
filtering and optional IEEE 802.11n
integrated access points that can be
managed using LWAPP.
Configuring Analog Voice
Ports and Configuring Digital
Voice Ports
12.3(14)T Hardware Echo Cancellation Two echo cancellation modules
(EC-MFT-32 and EC-MFT-64)
provide hardware echo cancellation
with a hard-coded tail length of
128 milliseconds for VWIC2s.
Configuring Hardware Echo
Cancellation on T1/E1
Multiflex Voice/WAN
Interface Cards
12.3(11)T NextPort-Based Voice-Tuning and
Dual-Filter G.168 Echo Canceller
NextPort dual-filter G.168 echo
canceller (EC) improves voice quality
in VoIP connections by providing
relatively less residual echo leakage,
better nonlinear processing (NLP)
timing, less clipping, and better
comfort noise generation (CNG) in
most environments.
NextPort-Based Voice Tuning
and Echo Cancellation

9
CCDE, CCENT, Cisco Eos, Cisco Lumin, Cisco Nexus, Cisco StadiumVision, Cisco TelePresence, the Cisco logo, DCE, and Welcome to the Human
Network are trademarks; Changing the Way We Work, Live, Play, and Learn and Cisco Store are service marks; and Access Registrar, Aironet,
AsyncOS, Bringing the Meeting To You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified Internetwork
Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Collaboration Without Limitation,
EtherFast, EtherSwitch, Event Center, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQ
Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, IronPort, the IronPort logo, LightStream, Linksys, MediaTone, MeetingPlace,
MeetingPlace Chime Sound, MGX, Networkers, Networking Academy, Network Registrar, PCNow, PIX, PowerPanels, ProConnect, ScriptShare,
SenderBase, SMARTnet, Spectrum Expert, StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and the WebEx logo
are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
coincidental.
12.3(7)T IP Communications High-Density
Digital Voice/Fax Network Module
This feature supports high-density
digital voice and low-density analog
voice connectivity along with data and
integrated access connectivity. The
network modules offer built-in T1/E1
ports, and include a single VIC/voice
WAN interface card (VWIC) slot for
FXS, FXO, E&M,
software-configured CAMA, DID,
BRI, or E1 and T1 cards, up to a
maximum of four T1/E1 ports. The
network modules also support up to
32 HDLC channels with an aggregate
capacity of 2.048 Mbps.
Configuring Digital Voice
Ports
12.2(13)T Enhanced ITU-T G.168 Echo
Cancellation
This feature offers improved
standards for echo cancellation
performance. Configurable tail length
is increased, up to 64 ms. Minimum
ERL is configurable to greater than or
equal to 0 dB, 3 dB, or 6 dB. Echo
suppression is no longer needed
because of faster convergence.
Table 1 Voice Port Features by Cisco IOS Release (continued)
Release Feature Name Feature Description Where Documented

10

Voice Port Configuration Overview
Voice ports are found at the intersections of packet-based networks and traditional telephony networks,
and they facilitate the passing of voice and call signals between the two networks. Physically, voice ports
connect a router or access server to a line from a circuit-switched telephony device in a PBX or the
PSTN.
Basic software configuration for voice ports describes the type of connection being made and the type
of signaling to take place over this connection. In addition to the commands for basic configuration, there
are also commands that provide fine-tuning for voice quality, enable special features, and specify
parameters to match those of proprietary PBXs.
This document includes the following chapters:
Configuring Analog Voice Ports
Configuring Digital Voice Ports
Fine-Tuning Analog and Digital Voice Ports
Configuring Hardware Echo Cancellation on T1/E1 Multiflex Voice/WAN Interface Cards
NextPort-Based Voice Tuning and Echo Cancellation
Verifying Analog and Digital Voice-Port Configurations
Troubleshooting Analog and Digital Voice Port Configurations
Not all voice-port commands are covered in this document. Some are described in the Cisco IOS ISDN
Voice Configuration Guide, Release 12.4 or the Trunk Management Features document, Cisco IOS Voice
Configuration Library, Release 12.4. The voice-port configuration commands included in this document
are fully documented in the Cisco IOS Voice Command Reference.
Finding Support Information for Platforms and Cisco IOS Software Images

2
Voice ports on routers and access servers emulate physical telephony switch connections so that voice
calls and their associated signaling can be transferred intact between a packet network and a
circuit-switched network or device. For a voice call to occur, certain information must be passed between
the telephony devices at either end of the call, such as the devices on-hook status, the lines availability,
and whether an incoming call is trying to reach a device. This information is referred to as signaling, and
to process it properly, the devices at both ends of the call segment (that is, those directly connected to
each other) must use the same type of signaling.
The devices in the packet network must be configured to convey signaling information in a way that the
circuit-switched network can understand. They must also be able to understand signaling information
received from the circuit-switched network. This is accomplished by installing appropriate voice
hardware in the router or access server and by configuring the voice ports that connect to telephony
devices or the circuit-switched network.
The following illustrations show examples of how voice ports are used.
Figure 1 shows one voice port connecting a telephone to the WAN through the router.
Figure 2 shows one voice port connected to the PSTN and another to a telephone; the router acts like
a small PBX.
Figure 3 shows how two PBXs can be connected over a WAN to provide toll bypass.
Figure 1 Telephone to WAN
Figure 2 Telephone to PSTN
Figure 3 PBX-to-PBX over a WAN
V
Voice port
1/0/0
Serial or
Ethernet port
WAN
3
7
7
5
4
V
Voice port
1/0/0
Voice port
0/0/1
PSTN
3
7
7
5
5
V
Voice port
1/0/0
PBX PBX
Serial or
Ethernet port
Serial or
Ethernet port
V
Voice port
1/0/0
WAN
3
7
7
5
6

3
Cisco provides a variety of Cisco IOS commands for flexibility in configuring voice ports to match the
physical attributes of the voice connections that are being made. Some of these connections are made
using analog means of transmission, while others use digital transmission. Table 1 shows the analog and
digital voice-port connection support of the router platforms discussed in this document.
Telephony Signaling Interfaces
Voice ports on routers and access servers physically connect the router or access server to telephony
devices such as telephones, fax machines, PBXs, and PSTN central office (CO) switches. These devices
may use any of several types of signaling interfaces to generate information about on-hook status,
ringing, and line seizure.
The routers voice-port hardware and software need to be configured to transmit and receive the same
type of signaling being used by the device with which they are interfacing so that calls can be exchanged
smoothly between the packet network and the circuit-switched network.
The signaling interfaces discussed in this document include foreign exchange office (FXO), foreign
exchange station (FXS), and receive and transmit (E&M), which are types of analog interfaces. Some
digital connections emulate FXO, FXS, and E&M interfaces, and they are discussed in the FXS and
FXO Interfaces section on page 4 and the E&M Interfaces section on page 5. It is important to know
which signaling method the telephony side of the connection is using, and to match the router
configuration and voice interface hardware to that signaling method.
The next three illustrations show how the different signaling interfaces are associated with different uses
of voice ports. In Figure 4, FXS signaling is used for end-user telephony equipment, such as a telephone
or fax machine. Figure 5 shows an FXS connection to a telephone and an FXO connection to the PSTN
at the far side of a WAN; this might be a telephone at a local office going over a WAN to a router at
headquarters that connects to the PSTN. In Figure 6, two PBXs are connected across a WAN by E&M
interfaces. This illustrates the path over a WAN between two geographically separated offices in the
same company.
Table 1 Analog and Digital Voice-Port Support on Cisco Platforms
Platform Analog Digital
Cisco 880 series (includes IAD881B,
IAD881F, C881SRST, IAD888B,
IAD888F, and C888SRST)
Yes Yes
Cisco 1750 Yes No
Cisco 2600 series Yes Yes
Cisco 3600 series Yes Yes
Cisco 3700 series Yea Yes
Cisco 7200 series No Yes
Cisco 7500 series No Yes
Cisco AS5300 No Yes
Cisco AS5350 No Yes
Cisco AS5400 No Yes
Cisco AS5800 No Yes
Cisco AS5850 No Yes
Cisco MC3810 Yes Yes

4
Figure 4 FXS Signaling Interfaces
Figure 5 FXS and FXO Signaling Interfaces
Figure 6 E&M Signaling Interfaces
FXS and FXO Interfaces
An FXS interface connects the router or access server to end-user equipment such as telephones, fax
machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an
RJ-11 connector for basic telephone equipment, keysets, and PBXs.
An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not
support E&M signaling (when local telecommunications authority permits). This interface is of value
for off-premise station applications. A standard RJ-11 modular telephone cable connects the FXO voice
interface card to the PSTN or PBX through a telephone wall outlet.
FXO and FXS interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one
of two access signaling methods: loop-start or ground-start. The type of access signaling is determined
by the type of service from the CO; standard home telephone lines use loop-start, but business telephones
can order ground-start lines instead.
V
Voice port
1/0/0
FXS FXS
3
7
7
5
7
Serial or
Ethernet port
Serial or
Ethernet port
V
Voice port
1/0/0
WAN
V
Voice port
1/0/0
FXS FXO
3
7
7
5
8
Serial or
Ethernet port
Serial or
Ethernet port
V
Voice port
1/0/0
WAN
PSTN
V
Voice port
1/0/0
PBX PBX
E&M E&M
3
7
7
5
9
Serial or
Ethernet port
Serial or
Ethernet port
V
Voice port
1/0/0
WAN

5
Loop-start is the more common of the access signaling techniques. When a handset is picked up (the
telephone goes off-hook), this action closes the circuit that draws current from the telephone company
CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is
signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the
telephone to ring.
Loop-start has two disadvantages, however, that usually are not a problem on residential telephones but
that become significant with the higher call volume experienced on business telephones. Loop-start
signaling has no means of preventing two sides from seizing the same line simultaneously, a condition
known as glare. Also, loop-start signaling does not provide switch-side disconnect supervision for FXO
calls. The telephony switch (the connection in the PSTN, another PBX, or key system) expects the
routers FXO interface, which looks like a telephone to the switch, to hang up the calls it receives through
its FXO port. However, this function is not built into the router for received calls; it operates only for
calls originating from the FXO port.
Another access signaling method used by FXO and FXS interfaces to indicate on-hook or off-hook status
to the CO is ground-start signaling. It works by using ground and current detectors that allow the network
to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for
positive recognition of connects and disconnects. For this reason, ground-start signaling is typically used
on trunk lines between PBXs and in businesses where call volume on loop-start lines can result in glare.
See the Configuring Disconnect Supervision and Configuring FXO Supervisory Disconnect Tones
sections in the Fine-Tuning Analog and Digital Voice Ports chapter for voice port commands that
configure additional recognition of disconnect signaling.
In most cases, the default voice port command values are sufficient to configure FXO and FXS voice
ports.
E&M Interfaces
Trunk circuits connect telephone switches to one another; they do not connect end-user equipment to the
network. The most common form of analog trunk circuit is the E&M interface, which uses special
signaling paths that are separate from the trunks audio path to convey information about the calls. The
signaling paths are known as the E-lead and the M-lead. The name E&M is thought to derive from the
phrase Ear and Mouth or rEceive and transMit although it could also come from Earth and Magnet. The
history of these names dates back to the days of telegraphy, when the CO side had a key that grounded
the E circuit, and the other side had a sounder with an electromagnet attached to a battery. Descriptions
such as Ear and Mouth were adopted to help field personnel determine the direction of a signal in a wire.
E&M connections from routers to telephone switches or to PBXs are preferable to FXS/FXO
connections because E&M provides better answer and disconnect supervision.
Like a serial port, an E&M interface has a data terminal equipment/data communications equipment
(DTE/DCE) type of reference. In telecommunications, the trunking side is similar to the DCE, and is
usually associated with CO functionality. The router acts as this side of the interface. The other side is
referred to as the signaling side, like a DTE, and is usually a device such as a PBX. Five distinct physical
configurations for the signaling part of the interface (Types I-V) use different methods to signal
on-hook/off-hook status, as shown in Table 2. Cisco voice implementation supports E&M Types I, II,
III, and V.

6
The physical E&M interface is an RJ-48 connector that connects to PBX trunk lines, which are classified
as either two-wire or four-wire. This refers to whether the audio path is full duplex on one pair of wires
(two-wire) or on two pair of wires (four-wire). A connection may be called a four-wire E&M circuit
although it actually has six to eight physical wires. It is an analog connection although an analog E&M
circuit may be emulated on a digital line. For more information on digital voice port configuration of
E&M signaling, see the DS0 Groups on Digital T1/E1 Voice Ports section in the Configuring Digital
Voice Ports chapter.
PBXs built by different manufacturers can indicate on-hook/off-hook status and telephone line seizure
on the E&M interface by using any of the following three types of access signaling:
Immediate-start is the simplest method of E&M access signaling. The calling side seizes the line by
going off-hook on its E-lead and sends address information as dual-tone multifrequency (DTMF)
digits (or as dialed pulses on Cisco 2600 and Cisco 3600 series routers) following a short,
fixed-length pause.
Wink-start is the most commonly used method for E&M access signaling, and is the default for
E&M voice ports. Wink-start was developed to minimize glare, a condition found in immediate-start
E&M, in which both ends attempt to seize a trunk at the same time. In wink-start, the calling side
seizes the line by going off-hook on its E-lead, then waits for a short temporary off-hook pulse, or
wink, from the other end on its M-lead before sending address information. The switch interprets
the pulse as an indication to proceed and then sends the dialed digits as DTMF or dialed pulses.
In delay-dial signaling, the calling station seizes the line by going off-hook on its E-lead. After a
timed interval, the calling side looks at the status of the called side. If the called side is on-hook, the
calling side starts sending information as DTMF digits; otherwise, the calling side waits until the
called side goes on-hook and then starts sending address information.
Toll Fraud Prevention
When a Cisco router platform is installed with a voice-capable Cisco IOS software image, appropriate
features must be enabled on the platform to prevent potential toll fraud exploitation by unauthorized
users. Deploy these features on all Cisco router Unified Communications applications that process voice
calls, such as Cisco Unified Communications Manager Express (CME), Cisco Survivable Remote Site
Telephony (SRST), Cisco Unified Border Element (UBE), Cisco IOS-based router and standalone
analog and digital PBX and public-switched telephone network (PSTN) gateways, and Cisco
contact-center VoiceXML gateways. These features include, but are not limited to, the following:
Table 2 E&M Wiring and Signaling Methods
E&M Type E-Lead Configuration
M-Lead
Configuration
Signal Battery Lead
Configuration
Signal Ground Lead
Configuration
I Output, relay to
ground
Input, referenced to
ground

II Output, relay to SG Input, referenced to
ground
Feed for M,
connected to 48V
Return for E,
galvanically isolated
from ground
III Output, relay to
ground
ground
Connected to 48V Connected to ground
V Output, relay to
ground
48V

7
Disable secondary dial tone on voice portsBy default, secondary dial tone is presented on voice
ports on Cisco router gateways. Use private line automatic ringdown (PLAR) for foreign exchange
office (FXO) ports and direct-inward-dial (DID) for T1/E1 ports to prevent secondary dial tone from
being presented to inbound callers.
Cisco router access control lists (ACLs)Define ACLs to allow only explicitly valid sources of
calls to the router or gateway, and therefore to prevent unauthorized Session Initiation Protocol (SIP)
or H.323 calls from unknown parties to be processed and connected by the router or gateway.
Close unused SIP and H.323 portsIf either the SIP or H.323 protocol is not used in your
deployment, close the associated protocol ports. If a Cisco voice gateway has dial peers configured
to route calls outbound to the PSTN using either time division multiplex (TDM) trunks or IP, close
the unused H.323 or SIP ports so that calls from unauthorized endpoints cannot connect calls. If the
protocols are used and the ports must remain open, use ACLs to limit access to legitimate sources.
Change SIP port 5060If SIP is actively used, consider changing the port to something other than
well-known port 5060.
SIP registrationIf SIP registration is available on SIP trunks, turn on this feature because it
provides an extra level of authentication and validation that only legitimate sources can connect
calls. If it is not available, ensure that the appropriate ACLs are in place.
SIP Digest AuthenticationIf the SIP Digest Authentication feature is available for either
registrations or invites, turn this feature on because it provides an extra level of authentication and
validation that only legitimate sources can connect calls.
Explicit incoming and outgoing dial peersUse explicit dial peers to control the types and
parameters of calls allowed by the router, especially in IP-to-IP connections used on CME, SRST,
and Cisco UBE. Incoming dial peers offer additional control on the sources of calls, and outgoing
dial peers on the destinations. Incoming dial peers are always used for calls. If a dial peer is not
explicitly defined, the implicit dial peer 0 is used to allow all calls.
Explicit destination patternsUse dial peers with more granularity than .T for destination patterns
to block disallowed off-net call destinations. Use class of restriction (COR) on dial peers with
specific destination patterns to allow even more granular control of calls to different destinations on
the PSTN.
Translation rulesUse translation rules to manipulate dialed digits before calls connect to the PSTN
to provide better control over who may dial PSTN destinations. Legitimate users dial an access code
and an augmented number for PSTN for certain PSTN (for example, international) locations.
Tcl and VoiceXML scriptsAttach a Tcl/VoiceXML script to dial peers to do database lookups or
additional off-router authorization checks to allow or deny call flows based on origination or
destination numbers. Tcl/VoiceXML scripts can also be used to add a prefix to inbound DID calls.
If the prefix plus DID matches internal extensions, then the call is completed. Otherwise, a prompt
can be played to the caller that an invalid number has been dialed.
Host name validationUse the permit hostname feature to validate initial SIP Invites that contain
a fully qualified domain name (FQDN) host name in the Request Uniform Resource Identifier
(Request URI) against a configured list of legitimate source hostnames.
Dynamic Domain Name Service (DNS)If you are using DNS as the session target on dial peers,
the actual IP address destination of call connections can vary from one call to the next. Use voice
source groups and ACLs to restrict the valid address ranges expected in DNS responses (which are
used subsequently for call setup destinations).
For more configuration guidance, see the Cisco IOS Unified Communications Toll Fraud Prevention
paper.

8
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Analog voice port interfaces connect routers in packet-based networks to analog two-wire or four-wire
analog circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and
four-wire circuits connect to PBXs. Connections to the PSTN central office (CO) are typically made with
digital interfaces.
This chapter describes how to configure analog voice ports and covers the following topics:
Prerequisites for Configuring Analog Voice Ports, page 19
Information About Analog Voice Hardware, page 20
Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports, page 22
Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with
High-Performance Compression Modules, page 26
Three other sections in this document provide help with fine-tuning and troubleshooting:
Fine-Tuning Analog and Digital Voice Ports, page 51
Verifying Analog and Digital Voice-Port Configurations, page 125
Troubleshooting Analog and Digital Voice Port Configurations, page 137
Prerequisites for Configuring Analog Voice Ports
Obtain two- or four-wire line service from your service provider or from a PBX.
Complete your companys dial plan.
Establish a working telephony network based on your companys dial plan.
Install at least one other network module or WAN interface card to provide the connection to the
network LAN or WAN.
Establish a working IP and Frame Relay or ATM network. For more information about configuring
IP, refer to the Cisco IOS IP Configuration Guide, Release 12.4.
Install appropriate voice processing and voice interface hardware on the router. See the Information
About Analog Voice Hardware section on page 20.
Gather the following information about the telephony connection of the voice port:
Information About Analog Voice Hardware
20
Telephony signaling interface: FXO, FXS, or E&M
Locale code (usually the country) for call progress tones
If FXO, type of dialing: DTMF (touch-tone) or pulse
If FXO, type of start signal: loop-start or ground-start
If E&M, type: I, II, III, or V
If E&M, type of line: two-wire or four-wire
If E&M, type of start signal: wink, immediate, delay-dial
If you are connecting a voice-port interface to a PBX, it is important to understand the PBXs wiring
scheme and timing parameters. This information should be available from your PBX vendor or the
reference manuals that accompany your PBX.
Note The slot and port numbering of interface cards differs for each of the voice-enabled routers. For the
specific slot and port designations for your hardware platform, refer to the Cisco Interface Cards
Hardware Installation Guide. More current information may be available in the release notes for the
Cisco IOS software you are using.
This section describes the general types of analog voice port hardware available for the following router
platforms:
Cisco 880 Series Routers, page 20
Cisco 1750 Modular Router, page 21
Cisco 2600 Series, Cisco 3600 Series, and Cisco 3700 Series Routers, page 21
Cisco MC3810, page 22
Note For current information about supported hardware, refer to the release notes for the platform and
Cisco IOS release being used.
Cisco 880 Series Routers
Beginning with Cisco IOS Release 12.4(15)XZ, the Cisco 880 series fixed router platforms support the
implementation of analog (FXS/DID/FXO) and digital (BRI S/T) voice ports. The IAD881B, IAD881F,
IAD888B, and IAD888F models support voice interface FXS or BRI. The IAD881F and IAD888F
models have four FXS ports and the IAD881B and IAD888B models support two ports for ISDN BRI
digital voice interface.
In the IAD881B and IAD888B models, the voice BRI interface presents an ISDN S/T interface to
connect either to an NT1 terminating an ISDN telephone network (TE-side) or to a TE user device such
as an ISDN telephone or PBX (NT-side). In the IAD881B and IAD888B models, the BRI interface is
available as the primary voice interface and is intended to be connected to a PBX (network side trunk).
All the voice interfaces are onboard though they are recognized as a 4-port FXS VIC and a 2-port BRI
VIC in order to leverage existing voice drivers.
21
Note If the primary voice interface is FXS and the backup is BRI, then ports 0, 1, 2, and 3 are analog voice
ports, and ports 4 and 5 are digital. If the primary voice interface is BRI, then ports 1, 2, 3, and 4 are
digital.
The C881 and C888 SRST models automatically detect a failure occuring in the network and initiate a
process to auto-configure the router. This process provides call-processing backup redundancy for the
IP and FXS phones and helps to ensure that telephony capabilities stay operational. All the IP or analog
phones hanging off of a telecommuter site are controlled by the headquarters office call control
(Cisco Unified CallManager or CallManager Express). In case of a WAN failure, the telecommuter
router allows all phones to re-register to it in SRST mode and allow all inbound and outbound dialing to
be routed off to the PSTN (using back up FXO or BRI port). Upon restoration of WAN connectivity, the
system automatically shifts call processing back to the primary Cisco Unified Call Manager cluster.
Cisco 1750 Modular Router
The Cisco 1750 modular router provides VoIP functionality and can carry voice traffic (for example,
telephone calls and faxes) over an IP network. To make a voice connection, the router must have a
supported voice interface card (VIC) installed. The Cisco 1750 router supports two slots for either WAN
interface cards (WICs) or VICs and supports one VIC-only slot. For analog connections, two-port VICs
are available to support FXO, FXS, and E&M signaling. VICs provide direct connections to telephone
equipment (analog phones, analog fax machines, key systems, or PBXs) or to a PSTN.
Cisco 2600 Series, Cisco 3600 Series, and Cisco 3700 Series Routers
The Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers are modular, multifunction
platforms that combine dial access, routing, LAN-to-LAN services, and multiservice integration of
voice, video, and data in the same device.
Voice network modules installed in Cisco 2600 series, Cisco 3600 series, or Cisco 3700 series routers
convert telephone voice signals into data packets that can be transmitted over an IP network. The voice
network modules have no connectors; VICs installed in the network modules provide connections to the
telephone equipment or network. VICs work with existing telephone and fax equipment and are
compatible with H.323 standards for audio and video conferencing.
For analog telephone connections, low-density voice/fax network modules that contain either one or two
VIC slots are installed in the network module slots. Each VIC is specific to a particular telephone
signaling interface (FXS, FXO, or E&M); therefore, the VIC determines the type of signaling on that
module.
For more information, refer to the following:
Hardware installation documents for Cisco 2600 series
Cisco Network Modules Hardware Installation Guide
Cisco Interface Cards Installation Guide
How to Configure Analog Voice Ports
22
Cisco MC3810
To support analog voice circuits, a Cisco MC3810 must be equipped with an AVM, which supports six
analog voice ports. When you install specific signaling modules known as analog personality
modules (APMs), the analog voice ports may be equipped for the following signaling types in various
combinations: FXS, FXO, and E&M. For FXS, the analog voice ports use an RJ-11 connector interface
to connect to analog telephones or fax machines (two-wire) or to a key system (four-wire). For FXO, the
analog voice ports use an RJ-11 physical interface to connect to a CO trunk. For E&M connections, the
analog voice ports use an RJ-1CX physical interface to connect to an analog PBX (two-wire or
four-wire).
Optional high-performance voice compression modules (HCMs) can replace standard voice compression
modules (VCMs) to operate according to the voice compression coding algorithm (codec) specified
when the Cisco MC3810 concentrator is configured. The HCM2 provides four voice channels at high
codec complexity and eight channels at medium complexity. The HCM6 provides 12 voice channels at
high complexity and 24 channels at medium complexity. One or two HCMs can be installed in a
Cisco MC3810, but an HCM may not be combined with a VCM in one chassis.
For more information, refer to the Cisco MC3810 Multiservice Concentrator Hardware Installation
Guide.
Cisco IOS release you are using.
To configure analog voice ports, complete the following tasks:
Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports
Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with
High-Performance Compression Modules
Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports
This section describes commands for basic analog voice port configuration.
All the data recommended in the Prerequisites for Configuring Analog Voice Ports section on page 19
should be gathered before you start this procedure.
If you are configuring a Cisco MC3810 that has HCMs, you should also configure the codec complexity
by performing the tasks in the Configuring Codec Complexity for Analog Voice Ports on the
Cisco MC3810 with High-Performance Compression Modules section on page 26.
Note If you have a Cisco MC3810 or Cisco 3660 router, the compand-type a-law command must be
configured on the analog ports only. The Cisco 2660, Cisco 3620, and Cisco 3640 routers do not require
the configuration of the compand-type a-law command. However, if you request a list of commands,
the compand-type a-law command will display.
23
In addition to the basic voice port parameters described in this section, there are commands that allow
voice port configurations to be fine-tuned. In most cases, the default values for fine-tuning commands
are sufficient for establishing FXO and FXS voice port configurations. E&M voice ports are more likely
to require some configuration. If it is necessary to change some of the voice port values to improve voice
quality or to match parameters on proprietary PBXs to which you are connecting, use the commands in
this section and also in the Fine-Tuning Analog and Digital Voice Ports chapter.
After the voice port has been configured, make sure that the ports are operational by performing the tasks
described in the following chapters:
For more information on these and other voice port commands, refer to the Cisco IOS Voice Command
Reference.
Note The commands, keywords, and arguments that you are able to use may differ slightly from those
presented here, based on your platform, Cisco IOS release, and configuration. When in doubt, use
Cisco IOS command help to determine the syntax choices that are available.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice-port port (for Cisco 880 series)
or
voice-port slot/port (for Cisco 1750 and Cisco MC3810)
or
voice-port slot/subunit/port (for Cisco 2600, Cisco 3600, and Cisco 3700 series)
4. signal {loop-start | ground-start} (for FXO and FXS)
or
signal {wink-start | immediate-start | delay-dial} (for E&M)
5. cptone locale
6. dial-type {dtmf | pulse}
7. operation {2-wire | 4-wire}
8. type {1 | 2 | 3 | 5}
9. ring frequency {25 | 50} (Cisco 1750 router and Cisco 2600, Cisco 3600, and Cisco 3700 series)
or
ring frequency {20 | 30} (Cisco MC3810)
10. ring number number
11. ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07
| pattern08 | pattern09 | pattern10 | pattern11 | pattern12] | [define pulse interval]}
12. description string
13. no shutdown
24
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Enables privileged EXEC mode.
Enter your password if prompted.
Step 2 configure terminal
Example:
Enters global configuration mode.
Step 3 voice-port slot/port
Example:
Router(config)# voice-port 1/0
or
voice-port slot/subunit/port
Example:
Router(config)# voice-port 1/0/0
Enters voice-port configuration mode.
Note The slash must be entered between slot
and port.
Valid entries vary by router platform; enter the
show voice port summary command for
available values.
Note For the Cisco 880 series platforms, the
command syntax does not include a slot
number, only the port is identified.
If the primary voice interface is FXS and
the backup is BRI, then ports 0, 1, 2, and 3
are analog voice ports, and ports 4 and 5
are digital. If the primary voice interface is
BRI, then ports 1, 2, 3, and 4 are digital.
Step 4 signal {loop-start | ground-start}
Example:
Router(config-voiceport)# signal ground-start
or
signal {wink-start | immediate-start | delay-dial}
Example:
Router(config-voiceport)# signal wink-start
Selects the access signaling type to match that of
the telephony connection you are making.
Note Configuring the signal keyword for one
voice port on a Cisco 2600 or Cisco 3600
series router VIC changes the signal value
for both ports on the VIC.
Step 5 cptone locale
Example:
Router(config-voiceport)# cptone us
Selects the two-letter locale for the voice call
progress tones and other locale-specific
parameters to be used on this voice port.
Cisco routers comply with the ISO 3166
locale name standards. To see valid choices,
enter a question mark (?) following the
cptone command.
The default is us.
25
Step 6 dial-type {dtmf | pulse}
Example:
Router(config-voiceport)# dial-type dtmf
(FXO only) Specifies the dialing method for
outgoing calls.
Step 7 operation {2-wire | 4-wire}
Example:
Router(config-voiceport)# operation 4-wire
(E&M only) Specifies the number of wires used
for voice transmission at this interface (the audio
path only, not the signaling path).
The default is 2-wire.
Step 8 type {1 | 2 | 3 | 5}
Example:
Router(config-voiceport)# type 2
(E&M only) Specifies the type of E&M interface
to which this voice port is connecting. See Table 2
in the Voice Port Configuration Overview chapter
for an explanation of E&M types.
The default is 1.
Step 9 ring frequency {25 | 50}
or
ring frequency {20 | 30}
Example:
Router(config-voiceport)# ring frequency 50
(FXS only) Selects the ring frequency, in hertz,
used on the FXS interface. This number must
match the connected telephony equipment and
may be country-dependent. If the ring frequency is
not set properly, the attached telephony device
may not ring or it may buzz.
The keyword default is 25 on the Cisco 1750
router, Cisco 2600 and Cisco 3600 series
routers; and 20 on the Cisco MC3810.
Step 10 ring number number
Example:
Router(config-voiceport)# ring number 1
(FXO only) Specifies the maximum number of
rings to be detected before an incoming call is
answered by the router.
The default is 1.
Step 11 ring cadence {[pattern01 | pattern02 | pattern03 |
pattern04 | pattern05 | pattern06 | pattern07 |
pattern12] | [define pulse interval]}
Example:
Router(config-voiceport)# ring cadence pattern01
(FXS only) Specifies an existing pattern for ring,
or defines a new one. Each pattern specifies a
ring-pulse time and a ring-interval time.
The default is the pattern specified by the
cptone locale that has been configured.
Step 12 description string
Example:
Router(config-voiceport)# description 255
Attaches a text string to the configuration that
describes the connection for this voice port. This
description appears in various displays and is
useful for tracking the purpose or use of the voice
port. The string argument is a character string
from 1 to 255 characters in length.
The default is that there is no text string
(describing the voice port) attached to the
configuration.
Step 13 no shutdown
Example:
Router(config-voiceport)# no shutdown
Activates the voice port. If a voice port is not being
used, shut the voice port down with the shutdown
command.
Command Purpose
26
Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810
with High-Performance Compression Modules
The term codec stands for coder-decoder. A codec is a particular method of transforming analog voice
into a digital bit stream (and vice versa) and also refers to the type of compression used. Several different
codecs have been developed to perform these functions, and each one is known by the number of the
International Telecommunication Union-Telecommunication Standardization Sector (ITU-T) standard
in which it is defined. For example, two common codecs are the G.711 and the G.729 codecs. The various
codecs use different algorithms to encode analog voice into digital bit-streams and have different bit
rates, frame sizes, and coding delays associated with them. The codecs also differ in the type of
perceived voice quality they achieve. Specialized hardware and software in the digital signal processors
(DSPs) perform codec transformation and compression functions, and different DSPs may offer different
selections of codecs.
Select the same type of codec as the one that is used at the other end of the call. For instance, if a call
was coded with a G.729 codec, it must be decoded with a G.729 codec. Codec choice is configured in
dial peers. For more information, refer to the Dial Peer Configuration on Voice Gateway Routers
document.
Codec complexity refers to the amount of processing power that a codec compression method requires.
The greater the codec complexity, the fewer the calls that the DSP interfaces can handle. Codec
complexity is either medium or high. The default is medium. All medium-complexity codecs can also
run in high-complexity mode, but fewer (usually half as many) channels are available per DSP. The
codec complexity value determines the choice of codecs that are available in the dial peers when the
codec command has been configured. For details on the number of calls that can be handled
simultaneously using each of the codec standards, refer to the entries for the codec and codec
complexity commands in the Cisco IOS Voice Command Reference.
To configure codec complexity on the Cisco MC3810 using HCMs, use the following commands:
SUMMARY STEPS
1. enable
2. show voice dsp
4. voice-card 0
5. codec complexity {high | medium}
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Step 2 show voice dsp
Example:
Router# show voice dsp
Checks the DSP voice channel activity. If any DSP
voice channels are in the busy state, the codec
complexity cannot be changed. When all the DSP
channels are in the idle state, continue to Step 3.
27
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coincidental.
Example:
Step 4 voice-card 0
Example:
Router(config)# voice-card 0
Enters voice-card configuration mode and
specifies voice card 0.
Step 5 codec complexity {high | medium}
Example:
Router(config-voicecard)# codec complexity high
Specifies codec complexity based on the codec
standard being used. This setting restricts the
codecs available in dial peer configuration. All
voice cards in a router must use the same codec
complexity setting.
Note If two HCMs are installed, this command
configures both HCMs at once.
Command Purpose
28

The digital voice port commands discussed in this section configure channelized T1 or E1 connections;
for information on ISDN connections, refer to the Cisco IOS ISDN Voice Configuration Guide.
The T1 or E1 lines that connect a telephony network to the digital voice ports on a router or access server
contain channels for voice calls; a T1 line contains 24 full-duplex channels or timeslots, and an E1 line
contains 30. The signal on each channel is transmitted at 64 kbps, a standard known as Digital Signal 0
(DS0); the channels are known as DS0 channels. The ds0-group command creates a logical voice port
(a DS0 group) from some or all of the DS0 channels, which allows you to address those channels easily,
as a group, in voice-port configuration commands.
Digital voice ports are found at the intersection of a packet voice network and a digital, circuit-switched
telephone network. The digital voice port interfaces that connect the router or access server to T1 or E1 lines
pass voice data and signaling between the packet network and the circuit-switched network.
Signaling is the exchange of information about calls and connections between two ends of a
communication path. For instance, signaling communicates to the calls endpoints whether a line is idle
or busy, whether a device is on-hook or off-hook, and whether a connection is being attempted. An
endpoint can be a central office (CO) switch, a PBX, a telephony device such as a telephone or fax
machine, or a voice-equipped router acting as a gateway. There are two aspects to consider about
signaling on digital lines: one aspect is the actual information about line and device states that is
transmitted, and the second aspect is the method used to transmit the information on the digital lines.
The actual information about line and device states is communicated over digital lines using signaling
methods that emulate the methods used in analog circuit-switched networks: Foreign Exchange Service
(FXS), Foreign Exchange Office (FXO), and Ear and Mouth (E&M).
The method used to transmit the information describes the way that the emulated analog signaling is
transmitted over digital lines, which may be common-channel signaling (CCS) or channel-associated
signaling (CAS). CCS sends signaling information down a dedicated channel and CAS takes place
within the voice channel itself. This chapter describes CAS, which is sometimes called robbed-bit
signaling because user bandwidth is robbed by the network for signaling. A bit is taken from every sixth
frame of voice data to communicate on- or off-hook status, wink, ground-start, dialed digits, and other
information about the call.
In addition to setting up and tearing down calls, CAS provides the receipt and capture of dialed number
identification (DNIS) and automatic number identification (ANI) information, which are used to support
authentication and other functions. The main disadvantage of CAS is its use of user bandwidth to
perform these signaling functions.

Prerequisites for Configuring Digital Voice Ports
30
For signaling to pass between the packet network and the circuit-switched network, both networks must
use the same type of signaling. The voice ports on Cisco routers and access servers can be configured to
match the signaling of most COs and PBXs, as explained in this document.
This chapter discusses the following topics:
Prerequisites for Configuring Digital Voice Ports, page 30
Information About Digital Voice Hardware, page 31
How to Configure Digital T1/E1 Voice Ports, page 37
Prerequisites for Configuring Digital Voice Ports
Digital T1 or E1 packet voice capability requires specific service, software, and hardware:
Obtain T1 or E1 service from the service provider or from your PBX.
Create your companys dial plan.
Establish a working telephony network based on your companys dial plan.
Establish a connection to the network LAN or WAN.
Set up a working IP and Frame Relay or ATM network. For more information about configuring IP,
refer to the Cisco IOS IP Configuration Guide.
Install appropriate voice processing and voice interface hardware on the router. See the Information
About Digital Voice Hardware section on page 31.
(Cisco 2600 and Cisco 3600 series routers) For digital T1 packet voice trunk network modules,
install Cisco IOS Release 12.2(1) or a later release. The minimum DRAM memory requirements are
as follows:
32 MB, with one or two T1 lines
48 MB, with three or four T1 lines
64 MB, with five to ten T1 lines
128 MB, with more than ten T1 lines
The memory required for high-volume applications may be greater than that listed. Support for
digital T1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature
set requires 8 MB of flash memory; other Plus feature sets require 16 MB.
(Cisco 2600 and Cisco 3600 series routers) For digital E1 packet voice trunk network modules,
install Cisco IOS Release 12.2(1) or a later release. The minimum DRAM memory requirements are:
48 MB, with one or two E1s
64 MB, with three to eight E1s
128 MB, with 9 to 12 E1s
For high-volume applications, the memory required may be greater than these minimum values.
Support for digital E1 packet voice trunk network modules is included in Plus feature sets. The
IP Plus feature set requires 16 MB of flash memory.
Before you can run the IP Communications High-Density Digital Voice/Fax Network Module
feature on T1/E1 interfaces, you must install an IP Plus image (minimum) of Cisco IOS
Release 12.3(7)T or a later release.
(Cisco MC3810 concentrators) HCMs require Cisco IOS Release 12.2(1) or a later release.

Information About Digital Voice Hardware
31
(Cisco 7200 and Cisco 7500 series routers) For digital T1/E1 voice port adapters, install Cisco IOS
Release 12.2(1) or a later release. The minimum DRAM memory requirement to support T1/E1
high-capacity digital voice port adapters is 64 MB.
The memory required for high-volume applications may be greater than that listed. Support for
T1/E1 high-capacity digital voice port adapters is included in Plus feature sets. The IP Plus feature
set requires 16 MB of flash memory.
Gather the following information about the telephony network connection of the voice port:
Line interface: T1 or E1
Signaling interface: FXO, FXS, or E&M. If the interfaces are PRI or BRI, refer to the Cisco IOS
ISDN Voice Configuration Guide, and Cisco IOS Terminal Services Configuration Guide.
Line coding: AMI or B8ZS for T1, and AMI or HDB3 for E1
Framing format: SF (D4) or ESF for T1, and CRC4 or no-CRC4 for E1
Number of channels
Note The slot and port numbering of interface cards differs for each of the voice-enabled routers. For specific
slot and port designations, refer to the hardware installation documentation for your router platform.
More current information may be available in the release notes that accompany the Cisco IOS software
you are using.
After the controllers have been configured, the show voice port summary command can be used to
determine available voice port numbers. If the show voice port command and a specific port number is
entered, the default voice-port configuration for that port displays.
The following is show voice port summary sample output for a Cisco MC3810:
Router# show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:18 19 fxo-ls up dorm idle on-hook y
0:23 24 e&m-imd up dorm idle idle y
This section briefly describes digital voice hardware on the following platforms:
Cisco 880 Series Routers, page 32
Cisco 2600 Series, Cisco 3600 Series, and Cisco 3700 Series Routers, page 32
Cisco 7200 and Cisco 7500 Series Routers, page 33
Cisco AS5300, page 34
Cisco AS5350 and Cisco AS5400 Universal Gateways, page 34
Cisco AS5800, page 35
Cisco AS5850 Universal Gateway, page 35

32
Cisco Catalyst 6500 Series Switches and Cisco 7600 Series Routers with Communication Media
Modules, page 36
Cisco MC3810, page 36
Cisco 880 Series Routers
Beginning with Cisco IOS Release 12.4(15)XZ, the Cisco 880 series fixed router platforms support the
implementation of analog (FXS/DID/FXO) and digital (BRI S/T) voice ports. The IAD881B, IAD881F,
IAD888B, and IAD888F models support voice interface FXS or BRI. The IAD881F and IAD888F
models have four FXS ports and the IAD881B and IAD888B models support two ports for ISDN BRI
digital voice interface.
In the IAD881B and IAD888B models, the voice BRI interface presents an ISDN S/T interface to
connect either to an NT1 terminating an ISDN telephone network (TE-side) or to a TE user device such
as an ISDN telephone or PBX (NT-side). In the IAD881B and IAD888B models, the BRI interface is
available as the primary voice interface and is intended to be connected to a PBX (network side trunk).
All the voice interfaces are onboard though they are recognized as a 4-port FXS VIC and a 2-port BRI
VIC in order to leverage existing voice drivers.
Note If the primary voice interface is FXS and the backup is BRI, then ports 0, 1, 2, and 3 are analog voice
ports, and ports 4 and 5 are digital. If the primary voice interface is BRI, then ports 1, 2, 3, and 4 are
digital.
The C881and C888 SRST models automatically detect a failure occuring in the network and initiate a
process to auto-configure the router. This process provides call-processing backup redundancy for the
IP and FXS phones and helps to ensure that telephony capabilities stay operational. All the IP or analog
phones hanging off of a telecommuter site are controlled by the headquarters office call control
(Cisco Unified CallManager or CallManager Express). In case of a WAN failure, the telecommuter
router allows all phones to re-register to it in SRST mode and allow all inbound and outbound dialing to
be routed off to the PSTN (using back up FXO or BRI port). Upon restoration of WAN connectivity, the
system automatically shifts call processing back to the primary Cisco Unified Call Manager cluster.
Cisco 2600 Series, Cisco 3600 Series, and Cisco 3700 Series Routers
Digital voice hardware on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series modular access
routers includes the high-density voice (HDV) network module and the multiflex trunk (MFT)
voice/WAN interface card (VWIC). When an HDV is used in conjunction with an MFT and packet voice
DSP modules (PVDMs), the HDV module is also called a digital packet voice trunk network module.
The digital T1 or E1 packet voice trunk network module supports T1 or E1 applications, including
fractional use. The T1 version integrates a fully managed DSU/CSU, and the E1 version includes a fully
managed DSU. The digital T1 or E1 packet voice trunk network module provides per-channel T1 or E1
data rates of 64 or 56 kbps for WAN services (Frame Relay or leased line).
Digital T1 or E1 packet voice trunk network modules allow enterprises or service providers, using the
voice-equipped routers as customer premises equipment (CPE), to deploy digital voice and fax relay.
These network modules receive constant bit-rate telephony information over T1 or E1 interfaces and

33
convert that information to a compressed format so that it can be sent over a packet network. The digital
T1 or E1 packet voice trunk network modules can connect either to a PBX (or similar telephony device)
or to a CO to provide PSTN connectivity.
The MFT VWICs that are used in the packet voice trunk network modules are available in one- and
two-port configurations for T1 and for E1, and in two-port configurations with drop-and-insert capability
for T1 and E1. MFTs support the following kinds of traffic:
Data. As WICs for T1 or E1 applications, including fractional data line use, the T1 version includes
a fully managed DSU/CSU, and the E1 version includes a fully managed DSU.
Packet voice. As VWICs included with the digital T1 or E1 packet voice trunk network module to
provide connections to PBXs and COs, the MFTs enable packet voice applications.
Multiplexed voice and data. Some two-port T1 or E1 VWICs can provide drop-and-insert
multiplexing services with integrated DSU/CSUs. For example, when used with a digital T1 packet
voice trunk network module, drop-and-insert allows 64-kbps DS0 channels to be taken from one T1
and digitally cross-connected to 64-kbps DS0 channels on another T1. Drop and insert, sometimes
called time-division multiplex (TDM) cross-connect, uses circuit switching rather than the digital
signal processors (DSPs) that VoIP technology employs. (Drop-and-insert is described in the Trunk
Management Features document.
The digital T1 or E1 packet voice trunk network module contains five 72-pin Single In-line Memory
Module (SIMM) sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a
single 72-pin PVDM, and there must be at least one packet voice data module (PVDM-12) in the network
module to process voice calls. Each PVDM holds three DSPs, so with five PVDM slots populated, a total
of 15 DSPs are provided. High-complexity codecs support two simultaneous calls on each DSP, and
medium-complexity codecs support four calls on each DSP. A digital T1 or E1 packet voice trunk
network module can support the following numbers of channels:
When the digital T1 or E1 packet voice trunk network module is configured for high-complexity
codec mode, up to six voice or fax calls can be completed per PVDM-12, using the following codecs:
G.711, G.726, G.729, G729 Annex A (E1), G.729 Annex B, G.723.1, G723.1 Annex A (T1), G.728,
and fax relay.
When the digital T1 or E1 packet voice trunk network module is configured for medium-complexity
codec mode, up to 12 voice or fax calls can be completed per PVDM-12, using the following codecs:
G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay.
For more information, refer to the following:
Cisco Network Modules Hardware Installation Guide
Cisco Interface Cards Installation Guide
Cisco 7200 and Cisco 7500 Series Routers
Cisco 7200 and Cisco 7500 series routers support multimedia routing and bridging with a wide variety
of protocols and media types. The Cisco 7000 family Versatile Interface Processor (VIP) is based on a
reduced instruction set computing (RISC) engine optimized for I/O functions. To this engine are attached
one or two port adapters or daughter boards, which provide the media-specific interfaces to the network.
The network interfaces provide connections between the routers peripheral component interconnect
(PCI) buses and external networks. Port adapters can be placed in any available port adapter slot, in any
desired combination.

34
T1/E1 high-capacity digital voice port adapters for Cisco 7200 and Cisco 7500 series routers allow
enterprises or service providers, using the equipped routers as CPE, to deploy digital voice and fax relay.
These port adapters receive constant bit-rate telephony information over T1/E1 interfaces and can
convert that information to a compressed format for transmission as VoIP. Two types of digital voice port
adapters are supported on Cisco 7200 and Cisco 7500 series routers: two-port high-capacity (up to 48 or
120 channels of compressed voice, depending on codec choice), and two-port moderate capacity (up to
24 or 48 channels of compressed voice). These single-width port adapters incorporate two universal
ports configurable for either T1 or E1 connection, for use with high-performance DSPs. Integrated
CSU/DSUs, echo cancellation, and DS0 drop-and-insert functionality eliminate the need for external
line termination devices and multiplexers.
For more information, refer to the following publications:
Cisco 7200 VXR Installation and Configuration Guide
Cisco 7500 Series Installation and Configuration Guide
Two-Port T1/E1 Moderate-Capacity and High-Capacity Digital Voice Port Adapter Installation and
Configuration
Cisco AS5300
The Cisco AS5300 includes three expansion slots. One slot is for either an Octal T1/E1/PRI feature card
(eight ports) or a Quad T1/E1/PRI feature card (four ports), and the other two can be used for voice/fax
or modem feature cards. Because a single voice/fax feature card (VFC) can support up to 48 (T1) or 60
(E1) voice calls, the Cisco AS5300 can support a total of 96 or 120 simultaneous voice calls.
Cisco AS5300 VFCs are coprocessor cards, each with a powerful reduced instruction set computing
(RISC) engine and dedicated, high-performance DSPs to ensure predictable, real-time voice processing.
The design couples this coprocessor with direct access to the Cisco AS5300 routing engine for
streamlined packet forwarding.
Hardware installation documents for Cisco AS5300
Configuration documents for Cisco AS5300
Cisco AS5350 and Cisco AS5400 Universal Gateways
The Cisco AS5350 and Cisco AS5400 universal gateways are versatile data and voice communications
platforms that provide the functions of a gateway, router, and digital modems in a single modular chassis.
The gateways are intended for Internet service providers (ISPs), telecommunications carriers, and other
service providers that offer managed Internet connections, and also medium to large sites that provide
both digital and analog access to users on an enterprise network.
The cards that reside in the Cisco AS5350 and AS5400 chassis, sometimes referred to as dial feature
cards (DFCs), are of two types: trunk cards, which provide an E1, T1, or T3 interface, and universal port
cards, which host the universal DSPs that dynamically handle voice, dial, and fax calls.

35
Cisco AS5350 and AS5400 Universal Gateway Card Installation Guide
Cisco AS5350 and AS5400 Universal Gateway Software Configuration Guide
Cisco AS5800
The Cisco AS5800 has two primary system components: the Cisco 5814 dial shelf (DS), which holds
channelized trunk cards and connects to the PSTN, and the Cisco 7206 router shelf (RS), which holds
port adapters and connects to the IP backbone.
The dial shelf acts as the access concentrator by accepting and consolidating all types of remote traffic,
including voice, dial-in analog and digital ISDN data, and industry-standard WAN and remote
connection types. The dial shelf also contains controller cards voice feature cards, modem feature cards,
trunk cards, and dial shelf interconnect cards.
One or two dial shelf controllers (DSCs) provide clock and power control to the dial shelf cards. Each
DSC contains a block of logic that is referred to as the common logic and system clocks. This block of
logic can use a variety of sources to generate the system timing, including an E1 or T1/T3 input signal
from the BNC connector on the front panel of the DSC. The configuration commands for the master
clock specify the various clock sources and a priority for each source (see the Clock Sources on Digital
T1/E1 Voice Ports section on page 49).
The Cisco AS5800 voice feature card is a multi-DSP coprocessing board and software package that adds
VoIP capabilities to the Cisco AS5800 platform. The Cisco AS5800 voice feature card, when used with
other cards such as LAN/WAN and modem cards, provides a gateway for up to 192 packetized voice/fax
calls and 360 data calls per card. A Cisco AS5800 can support up to 1344 voice calls in split-dial-shelf
configuration with two 7206VXR router shelves.
Cisco AS5800 Access Server Hardware Installation Guide
Cisco AS5800 Operation, Administration, Maintenance, and Provisioning Guide
Cisco AS5850 Universal Gateway
The Cisco AS5850 is a high-density ISDN and port WAN aggregation system that provides both digital
and analog call termination. It is intended to be used in service-provider dial point-of-presence (POP) or
centralized-enterprise dial environments. The feature cards and the route switch controller (RSC)
communicate over a nonblocking interconnect that supports Fast Ethernet and full-duplex service.
The Cisco AS5850 contains ingress interfaces (CT3 and CE1/PRI) that terminate ISDN and modem calls
and break out individual calls (DS0s) from the appropriate telco services. Digital or ISDN calls are
terminated on the trunk-card HDLC controllers, and analog calls are sent to port resources on the same
card or on separate port cards. As a result, any DS0 can be mapped to any HDLC controller or port
module. Unlike the Cisco AS5800, trunk-termination and port-handling services can be performed on
the same card in the same slot.
Cisco AS5850 Hardware Installation Guide
Cisco AS5850 Universal Gateway Operations, Administration, Maintenance, and Provisioning
Guide

36
Cisco Catalyst 6500 Series Switches and Cisco 7600 Series Routers with
Communication Media Modules
The Communication Media Module (CMM) acts as the VoIP gateway and media services module by
using Media Gateway Control Protocol (MGCP), H.323, and SIP protocols with Cisco CallManager and
other call agents. The CMM can support single or multiple Cisco CallManagers in an IP communication
network.
These VoIP gateway and media services features are provided through the four different types of CMM
port adapters as shown in Table 1.
For specific configuration information for the Catalyst 6500 series and Cisco 7600 series, see the
following documents:
Cisco 6500 and 7600 series Manager Installation Guide, Release 2.1
Cisco 6500 and 7600 series Manager User Guide, Release 2.1
Cisco 6500 and 7600 series Manager Release Notes, Release 2.1
For specific installation and configuration information for the CMM, see the following document:
Catalyst 6500 Series and Cisco 7600 Series CMM Installation and Verification Note
Cisco Communication Media Module Voice Features for Catalyst 6500 Series and Cisco 7600 Series
Cisco MC3810
To support a T1 or E1 digital voice interface, the Cisco MC3810 must be equipped with a digital voice
interface card (DVM). The DVM interfaces with a digital PBX, channel bank, or video codec. It supports
up to 24 channels of compressed digital voice at 8 kbps, or it can cross-connect channelized data from
user equipment directly onto the routers trunk port for connection to a carrier network.
Table 1 CMM Port Adapters
CMM Port Adapters Description
WS-SVC-CMM-6T1
WS-SVC-CMM-6E1
The 6-port T1 and E1 port adapters have onboard digital signal processor
(DSP) resources that allow you to connect the interfaces to the public
switched telephone network (PSTN) or private branch exchanges (PBXs)
through T1/E1R2 Channel Associated Signaling (CAS) or T1/E1 ISDN
Primary Rate Interface (PRI). The DSP resources on the port adapters
provide packetization, echo cancellation, fax relay, tone detection and
generation, concealment, and jitter buffers.
WS-SVC-CMM-24FXS The 24-port FXS port adapter has onboard DSP resources that allow the FXS
interfaces to emulate the central office (CO) or PBX analog trunk lines by
providing service to analog phones and fax machines, which behave as if
connected to a standard CO or PBX line.
WS-SVC-CMM-ACT The ACT port adapter has DSP resources for conferencing, transcoding, and
media termination point (MTP) services. A CMM with an ACT port adapter
supports a single conference with up to 64 participants. A single ACT port
adapter supports up to 128 audio conference ports, which can be distributed
among different conferences of two or more parties.

How to Configure Digital T1/E1 Voice Ports
37
The DVM is available with a balanced interface using an RJ-48 connector or with an unbalanced
interface using BNC connectors.
Optional HCMs can replace standard VCMs to operate according to the voice compression coding
algorithm (codec) specified when the Cisco MC3810 is configured. The HCM2 provides 4 voice
channels at high codec complexity and 8 channels at medium complexity. The HCM6 provides 12 voice
channels at high complexity and 24 channels at medium complexity. You can install one or two HCMs
in a Cisco MC3810, but an HCM cannot be combined with a VCM in the same chassis.
Cisco MC3810 Multiservice Concentrator Hardware Installation Guide
Cisco MC3810 Multiservice Concentrator Configuration Guide
This section describes commands for the basic configuration of digital voice ports. Make sure you have
all the data recommended in the Prerequisites for Configuring Digital Voice Ports section on page 30
before starting these procedures.
The basic steps for configuring digital voice ports are described in the next three sections. They are
grouped by the configuration mode from which they are executed, as follows:
Configuring Codec Complexity on Digital T1/E1 Voice Ports, page 38
Codec complexity refers to the amount of processing power assigned to a codec method on a voice
port. On most router platforms that support codec complexity, codec complexity is selected in
voice-card configuration mode, although it is selected in DSP interface mode on the Cisco 7200 and
Cisco 7500 series.
Configuring Controller Settings for Digital T1/E1 Voice Ports, page 47
Specific line characteristics must be configured to match those of the PSTN line that is being
connected to the voice port. These are typically configured in controller configuration mode.
Configuring Basic Voice Port Parameters for Digital T1/E1 Voice Ports, page 57
Voice-port configuration mode allows many of the basic voice call attributes to be configured to
match those of the PSTN or PBX connection being made on this voice port.
In addition to the basic voice port parameters, there are commands that allow for the fine- tuning of the
voice port configurations or for configuration of optional features. In most cases, the default values for
these commands are sufficient for establishing voice port configurations. If it is necessary to change
some of these parameters to improve voice quality or to match parameters in proprietary PBXs to which
you are connecting, use the commands in the Fine-Tuning Analog and Digital Voice Ports section on
page 51.
After voice port configuration, make sure the ports are operational by following the steps described in
these chapters:
Verifying Analog and Digital Voice-Port Configurations, page 125
For more information on voice port commands, refer to the Cisco IOS Voice Command Reference.

38
Configuring Codec Complexity on Digital T1/E1 Voice Ports
This section provides two configuration task tables: one for the Cisco 2600, Cisco 3600, and Cisco 3700
series routers and the Cisco MC3810 concentrator, which use voice-card configuration mode, and the
second for the Cisco 7200 and Cisco 7500 series routers, which use DSP interface configuration mode.
The task tables can be found in the following sections:
Configuring Codec Complexity on Cisco 880 Series, Cisco 2600, Cisco 3600, Cisco 3700 Series
and Cisco MC3810, page 38
Configuring Codec Complexity on Cisco 7200 Series and Cisco 7500 Series Routers, page 45
Configuring Codec Complexity on Cisco 880 Series, Cisco 2600, Cisco 3600, Cisco 3700 Series and
Cisco MC3810
On the Cisco 880 series, Cisco 2600, Cisco 3600, Cisco 3700, Cisco 7200, and Cisco 7500 routers, codec
complexity can be configured separately for each T1/E1 digital packet voice trunk network module or
port adapter. On a Cisco MC3810, the codec complexity setting applies to both HCMs if two HCMs are
installed.
Note On Cisco 2600, Cisco 3600, and Cisco 3700 series routers with digital T1/E1 packet voice trunk network
modules, codec complexity cannot be configured if DS0 or PRI groups are configured. If DS0 or PRI
groups are configured, see the Changing Codec Complexity on Cisco 880 Series, Cisco 2600 Series,
Cisco 3600 Series, Cisco 3700 Series, and Cisco MC3810 section on page 39.
To configure codec complexity for digital voice ports on the Cisco 880 series, Cisco 2600 series,
Cisco 3600 series, and Cisco 3700 series routers, and for voice ports on HCMs on the Cisco MC3810,
use the following commands:
SUMMARY STEPS
1. enable
2. show voice dsp
4. voice-card slot

39
DETAILED STEPS
Changing Codec Complexity on Cisco 880 Series, Cisco 2600 Series, Cisco 3600 Series, Cisco
3700 Series, and Cisco MC3810
To change codec complexity after the controller and voice ports have already been configured, use the
following commands:
Note Use the show voice dsp command to check the DSP voice channel activity. If any DSP voice channels
are in the busy state, the codec complexity cannot be changed. You must clear all calls before performing
the following task.
SUMMARY STEPS
1. enable
3. voice-port slot/port:ds0-group-number
4. shutdown
5. exit
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Checks the DSP voice channel activity. If any DSP voice
channels are in the busy state, codec complexity cannot be
changed. When all DSP channels are in the idle state, continue
to Step 2.
Example:
Step 4 voice-card slot
Example:
Enters voice card-configuration mode for the card or cards in
the slot specified. Range is 0 to 5.
Step 5 codec complexity {high | medium}
Example:
Router(config-voicecard)# codec complexity
high
Specifies codec complexity based on the codec standard being
used. This setting restricts the codecs available in dial peer
configuration. All voice cards in a router must use the same
codec complexity setting. Default is medium.
Note On the Cisco MC3810, this command is valid only
with one or more HCMs installed, and voice card 0
must be specified. If two HCMs are installed, this
command configures both HCMs at once.

40
6. controller {t1 | e1} slot/port
7. no ds0-group ds0-group-number
or
no pri-group timeslots timeslot-list
8. exit
9. voice-card slot
10. codec complexity {high | medium} [ecan-extended]
11. exit
12. Repeat Step 6, then continue with Step 13.
13. ds0-group ds0-group-number timeslots timeslot-list type {e&m-immediate | e&m-delay |
e&m-wink-start | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}
or
pri-group timeslots timeslot-list
14. exit
15. Repeat Step 3, then continue with Step 16.
16. no shutdown
17. end
DETAILED STEPS
Command or Action Purpose
Step 1 enable
Example:
Router> enable
Example:

41
Step 3 voice-port slot/port:ds0-group-number
Example:
Router(config)# voice-port 1/0:23
Enters voice-port configuration mode on the selected
slot, port, and DS0 group.
Note The syntax of this command is
platform-specific. For the syntax for your
platform, refer to the Cisco IOS Voice
CommandReference.
Note For the Cisco 880 series platforms, the
command syntax does not include a slot
number, only the port is identified.
If the primary voice interface is FXS and the
backup is BRI, then ports 0, 1, 2, and 3 are
analog voice ports, and ports 4 and 5 are
digital. If the primary voice interface is BRI,
then ports 1, 2, 3, and 4 are digital.
Step 4 shutdown
Example:
Router(config-voiceport)# shutdown
Shuts down all voice ports assigned to the T1 interface
on the voice card.
Step 5 exit
Example:
Router(config-voiceport)# exit
Exits voice-port configuration mode.
Step 6 controller {t1 | e1} slot/port
Example:
Router(config)# controller t1 1/0
Enters controller configuration mode on the T1
controller on the selected slot and port.
Step 7 no ds0-group ds0-group-number
or
no pri-group timeslots timeslot-list
Example:
Router(config-controller)# no ds0-group 1
Example:
Router(config-controller)# no pri-group timeslots
1,7,9
Removes the related DS0 groups.
or
Removes the related PRI group.
Step 8 exit
Example:
Router(config-controller) exit
Exits controller configuration mode and returns to
global configuration mode.
Example:
Enters voice-card configuration mode on the specified
slot.
slotSlot number of the voice card. Range is
0 to 6, depending on platform.

42
Step 10 codec complexity {high | medium} [ecan-extended]
Example:
Router(voice-card)# codec complexity high
ecan-extended
Changes codec complexity or changes the echo
canceller (EC) from the proprietary Cisco G.165 EC to
the G.168 extended EC.
highSupports up to six voice or fax calls per
DSP module (PVDM-12), using the codecs:
G.723, G.728, G.729, G.729 Annex B, GSMEFR,
GSMFR, fax relay, or any of the medium
complexity codecs.
mediumSupports up to 12 voice or fax calls per
DSP module (PVDM-12), using the codecs:
G.711, G.726, G.729 Annex A, G.729 Annex A
with Annex B, and fax relay. Default value.
ecan-extended(Optional) Selects the G.168
extended echo canceller. For more information,
see the How to Configure the Extended G.168
Echo Canceller section.
Specifying the codec complexity restricts the codecs
available in dial-peer configuration mode. All voice
cards in a gateway must use the same codec
complexity.
Step 11 exit
Example:
Router(voice-card) exit
Exits voice-card configuration mode and returns to
Step 12 Repeat Step 6, then continue with Step 13.
Step 13 ds0-group ds0-group-number timeslots timeslot-list
type {e&m-immediate | e&m-delay | e&m-wink-start |
fxs-ground-start | fxs-loop-start |
fxo-ground-start | fxo-loop-start}
or
pri-group timeslots timeslot-list
Example:
Router(config-controller)# ds0-group 0 timeslots
1-24 type e&m-wink-start
Example:
Router(config-controller)# pri-group timeslots
1,7,9
Defines the T1 or E1 channels for use by compressed
voice calls and the signaling method that the router
uses to connect to the PBX or CO.
Note If you are configuring PRI groups instead of
DS0 groups, omit this step and proceed to
Step 15.
or
Specifies an ISDN PRI on a channelized T1 or E1
controller.
Note When configuring PRI groups, you must also
configure the isdn switch-type command.
Also, only one PRI group can be configured on
a controller.
Step 14 exit
Example:
Router(config-controller)# exit
Exits controller configuration mode and completes the
process for adding back the PRI groups or DS0 groups.
Step 15 Repeat Step 3, then continue with Step 16.

43
Configuring the Flex Option on Codec Complexity for the Cisco 2600 XM, Cisco 2691, and Cisco 3700
Series
The IP Communications High-Density Digital Voice/Fax Network Module feature enables the flex
option for configuring codec complexity.
On the Cisco 2600 XM, Cisco 2691, Cisco 3700 series routers, codec complexity can be configured
using the flex option for configuring codec complexity. This option allows the DSP to process up to 16
channels. In addition to continuing support for configuring a fixed number of channels per DSP, the flex
option enables the DSP to handle a flexible number of channels. The total number of supported channels
varies from 6 to 16, depending on which codec is used for a call. Therefore, the channel density varies
from 6 per DSP (high-complexity codec) to 16 per DSP (g.711 codec).
Requirements
The following requirements apply to the IP Communications High-Density Digital Voice/Fax Network
Module feature.
When the IP Communications High-Density Digital Voice/Fax Network Module feature is used in a
Cisco CallManager network, the CCM 4.0(1) SR1 or CCM 3.3(4) release must be installed.
Software echo cancellation is the default configurationG.168-compliant echo cancellation is
enabled by default with a coverage of 64 milliseconds.
Only Packet Fax/Voice DSP modules (PVDM2s) are supported on the IP Communications
High-Density Digital Voice/Fax Network Module.
Only voice interface cards that start with VIC2 are supported in the IP Communications
High-Density Digital Voice/Fax Network Module feature except for VIC-1J1, VIC-2DID, and
VIC-4FXS/DID.
The direct inward dial (DID) feature in VIC-4FXS/DID is not supported.
The CAMA card (VIC-2CAMA) is not supported. Any port on the VIC2-2FXO and the VIC2-4FXO
can be software configured to support analog CAMA for dedicated E-911 services (North America
only).
Codec Combinations for DSP Sharing
When network modules or PVDM2s on the motherboard are configured for DSP sharing, the codec
complexity has to match. A local resource sharing or importing from a remote network module must
match its characteristics, that is, a high-complexity network module can only share from another
high-complexity network module, whereas a flex-complexity network module can share DSPs from both
high-complexity and flex-complexity network modules. Table 2 summarizes the codec combinations for
DSP-sharing.
Step 16 no shutdown
Example:
Router(config-controller)# no shutdown
Saves the controller configurations on the slot and port
specified.
Step 17 end
Example:
Router(config-controller)# end
Exits controller configuration mode and completes the
process for bringing the T1 controller back up.

44
Caution If you are configuring a Cisco 2600 XM router, you should not use the network-clock-participate
command for slot 1 of the router. This may cause a disruption in service to the router.
Using Flex Mode
In flex mode, you can connect (or configure in the case of DS0 groups and PRI groups) more voice
channels to the module than the DSPs can accommodate. This is referred to as oversubscription. If all
voice channels should go active simultaneously, the DSPs will be oversubscribed and calls that are
unable to allocate a DSP resource will fail to connect.
To enable the IP Communications Voice/Fax Network Module feature, perform this task to configure the
voice card for the flex option in codec complexity.
SUMMARY STEPS
1. enable
3. voice-card slot
4. codec complexity flex [reservation-fixed {high | medium}]
5. voice local-bypass
6. exit
DETAILED STEPS
Table 2 Codec Complexity Settings for DSP Resource Sharing Between Local and Remote
Sources
Local DSP Resource (Import)
Remote DSP Resource (Export)
High complexity Medium complexity Flexible complexity
High complexity Yes No No
Medium complexity Yes Yes No
Flexible complexity Yes No Yes
Step 1 enable
Example:
Router> enable
Example:

45
Configuring Codec Complexity on Cisco 7200 Series and Cisco 7500 Series Routers
On Cisco 7200 series and Cisco 7500 series routers, codec complexity is configured in the DSP
interface.
Example:
Enters voice-card configuration mode and specifies the slot
location.
For the slot argument, specify a value from 1 to 4,
depending on your router.
Step 4 codec complexity flex [reservation-fixed {high
| medium}]
Example:
Router(config-voicecard)# codec complexity flex
Specifies the flex option for codec complexity.
flexUp to 16 calls can be completed per DSP. The
number of supported calls varies from 6 to 16,
depending on the codec used for a call. In this mode,
reservation for analog VICs may be needed for certain
appplications such as CAMA E-911 calls because
oversubscription of DSPs is possible. If this is true,
then the reservation-fixed option may be enabled.
There is no reservation by default.
reservation-fixedAppears as an option only
when there is an analog VIC present. Ensures that
sufficient DSP resources are available to handle a
call. If you enter this keyword, then specify if the
complexity should be high or medium.
Note You cannot change codec complexity while
DS0 groups are defined. If they are already set
up, perform the steps in the Changing Codec
Complexity on Cisco 880 Series, Cisco 2600
Series, Cisco 3600 Series, Cisco 3700 Series,
and Cisco MC3810 section on page 39.
Step 5 voice local-bypass
Example:
Router(config-voicecard)# voice local-bypass
Configures local calls to bypass the DSP. This is the default.
Using this command enables intranetwork-module
hairpinning (no DSPs).
Note For POTS-to-POTS calls between two network
modules, hairpinning is not supported. If the
connection manager in Cisco IOS software does not
automatically handle this, it might be necessary to
disable local-bypass so that DSPs are used for these
calls.
Step 6 exit
Example:
Router(config-voicecard)# exit
Exits voice-card configuration mode and returns the router
to global configuration mode.

46
Note Use the show interfaces dspfarm command to check the DSP voice channel activity. If any DSP voice
channels are in the busy state, the codec complexity cannot be changed. You must clear all calls before
performing the following task.
SUMMARY STEPS
1. enable
3. dspint dspfarm slot/0
or
dspint dspfarm slot/port-adapter/port
4. codec {high | medium} [ecan-extended]
5. exit
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:
Step 3 dspint dspfarm slot/0
or
dspint dspfarm slot/port-adapter/port
Example:
Router(config)# dspint dspfarm 2/0
Enters DSP interface configuration mode for the
Cisco 7200 series.
or
Enters DSP interface configuration mode for the
Cisco 7500 series.

47
Cisco 7200 Series
On the Cisco 7200 series, the PA-MCX-2TE1 port adapter (PA) card can be used for making voice calls.
This PA does not have any DSPs but uses the DSP resources of the PA-VXC-2TE1+ card present in
another slot. If the PA-MCX card is used, codec complexity is configured for PA-VXC, while all other
echo cancellation configurations are done for PA-MCX.
The PA-MCX card borrows the DSP resources from the PA-VXC, PA-VXB, or PA-VXA card. If one of
the PA-VXC, PA-VXB, or PA-VXA cards has extended echo cancellation configured on the DSP
interface, extended echo cancellation is enabled for the PA-MCX card. It is recommended that you have
the same codec complexity and echo cancellation configuration on all the PA-VXC, PA-VXB, or
PA-VXA cards in the router.
Cisco AS5300
Codec support on the Cisco AS5300 is determined by the capability list on the voice feature card, which
defines the set of codecs that can be negotiated for a voice call. The capability list is created and
populated when VCWare is unbundled and DSPWare is added to VFC flash memory. The capability list
does not indicate codec preference; it simply reports the codecs that are available. The session
application decides which codec to use. Codec support is configured on dial peers rather than on voice
ports; refer to the Dial Peer Configuration on Voice Gateway Routers document.
Cisco AS5800
Codec support is selected on Cisco AS5800 access servers during dial peer configuration. Refer to the
Dial Peer Configuration on Voice Gateway Routers document.
Configuring Controller Settings for Digital T1/E1 Voice Ports
The controller configuration for digital T1/E1 voice ports must match the line characteristics of the
telephony network connection so that voice and signaling can be transferred between them and so that
logical voice ports, or DS0 groups, may be established.
Step 4 codec {high | medium} [ecan-extended]
Example:
Router(config-dspfarm)# codec medium ecan-extended
Sets the codec complexity.
The optional ecan-extended keyword selects
the G.168 extended echo canceller. This
keyword is supported only in Cisco IOS
Release 12.2(13)T. For more information, see
the How to Configure the Extended G.168
Echo Canceller section.
This command affects the choice of codecs
available when the codec command is used in
dial-peer configuration mode.
Step 5 exit
Example:
Router(config-dspfarm)# exit
Exits to global configuration mode.

48
Figure 1 shows how a ds0-group command gathers some of the DS0 time slots from a T1 line into a
group that becomes a single logical voice port that can later be addressed as a single entity in voice port
configurations. Other DS0 groups for voice can be created from the remaining time slots shown in the
figure, or the time slots can be used for data or serial pass-through.
Note All controller commands shown in Figure 1, other than ds0-group, apply to all time slots in the T1 line.
Figure 1 T1 Controller Configuration on Cisco 2600 or Cisco 3600 Series Routers
Voice port controller configuration includes setting the parameters described in the following sections:
Framing Formats on Digital T1/E1 Voice Ports, page 48
Clock Sources on Digital T1/E1 Voice Ports, page 49
Network Clock Timing, page 52
Line Coding on Digital T1/E1 Voice Ports, page 54
DS0 Groups on Digital T1/E1 Voice Ports, page 54
Another controller command that might be needed, cablelength, is discussed in the Cisco IOS Interface
and Hardware Component Command Reference.
Framing Formats on Digital T1/E1 Voice Ports
The framing format parameter describes the way that bits are robbed from specific frames to be used for
signaling purposes. The controller must be configured to use the same framing format as the line from
the PBX or CO that connects to the voice port you are configuring.
V V
Network module slot 1
VWIC slot 0
T1
3
7
7
6
0
controller t1 1/0
framing esf
clock source line
linecode b8zs
Configures T1
controller 1/0
Creates DS0 group, or
logical voice port, 1/0:1
by grouping 12
time slots together
ds0-group 1 timeslots 1-12 type e&m-wink-start

49
Digital T1 lines use SF or ESF framing formats. SF provides two-state, continuous supervision
signaling, in which bit values of 0 are used to represent on-hook and bit values of 1 are used to represent
off-hook. ESF robs four bits instead of two, yet has little impact on voice quality. ESF is required for
64-kbps operation on DS0 and is recommended for PRI configurations.
E1 lines can be configured for CRC4 or no cyclic redundancy check, with an optional argument for E1
lines in Australia.
Clock Sources on Digital T1/E1 Voice Ports
Digital T1/E1 interfaces use timers called clocks to ensure that voice packets are delivered and
assembled properly. All interfaces handling the same packets must be configured to use the same source
of timing so that packets are not lost or delivered late. The timing source that is configured can be
external (from the line) or internal to the routers digital interface.
If the timing source is internal, timing derives from the onboard phase-lock loop (PLL) chip in the digital
voice interface. If the timing source is line (external), then timing derives from the PBX or PSTN CO to
which the voice port is connected. It is generally preferable to derive timing from the PSTN becauseits
clocks are maintained at an extremely accurate level. This is the default setting for the clocks. When two
or more controllers are configured, one should be designated as the primary clock source; it will drive
the other controllers.
The line keyword specifies that the clock source is derived from the active line rather than from the
free-running internal clock. The following rules apply to clock sourcing on the controller ports:
When both ports are set to line clocking with no primary specification, port 0 is the default primary
clock source and port 1 is the default secondary clock source.
When both ports are set to line and one port is set as the primary clock source, the other port is by
default the backup or secondary source and is loop-timed.
If one port is set to clock source line or clock source line primary and the other is set to clock source
internal, the internal port recovers clock from the clock source line port if the clock source line port
is up. If it is down, then the internal port generates its own clock.
If both ports are set to clock source internal, there is only one clock source: internal.
This section describes the five basic timing scenarios that can occur when a digital voice port is
connected to a PBX or CO. In all the examples that follow, the PSTN (or CO) and the PBX are
interchangeable for purposes of providing or receiving clocking.
Single voice port providing clockingIn this scenario, the digital voice hardware is the clock source
for the connected device, as shown in Figure 2. The PLL generates the clock internally and drives
the clocking on the line. Generally, this method is useful only when connecting to a PBX, key
system, or channel bank. A Cisco VoIP gateway rarely provides clocking to the CO because CO
clocking is much more reliable. The following configuration sets up this clocking method for a
digital E1 voice port:
controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start

50
Figure 2 Single Voice Port Providing Clocking
Single voice port receiving internal clockingIn this scenario, the digital voice hardware receives
clocking from the connected device (CO telephony switch or PBX) (see Figure 3). The PLL
clocking is driven by the clock reference on the receive (Rx) side of the digital line connection.
Figure 3 Single E1 Port Receiving Clocking from the Line
The following configuration sets up this clocking method:
controller T1 1/0
framing esf
linecoding ami
clock source line
Dual voice ports receiving clocking from the LineIn this scenario, the digital voice port has two
reference clocks, one from the PBX and another from the CO, as shown in Figure 4.
Figure 4 Dual E1 Ports Receiving Clocking from the Line
Because the PLL can derive clocking from only one source, this case is more complex than the two
preceding examples. Before looking at the details, consider the following as they pertain to the
clocking method:
Looped-time clockingThe voice port takes the clock received on its Rx (receive) pair and
regenerates it on its Tx (transmit) pair. While the port receives clocking, the port is not driving
the PLL on the card but is spoofing (that is, fooling) the port so that the connected device has
a viable clock and does not see slips (that is, loss of data bits). PBXs are not designed to accept
slips on a T1 or E1 line, and such slips cause a PBX to drop the link into failure mode. While
in looped-time mode, the router often sees slips, but because these are controlled slips, they
usually do not force failures of the routers voice port.
SlipsThese messages indicate that the voice port is receiving clock information that is out of
phase (out of synchronization). Because the router has only a single PLL, it can experience
controlled slips while it receives clocking from two different time sources. The router can
usually handle controlled slips because its single-PLL architecture anticipates them.
E1 0
PBX
Clock
2
6
9
1
9
E1 0
Clock
2
6
9
2
0
PSTN
E1 0
E1 1
Looped time
Clock
Clock
2
6
9
2
1
PSTN
PBX

51
Note Physical layer issues, such as bad cabling or faulty clocking references, can cause slips.
Eliminate these slips by addressing the physical layer or clock reference problems.
In the dual voice ports receiving clocking from the line scenario, the PLL derives clocking from the
CO and puts the voice port connected to the PBX into looped-time mode. This is usually the best
method because the CO provides an excellent clock source (and the PLL usually requires that the
CO provide that source) and a PBX usually must receive clocking from the other voice port.
The following configuration sets up this clocking method (controller E1 1/0 is connected to the CO;
controller E1 1/1 is connected to the PBX:
controller E1 1/0
framing crc4
linecoding hdb3
clock source line primary
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source line
The clock source line primary command tells the router to use this voice port to drive the PLL. All
other voice ports configured as clock source line are then put into an implicit loop-timed mode. If
the primary voice port fails or goes down, the other voice port instead receives the clock that drives
the PLL. In this configuration, port 1/1 might see controlled slips, but these should not force it down.
This method prevents the PBX from seeing slips.
Note When two T1/E1 lines terminate on a two-port interface card, such as the VWIC-2MFT, and both
controllers are set for line clocking but the lines are not within clocking tolerance of one another,
one of the controllers is likely to experience slips. To prevent slips, ensure that the two T1 or E1
lines are within clocking tolerance of one another, even if the lines are from different providers.
Dual voice ports (one receives clocking and one provides clocking)In this scenario, the digital
voice hardware receives clocking for the PLL from E1 0 and uses this clock as a reference to clock
E1 1 (see Figure 5). If controller E1 0 fails, the PLL internally generates the clock reference to drive
E1 1.
Figure 5 Dual E1 PortsOne Receiving and One Providing Clocking
controller E1 1/0
framing crc4
linecoding hdb3
E1 0
E1 1
Clock
Clock
2
6
9
2
2
PSTN
PBX

52
clock source line
!
controller E1 1/1
framing crc4
linecoding hdb3
Dual voice ports (router provides both clocks)In this scenario, the router generates the clock for
the PLL and, therefore, for both voice ports (see Figure 6).
Figure 6 Dual E1 PortsBoth Clocks from the Router
controller E1 1/0
framing crc4
linecoding hdb3
!
controller E1 1/1
framing esf
linecoding b8zs
Network Clock Timing
Voice systems that pass digitized (pulse code modulation or PCM) speech have always relied on the
clocking signal being embedded in the received bit stream. This reliance allows connected devices to
recover the clock signal from the bit stream, and then use this recovered clock signal to ensure that data
on different channels keep the same timing relationship with other channels.
If a common clock source is not used between devices, the binary values in the bit streams may be
misinterpreted because the device samples the signal at the wrong moment. As an example, if the local
timing of a receiving device is using a slightly shorter time period than the timing of the sending device,
a string of eight continuous binary 1s may be interpreted as nine continuous 1s. If this data is then re-sent
to further downstream devices that used varying timing references, the error could be compounded. By
ensuring that each device in the network uses the same clocking signal, you can ensure the integrity of
the traffic.
If timing between devices is not maintained, a condition known as clock slip can occur. Clock slip is the
repetition or deletion of a block of bits in a synchronous bit stream due to a discrepancy in the read and
write rates at a buffer.
Slips are caused by the inability of an equipment buffer store (or other mechanisms) to accommodate
differences between the phases or frequencies of the incoming and outgoing signals in cases where the
timing of the outgoing signal is not derived from that of the incoming signal.
E1 1/0
E1 1/1
Clock
2
6
9
2
3
Clock
PSTN
PBX

53
A T1 or E1 interface sends traffic inside repeating bit patterns called frames. Each frame is a fixed
number of bits, allowing the device to see the start and end of a frame. The receiving device also knows
exactly when to expect the end of a frame simply by counting the appropriate number of bits that have
come in. Therefore, if the timing between the sending and receiving device is not the same, the receiving
device may sample the bit stream at the wrong moment, resulting in an incorrect value being returned.
Even though Cisco IOS software can be used to control the clocking on these platforms, the default
clocking mode is effectively free running, meaning that the received clock signal from an interface is not
connected to the backplane of the router and used for internal synchronization between the rest of the
router and its interfaces. The router will use its internal clock source to pass traffic across the backplane
and other interfaces.
For data applications, this clocking generally does not present a problem as a packet is buffered in
internal memory and is then copied to the transmit buffer of the destination interface. The reading and
writing of packets to memory effectively removes the need for any clock synchronization between ports.
Digital voice ports have a different issue. It would appear that unless otherwise configured, Cisco IOS
software uses the backplane (or internal) clocking to control the reading and writing of data to the DSPs.
If a PCM stream comes in on a digital voice port, it will be using the external clocking for the received
bit stream. However, this bit stream will not necessarily be using the same reference as the router
backplane, meaning the DSPs may misinterpret the data coming in from the controller.
This clocking mismatch is seen on the routers E1 or T1 controller as a clock slipthe router is using
its internal clock source to send the traffic out the interface but the traffic coming in to the interface is
using a completely different clock reference. Eventually, the difference in the timing relationship
between the transmit and receive signal becomes so great that the controller registers a slip in the
received frame.
To eliminate the problem, change the default clocking behavior through Cisco IOS configuration
commands. It is absolutely critical to set up the clocking commands properly.
Even though these commands are optional, we strongly recommend you enter them as part of your
configuration to ensure proper network clock synchronization:
network-clock-participate [slot slot-number | wic wic-slot | aim aim-slot-number]
network-clock-select priority {bri | t1 | e1} slot/port
The network-clock-participate command allows the router to use the clock from the line via the
specified slot/WIC/AIM and synchronize the onboard clock to the same reference.
If multiple VWICS are installed, the commands must be repeated for each installed card. The system
clocking can be confirmed using the show network clocks command.
Caution If you are configuring a Cisco 2600 XM voice gateway with an NM-HDV2 or NM-HD-2VE installed in
slot 1, do not use the network-clock-participate slot 1 command in the configuration. In this particular
hardware scenario, the network-clock-participate slot 1 command is not necessary.
If the network-clock-participate slot 1 command is configured, voice and data connectivity on
interfaces terminating on the NM-HDV2 or NM-HD-2VE network module may fail to operate properly.
Data connectivity to peer devices may not be possible, and even loopback plug tests to the serial interface
spawned via a channel group configured on the local T1/E1 controller will fail. Voice groups such as
CAS DS0 groups and ISDN PRI groups may fail to signal properly. The T1/E1 controller may
accumulate large amounts of timing slips and Path Code Violations (PCVs) and Line Code Violations
(LCVs).

54
Line Coding on Digital T1/E1 Voice Ports
Digital T1/E1 interfaces require that line encoding be configured to match that of the PBX or CO that is
being connected to the voice port. Line encoding defines the type of framing used on the line.
T1 line encoding methods include AMI and B8ZS. AMI is used on older T1 circuits and references signal
transitions with a binary 1, or mark. B8ZS, a more reliable method, is more popular and is
recommended for PRI configurations as well. B8ZS encodes a sequence of eight zeros in a unique binary
sequence to detect line-coding violations.
Supported E1 line encoding methods are AMI and HDB3, which is a form of zero-suppression line
coding.
DS0 Groups on Digital T1/E1 Voice Ports
For digital voice ports, a single command, ds0-group, performs the following functions:
Defines the T1/E1 channels for compressed voice calls.
Automatically creates a logical voice port.
The numbering for the logical voice port created as a result of this command is
controller:ds0-group-number, where controller is defined as the platform-specific address for a
particular controller. On a Cisco 3640 router, for example, ds0-group 1 timeslots 1-24 type
e&m-wink automatically creates the voice port 1/0:1 when issued in the configuration mode for
controller 1/0. On a Cisco MC3810 universal concentrator, when you are in the configuration mode
for controller 0, the ds0-group 1 timeslots 1-24 type e&m-wink command creates logical voice
port 0:1.
To map individual DS0s, define additional DS0 groups under the T1/E1 controller, specifying
different time slots. Defining additional DS0 groups also creates individual DS0 voice ports.
Defines the emulated analog signaling method that the router uses to connect to the PBX or PSTN.
Most digital T1/E1 connections used for switch-to-switch (or switch-to-router) trunks are E&M
connections, but FXS and FXO connections are also supported. These are normally used to provide
emulated-OPX (Off-Premises eXtension) from a PBX to remote stations. FXO ports connect to FXS
ports. The FXO or FXS connection between the router and switch (CO or PBX) must use matching
signaling, or calls cannot connect properly. Either ground-start or loop-start signaling is appropriate
for these connections. Ground-start provides better disconnect supervision to detect when a remote
user has hung up the telephone, but ground-start is not available on all PBXs.
Digital ground start differs from digital E&M because the A and B bits do not track each other as
they do in digital E&M signaling (that is, A is not necessarily equal to B). When the CO delivers a
call, it seizes a channel (goes off-hook) by setting the A bit to 0. The CO equipment also simulates
ringing by toggling the B bit. The terminating equipment goes off-hook when it is ready to answer
the call. Digits are usually not delivered for incoming calls.
E&M connections can use one of three different signaling types to acknowledge on-hook and
off-hook states: wink start, immediate-start, and delay-start. E&M wink start is usually preferred,
but not all COs and PBXs can handle wink-start signaling. The E&M connection between the router
and switch (CO or PBX) must match the CO or PBX E&M signaling type, or calls cannot be
connected properly.
E&M signaling is normally used for trunks. It is normally the only way that a CO switch can provide
two-way dialing with DID. In all the E&M protocols, off-hook is indicated by A=B=1 and on-hook
is indicated by A=B=0 (robbed-bit signaling). If dial pulse dialing is used, the A and B bits are
pulsed to indicate the addressing digits. The are several further important subclasses of E&M
robbed-bit signaling:

55
E&M wink-startFeature Group B
In the original wink start handshaking protocol, the terminating side responds to an off-hook
from the originating side with a short wink (transition from on-hook to off-hook and back
again). This wink tells the originating side that the terminating side is ready to receive
addressing digits. After receiving addressing digits, the terminating side then goes off-hook for
the duration of the call. The originating endpoint maintains off-hook for the duration of the call.
E&M wink-startFeature Group D
In Feature Group D wink-start with wink acknowledge handshaking protocol, the terminating
side responds to an off-hook from the originating side with a short wink (transition from
on-hook to off-hook and back again) just as in the original wink-start. This wink tells the
originating side that the terminating side is ready to receive addressing digits. After receiving
addressing digits, the terminating side provides another wink (called an acknowledgment wink)
that tells the originating side that the terminating side has received the dialed digits. The
terminating side then goes off-hook to indicate connection. This last indication can be due to
the ultimate called endpoints having answered. The originating endpoint maintains an off-hook
condition for the duration of the call.
E&M immediate-start
In the immediate-start protocol, the originating side does not wait for a wink before sending
addressing information. After receiving addressing digits, the terminating side then goes
off-hook for the duration of the call. The originating endpoint maintains off-hook for the
duration of the call.
Note Feature Group D is supported on Cisco AS5300 platforms, and on Cisco 2600, Cisco 3600, and
Cisco 7200 series with digital T1 packet voice trunk network modules. Feature Group D is not supported
on E1 or analog voice ports.
To configure controller settings for digital T1/E1 voice ports, use the following commands:
SUMMARY STEPS
1. enable
3. card type {t1 | e1} slot
or
card type {t1 | e1} slot/port
or
controller {t1 | e1} number
or
controller {t1 | e1} shelf/slot/port
5. framing {sf | esf}
or
framing {crc4 | no-crc4} [australia]
6. clock source {line [primary | secondary] | internal}
7. linecode {ami | b8zs}
or
linecode {ami | hdb3}

56
8. ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd |
e&m-immediate-start | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start |
fxs-ground-start | fxs-loop-start}
9. no shutdown
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Step 3 card type {t1 | e1} slot
Example:
Router(config)# card type t1 0
Defines the card as T1 or E1 and identifies the
location.
or
controller {t1 | e1} number
or
controller {t1 | e1} shelf/slot/port
Example:
or
Example:
Router(config)# controller t1 1
or
Example:
Router(config)# controller t1 1/0/0
Enters controller configuration mode and specifies
either T1 or E1 for the line.
For the Cisco 2600, Cisco 3600 series,
Cisco MC3810, and Cisco 7200 series,
identifies the slot and port.
For the Cisco AS5300, identifies the port
number.
For the Cisco AS5800 and Cisco 7500 series,
identifies the shelf, slot, and port number.

57
Configuring Basic Voice Port Parameters for Digital T1/E1 Voice Ports
For FXO and FXS connections the default voice-port parameter values are often adequate. However, for
E&M connections, it is important to match the characteristics of your PBX, so voice port parameters may
need to be reconfigured from their defaults.
Step 5 framing {sf | esf}
or
framing {crc4 | no-crc4} [australia]
Example:
Router(config-controller)# framing esf
or
Example:
Router(config-controller)# framing crc4
Selects frame type for T1 or E1 line.
For T1, the frame type can be sf or esf. Default
for T1 is sf.
For E!, the frame type can be crc4 or no crc4
or australia. Default for E1 is crc4.
Step 6 clock source {line [primary | secondary] | internal}
Example:
Router(config-controller)# clock source line primary
Configures the clock source.
Default is line.
For more information about clock sources, see
the Clock Sources on Digital T1/E1 Voice
Ports section on page 49.
Step 7 linecode {ami | b8zs}
or
linecode {ami | hdb3}
Example:
Router(config-controller)# linecode b8zs
or
Example:
Router(config-controller)# linecode hdb3
Specifies the line encoding to use for T1 or E1
line.
For T1, the line encoding can be ami or b8zs.
Default for T1 is ami.
For E1, the line encoding can be ami or hdb3.
Default for E1 is hdb3.
Step 8 ds0-group ds0-group-number timeslots timeslot-list
type {e&m-delay-dial | e&m-fgd | e&m-immediate-start
|e&m-wink-start | ext-sig | fgd-eana |
fxo-ground-start | fxo-loop-start | fxs-ground-start |
fxs-loop-start}
Example:
Router(config-controller)# ds0-group 30 timeslots 0
type e&m-immediate-start
Defines the T1 channels for use by compressed
voice calls and the signaling method that the
router uses to connect to the PBX or CO.
Note This step shows the basic syntax and
signaling types available with the
ds0-group command. For the complete
syntax, refer to the Cisco IOS Voice
Command Reference.
Step 9 no shutdown
Example:
Router(config-controller)# no shutdown
Activates the controller.
Command Purpose

58
Each voice port that you address in digital voice port configuration is one of the logical voice ports that
you created with the ds0-group command.
Companding (from compression and expansion), used in Step 6 of the following table, is the part of the
PCM process in which analog signal values are logically rounded to discrete scale-step values on a
nonlinear scale. The decimal step number is then coded in its binary equivalent prior to transmission.
The process is reversed at the receiving terminal using the same nonlinear scale.
To configure basic parameters for digital T1/E1 voice ports, use the following commands:
SUMMARY STEPS
1. enable
3. voice-port port
or
voice-port slot/port:ds0-group-number
or
voice-port slot/port-adapter:ds0-group-number
or
voice-port slot/port-adapter/slot:ds0-group-number
or
voice-port controller:{ds0-group-number | D}
or
voice-port slot/controller:{ds0-group-number | D}
or
voice-port shelf/slot/port:ds0-group-number
4. type {1 | 2 | 3 | 5}
5. cptone locale
6. compand-type {u-law | a-law}
7. ring frequency {25 | 50}
8. ring number number
9. ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07
| pattern08 | pattern09 | pattern10 | pattern11 | pattern12] [define pulse interval]}
10. description string
11. no shutdown

59
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Step 3 voice-port port
or
voice-port slot/port:ds0-group-number
or
voice-port slot/port-adapter:ds0-group-number
or
voice-port slot/port-adapter/slot:ds0-group-number
or
voice-port controller:{ds0-group-number | D}
or
voice-port slot/controller:{ds0-group-number | D}
or
voice-port shelf/slot/port:ds0-group-number
Example:
Router(config)# voice-port 1:0
or
Example:
or
Example:
Router(config)# voice-port 1/1/1:1
or
Example:
Router(config)# voice-port 1:1
or
Example:
Router(config)# voice-port 1/0 D
or
Example:
Router(config)# voice-port 1/2/0:1
Enters voice-port configuration mode and
identifies the port to be configured.
For the Cisco 880 series, specify the port
number.
For the Cisco 2600, Cisco 3600, and
Cisco 3700 series, specify the slot, port, and
DS0 group number.
For the Cisco 7200 series, specify the slot,
port adapter,and DS0 group number.
For the Cisco 7500 series, specify the slot,
port adapter, slot, and DS0 group number.
For the Cisco AS5300, specify the controller
and DS0 group number or the keyword D.
For the Cisco AS5350, Cisco AS5400, and
Cisco AS5850 universal gateways, specify the
slot, controller, and DS0 group number or the
keyword D.
For the Cisco AS5800, specify the shelf, slot,
port, and DS0 group number.

60
Step 4 type {1 | 2 | 3 | 5}
Example:
(E&M only) Specifies the type of E&M interface
to which this voice port is connected. See Table 3
in the Vo ice Port Configuration Overview chapter
for an explanation of E&M types.
Default is 1.
Step 5 cptone locale
Example:
Router(config-voiceport)# cptone us
Selects a two-letter locale keyword for the voice
call progress tones and other locale-specific
parameters to be used on this voice port. Voice call
progress tones include dial tone, busy tone, and
ringback tone, which vary with geographical
region.
Other parameters include ring cadence and
compand type. Cisco routers comply with the
ISO3166 locale name standards; to see valid
choices, enter a question mark (?) following
the cptone command.
Default is us.
Step 6 compand-type {u-law | a-law}
Example:
Router(config-voiceport)# compand-type u-law
(Cisco 2600 and Cisco 3600 series routers.)
Specifies the companding standard used. This
command is used in cases when the DSP is not
used, such as local cross-connects, and overwrites
the compand-type value set by the cptone
command.
The default for E1 is a-law.
The default for T1 is u-law.
Note If you have a Cisco 3660 router, the
compand-type a-law command must be
configured on the analog ports only. The
Cisco 2660, 3620, and 3640 routers do not
require the compand-type a-law
command configured. However, if you
request a list of commands, the
compand-type a-law command will
display.
Step 7 ring frequency {25 | 50}
Example:
(FXS only) Selects the ring frequency, in hertz,
used on the FXS interface. This number must
match the connected telephony equipment, and
can be country-dependent. If the ring frequency is
not set properly, the attached telephony device
may not ring or it may buzz.
Default is 25.
Step 8 ring number number
Example:
Router(config-voiceport)# ring number 1
(FXO only) Specifies the maximum number of
rings to be detected before an incoming call is
answered by the router.
Default is 1.
Command Purpose

61
coincidental.
Step 9 ring cadence {[pattern01 | pattern02 | pattern03 |
pattern12] [define pulse interval]}
Example:
Router(config-voiceport)# ring cadence pattern01
define 12 15
(FXS only) Specifies an existing pattern for ring,
or defines a new one. Each pattern specifies a
ring-pulse time and a ring-interval time. The
keywords and arguments are as follows:
pattern01 through pattern12Specifies
preset ring cadence patterns. Enter ring
cadence ? to see ring pattern explanations.
define pulse intervalSpecifies a
user-defined pattern as follows:
pulse is a number (1 or 2 digits from 1 to
50) specifying ring pulse (on) time in
hundreds of milliseconds.
interval is a number (1 or 2 digits from 1
to 50) specifying ring interval (off) time
in hundreds of milliseconds.
The default is the pattern specified by the
configured cptone locale command.
Step 10 description string
Example:
Router(config-voiceport)# description 1
Attaches a text string to the configuration that
describes the connection for this voice port. This
description appears in various displays and is
useful for tracking the purpose or use of the voice
port. The string argument is a character string
from 1 to 255 characters in length.
The default is that no description is attached
to the configuration.
Step 11 no shutdown
Example:
Router(config-voiceport)# no shutdown
Activates the voice port.
Command Purpose

62

The default parameter values for voice ports are usually sufficient for most networks. Depending on the
specifics of your particular network, however, you may need to adjust certain parameters that are
configured on voice ports. Collectively, these commands are referred to as voice port tuning commands.
Information About Fine-Tuning Analog and Digital Voice Ports
This section provides basic information about some the features that can be fine-tuned to improve the
performance of your voice network:
Channel Bank Support for T1/E1 Voice PortsProvides support for the time-division multiplexing
(TDM) cross-connect functionality between analog voice ports and digital DS0s on the same
NM-HD-2VE using channel associated signaling (CAS).
Auto Cut-ThroughAllows you to connect to PBXs that do not provide an M-lead response.
Modification of Bit Patterns for Digital Voice PortsEnables commands for digital voice ports to
modify sent or received bit patterns. Different versions of E&M use different ABCD signaling bits
to represent idle and seize.
ANI for Outbound CallingAllows the automatic number identification (ANI) to be sent for
outgoing calls on the Cisco AS5300 (if T1 CAS is configured with the Feature Group-D
(FGD)Exchange Access North American (FGD-EANA) signaling).
Disconnect SupervisionConfigures the router to recognize the type of signaling in use by the PBX
or PSTN switch connected to the voice port. These methods include the following:
Battery reversal disconnect
Battery denial disconnect
Supervisory tone disconnect (STD)

How to Configure Fine-Tuning Features for Analog and Digital Voice Ports
64
FXO Supervisory Disconnect TonesPrevents an analog FXO port from remaining in an off-hook
state after an incoming call is ended. FXO supervisory disconnect tone enables interoperability with
PSTN and PBX systems whether or not they transmit supervisory tones.
Timeouts ParametersModifies values for timeouts. For example, you can adjust the wait time for
the caller input of the initial digit and the subsequent digit of the dialed string. If the wait time
expires before the destination is identified, a tone sounds and the call ends.
Timing ParametersChanges a wide range of timing values. For example, you can specify the
minimum delay time, in milliseconds, from outgoing seizure to outdial address.
DTMF TimerModifies the value for the DTMF interdigit timer.
Comfort Noise and Music Threshold for VADSpecifies the minimal decibel level of music played
when calls are put on hold and creates subtle background noise to fill silent gaps during calls when
VAD is enabled on voice dial peers. If comfort noise is not generated, the resulting silence can fool
the caller into thinking the call is disconnected instead of being merely idle.
How to Configure Fine-Tuning Features for Analog and Digital
Voice Ports
To configure the voice port tuning features for analog and digital voice ports, complete these tasks:
Configuring Channel Bank Support for T1/E1 Voice Ports, page 64
Configuring Auto Cut-Through, page 67
Modifying Bit Patterns for Digital Voice Ports, page 68
Configuring ANI for Outbound Calling, page 70
Configuring Disconnect Supervision, page 72
Configuring FXO Supervisory Disconnect Tones, page 74
Configuring Timeouts Parameters, page 77
Changing Timing Parameters, page 79
Configuring the DTMF Timer, page 82
Configuring Comfort Noise and Music Threshold for VAD, page 83
Cisco IOS command help to determine the syntax choices that are available. Full descriptions of the
commands in this section can be found in the Cisco IOS Voice Command Reference.
Configuring Channel Bank Support for T1/E1 Voice Ports
The channel bank feature provides support for the time-division multiplexing (TDM) cross-connect
functionality between analog voice ports and digital DS0s on the same NM-HD-2VE using channel
associated signaling (CAS).
To establish a channel bank connection between an analog voice port and a T1 DS0, configure the
connect (voice-port) command in global configuration mode. To verify the channel bank connection, use
the show connection all command.

65
Restrictions for Channel Bank Support
The configuration for cross-connect must be on the same network module.
A maximum of four Foreign Exchange Service (FXS) or Foreign Exchange Office (FXO) ports can
be cross-connected to a T1 interface.
A BRI-to-PRI cross-connect cannot be configured.
Analog-to-BRI/PRI cross-connect cannot be configured; the only connection for analog is
analog-to-T1/E1 CAS (ds0-group).
The local-bypass command has no effect when cross-connect is configured. It is applicable only to
calls that are hairpinned via POTS-to-POTS dial peers.
The DS0 group must contain only one time slot. The signaling type of the DS0 group must match
that of the analog voice port.
If the channel bank feature is used for the T1 controller, the rest of the unused DS0 group cannot be
used for fractional PRI signaling.
SUMMARY STEPS
1. enable
4. ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd |
e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start |
fxs-loop-start}
5. exit
6. voice-port slot/port
7. operation {2-wire | 4-wire}
8. type {1 | 2 | 3 | 5}
9. signal {loop-start | ground-start}
or
signal {wink-start | immediate | delay-dial}
10. exit
11. connect connection-name voice-port voice-port-number {t1 | e1} controller-number
ds0-group-number
12. exit

66
DETAILED STEPS

Step 1 enable
Example:
Router> enable
Example:
Example:
Enters controller configuration mode and identifies the
controller type (T1 or E1) and a slot and port for
configuration commands that specifically apply to the T1 or
E1 interface.
Valid values for the slot and port arguments are 0 and 1.
Step 4 ds0-group ds0-group-number timeslots
timeslot-list type {e&m-delay-dial | e&m-fgd |
e&m-immediate-start | e&m-wink-start |
fxs-ground-start | fxs-loop-start |
fxo-ground-start | fxo-loop-start}
Example:
Router(config-controller)# ds0-group 1
timeslots 1 type e&m-wink-start
Defines the T1 or E1 channels for use by compressed voice
calls and the signaling method the router uses to connect to
the PBX or central office (CO).
The ds0-group command automatically creates a
logical voice port.
ds0-group-numberValue from 0 to 23 that identifies
the DS0 group.
timeslot-listSingle number, numbers separated by
commas, or a pair of numbers separated by a hyphen to
indicate a range of time slots. For T1, allowable values
are 1 to 24; for E1, allowable values are 1 to 31.
The signaling method selection for type depends on the
connection that you are making:
Ear and Mouth (E&M) connects PBX trunk lines (tie
lines) and telephone equipment. The wink and delay
settings both specify confirming signals between the
sending and receiving ends, or the immediate setting
stipulates no special off-hook/on-hook signal.
FXO connects a CO to a standard PBX interface where
permitted by local regulations.
FXS connects basic telephone equipment and PBXs.
Step 5 exit
Example:
Router(config-controller)# exit
Exits controller configuration mode and returns to global
configuration mode.
Example:
Enters voice-port configuration mode and identifies a slot
and port for configuration parameters.

67
Configuring Auto Cut-Through
The auto-cut-through command allows you to connect to PBXs that do not provide an M-lead response.
To configure auto-cut-through, complete the following task:
SUMMARY STEPS
1. enable
Step 7 operation {2-wire | 4-wire}
Example:
Router(config-voiceport)# operation 4-wire
Selects a specific cabling scheme for E&M ports:
This command is not applicable to FXS or FXO
interfaces because they are, by definition, 2-wire
interfaces.
Using this command on a voice port changes the
operation of both voice ports on a VPM card. The voice
port must be shut down and then opened again for the
new value to take effect.
Step 8 type {1 | 2 | 3 | 5}
Example:
Specifies the E&M interface type.
Step 9 signal {loop-start | ground-start}
or
signal {wink-start | immediate | delay-dial}
Example:
Router(config-voiceport)# signal loop-start
or
Router(config-voiceport)# signal wink-start
Defines the signal type to be used.
Step 10 exit
Example:
Exits voice-port configuration mode and returns to global
configuration mode.
Step 11 connect connection-name voice-port
voice-port-number {t1 | e1} controller-number
ds0-group-number
Example:
Router(config)# connect connect1 voice-port
1/1/0 t1 1/0 0
Creates a named connection between two voice ports
associated with T1 or E1 interfaces where you have already
defined the groups by using the ds0-group command.
Step 12 exit
Example:
Router(config)# exit
Exits the current configuration session and returns to
privileged EXEC mode.

68
4. auto-cut-through
5. exit
DETAILED STEPS
Modifying Bit Patterns for Digital Voice Ports
The bit modification commands for digital voice ports modify sent or received bit patterns. Different
versions of E&M use different ABCD signaling bits to represent idle and seize. For example, North
American CAS E&M represents idle as 0XXX and seize as 1XXX, where X indicates that the state of
the BCD bits is ignored. In MELCAS E&M, idle is 1101 and seize is 0101.
To manipulate bit patterns to match particular E&M schemes, use the following commands:
SUMMARY STEPS
1. enable
4. condition {tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit} {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit} {on |
off | invert}
Step 1 enable
Example:
Router> enable
Example:
Example:
Note The syntax of this command is platform-specific.
For the syntax for your platform, refer to the
Cisco IOS Voice Command Reference.
Step 4 auto-cut-through
Example:
Router(config-voiceport)# auto-cut-through
(E&M only) Enables call completion on a router if a PBX
does not provide an M-lead response.
Step 5 exit
Example:
Exits voice-port configuration mode and completes the
configuration.

69
5. define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000
| 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
6. ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
7. exit
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Example:
Command Reference.
Step 4 condition {tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit}
{rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit} {on | off
| invert}
Example:
Router(config-voiceport)# condition tx-a-bit on
Manipulates sent or received bit patterns to match
expected patterns on a connected device. Repeat the
command for each transmit or receive bit to be
modified, but be careful not to destroy the
information content of the bit pattern.
The default is that the signaling format is not
manipulated (for all transmit or receive A, B, C,
and D bits).
Note The show voice port command reports at the
protocol level, and the show controller
command reports at the driver level. The
driver is not notified of any bit manipulation
using the condition command. As a result,
the show controller command output does
not account for the bit conditioning.

70
Configuring ANI for Outbound Calling
On the Cisco AS5300 platform, if T1 CAS is configured with the Feature Group-D (FGD)Exchange
Access North American (FGD-EANA) signaling, the automatic number identification (ANI) can be sent
for outgoing calls by using the calling-number outbound command.
FGD-EANA is a FGD signaling protocol of type EANA, which provides certain call services, such as
emergency (USA 911) calls. ANI is a Signaling System 7 (SS7) feature in which a series of digits, analog
or digital, are included in the call to identify the telephone number of the calling device. In other words,
ANI identifies the number of the calling party. ANI digits are used for billing purposes by Internet
service providers (ISPs), among other things. The commands in this section can be issued in voice-port
or dial-peer configuration mode, because the syntax is the same.
To configure your digital T1/E1 packet voice trunk network module to generate outbound ANI digits on
a Cisco AS5300, use the following commands:
SUMMARY STEPS
1. enable
Step 5 define {tx-bits | rx-bits} {seize | idle} {0000 |
0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000
| 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
Example:
Router(config-voiceport)# define tx-bits seize 0000
(Digital E1 E&M voice ports on Cisco 2600 and
Cisco 3600 series routers only) Defines specific
transmit or receive signaling bits to match the bit
patterns required by a connected device for North
American E&M and E&M MELCAS voice
signaling, if patterns different from the preset
defaults are required.
Also specifies which bits a voice port monitors
and which bits it ignores, if patterns that are
different from the defaults are required.
See the define command for the default
signaling patterns as defined in American
National Standards Institute (ANSI) and
European Conference of Posts and
Telecommunication Administration (CEPT)
standards.
Step 6 ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
Example:
Router(config-voiceport)# ignore rx-a-bit
(Digital E1 E&M voice ports on Cisco 2600 and
Cisco 3600 series routers only) Configures the voice
port to ignore the specified receive bit for North
American E&M or E&M MELCAS, if patterns
different from the defaults are required. See the
Cisco IOS Voice Command Reference for the default
signaling patterns as defined in ANSI and CEPT
standards.
Step 7 exit
Example:
Exits voice-port configuration mode and completes
the configuration.
Command Purpose

71
4. calling-number outbound range string1 string2
5. calling-number outbound sequence [string1] [string2] [string3] [string4] [string5]
6. calling-number outbound null
7. exit
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Example:
Command Reference.
Step 4 calling-number outbound range string1 string2
Example:
Router(config-voiceport)# calling-number outbound
range 3000 4000
(Cisco AS5300 only) Specifies ANI to be sent out
when the T1-CAS fgd-eana command is configured
as signaling type. The string1 and string2 arguments
are valid E.164 telephone number strings. Both
strings must be of the same length and cannot be
more than 32 digits long.
Only the last four digits are used for specifying
the range (string1 to string2) and for generating
the sequence of ANI by rotating through the
range until string2 is reached and then starting
from string1 again. If strings are fewer than four
digits in length, then entire strings are used.
Step 5 calling-number outbound sequence [string1] [string2]
[string3] [string4] [string5]
Example:
sequence 2000 3000 4000
(Cisco AS5300 only) Specifies ANI to be sent out
when the T1-CAS fgd-eana command is configured
as signaling type. This option configures a sequence
of discrete strings (string1...string5) to be passed out
as ANI for successive calls using the dial peer or
voice port. Limit is five strings. All strings must be
valid E.164 numbers, up to 32 digits in length.

72
Configuring Disconnect Supervision
PBX and PSTN switches use several different methods to indicate that a call should be disconnected
because one or both parties have hung up. The commands in this section are used to configure the router
to recognize the type of signaling in use by the PBX or PSTN switch connected to the voice port. These
methods include the following:
Battery reversal disconnect
Battery denial disconnect
Supervisory tone disconnect (STD)
Battery reversal occurs when the connected switch changes the polarity of the line in order to indicate
changes in call state (such as off-hook or, in this case, call disconnect). This is the signaling looked for
when the battery reversal command is enabled on the voice port, which is the default configuration.
Battery denial (sometimes called power denial) occurs when the connected switch provides a short
(approximately 600 milliseconds) interruption of line power to indicate a change in call state. This is the
signaling looked for when the supervisory disconnect command is enabled on the voice port, which is
the default configuration.
Supervisory tone disconnect occurs when the connected switch provides a special tone to indicate a
change in call state. Some PBXs and PSTN CO switches provide a 600-millisecond interruption of line
power as a supervisory disconnect, and others provide STD. This is the signal that the router is looking
for when the no supervisory disconnect command is configured on the voice port.
Note In some circumstances, you can use the FXO Disconnect Supervision feature to enable analog FXO ports
to monitor call progress tones for disconnect supervision that are returned from a PBX or from the PSTN.
For more information, see the Configuring FXO Supervisory Disconnect Tones section on page 74.
To change parameters related to disconnect supervision, use the following commands:
SUMMARY STEPS
1. enable
4. no battery-reversal
5. no supervisory disconnect
Step 6 calling-number outbound null
Example:
null
(Cisco AS5300 only) Suppresses ANI. No ANI is
passed when this voice port is selected.
Step 7 exit
Example:
the configuration.
Command Purpose

73
6. disconnect-ack
7. exit
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Example:
Command Reference.
Step 4 no battery-reversal
Example:
Router(config-voiceport)# no battery-reversal
(Analog only) Enables battery reversal. The default is
that battery reversal is enabled.
For FXO portsUse the no battery-reversal
command to configure a loop-start voice port not
to disconnect when it detects a second battery
reversal. The default is to disconnect when a
second battery reversal is detected.
Note This functionality is supported on Cisco 1750,
Cisco 2600 series, and Cisco 3600 series
routers; only analog voice ports on VIC-2FXO
cards are able to detect battery reversal.
Also use the no battery-reversal command when
a connected FXO port does not support battery
reversal detection.
For FXS portsUse the no battery-reversal
command to configure the voice port not to reverse
battery when it connects calls. The default is to
reverse battery when a call is connected, then
return to normal when the call is over, providing
positive disconnect.
See also the disconnect-ack command (Step 6).
Step 5 no supervisory disconnect
Example:
Router(config-voiceport)# no supervisory disconnect
(FXO only) Enables the PBX or PSTN switch to
provide STD. The supervisory disconnect command
is enabled by default.

74
Configuring FXO Supervisory Disconnect Tones
If the FXO supervisory disconnect tone is configured and a detectable tone from the PSTN or PBX is
detected by the digital signal processor (DSP), the analog FXO port goes on-hook. This feature prevents
an analog FXO port from remaining in an off-hook state after an incoming call is ended. FXO
supervisory disconnect tone enables interoperability with PSTN and PBX systems whether or not they
transmit supervisory tones.
To configure a voice port to detect incoming tones, you need to know the parameters of the tones
expected from the PBX or PSTN. Then create a voice class that defines the tone detection parameters,
and, finally, apply the voice class to the applicable analog FXO voice ports. This procedure configures
the voice port to go on-hook when it detects the specified tones. The parameters of the tones need to be
precisely specified to prevent unwanted disconnects because of nonsupervisory tones or noise detection.
A supervisory disconnect tone is normally a dual tone consisting of two frequencies; however, tones of
only one frequency can also be detected. Use caution if you configure voice ports to detect nondual
tones, because unwanted disconnects can result from detection of random tone frequencies. You can
configure a voice port to detect a tone with one on/off time cycle, or you can configure it to detect tones
in a cadence pattern with up to four on/off time cycles.
Note In the following procedure, the following commands were not supported until Cisco IOS
Release 12.2(2)T: freq-max-deviation, freq-max-power, freq-min-power, freq-power-twist, and
freq-max-delay.
To create a voice class that defines the specific tone or tones to be detected and then apply the voice class
to the voice port, use the following commands:
SUMMARY STEPS
1. enable
3. voice class dualtone tag
4. freq-pair tone-id frequency-1 frequency-2
5. freq-max-deviation hertz
Step 6 disconnect-ack
Example:
Router(config-voiceport)# disconnect-ack
(FXS only) Configures the voice port to return an
acknowledgment upon receipt of a disconnect signal.
The FXS port removes line power if the equipment on
the FXS loop-start trunk disconnects first. This is the
default.
The no disconnect-ack command prevents the FXS
port from responding to the on-hook disconnect with a
removal of line power.
Step 7 exit
Example:
configuration.
Command Purpose

75
6. freq-max-power dBmO
7. freq-min-power dBmO
8. freq-power-twist dBmO
9. freq-max-delay milliseconds
10. cadence-min-on-time milliseconds
11. cadence-max-off-time milliseconds
12. cadence-list cadence-id cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time]
[cycle-3-on-time cycle-3-off-time] [cycle-4-on-time cycle-4-off-time]
13. cadence-variation milliseconds
14. exit
15. voice-port slot/subunit/port
16. supervisory disconnect dualtone {mid-call | pre-connect} voice-class tag
17. supervisory disconnect anytone
18. exit
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Step 3 voice class dualtone tag
Example:
Router(config)# voice class dualtone 1
Enters voice-class configuration mode and creates
a voice class for defining one tone detection
pattern. Range is 1 to 10000. The tag number must
be unique on the router.
For more information about configuring voice
classes, refer to the Dial Peer Configuration
onVoice Gateway Routers.
Step 4 freq-pair tone-id frequency-1 frequency-2
Example:
Router(config-voice-class)# freq-pair 16 300 0
Specifies the two frequencies, in Hz, for a tone to
be detected (or one frequency if a nondual tone is
to be detected). If the tone to be detected contains
only one frequency, enter 0 for frequency-2.
Note Repeat this command for each additional
tone to be specified.
Step 5 freq-max-deviation hertz
Example:
Router(config-voice-class)# freq-max-deviation 10
Specifies the maximum frequency deviation that
will be detected, in Hz. Range is 10 to 125. Default
is 10.

76
Step 6 freq-max-power dBmO
Example:
Router(config-voice-class)# freq-max-power 20
Specifies the maximum tone power that will be
detected, in dBmO. Range is 0 to 20. Default is 10.
Step 7 freq-min-power dBmO
Example:
Router(config-voice-class)# freq-min-power 35
Specifies the minimum tone power that will be
detected, in dBmO. Range is 10 to 35. Default is
30.
Step 8 freq-power-twist dBmO
Example:
Router(config-voice-class)# freq-power-twist 15
Specifies the power difference allowed between
the two frequencies, in dBmO. Range is 0 to 15.
Default is 6.
Step 9 freq-max-delay time
Example:
Router(config-voice-class)# freq-max-delay 10
Specifies the timing difference allowed between
the two frequencies, in 10-millisecond increments.
Range is 10 to 100 (100 ms to 1 second). Default
is 20 (200 ms).
Step 10 cadence-min-on-time time
Example:
Router(config-voice-class)# cadence-min-on-time 10
Specifies the minimum tone on time that will be
detected, in 10-millisecond increments. Range is
0 to 100 (0 ms to 1 second).
Step 11 cadence-max-off-time time
Example:
Router(config-voice-class)# cadence-max-off-time 2000
Specifies the maximum tone off time that will be
detected, in 10-millisecond increments. Range is
0 to 5000 (0 ms to 50 seconds).
Step 12 cadence-list cadence-id cycle-1-on-time
cycle-1-off-time [cycle-2-on-time cycle-2-off-time]
[cycle-3-on-time cycle-3-off-time] [cycle-4-on-time
cycle-4-off-time]
Example:
Router(config-voice-class)# cadence-list 1 0 1000
(Optional) Specifies a tone cadence pattern to be
detected. Specify an on time and off time for each
cycle of the cadence pattern.
The arguments are as follows:
cadence-idRange is 1 to 10. There is no
default.
cycle-N-on-timeRange is 0 to 1000 (0 ms to
10 seconds). Default is 0.
cycle-N-off-timeRange is 0 to 1000 (0 ms to
10 seconds). Default is 0.
Step 13 cadence-variation time
Example:
Router(config-voice-class)# cadence-variation 200
(Optional) Specifies the maximum time that the
tone onset can vary from the specified onset time
and still be detected, in 10-millisecond
increments. Range is 0 to 200 (0 ms to 2 seconds).
Default is 0.
Step 14 exit
Example:
Router(config-voice-class)# exit
Exits voice class configuration mode.
Command Purpose

77
Configuring Timeouts Parameters
To change timeouts parameters, use the following commands:
SUMMARY STEPS
1. enable
4. timeouts call-disconnect seconds
5. timeouts initial seconds
6. timeouts interdigit seconds
7. timeouts ringing {seconds | infinity}
8. timeouts wait-release {seconds | infinity}
9. exit
Step 15 voice-port slot/subunit/port
Example:
Router(config)# voice-port 0/1/0
Step 16 supervisory disconnect dualtone {mid-call |
pre-connect} voice-class tag
Example:
Router(config-voiceport)# supervisory disconnect
dualtone mid-call voice-class 1
Assigns an FXO supervisory disconnect tone
voice class to the voice port.
Step 17 supervisory disconnect anytone
Example:
Router(config-voiceport)# supervisory disconnect
anytone
Configures the voice port to disconnect on receipt
of any tone.
Step 18 exit
Example:
Exits voice-port configuration mode and
completes the configuration.
Command Purpose

78
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Example:
Command Reference.
Step 4 timeouts call-disconnect seconds
Example:
Router(config-voiceport)# timeouts call-disconnect 60
Configures the call disconnect timeout value in
seconds. Range is 0 to 120. Default is 60.
Step 5 timeouts initial seconds
Example:
Router(config-voiceport)# timeouts initial 10
Sets the number of seconds that the system waits
between the caller input of the initial digit and the
subsequent digit of the dialed string. If the wait time
expires before the destination is identified, a tone
sounds and the call ends.
The seconds argument is the initial timeout
duration. Range is 0 to 120. Default is 10.
Step 6 timeouts interdigit seconds
Example:
Router(config-voiceport)# timeouts interdigit 10
Configures the number of seconds that the system
waits after the caller has input the initial digit or a
subsequent digit of the dialed string. If the timeout
ends before the destination is identified, a tone
sounds and the call ends. This value is important
when you are using variable-length dial peer
destination patterns (dial plans).
The seconds argument is the interdigit timeout
wait time in seconds. Range is 0 to 120. Default
is 10.
Step 7 timeouts ringing {seconds | infinity}
Example:
Router(config-voiceport)# timeouts ringing infinity
Specifies the duration that the voice port allows
ringing to continue if a call is not answered.
Default for seconds is 180.

79
Changing Timing Parameters
To change timing parameters, use the following commands:
SUMMARY STEPS
1. enable
4. timing clear-wait milliseconds
5. timing delay-duration milliseconds
6. timing delay-start milliseconds
7. timing delay-with-integrity milliseconds
8. timing dial-pulse min-delay milliseconds
9. timing dialout-delay millisecond
10. timing digit milliseconds
11. timing guard-out milliseconds
12. timing hookflash-out milliseconds
13. timing interdigit milliseconds
14. timing percentbreak percent
15. timing pulse pulses-per-second
16. timing pulse-digit milliseconds
17. timing pulse-interdigit milliseconds
18. timing wink-duration milliseconds
19. timing wink-wait milliseconds
20. exit
Step 8 timeouts wait-release {seconds | infinity}
Example:
Router(config-voiceport)# timeouts wait-release 30
Specifies the duration that a voice port stays in the
call-failure state while the Cisco device sends a busy
tone, reorder tone, or an out-of-service tone to the
port.
Default for seconds is 30.
Step 9 exit
Example:
the configuration.
Command Purpose

80
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Example:
Command Reference.
Step 4 timing clear-wait milliseconds
Example:
Router(config-voiceport)# timing clear-wait 200
(E&M only) Specifies the minimum amount of time,
in milliseconds, between the inactive seizure signal
and clearing of the call.
Range is 200 to 2000. Default is 400.
Step 5 timing delay-duration milliseconds
Example:
Router(config-voiceport)# timing delay-duration 100
(E&M only) Specifies the delay signal duration for
delay-dial signaling, in milliseconds.
Step 6 timing delay-start milliseconds
Example:
Router(config-voiceport)# timing delay-start
milliseconds
(E&M only) Specifies minimum delay time, in
milliseconds, from outgoing seizure to outdial
address.
Step 7 timing delay-with-integrity milliseconds
Example:
Router(config-voiceport)# timing delay-with-integrity
0
(Cisco MC3810 E&M ports only) Specifies duration
of the wink pulse for the delay dial, in milliseconds.
Step 8 timing dial-pulse min-delay milliseconds
Example:
Router(config-voiceport)# timing dial-pulse min-delay
300
Specifies time, in milliseconds, between the
generation of wink-like pulses when the type is
pulse.
Range is 0 to 5000. Default is 300 for Cisco 3600
series and 140 for Cisco MC3810.
Step 9 timing dialout-delay milliseconds
Example:
Router(config-voiceport)# timing dialout-delay 100
(Cisco MC3810 only) Specifies dial-out delay, in
milliseconds, for the sending digit or cut-through on
an FXO trunk or an E&M immediate trunk.

81
Step 10 timing digit milliseconds
Example:
Router(config-voiceport)# timing digit 50
Specifies the DTMF digit signal duration in
milliseconds.
Step 11 timing guard-out milliseconds
Example:
Router(config-voiceport)# timing guard-out 300
(FXO ports only) Specifies the duration in
milliseconds of the guard-out period that prevents
this port from seizing a remote FXS port before the
remote port detects a disconnect signal.
Step 12 timing hookflash-out milliseconds
Example:
Router(config-voiceport)# timing hookflash-out 500
Specifies the duration, in milliseconds, of the
hookflash.
Step 13 timing interdigit milliseconds
Example:
Router(config-voiceport)# timing interdigit 100
Specifies the dual-tone multifrequency (DTMF)
interdigit duration, in milliseconds.
Step 14 timing percentbreak percent
Example:
Router(config-voiceport)# timing percentbreak 20
(Cisco MC3810 FXO and E&M ports only) Specifies
the percentage of the break period for the dialing
pulses, if different from the default.
Step 15 timing pulse pulses-per-second
Example:
Router(config-voiceport)# timing pulse 20
(FXO and E&M only) Specifies the pulse dialing rate
in pulses per second.
Step 16 timing pulse-digit milliseconds
Example:
Router(config-voiceport)# timing pulse-digit 10
(FXO only) Configures the pulse digit signal
duration.
Step 17 timing pulse-interdigit milliseconds
Example:
Router(config-voiceport)# timing pulse-interdigit 500
(FXO and E&M only) Specifies pulse dialing
interdigit timing in milliseconds.
Step 18 timing wink-duration milliseconds
Example:
Router(config-voiceport)# timing wink-duration 200
(E&M only) Specifies maximum wink-signal
duration, in milliseconds, for a wink-start signal.
Command Purpose

82
Configuring the DTMF Timer
To configure the DTMF timer, use the following commands:
SUMMARY STEPS
1. enable
3. controller T1 number
4. ds0-group channel-number timeslots range type signaling-type dtmf dnis
5. cas-custom channel
6. dtmf timer-inter-digit milliseconds
7. exit
DETAILED STEPS
Step 19 timing wink-wait milliseconds
Example:
Router(config-voiceport)# timing wink-wait 200
(E&M only) Specifies maximum wink-wait duration,
in milliseconds, for a wink-start signal.
Step 20 exit
Example:
the configuration.
Command Purpose
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Step 3 controller T1 number
Example:
Router(config)# controller T1 1
Configures a T1 controller and enters controller
configuration mode.

83
Configuring Comfort Noise and Music Threshold for VAD
In normal voice conversations, only one person speaks at a time. Circuit-switched telephone networks
dedicate a bidirectional 64 kbps channel for the duration of each conversation, regardless of whether
anyone is speaking at the moment. This means that, in a normal voice conversation, at least 50 percent
of the bandwidth is wasted when one or both parties are silent. This figure can actually be much higher
when normal pauses and breaks in conversation are taken into account.
Packet-switched voice networks can use this wasted bandwidth for other purposes when voice activity
detection (VAD) is configured. VAD works by detecting the magnitude of speech in decibels and
deciding when to stop segmenting voice packets into frames. VAD has some technological problems,
however, which include the following:
General difficulties determining when speech ends
Clipped speech when VAD is slow to detect that speech is beginning again
Automatic disabling of VAD when conversations take place in noisy surroundings
VAD is configured in dial peers; by default it is enabled. Two parameters associated with VAD, music
threshold and comfort noise, are configured on voice ports.
If VAD is enabled, use the following commands to adjust music threshold and comfort noise:
Step 4 ds0-group channel-number timeslots range type
signaling-type dtmf dnis
Example:
Router(config-controller)# ds0-group 0 timeslots
1-4 type e&m-immediate-start dtmf dnis
Configures channelized T1 time slots, which enables a
Cisco AS5300 modem to answer and send an analog
call.
Step 5 cas-custom channel
Example:
Router(config-controller)# cas-custom 2
Enters cas-controller configuration mode and
customizes signaling parameters for a particular E1 or
T1 channel group on a channelized line.
Step 6 dtmf timer-inter-digit milliseconds
Example:
Router(conf-ctrl-cas)# dtmf timer-inter-digit 100
Configures the DTMF interdigit timer for a DS0 group.
Step 7 exit
Example:
Router(conf-ctrl-cas)# exit
Exits cas-controller configuration mode and completes
the configuration.
Command Purpose

84
SUMMARY STEPS
1. enable
3. dial-peer voice tag voip
4. vad [aggressive]
5. exit
6. voice vad-time milliseconds
8. music-threshold number
9. comfort-noise
10. exit
DETAILED STEPS
Command Purpose
Step 1 enable
Example:
Router> enable
Example:
Step 3 dial-peer voice tag voip
Example:
Router(config)# dial-peer voice 555 voip
Enters dial-peer configuration mode.
Step 4 vad [aggressive]
Example:
Router(config-dial-peer)# vad
Enables VAD for calls using this dial peer.
Note VAD is enabled by default. Use the vad
command only if you have previously disabled
the feature by using the no vad command.
Step 5 exit
Example:
Router(config-dial-peer)# exit
Exits dial-peer configuration mode.
Step 6 voice vad-time milliseconds
Example:
Router(config)# voice vad-time 500
Modifies the minimum silence detection time for VAD.

85
coincidental.
Example:
platform-specific. For information, refer to the
Step 8 music-threshold number
Example:
Router(config-voiceport)# music-threshold -70
Specifies the minimal decibel level of music played
when calls are put on hold. The decibel level affects
how VAD treats the music data.
Valid values range from 70 to 30. If the music
threshold is set too high and VAD is configured,
the remote end hears no music; if the level is set
too low, there is unnecessary voice traffic. Default
is 38.
Step 9 comfort-noise
Example:
Router(config-voiceport)# comfort-noise
Creates subtle background noise to fill silent gaps
during calls when VAD is enabled on voice dial peers.
If comfort noise is not generated, the resulting silence
can fool the caller into thinking the call is disconnected
instead of being merely idle.
Comfort noise is enabled by default.
Step 10 exit
Example:
Command Purpose

86

Echo cancellation is a key function in packet voice. Much of the perceived quality of the connection
depends on the performance of the echo canceller. The G.168 extended echo cancellation (EC) provides
an alternative to the proprietary Cisco G.165 EC with improved performance for trunking gateway
applications.
The following sections provide configuration information for echo cancellation:
Information About Echo Cancellation, page 87
How to Configure the Extended G.168 Echo Canceller, page 95
Configuration Examples for Extended G.168 Echo Cancellation, page 104
Information About Echo Cancellation
To use the feature, you should understand the following concepts:
Voice Call Transmit and Receive Paths, page 87
Echo Cancellation, page 88
Echo Canceller Operation, page 89
Echo Canceller Components, page 90
Echo Canceller Coverage, page 90
ITU-T Echo Cancellation History, page 91
Extended G.168 Echo Canceller Features, page 92
Extended EC Comparison, page 92
Extended Echo Canceller Support by Platform, page 93
Voice Call Transmit and Receive Paths
Every voice conversation has at least two participants. From the perspective of each participant, there
are two voice paths in every call:

88
Transmit path (also called the send or Tx path)The transmit path is created when a person speaks.
The sound is transmitted from the mouth of the speaker to the ear of the listener.
Receive path (also called the return or Rx path)The receive path is created when a person hears
the conversation. The sound is received by the ear of the listener from the mouth of the speaker.
Figure 1 shows a simple voice call between caller A and caller B. The top line represents the Tx path for
caller A, which becomes the Rx path for caller B. The bottom line represents the Tx path for caller B,
which becomes the Rx path for caller A.
Figure 1 Transmit and Receive Paths in a Voice Network
Echo Cancellation
Echo is the sound of your own voice reverberating in the telephone receiver while you are talking. When
timed properly, echo is not a problem in the conversation; however, if the echo interval exceeds
approximately 25 milliseconds (ms), it can be distracting to the speaker. In the traditional telephony
network, echo is generally caused by an impedance mismatch when the four-wire network is converted
to the two-wire local loop. Echo is controlled by echo cancellers (ECs).
A packet voice gateway, which operates between a digital packet network and the PSTN, can include
both digital (time division multiplexing [TDM]) and analog links. The analog circuit is known as the tail
circuit. It forms the tail or termination of the call from the perspective of the person experiencing the
echo. The tail circuit is everything connected to the PSTN side of a packet voice gatewayall the
switches, multiplexers, cabling, and PBXs between the voice gateway and the telephone.
Figure 2 shows a common voice network where echo cancellation might be used.
Figure 2 Echo Cancellation Network
An echo canceller reduces the level of echoes that leak from the Rx path (from the gateway out into the
tail circuit) into the Tx path (from the tail circuit into the gateway). From the perspective of the echo
canceller in a voice gateway, the Rx signal is a voice coming across the network from another location.
The Tx signal is a mixture of the voice call in the other location and the echo of the original voice, which
comes from the tail circuit on the initiating end and is sent to the receiving end.
8
8
5
6
0
Voice network
Tx
Rx
Rx
Tx
A B
B's voice
A's voice
V V
IP
6
2
0
8
6
Analog Analog
IP phone
Digital Digital
E0 E0
PSTN
PSTN
IP network

89
Echo cancellers face into the PSTN tail circuit. They eliminate echoes in the tail circuit on its side of the
network.The echo canceller in the originating gateway looks out into the tail circuit and is responsible
for eliminating the echo signal from the initiation Tx signal and allowing a voice call to go through
unimpeded. By design, ECs are limited by the total amount of time they wait for the reflected speech to
be received, which is known as an echo tail. The echo tail is normally 32 ms.
Note Delay and jitter in the WAN do not affect the operation of the echo canceller because the tail circuit,
where the echo canceller operates, is static.
Echo cancellation is implemented in digital signal processor (DSP) firmware (DSPWare) on Cisco voice
gateways and is independent of other functions implemented in the DSP (the DSP protocol and
compression algorithm). In voice packet-based networks, ECs are built into the low-bit-rate codecs and
are operated on each DSP.
Figure 3 shows a typical DSP channel configured for voice processing.
Figure 3 DSP Channel Configured for Voice Processing
Echo Canceller Operation
An echo canceller removes the echo portion of the signal coming out of the tail circuit and headed into
the WAN. It does so by learning the electrical characteristics of the tail circuit and forming its own model
of the tail circuit in its memory, and creating an estimated echo signal based on the current and past Rx
signal. It subtracts the estimated echo from the actual Tx signal coming out of the tail circuit. The quality
of the estimation is continuously improved by monitoring the estimation error.
Following are descriptions of the primary measurements of relative signal levels used by echo cancellers.
They are all expressed in decibels (dB).
Echo return loss (ERL)Reduction in the echo level produced by the tail circuit without the use of
an echo canceller. If an Rx speech signal enters the tail circuit from the network at a level of X dB,
the echo coming back from the tail circuit into the echo canceller is X less ERL.
Telephony
interface
Tone generation
and detection
DTMF
call progress
PCM interface
Echo canceller
Fax detection
Speech coder
Packetizer
and
dejitterer
Management
interface
Host
interface
RTOS
Data flow
Control flow
5
6
4
0
6
Voice activity
detection

90
Echo return loss enhancement (ERLE)Additional reduction in echo level accomplished by the
echo canceller. An echo canceller is not a perfect device; the best it can do is attenuate the level of
the returning echo. ERLE is a measure of this echo attenuation. It is the difference between the echo
level arriving from the tail circuit at the echo canceller and the level of the signal leaving the echo
canceller.
Acombined (ACOM)Total ERL seen across the terminals of the echo canceller. ACOM is the sum
of ERL + ERLE, or the total ERL seen by the network.
For more information about the echo canceller, refer to the Echo Analysis for Voice over IP document.
Echo Canceller Components
A typical echo canceller includes two components: convolution processor (CP) and a nonlinear
processor (NLP).
Convolution Processor
The CP first stage captures and stores the outgoing signal toward the far-end hybrid. The CP then
switches to monitoring mode and, when the echo signal returns, estimates the level of the incoming echo
signal and subtracts the attenuated original voice signal from the echo signal.
The time required to adjust the level of attenuation needed in the original signal is called the convergence
time. Because the convergence process requires that the voice signal be stored in memory, the EC has
limited coverage of tail circuit delay, normally 64, 96, and up to 128 ms. After convergence, the CP
provides about 18 dB of ERLE. Because a typical analog phone circuit provides at least 12 dB of ERL
(that is, the echo path loss between the echo canceller and the far-end hybrid), the expected permanent
ERL of the converged echo canceller is about 30 dB or greater.
Nonlinear Processor
In single-talk mode, that is, when one person is talking and the other is silent, the NLP replaces the
residual echo at the output of the echo canceller with comfort noise based on the actual background noise
of the voice path. The background noise normally changes over the course of a phone conversation, so
the NLP must adapt over time. The NLP provides an additional loss of at least 25 dB when activated. In
double-talk mode, the NLP must be deactivated because it would create a one-way voice effect by adding
25 to 30 dB of loss in only one direction.
To completely eliminate the perception of echo, the talker echo loudness rating (TELR) should be greater
than 65 dB in all situations. To reflect this reality, ITU-T standard G.168 requires an ERL equal to or
greater than 55 dB. Segmentation local reference (SLR), receive loudness rating (RLR), and cell loss
ratio (CLR) along the echo path should allow another 10 dB to meet the expected TELR. CP, NLP and
loudness ratings (LRs) must be optimized to make sure that echo is canceled effectively.
Echo Canceller Coverage
Echo canceller coverage (also known as tail coverage or tail length) is the length of time that the echo
canceller stores its approximation of an echo in memory. It is the maximum echo delay that an echo
canceller is able to eliminate.
The echo canceller faces into a static tail circuit with input and an output. If a word enters a tail circuit,
the echo is a series of delayed and attenuated versions of that word, depending on the number of echo
sources and the delays associated with them. After a certain period of time, no signal comes out. This

91
time period is known as the ringing time of the tail circuitthe time required for all of the ripples to
disperse. To fully eliminate all echoes, the coverage of the echo canceller must be as long as the ringing
time of the tail circuit.
ITU-T Echo Cancellation History
ITU-T standard G.164 defines the performance of echo suppressors, which are the predecessors of echo
cancellation technology. G.164 also defines the disabling of echo suppressors in the presence of 2100 Hz
tones (which precede low bit rate modems).
ITU-T standard G.165 defines echo cancellation and provides a number of objective tests that ensure a
minimum level of performance. These tests check convergence speed of the echo canceller, stability of
the echo canceller filter, performance of the nonlinear processor, and a limited amount of double talk
testing. The signal used to perform these tests is white noise. Additionally, G.165 defines the disabling
of echo cancellers in the presence of 2100 Hz signals with periodic phase reversals in order to support
echo cancelling modem technology (V.34, for example), which does not work if line echo cancellation
is performed in the connection.
ITU-T standard G.168 allows more rigorous testing and satisfies more testing requirements. White noise
is replaced with a pseudospeech signal for the convergence tests. Most echo cancellation algorithms use
a least mean square (LMS) algorithm to adapt the echo cancellation filter. LMS works best with random
signals, and slows down with more correlated signals such as speech. Using the pseudospeech signal in
testing provides a more realistic portrayal of the echo cancellers performance in real use.
In Cisco IOS Release 12.3(4)XD and later releases, the G.168 EC is the default and you can no longer
select the Cisco G.165 EC on any supported platform except the Cisco AS5300. The Cisco AS5300 still
supports the Cisco G.165 EC and the extended G.168 EC. Table 1 provides a summary of the Extended
ITU-T G.168 Echo Cancellation feature availability in Cisco IOS releases.
Table 1 Feature History for Extended ITU-T G.168 Echo Cancellation
Release Modification
12.2(13)T This feature was introduced.
12.2(13)ZH The extended G.168 EC became the default on the Cisco 1700 series and
the Cisco ICS 7750.
12.2(15)ZJ The extended G.168 EC became the default on the Cisco 2600 series,
Cisco 3600 series, Cisco 3700 series.
Note The extended G.168 EC is not supported on the High-Density
Analog Network Modules (NM-HDA) and Asynchronous Interface
Module (AIM)-Voice modules on the Cisco 2600 series in this
release.
12.3(1) The extended G.168 EC became the default on the Cisco IAD2420,
Cisco MC3810, and Cisco VG200.
12.3(4)T The extended G.168 EC became the default on the Cisco 7200 series and
Cisco Catalyst 4000 AGM.
12.3(4)XD The G.168 extended EC became the only EC on all voice packet platforms
that support the extended G.168 EC; the Cisco G.165 EC is no longer a
selectable option.
Note The Cisco AS5300 still supported choosing between the
Cisco G.165 EC and the extended G.168 EC.

92
Finding Support Information for Platforms and Cisco IOS and Catalyst OS Software Images
Extended G.168 Echo Canceller Features
Configuration and reporting of extended echo path capacity and worst-case ERL
Test mode support for manually freezing, thawing, and clearing the EC h-register
Reporting of statistics for location of the largest reflector and the internal state of the EC
No changes to platformImproves platform functionality by updating the EC module through a
DSPWare upgrade and a Cisco IOS software upgrade
Enabling and disabling of nonlinear processorEnables and disables NLP spectrally matched
comfort noise
ERL configurationCan be set to three values: 0 dB, 3 dB, and 6 dB
Expansion of EC capacityEC capacity is expanded to 64 ms (128 ms in Release 12.4(20)T or later)
Extended EC Comparison
Table 2 contains comparison information for G.165 and G.168 echo cancellation.
12.3(3) The G.148 extended EC was configurable with no codec restriction on the
Cisco AS5300.
12.3(8)XY The G.168 extended EC was supported for the WS-SVC-CMM-6T1,
WS-SVC-CMM-6E1, and WS-SVC-CMM-24FXS port adapters on the
Cisco Communication Media Module (WS-SVC-CMM).
12.3(9) The extended G.168 EC was supported for the NM-HDA and AIM-Voice
modules on the Cisco 2600 series.
12.3(11)T The dual-filter G.168 echo canceller capability was added to NextPort SPE
firmware (SPEware) version 10.2.2 and later versions with Cisco IOS
Release 12.3(11)T and later releases. See the chapter NextPort-Based Voice
Tuning and Echo Cancellation.
12.4(20)T Software-configurable echo cancel coverage was extended to 80, 96, 112,
and 128 ms. The default parameter is 128 ms.
Table 2 Echo Canceller Comparison
Feature G.165 EC G.168 EC
Tail Coverage Up to 32 ms Up to 64 ms (128 ms in Release
12.4(20)T or later
Minimum ERL Greater than or equal to 6 dB Configurable to greater than or equal to
0 dB, 3 dB, or 6 dB
Echo Suppression Up to 10 seconds Not required because of faster
convergence

93
Extended Echo Canceller Support by Platform
Table 3 lists the support for the extended G.168 EC by platform, network module, high-complexity and
medium-complexity codecs, and minimum Cisco IOS release.
Table 3 Extended Echo Canceller Algorithm Coverage by Platform
Platform Network Module High Complexity Codec
Medium Complexity
Codec Comments
Analog Digital Analog Digital
Cisco 1700 series 12.2(8)YN
12.2(13)T
12.2(8)YN
12.2(13)T
12.2(8)Y
N
12.3(2)T
12.2(8)YN
12.3(2)T
Flexi6 support in
Cisco IOS
Release 12.2(8)YN.
Cisco 2600 series
Cisco 2600XM
Cisco 3600 series
Cisco 3700 series
Cisco VG200
NM-HDV (C549) 12.2(13)T 12.2(13)T Full support.
Cisco 2600 series
Cisco 2691
Cisco 3600 series
Cisco 3700 series
Cisco VG200
NM-1V,
NM-2V
(C542)
Not supported.
Cisco 2600XM
Cisco 2691
Cisco 3640
Cisco 3660
Cisco 3700 series
NM-HDxx 12.3(4)XD 12.3(4)XD 12.3(4)X
D
12.3(4)XD
Cisco 2600XM
Cisco 2691
Cisco 3640
Cisco 3660
Cisco 3700 series
AIM-Voice (C5421),
AIM-Voice-30 (C542)
12.2(15)ZJ
12.3(4)T
12.2(15)ZJ
12.3(4)T
AIM.
Cisco 2600XM
Cisco 2691
Cisco 3640
Cisco 3660
Cisco 3700 series
NM-HDA (C5421) 12.2(15)ZJ
12.3(4)T
12.2(15)Z
J,
12.3(4)T
12.2(15)ZJ
12.3(4)T
NM-HDA.
Note G.728 high
complexity is
not supported.
Cisco 2600 series NM-HDA (C5421) 12.3(9) 12.3(9)
Cisco 2600 series AIM-Voice (C5421) 12.3(9) 12.3(9)
Cisco 7200 series PA-VXx-2TE1+ and
PA-MCX-nTE1
12.2(13)T 12.2(13)T PA-MCX-nTE1 port
adapters do not have
their own DSPs, so they
use the DSPs of
PA-VXx-2TE1+ port
adapters.
Cisco 7500 series 12.2(13)T No medium complexity.

94
Cisco 7600 series Communiction Media
Module
(WS-SVC-CMM) with
one of the following
port adapters:
WS-SVC-CMM-6T1
WS-SVC-CMM-6E1
12.3(8)XY
12.3(14)T

WS-SVC-CMM-24FXS 12.3(8)XY
12.3(14)T

Cisco AS5300 12.2(13)T
(restricted)
12.3(3)
(unrestricted)
1-channel DSP on C549
with extended EC, any
codec (unrestricted).
Cisco AS5350
Cisco AS5400
Cisco AS5850
NextPort DFC modules:
DFC60
DFC108
1 CT3_UPC 216
UPC324
Digital -
12.3(11)T
12.3(11)T See the NextPort-Based
Voice Tuning and Echo
Cancellation chapter in
this guide.
Cisco Catalyst
4000
AGM 12.3(4)T 12.3(4)T High-complexity
analog and
medium-complexity
digital is planned.
Cisco Catalyst
6000
Cisco 6624 A002040-
00002
A002040-
00002

Cisco 6608 A004040-
00002
A004040-
00002
Cisco Catalyst
6500 series
Communiction Media
Module
(WS-SVC-CMM) with
one of the following
port adapters:
WS-SVC-CMM-6T1
WS-SVC-CMM-6E1
12.3(8)XY
12.3(14)T

WS-SVC-CMM-24FXS 12.3(8)XY
12.3(14)T

Cisco IAD2420 12.2(13)T 12.2(13)T 12.3(1)
mainline
12.3(1)
mainline
Cisco IAD243x VIC2-4FXO onboard

T1
12.3(4)XD 12.3(4)XD 12.3(4)X
D
12.3(4)XD
Cisco ICS 7750 12.2(13)T 12.2(13)T 12.2(13)T 12.2(13)T Flexi6 support.
Cisco MC3810 HCM 549 12.2(13)T 12.2(13)T 12.3(1)
mainline
12.3(1)
mainline
Table 3 Extended Echo Canceller Algorithm Coverage by Platform (continued)

Platform Network Module High Complexity Codec
Medium Complexity
Codec Comments
Analog Digital Analog Digital

How to Configure the Extended G.168 Echo Canceller
95
How to Configure the Extended G.168 Echo Canceller
The Extended ITU-T standard G.168 Echo Cancellation feature provides an alternative to the default
proprietary Cisco G.165 EC. Beginning in Release 12.4(20)T, the extended EC provides improved
performance for trunking gateway applications and provides a configurable tail length that supports up
to 128 ms of echo cancellation. The G.165 EC is not configurable in Cisco IOS Release 12.3(4)XD and
later releases, except on the Cisco AS5300.
Note Extended echo cancellation is configured differently depending on the version of Cisco IOS software
that you are using. If you are using Cisco IOS Release 12.3(4)XD or a later release, you do not have to
use any Cisco IOS commands to enable the Extended ITU-T standard G.168 Echo Cancellation feature
because the extended G.168 EC is the only available echo canceller. You have the option of disabling the
extended EC, but it is highly recommended that you leave it enabled.
To configure the NextPort dual-filter G.168 echo canceller, see the NextPort-Based Voice Tuning and
Echo Cancellation chapter in this guide.
This section contains the following procedures:
Changing Echo Cancellers on Digital Voice Ports in Releases Prior to Cisco IOS Release 12.3(4)XD,
page 96
Configuring the Extended G.168 EC on the Cisco AS5300, page 98
Modifying Echo Cancellation Default Settings, page 101
Table 4 lists the Cisco IOS commands that are used for selecting the extended G.168 EC based on your
platform and Cisco IOS release.
Table 4 Cisco IOS Commands for Selecting Extended E.168 EC by Platform and Cisco IOS
Release
Cisco IOS Release Cisco IOS Command
Cisco 1700 Series and Cisco ICS 7750
12.2(13)T
Router(config)# voice echo-canceller extended
12.2(13)ZH
12.2(15)ZJ
12.3(1)
Router(voice-card)# codec complexity medium
12.3(4)T and later No configuration necessary. G.168 EC enabled by default.
Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series, Cisco MC3810, Cisco VG200
12.2(13)T
12.2(13)ZH
12.3(1)
Router(voice-card)# codec complexity medium ecan-extended
or
Router(voice-card)# codec complexity high ecan-extended
12.2(15)ZJ
12.3(4)T
12.3(4)XD and later No configuration necessary. G.168 EC enabled by default.
Cisco 7200 Series and Cisco 7500 Series
12.2(13)T
Router(config-dspfarm)# codec complexity medium ecan-extended
12.2(13)ZH and later No configuration necessary. G.168 EC enabled by default.
Cisco 7600 Series with Communication Media Modules

Restrictions for G.168 Extended Echo Canceller
96
Not all Cisco platforms that use C542 or C549 DSPs support the extended EC.
The G.168 extended EC is not supported on the Cisco AS5300 in Cisco IOS Release 12.2(13)ZH.
The Cisco 1700 series does not support the T1/E1 card in Cisco IOS Release 12.2(13)T.
Changing Echo Cancellers on Digital Voice Ports in Releases Prior to Cisco IOS
Release 12.3(4)XD
To use the G.168 extended EC in a release prior to Cisco IOS Release 12.3(4)XD, perform one of the
following tasks depending on your hardware platform:
Cisco 1700 or Cisco ICS 7750
Enabling the Extended G.168 EC on the Cisco 1700 Series and Cisco ICS 7750 in Cisco IOS
Release 12.2(13)T, page 97
Release 12.2(13)ZH, page 97
Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series, Cisco MC3810, and Cisco VG200
Changing Codec Complexity on Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series, and
Cisco MC3810 in the Con figuring Digital Voice Ports chapter.
Cisco 7200 Series and Cisco 7500 Series
Configuring Codec Complexity on Cisco 7200 Series and Cisco 7500 Series Routers in the
Configuring Digital Voice Ports chapter
Note See Table 3 for extended EC algorithm coverage by platform.
12.3(8)XY and later No configuration necessary. G.168 EC enabled by default.
Cisco AS5300
12.2(13)T
Router(config)# voice echo-canceller extended codec small codec
large codec
12.3(3)
Router(config)# voice echo-canceller extended [codec small codec
large codec]
Cisco Catalyst 6500 Series with Communication Media Modules
12.3(8)XY and later No configuration necessary. G.168 EC enabled by default.
Cisco Catalyst 4000 AGM
12.3(4)T and later No configuration necessary. G.168 EC enabled by default.
Table 4 Cisco IOS Commands for Selecting Extended E.168 EC by Platform and Cisco IOS
Release
Cisco IOS Release Cisco IOS Command

97
Release 12.2(13)T
To change codec complexity on the Cisco 1700 series and Cisco ICS 7750 and switch between the
proprietary Cisco EC and the extended G.168 EC, use the following commands.
Note You must clear all calls on the system before using the following commands. If there are active calls on
the system, the commands are ignored and a warning message is issued.
SUMMARY STEPS
1. enable
3. voice echo-canceller extended
4. exit
DETAILED STEPS
Enabling the Extended G.168 EC on the Cisco 1700 Series and Cisco ICS 7750 in Cisco IOS Release 12.2(13)ZH
The codec complexity medium command enables the extended echo canceller by default on the
Cisco 1700 series and the Cisco ICS 7750 in Cisco IOS Release 12.2(13)ZH.
Step 1 enable
Example:
Router> enable
Example:
Step 3 voice echo-canceller extended
Example:
Enables the G.168 extended echo canceller on the
Cisco 1700 series or Cisco ICS 7750.
You do not have to shut down all the voice ports
on the Cisco 1700 or Cisco ICS 7750 to switch
the echo canceller, but you should make sure
that when you switch the echo canceller, there
are no active calls on the router.
To return to the proprietary Cisco G.165 default
EC, use the no form of the command.
Step 4 exit
Example:
Exits global configuration mode.

98
Note You must clear all calls on the system before using the following commands. If there are active calls on
the system, the commands are ignored and a warning message is issued.
SUMMARY STEPS
1. enable
3. voice-card slot
5. end
DETAILED STEPS
Configuring the Extended G.168 EC on the Cisco AS5300
Perform this task to enable the Extended ITU-T standard G.168 Echo Cancellation feature on the
Cisco AS5300. You must designate which EC to use on the Cisco AS5300 C542 and C549 DSPM
high-complexity platform. You can use the extended EC with codec restrictions on the C542 DSP
firmware or with or without codec restrictions on the C549 DSP firmware.
Step 1 enable
Example:
Router> enable
Enables higher privilege levels, such as privileged
EXEC mode.
Example:
Example:
Enters voice card configuration mode on the
specified slot.
Step 4 codec complexity medium
Example:
Enables the extended EC (default).
Step 5 end
Example:
Router(voice-card)# end
Exits voice-card configuration mode and completes
the steps for configuring the extended EC on the
Cisco 1700 series and Cisco ICS 7750.

99
Note A firmware upgrade can be made by upgrading Cisco VCWare. For upgrade information, refer to the
Combined Version Release Notes and Compatibility Matrix for Cisco VCWare on Cisco AS5300
Universal Access Servers/Voice Gateways.
Prerequisites for Extended G.168 EC on Cisco AS5300
Extended EC with Restricted Codecs on C542 or C549 DSP
1. Create a backup of your router configuration.
2. Determine which codecs are required when enabling the extended echo canceller. For this
information, see the codec restriction options for the voice echo-canceller extended command.
a. Specify a single small codec (g711 or g726) and a single large codec (g723, g728, GSM FR,
GSM EFR, g729, or fax-relay). Any call setup involving other codecs is rejected.
b. If fax-relay is not selected as the large codec, the VoIP dial peer requires that you use the
fax rate disabled command in dial-peer configuration mode to reset the dial peer for voice
calls.
3. Review your existing configuration and look for all dial peers that select codecs or fax-relay
specification that are different from the codecs that you decide on. After choosing the codecs to be
supported by the extended echo canceller, either remove all dial peers with different codecs not
supported by your new configuration or modify the dial-peer codec selection by selecting a voice
codec or fax-relay that is supported by the new configuration.
4. Ensure that modem relay is not configured in any of the dial-peer configurations. If modem relay is
configured, it should be disabled using the no modem relay command.
Extended EC with Unrestricted Codecs on C549 DSP
1. Create a backup of your router configuration.
2. Ensure that modem relay is not configured in any of the dial-peer configurations. If modem relay is
configured, it should be disabled using the no modem relay command.
Restrictions for Extended G.168 EC on the Cisco AS5300
The extended G.168 EC can be used only in one of the following ways on the Cisco AS5300:
With a restricted set of codecs with C542 or C549 DSP firmware. Two channels of voice are
supported per DSP, and full call handling capacity is supported.
With no restrictions on codecs with C549 DSP firmware. One channel of voice is supported per
DSP, and call handling capacity is reduced by half.
Not all Cisco platforms that use C542 or C549 DSPs support the extended EC. Other platforms
continue to use the proprietary Cisco G.165 EC if they do not support the extended EC.
SUMMARY STEPS
1. enable
3. no dial-peer voice tag voip
4. dial-peer voice tag voip

100
5. codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 |
g726r32 | g726r53 |g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr}
[bytes payload-size]
6. exit
7. voice echo-canceller extended
or
voice echo-canceller extended [codec small codec large codec]
8. exit
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:
Step 3 no dial-peer voice tag voip
Example:
Router(config)# no dial-peer voice 1 voip
(Optional) Removes VoIP dial peers, one dial peer
at a time.
When configuring the extended EC in global
configuration mode, you must remove or
modify all existing VoIP dial peers before the
voice echo-canceller extended command is
accepted.
Step 4 dial-peer voice tag voip
Example:
Router(config)# dial-peer voice 1 voip
Enters dial-peer configuration mode so that you
can modify a codec type.
Step 5 codec {g711alaw | g711ulaw | g723ar53 | g723ar63 |
g723r53 | g723r63 | g726r16 | g726r24 | g726r32 |
g726r53 | g726r63 | g728 | g729abr8 | g729ar8 |
g729br8 | g729r8 | gsmefr | gsmfr} [bytes
payload-size]
Example:
Router(config-dialpeer)# codec g711alaw
Specifies the voice codec rate for the dial peer.
Step 6 exit
Example:
Router(config-dialpeer)# exit
Exits dial-peer configuration mode and returns to

101
Modifying Echo Cancellation Default Settings
Perform this task to modify the default settings for echo cancellation parameters.
SUMMARY STEPS
1. enable
3. voice-port slot/port:ds0-group-number
4. echo-cancel enable
5. echo-cancel coverage {24 | 32 | 48 | 64 | 80 | 96 | 112 | 128}
6. echo-cancel erl worst-case {0 | 3 | 6}
7. non-linear
8. echo-cancel suppressor seconds
9. end
Step 7 voice echo-canceller extended
or
voice echo-canceller extended [codec small codec large
codec]
Example:
Example:
Router(config)# voice echo-canceller extended codec
small g711 large fax-relay
Enables the extended echo canceller with no
restriction on codecs.
or
Enables the extended echo canceller with
restricted codecs.
The following codec choices are valid:
Small footprint codecG.711 or G.726.
Large footprint codecG.729, G.726,
G.728, G.723, fax-relay, GSM FR, or
GSM EFR.
Note The voice echo-canceller extended
command enables the extended EC
without codec restrictions on a
Cisco AS5300 with C549 DSP firmware.
One channel of voice is supported per
DSP. Any codec is supported.
Note The voice echo-canceller extended codec
command restricts codecs on the
Cisco AS5300 with C542 and C549 DSP
firmware. Two channels of voice are
supported per DSP. Only specific codecs
are supported.
Step 8 exit
Example:
Exits global configuration mode.

102
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Enables higher privilege levels, such as privileged EXEC
mode.
Example:
Step 3 voice-port slot/port:ds0-group-number
Example:
Enters voice-port configuration mode on the selected slot,
port, and DS0 group.
Note The syntax of this command is platform-specific. For
the syntax for your platform, refer to the Cisco IOS
Voice Command Reference.
Step 4 echo-cancel enable
Example:
Router(config-voiceport)# echo-cancel enable
Enables echo cancellation.
Echo cancellation is enabled by default. Use the no form
of this command to disable echo cancellation.
The extended G.168 EC is the default EC for all
supported platforms in Cisco IOS Release 12.3(4)XD
and later releases, except the Cisco AS5300.
On the Cisco AS5300, the Cisco G.165 EC is enabled by
default with echo suppression disabled.
Note This command is supported only when the
echo-cancel coverage command is enabled.
Step 5 echo-cancel coverage {24 | 32 | 48 | 64 | 80 |
96 | 112 | 128}
Example:
Router(config-voiceport)# echo-cancel coverage
64
Adjusts the size of the echo canceller (echo path capacity
coverage).
128 is the default in Cisco IOS Release 12.4(20)T and
later releases. Prior to this release, the default is 64 ms.
Note This command is supported only when echo
cancellation is enabled. See Step 4 of this procedure.

103
Step 6 echo-cancel erl worst-case [0 | 3 | 6]
Example:
Router(config-voiceport)# echo-cancel erl
worst-case 6
(Optional) Determines worst-case ERL in dB.
Worst-case ERL is the minimum expected attenuation in
the voice path. For example, if you have a worst-case
ERL of 6 (erl worst-case 6), when you speak into the
phone you can expect at least 6 dB of attenuation on the
signal by the time it gets back to the original source
(echo). In general you do not need to change this value
from 6, which is the default.
Worst-case ERL does not directly modify the inbound or
outbound signals. This is purely a configuration
parameter for the EC to help it distinguish between echo
and a new signal.
Note This command is supported for the extended G.168
EC only; it is not supported for the G.165 EC.
Step 7 non-linear
Example:
Router(config-voiceport)# non-linear
(Optional) Selects nonlinear processing (residual echo
suppression) in the EC, which either shuts off any signal or
mixes in comfort noise if no near-end speech is detected.
cancellation is enabled. See Step 4.
Nonlinear processing is enabled when the extended
G.168 EC is enabled. Use the no form of this command
to disable the NLP.

Configuration Examples for Extended G.168 Echo Cancellation
104
This section includes the following examples:
Enabling the Extended EC on the Cisco 7200 Series or Cisco 7500 Series: Example, page 106
Enabling the Extended Echo Canceller on the Cisco 2600 Series, Cisco 3600 Series,
Cisco 3700 Series, and Cisco VG200 in Releases Prior to Cisco IOS Release 12.3(4)XD: Example,
page 105
Enabling the Extended EC on the Cisco 7200 Series or Cisco 7500 Series: Example, page 106
Enabling the Extended Echo Canceller on the Cisco AS5300: Example, page 106
Adjusting the Echo Canceller Size: Example, page 107
Worst-Case Echo Return Loss: Example, page 107
Step 8 echo-cancel suppressor seconds
Example:
Router(config-voiceport)# echo-cancel
suppressor 10
(Optional) Applies echo suppression for the number of
seconds specified when using the G.165 EC.
This command cannot be used with the extended G.168
EC in Cisco IOS Release 12.2(15)ZJ or later releases, or
on NextPort (Cisco AS5350 and Cisco AS5400)
platforms.
Note This command is required to configure the Extended
ITU-T standard G.168 Echo Cancellation feature in
Cisco IOS Release 12.2(13)T.
For the AS5300, the Cisco G.165 EC is enabled by
default with echo suppression disabled. The echo
suppressor can be used only on T1 DSPs when the
default Cisco G.165 EC is used.
This command enables echo cancellation for voice that
is sent out an interface and received back on the same
interface within the configured amount of time.
This command reduces the initial echo before the echo
canceller can converge. In case of double-talk in the first
number of seconds, the code automatically disables the
suppressor.
The echo-cancel suppressor command is visible when
the G.168 extended EC is selected but it has no effect.
cancellation is enabled.
Step 9 end
Example:
Router(config-voiceport)# end
configuration.

105
Enabling the Extended EC on the Cisco 1700 Series and Cisco ICS 7750: Example
The following example enables the G.168 extended EC on a Cisco 1700 series or a Cisco ICS 7750. The
extended EC is enabled by default when the medium keyword is used in Cisco IOS Release 12.2(13)ZH
and later.
voice-card 1
codec complexity medium
Enabling the Extended Echo Canceller on the Cisco 2600 Series, Cisco 3600
Series, Cisco 3700 Series, and Cisco VG200 in Releases Prior to Cisco IOS
Release 12.3(4)XD: Example
The following example shows that the echo canceller has been changed from the default proprietary
Cisco EC to the extended EC in Cisco IOS releases prior to 12.3(4)XD.
Note The extended G.168 EC is the only EC in Cisco IOS Release 12.3(4)XD and later releases. Because it is
enabled by default, it does not display in the configuration output in Cisco IOS Release 12.3(4)XD and
later releases.
The following is example output from an originating Cisco 3640:
.
.
.
voice-card 1
codec complexity high ecan-extended
.
.
.
controller T1 1/0
framing esf
linecode b8zs
pri-group timeslots 1-24
!
voice-port 1/0:23
.
.
.
dial-peer voice 104001 voip
destination-pattern 104001
session target ipv4:10.2.0.104
dtmf-relay cisco-rtp
codec g711alaw
fax rate 14400
fax protocol cisco
.
.
.

106
Enabling the Extended EC on the Cisco 7200 Series or Cisco 7500 Series:
Example
The following example changes codec complexity on a Cisco 7200 series or 7500 series:
dspint dspfarm 2/0
codec medium ecan-extended
Enabling the Extended Echo Canceller on the Cisco AS5300: Example
The following example enables the extended G.168 EC with restricted codecs on the Cisco AS5300 with
C542 or C549 DSP firmware:
!
version 12.3
no service pad
service timestamps debug datetime msec
service timestamps log uptime
no service password-encryption
service internal
!
hostname router
!
boot-start-marker
boot-end-marker
!
enable secret 5 $123
enable password temp
!
!
resource-pool disable
!
no aaa new-model
ip subnet-zero
ip rcmd rcp-enable
ip rcmd rsh-enable
ip domain name cisco.com
ip host router1 10.10.101.14
!
!
isdn switch-type primary-5ess
!
voice echo-canceller extended codec small g711 large fax-relay
!
!
!
fax interface-type fax-mail
!
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
.
.
.

107
Adjusting the Echo Canceller Size: Example
The following example adjusts the size of the extended EC to 64 ms on Cisco 3600 series routers:
voice-port 1/0:0
echo-cancel coverage 64
Worst-Case Echo Return Loss: Example
The following example sets the worst-case echo return loss to 3:
voice-port 0:D
echo-canceller erl worst-case 3
playout-delay mode fixed
no comfort-noise
CCDE, CCENT, Cisco Eos, Cisco Lumin, Cisco Nexus, Cisco StadiumVision, the Cisco logo, DCE, and Welcome to the Human Network are
trademarks; Changing the Way We Work, Live, Play, and Learn is a service mark; and Access Registrar, Aironet, AsyncOS, Bringing the Meeting To
You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS,
Event Center, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQ Expertise, the iQ logo,
iQ Net Readiness Scorecard, iQuick Study, IronPort, the IronPort logo, LightStream, Linksys, MediaTone, MeetingPlace, MGX, Networkers,
Networking Academy, Network Registrar, PCNow, PIX, PowerPanels, ProConnect, ScriptShare, SenderBase, SMARTnet, Spectrum Expert,
StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and the WebEx logo are registered trademarks of
Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
coincidental.

108

Configuring Hardware Echo Cancellation on
T1/E1 Multiflex Voice/WAN Interface Cards
The multiflex trunk (MFT) dedicated echo cancellation modules (dedicated ECAN modules) are
daughter cards that attach to the second generation multiflex voice/WAN interface cards (MFT VWIC2
family). The dedicated ECAN modules are available in 32-channel and 64-channel configurations
(EC-MFT-32 and EC-MFT-64), which match the requirements of the 1- and 2-port T1/E1 MFT VWIC2s,
respectively. This chapter describes the configuration to enable additional echo cancellation
effectiveness:
Control of the echo canceller provided through the size of the echo cancellation buffer, ranging from
24 milliseconds (ms) to 128 ms
Processing and memory resources to ensure robust echo canceller coverage independent from the
configuration of the echo canceller or the demand placed on the general voice DSP resources
Prerequisites for Hardware Echo Cancellation on T1/E1
Multiflex Voice/WAN Interface Cards
Cisco IOS Image
To run hardware echo cancellation on T1/E1 interfaces, you must install an IP Plus or IP Voice image
(minimum) of Cisco IOS Release 12.3(14)T or a later release.
Baseboard and Daughter Card Configuration
Hardware echo cancellation is restricted to the same baseboard voice/WAN interface card (VWIC) on
which the daughter card (EC-MFT-32 and EC-MFT-64) is installed and cannot be shared by other T1/E1
controllers.

Information About Hardware Echo Cancellation on T1/E1 Multiflex Voice/WAN Interface Cards
110
Hardware Echo Cancellation Tail Length
If you are using hardware echo cancellation, the value for tail length is set to 128 ms. This is not
configurable and cannot be changed.
Information About Hardware Echo Cancellation on T1/E1
Table 1 lists the names and descriptions of the baseboard modules and daughter cards that offer hardware
echo cancellation on T1/E1 multiflex voice/WAN interface cards.
Figure 1 shows a sample network topology with a Cisco 3725 router using hardware echo cancellation.
Table 1 Modules with T1/E1 Multiflex Voice/WAN Interface Cards with Echo Cancellation
Functionality
Module Name Description
Baseboard VIC2-1MFT-T1/E1
1
1. The VIC2-1MFT-T1/E1 and VIC2-2MFT-T1/E1 voice modules do not support WAN connectivity.
1-port RJ-48 multiflex voice trunk (T1/E1)
VIC2-2MFT-T1/E1
1
2-port RJ-48 multiflex voice trunk (T1/E1)
VWIC2-1MFT-T1/E1 1-port RJ-48 multiflex voice/WAN trunk (T1/E1)
VWIC2-2MFT-T1/E1 2-port RJ-48 multiflex voice/WAN trunk (T1/E1)
VWIC2-1MFT-G703
2
2. Although unstructured G.703 operation is particular to E1 operation, the VWIC2-1/2MFT-G703 cards also support structured
T1 operation.
1-port RJ-48 multiflex trunk (E1 G.703)
VWIC2-2MFT-G703
2
2-port RJ-48 multiflex trunk (E1 G.703)
Daughter cards for
hardware echo
cancellation
EC-MFT-32 32-channel echo canceller module for multiflex trunk
EC-MFT-64 64-channel echo canceller module for multiflex trunk

How to Configure Hardware Echo Cancellation on T1/E1 Multiflex Voice/WAN Interface Cards
111
Figure 1 Sample Network Topology for the T1/E1 Multiflex Voice/WAN Interface Cards with
Echo Cancellation Module
How to Configure Hardware Echo Cancellation on T1/E1
To configure hardware echo cancellation on T1/E1 multiflex voice/WAN interface cards, complete the
following tasks.
SUMMARY STEPS
1. enable
3. card type {e1 | t1} slot subslot
4. voice-card slot
5. voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
6. echo-cancel enable type [hardware | software]
7. echo-cancel coverage {24 | 32 | 48 | 64}
8. exit
V
Applications
server
Cisco CallManager
7200
T1
FXS
NM-2V/NM-HDA
Calling party
Called
party
1
2
7
1
5
9
IP IP
IP
IP
IP
IP
V
PSTN
Echo
Echo
Cisco 3725
VWIC2-1MFT-T1/E1 with 32-channel HWECAN
NMD-36ESW with ILP
Line powered
IP
IP
IP
WAN
...
...
...
...

112
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:
Step 3 card type {e1 | t1} slot subslot
Example:
Router(config)# card type t1 1 0
Sets or changes the card type to E1 or T1.
slotSpecifies the slot number. Range can be 0 to 6,
depending on the platform.
subslotSpecifies the VWIC slot number. Range can
be 0 to 3, depending on the host module or platform.
When the command is used for the first time, the
configuration takes effect immediately.
A subsequent change in the card type will not take
effect unless you enter the reload command or reboot
the router.
Note When you are using the card type command to
change the configuration of an installed card, you
must enter the no card type {e1 | t1} slot subslot
command first. Then enter the card type {e1 | t1}
slot subslot command for the new configuration
information.
Example:
Router(config)# voice card 1
Enters voice card configuration mode.
Specify the slot location using a value from 0 to 5.

113
Step 5 voice-port {slot-number/subunit-number/port |
slot/port:ds0-group-number}
Example:
Router(voice-card)# voice-port 3/0:0
Enters voice port configuration mode and specifies the
voice port.
The slot-number argument identifies the slot where the
voice interface card (VIC) is installed. Valid entries are
from 0 to 3, depending on the slot in which it has been
installed.
The subunit-number identifies the subunit on the VIC
where the voice port is located. Valid entries are 0 or 1.
The port argument identifies the voice port number.
Valid entries are 0 and 1.
or
The slot argument is the slot in which the voice port
adapter is installed. Valid entries are from 0 to 3.
The port argument is the voice interface card location.
Valid entries are 0 to 3.
The ds0-group-number argument indicates the defined
DS0 group number. Each defined DS0 group number is
represented on a separate voice port. This allows you to
define individual DS0s on the digital T1/E1 card.
Note The commands, keywords, and arguments that you
are able to use may differ slightly from those
presented here, based on your platform, Cisco IOS
release, and configuration. When in doubt, use
Cisco IOS command help to determine the syntax
choices that are available.
Step 6 echo-cancel enable type [hardware | software]
Example:
router(config-voiceport)# echo-cancel enable
type hardware
Enables hardware echo cancellation.
The hardware keyword is the default. Echo cancel
coverage is hardcoded for 128 ms.
This command is needed only to configure the software
keyword to effect software-based (DSP) echo
cancellation or to restore the hardware default.
Note The hardware and software keywords are available
only when the optional hardware echo cancellation
module (EC-MFT-32 or EC-MFT-64) is installed on
the multiflex VWIC.

114
Examples
This section provides the following examples for verifying echo cancellation:
show echo-cancel hardware status: Example, page 114
show call active voice echo-canceller summary: Example, page 114
show call active voice echo-canceller CallID: Example, page 115
show echo-cancel hardware status: Example
The output in this section shows that hardware echo cancellation is enabled on slot 1.
Router_3725# show echo-cancel hardware status 1

VWIC HWECAN 1/0 is UP.
Software version:4.4.803 , Date:Feb 6 16:58:57 2004
Tail length:128 Tone disabler type:G.165 Fax notify: Off
Device:VWIC_8MBPS_1TIEC_TL128_MS_1P Max Channels:32
Only Port0 have Local HWECAN Connectivity.
Total assigned channel(s):1
Total device(s) in the slot 1
show call active voice echo-canceller summary: Example
The output in this section shows summary information for the hardware echo cancellation.
Router_3725# show call active voice echo-canceller summary
96 | 112 | 128}
Example:
Router (config-voiceport) # echo-cancel
coverage 96
Adjusts the echo canceller by the specified number of
milliseconds.
These coverage options are applicable only if you
configured the echo-cancel enable type software
command in the previous step.
If you configured the echo-cancel enable type
hardware command in the previous step, this value is
set to 128 ms.
Beginning with Release 12.4(20)T, the default for
software echo cancellation is 128 ms. Prior to
Release 12.4(20)T, the default is 64 ms.
Step 8 exit
Example:
Exits controller cofiguration mode and returns the router to
privileged EXEC mode.
ECAN CH ASSIGNED DSP ID VOICEPORT EC NLP COV LAW
========================================================================================
1 yes 1/1 1/0:1.1 on off on u-Law

115
1 active call found
number of hardware ecan channels:1
number of software ecan channels:0
show call active voice echo-canceller CallID: Example
The output in this section shows hardware echo canceller information for an active voice call.
Router# show call active voice echo-canceller E71

Device:VWIC HWECAN 1/0 Channel Id = 1 Tail = 128Ms
Software version:4.4.803 , Date:Feb 6 16:58:57 2004
Echo Canceller:On Tail-length:128ms
H-Register:Update Modem tone disable:Ignore 2100Hz tone
Worst ERL :6dB Residual Control:Cancel only
High level compensation:Off
Tx Power = 0.0dB Tx Avg Power = 0.0dB
Rx Power = 0.0dB Rx Avg Power = 0.0dB
ERL = 1.0dB ACOM = 0.0
3 Reflectors(Tails) = (90, 0, 0)Ms, Max Reflector = 90Ms
Ecan Status words 0x1C, 0x00
EC Lib version:9155
More detailed syntax information about the commands used in this chapter is documented in the Cisco
IOS Voice Command Reference.
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trademarks; Changing the Way We Work, Live, Play, and Learn is a service mark; and Access Registrar, Aironet, AsyncOS, Bringing the Meeting To
You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS,
Event Center, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQ Expertise, the iQ logo,
iQ Net Readiness Scorecard, iQuick Study, IronPort, the IronPort logo, LightStream, Linksys, MediaTone, MeetingPlace, MGX, Networkers,
Networking Academy, Network Registrar, PCNow, PIX, PowerPanels, ProConnect, ScriptShare, SenderBase, SMARTnet, Spectrum Expert,
StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and the WebEx logo are registered trademarks of
Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
coincidental.
Call ID Port DSP/Ch Codec Ecan-type Tail Called # Dial-peers
========================================================================================
0xE71 1/0:1.1 1/1 g729r8 HW 128ms 1000 1/10

116

NextPort-Based Voice Tuning and Echo
Cancellation
This chapter describes how to dynamically configure voice services on the NextPort-based platforms:
Cisco AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850. This chapter contains the
following sections:
Prerequisites for Nextport-Based Voice Tuning and NextPort Dual-Filter G.168 Echo Canceller,
page 117
Information About NextPort Voice Services, page 117
How to Configure Nextport-Based Voice Tuning and the NextPort Dual-Filter G.168 Echo
Canceller, page 120
Prerequisites for Nextport-Based Voice Tuning and NextPort
To use the Nextport-Based Voice Tuning and Background Noise Statistics feature, you must be running
NextPort service processing element (SPE) firmware version 8.8.1 or a later version and Cisco IOS
Release 12.3(4)T or a later release.
To use the NextPort dual-filter G.168 echo canceller, you must be running SPE firmware version 10.2.2
or a later version and Cisco IOS Release 12.3(11)T or a later release.
Information About NextPort Voice Services
To configure the features presented in this chapter, you should understand the following concepts:
NextPort Dual-Filter G.168 Echo Canceller, page 118
NextPort SPE Firmware, page 119
Voice Tuning, page 119
Background Noise, page 120

118
NextPort Dual-Filter G.168 Echo Canceller
The CSMV6 dial feature card (DFC) for NextPort platforms offers dual-filter G.168 echo canceller
capability. The NextPort dual-filter G.168 echo canceller (EC) improves voice quality in VoIP
connections by providing relatively less residual echo leakage, better nonlinear processing (NLP)
timing, less clipping, and better comfort noise generation (CNG) in most environments.
The dual-filter G.168 echo canceller features two concurrently operating adaptive filters (which control
echo tail coverage) and two double-talk detection functions. In addition, the comfort noise model uses
Hoth noise spectrum shaping to better replicate the true noise spectrum.
Note Detailed information for all Cisco IOS commands mentioned in this section can be found in the
The NextPort dual-filter G.168 echo canceller uses the same voice-tuning (VC tune) interface for
configuring voicecap parameters as the Cisco-proprietary G.164 echo canceller. To adjust the dual-filter
echo canceller, use a voicecap or the Cisco IOS command-line interface (CLI) during configuration. You
can also adjust settings while the system is running by using the show port log and show port
operational-status commands. However because of the differences in internal operation of these ECs,
there are some changes in the set of available parameters for voice tuning.
See the echo-cancel coverage command for updated Cisco IOS command usage with this feature. The
NextPort dual-filter G.168 echo canceller adds the following benefits on NextPort platforms:
Configurable parametersRange checking that is performed on the voicecap parameters in the I960
NextPort layer has been updated. (Voicecap parameters in raw mode are never range-checked.)
Up to 128 ms of echo tail coverageBeginning with Cisco IOS Release 12.4(20)T, the NextPort
dual-filter G.168 echo canceller supports echo tails from 24-ms to 128-ms in 16-ms increments. The
echo-cancel coverage command limits the echo canceller coverage to 128-ms on NextPort
platforms. For backward compatibility, a voicecap used in "raw mode" will still configure older
SPEware to settings greater than 64-ms when used with newer releases of Cisco IOS software. For
situations when new SPEware is loaded onto an older Cisco IOS release, the NextPort dual-filter
G.168 echo canceller automatically sets coverage time to 64 ms.
Updated set of reported statisticsText in the show voice port command output has been changed
to describe voicecap parameters and reported statistics. The show port operational-status
command output has been updated to report TX/RX mean speech level statistics.
Power statistics (RX and TX)These statistics average only the power that is received during signal
periods that are classified as speech.
Unchanged configuration stepsUse voicecaps and the echo-cancel coverage command to
configure this feature. See the Voicecap Strings section on page 119.
SPE firmware and Cisco IOS software packaging supportThe SPEware that contains the
dual-filter G.168 echo canceller is field-upgradeable and can be used interchangeably with previous
firmware versions with no effect on platform call density. The new SPEware interoperates with any
Cisco IOS software release that supports voicecaps.
Note When older Cisco IOS software releases are used, voicecaps must be used in raw mode for some
parameters. Some statistics may not be displayed or recorded properly with older software
releases.

119
NextPort SPE Firmware
NextPort SPE firmware is software that drives the digital signal processor (DSP) portion of the NextPort
dial feature cards (DFCs). NextPort firmware is bundled with Cisco IOS software.
NextPort SPE firmware runs on the NextPort DFC60, DFC108, 1 CT3_UPC 216, and UPC324 DFCs on
the Cisco AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850 platforms. The ports on these
modules can support modem, voice, fax, and digital services and can be aggregated at any of the
following levels:
Slot level of the NextPort module
SPE level within the NextPort module
Individual port level
Note To use the NextPort Voice Tuning and Background Noise Statistics feature, you must use the default
bundled NextPort SPE firmware code that runs with Cisco IOS software. The NextPort-Based Voice
Tuning and Background Noise feature uses SPE firmware version 8.8.1 or a later version. The NextPort
dual-filter G.168 echo canceller uses NextPort firmware version 10.2.2, which is bundled with
Cisco IOS Release 12.3(11)T. NextPort firmware version 10.2.2 can be used with Cisco IOS
Release 12.3(7)T, 12.3(10), and later releases.
For more information about NextPort SPE firmware, see the NextPort SPE Release Notes on Cisco.com.
Voicecap Strings
Additional configuration of voice services on NextPort DFCs is achieved by configuring the voice tuning
configuration capability (called voicecaps) using voicecap strings. Voicecap strings are created with the
the voicecap entry command and are applied with the voicecap configure command.
Voice Tuning
Note You must have specific knowledge of the behavior of the telephone network in order to use these
voicecap capabilities. See the Nextport-Based Voice Tuning White Paper on Cisco.com for a brief
description of telephone network behavior.
This feature allows the following parameters, among others, to be configured:
PSTN gainsPSTN gains adjust the power levels at the PSTN side of a VoIP connection to make
up for loss plan imbalances and to ensure minimum echo return losses (ERLs) in a call. PSTN gain
is configured with the CLI rather than with voicecaps.
IP gainsIP gains adjust IP-side levels and are applied to the signal before it is propagated through
the echo canceller. This point is also known as the reference signal.
Dynamic attenuationDynamic attenuation mitigates low volume calls when attenuation has been
added on the PSTN call leg to compensate for low ERL calls.
Comfort noise generationCNG enables and disables EC comfort noise.
Minimum ERLMinimum ERL switches off near-end talker clipping and poor echo canceller
performance.

How to Configure Nextport-Based Voice Tuning and the NextPort Dual-Filter G.168 Echo Canceller
120
Background Noise
The NextPort Voice Tuning and Background Noise Statistics feature reports EC background noise level,
voice activity detection (VAD) background noise level, ERL level, and Acombined (ACOM) statistics
by averaging the combined values that are computed over the duration of the call. These statistics are
appended to the end of each entry in the voice log, which you can see in the output from the show port
log and show port operational-status commands.
How to Configure Nextport-Based Voice Tuning and the
NextPort Dual-Filter G.168 Echo Canceller
To configure the Nextport-Based Voice Tuning and the NextPort Dual-Filter G.168 Echo Canceller
feature, complete the following tasks:
Downloading NextPort SPE Firmware, page 101 (required)
Creating and Applying Voicecaps, page 103 (required)
Setting Voice Tuning Parameters with V Registers, page 105 (optional)
Downloading NextPort SPE Firmware
To download NextPort SPE firmware, use the following commands:
SUMMARY STEPS
1. enable
3. spe {first slot | first slot/spe} {last slot | last slot/spe}
4. firmware location [IFS:[/]]filename
5. end
6. copy running-config startup-config
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:

121
Step 3 spe {first slot | first slot/spe} {last slot |
last slot/spe}
Example:
Router(config)# spe 1 1/17
Enters SPE configuration mode and sets the range of SPEs.
first slot and last slotIdentifies slots for the range.
For the Cisco AS5350, slot values range from 1 to 3.
For the Cisco AS5400, slot values range from 1 to 7.
All ports on the specified slot are affected.
first slot/spe and last slot/speIdentifies slots for the
range. For the Cisco AS5350, slot values range from 1
to 3. For the Cisco AS5400, slot values range from 1 to
7. SPE values range from 1 to 17. You must include the
slash mark. All ports on the specified slot and SPE are
affected.
Step 4 firmware location [IFS:[/]]filename
Example:
Router(config-spe)# firmware location
flash:np.8.8.1.spe
Downloads SPE modem code to all modems in a particular
slot (that is, all modems on a feature card that contains 18
6-port modem modules).
IFS(Optional) Cisco IOS file specification (IFS),
which can be any valid IFS on any local file system.
Examples of legal specifications include:
bootflash:Loads the firmware from a separate
flash memory device.
flash:Loads the firmware from the flash
NVRAM located within the router.
null:Specifies a firmware file from null: File
System.
system:/Loads the firmware from a built-in file
within the Cisco IOS image. The optional forward
slash (/) and system path must be entered with this
specification.
filenameThe firmware filename. When the filename
is entered without an IFS specification, this name
defaults to the file in flash memory.
Use the dir all-filesystems EXEC command to display
legal IFSs.
The no form of the command reverts the router back to
the system-embedded default. When the access server
is booted, the firmware location command displays the
location for the firmware that is embedded in the
Cisco IOS image. If the firmware location command
is issued to download a firmware image from flash and
then the no version of the exact command is
subsequently issued, then the firmware location
command downloads the embedded firmware in
Cisco IOS software.

122
Creating and Applying Voicecaps
Voicecaps can be used to configure any of the voice service parameters. Voicecaps are, however,
primarily used to configure only those parameters that do not have associated Cisco IOS commands.
Restrictions
Voicecaps are configured in global configuration mode. A maximum of five voicecap entries can be
defined.
Applying a voicecap is possible only in voice-port configuration mode. Once applied to a voice port,
the voicecap affects all calls associated with that voice port.
To achieve the specified functionality, an SPE image capable of voice tuning must be used in
conjunction with the Cisco IOS software and module controller software.
For backward compatibility, a voicecap used in raw mode will configure older SPEware to allow
echo canceller coverage settings greater than 64 ms when the older SPEware is used with newer
releases of Cisco IOS software. For situations when new SPEware is loaded onto an older Cisco IOS
software release, the NextPort dual-filter G.168 echo canceller automatically sets coverage time to
64 ms.
Note Voicecap parameters in raw mode are never range-checked.
For a list of available voicecap parameters and code words that are used with the NextPort dual-filter
G.168 echo canceller feature, see the NextPort Dual-Filter G.168 Echo Canceller White Paper.
Step 5 end
Example:
Router(config-spe)# end
Completes the download and exits SPE configuration mode.
Step 6 copy running-config startup-config
Example:
Router# copy running-config startup-config
Copies the configuration from running RAM into NVRAM.
The download occurs when the modems become
available and the display shows the SPE firmware
upgrade option defined (default: busyout).
The spe command generates NVRAM modem
download and configuration file entries.
Note If the configuration is not saved as described in this
step, download of the firmware specified with the
spe command will not occur after the next reboot.
For detailed information on the spe command, see the
following Cisco document:
SPE and Firmware Download Enhancements.

123
Setting Voice Tuning Parameters with V Registers
The following sections contain information about voice tuning parameters with V registers:
Note All data specified in dB is entered in the form (dB * 10). So, for example, to specify 6.0 dB, 60 must be
entered.
Set PSTN Gains, page 123
Set IP Gains, page 123
Set Dynamic Attenuation, page 123
Set Comfort Noise Generation, page 124
Set Minimum ERL, page 124
Set PSTN Gains
Note Too much attenuation may cause some calls to have low volume speech. For more information, see the
dynamic attenuation feature described in the Nextport-Based Voice Tuning White Paper.
To fix poor loss plans and to ensure minimum ERLs, use PSTN gains to adjust the power levels on the
PSTN call leg. To adjust these levels, make sure that the sum of the output attenuation, the input
attenuation, and the lowest expected functional ERL is greater than the MinERL parameter setting
described below. You should balance the power levels of the far-end talker and near-end talker as seen
at the echo canceller.
Note Use the Cisco IOS CLI, not voicecap indexes, to set PSTN gains.
Set IP Gains
To adjust IP-side levels that are applied to the signal before it is propagated through the echo canceller,
use IP gains. IP gains are controlled with the following V registers. The valid range for both input and
output gain is 14 dB to 14 dB.
v261IP output gain.
v263IP input gain.
Note There have been some instances where the IP-side power has been too high. Using index v263 can
mitigate this problem.
Set Dynamic Attenuation
To mitigate low volume calls when attenuation has been added on the PSTN call leg to compensate for
low ERL calls, use dynamic attenuation. Dynamic attenuation is controlled with the following
V registers:
v289Dynamic EC Attenuation Feature Enable. Set to 1 to enable. Set to 0 to disable.

124
v290Dynamic EC Attenuation Minimum ERL Value. Valid range is from 0 dB to 60 dB.
v291Dynamic EC Attenuation Final Rout Gain. Set to the lowest level desired for PSTN output
attenuation. This value is usually set to 0. Valid range is from 14 dB to 6 dB.
v292Dynamic EC Attenuation Final Sin Gain. Set to the lowest level desired for PSTN input
attenuation. This value is usually set to 0. Valid range is from 14 dB to 6 dB.
Set Comfort Noise Generation
During periods of far-end single talk, the echo canceller engages the non-linear processor (NLP) to
suppress residual echo. However, it will also suppress any noise signal that is coming from the near-end
side. This can lead to dead silence, which the listener may confuse with a dropped call. To overcome this
condition, comfort noise generation (CNG) is added.
To choose between NLP silence or background noise reproduction, use comfort noise generation. CNG
is controlled with the following V register:
v294CNG Enable. Set to 1 to enable. Set to 0 to disable.
Set Minimum ERL
The echo canceller uses the minimum ERL (MinERL) value to decide whether the incoming signal on
the PSTN call leg is an echo or a near-end talker. If this value is too high, the echo canceller will not
properly identify echo and will not adapt. If this value is too low, clipping of the near-end talker may
occur.
To reduce near-end talker clipping and poor echo canceller performance, use minimum ERL. MinERL
is controlled with the following V register:
v270Sets the level that the echo canceller expects the lowest ERL of the PSTN to be. The valid range
is from 0 dB to 20 dB. The default is 6.
Creating and Applying Voice Caps
To create and apply voice caps and voice cap entries, complete the following tasks:
SUMMARY STEPS
1. enable
3. voicecap entry name string
4. voice-port slot/port:D
5. voicecap configure name
6. exit

125
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:
Step 3 voicecap entry name string
Example:
Router(config)# voicecap entry qualityERL
v270=120
Creates a voicecap.
The name argument is a word that uniquely identifies
this voicecap.
The string argument is a string of characters composed
of a series of voicecap register entries, similar to a
modemcap. Each entry is of the form
vINDEX=VALUE, where INDEX refers to a specific
V register, and VALUE designates the value to set that
V register to.
For more information about V_register entries, see the
Setting Voice Tuning Parameters with V Registers
section on page 123.
The example creates a simple voicecap string named
qualityERL with V register 270 set to 120.
Step 4 voice-port slot/port:D
Example:
Router(config)# voice-port 3/0:D
Enters voice-port configuration mode on the selected slot
and port.
Step 5 voicecap configure name
Example:
Router(config-voiceport)# voicecap configure
qualityERL
Applies a voicecap.
The name argument designates which of the newly
created voicecaps to use on this voice port. This
character value must be identical to the value entered
when you created the voicecap entry. In this case, the
value is qualityERL.
Note To configure multiple voice ports, repeat Step 4 and
Step 5 for each voice port.
Step 6 exit
Example:
configuration.

126
Verifying Voicecap Configurations
Use the following show commands in privileged EXEC mode to verify your configuration. Relevant
fields are shown in bold.
SUMMARY STEPS
1. show voice port
2. show port operational-status slot/port
3. show port voice log
DETAILED STEPS
Step 1 show voice port
Use this command to display configured voicecaps, for example:
Router# show voice port
ISDN 2/0:D - 2/0:D
Type of VoicePort is ISDN
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
.
.
.
Station name None, Station number None
Translation profile (Incoming):
Translation profile (Outgoing):
Voicecap:EXAMPLE
Step 2 show port operational-status slot/port
Use this command to display background noise level information on current calls. Significant fields are
shown in bold in the following example:
Router# show port operational-status 1/0
Slot/SPE/Port -- 1/0
Service Type :Voice service
Voice Codec :G.711 u-law
Echo Canceler Length :8 ms
Echo Cancellation Control :Echo cancellation - enabled
Echo update - enabled
Non-linear processor - enabled
Echo reset coefficients - disabled
High pass filter enable - disabled
Digit detection enable :DTMF signaling - enabled
Voice activity detection :Disabled
Comfort noise generation :Generate comfort noise
Digit relay enable :OOB Digit relay - disabled
IB Digit relay - disabled
Information field size :20 ms
Playout de-jitter mode :adaptive
Encapsulation protocol :RTP
Input Gain :0.0 dB
Output Gain :0.0 dB
Tx/Rx SSRC :20/0

127
Current playout delay :65 ms
Min/Max playout delay :65/105 ms
Clock offset :142003 ms
Predictive concealment :0 ms
Interpolative concealment :0 ms
Silence concealment :0 ms
Buffer overflow discards :1
End-point detection errors :0
Tx/Rx Voice packets :1337/1341
Tx/Rx signaling packets :0/0
Tx/Rx comfort noise packets :0/0
Tx/Rx duration :26745/26745 ms
Tx/Rx voice duration :0/0 ms
Out of sequence packets :0
Bad protocol headers :0
Num. of late packets :0
Num. of early packets :1
Tx/Rx Power :-87.0/-57.3 dBm
Tx/Rx Talker Level :-86.3/-57.0 dBm
TX/RX Mean Speech level :-86.3/-57.0 dBm
VAD Background noise level :6.2 dBm
ERL level :127.0 dB
ACOM level :127.0 dB
Tx/Rx current activity :silence/silence
Tx/Rx byte count :213920/214240
ECAN Background noise level :-83.4 dBm
Latest SSRC value :391643394
Number of SSRC changes :1
Number of payload violations :0
Step 3 show port voice log
Use this command to display background noise level information on completed calls. Significant fields
are shown in bold in the following example:
Router# show port voice log
Port 1/00 Events Log
*Aug 22 07:59:27.515:Voice Terminate event:
Disconnect Reason : normal call clearing (16)
Call Timer : 57 secs
Current playout delay : 65 ms
Min/Max playout delay : 65/105 ms
Clock offset : 142003 ms
Predictive concealment : 0 ms
Interpolative concealment : 0 ms
Silence concealment : 0 ms
Buffer overflow discards : 1
End-point detection errors : 0
Tx/Rx Voice packets : 2813/2816
Tx/Rx signaling packets : 0/0
Tx/Rx comfort noise packets : 0/0
Tx/Rx duration : 56260/56260 ms
Tx/Rx voice duration : 0/0 ms
Out of sequence packets : 0
Bad protocol headers : 0
Num. of late packets : 0
Num. of early packets : 1
Tx/Rx Power : -87.0/-57.3 dBm
Tx/Rx Mean Speech Level : -86.7/-57.0 dBm
Tx/Rx Talker Level : -86.3/-57.0 dBm
Average VAD Background noise level : 6.2 dBm
Average ERL level : 127.0 dB
Average ACOM level : 127.0 dB

128
Tx/Rx current activity : silence/silence
Tx/Rx byte count : 450080/450240
Average ECAN Background noise level: -83.4 dBm
*Aug 22 07:59:27.515:Voice SSRC change events:
Latest ssrc value : 391643394
Total ssrc changes : 1
Troubleshooting NextPort Voicecaps
Use the following debug and show commands in privileged EXEC mode to debug the application of a
voicecap and to check debugging output:
SUMMARY STEPS
1. debug nextport vsmgr detail
2. debug dspapi detail
3. show debug
DETAILED STEPS
Step 1 debug nextport vsmgr detail
Use this command to turn on debugging for NextPort voice services, for example:
Router# debug nextport vsmgr detail
NextPort Voice Service Manager:
NP Voice Service Manager Detail debugging is on
.
.
.
Step 2 debug dspapi detail
Use this command to turn on debugging for DSP API message event details, for example:
Router# debug dspapi detail
DSP API:
DSP API Command debugging is on
DSP API Detail debugging is on
.
.
.
Step 3 show debug
Use this command to check voicecap application debugging. The significant field in the output is
highlighted in bold in the following example:
Router# show debug
NextPort Voice Service Manager:
NP Voice Service Manager Detail debugging is on
DSP API:
DSP API Command debugging is on

129
DSP API Detail debugging is on
*Aug 22 08:34:47.399:dspapi [2/1:1 (4)] dsp_init
*Aug 22 08:34:47.399:dspapi [2/1:1 (4)] dsp_voice_config_params:10 params
*Aug 22 08:34:47.399: [0] ENCAP RTP:t_ssrc=20 r_ssrc=0 t_vpxcc=0 r_vpxcc=0
ifp_payload_type=122 sid_support=19 tse_payload=101
seq_num_start=6303 redundancy=0 cc_payload_typ
Router# e=125 fax_payload_type=122 alaw_pcm_switchover=8 mulaw_pcm_switchover=0
dtmf_payload_type=121, nte_rcv_payload_type=101dynamic_payload=0, codec=5
*Aug 22 08:34:47.399: [1] PO_JITTER:mode=2 initial=60 max=200 min=40 fax_nom=300
*Aug 22 08:34:47.399: [2] INBAND_SIG:mode=0x1 enable
*Aug 22 08:34:47.399: [3] ECHO_CANCEL:flags=0x17 echo_len=256
*Aug 22 08:34:47.399: [4] IDLE_CODE_DET:enable = 0 code=0x0 duration=6000
*Aug 22 08:34:47.403: [5] GAIN:input=0 output=0
*Aug 22 08:34:47.403: [6] CNG:
Router# 1
*Aug 22 08:34:47.403: [7] INFO_FIELD_SIZE:160
*Aug 22 08:34:47.403: [8] DIGIT_RELAY:2
*Aug 22 08:34:47.403: [9] VOICECAP:EXAMPLE
*Aug 22 08:34:47.403:dspapi [2/1:1 (4)] dsp_start_service:G711_U (5)
*Aug 22 08:34:47.403:Matched voicecap:v0=0 v1=1
*Aug 22 08:34:47.403:msg length = 0x001D
*Aug 22 08:34:47.403:session ID = 0x006D
*Aug 22 08:34:47.403: msg tag = 0x0000
*Aug 22 08:34:47.403: msg ID = 0xF201
Configuring the NextPort Dual-Filter G.168 Echo Canceller
The NextPort dual-filter G.168 echo canceller is enabled by default on NextPort platforms in Cisco IOS
Release 12.3(11)T and later releases. However, you can adjust the echo canceller tail coverage time at
the voice-port interface by completing the following tasks:
SUMMARY STEPS
1. enable
3. voice-port slot/port:D
4. echo-cancel coverage {24 | 32 | 48 | 64 | 80 | 96 | 112 | 128}
5. exit
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:

130
Configuration Examples for Nextport-Based Voice Tuning and the NextPort
The following sections contain examples that could be used to optimize a NextPort-based gateway for a
given telephone network:
High ERL in the Network: Example, page 130
Low ERL in the Network: Example, page 131
Clipped or Squelched Speech and Low ERL in the Network: Example, page 131
Dynamic Attenuation: Example, page 131
Echo Canceller Tail Coverage: Example, page 132
High ERL in the Network: Example
Register v270 is used to set the limit for the minimum expected ERLs that the gateway will encounter.
If the gateway encounters ERLs that are lower than the v270 setting, the echo canceller performance will
be suboptimal.
The default setting for v270 is 6 dB. This setting should work well for usual telephone networks.
However, when the gateway is used on a well-managed telephone network with organized loss plans in
place, the ERL is often greater than 6 dB. In these cases, v270 can be raised. Making this change reduces
any clipping of the near-end signal; however, it will underperform if a low ERL is encountered.
In this example, the network is designed to have an ERL of 12 dB or greater. In this case, the following
voicecap may improve performance. Notice that the value for v270 is entered as decibels multiplied by
10.
Router> enable
Router(config)# voicecap entry qualityERL v270=120
Router(config)# end
Step 3 voice-port slot/port:D
Example:
and port.
96 | 112 | 128}
Example:
Router(config-voiceport)# echo-cancel coverage
64
Adjusts the size of the echo canceller (EC) and selects the
extended EC when the Cisco default EC is present.
Starting with Cisco IOS Release 12.4(20)T, the default
coverage time and maximum possible coverage is
128 ms for both 8.x and 10.2.2 SPEware versions.
Step 5 exit
Example:
configuration.

131
Low ERL in the Network: Example
Unlike the scenario described in the previous example, some telephone networks may not always
produce sufficient ERLs to meet the default setting of 6 dB. The best way to solve this problem would
be to institute better loss plans on the telephone network. However, because this is not always possible,
a voicecap can be used to alleviate the problem.
A low ERL means that the echo canceller must do a much deeper cancellation to remove sufficient echo.
Also, lowering the minimum ERL can increase the incidence of clipping. The best way to improve this
situation is to make up for the telephone networks lack of loss plan by adding some loss in the gateway.
If the lowest ERL seen is 4 dB, adding 1 dB of output attenuation and 1 dB of input gain will ensure
that the echo canceller never sees more than a 6-dB effective ERL.
Adding this attenuation can be done by entering a voicecap, but using the CLI is the recommended
approach in this example. The following commands set the gains that are needed for this example:
Router> enable
Router(config-voiceport)# output attenuation 1
Router(config-voiceport)# input gain -1
Clipped or Squelched Speech and Low ERL in the Network: Example
In this example, a network in which signal level imbalance is already causing clipping to occur with a
MinERL setting of 6 dB and in which ERLs of less than 6 dB are already occurring, a dual approach
must be taken. To stop the clipping, the MinERL setting should be lowered to 12 dB. If 4-dB ERLs are
occurring, 4 dB of attenuation must be added to the input and the output to ensure that there is 12 dB of
effective ERL at the echo canceller (4 dB of real ERL, plus 4 dB of output attenuation, plus 4 dB of input
attenuation equals 12 dB of effective ERL). To create these settings, the following commands are used:
Router> enable
Router(config)# voicecap entry qualityERL v270=120
Router(config-voiceport)# voicecap configure qualityERL
Dynamic Attenuation: Example
It is possible that worst-case settings are only required when the primary telephone circuits are all used
and an alternate carrier with a poor loss plan must be used instead. For these cases, the dynamic
attenuation feature removes the attenuation when the ERL is sufficient. In the following example, 4 dB
of input and output attenuation is to be removed when it is not necessary to ensure the minimum ERL
setting. To do this, the dynamic attenuation feature (using v289) is enabled. The required ERL must be
set before attenuation is removed (using v290), and minimum attenuation levels for the input and output
(using v291 and v292) must also be set. In this example, attenuation is set to 15 dB and then removed.
Router> enable
Router(config)# voicecap entry dynatten v270=120 v289=1 v290=15 v291=0 v292=0
Router(config-voiceport)# voicecap configure dynatten

132
Echo Canceller Tail Coverage: Example
The following example adjusts echo canceller tail coverage to 64 ms on the Cisco AS5400:
Router(config) voice-port 1/0:0
Router(config-voiceport)# echo-cancel coverage 64
Enabling NextPort Echo Canceller Control for G.711 Encoded VoIP Packets
This section describes how to enable NextPort echo canceller control on the Cisco AS5350, AS5400,
AS5400HPX, and AS5850 universal gateways when these gateways detect 2100 Hz tones, received in
G.711 encoded VoIP packets. You can enable NextPort voicecaps to control the echo canceller from
either the PSTN or IP side of the network.
Note NextPort control over the echo canceller is possible only in G.711 codec modes. Cisco recommends that
you do not enable NextPort control over the echo canceller in conjunction with modem pass-through.
IP tone detection and NextPort control over the echo canceller is enabled using the command-line
interface. Use the following commands to enable NextPort control over the echo canceller by creating a
voicecap entry and applying it to the voice port.
SUMMARY STEPS
1. enable
3. voicecap entry name string
5. voicecap configure name
6. exit
DETAILED STEPS
Step 1 enable
Example:
Router> enable
Example:

133
Troubleshooting NextPort Echo Canceller Control for G.711 Encoded VoIP Packets
You can display the EST trace messages that show the tone detections and the resultant echo operations
if you enter the debug trace module f080 0010 s/d/m command. NextPort enables and disables the NLP
and the echo canceller based on reception of 2100 Hz answer tones from the IP side or PSTN side and
generates EST trace messages for each tone detected and its echo operation. NextPort also detects the
250 ms of silence and generates EST trace messages to indicate such detection and to indicate that the
echo state has been restored.
Step 1 Router# debug trace module f080 0010 s/d/m
Step 3 voicecap entry name string
Example:
Router(config)# voicecap entry npecho_ctrl
v2=512 v51=32769
Creates a voicecap entry.
The name argument is a word that uniquely identifies
this voicecap.
The string argument is a series of voicecap register
entries, similar to a modemcap. Each entry is of the
form vINDEX=VALUE, where INDEX refers to a
specific V register, and VALUE designates the value to
which the V register should be set.
Example settings:
The v2=512 setting enables the 250-ms silence
detection. This setting is optional. When this
setting is used in conjunction with the v51 = 32769
setting, NextPort restores the echo canceller to its
original state after it detects the 250-ms silence.
The v51=32769 setting enables IP side tone
detection/notification and allows NextPort to
disable the NLP or the echo canceller upon
reception of 2100 Hz answer tones from the IP
side. This setting is required in Cisco IOS
Release 12.3T.
Example:
and port.
Step 5 voicecap configure name
Example:
Router(config-voiceport)# voicecap configure
npecho_ctrl
Applies a voicecap entry to the voice port.
The name argument designates which of the newly
created voicecaps to use on this voice port. This
character value must be identical to the value entered
when you created the voicecap entry.
Note To configure multiple voice ports, repeat Step 4 and
Step 5 for each voice port.
Step 6 exit
Example:

134
Use this command to display the EST trace messages. In the command, s/d/m is defined as follows:
s = slot
d = dfc
m = module number
When the default configuration values for Index 51 and Index 52 are used, IP tone detection and
notification are disabled, and all existing features continue to function normally.
The following example shows EST trace messages collected from the console:
Router#
*Apr 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:51.735: Data Len : 56
*Apr 26 21:40:51.735: Data : Session 0x0144 Received Early ANS tone 0x01 from
IP side
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735:
Router# Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Len : 63
*Apr 26 21:40:51.735: Data : Session 0x0144 Received Tone Off ntf for code 0x01
from IP side
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
Router#*Apr 26 21:40:51.735: Data Len : 45
*Apr 26 21:40:51.735: Data : Session 0x0144 Received ANS tone 0x03 from IP
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Len : 47
*Apr 26 21:40:51.735: Data : Session 0x0144 Non-linear Processor Is Disabled
*Apr
Router# 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Len : 63
from IP side
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735:
Router# Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Len : 47
*Apr 26 21:40:51.735: Data : Session 0x0144 Received ANSam tone 0x07 from IP
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000

135
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:5
Router#1.735: Data Len : 63
from IP side
*Apr 26 21:40:51.739: Port : 3/00
*Apr 26 21:40:51.739: Address : 0x3000000
*Apr 26 21:40:51.739: Trace Event: 0x2
*Apr 26 21:40:51.739: Data Len : 48
*Apr 26 21:40:51.739: Data : Session 0x0144 Received /ANSam tone 0x0f from IP
Router#*Apr 26 21:40:51.739: 00:00:13: Port Trace Event:
*Apr 26 21:40:51.739: Port : 3/00
*Apr 26 21:40:51.739: Address : 0x3000000
*Apr 26 21:40:51.739: Trace Event: 0x2
*Apr 26 21:40:51.739: Data Len : 31
*Apr 26 21:40:51.739: Data : Session 0x0144 ECAN Is Disabled
*Apr 26 21:40:51.739: Port : 3/00
*Apr 26 21:40:51.739: Address : 0x3000000
Router#*Apr 26 21:40:51.739: Trace Event: 0x2
*Apr 26 21:40:51.739: Data Len : 63
*Apr 26 21:40:51.739: Data : Session 0x0144 Received Tone Off ntf for code 0x0f
from IP side
*Apr 26 21:46:36.431: Port : 3/00
*Apr 26 21:46:36.431: Address : 0x3000000
*Apr 26 21:46:36.431: Trace Event: 0x2
*Apr 26 21:46:36.431: Data Len : 43
*Apr 26 21:46:36.431: Data : Session 0x0144 detected 250 msec of silence
*Apr 26 21:46:36.431: Port : 3/00
*Apr 26 21:46:36.431: Address : 0x3000000
*Apr 26 21:46:36.435: Trace Event: 0x2
*Apr 26 21:46:36.435: Data Len : 41
*Apr 26 21:46:36.435: Data : Session 0x0144 Ecan State 0x0007 Restored
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coincidental.

136

Verifying Analog and Digital Voice-Port
Configurations
This chapter describes some basic procedures and specific CLI commands you can use to verify the set
up and configuration of the analog and digital voice ports on the routers in your voice network.
Information About Verifying Analog and Digital Voice-Port
Configurations
After configuring the voice ports on your router, there are two simple verifications you can use to check
the operation of the telephony handset. There are also eight CLI commands you can use to verify proper
operation of the configuration.
How to Verify Analog and Digital Voice-Port Configurations
To verify the operability of the analog and digital voice-port configurations on your voice network,
complete the following tasks.
SUMMARY STEPS
1. Check for a dial tone.
2. Check for DTMF detection.
3. show voice port summary
4. show voice port
5. show running-config
6. show controller
7. show voice dsp
8. show voice call summary
9. show call active voice

138
10. show call history voice
DETAILED STEPS
Step 1 Check for a dial tone.
Pick up the handset of an attached telephony device and check for a dial tone.
Step 2 Check for Dual-Tone Multifrequency (DTMF) detection.
If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the
voice port is probably configured properly.
Step 3 show voice port summary
Use this command to identify the port numbers of voice interfaces installed in your router. For examples
of the output, see the Examples section on page 139.
Step 4 show voice port
Use this command to verify voice-port parameter settings. Refer to Table 2 for the appropriate syntax
for your platform. For sample output, see the show voice port Command: Examples section on
page 140.
Step 5 show running-config
Use this command to verify the codec complexity setting for digital T1/E1 connections. If medium
complexity is specified for the voice card, the codec complexity command is not displayed. If high
complexity is specified, the codec complexity high command is displayed. The following example
shows output when high complexity is specified:
Router# show running-config
.
.
.
hostname router-alpha

Table 2 Show Voice Port Command Syntax
Platform Voice Port Type Command Syntax
Cisco 1750 Analog show voice port [slot/port | summary]
Cisco 2600 series
Cisco 3600 series
Cisco 3700 series
Analog show voice port [slot/port | summary]
Digital show voice port [slot/port:ds0-group-number | summary]
Cisco MC3810 Analog show voice port [slot/port | summary]
Digital show voice port [slot:ds0-group-number | summary]
Cisco AS5300 Digital show voice port [controller:{ds0-group-number | D}]
[summary]
Cisco AS5350
Cisco AS5400
Cisco AS5850
Digital show voice port [slot/controller:{ds0-group-number | D}]
[summary]
Cisco AS5800 Digital show voice port {shelf/slot/port:ds0-group-number}
Cisco 7200 series Digital show voice port {slot/port-adapter:ds0-group-number}
Cisco 7500 series Digital show voice port {slot/port-adapter/slot:ds0-group-number}

139
voice-card 0
codec complexity high
.
.
.
Step 6 show controller
Use this command to verify that the digital T1/E1 controller is up and that no alarms have been reported,
and to display information about clock sources and other controller settings. For output examples, see
the show controller Command: Examples section on page 143.
Router# show controller {t1 | e1} controller-number
Use this command to display voice-channel configuration information for all DSP channels. For output
examples, see the show voice dsp Command: Examples section on page 144.
Step 8 show voice call summary
Use this command to verify the call status for all voice ports. For output examples, see the show voice
call summary Command: Examples section on page 145.
Router# show voice call summary
Step 9 show call active voice
Use this command to display the contents of the active call table, which shows all of the calls currently
connected through the router or concentrator. For output examples, see the show call active voice
Command: Example section on page 146.
Router# show call active voice
Step 10 show call history voice
Use this command to display the contents of the call history table. To limit the display to the most recent
calls connected through this router, use the last keyword and define the number of calls to display with
the number argument. To limit the display to a shortened version of the call history table, use the brief
keyword. For output examples, see the show call history voice Command: Example section on
page 147.
Router# show call history voice [last | number | brief]
Examples
This section contains output examples for the following commands on different platforms and for
different configurations:
show voice port summary Command: Examples
show voice port Command: Examples
show controller Command: Examples
show voice dsp Command: Examples
show voice call summary Command: Examples

140
show call active voice Command: Example
show call history voice Command: Example
show voice port summary Command: Examples
Cisco 3640 Router Analog Voice Port
The following output is from a Cisco 3640 router:
IN OUT
====== == ========== ===== ==== ======== ======== ==
2/0/0 -- e&m-wnk up dorm idle idle y
2/0/1 -- e&m-wnk up dorm idle idle y
2/1/0 -- fxs-ls up dorm on-hook idle y
2/1/1 -- fxs-ls up dorm on-hook idle y
Cisco MC3810 Digital Voice Port
The following output is from a Cisco MC3810:
IN OUT
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:23 24 e&m-imd up dorm idle idle y
1/1 -- fxs-ls up dorm on-hook idle y
1/2 -- fxs-ls up dorm on-hook idle y
1/3 -- e&m-imd up dorm idle idle y
1/4 -- e&m-imd up dorm idle idle y
1/5 -- fxo-ls up dorm idle on-hook y
1/6 -- fxo-ls up dorm idle on-hook y
show voice port Command: Examples
Cisco 3600 Series Router Analog E&M Voice Port
The following output is from a Cisco 3600 series router analog E&M voice port:
Router# show voice port 1/0
E&M Slot is 1, Sub-unit is 0, Port is 0
Type of VoicePort is E&M
Operation State is unknown
Administrative State is unknown
The Interface Down Failure Cause is 0
Alias is NULL
Noise Regeneration is disabled
Non Linear Processing is disabled
Music On Hold Threshold is Set to 0 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is disabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal

141
Connection Number is
Initial Time Out is set to 0 s
Interdigit Time Out is set to 0 s
Analog Info Follows:
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0

Voice card specific Info Follows:
Signal Type is wink-start
Operation Type is 2-wire
Impedance is set to 600r Ohm
E&M Type is unknown
Dial Type is dtmf
In Seizure is inactive
Out Seizure is inactive
Digit Duration Timing is set to 0 ms
InterDigit Duration Timing is set to 0 ms
Pulse Rate Timing is set to 0 pulses/second
InterDigit Pulse Duration Timing is set to 0 ms
Clear Wait Duration Timing is set to 0 ms
Wink Wait Duration Timing is set to 0 ms
Wink Duration Timing is set to 0 ms
Delay Start Timing is set to 0 ms
Delay Duration Timing is set to 0 ms
Cisco 3600 Series Router Analog FXS Voice Port
The following output is from a Cisco 3600 series router analog Foreign Exchange Service (FXS) voice
port:
Router# show voice port 1/2
Voice port 1/2 Slot is 1, Port is 2
Type of VoicePort is FXS
Operation State is UP
Noise Regeneration is enabled
Non Linear Processing is enabled
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Coder Type is g729ar8
Companding Type is u-law
Voice Activity Detection is disabled
Ringing Time Out is 180 s
Wait Release Time Out is 30 s
Nominal Playout Delay is 80 milliseconds
Maximum Playout Delay is 160 milliseconds
Analog Info Follows:
Region Tone is set for northamerica
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm

142
Analog interface A-D gain offset = -3 dB
Analog interface D-A gain offset = -3 dB
Signal Type is loopStart
Ring Frequency is 20 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is active
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Ring Cadence are [20 40] * 100 msec
InterDigit Pulse Duration Timing is set to 500 ms
Cisco 3600 Series Router Digital E&M Voice Port
The following output is from a Cisco 3600 series router digital E&M voice port:
Router# show voice port 1/0:1

receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
Music On Hold Threshold is Set to -38 dBm
Region Tone is set for US
Cisco AS5300 T1 CAS Voice Port
The following output is from a Cisco AS5300 T1 channel-associated signaling (CAS) voice port:
Router# show voice port

DS0 Group 1:0 - 1:0
Type of VoicePort is CAS
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms

143
Call-Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Companding Type is u-law
Wait Release Time Out is 30 s
Station name None, Station number None


DS0 channel specific status info:
IN OUT
PORT CH SIG-TYPE OPER STATUS STATUS TIP RING
Cisco 7200 Series Router Digital E&M Voice Port
The following output is from a Cisco 7200 series router digital E&M voice port:
Router# show voice port 1/0:1
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1 << voice-port 1/0:1
show controller Command: Examples
Cisco 3600 Series Router T1 Controller
The following output is from a Cisco 3600 series router with a T1 controller:
Router# show controller T1 1/1/0
T1 1/0/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
No alarms detected.
alarm-trigger is not set
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
Data in current interval (180 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations

144
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Cisco MC3810 E1 Controller
The following output is from a Cisco MC3810 with an E1 controller:
Router# show controller e1 1/0
E1 1/0 is up.
Applique type is Channelized E1
Cablelength is short 133
Description: E1 WIC card Alpha
No alarms detected.
Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.
Cisco AS5800 T1 Controller
The following output is from a Cisco AS5800 with a T1 controller:
Router# show controller t1 2
T1 2 is up.
No alarms detected.
Version info of slot 0: HW: 2, Firmware: 16, PLD Rev: 0

Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x42,
Board Hardware Version 1.0, Item Number 73-2217-4,
Board Revision A0, Serial Number 06467665,
PLD/ISP Version 0.0, Manufacture Date 14-Nov-1997.

Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
show voice dsp Command: Examples
Digital Voice Port on a Cisco 3640
The following output is from a Cisco 3640 router when a digital voice port is configured:
TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 010 00 g729r8 3.3 busy idle 0 0 1/015 1 0 67400/85384
01 g729r8 .8 busy idle 0 0 1/015 7 0 67566/83623
02 g729r8 busy idle 0 0 1/015 13 0 65675/81851
03 g729r8 busy idle 0 0 1/015 20 0 65530/83610
C549 011 00 g729r8 3.3 busy idle 0 0 1/015 2 0 66820/84799
01 g729r8 .8 busy idle 0 0 1/015 8 0 59028/66946
02 g729r8 busy idle 0 0 1/015 14 0 65591/81084
03 g729r8 busy idle 0 0 1/015 21 0 66336/82739
C549 012 00 g729r8 3.3 busy idle 0 0 1/015 3 0 59036/65245
01 g729r8 .8 busy idle 0 0 1/015 9 0 65826/81950
02 g729r8 busy idle 0 0 1/015 15 0 65606/80733
03 g729r8 busy idle 0 0 1/015 22 0 65577/83532

145
C549 013 00 g729r8 3.3 busy idle 0 0 1/015 4 0 67655/82974
01 g729r8 .8 busy idle 0 0 1/015 10 0 65647/82088
02 g729r8 busy idle 0 0 1/015 17 0 66366/80894
03 g729r8 busy idle 0 0 1/015 23 0 66339/82628
C549 014 00 g729r8 3.3 busy idle 0 0 1/015 5 0 68439/84677
01 g729r8 .8 busy idle 0 0 1/015 11 0 65664/81737
02 g729r8 busy idle 0 0 1/015 18 0 65607/81820
03 g729r8 busy idle 0 0 1/015 24 0 65589/83889
C549 015 00 g729r8 3.3 busy idle 0 0 1/015 6 0 66889/83331
01 g729r8 .8 busy idle 0 0 1/015 12 0 65690/81700
02 g729r8 busy idle 0 0 1/015 19 0 66422/82099
03 g729r8 busy idle 0 0 1/015 25 0 65566/83852
TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 007 00 {medium} 3.3 IDLE idle 0 0 1/0:1 4 0 0/0
.13
.13
.13
.13
.13
.13
C542 001 01 g711ulaw 3.3 IDLE idle 0 0 2/0/0 0 512/519
.13
C542 002 01 g711ulaw 3.3 IDLE idle 0 0 2/0/1 0 505/502
.13
C542 003 01 g711alaw 3.3 IDLE idle 0 0 2/1/0 0 28756/28966
.13
C542 004 01 g711ulaw 3.3 IDLE idle 0 0 2/1/1 0 834/838
.13
show voice call summary Command: Examples
Cisco MC3810 Analog Voice Port
The following output is from a Cisco MC3810:
PORT CODEC VAD VTSP STATE VPM STATE
========= ======== === ===================== ========================
1/1 g729r8 y S_CONNECT FXSLS_CONNECT
1/2 - - - FXSLS_ONHOOK
1/3 - - - EM_ONHOOK
1/4 - - - EM_ONHOOK
1/5 - - - FXOLS_ONHOOK
1/6 - - - FXOLS_ONHOOK
Cisco 3600 Series Router Digital Voice Port
The following output is from a Cisco 3600 series router:
PORT CODEC VAD VTSP STATE VPM STATE
========= ======== === ===================== ========================

146
1/015.1 g729r8 y S_CONNECT S_TSP_CONNECT
show call active voice Command: Example
Cisco 7200 Series Router
Router# show call active voice
GENERIC:
SetupTime=94523746 ms
Index=448
PeerAddress=##73072
PeerSubAddress=
PeerId=70000
PeerIfIndex=37
LogicalIfIndex=0
ConnectTime=94524043
DisconectTime=94546241
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=6251
TransmitBytes=125020
ReceivePackets=3300
ReceiveBytes=66000
VOIP:
ConnectionId[0x142E62FB 0x5C6705AF 0x0 0x385722B0]
RemoteIPAddress=172.16.235.18
RemoteUDPPort=16580
RoundTripDelay=29 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:172.16.235.18
OnTimeRvPlayout=63690
GapFillWithSilence=0 ms
GapFillWithPrediction=180 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=70 ms
LoWaterPlayoutDelay=30 ms
ReceiveDelay=40 ms

147
LostPackets=0 ms
EarlyPackets=1 ms
LatePackets=18 ms
VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
SignalingType=cas
show call history voice Command: Example
Cisco 7200 Series Router
Router# show call history voice
GENERIC:
SetupTime=94893250 ms
Index=450
PeerAddress=##52258
PeerSubAddress=
PeerId=50000
PeerIfIndex=35
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing.
ConnectTime=94893780
DisconectTime=95015500
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=32258
TransmitBytes=645160
ReceivePackets=20061
ReceiveBytes=401220
VOIP:
ConnectionId[0x142E62FB 0x5C6705B3 0x0 0x388F851C]
RemoteIPAddress=172.16.235.18
RemoteUDPPort=16552
RoundTripDelay=23 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:172.16.235.18
OnTimeRvPlayout=398000
GapFillWithSilence=0 ms
GapFillWithPrediction=1440 ms

148
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=97 ms
LoWaterPlayoutDelay=30 ms
ReceiveDelay=49 ms
LostPackets=1 ms
EarlyPackets=1 ms
LatePackets=132 ms
VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
coincidental.

Troubleshooting Analog and Digital Voice Port
Configurations
This chapter provides information to assist you in analyzing and troubleshooting voice port problems.
Troubleshooting Chart
Table 1 lists some problems that you might encounter after configuring voice ports. It also provides some
suggested remedies.
Table 1 Troubleshooting Voice Port Configurations
Problem Suggested Action
No connectivity Ping the associated IP address to confirm connectivity. If you
cannot successfully ping your destination, refer to the Cisco IOS
IP Configuration Guide.
No connectivity Enter the show controller t1 or show controller e1 command
with the controller number for the voice port you are
troubleshooting. This will tell you:
If the controller is up. If it is not, use the no shutdown
command to make it active.
Whether alarms have been reported.
What parameter values have been set for the controller
(framing, clock source, line code, cable length). If these
values do not match those of the telephony connection you
are making, reconfigure the controller.
See the show controller Command: Examples section in the
Verifying Analog and Digital Voice-Port Configurations chapter
for output.


150
No connectivity Enter the show voice port command with the voice port number
that you are troubleshooting, which will tell you:
If the voice port is up. If it is not, use the no shutdown
command to make it active.
What parameter values have been set for the voice port,
including default values (these do not appear in the output for
the show running-config command). If these values do not
match those of the telephony connection you are making,
reconfigure the voice port.
See the show voice port Command: Examples section in the
Verifying Analog and Digital Voice-Port Configurations chapter
for sample output.
Telephony device buzzes or does
not ring
Use the show voice port command to confirm that the ring
frequency command is configured correctly. It must match the
connected telephony equipment and may be country-dependent.
Distorted speech Use the show voice port command to confirm the cptone
keyword setting (also called region tone) is US.
Setting a wrong cptone could result in faulty voice reproduction
during analog-to-digital or digital-to-analog conversions.
Music on hold is not heard Reduce the configured level for the music-threshold command.
Background noise is not heard Enable the comfort-noise command.
Long pauses occur in
conversation; like speaking on a
walkie-talkie
Overall delay is probably excessive; the standard for adequate
voice quality is 150 milliseconds (ms) one-way transit delay.
Measure delay by using ping tests at various times of the day with
different network traffic loads. If delay must be reduced, areas to
examine include propagation delay of signals between the
sending and receiving endpoints, voice encoding delay, and the
voice packetization time for various VoIP codecs.
Jerky or choppy speech Variable delay, or jitter, is being introduced by congestion in the
packet network. Two possible remedies are to:
Reduce the amount of congestion in your packet network.
Pings between VoIP endpoints will give an idea of the
round-trip delay of a link, which should never exceed 300 ms.
Network queuing and dropped packets should also be
examined.
Increase the size of the jitter buffer with the playout-delay
command. (Refer to the Cisco IOS Voice Troubleshooting and
Monitoring Guide.)
Clipped or fuzzy speech Reduce input gain. (Refer to the Cisco IOS Voice
Troubleshooting and Monitoring Guide.)
Change the voice activity detection (VAD) level. Sometimes
VAD cuts the sound too early and the speaker's voice is
clipped. You can also change the time that VAD waits for
silence.
Table 1 Troubleshooting Voice Port Configurations (continued)

151
Note For information on test commands that force voice ports into specific states for testing refer to the
Cisco IOS Voice Troubleshooting and Monitoring Guide.
coincidental.
Clipped speech Reduce the input level at the listeners router. (Refer to the Cisco
IOS Voice Troubleshooting and Monitoring Guide.)
Volume too low or missed
Dual-Tone Multifrequency
(DTMF)
Increase speakers output level or listeners input level. (Refer to
the Cisco IOS Voice Troubleshooting and Monitoring Guide.)
Echo interval is greater than 25 ms
(sounds like a separate voice)
Configure the echo-cancel enable command and increase the
value for the echo-cancel coverage keyword. (See the
Configuring Echo Cancellation section on page 75.)
Too much echo Reduce the output level at the speakers voice port. (Refer to the
Cisco IOS Voice Troubleshooting and Monitoring Guide.)
Table 1 Troubleshooting Voice Port Configurations (continued)


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Table 3 Extended Echo Canceller Algorithm Coverage by Platform (continued)

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