PDF - Complete Book

PDF - Complete Book
Cisco IOS
Voice, Video, and Fax
Command Reference
Release 12.2
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Cisco IOS Voice, Video, and Fax Command Reference
Copyright © 2001–2006 Cisco Systems, Inc.
All rights reserved.
CONTENTS
About Cisco IOS Software Documentation
Using Cisco IOS Software
v
xiii
Cisco IOS Voice, Video, and Fax Commands:
A Through C VR-1
Cisco IOS Voice, Video, and Fax Commands:
D Through F VR-233
Cisco IOS Voice, Video, and Fax Commands:
G Through P VR-339
Cisco IOS Voice, Video, and Fax Commands:
R Through Sh VR-561
Cisco IOS Voice, Video, and Fax Commands:
Si Through Z VR-871
INDEX
Cisco IOS Voice, Video, and Fax Command Reference
iii
Contents
Cisco IOS Voice, Video, and Fax Command Reference
iv
About Cisco IOS Software Documentation
This chapter discusses the objectives, audience, organization, and conventions of Cisco IOS software
documentation. It also provides sources for obtaining documentation from Cisco Systems.
Documentation Objectives
Cisco IOS software documentation describes the tasks and commands necessary to configure and
maintain Cisco networking devices.
Audience
The Cisco IOS software documentation set is intended primarily for users who configure and maintain
Cisco networking devices (such as routers and switches) but who may not be familiar with the tasks,
the relationship between tasks, or the Cisco IOS software commands necessary to perform particular
tasks. The Cisco IOS software documentation set is also intended for those users experienced with
Cisco IOS software who need to know about new features, new configuration options, and new software
characteristics in the current Cisco IOS software release.
Documentation Organization
The Cisco IOS software documentation set consists of documentation modules and master indexes. In
addition to the main documentation set, there are supporting documents and resources.
Documentation Modules
The Cisco IOS documentation modules consist of configuration guides and corresponding command
reference publications. Chapters in a configuration guide describe protocols, configuration tasks, and
Cisco IOS software functionality and contain comprehensive configuration examples. Chapters in a
command reference publication provide complete Cisco IOS command syntax information. Use each
configuration guide in conjunction with its corresponding command reference publication.
Cisco IOS Voice, Video, and Fax Command Reference
v
About Cisco IOS Software Documentation
Documentation Organization
Figure 1 shows the Cisco IOS software documentation modules.
Note
Figure 1
The abbreviations (for example, FC and FR) next to the book icons are page designators,
which are defined in a key in the index of each document to help you with navigation. The
bullets under each module list the major technology areas discussed in the corresponding
books.
Cisco IOS Software Documentation Modules
IPC
FC
Cisco IOS
Configuration
Fundamentals
Configuration
Guide
Cisco IOS
Configuration
Fundamentals
Command
Reference
FR
IP2R
Module FC/FR:
• Cisco IOS User
Interfaces
• File Management
• System Management
WR
Cisco IOS
Wide-Area
Networking
Command
Reference
Module WC/WR:
• ATM
• Broadband Access
• Frame Relay
• SMDS
• X.25 and LAPB
Cisco IOS
IP Command
Reference,
Volume 1 of 3:
Addressing
and Services
Cisco IOS
IP Command
Reference,
Volume 2 of 3:
Routing
Protocols
P2C
IP3R
Cisco IOS
IP Command
Reference,
Volume 3 of 3:
Multicast
Cisco IOS
Interface
Configuration
Guide
IR
Cisco IOS
Interface
Command
Reference
P3C
Cisco IOS
AppleTalk and
Novell IPX
Configuration
Guide
P2R
Module IPC/IP1R/IP2R/IP3R:
• IP Addressing and Services
• IP Routing Protocols
• IP Multicast
IC
Cisco IOS
Wide-Area
Networking
Configuration
Guide
IP1R
Cisco IOS
AppleTalk and
Novell IPX
Command
Reference
P3R
Module P2C/P2R:
• AppleTalk
• Novell IPX
MWC
Cisco IOS
Mobile
Wireless
Configuration
Guide
MWR
Module IC/IR:
• LAN Interfaces
• Serial Interfaces
• Logical Interfaces
Cisco IOS
Mobile
Wireless
Command
Reference
Module MWC/MWR:
• General Packet
Radio Service
Cisco IOS
Apollo Domain,
Banyan VINES,
DECnet, ISO
CLNS, and XNS
Configuration
Guide
SC
Cisco IOS
Apollo Domain,
Banyan VINES,
DECnet, ISO
CLNS, and XNS
Command
Reference
Module P3C/P3R:
• Apollo Domain
• Banyan VINES
• DECnet
• ISO CLNS
• XNS
Cisco IOS
Security
Configuration
Guide
SR
Cisco IOS
Security
Command
Reference
Module SC/SR:
• AAA Security Services
• Security Server Protocols
• Traffic Filtering and Firewalls
• IP Security and Encryption
• Passwords and Privileges
• Neighbor Router Authentication
• IP Security Options
• Supported AV Pairs
47953
WC
Cisco IOS
IP
Configuration
Guide
Cisco IOS Voice, Video, and Fax Command Reference
vi
About Cisco IOS Software Documentation
Documentation Organization
Cisco IOS
Dial
Technologies
Configuration
Guide
TC
BC
Cisco IOS
Terminal
Services
Configuration
Guide
Cisco IOS
Bridging and
IBM Networking
Configuration
Guide
B2R
B1R
DR
Cisco IOS
Dial
Technologies
Command
Reference
TR
Module DC/DR:
• Preparing for Dial Access
• Modem and Dial Shelf Configuration
and Management
• ISDN Configuration
• Signalling Configuration
• Dial-on-Demand Routing
Configuration
• Dial-Backup Configuration
• Dial-Related Addressing Services
• Virtual Templates, Profiles, and
Networks
• PPP Configuration
• Callback and Bandwidth Allocation
Configuration
• Dial Access Specialized Features
• Dial Access Scenarios
VC
Cisco IOS
Voice, Video,
and Fax
Configuration
Guide
VR
Cisco IOS
Voice, Video,
and Fax
Command
Reference
Module VC/VR:
• Voice over IP
• Call Control Signalling
• Voice over
Frame Relay
• Voice over ATM
• Telephony Applications
• Trunk Management
• Fax, Video, and
Modem Support
Cisco IOS
Terminal
Services
Command
Reference
Module TC/TR:
• ARA
• LAT
• NASI
• Telnet
• TN3270
• XRemote
• X.28 PAD
• Protocol Translation
QC
Cisco IOS
Quality of
Service
Solutions
Configuration
Guide
QR
Cisco IOS
Quality of
Service
Solutions
Command
Reference
Module QC/QR:
• Packet Classification
• Congestion Management
• Congestion Avoidance
• Policing and Shaping
• Signalling
• Link Efficiency
Mechanisms
Cisco IOS
Bridging
and IBM
Networking
Command
Reference,
Volume 1 of 2
Cisco IOS
Bridging
and IBM
Networking
Command
Reference,
Volume 2 of 2
Module BC/B1R:
• Transparent
Bridging
• SRB
• Token Ring
Inter-Switch Link
• Token Ring Route
Switch Module
• RSRB
• DLSw+
• Serial Tunnel and
Block Serial Tunnel
• LLC2 and SDLC
• IBM Network
Media Translation
• SNA Frame Relay
Access
• NCIA Client/Server
• Airline Product Set
XC
Module BC/B2R:
• DSPU and SNA
Service Point
• SNA Switching
Services
• Cisco Transaction
Connection
• Cisco Mainframe
Channel Connection
• CLAW and TCP/IP
Offload
• CSNA, CMPC,
and CMPC+
• TN3270 Server
Cisco IOS
Switching
Services
Configuration
Guide
XR
Cisco IOS
Switching
Services
Command
Reference
Module XC/XR:
• Cisco IOS
Switching Paths
• NetFlow Switching
• Multiprotocol Label Switching
• Multilayer Switching
• Multicast Distributed Switching
• Virtual LANs
• LAN Emulation
47954
DC
Cisco IOS Voice, Video, and Fax Command Reference
vii
About Cisco IOS Software Documentation
Documentation Organization
Master Indexes
Two master indexes provide indexing information for the Cisco IOS software documentation set:
an index for the configuration guides and an index for the command references. Individual books also
contain a book-specific index.
The master indexes provide a quick way for you to find a command when you know the command name
but not which module contains the command. When you use the online master indexes, you can click
the page number for an index entry and go to that page in the online document.
Supporting Documents and Resources
The following documents and resources support the Cisco IOS software documentation set:
•
Cisco IOS Command Summary (two volumes)—This publication explains the function and syntax
of the Cisco IOS software commands. For more information about defaults and usage guidelines,
refer to the Cisco IOS command reference publications.
•
Cisco IOS System Error Messages—This publication lists and describes Cisco IOS system error
messages. Not all system error messages indicate problems with your system. Some are purely
informational, and others may help diagnose problems with communications lines, internal
hardware, or the system software.
•
Cisco IOS Debug Command Reference—This publication contains an alphabetical listing of the
debug commands and their descriptions. Documentation for each command includes a brief
description of its use, command syntax, usage guidelines, and sample output.
•
Dictionary of Internetworking Terms and Acronyms—This Cisco publication compiles and defines
the terms and acronyms used in the internetworking industry.
•
New feature documentation—The Cisco IOS software documentation set documents the mainline
release of Cisco IOS software (for example, Cisco IOS Release 12.2). New software features are
introduced in early deployment releases (for example, the Cisco IOS “T” release train for 12.2,
12.2(x)T). Documentation for these new features can be found in standalone documents called
“feature modules.” Feature module documentation describes new Cisco IOS software and hardware
networking functionality and is available on Cisco.com and the Documentation CD-ROM.
•
Release notes—This documentation describes system requirements, provides information about
new and changed features, and includes other useful information about specific software releases.
See the section “Using Software Release Notes” in the chapter “Using Cisco IOS Software” for
more information.
•
Caveats documentation—This documentation provides information about Cisco IOS software
defects in specific software releases.
•
RFCs—RFCs are standards documents maintained by the Internet Engineering Task Force (IETF).
Cisco IOS software documentation references supported RFCs when applicable. The full text of
referenced RFCs may be obtained on the World Wide Web at http://www.rfc-editor.org/.
•
MIBs—MIBs are used for network monitoring. For lists of supported MIBs by platform and
release, and to download MIB files, see the Cisco MIB website on Cisco.com at
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
Cisco IOS Voice, Video, and Fax Command Reference
viii
About Cisco IOS Software Documentation
Document Conventions
Document Conventions
Within Cisco IOS software documentation, the term router is generally used to refer to a variety of Cisco
products (for example, routers, access servers, and switches). Routers, access servers, and other
networking devices that support Cisco IOS software are shown interchangeably within examples. These
products are used only for illustrative purposes; that is, an example that shows one product does not
necessarily indicate that other products are not supported.
The Cisco IOS documentation set uses the following conventions:
Convention
Description
^ or Ctrl
The ^ and Ctrl symbols represent the Control key. For example, the key combination ^D or Ctrl-D
means 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 an SNMP
community string to public, do not use quotation marks around the string or the string will include the
quotation marks.
Command syntax descriptions use the following conventions:
Convention
Description
boldface
Boldface text indicates commands and keywords that you enter literally as shown.
italics
Italic text indicates arguments for which you supply values.
[x]
Square brackets enclose an optional element (keyword or argument).
|
A vertical line indicates a choice within an optional or required set of keywords or arguments.
[x | y]
Square brackets enclosing keywords or arguments separated by a vertical line indicate an optional
choice.
{x | y}
Braces enclosing keywords or arguments separated by a vertical line indicate a required choice.
Nested sets of square brackets or braces indicate optional or required choices within optional or
required elements. For example:
Convention
Description
[x {y | z}]
Braces and a vertical line within square brackets indicate a required choice within an optional element.
Examples use the following conventions:
Convention
Description
screen
Examples of information displayed on the screen are set in Courier font.
boldface screen
Examples of text that you must enter are set in Courier bold font.
<
Angle brackets enclose text that is not printed to the screen, such as passwords.
>
Cisco IOS Voice, Video, and Fax Command Reference
ix
About Cisco IOS Software Documentation
Obtaining Documentation
Convention
Description
!
An exclamation point at the beginning of a line indicates a comment line. (Exclamation points are also
displayed by the Cisco IOS software for certain processes.)
[
]
Square brackets enclose default responses to system prompts.
The following conventions are used to attract the attention of the reader:
Caution
Note
Timesaver
Means reader be careful. In this situation, you might do something that could result in
equipment damage or loss of data.
Means reader take note. Notes contain helpful suggestions or references to materials not
contained in this manual.
Means the described action saves time. You can save time by performing the action
described in the paragraph.
Obtaining Documentation
The following sections provide sources for obtaining documentation from Cisco Systems.
World Wide Web
The most current Cisco documentation is available on the World Wide Web at the following website:
http://www.cisco.com
Translated documentation is available at the following website:
http://www.cisco.com/public/countries_languages.html
Documentation CD-ROM
Cisco documentation and additional literature are available in a CD-ROM package, which ships
with your product. The Documentation CD-ROM is updated monthly and may be more current than
printed documentation. The CD-ROM package is available as a single unit or through an
annual subscription.
Cisco IOS Voice, Video, and Fax Command Reference
x
About Cisco IOS Software Documentation
Documentation Feedback
Ordering Documentation
Cisco documentation can be ordered in the following ways:
•
Registered Cisco Direct Customers can order Cisco product documentation from the Networking
Products MarketPlace:
http://www.cisco.com/cgi-bin/order/order_root.pl
•
Registered Cisco.com users can order the Documentation CD-ROM through the online
Subscription Store:
http://www.cisco.com/go/subscription
•
Nonregistered Cisco.com users can order documentation through a local account representative by
calling Cisco corporate headquarters (California, USA) at 408 526-7208 or, in North America, by
calling 800 553-NETS(6387).
Documentation Feedback
If you are reading Cisco product documentation on the World Wide Web, you can submit technical
comments electronically. Click Feedback in the toolbar and select Documentation. After you complete
the form, click Submit to send it to Cisco.
You can e-mail your comments to [email protected]
To submit your comments by mail, use the response card behind the front cover of your document, or
write to the following address:
Cisco Systems, Inc.
Document Resource Connection
170 West Tasman Drive
San Jose, CA 95134-9883
We appreciate your comments.
Obtaining Technical Assistance
Cisco provides Cisco.com as a starting point for all technical assistance. Customers and partners can
obtain documentation, troubleshooting tips, and sample configurations from online tools. For
Cisco.com registered users, additional troubleshooting tools are available from the TAC website.
Cisco.com
Cisco.com is the foundation of a suite of interactive, networked services that provides immediate, open
access to Cisco information and resources at anytime, from anywhere in the world. This highly
integrated Internet application is a powerful, easy-to-use tool for doing business with Cisco.
Cisco.com provides a broad range of features and services to help customers and partners streamline
business processes and improve productivity. Through Cisco.com, you can find information about Cisco
and our networking solutions, services, and programs. In addition, you can resolve technical issues with
online technical support, download and test software packages, and order Cisco learning materials and
merchandise. Valuable online skill assessment, training, and certification programs are also available.
Cisco IOS Voice, Video, and Fax Command Reference
xi
About Cisco IOS Software Documentation
Obtaining Technical Assistance
Customers and partners can self-register on Cisco.com to obtain additional personalized information
and services. Registered users can order products, check on the status of an order, access technical
support, and view benefits specific to their relationships with Cisco.
To access Cisco.com, go to the following website:
http://www.cisco.com
Technical Assistance Center
The Cisco TAC website is available to all customers who need technical assistance with a Cisco product
or technology that is under warranty or covered by a maintenance contract.
Contacting TAC by Using the Cisco TAC Website
If you have a priority level 3 (P3) or priority level 4 (P4) problem, contact TAC by going to the TAC
website:
http://www.cisco.com/tac
P3 and P4 level problems are defined as follows:
•
P3—Your network performance is degraded. Network functionality is noticeably impaired, but
most business operations continue.
•
P4—You need information or assistance on Cisco product capabilities, product installation, or basic
product configuration.
In each of the above cases, use the Cisco TAC website to quickly find answers to your questions.
To register for Cisco.com, go to the following website:
http://www.cisco.com/register/
If you cannot resolve your technical issue by using the TAC online resources, Cisco.com registered
users can open a case online by using the TAC Case Open tool at the following website:
http://www.cisco.com/tac/caseopen
Contacting TAC by Telephone
If you have a priority level 1 (P1) or priority level 2 (P2) problem, contact TAC by telephone and
immediately open a case. To obtain a directory of toll-free numbers for your country, go to the following
website:
http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml
P1 and P2 level problems are defined as follows:
•
P1—Your production network is down, causing a critical impact to business operations if service
is not restored quickly. No workaround is available.
•
P2—Your production network is severely degraded, affecting significant aspects of your business
operations. No workaround is available.
Cisco IOS Voice, Video, and Fax Command Reference
xii
Using Cisco IOS Software
This chapter provides helpful tips for understanding and configuring Cisco IOS software using the
command-line interface (CLI). It contains the following sections:
•
Understanding Command Modes
•
Getting Help
•
Using the no and default Forms of Commands
•
Saving Configuration Changes
•
Filtering Output from the show and more Commands
•
Identifying Supported Platforms
For an overview of Cisco IOS software configuration, refer to the Cisco IOS Configuration
Fundamentals Configuration Guide.
For information on the conventions used in the Cisco IOS software documentation set, see the chapter
“About Cisco IOS Software Documentation” located at the beginning of this book.
Understanding Command Modes
You use the CLI to access Cisco IOS software. Because the CLI is divided into many different modes,
the commands available to you at any given time depend on the mode you are currently in. Entering a
question mark (?) at the CLI prompt allows you to obtain a list of commands available for each
command mode.
When you log in to the CLI, you are in user EXEC mode. User EXEC mode contains only a limited
subset of commands. To have access to all commands, you must enter privileged EXEC mode, normally
by using a password. From privileged EXEC mode you can issue any EXEC command—user or
privileged mode—or you can enter global configuration mode. Most EXEC commands are one-time
commands. For example, show commands show important status information, and clear commands
clear counters or interfaces. The EXEC commands are not saved when the software reboots.
Configuration modes allow you to make changes to the running configuration. If you later save the
running configuration to the startup configuration, these changed commands are stored when the
software is rebooted. To enter specific configuration modes, you must start at global configuration
mode. From global configuration mode, you can enter interface configuration mode and a variety of
other modes, such as protocol-specific modes.
ROM monitor mode is a separate mode used when the Cisco IOS 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.
Cisco IOS Voice, Video, and Fax Command Reference
xiii
Using Cisco IOS Software
Getting Help
Table 1 describes how to access and exit various common command modes of the Cisco IOS software.
It also shows examples of the prompts displayed for each mode.
Table 1
Accessing and Exiting Command Modes
Command
Mode
Access Method
Prompt
Exit Method
User EXEC
Log in.
Router>
Use the logout command.
Privileged
EXEC
From user EXEC mode,
use the enable EXEC
command.
Router#
To return to user EXEC mode, use the disable
command.
Global
configuration
From privileged EXEC
mode, use the configure
terminal privileged
EXEC command.
Router(config)#
To return to privileged EXEC mode from global
configuration mode, use the exit or end command,
or press Ctrl-Z.
Interface
configuration
Router(config-if)#
From global
configuration mode,
specify an interface using
an interface command.
To return to global configuration mode, use the exit
command.
>
From privileged EXEC
mode, use the reload
EXEC command. Press
the Break key during the
first 60 seconds while the
system is booting.
To exit ROM monitor mode, use the continue
command.
ROM monitor
To return to privileged EXEC mode, use the end
command, or press Ctrl-Z.
For more information on command modes, refer to the “Using the Command-Line Interface” chapter in
the Cisco IOS Configuration Fundamentals Configuration Guide.
Getting Help
Entering a question mark (?) at the CLI prompt displays a list of commands available for each command
mode. You can also get a list of keywords and arguments associated with any command by using the
context-sensitive help feature.
To get help specific to a command mode, a command, a keyword, or an argument, use one of the
following commands:
Command
Purpose
help
Provides a brief description of the help system in any command mode.
abbreviated-command-entry?
Provides a list of commands that begin with a particular character string. (No space
between command and question mark.)
abbreviated-command-entry<Tab>
Completes a partial command name.
?
Lists all commands available for a particular command mode.
command ?
Lists the keywords or arguments that you must enter next on the command line.
(Space between command and question mark.)
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Using Cisco IOS Software
Getting Help
Example: How to Find Command Options
This section provides an example of how to display syntax for a command. The syntax can consist of
optional or required keywords and arguments. To display keywords and arguments for a command, enter
a question mark (?) at the configuration prompt or after entering part of a command followed by a space.
The Cisco IOS software displays a list and brief description of available keywords and arguments. For
example, if you were in global configuration mode and wanted to see all the keywords or arguments for
the arap command, you would type arap ?.
The <cr> symbol in command help output stands for “carriage return.” On older keyboards, the carriage
return key is the Return key. On most modern keyboards, the carriage return key is the Enter key. The
<cr> symbol at the end of command help output indicates that you have the option to press Enter to
complete the command and that the arguments and keywords in the list preceding the <cr> symbol are
optional. The <cr> symbol by itself indicates that no more arguments or keywords are available and that
you must press Enter to complete the command.
Table 2 shows examples of how you can use the question mark (?) to assist you in entering commands.
The table steps you through configuring an IP address on a serial interface on a Cisco 7206 router that
is running Cisco IOS Release 12.0(3).
Table 2
How to Find Command Options
Command
Comment
Router> enable
Password: <password>
Router#
Enter the enable command and
password to access privileged EXEC
commands. You are in privileged
EXEC mode when the prompt changes
to Router#.
Router# configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)#
Enter the configure terminal
privileged EXEC command to enter
global configuration mode. You are in
global configuration mode when the
prompt changes to Router(config)#.
Router(config)# interface serial ?
<0-6>
Serial interface number
Router(config)# interface serial 4 ?
/
Router(config)# interface serial 4/ ?
<0-3>
Serial interface number
Router(config)# interface serial 4/0
Router(config-if)#
Enter interface configuration mode by
specifying the serial interface that you
want to configure using the interface
serial global configuration command.
Enter ? to display what you must enter
next on the command line. In this
example, you must enter the serial
interface slot number and port number,
separated by a forward slash.
You are in interface configuration mode
when the prompt changes to
Router(config-if)#.
Cisco IOS Voice, Video, and Fax Command Reference
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Using Cisco IOS Software
Getting Help
Table 2
How to Find Command Options (continued)
Command
Comment
Router(config-if)# ?
Interface configuration commands:
.
.
.
ip
Interface Internet Protocol config commands
keepalive
Enable keepalive
lan-name
LAN Name command
llc2
LLC2 Interface Subcommands
load-interval
Specify interval for load calculation for an
interface
locaddr-priority
Assign a priority group
logging
Configure logging for interface
loopback
Configure internal loopback on an interface
mac-address
Manually set interface MAC address
mls
mls router sub/interface commands
mpoa
MPOA interface configuration commands
mtu
Set the interface Maximum Transmission Unit (MTU)
netbios
Use a defined NETBIOS access list or enable
name-caching
no
Negate a command or set its defaults
nrzi-encoding
Enable use of NRZI encoding
ntp
Configure NTP
.
.
.
Router(config-if)#
Enter ? to display a list of all the
interface configuration commands
available for the serial interface. This
example shows only some of the
available interface configuration
commands.
Router(config-if)# ip ?
Interface IP configuration subcommands:
access-group
Specify access control for packets
accounting
Enable IP accounting on this interface
address
Set the IP address of an interface
authentication
authentication subcommands
bandwidth-percent
Set EIGRP bandwidth limit
broadcast-address
Set the broadcast address of an interface
cgmp
Enable/disable CGMP
directed-broadcast Enable forwarding of directed broadcasts
dvmrp
DVMRP interface commands
hello-interval
Configures IP-EIGRP hello interval
helper-address
Specify a destination address for UDP broadcasts
hold-time
Configures IP-EIGRP hold time
.
.
.
Router(config-if)# ip
Enter the command that you want to
configure for the interface. This
example uses the ip command.
Cisco IOS Voice, Video, and Fax Command Reference
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Enter ? to display what you must enter
next on the command line. This
example shows only some of the
available interface IP configuration
commands.
Using Cisco IOS Software
Using the no and default Forms of Commands
Table 2
How to Find Command Options (continued)
Command
Comment
Router(config-if)# ip address ?
A.B.C.D
IP address
negotiated
IP Address negotiated over PPP
Router(config-if)# ip address
Enter the command that you want to
configure for the interface. This
example uses the ip address command.
Enter ? to display what you must enter
next on the command line. In this
example, you must enter an IP address
or the negotiated keyword.
A carriage return (<cr>) is not
displayed; therefore, you must enter
additional keywords or arguments to
complete the command.
Enter the keyword or argument you
want to use. This example uses the
172.16.0.1 IP address.
Router(config-if)# ip address 172.16.0.1 ?
A.B.C.D
IP subnet mask
Router(config-if)# ip address 172.16.0.1
Enter ? to display what you must enter
next on the command line. In this
example, you must enter an IP subnet
mask.
A <cr> is not displayed; therefore, you
must enter additional keywords or
arguments to complete the command.
Router(config-if)# ip address 172.16.0.1 255.255.255.0 ?
secondary
Make this IP address a secondary address
<cr>
Router(config-if)# ip address 172.16.0.1 255.255.255.0
Enter the IP subnet mask. This example
uses the 255.255.255.0 IP subnet mask.
Enter ? to display what you must enter
next on the command line. In this
example, you can enter the secondary
keyword, or you can press Enter.
A <cr> is displayed; you can press
Enter to complete the command, or
you can enter another keyword.
Router(config-if)# ip address 172.16.0.1 255.255.255.0
Router(config-if)#
In this example, Enter is pressed to
complete the command.
Using the no and default Forms of Commands
Almost every configuration command has a no form. In general, use the no form to disable a function.
Use the command without the no keyword to reenable a disabled function or to enable a function that
is disabled by default. For example, IP routing is enabled by default. To disable IP routing, use the no
ip routing command; to reenable IP routing, use the ip routing command. The Cisco IOS software
command reference publications provide the complete syntax for the configuration commands and
describe what the no form of a command does.
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Using Cisco IOS Software
Saving Configuration Changes
Configuration commands also can have a default form, which returns the command settings to the
default values. Most commands are disabled by default, so in such cases using the default form has the
same result as using the no form of the command. However, some commands are enabled by default and
have variables set to certain default values. In these cases, the default form of the command enables the
command and sets the variables to their default values. The Cisco IOS software command reference
publications describe the effect of the default form of a command if the command functions differently
than the no form.
Saving Configuration Changes
Use the copy system:running-config nvram:startup-config command to save your configuration
changes to the startup configuration so that the changes will not be lost if the software reloads or a
power outage occurs. For example:
Router# copy system:running-config nvram:startup-config
Building configuration...
It might take a minute or two to save the configuration. After the configuration has been saved, the
following output appears:
[OK]
Router#
On most platforms, this task saves the configuration to NVRAM. On the Class A Flash file system
platforms, this task saves the configuration to the location specified by the CONFIG_FILE environment
variable. The CONFIG_FILE variable defaults to NVRAM.
Filtering Output from the show and more Commands
In Cisco IOS Release 12.0(1)T and later releases, you can search and filter the output of show and more
commands. This functionality is useful if you need to sort through large amounts of output or if you
want to exclude output that you need not see.
To use this functionality, enter a show or more command followed by the “pipe” character (|); one of
the keywords begin, include, or exclude; and a regular expression on which you want to search or filter
(the expression is case-sensitive):
command | {begin | include | exclude} regular-expression
The output matches certain lines of information in the configuration file. The following example
illustrates how to use output modifiers with the show interface command when you want the output to
include only lines in which the expression “protocol” appears:
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
For more information on the search and filter functionality, refer to the “Using the Command-Line
Interface” chapter in the Cisco IOS Configuration Fundamentals Configuration Guide.
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Using Cisco IOS Software
Identifying Supported Platforms
Identifying Supported Platforms
Cisco IOS software is packaged in feature sets consisting of software images that support specific
platforms. The feature sets available for a specific platform depend on which Cisco IOS software
images are included in a release. To identify the set of software images available in a specific release
or to find out if a feature is available in a given Cisco IOS software image, see the following sections:
•
Using Feature Navigator
•
Using Software Release Notes
Using Feature Navigator
Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software
images support a particular set of features and which features are supported in a particular Cisco IOS
image.
Feature Navigator is available 24 hours a day, 7 days a week. To access Feature Navigator, you must
have an account on Cisco.com. If you have forgotten or lost your account information, e-mail the
Contact Database Administration group at [email protected] If you do not have an account on
Cisco.com, go to http://www.cisco.com/register and follow the directions to establish an account.
To use Feature Navigator, you must have a JavaScript-enabled web browser such as Netscape 3.0 or
later, or Internet Explorer 4.0 or later. Internet Explorer 4.0 always has JavaScript enabled. To enable
JavaScript for Netscape 3.x or Netscape 4.x, follow the instructions provided with the web browser. For
JavaScript support and enabling instructions for other browsers, check with the browser vendor.
Feature Navigator is updated when major Cisco IOS software releases and technology releases occur.
You can access Feature Navigator at the following URL:
http://www.cisco.com/go/fn
Using Software Release Notes
Cisco IOS software releases include release notes that provide the following information:
•
Platform support information
•
Memory recommendations
•
Microcode support information
•
Feature set tables
•
Feature descriptions
•
Open and resolved severity 1 and 2 caveats for all platforms
Release notes are intended to be release-specific for the most current release, and the information
provided in these documents may not be cumulative in providing information about features that first
appeared in previous releases.
Cisco IOS Voice, Video, and Fax Command Reference
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Using Cisco IOS Software
Identifying Supported Platforms
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands:
A Through C
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax
applications. The commands are presented in alphabetical order. Some commands required for
configuring voice, video, and fax may be found in other Cisco IOS command references. Use the
command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS
Voice, Video, and Fax Configuration Guide.
Cisco IOS Voice, Video, and Fax Command Reference
VR-1
Cisco IOS Voice, Video, and Fax Commands: A Through C
aaa nas port voip
aaa nas port voip
To send out the standard NAS-Port attribute (RADIUS IETF Attribute 5) on voice interfaces, use the aaa
nas port voip command in global configuration mode. To disable the command, use the no form of the
command.
aaa nas port voip
no aaa nas port voip
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.2(1)T
This command was introduced on the Cisco AS5300.
Usage Guidelines
Note
Examples
This command brings back the original behavior of the AAA NAS-Port on Voice over IP (VoIP)
interfaces. By default this feature should not be enabled.
Some customers using the Cisco AS5300 voice gateway have had the Debit Card
application stop working after upgrading from 12.1(5)T to 12.1(5.3)T.
The following example shows how to return to the original behavior of the AAA NAS-Port:
aaa nas port voip
Related Commands
Command
Description
aaa nas port extended
Replaces the NAS-port attribute with RADIUS IETF attribute 26 and
displays extended field information.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
acc-qos
acc-qos
To define the acceptable quality of service (QoS) for any inbound and outbound call on a Voice over IP
(VoIP) dial peer, use the acc-qos command in dial peer configuration mode. To restore the default QoS
setting, use the no form of this command.
acc-qos {best-effort | controlled-load | guaranteed-delay}
no acc-qos
Syntax Description
best-effort
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth
reservation. This is the default.
controlled-load
Indicates that RSVP guarantees a single level of preferential service, presumed
to correlate to a delay boundary. The controlled load service uses admission (or
capacity) control to ensure that preferential service is received even when the
bandwidth is overloaded.
guaranteed-delay
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and
preferential queueing if the bandwidth reserved is not exceeded.
Defaults
best-effort
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.1(5)T
The description of the command was modified.
Usage Guidelines
This command is applicable only to VoIP dial peers.
When VoIP dial peers are used, the Cisco IOS software uses RSVP to reserve a certain amount of
bandwidth so that the selected QoS can be provided by the network. Call setup is aborted if the RSVP
resource reservation does not satisfy the acceptable QoS for both peers.
To select the most appropriate value for this command, you need to be familiar with the amount of traffic
this connection supports and what kind of impact you are willing to have on it. The Cisco IOS software
generates a trap message when the bandwidth required to provide the selected quality of service is not
available.
Examples
The following example selects guaranteed-delay as the acceptable QoS for inbound and outbound calls
on VoIP dial peer 10:
dial-peer voice 10 voip
acc-qos guaranteed-delay
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
acc-qos
Related Commands
Command
Description
req-qos
Requests a particular QoS using RSVP to be used in reaching a specified dial
peer in VoIP.
Cisco IOS Voice, Video, and Fax Command Reference
VR-4
Cisco IOS Voice, Video, and Fax Commands: A Through C
alarm-trigger
alarm-trigger
To configure a T1 or E1 controller to send an alarm to the public switched telephone network (PSTN) or
switch if specified T1 or E1 DS0 groups are out of service, use the alarm-trigger command in controller
configuration mode. To configure a T1 or E1 controller not to send an alarm, use the no form of this
command.
alarm-trigger blue ds0-group-list
no alarm-trigger
Syntax Description
blue
Specifies the alarm type to be sent is “blue,” also known as an Alarm Indication
Signal (AIS).
ds0-group-list
Specifies the DS0 group or groups to be monitored for permanent trunk
connection status or busyout status.
Defaults
No alarm is sent.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600, 3600, and MC3810
multiservice concentrator.
Usage Guidelines
Any monitored time slot can be used for either permanent trunk connections or switched connections.
Permanent virtual circuits (PVCs) and switched virtual circuits (SVCs) can be combined on a T1 or E1
controller and monitored for alarm conditioning.
An alarm is sent only if all of the time slots configured for alarm conditioning on a T1 or E1 controller
are out of service. If one monitored time slot remains in service or returns to service, no alarm is sent.
Examples
The following example configures T1 0 to send a blue (AIS) alarm if DS0 groups 0 and 1 are out of
service:
controller t1 0
alarm-trigger blue 0,1
exit
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
alarm-trigger
Related Commands
Command
Description
busyout monitor
Configures a voice port to monitor an interface for events that would
trigger a voice-port busyout.
connection trunk
Creates a permanent trunk connection (private line or tie-line) between a
voice port and a PBX.
voice class permanent
Creates a voice class for a Cisco or FRF-11 permanent trunk.
Cisco IOS Voice, Video, and Fax Command Reference
VR-6
Cisco IOS Voice, Video, and Fax Commands: A Through C
alias static
alias static
To create a static entry in the local alias table, use the alias static command in gatekeeper configuration
mode. To remove a static entry, use the no form of this command.
alias static ip-signaling-addr [port] gkid gatekeeper-name [ras ip-ras-addr port] [terminal | mcu |
gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id]
no alias static ip-signaling-addr [port] gkid gatekeeper-name [ras ip-ras-addr port] [terminal |
mcu | gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id]
Syntax Description
ip-signaling-addr
IP address of the H.323 node, used as the address to signal when establishing
a call.
port
(Optional) Port number other than the endpoint Call Signaling well-known
port number (1720).
gkid gatekeeper-name Name of the local gatekeeper of whose zone this node is a member.
ras ip-ras-addr
(Optional) Node remote access server (RAS) signaling address. If omitted,
the ip-signaling-addr parameter is used in conjunction with the RAS
well-known port.
port
(Optional) Port number other than the RAS well-known port number (1719).
terminal
(Optional) Indicates that the alias refers to a terminal.
mcu
(Optional) Indicates that the alias refers to a multiple control unit (MCU).
gateway
(Optional) Indicates that the alias refers to a gateway.
h320
(Optional) Indicates that the alias refers to an H.320 node.
h323-proxy
(Optional) Indicates that the alias refers to an H.323 proxy.
voip
(Optional) Indicates that the alias refers to VoIP.
e164 e164-address
(Optional) Specifies the node E.164 address. This keyword and argument can
be used more than once to specify as many E.164 addresses as needed. Note
that there is a maximum number of 128 characters that can be entered for this
address. To avoid exceeding this limit, you can enter multiple alias static
commands with the same call signaling address and different aliases.
h323id h323-id
(Optional) Specifies the node H.323 alias. This keyword and argument can be
used more than once to specify as many H.323 identification (ID) aliases as
needed. Note that there is a maximum number of 256 characters that can be
entered for this address. To avoid exceeding this limit, you can enter multiple
alias static commands with the same call signaling address and different
aliases.
Defaults
No static aliases exist.
Command Modes
Gatekeeper configuration
Cisco IOS Voice, Video, and Fax Command Reference
VR-7
Cisco IOS Voice, Video, and Fax Commands: A Through C
alias static
Command History
Usage Guidelines
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
The local alias table can be used to load static entries by performing as many of the commands as
necessary. Aliases for the same IP address can be added in different commands, if required.
Typically, static aliases are needed to access endpoints that do not belong to a zone (that is, they are not
registered with any gatekeeper) or whose gatekeeper is inaccessible for some reason.
Examples
The following example creates a static terminal alias in the local zone:
zone local gk.zone1.com zone1.com
alias static 191.7.8.5 gkid gk.zone1.com terminal e164 14085551212 h323id bobs_terminal
Cisco IOS Voice, Video, and Fax Command Reference
VR-8
Cisco IOS Voice, Video, and Fax Commands: A Through C
alt-dial
alt-dial
To configure an alternate dial-out string for dial peers on the Cisco MC3810 multiservice concentrator,
use the alt-dial command in dial peer configuration mode. To delete the alternate dial-out string, use the
no form of this command.
alt-dial string
no alt-dial string
Syntax Description
string
Defaults
No alternate dial-out string is configured.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
The alternate dial-out string.
This command applies to Cisco MC3810 multiservice concentrator plain old telephone service (POTS),
Voice over Frame Relay (VoFR), and Voice over ATM (VoATM) dial peers.
The alt-dial command is used for the on-net-to-off-net alternative dialing function. The string replaces
the destination-pattern string for dialing out.
Examples
The following example configures an alternate dial-out string of 9,5559871:
alt-dial 9,5559871
Cisco IOS Voice, Video, and Fax Command Reference
VR-9
Cisco IOS Voice, Video, and Fax Commands: A Through C
answer-address
answer-address
To specify the full E.164 telephone number to be used to identify the dial peer of an incoming call, use
the answer-address command in dial peer configuration mode. To disable the configured telephone
number, use the no form of this command.
answer-address [+]string[T]
no answer-address
Syntax Description
+
(Optional) Character indicating an E.164 standard number.
string
Series of digits that specify the E.164 or private dial plan telephone number.
Valid entries are as follows:
T
•
Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*),
which represent specific digits that can be entered.
•
Comma (,), which inserts a pause between digits.
•
Period (.), which matches any entered digit.
(Optional) Control character indicating that the answer-address value is a
variable-length dial string.
Defaults
The default value is enabled with a null string.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on Cisco 3600 series routers.
Usage Guidelines
Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP
network. Cisco IOS software identifies the dial peers of a call in one of two ways: either by identifying
the interface through which the call is received or through the telephone number configured with the
answer-address command. In the absence of a configured telephone number, the peer associated with
the interface will be associated with the incoming call.
For calls coming in from a POTS interface, the answer-address command is not used to select an
incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that
dial peer.
There are certain areas in the world (for example, in certain European countries) where valid telephone
numbers can vary in length. Use the optional control character T to indicate that a particular
answer-address value is a variable-length dial string. In this case, the system will not match the dialed
numbers until the interdigit timeout value has expired.
Cisco IOS Voice, Video, and Fax Command Reference
VR-10
Cisco IOS Voice, Video, and Fax Commands: A Through C
answer-address
Note
Examples
The Cisco IOS software does not check the validity of the E.164 telephone number;
it accepts any series of digits as a valid number.
The following example configures the E.164 telephone number 555-9626 as the dial peer of an incoming
call:
dial-peer voice 10 pots
answer-address +5559626
Related Commands
Command
Description
destination-pattern
Specifies either the prefix or the full E.164 telephone number (depending on
your dial plan) to be used for a dial peer.
port (dial peer)
Enables an interface on a PA-4R-DTR port adapter to operate as a
concentrator port.
prefix
Specifies the prefix of the dialed digits for this dial peer.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
application
application
To enable a specific interactive voice response (IVR) application on a dial peer, use the application
command in dial-peer configuration mode. To remove the application from the dial peer, use the no form
of this command.
application application-name [out-bound]
no application application-name [out-bound]
Syntax Description
application-name
Indicates the name of the predefined application you wish to enable
on the dial peer. For H.323 networks, the application is defined by a
Tool Command Language/interactive voice response (TCL/IVR)
filename and location. Incoming calls using plain old telephone
service (POTS) dial peers and outgoing calls using Multimedia Mail
over IP (MMoIP) dial peers are handed off to this application. For
Media Gateway Control Protocol (MGCP) or Simple Gateway
Control Protocol (SGCP) networks, see the usage guidelines below
for valid application names.
out-bound
The named application will handle the MMoIP dial peer in the
outgoing mode.
Defaults
No default behavior or values.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 series, 3600 series,
and AS5300.
12.0(5)T
The SGCPAPP application was supported initially on the
Cisco AS5300 universal access server in a private release that was not
generally available.
12.0(7)XK
Support for the SGCPAPP application was extended to the
Cisco MC3810 multiservice concentrator and the Cisco 3600 series
routers (except for the Cisco 3620) in a private release that was not
generally available.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
The MGCPAPP application was supported initially on the
Cisco AS5300 universal access server.
12.1(3)XI
The out-bound keyword was added for the store-and-forward fax
feature on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
application
Usage Guidelines
Use this command to associate a predefined session application with an incoming POTS dial peer or an
outgoing MMoIP dial peer. Calls using this incoming POTS dial peer or this outgoing MMoIP dial peer
will be handed to the predefined specified session application.
SGCP Networks
For SGCP networks, enter SGCPAPP in uppercase characters. This application can be applied only to
POTS dial peers. Note that SGCP dial peers do not use dial peer hunting.
Note
In Cisco IOS Release 12.2, you cannot mix SGCP and non-SGCP endpoints in the same T1
controller. You also cannot mix SGCP and non-SGCP endpoints in the same DS0 group.
MGCP Networks
For MGCP networks, enter MGCPAPP in upper-case characters. This application can be applied only
to POTS dial peers. Note that MGCP dial peers do not use dial peer hunting.
Examples
The following example shows how to define an application and how to apply it to an outbound MMoIP
dial peer for the fax onramp operation:
call application voice fax_on_vfc_onramp http://santa/username/clid_4digits_npw_3.tcl
dial-peer voice 3 mmoip
application fax_on_vfc_onramp out-bound
destination-pattern 57108..
session target mailto:[email protected]
The following example shows how to apply the MGCP application to a dial peer:
dial-peer voice 1 pots
application MGCPAPP
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and indicates the
location of the appropriate IVR script to be used with this
application.
mgcp
Starts the MGCP daemon.
sgcp
Starts and allocates resources for the SCGP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
arq reject-unknown-prefix
arq reject-unknown-prefix
To enable the gatekeeper to reject admission requests (ARQs) for zone prefixes that are not configured,
use the arq reject-unknown-prefix command in gatekeeper configuration mode. To reenable the
gatekeeper to accept and process all incoming ARQs, use the no form of this command.
arq reject-unknown-prefix
no arq reject-unknown-prefix
Syntax Description
This command has no arguments or keywords.
Defaults
The gatekeeper accepts and processes all incoming ARQs.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(6)Q, 11.3(7)NA
This command was introduced.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
Use the arq reject-unknown-prefix command to configure the gatekeeper to reject any incoming ARQs
for a destination E.164 address that does not match any of the configured zone prefixes.
When an endpoint or gateway initiates an H.323 call, it sends an ARQ to its gatekeeper. The gatekeeper
uses the configured list of zone prefixes to determine where to direct the call. If the called address does
not match any of the known zone prefixes, the gatekeeper attempts to hairpin the call out through a local
gateway. If you do not want your gateway to do this, then use the arq reject-unknown-prefix command.
(The term hairpin is used in telephony. It means to send a call back in the direction from which it came.
For example, if a call cannot be routed over IP to a gateway that is closer to the target phone, the call is
typically sent back out through the local zone, back the way it came.)
This command is typically used to either restrict local gateway calls to a known set of prefixes or
deliberately fail such calls so that an alternate choice on a gateway’s rotary dial peer is selected.
Examples
Consider a gatekeeper configured as follows:
zone
zone
zone
zone
local gk408 cisco.com
remote gk415 cisco.com 172.21.139.91
prefix gk408 1408.......
prefix gk415 1415.......
In this example configuration, the gatekeeper manages a zone containing gateways to the 408 area code,
and it knows about a peer gatekeeper that has gateways to the 415 area code. Using the zone prefix
command, the gatekeeper is then configured with the appropriate prefixes so that calls to those area
codes hop off in the optimal zone.
Cisco IOS Voice, Video, and Fax Command Reference
VR-14
Cisco IOS Voice, Video, and Fax Commands: A Through C
arq reject-unknown-prefix
If the arq request-unknown-prefix command is not configured, the gatekeeper handles calls in the
following way:
•
A call to the 408 area code is routed out through a local gateway.
•
A call to the 415 area code is routed to the gk415 zone, where it hops off on a local gateway.
•
A call to the 212 area code is routed to a local gateway in the gk408 zone.
If the arq reject-unknown-prefix command is configured, the gatekeeper handles calls in the following
way:
•
A call to the 408 area code is routed out through a local gateway.
•
A call to the 415 area code is routed to the gk415 zone, where it hops off on a local gateway.
•
A call to the 212 area code is rejected because the destination address does not match any configured
prefix.
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as
as
To define an application server for backhaul, use the as command in IUA configuration mode. To disable,
use the no form of this command.
Note
All of the ASPs in an application server must be removed before an application server can be
unconfigured.
as as-name {localip1 [localip2]} [local-sctp-port] | [fail-over-timer] [sctp-startup-rtx]
[sctp-streams] [sctp-t1init]
no as as-name
Syntax Description
as-name
Defines the protocol name (only ISDN is supported).
localip1
Defines the local IP address(es) for all the ASPs in a particular AS.
localip2
Defines the local IP address(es) for all the ASPs in a particular AS.
local-sctp-port
Defines a specific local SCTP port rather than an IUA well-known port.
fail-over-timer
(Optional) Configures the failover timer for a particular AS.
sctp-startup-rtx
(Optional) Configures the SCTP maximum startup retransmission timer.
sctp-streams
(Optional) Configures the number of SCTP streams for a particular AS.
sctp-t1init
(Optional) Configures the SCTP t1 init timer.
Defaults
No default behavior or values.
Command Modes
IUA configuration
Command History
Release
Modification
12.2(4)T
This command was introduced.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T and
support was added for the Cisco AS5300 platform.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T on
Cisco 2420, Cisco 2600, Cisco 3600, Cisco 3700, Cisco AS5300, Cisco
AS5350, Cisco AS5400, and Cisco AS5850 network access server (NAS)
platforms.
Usage Guidelines
A maximum of two local IP addresses can be specified (note that SCTP has built-in support for
multi-homed machines).
The default value of the SCTP streams is determined by the hardware that you have installed. The value
of failover timer is found in the show iua as all command output.
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as
The number of streams to assign to a given association is implementation dependent. During the
initialization of the IUA association, you need to specify the total number of streams that can be used.
Each D channel is associated with a specific stream within the association. With multiple trunk group
support, every interface can potentially be a separate D channel.
At start-up the IUA code checks for all the possible T1, E1, or T3 interfaces and sets the total number of
inbound and outbound streams supported accordingly. In most cases, there is only a need for one
association between the GW and the MGC. For the rare case that you are configuring multiple AS
associations to various MGCs, the overhead from the unused streams would have minimal impact. The
NFAS D channels are configured for one or more interfaces, where each interface is assigned a unique
stream ID.
The total number of streams for the association needs to include an additional stream for the SCTP
management messages. So during start-up the IUA code adds one to the total number of interfaces
(streams) found.
You have the option to manually configure the number of streams per association. In the backhaul
scenario, if the number of D channel links is limited to one, allowing the number of streams to be
configurable avoids the unnecessary allocation of streams in an association that will never be used. For
multiple associations between a GW and multiple MGCs, the configuration utility is useful in providing
only the necessary number of streams per association. The overhead from the streams allocated but not
used in the association is negligible.
If the number of streams is manually configured through the CLI, the IUA code cannot distinguish
between a start-up event, which automatically sets the streams to the number of interfaces, or if the value
is set manually during runtime. If you are configuring the number of SCTP streams manually, you must
add one plus the number of interfaces using the sctp-streams keyword. Otherwise, IUA needs to always
add one for the management stream, and the total number of streams increments by one after every
reload.
When you set the SCTP stream with CLI, you cannot change the inbound and outbound stream support
once the association is established with SCTP. The value takes effect when you first remove the IUA AS
configuration and then configure it back as the same AS or a new one. The other option is to reload the
router.
Examples
An application server (AS) and the application server process (ASP) should be configured first to allow
a National ISDN-2 with Cisco extensions (NI2+) to be bound to this transport layer protocol. The AS is
a logical representation of SCTP local end point. The local end point can have more than one IP address
but must use the same port number.
The following is an example of an AS configuration on a gateway:
Router(config-iua)# as as5400-3 10.1.2.34, 10.1.2.35 2577
In the configuration above, an AS named as5400-3 is configured to use two local IP addresses and a port
number of 2577.
The following output shows options available when you use this command:
Router(config-iua)# as as5400-3 fail-over ?
<1000-10000>
set Fail-Over time (in milliseconds) between 1 and 10 seconds
Router(config-iua)# as as5400-3 sctp-stre ?
<2-57>
Specify number of SCTP streams for association
Router(config-iua)# as as5400-3 sctp-startup ?
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as
<2-20>
Set SCTP Maximum Startup Retransmission Interval
Router(config-iua)# as as5400-3 sctp-t1init ?
<1000-60000>
Related Commands
Set SCTP T1 init timer (in milliseconds)
Command
Description
asp
Defines an ASP for backhaul.
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asp
asp
To define an ASP for backhaul, use the asp command in IUA configuration mode. To disable, use the no
form of this command.
Note
All of the ASPs in an application server must be removed before an application server can be
unconfigured.
asp asp-name as as-name {remoteip1 [remoteip2]} [remote-sctp-port] | [ip-precedence
[sctp-keepalives] [sctp-max-associations] [sctp-path-retransmissions] [sctp-t3-timeout]
no asp asp-name
Syntax Description
asp-name
Names the current ASP.
as
The application server to which the ASP belongs.
as-name
Name of the application server to which the ASP belongs.
remoteip1
Designates the remote IP address for this SCTP association.
remoteip2
Designates the remote IP address for this SCTP association.
remote-sctp-port
Connects to a remote SCTP port rather than the IUA well-known port.
ip-precedence
(Optional) Sets IP Precedence bits for protocol data units (PDUs). IP
precedence is expressed in the type of service (ToS) field of the show ip sctp
association parameters output. Default ToS is 0.
sctp-keepalives
(Optional) Modifies the keepalive behavior of an IP address in a particular
ASP. The default is 500 ms (see the show ip sctp association parameters
output under heartbeats).
sctp-max-associations
(Optional) Sets the SCTP max association retransmissions for a particular
ASP. The default value is 3.
sctp-pathretransmissions
(Optional) Sets the SCTP path retransmissions for a particular ASP. The
default value is 5.
sctp-t3-timeout
(Optional) Sets the SCTP T3 retransmission timeout for a particular ASP.
The default value is 900 ms.
Defaults
No default behavior or values.
Command Modes
IUA configuration
Command History
Release
Modification
12.2(4)T
This command was introduced.
12.2(11)T
This command was integrated into Cisco IOS Release Cisco IOS Release
12.2(11)T on Cisco 2420, Cisco 2600, Cisco 3600, Cisco 3700,
Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 network
access server (NAS) platforms.
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asp
Usage Guidelines
This command establishes SCTP associations. There can be only a maximum of three ASPs configured per
AS. IP precedence is expressed in the ToS field of show ip sctp association parameters output. Default
ToS is 0.
You can configure the precedence value in IUA in the range of 0 through 7 for a given IP address. Within
IUA, the upper three bits representing the IP precedence in the ToS byte (used in the IP header) is set
based on the user input before passing down the value to SCTP. In turn, SCTP passes the ToS byte value
to IP. The default value is 0 for “normal” IP precedence handling.
The asp-name argument specifies the name of this ASP. The ip-precedence keyword sets the precedence
and ToS field. The remote-ip_address argument specifies the IP address of the remote end-point (the
address of MGC, for example). The number argument can be any IP precedence bits in the range 1
through 255.
The no form of the command results in precedence bits not being explicitly set by SCTP. The default is
to set all bits in the ToS field to zero by SCTP.
In the case of a hot-standby PGW pair, from the GW perspective there is usually be one ASP active and
another in the INACTIVE state. The ASP_UP message is used to bring the ASP state on the GW to the
INACTIVE state, followed by the ASPTM message, ASP_ACTIVE to ready the IUA link for data
exchange (eventually the QPTM Establish Request message actually initiates the start of the D channel
for the given interface). In the event that the GW detects a failure on the active ASP, it can send a NTFY
message to the standby ASP to request that it become active.
Examples
An ASP can be viewed as a local representation of an SCTP association since it specifies a remote end
point that will be in communication with an AS local end point. An ASP is defined for a given AS. For
example, the following configuration defines a remote signaling controller asp-name at two IP addresses
for AS as-name. The remote SCTP port number is 2577:
AS as-name 10.4.8.69, 10.4.9.69 2477
ASP asp-name AS as-name 10.4.8.68 10.4.9.68 2577
Multiple ASPs can be defined for a single AS for the purpose of redundancy, but only one ASP can be
active. The ASPs are inactive and only become active after fail-over.
In the Cisco MGC solution, a signaling controller is always the client that initiates the association with
a gateway. During the initiation phase, you can request outbound and inbound stream numbers, but the
gateway only allows a number that is at least one digit higher than the number of interfaces (T1/E1)
allowed for the platform.
The following shows options for this command:
Router(config-iua)# asp asp-name ip-precedence 10.1.2.345 ?
<0-7>
default
specify precedence level (0 - 7)
use default value of IP precedence for this address
Router(config-iua)# asp asp-name sctp-keep ?
A.B.C.D
specify the IP address to enable/disable keep alives
Router(config-iua)# asp asp-name sctp-keep 10.1.2.234 ?
<1000-60000>
specify keep alive interval (in milliseconds)
Router(config-iua)# asp asp-name sctp-max-ass ?
<2-20>
default
specify maximum associations
use default value of max associations for this address
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asp
Router(config-iua)# asp asp-name sctp-path-retran ?
<2-10>
default
specify maximum path retransmissions
use default value of max path retrans for this address
Router(config-iua)# asp asp-name sctp-t3-time ?
<300-60000>
default
Related Commands
specify T3 retransmission timeout (in milliseconds)
use default value of T3 for this address
Command
Description
as
Defines an application server (AS) for backhaul.
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atm scramble-enable
atm scramble-enable
To enable scrambling on E1 links, use the atm scramble-enable command in interface configuration
mode. To disable scrambling, use the no form of this command.
atm scramble-enable
no atm scramble-enable
Syntax Description
This command has no arguments or keywords.
Defaults
By default, payload scrambling is set off.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM interface configuration on the
Cisco MC3810 multiservice concentrator.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Usage Guidelines
Enable scrambling on E1 links only. On T1 links, the default B8ZS line encoding normally ensures
sufficient reliability. Scrambling improves data reliability on E1 links by randomizing the ATM cell
payload frames to avoid continuous nonvariable bit patterns and to improve the efficiency of the ATM
cell delineation algorithms.
The scrambling setting must match that of the far end.
Examples
On a Cisco MC3810, the following example shows how to set the ATM0 E1 link to scramble payload:
interface atm0
atm scramble-enable
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atm video aesa
atm video aesa
To set the unique ATM end-station address (AESA) for an ATM video interface that is using switched
virtual circuit (SVC) mode, use the atm video aesa command in ATM interface configuration mode. To
remove any configured address for the interface, use the no form of this command.
atm video aesa [default | esi-address]
no atm video aesa
Syntax Description
default
(Optional) Automatically creates a network service access point (NSAP)
address for the interface, based on a prefix from the ATM switch (26
hexadecimal characters), the MAC address (12 hexadecimal characters)
as the end station identifier (ESI), and a selector byte (two hexadecimal
characters).
esi-address
(Optional) Defines the 12 hexadecimal characters used as the ESI. The
ATM switch provides the prefix (26 hexadecimal characters), and the
video selector byte provides the remaining two hexadecimal characters.
Defaults
default
Command Modes
ATM Interface configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM interface configuration on the
Cisco MC3810 multiservice concentrator.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Usage Guidelines
You cannot specify the ATM interface NSAP address in its entirety. The system creates either all of the
address or part of it, depending on how you use this command.
Examples
On a Cisco MC3810 multiservice concentrator, the following example shows the ATM interface NSAP
address set automatically:
interface atm0
atm video aesa default
On a Cisco MC3810 multiservice concentrator, the following example shows the ATM interface NSAP
address set to a specific ESI value:
interface atm0/1
atm video aesa 444444444444
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atm video aesa
Related Commands
Command
Description
show atm video-voice
address
Displays the NSAP address for the ATM interface.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
audio-prompt load
audio-prompt load
To initiate loading the selected audio file (.au), the file that contains the announcement prompt for the
caller from Flash memory into RAM, use the audio-prompt load command in privileged EXEC mode.
audio-prompt load name
Syntax Description
name
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(6)NA2
This command was introduced.
Usage Guidelines
Indicates the location of the audio file that you want to have loaded from
memory, Flash memory, or an FTP server.
The first time the interactive voice response (IVR) application plays a prompt, it reads it from the URL
(or the specified location for the .au file, such as Flash or TFP) into RAM. Then it plays the script from
RAM. An example of the sequence of events is as follows:
•
When the first caller is asked to enter the account and personal identification numbers (PINs), the
enter_account.au and enter_pin.au files are loaded into RAM from Flash memory.
•
When the next call comes in, these prompts are played from the RAM copy.
•
If all callers enter valid account numbers and PINs, then the auth_failed.au file is not loaded from
Flash memory into RAM memory.
The router will load the audio file only when the script initially plays that prompt after the router restarts.
If the audio file is changed, you must run this EXEC command to reread the file. This will generate an
error message if the file is not accessible or if there is a format error.
Note
Examples
With Cisco IOS Release 11.3(6)NA2, the URL pointer refers to the directory where Flash
memory is stored.
The following example shows how to load the enter_pin.au audio file from Flash memory into RAM:
audio-prompt load flash:enter_pin.au
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auto-cut-through
auto-cut-through
To enable call completion when a PBX does not provide an M-lead response, use the auto-cut-through
command in voice-port configuration mode. To disable the auto-cut-through operation, use the no form
of this command.
auto-cut-through
no auto-cut-through
Syntax Description
This command has no arguments or keywords.
Defaults
Auto-cut-through is enabled.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series
routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The auto-cut-through command applies to ear and mouth (E&M) voice ports only.
Examples
The following example shows enabling of call completion on a Cisco MC3810 multiservice concentrator
when a PBX does not provide an M-lead response:
voice-port 1/1
auto-cut-through
The following example shows enabling of call completion on a Cisco 2600 or 3600 router when a PBX
does not provide an M-lead response:
voice-port 1/0/0
auto-cut-through
Related Commands
Command
Description
show voice port
Displays voice port configuration information.
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backhaul-session-manager
backhaul-session-manager
To enter backhaul session manager configuration mode, use the backhaul-session-manager command
in global configuration mode.
backhaul-session-manager
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Usage Guidelines
Use the backhaul-session-manager command to enter the backhaul session manager configuration
mode. Table 3 lists the backhaul session manager configuration mode commands:
Table 3
Backhaul Session Manager Configuration Mode Commands
Command
Description
group
Creates a session-group and associates it with a
specified session-set.
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative
acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments
that are received before an EACK is sent.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer retransmit
Configures retransmission timeout.
group timer transfer
Configures state transfer timeout.
session group
Associates a transport session with a specified
session-group.
set
Creates a fault-tolerant or non-fault-tolerant
session-set with the client or server option.
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backhaul-session-manager
Examples
The following example shows how to enter backhaul-session-manager configuration mode:
Router(config)# backhaul-session-manager
Router(config-bsm)#
Related Commands
Command
Description
clear backhaul-session-manager
group
Resets the stastistics or traffic counters for a
specified session-group.
clear rudpv1 statistics
Clears the RUDP statistics and failure counters.
group
Creates a session-group and associates it with a
specified session-set.
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative
acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments
that are received before an EACK is sent.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer retransmit
Configures retransmission timeout.
group timer transfer
Configures state transfer timeout.
isdn bind-l3
Configures the ISDN serial interface for backhaul.
session group
Associates a transport session with a specified
session-group.
set
Creates a fault-tolerant or non-fault-tolerant
session-set with the client or server option.
show backhaul-session-manager
group
Displays status, statistics, or configuration of a
specified or all session-groups.
show backhaul-session-manager
session
Displays status, statistics, or configuration of
sessions.
show backhaul-session-manager set
Displays session-groups associated with a specific
or all session-sets.
show rudpv1
Displays RUDP statistics.
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bandwidth
bandwidth
To specify the maximum aggregate bandwidth for H.323 traffic, use the bandwidth command in
gatekeeper configuration mode. To remove the maximum aggregate bandwidth value, use the no form of
this command.
bandwidth {interzone | total | session} {default | zone zone-name} bandwidth-size
no bandwidth {interzone | total | session} {default | zone zone-name} bandwidth-size
Syntax Description
interzone
Specifies the maximum bandwidth for H.323 traffic between one zone and
another zone.
total
Specifies the maximum bandwidth for H.323 traffic within a zone and
between zones (intrazone and interzone).
session
Specifies the maximum bandwidth allowed for a single session in a specific
zone or in all zones.
default
Specifies the maximum bandwidth for all applicable zones, depending on the
keyword with which it is used.
zone
Specifies a particular zone.
zone-name
Names the particular zone.
bandwidth-size
Maximum bandwidth. For interzone and total, the range is from 1 to
10,000,000 kbps. For session, the range is from 1 to 5000 kbps.
Defaults
None
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 series and 3600 series
routers and on the AS5300 universal access server.
12.1(5)T
The bandwidth command replaced the zone bw command.
Usage Guidelines
The functionality of this command in previous Cisco IOS software releases was enabled by using the
zone bw command.
To specifiy maximum bandwidth for traffic between one zone and any other zone, use the default
keyword with the interzone keyword.
To specify maximum bandwidth for traffic within one zone or for traffic between that zone and another
zone (interzone and intrazone), use the default keyword with the total keyword.
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bandwidth
To specify maximum bandwidth for a single session within a specific zone, use the zone keyword with
the session keyword.
To specify maximum bandwidth for a single session within any zone, use the default keyword with the
session keyword.
Examples
The following example configures the default maximum bandwidth for traffic between one zone and
another zone to 5000 kbps:
gatekeeper
bandwidth interzone default 5000
The following example configures the default maximum bandwidth for all zones to 5000 kbps:
gatekeeper
bandwidth total default 5000
The following example configures the default maximum bandwidth for a single session within any zone
to 2000 kbps:
gatekeeper
bandwidth session default 2000
The following example configures the default maximum bandwidth for a single session with a specific
zone to 1000 kbps:
gatekeeper
bandwidth session zone denver 1000
Related Commands
Command
Description
bandwidth remote
Specifies the total bandwidth for H.323 traffic between this gatekeeper and
any other gatekeeper.
h323 interface
Defines on which port the proxy will listen.
h323 t120
Enables the T.120 capabilities on your router and specifies bypass or proxy
mode.
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bandwidth remote
bandwidth remote
To specify the total bandwidth for H.323 traffic between this gatekeeper and any other gatekeeper, use
the bandwidth remote command in gatekeeper configuration mode. To disable the total bandwidth
specified, use the no form of this command.
bandwidth remote bandwidth-size
no bandwidth remote bandwidth-size
Syntax Description
bandwidth-size
Defaults
None
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(5)T
This command was introduced on the Cisco 2600, 3600, and 7200 series
routers and on the MC3810 multiservice concentrator.
Maximum bandwidth. The range is from 1 to 10,000,000 kbps.
Usage Guidelines
The functionality of this command in previous Cisco IOS software releases was enabled by using the
zone gatekeeper command.
Examples
The following example configures the remote maximum bandwidth to 100,000 kbps:
gatekeeper
bandwidth remote 100000
Related Commands
Command
Description
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic from a zone
to another zone, within a zone, or for a session in a zone.
h323 interface
Defines which port the proxy will listen on.
h323 t120
Enables the T.120 capabilities on your router and specifies bypass or proxy
mode.
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battery-reversal
battery-reversal
To specify battery polarity reversal on a Foreign Exchange Office (FXO) or Foreign Exchange Station
(FXS) port, use the battery-reversal command in voice-port configuration mode. To disable battery
reversal, use the no form of this command.
battery-reversal
no battery-reversal
Syntax Description
This command has no arguments or keywords.
Defaults
Battery reversal is enabled.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco 2600 and 3600 series
routers and on the MC3810 multiservice concentrator.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The battery-reversal command applies to FXO and FXS voice ports. On Cisco 2600 and 3600 series
routers, only analog voice ports in VIC-2FXO-M1 and VIC-2FXO-M2 voice interface cards are able to
detect battery reversal; analog voice ports in VIC-2FXO and VIC-2FXO-EU voice interface cards do not
detect battery reversal. On digital voice ports, battery reversal is supported only on E1 Mercury
Exchange Limited Channel Associated Signaling (MELCAS); it is not supported in T1 channel
associated signaling (CAS) or E1 CAS.
FXS ports normally reverse battery upon call connection. If an FXS port is connected to an FXO port
that does not support battery reversal detection, you can use the no battery-reversal command on the
FXS port to prevent unexpected behavior.
FXO ports in loopstart mode normally disconnect calls when they detect a second battery reversal (back
to normal). You can use the no battery-reversal command on FXO ports to disable this action.
The battery-reversal command restores voice ports to their default battery-reversal operation.
Examples
The following example disables battery reversal on voice port 1/1 on a Cisco MC3810:
voice-port 1/1
no battery-reversal
The following example disables battery reversal on voice port 1/0/0 on a Cisco 2600 or 3600 series
router:
voice-port 1/0/0
no battery-reversal
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battery-reversal
Related Commands
Command
Description
show voice port
Displays voice port configuration information.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
block-caller
block-caller
To configure call blocking on caller ID, use the block-caller command in dial peer voice configuration
mode. To disable call blocking on caller ID, use the no form of this command.
block-caller number
no block-caller number
Syntax Description
number
Defaults
Call blocking is disabled; the router does not block any calls for any listed directory numbers (LDNs)
based on caller ID numbers.
Command Modes
Dial peer voice configuration
Command History
Release
Modification
12.1.(2)XF
This command was introduced on the Cisco 800 series routers.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Specifies the telephone number to block. You can use a period (.) as a digit
wildcard. For example, the command block-caller 5.51234 blocks all
numbers beginning with the digit 5, followed by any digit, and then
sequentially followed by the digits 5, 1, 2, 3, and 4.
This command is available on Cisco 800 series routers that have plain old telephone service (POTS)
ports. For each dial peer, you can enter up to ten caller ID numbers to block. The routers do not accept
additional caller ID numbers if ten numbers are already present. In that case, a number must be removed
before another caller ID number can be added for blocking.
If you do not specify the block-caller command for a local directory, all voice calls to that local directory
are accepted. If you specify the block-caller command for a local directory, the router verifies that the
incoming calling-party number does not match any caller ID numbers in that local directory before
processing or accepting the voice call. Each specified caller ID number and incoming calling-party
number is compared from right to left, up to the number of digits in the specified caller ID number or
incoming calling-party number, whichever has fewer digits.
This command is effective only if you subscribe to caller ID service. If you enable call blocking on caller
ID without subscribing to the caller ID service, the routers do not perform the verification process on
calling-party numbers and do not block any calls.
Examples
The following example configures a router to block calls from a caller whose caller ID number is
408-555-1234.
dial-peer voice 1 pots
block-caller 4085551234
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block-caller
Related Commands
Command
Description
caller-id
Identifies incoming calls with caller ID.
debug pots csm csm
Activates events from which an application can determine and display
the status and progress of calls to and from POTS ports.
isdn i-number
Configures several terminal devices to use one subscriber line.
pots call-waiting
Enables local call waiting on a router.
registered-caller ring
Configures the Nariwake service registered caller ring cadence.
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busyout forced
busyout forced
To force a voice port into the busyout state, use the busyout forced command in voice-port configuration
mode. To remove the voice port from the busyout state, use the no form of this command.
busyout forced
no busyout forced
Syntax Description
This command has no arguments or keywords.
Defaults
The voice-port is not in the busyout state.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series
routers. On the Cisco MC3810, the voice-port busyout command was
eliminated in favor of this command.
12.1(2)T
The modifications in Cisco IOS Release 12.0(7)XK were integrated
into Cisco IOS Release 12.1(2)T.
Usage Guidelines
If a voice port is in the forced busyout state, only the no busyout forced command can restore the voice
port to service.
To avoid conflicting command-line interface (CLI) commands, do not use the busyout forced command
and the ds0 busyout command on the same controller.
Examples
The following example forces analog voice port 1/1 on a Cisco MC3810 multiservice concentrator into
the busyout state:
voice-port 1/1
busyout forced
The following example forces digital voice port 0:8 on a Cisco MC3810 multiservice concentrator into
the busyout state:
voice-port 0:8
busyout forced
The following example forces analog voice port 3/1/1 on a Cisco 3600 router into the busyout state:
voice-port 3/1/1
busyout forced
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busyout forced
The following example forces digital voice port 0/0:12 on a Cisco 3600 router into the busyout state:
voice-port 0/0:12
busyout forced
Related Commands
Command
Description
busyout-monitor interface
Configures a voice port to monitor a serial interface for events that
would trigger a voice-port busyout.
busyout seize
Changes the busyout seize procedure for a voice port.
show voice busyout
Displays information about the voice busyout state.
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busyout monitor
busyout monitor
To place a voice port into the busyout monitor state, enter the busyout monitor command in voice-port
configuration mode. To remove the busyout monitor state from the voice port, use the no form of this
command.
busyout monitor {serial interface-number | ethernet interface-number} [in-service]
no busyout monitor {serial interface-number | ethernet interface-number}
Syntax Description
serial
Specifies monitoring of a serial interface. More than one interface can be
entered for a voice port.
ethernet
Specifies monitoring of an Ethernet interface. More than one interface can
be entered for a voice port.
interface-number
Identifies an interface to be monitored for the voice port busyout function.
Interface choices include serial port, serial port subinterface, Ethernet port,
and ATM interface.
in-service
(Optional) Configures the voice port to be busied out when any monitored
interface comes into service (its state changes to up). If the keyword is not
entered, the voice port is busied out when all monitored interfaces go out
of service (their state changes to down).
Defaults
The voice port does not monitor any interfaces.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(5)XE
This command was implemented on the Cisco 7200 series routers.
12.0(5)XK
This command was implemented on the Cisco 2600 and 3600 series
routers.
12.0(7)T
The Cisco 2600 and 3600 series router implementation was integrated into
Cisco IOS Release 12.0(7)T.
12.0(7)XK
The ability to monitor an Ethernet port was introduced and the in-service
keyword was added. The serial keyword was first supported on the Cisco
2600 and 3600 series routers.
12.1(1)T
The implementation of this command on the Cisco 7200 series routers was
integrated into Cisco IOS Release 12.1(1)T.
12.1(2)T
The serial and ethernet keywords were added, the in-service keyword was
integrated into Cisco IOS Release 12.1(2)T, and the interface number
argument was changed to go with the serial and Ethernet keywords.
12.1(3)T
The interface keyword was removed.
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busyout monitor
Usage Guidelines
When you place a voice port in the busyout monitor state, the voice port monitors the specified interface
and enters the busyout state when the interface is down. This down state forces the rerouting of calls.
The command monitors only the up or down status of an interface—not end-to-end TCP/IP connectivity.
When an interface is operational, a busied-out voice port returns to its normal state.
This feature can monitor LAN, WAN, and virtual interfaces as well as subinterfaces.
The Cisco 2600 and 3600 series routers and the MC3810 multiservice concentrator support ATM
interfaces. To monitor an ATM interface, enter ATM and the interface number.
A voice port can monitor multiple interfaces at the same time. To configure a voice port to monitor
multiple interfaces, reenter the busyout monitor command for each additional interface to be monitored.
If you specify more than one monitored interface for a voice port, all the monitored interfaces must be
down to trigger busyout on the voice port.
You can combine in-service and out-of-service monitoring on a voice port. The following rule describes
the actions if monitored interfaces change state.
A voice port is busied out if either of the following occurs:
Examples
•
Any interface monitored for coming into service comes up.
•
All interfaces monitored for going out of service go down.
The following example shows configuration of analog voice port 1/1 on a Cisco MC3810 multiservice
concentrator to busyout if serial ports 1 and 0:0 both go out of service:
voice-port 1/1
busyout monitor serial 0:0
busyout monitor serial 1
The following example shows configuration of analog voice port 1/2 on a Cisco MC3810 multiservice
concentrator to busy out if serial port 0 or 1 comes into service:
voice-port 1/2
busyout monitor serial 0 in-service
busyout monitor serial 1 in-service
The following example shows configuration of digital voice port 1/2/2 on a Cisco 3600 series router to
busy out if serial port 0 goes out of service:
voice-port 1/2/2
busyout monitor serial 0
The following example shows configuration of digital voice port 0:6 on a Cisco MC3810 multiservice
concentrator to busy out if both Ethernet port 0 and serial port 0 go out of service:
voice-port 0:6
busyout monitor ethernet 0
busyout monitor serial 0
The following example shows configuration of the voice port to monitor two serial interfaces and an
Ethernet interface. When all these interfaces are down, the voice port is busied out. When at least one
interface is operating, the voice port is put back into a normal state.
voice-port 3/0:0
busyout monitor ethernet 0/0
busyout monitor serial 1/0
busyout monitor serial 2/0
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busyout monitor
Related Commands
Command
Description
busyout forced
Forces a voice port into the busyout state.
busyout monitor probe
Configures a voice port to enter the busyout state if a Service
Assurance Agent (SAA) probe signal returned from a remote,
IP-addressable interface crosses a specified delay or loss threshold.
busyout seize
Changes the busyout seize procedure for a voice port.
show voice busyout
Displays information about the voice busyout state.
voice-port busyout
Places all voice ports associated with a serial or ATM interface into a
busyout state.
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busyout monitor probe
busyout monitor probe
To configure a voice port to enter the busyout state if a Service Assurance Agent (SAA) probe signal
returned from a remote, IP-addressable interface crosses a specified delay or loss threshold, use the
busyout monitor probe command in voice-port configuration mode. To configure a voice port not to
monitor SAA probe signals, use the no form of this command.
busyout monitor probe ip-address [codec codec-type] [icpif number | loss percent delay
milliseconds]
no busyout monitor probe ip-address
Syntax Description
ip-address
The IP address of a target interface for the SAA probe signal.
codec
(Optional) Configures the profile of the SAA probe signal to mimic the packet
size and interval of a specific codec type.
codec-type
(Optional) The codec type for the SAA probe signal.
Available options are as follows:
Defaults
•
g711a—G.711 A-law
•
g711u—G.711 U-law (the default)
•
g729—G.729
•
g729a—G.729
icpif
(Optional) Configures the busyout monitor probe to use an
Impairment/Calculated Planning Impairment Factor (ICPIF) loss/delay busyout
threshold, in accordance with ITU-T G.113. The ICPIF numbers represent
predefined combinations of loss and delay.
number
(Optional) The ICPIF threshold for initiating a busyout. The range is from 0 to 30.
Lower numbers are equivalent to lower loss and delay thresholds.
loss
(Optional) Configures the percentage-of-packets-lost threshold for initiating a
busyout.
percent
(Optional) The loss value (expressed as a percentage) for initiating a busyout. The
range is from 1 to 100.
delay
(Optional) Configures the average packet delay threshold for initiating a busyout.
milliseconds
(Optional) The delay threshold, in milliseconds, for initiating a busyout. The
range is from 1 to 2147483647.
If the busyout monitor probe command is not entered, the voice port does not monitor SAA probe
signals.
If the busyout monitor probe command is entered with no optional keywords or arguments, the default
codec type is G.711 alaw, and the default loss and delay thresholds are the threshold values configured
with the pstn fallback command.
Command Modes
Voice-port configuration
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busyout monitor probe
Command History
Usage Guidelines
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600 series and on the Cisco
MC3810 multiservice concentrator.
A voice port can monitor multiple interfaces at the same time. To configure a voice port to monitor
multiple interfaces, enter the busyout monitor probe command for each additional interface to be
monitored.
The busyout monitor probe command is effective only if the call fallback function is enabled on this
router and the SAA responder is enabled on the target router.
The SAA probe is transmitted periodically with a period determined by the call fallback function.
Refer to the PSTN Fallback feature module for Cisco IOS Release 12.1(3)T for details of the call fallback
function and ICPIF values.
Lower thresholds of ICPIF, loss, and delay result in earlier busyout when the link deteriorates, thereby
raising the voice minimum quality level. Higher thresholds prevent busyout until loss and delay are
greater, allowing transmission of lower-quality voice.
Caution
Examples
If thresholds are set too low, the link can alternate between in-service and out-of-service states,
causing repeated interruptions of traffic.
The following example configures analog voice port 1/1 on a Cisco MC3810 multiservice concentrator
to use an SAA probe with a G.711alaw profile to probe the link to two remote interfaces that have IP
addresses and to busy out the voice port. Both links have a loss exceeding 25 percent or a packet delay
of more than 1.5 seconds.
voice-port 1/1
busyout monitor probe 209.165.202.128 codec g711a loss 25 delay 1500
busyout monitor probe 209.165.202.129 codec g711a loss 25 delay 1500
Related Commands
Command
Description
busyout monitor
Places a voice port into the busyout monitor state.
pstn fallback
Forces a voice port into the busyout state.
show voice busyout
Displays information about the voice busyout state.
voice class busyout
Creates a voice class for local voice busyout functions.
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busyout seize
busyout seize
To change the busyout action for a Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS)
voice port, use the busyout seize command in voice-port configuration mode. To restore the default
busyout action, use the no form of this command.
busyout seize {ignore | repeat}
no busyout seize
Syntax Description
ignore
Specifies the type of ignore procedure, depending on the type of voice port
signaling. See Table 4 for more information.
repeat
Specifies the type of repeat procedure, depending on the type of voice port
signaling. See Table 4 for more information.
Defaults
See Table 4 for the default actions for different voice ports and signaling types.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on Cisco 2600 and 3600 series routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The busyout seize command is valid for both analog and digital voice ports. On digital voice ports, the
busyout actions are valid whether the busyout results from a voice-port busyout event or from the
ds0-busyout command.
The voice port returns to an idle state when the event that triggered the busyout disappears.
Table 4 describes the busyout actions for the busyout seize settings on each voice port type.
The busyout action for E and M voice ports is to seize the far end by setting lead busy.
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busyout seize
Table 4
Busyout Seize Actions for Voice Ports
Voice Port
Signaling Type
Procedure Setting
(busyout-option command)
FXS loop start
Default
Removes the power from the loop. For analog voice ports, this is equivalent
to removing the ground from the tip lead. For digital voice ports, the port
will generate the bit pattern equivalent to removing the ground from the tip
lead, or it will busy out if the bit pattern exists.
FXS loop start
Ignore
Ignores the ground on the ring lead.
FXS ground start Default
FXS ground start Ignore
FXS ground start Repeat
FXO loop start
Default
FXO loop start
Ignore
FXO loop start
Repeat
FXO ground start Default
FXO ground start Ignore
FXO ground start Repeat
Busyout Actions
Grounds the tip lead and stays at this state.
1.
Leaves the tip lead open.
2.
Ignores the ground on the ring lead.
1.
Grounds the tip lead.
2.
Waits for the far end to close the loop.
3.
The far end closes the loop.
4.
If the far end then opens the loop, FXS removes the ground from the
tip lead.
5.
FXS waits for several seconds before returning to Step 1.
Closes the loop and stays at this state.
1.
Leaves the loop open.
2.
Ignores the ringing current on the ring level.
1.
Closes the loop.
2.
After the detected far end starts the power denial procedure, FXO
opens the loop.
3.
After the detected far end has completed the power denial procedure,
FXO waits for several seconds before returning to Step 1.
Grounds the tip lead.
1.
Leaves the loop open.
2.
Ignores the running current on the ring lead, or the ground current on
the tip lead.
1.
Grounds the ring lead.
2.
Removes the ground from the ring lead and closes the loop after the
detected far end grounds the tip lead.
3.
When the detected far end removes the ground from tip lead, FXO
opens the loop.
4.
FXO waits for several seconds before returning to Step 1.
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busyout seize
Examples
The following example shows configuration of analog voice port 1/1 to perform the ignore actions when
busied out:
voice-port 1/1
busyout seize ignore
The following example shows configuration of digital voice port 0:2 to perform the repeat actions when
busied out:
voice-port 0:2
busyout seize repeat
Related Commands
Command
Description
busyout forced
Forces a voice port into the busyout state.
busyout-monitor interface
Configures a voice port to monitor an interface for events that would
trigger a voice port busyout.
ds0 busyout
Forces a DS0 time slot on a controller into the busyout state.
show voice busyout
Displays information about the voice busyout state.
voice-port busyout
Places all voice ports associated with a serial or ATM interface into a
busyout state.
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cac master
cac master
To configure the call admission control (CAC) operation as master, enter the cac master command in
voice-service configuration mode. To restore the default value, use the no form of this command.
cac master
no cac master
Syntax Description
No arguments or keywords
Defaults
The Cisco MC3810 multiservice concentrator is enabled as a CAC slave.
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(1)XA
The command was introduced for the Cisco MC3810 multiservice
concentrator.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
You should configure the Cisco MC3810 multiservice concentrators at opposite ends of an AAL2 trunk
for the opposite CAC operation—master at one end and slave at the other end.
A Cisco MC3810 multiservice concentrator configured as a master always performs CAC during
fax/modem upspeed. A Cisco MC3810 multiservice concentrator configured as a slave sends a request
for CAC to the CAC master.
Examples
The following example shows configuration of the CAC operation of a Cisco MC3810 multiservice
concentrator as master:
voice service voatm
session protocol aal2
cac master
The following example shows the CAC operation of a Cisco MC3810 multiservice concentrator being
returned to slave:
voice service voatm
session protocol aal2
no cac master
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cadence-list
cadence-list
To specify a tone cadence pattern to be detected, use the cadence-list command in voice-class
configuration mode. To delete a cadence pattern, use the no form of this command.
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]
no cadence-list cadence-id
Syntax Description
cadence-id
A tag to identify this cadence list. The range is from 1 to 10.
cycle-1-on-time
The tone duration for the first cycle of the cadence pattern. The range is from 0
to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-1-off-time
The silence duration for the first cycle of the cadence pattern. The range is from
0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-2-on-time
(Optional) The tone duration for the second cycle of the cadence pattern. The
range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-2-off-time
(Optional) The silence duration for the second cycle of the cadence pattern. The
range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-3-on-time
(Optional) The tone duration for the third cycle of the cadence pattern. The
range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-3-off-time
(Optional) The silence duration for the third cycle of the cadence pattern. The
range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-4-on-time
(Optional) The tone duration for the fourth cycle of the cadence pattern. The
range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-4-off-time
(Optional) The silence duration for the fourth cycle of the cadence pattern. The
range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
Defaults
No cadence pattern is configured.
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and Cisco 3600 series routers
and on the Cisco MC3810 multiservice concentrator.
Usage Guidelines
A cadence list enables the router to match a complex tone pattern from a PBX or public switched
telephone network (PSTN). A tone is detected if it matches any configured cadence list. You can create
up to ten cadence lists, enabling the router to detect up to ten different tone patterns. If the tone to be
detected consists of only one on-off cycle, you can configure this in either of two ways:
•
Create a cadence list using only the cycle-1-on-time and cycle-1-off-time variables.
•
Use the cadence-max-off-time and cadence-min-on-time commands.
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cadence-list
You must also configure the times of the cadence-max-off-time and cadence-min-on-time commands
to be compatible with the on and off times specified by the cadence-list command. The time of the
cadence-max-off-time must be equal to or greater than the longest off-time in the cadence list; the
cadence-min-on-time must be equal to or less than the shortest on-time in the cadence list.
Examples
The following example shows configuration of cadence list 1 with three on/off cycles and cadence list 2
with two on/off cycles for voice class 100:
voice class dualtone 100
cadence-list 1 100 100 300 300 100 200
cadence-list 2 100 200 100 400
Related Commands
Command
Description
cadence-max-off-time
Specifies the maximum off duration for detection of a tone.
cadence-min-on-time
Specifies the minimum on duration for detection of a tone.
voice class dualtone
Creates a voice class for FXO tone detection parameters.
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cadence-max-off-time
cadence-max-off-time
To specify the maximum off duration for detection of a tone, use the cadence-max-off-time command
in voice-class configuration mode. To restore the default, use the no form of this command.
cadence-max-off-time time
no cadence-max-off-time
Syntax Description
time
Defaults
No cadence maximum off time is configured.
Command Modes
Voice-class configuration
Related Commands
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600 series routers and on
the Cisco MC3810 multiservice concentrator.
The maximum off time of a tone that can be detected, in 10-millisecond
increments. The range is from 0 to 5000 (0 milliseconds to 50 seconds). The
default is 0.
Usage Guidelines
You must specify a time value greater than the off time of the tone to be detected. You must specify a
time value greater than 0 to enable detection of a tone. With the default (0), the router will detect only a
continuous tone.
Examples
The following example shows configuration of a maximum off duration of 20 seconds for voice class
100:
voice class dualtone 100
cadence-max-off-time 2000
Related Commands
Command
Description
cadence-min-on-time
Specifies the minimum on duration for detection of a tone.
cadence-variation
Specifies the cadence variation time allowed for detection of a tone.
voice class dualtone
Creates a voice class for FXO tone detection parameters.
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cadence-min-on-time
cadence-min-on-time
To specify the minimum on duration for detection of a tone, use the cadence-min-on-time command in
voice-class configuration mode. To restore the default, use the no form of this command.
cadence-min-on-time time
no cadence-min-on-time
Syntax Description
time
Defaults
No cadence minimum on time is configured.
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600 series routers and on
the Cisco MC3810 multiservice concentrator.
The minimum on time of a tone that can be detected, in 10-millisecond
increments. The range is from from 0 to 100 (0 milliseconds to 1 seconds). The
default is 0.
Usage Guidelines
You must specify a time value shorter than the on time of the tone to be detected. With the default (0), a
tone of any length will be detected.
Examples
The following example shows configuration of a minimum on duration of 30 milliseconds for voice class
100:
voice class dualtone 100
cadence-min-on-time 3
Related Commands
Command
Description
cadence-max-off-time
Specifies the maximum off duration for detection of a tone.
cadence-variation
Specifies the cadence variation time allowed for detection of a tone.
voice class dualtone
Creates a voice class for Foreign Exchange Office (FXO) tone detection
parameters.
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cadence-variation
cadence-variation
To specify the cadence variation time allowed for detection of a tone, use the cadence-variation
command in voice-class configuration mode. To restore the default cadence variation time, use the no
form of this command.
cadence-variation time
no cadence-variation
Syntax Description
time
Defaults
Zero for zero millisecond cadence variation allowed
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600 series routers and on
the Cisco MC3810 multiservice concentrator.
The maximum time by which the tone onset can vary from the specified onset
time and still be detected, in 10-millisecond increments. The range is from 0 to
200 (0 milliseconds to 2 seconds). The default is 0.
Usage Guidelines
You should specify a time value greater than the cadence variation of the tone to be detected. With the
default of 0, only those tones that match the configured cadence will be detected.
Examples
The following example shows configuration of a cadence variation time of 30 milliseconds for voice
class 100:
voice class dualtone 100
cadence-variation 3
Related Commands
Command
Description
cadence-max-off-time
Specifies the maximum off duration for detection of a tone.
cadence-min-on-time
Specifies the minimum on duration for detection of a tone.
voice class dualtone
Creates a voice class for FXO tone detection parameters.
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call application cache reload time
call application cache reload time
To configure the router to reload the Media Gateway Control Protocol (MGCP) scripts from cache on a
regular interval, use the call application cache reload time command in global configuration mode. To
set the value to the default, use the no form of this command.
call application cache reload time bg-minutes
no call application cache reload time
Syntax Description
bg-minutes
Specifies the number of minutes after which the background process
is awakened. This background process checks the time elapsed since
the script was last used and whether the script is current:
•
If the script has not been used in the last “unload time,” it will
unload the script and quit. The unload time is not configurable.
•
If the script has been used, the background process will load the
script from the URL. It compares the scripts, and if they do not
match, it begins using the new script for new calls.
Defaults
30 minutes
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300 universal access
server.
Examples
The following example displays the call application cache reload time command configured to specify
30 minutes before a background process is awakened:
call application cache reload time 30
Related Commands
Command
Description
call application voice load
Allows reload of an application that was loaded via the MGCP
scripting package.
show call application voice
Displays all TCL or MGCP scripts that are loaded.
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call application voice
call application voice
To create an application and to indicate the location of the corresponding Tool Command Language
(TCL) files that implement this application, use the call application voice command in global
configuration mode. To remove the defined application and all configured parameters associated with it,
use the no form of this command.
call application voice application-name location {word}
no call application voice application-name location {word}
Syntax Description
application-name
Character string that defines the name of the application.
location
Location of the TCL file in URL format. Valid storage locations are TFTP,
FTP, and Flash.
word
Text string that defines an attribute-value pair specified by the TCL script and
understood by the RADIUS server.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco AS5300 universal access server.
12.1(3)T
The word argument was added for attribute-value (AV) pairs.
Usage Guidelines
Note
Examples
Use this command when configuring interactive voice response (IVR) or one of the IVR-related features
(such as Debit Card) to define the name of an application and to identify the location of the TCL script
associated with this application.
The command no call application voice application-name removes the entire application and all
parameters, if configured.
This example shows how to define the application “prepaid” and the TFTP server location of the
associated TCL script:
call application voice prepaid tftp://keyer/debitcard.tcl
The following is an example of AV pair configuration:
set avsend(h323-ivr-out,)) “payphone:true”
set avsend(323-ivr-out,1) “creditTime:3400”
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call application voice
The AV pair (after the array is defined, as in the prior example) must be sent to the server, using the
authentication, authorization, and accounting (AAA) authenticate or AAA authorize verbs as follows:
aaa authenticate $account $password $avsend
The script would use this AV pair whenever it is needed to convey information to the RADIUS server
that cannot be represented by the standard vendor-specific attributes (VSAs).
Related Commands
Command
Description
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reloads the designated TCL script.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application.
call application voice warning-time
Defines the number of seconds of warning that a user
receives before the allowed calling time runs out for the
designated application.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice access-method
call application voice access-method
To specify the access method for two-stage dialing for the designated application, use the call
application voice access-method command in global configuration mode. To restore default values for
this command, use the no form of this command.
call application voice application-name access-method {prompt-user | redialer}
no call application voice application-name access-method
Syntax Description
application-name
The name of the application.
prompt-user
Specifies that no direct inward dialing (DID) is set in the incoming plain old
telephone service (POTS) dial peer and that a Tool Command Language (TCL)
script in the incoming POTS dial peer will be used for two-stage dialing.
redialer
Specifies that no DID is set in the incoming POTS dial peer and that the redialer
device will be used for two-stage dialing.
Defaults
Prompt-user when DID is not set in the dial peer
Command Modes
Global configuration mode
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
Use the call application voice access-method command to specify the access method for two-stage
dialing when DID is disabled in the POTS dial peer.
Examples
The following example specifies prompt-user as the access method for two-stage dialing for the
app_libretto_onramp9 IVR application:
call application voice app_libretto_onramp9 access-method prompt-user
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the application.
call application voice load
Reloads the designated TCL script.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice access-method
Command
Description
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the application.
call application voice redirect-number Defines the telephone number to which a call will be
redirected—for example, the operator telephone number of
the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter
the PIN for a designated application and passes that
information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application.
call application voice warning-time
Defines the number of warning seconds a user receives
before the allowed calling time runs out for the designated
application.
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call application voice accounting enable
call application voice accounting enable
To enable authentication, authorization, and accounting (AAA) accounting for a Tool Command
Language (TCL) application, use the call application voice accounting enable command in global
configuration mode. To disable accounting for a TCL application, use the no form of this command.
call application voice application-name accounting enable
no call application voice application-name accounting enable
Syntax Description
application-name
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
The name of the application.
This command enables AAA accounting services if an AAA accounting method list has been defined
using both the aaa accounting command and the mmoip aaa method fax accounting command.
This command applies to off-ramp store-and-forward fax functions on Cisco AS5300 universal access
server voice feature cards (VFCs). It is not used on modem cards.
Examples
The following example enables AAA accounting to be used with outbound store-and-forward fax:
configure terminal
call application voice app_libretto_onramp9 accounting enable
Related Commands
Command
Description
mmoip aaa method fax accounting
Defines the name of the method list to be used for AAA
accounting with store-and-forward fax.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice accounting-list
call application voice accounting-list
To define the accounting list name of the voice feature card (VFC), use the call application voice
accounting-list command in global configuration mode. To restore the default value, use the no form of
this command.
call application voice application-name accounting-list method-list-name
no call application voice application-name accounting-list method-list-name
Syntax Description
application-name
The name of the application.
method-list-name
Character string used to name a list of accounting methods to be used with
store-and-forward fax.
Defaults
No AAA accounting method list is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
This command defines the name of the accounting feature of the authentication, authorization, and
accounting (AAA) method list to be used with store-and-forward fax. The method list itself, which
defines the type of accounting services provided for store-and-forward fax, is defined using the aaa
accounting global configuration command. Unlike standard AAA (where each defined method list can
be applied to specific interfaces and lines), the AAA accounting method lists used in store-and-forward
fax are applied globally on the Cisco AS5300 universal access server.
After the accounting method lists have been defined, they are enabled by using the mmoip aaa
receive-accounting enable command.
This command applies to both on-ramp and off-ramp store-and-forward fax functions on Cisco AS5300
universal access server voice feature cards. It is not used on modem cards.
Examples
The following example defines a AAA accounting method list (called “sherman”) to be used with
store-and-forward fax:
aaa new-model
call application voice app_libretto_onramp9 accounting-list sherman
Related Commands
Command
Description
call application voice accounting enable Enables on-ramp AAA accounting services.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice authen-list
call application voice authen-list
To specify the name of an authentication method list for a Tool Command Language (TCL) application,
use the call application voice authen-list command in global configuration mode. To disable the
authentication method list for a TCL application, use the no form of this command.
call application voice application-name authen-list method-list-name
no call application voice application-name authen-list method-list-name
Syntax Description
application-name
The name of the application.
method-list-name
Character string used to name a list of authentication methods to be used with
T.38 fax relay and T.37 store-and-forward fax.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
This command defines the name of the authentication, authorization, and accounting (AAA)
authentication method list to be used with fax applications on voice feature cards. The method list itself,
which defines the type of authentication services provided for store-and-forward fax, is defined using
the aaa authentication global configuration command. Unlike standard AAA (where each defined
method list can be applied to specific interfaces and lines), AAA authentication method lists used with
fax applications are applied globally on the Cisco AS5300 universal access server.
After the authentication method lists have been defined, they are enabled by using the call application
voice authentication enable command.
Examples
The following example defines an AAA authentication method list (called “fax”) to be used with T.38
fax relay and T.37 store-and-forward fax:
configure terminal
call application voice app_libretto_onramp9 authen-list fax
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice authen-list
Related Commands
Command
Description
call application voice authentication
enable
Enables AAA authentication services for a TCL
application.
call application voice authen-method
Specifies the authentication method for a TCL
application.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice authen-method
call application voice authen-method
To specify an authentication, authorization, and accounting (AAA) authentication method for a Tool
Command Language (TCL) application, use the call application voice authen-method command in
global configuration mode. To disable the authentication method for a TCL application, use the no form
of this command.
call application voice application-name authen-method {prompt-user | ani | dnis | gateway |
redialer-id | redialer-dnis}
no call application voice application-name authen-method {prompt-user | ani | dnis | gateway |
redialer-id | redialer-dnis}
Syntax Description
application-name
The name of the application.
prompt-user
Indicates that the user is prompted for the TCL application account identifier.
ani
Indicates that the calling-party telephone number (automatic number
identification [ANI]) is used as the TCL application account identifier.
dnis
Indicates that the called party telephone number (dialed number identification
service [DNIS]) is used as the TCL application account identifier.
gateway
Indicates that the router-specific name derived from the host name and
domain name is used as the TCL application account identifier. It is displayed
in the following format: router-name.domain-name.
redialer-id
Indicates that the account string returned by the external redialer device is
used as the TCL application account identifier. In this case, the redialer ID is
either the redialer serial number or the redialer account number.
redialer-dnis
Indicates that the called party telephone number (dialed number identification
service or DNIS) is used as the TCL application account identifier captured
by the redialer if a redialer device is present.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
Normally, when AAA is being used for simple user authentication, AAA uses the username information
defined in the user profile for authentication. With T.37 store-and-forward fax and T.38 real-time fax,
you can specify that the ANI, DNIS, gateway identification (ID), redialer ID, or redialer DNIS be used
to identify the user for authentication or that the user be prompted for the TCL application.
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call application voice authen-method
Examples
The following example shows how to configure the router-specific name derived from the host name and
domain name as the TCL application account identifier for the app_libretto_onramp9 TCL application:
call application voice app_libretto_onramp9 authen-method gateway
Related Commands
Command
Description
call application voice authentication
enable
Enables AAA authentication services for a TCL
application.
call application voice authen-list
Specifies the name of an authentication method list for a
TCL application.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice authentication enable
call application voice authentication enable
To enable AAA authentication services for a tool command line (TCL) application, use the call
application voice authentication enable command in global configuration mode. To disable
authentication for a TCL application, use the no form of this command.
call application voice application-name authentication enable
no call application voice application-name authentication enable
Syntax Description
application-name
The name of the application.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal
access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
This command enables AAA authentication services for a TCL application if a AAA authentication
method list has been defined using both the aaa authentication command and the call application voice
authen-list command.
Examples
The following example enables a AAA authentication method list (called peabody) to be used with
outbound store and forward fax.
configure terminal
aaa new-model
call application voice app_onramp6 authen-list peabody
call application voice app_onramp6 authentication enable
Related Commands
Command
Description
call application voice authen-list
Specifies the name of an authentication method list for a TCL
application.
call application voice
authen-method
Specifies the authentication method for a TCL application.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice global-password
call application voice global-password
To define a password to be used with CiscoSecure for Windows NT when using store-and-forward fax
on a voice feature card, use the call application voice global-password command in global
configuration mode. To restore the default value, use the no form of this command.
call application voice application-name global-password password
no call application voice application-name global-password password
Syntax Description
application-name
The name of the application.
password
Character string used to define the CiscoSecure for Windows NT password to
be used with store-and-forward fax. The maximum length is 64 alphanumeric
characters.
Defaults
No password is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 universal access server.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
CiscoSecure for Windows NT might require a separate password to complete authentication, no matter
what security protocol you use. This command defines the password to be used with CiscoSecure for
Windows NT. All records on the Windows NT server use this defined password.
This command applies to on-ramp store-and-forward fax functions on Cisco AS5300 universal access
server voice feature cards. It is not used on modem cards.
Examples
The following example shows a password (abercrombie) being used by AAA for the
app_libretto_onramp9 TCL application:
call application voice app_libretto_onramp9 global-password abercrombie
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call application voice language
call application voice language
To define the language of the audio file for the specified application and to pass that information to the
specified application, use the call application voice language command in global configuration mode.
To remove the associated language of the audio file from the application, use the no form of this
command.
call application voice application-name language number language
no call application voice application-name language number language
Syntax Description
application-name
The name of the application to which the language parameters are being
passed.
number
Tag that uniquely identifies an audio file. Valid entries are from 0 to 9.
language
Defines the language of the associated audio file. Valid entries are as follows:
•
en—English
•
sp—Spanish
•
ch—Mandarin
•
aa—all
Defaults
No default behavior or values.
Command Modes
Global configuration mode
Command History
Release
Modification
12.0(7)T
This command was introduced.
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool
Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card)
to define the language of the audio file for the specified application and to pass that information to the
specified application.
Table 5 lists TCL script names and the corresponding parameters that are required for each TCL script.
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call application voice language
Table 5
TCL Scripts and Parameters
TCL Script Name
Description
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
personal identification number (PIN), respectively,
using automatic number identification (ANI) and
NULL. The numbers of digits allowed for the account
number and password, respectively, are configurable
through the command-line interface (CLI). If the
authentication fails, it allows the caller to retry. The
retry number is also configured through the CLI.
Parameters to Configure
•
call application voice
uid-len
minimum = 1, maximum =
20, default = 10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum = 5,
default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and PIN,
respectively, using ANI and NULL. If the
authentication fails, it allows the caller to retry. The
retry number is configured through CLI. The account
number and PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum = 5,
default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum = 5,
default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum = 5,
default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account and PIN, respectively. If authentication fails,
it allows the caller to retry. The retry number is
configured through the CLI. The account number and
PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum = 5,
default = 3
Examples
The following example shows how to define English and Spanish as the languages of the audio files
associated with the application (named “prepaid”):
call application voice prepaid language 1 en
call application voice prepaid language 2 sp
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call application voice language
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice load
Reload the designated TCL script.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application. the application.
call application voice warning-time
Defines the number of second’s warning a user receives
before the allowed calling time runs out for the
designated application.
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call application voice load
call application voice load
To reload the selected Tool Command Language (TCL) script from the URL, use the call application
voice load command in privileged EXEC mode.
call application voice load name
Syntax Description
name
Defaults
TCL or scripts are not loaded.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series routers,
Cisco 3600 series (except for the 3660), and on the Cisco AS5300.
12.1(3)T
Support for dynamic script loading of MGCP scripts was added.
Usage Guidelines
Defines the TCL script to use for the call. Enter the name of the TCL
or Media Gateway Control Protocol (MGCP) script you want this dial
peer to use.
The software checks the signature lock to ensure that it is a Cisco-supported TCL script.
If the TCL script does not have a valid Cisco-supported signature, the software fails to load the script
and generates the following error message:
00:02:54: %IVR-3-BAD_IVR_SIG: Script signature is invalid
Examples
The following example shows the loading of a MGCP script package:
Router# call application voice load mgcp-script-pkg
Related Commands
Command
Description
call application cache
reload time
Configures the interval for reloading MGCP scripts.
call application voice
Creates and calls the application that will interact with the IVR feature.
show call application
voice
Displays a list of the voice applications that are configured.
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call application voice pin-len
call application voice pin-len
To define the number of characters in the personal identification number (PIN) for the designated
application, use the call application voice pin-len command in global configuration mode. To restore
default values for this command, use the no form of this command.
call application voice application-name pin-len number
no call application voice application-name pin-len number
Syntax Description
application-name
The name of the application to which the PIN length parameter is being
passed.
number
Defines the number of allowable characters in PINs associated with the
specified application. Valid entries are from 0 to 10.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced in the Cisco 2600 series routers, Cisco 3600
series routers, and Cisco AS5300 universal access routers.
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the TCL script
being used—or one of the IVR-related features (such as Debit Card) to define the number of allowable
characters in a PIN for the specified application and to pass that information to the specified application.
Table 6 lists TCL script names and the corresponding parameters that are required for each TCL script.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice pin-len
Table 6
TCL Scripts and Parameters
TCL Script Name
Description
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
PIN, respectively, using automatic number
identification (ANI) and NULL. The number of digits
allowed for the account number and password,
respectively, are configurable through the
command-line interface (CLI). If the authentication
fails, it allows the caller to retry. The retry number is
also configured through the CLI.
Parameters to Configure
•
call application voice
uid-len
minimum = 1, maximum =
20, default = 10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and
PIN, respectively, using ANI and NULL. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account and PIN, respectively. If authentication fails,
it allows the caller to retry. The retry number is
configured through the CLI. The account number and
PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
Examples
The following example shows how to define a PIN length of four characters for the application (named
“prepaid”):
call application voice prepaid pin-len 4
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice pin-len
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reload this designated TCL script.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application.
call application voice warning-time
Defines the number of seconds of warning that a user
receives before the allowed calling time runs out for the
designated application.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice redirect-number
call application voice redirect-number
To define the telephone number to which a call will be redirected—for example, the operator telephone
number of the service provider—for the designated application, use the call application voice
redirect-number command in global configuration mode. To cancel this particular parameter, use the
no form of this command.
call application voice application-name redirect-number number
no call application voice application-name redirect-number number
Syntax Description
application-name
The name of the application to which the redirect telephone number
parameter is being passed.
number
Defines the designated operator telephone number of the service provider (or
any other number designated by the customer). This is the number that calls
are terminated to when, for example, debit time allowed has run out or the
debit amount is exceeded.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series routers, 3600 series
routers , and the AS5300 universal access server.
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool
Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card)
to define the telephone number to which a call will be redirected.
Cisco IOS Voice, Video, and Fax Command Reference
VR-72
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice redirect-number
Table 7 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 7
TCL Scripts and Parameters
TCL Script Name
Description
Parameters to Configure
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
personal identification number (PIN), respectively,
using automatic number identification (ANI) and
NULL. The number of digits allowed for the account
number and password, respectively, are configurable
through the command-line interface (CLI). If the
authentication fails, it allows the caller to retry. The
retry number is also configured through the CLI.
•
call application voice
uid-len
minimum = 1, maximum =
20, default =10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and
PIN, respectively, using ANI and NULL. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account and PIN, respectively. If the authentication
fails, it allows the caller to retry. The retry number is
configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
Examples
The following example shows how to define a redirect number for the application (named “prepaid”):
call application voice prepaid redirect-number 5551111
Cisco IOS Voice, Video, and Fax Command Reference
VR-73
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice redirect-number
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reloads the designated TCL script.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application.
call application voice warning-time
Defines the number of seconds of warning that a user
receives before the allowed calling time runs out for the
designated application.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice retry-count
call application voice retry-count
To define the number of times a caller is permitted to reenter the personal identification number (PIN)
for the designated application, use the call application voice retry-count command in global
configuration mode. To cancel this particular parameter, use the no form of this command.
call application voice application-name retry-count number
no call application voice application-name retry-count number
Syntax Description
application-name
The name of the application to which the number of possible retries is being
passed.
number
Defines the number of times the caller is permitted to reenter personal
identification number (PIN) digits.Valid entries for this parameter are from 1
to 5.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series routers, 3600 series,
and on the AS5300.
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool
Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card)
to define how many times a user can reenter a PIN.
Cisco IOS Voice, Video, and Fax Command Reference
VR-75
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice retry-count
Table 8 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 8
TCL Scripts and Parameters
TCL Script Name
Description Summary
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
PIN, respectively, using automatic number
identification (ANI) and NULL. The number of digits
allowed for the account number and password,
respectively, are configurable through the
command-line interface (CLI). If the authentication
fails, it allows the caller to retry. The retry number is
also configured through the CLI.
Parameters to Configure
•
call application voice
uid-len
minimum = 1, maximum =
20, default =10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and
PIN, respectively, using ANI and NULL. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account and PIN, respectively. If authentication fails,
it allows the caller to retry. The retry number is
configured through the CLI. The account number and
PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
Examples
The following example shows how to define that a user can re-enter a PIN three times before being
disconnected for the application (named “prepaid”):
call application voice prepaid retry-count 3
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice retry-count
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reloads the designated TCL script.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application. the application.
call application voice warning-time
Defines the number of second’s warning a user receives
before the allowed calling time runs out for the
designated application.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice set-location
call application voice set-location
To define the location, language, and category of the audio files for the specified application, use the call
application voice set-location command in global configuration mode. To cancel this particular
parameter, use the no form of this command.
call application voice application-name set-location language category location
no call application voice application-name set-location language category location
Syntax Description
application-name
The name of the application to which the set-location parameters are being
passed.
language
Defines the language associated with the audio files. Possible values for this
parameter are as follows:
•
en = English
•
ch = Mandarin
•
sp = Spanish
category
Defines a particular category group. Audio files can be divided into category
groups (from 0 to 4). For example, audio files representing the days and
months can be category 1, audio files representing units of currency can be
category 2, audio files representing units of time—seconds, minutes, and
hours—can be category 3. The minimum is 0; the maximum is 4 (0 means all).
location
Defines the location (audio file URL or directory in the TFTP server) where
the audio files are stored.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series routers, 3600 series,
and on the AS5300.
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the TCL script
being used—or one of the IVR-related features (such as Debit Card) to define the location, language,
and category of the audio files for the designated application and pass that information to the application.
Cisco IOS Voice, Video, and Fax Command Reference
VR-78
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice set-location
Table 9 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 9
TCL Scripts and Parameters
TCL Script Name
Description
Parameters to Configure
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
PIN, respectively, using automatic number
identification (ANI) and NULL. The number of digits
allowed for the account number and password,
respectively, are configurable through the
command-line interface (CLI). If the authentication
fails, it allows the caller to retry. The retry number is
also configured through the CLI.
•
call application voice
uid-len
minimum = 1, maximum =
20, default =10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and
PIN, respectively, using ANI and NULL. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account and PIN respectively. If the authentication
fails, it allows the caller to retry. The retry number is
configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
Examples
The following example shows how to configure the call application voice set-location command for the
application (named “prepaid”). In this example, the language defined is English, the category into which
the audio files are group is Category 0 (meaning all), and the location is the keyer directory on the TFTP
server.
call application voice prepaid set-location en 0 tftp://keyer/
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice set-location
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reloads the designated TCL script.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application. the application.
call application voice warning-time
Defines the number of seconds of warning that a user is
warned before their allowed calling time runs out for the
designated application.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice uid-length
call application voice uid-length
To define the number of characters in the user identification number (UID) for the designated
application, use the call application voice uid-length command in global configuration mode. To delete
the specification of the number of characters in the user identification number, use the no form of this
command.
call application voice application-name uid-length number
no call application voice application-name uid-length number
Syntax Description
application-name
The name of the application to which the UID length parameter is being
passed.
number
Defines the number of allowable characters in UIDs associated with the
specified application. Valid entries are from 1 to 20.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series routers, Cisco 3600
series, and on the Cisco AS5300.
Usage Guidelines
Use this command when configuring interactive voice response (IVR), depending on the Tool Command
Language (TCL) script being used or one of the IVR-related features (such as Debit Card) to define the
number of allowable characters in a UID for the specified application and to pass that information to the
specified application.
Cisco IOS Voice, Video, and Fax Command Reference
VR-81
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice uid-length
Table 7 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 10
TCL Script Names and Parameters
TCL Script Name
Description
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
PIN, respectively, using automatic number
identification (ANI) and NULL. The number of digits
allowed for the account number and password,
respectively, are configurable through the
command-line interface (CLI). If the authentication
fails, it allows the caller to retry. The retry number is
also configured through the CLI.
Parameters to Configure
•
call application voice
uid-len
minimum = 1, maximum =
20, default =10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and
personal identification number (PIN), respectively,
using ANI and NULL. If the authentication fails, it
allows the caller to retry. The retry number is
configured through the CLI. The account number and
PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If the
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
Examples
The following example shows how to configure four allowable characters in the UID for the application
(named “prepaid”):
call application voice prepaid uid-len 4
Cisco IOS Voice, Video, and Fax Command Reference
VR-82
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice uid-length
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reloads the designated TCL script.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice warning-time
Defines the number of seconds of warning that a user
receives before the allowed calling time runs out for the
designated application.
Cisco IOS Voice, Video, and Fax Command Reference
VR-83
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice warning-time
call application voice warning-time
To define the number of seconds of warning that a user receives before the allowed calling time runs out,
use the call application voice warning-time command in global configuration mode. To restore default
values for this command, use the no form of this command.
call application voice application-name warning-time number
no call application voice application-name warning-time number
Syntax Description
application-name
The name of the application to which the warning time parameter is being
passed.
number
Defines the length of the warning period, in seconds, before the allowed
calling time runs out. Valid entries are from 10 to 600.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series routers, 3600 series
routers, and AS5300 universal access server.
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool
Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card)
to define how many seconds in the warning period before the allowed calling time runs out for the
specified application and to pass that information to the specified application.
Table 11 lists TCL script names and the corresponding parameters that are required for each TCL script.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice warning-time
.
Table 11
TCL Scripts and Parameters
TCL Script Name
Description
Parameters to Configure
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and
PIN, respectively, using automatic number
identification (ANI) and NULL. The number of digits
allowed for the account number and password,
respectively, are configurable through the
command-line interface (CLI). If the authentication
fails, it allows the caller to retry. The retry number is
also configured through the CLI.
•
call application voice
uid-len
minimum = 1, maximum =
20, default =10
•
call application voice
pin-len
minimum = 0, maximum =
10, default = 4
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_col_npw_cli.tcl
This script authenticates the account number and
personal identification number (PIN), respectively,
using ANI and NULL. If the authentication fails, it
allows the caller to retry. The retry number is
configured through the CLI. The account number and
PIN are collected separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN
using ANI and dialed number identification service
(DNIS). If the authentication fails, it allows the caller
to retry. The retry number is configured through the
CLI. The account number and PIN are collected
separately.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for
account and PIN, respectively. If the authentication
fails, it allows the caller to retry. The retry number is
configured through the CLI.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for
account numbers and PINs, respectively. If
authentication fails, it allows the caller to retry. The
retry number is configured through the CLI. The
account number and PIN are collected together.
•
call application voice
retry-count
minimum = 1, maximum =
5, default = 3
Examples
The following example shows how to configure a 30-second warning time for the application (named
“prepaid”):
call application voice prepaid warning-time 30
Cisco IOS Voice, Video, and Fax Command Reference
VR-85
Cisco IOS Voice, Video, and Fax Commands: A Through C
call application voice warning-time
Related Commands
Command
Description
call application voice language
Defines the language of the audio file for the designated
application and passes that information to the
application.
call application voice load
Reloads the designated TCL script.
call application voice location
Defines the name to be used for an application and
indicates the location of the appropriate IVR script to be
used with this application.
call application voice pin-len
Defines the number of characters in the PIN for the
application and passes that information to the
application.
call application voice redirect-number
Defines the telephone number to which a call will be
redirected—for example, the operator telephone number
of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to
reenter the PIN for a designated application and passes
that information to the application.
call application voice set-location
Defines the location, language, and category of the audio
files for the designated application and passes that
information to the application.
call application voice uid-len
Defines the number of characters in the UID for the
designated application and passes that information to the
application. the application.
Cisco IOS Voice, Video, and Fax Command Reference
VR-86
Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback active
call fallback active
To enable a call request to fall back to alternate dial peers in case of network congestion, use the call
fallback active command in global configuration mode. To disable public switched telephone network
(PSTN) fall back, use the no form of this command.
call fallback active
no call fallback active
Syntax Description
This command has no arguments or keywords.
Defaults
This command is disabled by default.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series
routers, 3600 series, and on the MC3810 multiservice
concentrator.
Usage Guidelines
Enabling the call fallback active command determines whether calls should be accepted or rejected
based on probing of network conditions. The command call fallback active checks each H.323 call
request and rejects the call if the network congestion parameters are greater than the value of the
configured threshold parameters of the destination. If this is the case, alternative dial peers are tried from
the session application layer.
Use the call fallback threshold delay loss or call fallback threshold icpif command to set the threshold
parameters.
Connected calls are not affected by this feature.
Examples
The following example enables the call fallback active command:
call fallback active
Related Commands
Command
Description
call fallback cache-size
Specifies the call fallback cache size for network traffic
probe entries.
call fallback cache-timeout
Specifies the time after which the cache entries of
network conditions are purged.
call fallback instantaneous-value-weight Configures the call fallback subsystem to take an average
from the last two cache entries for call requests.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback active
Command
Description
call fallback jitter-probe num-packets
Specifies the number of packets in a jitter probe used to
determine network conditions.
call fallback jitter-probe precedence
Specifies the priority of the jitter-probe transmission.
call fallback jitter-probe priority-queue Assigns a priority-queue for jitter-probe transmissions.
call fallback key-chain
Specifies use of MD5 authentication for sending and
receiving SAA probes.
call fallback map address-list
Configures the call fallback router to keep a cache table
by IP addresses of distances for several destination peers
sitting behind the router.
call fallback map subnet
Configures the call fallback router to keep a cache table
by subnet addresses of distances for several destination
peers sitting behind the router.
call fallback probe-timeout
Sets the timeout for a SAA probe for call fallback
purposes.
call fallback threshold delay loss
Configures the call fallback threshold to use solely packet
delay and loss values.
call fallback threshold icpif
Configures call fallback to use the Impairment/Calculated
Planning Impairment Factor (ICPIF) threshold.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback cache-size
call fallback cache-size
To specify the call fallback cache size for network traffic probe entries, use the call fallback cache-size
command in global configuration mode. To restore the default value, use the no form of this command.
call fallback cache-size number
no call fallback cache-size number
Syntax Description
number
Defaults
128 entries
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers,
3600 series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Specifies the cache size in number of entries. The valid range
is from 1 to 256.
The cache size can be changed only when the call fallback active command is not enabled.
The pverflow process deletes up to one-fourth of the cache entries to allow for additional calls beyond
the specified cache size. The cache entries chosen for deletion are the oldest entries in the cache.
Examples
The following example specifies 120 cache entries:
call fallback cache-size 120
When call fallback is already configured, the output is as follows:
call fall cache-size 128
Cache size left unchanged (can be changed only when Fallback is OFF (use no call fallback)
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback cache-timeout
Specifies the time after which the cache entry is purged.
show call fallback cache
Displays the current IPCIF estimates for all IP addresses in the
cache.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
VR-89
Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback cache-timeout
call fallback cache-timeout
To specify the time after which the cache entries of network conditions are purged, use the call fallback
cache-timeout command in global configuration mode. To disable, use the no form of this command.
call fallback cache-timeout seconds
no call fallback cache-timeout seconds
Syntax Description
seconds
Defaults
600 seconds
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Specifies the cache timeout value in seconds. The valid range is from
1 to 2,147,483.
Enabling the call fallback cache-timeout command sends a Service Assurance Agent (SAA) probe out
to the network to determine the amount of congestion in terms of configured thresholds.The network
condition terms are based on delay and loss, or Impairment/Calculated Planning Impairment Factor
(ICPIF) thresholds. Use the call fallback threshold delay loss or call fallback threshold icpif
command to set the threshold parameters.
The cache keeps entries for every network congestion-checking probe sent and received between
timeouts. The cache updates after each probe returns the current condition of network traffic. To set the
probe frequency, use the call fallback probe-timeout command.
A call comes into the router. The router matches a dial peer and obtains the destination information. The
router calls the fall back subsystem to look up the specified destination in its network traffic cache. If
the delay and loss or ICPIF threshold exists and is current, then the router uses that value to decide
whether to permit the call into the VoIP network. If the router determines that the network congestion is
below the configured threshold (by looking at the value in the cache), then the call is connected.
After each call request, the timer is reset. Purging of the cache occurs only when the cache has received
no call requests during the timeout (seconds) period. When the cache timeout expires, the entire cache
is deleted, and a probe is sent to start a new cache entry. A call cannot be completed until this probe
returns with network traffic information.
The network congestion probes continue in the background as long as the entry for the last call request
remains in the cache.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback cache-timeout
Examples
The following example specifies 1200 seconds before the cache times out:
call fallback cache-timeout 1200
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback cache-size
Specifies the call fallback cache size.
call fallback probe-timeout
Specifies the call fallback probe timeout.
call fallback threshold delay loss
Configures the call fallback threshold to use only packet delay
and loss values.
call fallback threshold icpif
Configures the call fallback to use the ICPIF threshold.
show call fallback cache
Displays the current ICPIF estimates for all IP addresses in the
cache.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback instantaneous-value-weight
call fallback instantaneous-value-weight
To configure the call fallback subsystem to take an average from the last two probes registered in the
cache for call requests, use the call fallback instantaneous-value-weight command in global
configuration mode. To return to the default values, use the no form of this command.
call fallback instantaneous-value-weight weight
no call fallback instantaneous-value-weight weight
Syntax Description
weight
Defaults
66 percent
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Specifies the instantaneous value weight. The valid range is from 0
to 100 percent.
Probes returning with network congestion information are logged into the cache to determine whether
the next call request will be granted. The network can be busy regularly, and the cache entries reflect
these heavy traffic conditions. However, one probe may return with low traffic conditions, which are in
contrast to normal conditions. All call requests received between the time of this probe and the next use
this entry to determine call acceptance. These calls are allowed through the network, but before the next
probe is sent and received, the normal heavy traffic conditions may have returned. The calls sent through
congest the network and result in worse traffic conditions.
Use the call fallback instantaneous-value-weight command to recover gradually from heavy traffic
network conditions. While the system waits for a call, probes are received updating the cache. When a
new probe is received, the weight calculates how much to rely upon the new probe and how much to rely
upon the previous cache entry. If the weight is set to 50(%), the system enters a cache entry based on an
average from the new probe and the most recent entry in the cache. Call requests use this blended entry
to determine acceptance. This system allows the call fallback subsystem to keep conservative measures
of network congestion.
The configured weight applies to the new probe first. If the call fallback instantaneous-value-weight
command is configured with the default weight of 66(%), the new probe is given a higher value than the
earlier one in calculating the average for the new cache entry.
Examples
The following example specifies a fall back value weight of 50 percent:
call fallback instantaneous-value-weight 50
Cisco IOS Voice, Video, and Fax Command Reference
VR-92
Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback instantaneous-value-weight
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
VR-93
Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback jitter-probe num-packets
call fallback jitter-probe num-packets
To specify the number of packets in a jitter probe used to determine network conditions, use the
call fallback jitter-probe num-packets command in global configuration mode. To restore the default
value, use the no form of this command.
call fallback jitter-probe num-packets number-of-packets
no call fallback jitter-probe num-packets number-of-packets
Syntax Description
number-of-packets
Defaults
15 packets
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Specifies the number of packets value. The valid range is from 2 to 50.
A jitter probe, consisting of 2 to 50 packets, details the conditions of the network. More than one packet
is used by the probe to calculate an average of delay and loss or Impairment/Calculated Planning
Impairment Factor (ICPIF). After the packets return to the probe, the probe delivers the traffic
information to the cache, where it is logged for call acceptance or denial. Use the call fallback
threshold delay loss or call fallback threshold icpif command to set the threshold parameters.
To get a more realistic estimate of the network congestion, increase the number of packets. More probing
packets give better estimates of network conditions, but also negatively affect the bandwidth for other
network operations. Use fewer packets when you need to focus on bandwidth.
Examples
The following example specifies 20 packets for jitter:
call fallback jitter-probe num-packets 20
If the call fallback command has been enabled before configuring the number of jitter-probe packets,
the output is as follows:
call fallback jitter-probe num-packets 20
The new num-packets will take effect only for new probes
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback jitter-probe num-packets
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in
case of network congestion.
call fallback jitter-probe precedence Specifies the jitter-probe precedence.
call fallback jitter-probe
priority-queue
Assigns a priority queue for jitter-probe transmissions.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback jitter-probe precedence
call fallback jitter-probe precedence
To specify the treatment of the jitter-probe transmission, use the call fallback jitter-probe precedence
command in global configuration mode. To restore the default value, use the no form of this command.
call fallback jitter-probe precedence precedence-value
no call fallback jitter-probe precedence precedence-value
Syntax Description
precedence-value
Defaults
Precedence of 2
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Specifies the jitter-probe precedence. The valid range is from 0 to 6.
In every IP packet, there is a precedence header. Precedence is used by various queueing mechanisms in
different companies’ routers to determine the priority of allowing traffic through the system.
Use the call fallback jitter-probe precedence command if there are different queueing mechanisms in
your network. Enabling the call fallback jitter-probe precedence command sets the precedence for
jitter probes to pass through your network.
If you require your probes to be sent and returned quickly, set the precedence to a low number (zero or
one): the lower the precedence, the higher the priority given.
Examples
The following example specifies a jitter-probe precedence of 5, or low priority:
call fallback jitter-probe precedence 5
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback jitter-probe
num-packets
Specifies the number of packets in a jitter probe used to
determine network conditions.
call fallback jitter-probe
priority-queue
Assigns a priority queue for jitter-probe transmissions.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback jitter-probe priority-queue
call fallback jitter-probe priority-queue
To assign a priority queue, use the call fallback jitter-probe priority-queue command in global
configuration mode. To return to default values, use the no form of this command.
call fallback jitter-probe priority-queue
no call fallback jitter-probe priority-queue
Syntax Description
This command has no arguments or keywords.
Defaults
This command is disabled by default.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
This command is applicable only if the queueing method used is IP RTP Priority. This command is
unnecessary when low latency queueing (LLQ) is used because these packets follow the priority queue
path (or not) based on the LLQ classification criteria and not this command.
The command works by choosing between sending the probe on an odd or even RTP port number. The
Service Assurance Agent (SAA) probe packets go out on randomly selected ports chosen from within
the top end of the audio User Datagram Protocol (UDP) defined port range (16384–32767). The port pair
(Real-Time Transport Protocol [RTP] & Real-Time Transport Control Protocol [RTCP] port) is selected,
and, by default, SAA probes for call fallback use the RTCP port (odd) to avoid going into the priority
queue, if enabled. If call fallback is configured to use the priority queue, the RTP port (even) is selected.
Examples
The following example specifies the setting of the call fallback jitter-probe priority queue command:
call fallback jitter-probe priority-queue
Warning:In order for this command to have any affect on the probes, IP priority queueing
must be set for UDP voice ports 16384-32767.
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback jitter-probe
num-packets
Specifies the number of packets in a jitter probe used to
determine network conditions.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback jitter-probe priority-queue
Command
Description
call fallback jitter-probe
precedence
Specifies the jitter-probe precedence.
ip rtp priority
Provides a strict priority queueing scheme for delay-sensitive
data.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback key-chain
call fallback key-chain
To specify use of Message Digest 5 (MD5) authentication for sending and receiving Service Assurance
Agent (SAA) probes, use the call fallback key-chain command in global configuration mode. To disable
MD5 use, use the no form of this command.
call fallback key-chain name-of-chain
no call fallback key-chain name-of-chain
Syntax Description
name-of-chain
Defaults
No call fallback key chain is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers,
3600 series, and on the MC3810 multiservice concentrator.
Specifies the name of the chain. This line is to be alphanumeric
and case-sensitive text.
Usage Guidelines
This command is used to enable Service Assurance Agent (SAA) probe authentication using MD5. If
authentication is used, the keys on the sender and receiver routers must match.
Examples
The following example specifies “secret” as the fall back key chain:
call fallback key-chain secret
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
key chain
Enables authentication for routing protocols by identifying a
group of authentication keys.
key-string
Specifies the authentication string for a key.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback map target address-list
call fallback map target address-list
To configure the call fallback router to keep a cache table by IP addresses of distances for several
destination peers sitting behind the router, use the call fallback map target address-list command in
global configuration mode. To restore the default values, use the no form of this command.
call fallback map map target ip-address address-list ip-address1 ip-address2 ... ip-address7
no call fallback map map target ip-address address-list ip-address1 ip-address2 ... ip-address7
Syntax Description
map
Specifies the fall back map. The valid range is from 1 to 16.
target ip-address
Specifies the target IP address.
ip-address1 ... ip-address7
Lists the IP addresses that will be kept in the cache table. The
maximum number of IP addresses is seven.
Defaults
No call fallback maps are defined.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers,
3600 series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Use this command when several destination peers are connected to a single access point.
Call fallback map setup allows the decongestion of traffic caused by a high volume of call probes sent
across a network to query a large number of dial peers. One router/common node can keep the distances
in a cache table to numerous IP addresses or destination peers in a network. When the fall back is queried
for network congestion to a particular IP address (that is, the common node), the map addresses are
searched to find the target IP address. If a match is determined, the probes are sent to the target address
rather than to the particular IP address.
In Figure 2, the three routers (1, 2, and 3) keep the cache tables of distances for the destination peers
behind them. When a call probe comes from somewhere in the IP cloud, the cache routers check their
distance tables for the IP address or destination peer where the call probe is destined. This distance
checking limits congestion on the networks behind these routers by directing the probe to the particular
IP address and not to the entire network.
Cisco IOS Voice, Video, and Fax Command Reference
VR-100
Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback map target address-list
Call Fallback Map with IP Addresses
172.16.10.8
172.16.10.7
172.26.10.7
172.31.10.7
172.16.10.6
172.16.10.5
172.26.10.5
172.16.10.4
172.26.10.4
172.31.10.4
172.26.10.3
172.31.10.3
172.16.10.2
172.16.10.1
172.26.10.6
172.31.10.6
172.31.10.5
172.16.10.3
172.26.10.8
172.31.10.8
172.26.10.2
172.31.10.2
172.26.10.1
172.31.10.1
Router 1
Router 2
Router 3
IP
cloud
Examples
30467
Figure 2
The following example specifies call fallback map target address-list configurations for 172.31.10.1
and 172.26.10.1:
call fallback map 1 target 172.31.10.1
address-list 172.31.10.2 172.31.10.3 172.31.10.4 172.31.10.5
172.31.10.6 172.31.10.7 172.31.10.8
call fallback map 2 target 172.26.10.1
address-list 172.26.10.2 172.26.10.3 172.26.10.4 172.26.10.5
172.26.10.6 172.26.10.7 172.26.10.8
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback map target subnet
Specifies the call fallback router to keep a cache table (by
subnet addresses) of distances for several destination peers
sitting behind the router.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback map target subnet
call fallback map target subnet
To configure the call fallback router to keep a cache table by subnet addresses of distances for several
destination peers sitting behind the router, use the call fallback map target subnet command in global
configuration mode. To restore the default values, use the no form of this command.
call fallback map map target ip-address subnet ip-network netmask
no call fallback map map target ip-address subnet ip-network netmask
Syntax Description
map
Specifies the fall back map. The valid range is from 1 to 16.
target ip-address
Specifies the target IP address.
subnet ip-network
Specifies the subnet IP address.
netmask
Specifies the network mask number.
Defaults
No call fallback maps are defined.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Use this command when several destination peers are sitting behind one common node.
call fallback map setup allows the decongestion of traffic caused by a high volume of call probes sent
across a network to query a large number of dial peers. One router/common node can keep the distances
in a cache table to numerous IP addresses within a subnet (destination peers) in a network. When the fall
back is queried for network congestion to a particular IP address (that is, the common node), the map
addresses are searched to find the target IP address. If a match is determined, the probes are sent to the
target address rather than to the particular IP address.
In Figure 3, the three routers (1, 2, and 3) keep the cache tables of distances for the destination peers
behind them. When a call probe comes from somewhere in the IP cloud, the cache routers check their
distance tables for the subnet address/destination peer where the call probe is destined. This distance
checking limits congestion on the networks behind these routers by directing the probe to the particular
subnet address and not to the entire network.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback map target subnet
Call Fallback Map with Subnet Addresses
209.165.200.254
209.165.201.30
209.165.202.158
209.165.200.233
209.165.201.8
209.165.202.136
209.165.200.231
209.165.201.7
209.165.200.230
209.165.200.229
209.165.202.135
209.165.201.6
209.165.201.5
209.165.200.228
209.165.200.227
209.165.202.133
209.165.201.4
209.165.201.3
209.165.200.226
209.165.200.225
255.255.255.224
209.165.202.134
209.165.202.132
209.165.202.131
209.165.201.2
209.165.201.1
255.255.255.224
Router 1
209.165.202.130
209.165.202.129
255.255.255.224
Router 2
Router 3
IP
cloud
Examples
30466
Figure 3
The following example specifies the call fallback map target subnet command configuration for
209.165.201.225:
call fall back map 1 209.165.201.225 subnet
209.165.201.224 255.255.255.224
call fall back map 2 209.165.202.225 subnet
209.165.202.224 255.255.255.224
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback map target
address-list
Specifies the call fallback router to keep a cache table (by IP
addresses) of distances for several destination peers sitting
behind the router.
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback monitor
call fallback monitor
To enable the monitoring of destinations with no provision for call fallback to alternate dial peers, use
the call fallback monitor command in global configuration mode. To disable monitoring without fall
back, use the no form of this command.
call fallback monitor
no call fallback monitor
Syntax Description
This command has no arguments or keywords.
Defaults
This command is disabled by default.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
The call fallback monitor command is used as a statistics collector of network conditions based on
probes (detailing network traffic) and connected calls. There is no H.323 call checking and rejecting as
with the call fallback active command. All call requests are granted, regardless of network traffic
conditions.
Configure the call fallback threshold delay loss or call fallback threshold icpif command to set
threshold parameters. The thresholds are ignored, but for statistics collecting, configuring one of the
thresholds allows you to monitor cache entries for either delay and loss or Impairment/Calculated
Planning Impairment Factor (ICPIF) values.
Examples
The following example shows that the call fallback monitor command has been enabled:
call fallback monitor
Related Commands
Command
Description
show call fallback config
Displays the call fallback configuration.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
call fallback probe-timeout
call fallback probe-timeout
To set the timeout for a Service Assurance Agent (SAA) probe for call fallback purposes, use the call
fallback probe-timeout command in global configuration command. To restore the default value, use
the no form of this command.
call fallback probe-timeout seconds
no call fallback probe-timeout seconds
Syntax Description
seconds
Defaults
30 seconds
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers,
3600 series, and on the MC3810.
Usage Guidelines
Specifies the interval in seconds. The valid range is from 1 to
2,147,483.
SAA probes collect network traffic information based on configured delay and loss or
Impairment/Calculated Planning Impairment Factor (ICPIF) values and report this information to the
cache for call request determinations. Use the call fallback threshold delay loss or call fallback
threshold icpif command to set the threshold parameters.
When the probe timeout expires, a new probe is sent to collect network statistics. To reduce the
bandwidth taken up by the probes, increase the probe-timeout interval (seconds). Probes do not have a
great affect upon bandwidth unless several thousand destinations are involved. If this is the case in your
network, use a longer timeout. If you need more network traffic information and bandwidth is not an
issue, use a lower timeout. The default interval, 30 seconds, is a low timeout.
When the call fallback cache-timeout command is configured or expires, new probes are initiated for
data collection.
Examples
The following example configures a 120-second interval:
call fallback probe-timeout 120
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
show call fallback config
Displays the call fallback configuration.
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call fallback threshold delay loss
call fallback threshold delay loss
To configure the call fallback threshold to use only packet delay and loss values, use the call fallback
threshold delay loss command in global configuration mode. To restore the default value, use the no
form of this command.
call fallback threshold delay delay-value loss loss-value
no call fallback threshold delay delay-value loss loss-value
Syntax Description
delay-value
Sets the delay value. The valid range is from 1 to 2,147,483,647
milliseconds.
loss-value
Sets the loss value. The valid range is from 0 to 100 percent.
Defaults
There are no values set for delay and loss by default.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
For voice traffic, delays and loss of voice packets create unhappy customers. During times of heavy voice
traffic, two parties in a conversation may notice a significant delay in transmission or hear only part of
a conversation due to loss of voice packets.
Use the call fallback threshold delay loss command to configure parameters for voice quality. Lower
values of delay and loss allow higher quality of voice. Call requests match the network information in
the cache with the configured thresholds of delay and loss. If you enable call fallback active, the call
fallback subsystem uses the last cache entry compared with the configured delay and loss threshold to
determine whether the call is connected or denied. If you enable call fallback monitor, all calls are
connected, regardless of the configured threshold or voice quality. In this case, configuring the call
fallback threshold delay loss command allows you to collect network statistics for further tracking.
Note
The call fallback threshold delay loss command differs from the call fallback threshold
icpif command because the call fallback threshold delay loss command uses only packet
delay and loss parameters. The call fallback threshold icpif command uses packet delay and
loss plus other ITU G.113 factors to gather impairment information.
Setting this command does not affect bandwidth. Available bandwidth for call requests is determined by
the call fallback subsystem using probes. The number of probes on the network affects bandwidth.
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call fallback threshold delay loss
Examples
The following example configures a threshold delay of 20 milliseconds and a threshold loss of 50 percent:
call fallback threshold delay 20 loss 50
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback threshold icpif
Specifies the ICPIF threshold.
show call fallback config
Displays the call fallback configuration.
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call fallback threshold icpif
call fallback threshold icpif
To configure call fallback to use the Impairment/Calculated Planning Impairment Factor (ICPIF)
threshold, use the call fallback threshold icpif command in global configuration mode. To restore the
default value, use the no form of this command.
call fallback threshold icpif threshold-value
no call fallback threshold icpif threshold-value
Syntax Description
threshold-value
Defaults
ICPIF threshold of 5
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
Sets the threshold value. The valid range is from 0 to 34.
During times of heavy voice traffic, two parties in a conversation may notice a significant delay in
transmission or hear only part of a conversation because of loss of voice packets.
Use the call fallback threshold icpif command to configure parameters for voice quality. A low ICPIF
value allows for higher quality of voice. Call requests match the network information in the cache with
the configured ICPIF threshold. If you enable the call fallback active command, the call fallback
subsystem uses the last cache entry compared with the configured ICPIF threshold to determine whether
the call is connected or denied. If you enable the call fallback monitor command, all calls are connected
regardless of the configured threshold or voice quality. In this case, configuring the call fallback
threshold icpif command allows you to collect network statistics for further tracking.
A lower value of ICPIF tolerates less delay and loss (according to ICPIF calculations) of voice packets.
Use lower values for higher quality of voice. Configuring a value of 34 equates to 100 percent packet
loss.
The ICPIF is calculated and used according to the International Telecommunication Union (ITU) G.113
specifications.
Note
The call fallback threshold delay loss command differs from the call fallback threshold icpif
command because the call fallback threshold delay loss command uses only packet delay and loss
parameters. The call fallback threshold icpif command uses packet delay and loss plus other ITU
G.113 factors to gather impairment information.
Setting this command does not affect bandwidth. Available bandwidth for call requests is determined by
the call fallback subsystem using probes. The number of probes on the network affect bandwidth.
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call fallback threshold icpif
Examples
The following example sets the ICPIF threshold to 20:
call fallback threshold icpif 20
Related Commands
Command
Description
call fallback active
Enables a call request to fall back to alternate dial peers in case
of network congestion.
call fallback threshold delay loss
Specifies the call fallback threshold delay and loss values.
show call fallback config
Displays the call fallback configuration.
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call rsvp-sync
call rsvp-sync
To enable synchronization between Resource Reservation Protocol (RSVP) signaling and the voice
signaling protocol, use the call rsvp-sync command in global configuration mode. To disable
synchronization, use the no form of this command.
call rsvp-sync
no call rsvp-sync
Syntax Description
This command has no keywords or arguments.
Defaults
Synchronization is enabled between RSVP and the voice signaling protocol (for example, H.323).
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600 series routers, 3600
series, and 7200 series and on the AS5300, AS5800, and MC3810.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
The call rsvp-sync command is enabled by default.
Examples
The following example enables synchronization between RSVP and the voice signaling protocol:
call rsvp-sync
Related Commands
Command
Description
call rsvp-sync resv-timer
Sets the timer for reservation requests.
call start
Forces the H.323 Version 2 gateway to use fast connect or slow
connect procedures for a dial peer.
debug call rsvp-sync events
Displays the events that occur during RSVP synchronization.
h323 call start
Forces an H.323 Version 2 gateway to use fast connect or slow
connect procedures for all VoIP services.
ip rsvp bandwidth
Enables the use of RSVP on an interface.
show call rsvp-sync conf
Displays the RSVP synchronization configuration.
show call rsvp-sync stats
Displays statistics for calls that have attempted RSVP reservation.
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call rsvp-sync resv-timer
call rsvp-sync resv-timer
To set the timer on the terminating VoIP gateway for completing RSVP reservation setups, use the
call rsvp-sync resv-timer command in global configuration mode. To restore the default value, use the
no form of this command.
call rsvp-sync resv-timer seconds
no call rsvp-sync resv-timer
Syntax Description
seconds
Defaults
The timer default is 10 seconds.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600 series routers, 3600
series, and 7200 series and on the AS5300, AS5800, and MC3810.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Number of seconds in which the reservation setup must be completed,
in both directions. The value range is from 1 to 60 seconds.
Usage Guidelines
The reservation timer is started on the terminating gateway when the session protocol receives an
indication of the incoming call. This timer is not set on the originating gateway because the resource
reservation is confirmed at the terminating gateway. If the reservation timer expires before the RSVP
setup is complete, the outcome of the call depends on the acceptable quality of service (QoS) level
configured in the dial peer; either the call proceeds without any bandwidth reservation or it is released.
The timer must be set long enough to allow calls to complete but short enough to free up resources. The
optimum number of seconds depends on the number of hops between the participating gateways and the
delay characteristics of the network.
Examples
The following example sets the reservation timer to 30 seconds:
call rsvp-sync resv-timer 30
Related Commands
Command
Description
call rsvp-sync
Enables synchronization of RSVP and the H.323 voice signaling
protocol.
debug call rsvp-sync events
Displays the events that occur during RSVP synchronization.
show call rsvp-sync conf
Displays the RSVP synchronization configuration.
show call rsvp-sync stats
Displays statistics for calls that have attempted RSVP reservation.
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call start
call start
To force the H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer, use
the call start command in voice-class configuration mode. To restore the default condition, use the no
form of this command.
call start {fast | slow | system}
no call start
Syntax Description
fast
Gateway uses H.323 Version 2 (fast connect) procedures.
slow
Gateway uses H.323 Version 1 (slow connect) procedures.
system
Gateway defaults to the voice service configuration that is defined using the
h323 call start command in voice-service configuration mode.
Defaults
The default is system.
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600 series routers, 3600
series, and 7200 series and on the AS5300, AS5800, and MC3810.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
In Cisco IOS Release 12.1(3)XI and later, H.323 Voice over IP (VoIP) gateways by default use H.323
Version 2 (fast connect) for all calls, including those initiating RSVP. Previously, gateways used only
slow connect procedures for RSVP calls. To enable Cisco IOS Release 12.1(3)XI gateways to be
backward compatible with earlier releases of Cisco IOS Release 12.1 T, the call start command allows
the originating gateway to initiate calls using slow connect.
The call start command is configured as part of the voice class assigned to an individual VoIP dial peer.
It takes precedence over the h323 call start voice-service configuration command, which applies
globally to all VoIP calls, unless the system keyword is selected. If the system keyword is used for the
call start voice-class command, the gateway defaults to the voice-service configuration.
Examples
The following example selects slow connect for voice class 1000:
voice class h323 1000
call start slow
!
dial-peer voice 210 voip
voice-class h323 1000
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call-waiting
call-waiting
To enable call waiting, use the call-waiting command in interface configuration mode. To disable call
waiting, use the no form of this command.
call-waiting
no call-waiting
Syntax Description
This command has no arguments or keywords.
Defaults
Call waiting is enabled.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command is applicable to Cisco 800 series routers.
You must specify this command when creating a dial peer. This command will not work if it is not
specified within the context of a dial peer. For information on creating a dial peer, refer to the Cisco 800
Series Routers Software Configuration Guide.
Examples
The following example disables call waiting:
no call-waiting
Related Commands
Command
Description
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN
directory number (depending on the dial plan) to be used for a dial peer.
dial peer voice
Enters dial peer configuration mode, defines the type of dial peer, and
defines the tag number associated with a dial peer.
port (dial peer)
Enables an interface on a PA-4R-DTR port adapter to operate as a
concentrator port.
ring
Sets up a distinctive ring for telephones, fax machines, or modems connected
to a Cisco 800 series router.
show dial peer voice
Displays configuration information and call statistics for dial peers.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
called-number (dial-peer)
called-number (dial-peer)
To enable an incoming Voice over Frame Relay (VoFR) call leg to get bridged to the correct plain old
telephone service (POTS) call leg when a static FRF.11 trunk connection is used, use the called-number
command in dial peer configuration mode. To disable a static trunk connection, use the no form of this
command.
called-number string
no called-number
Syntax Description
string
Defaults
This command is disabled.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.0(4)T
This command was introduced on the Cisco 2600 and 3600 series.
Usage Guidelines
A string of digits, including wildcards, that specifies the telephone
number of the voice port dial peer.
This command applies to the Cisco 2600 and 3600 series routers only. It is ignored on the Cisco MC3810
multiservice concentrator and on the Cisco 7200 series routers.
The called-number command is used only when the dial peer type is VoFR and you are using the
frf11-trunk (FRF.11) session protocol. It is ignored at all times on the Cisco MC3810 multiservice
concentrator and on all other platforms when using the Cisco-switched session protocol.
Because FRF.11 does not provide any end-to-end messaging to manage a trunk, the called-number
command is necessary to allow the router to establish an incoming trunk connection. The E.164 number
is used to find a matching dial peer during call setup.
Examples
The following example shows how to configure a Cisco 2600 series routers or 3600 series router for a
static FRF.11 trunk connection to a specific telephone number (555-2150), beginning in global
configuration mode:
voice-port 1/0/0
connection trunk 55Router0
exit
dial-peer voice 100 pots
destination pattern 5552150
exit
dial-peer voice 200 vofr
session protocol frf11-trunk
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called-number (dial-peer)
called-number 5552150
destination pattern 55Router0
Related Commands
Command
Description
codec (dial peer)
Specifies the voice coder rate of speech for a VoFR dial peer.
connection
Specifies a connection mode for a voice port.
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN
directory number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (VoFR)
Enables the generation of FRF.11 Annex A frames for a dial peer.
fax-rate
Establishes the rate at which a fax will be sent to the specified dial peer.
preference
Indicates the preferred order of a dial peer within a rotary hunt group.
session protocol
Establishes a session protocol for calls between the local and remote routers
via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination
gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
vad (dial peer)
Enables voice-activated dialing (VAD) for the calls using a particular dial
peer.
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caller-id
caller-id
To enable caller ID, use the caller-id command in dial peer configuration mode. To disable caller ID,
use the no form of the command.
caller-id
no caller-id
Syntax Description
This command contains no arguments or keywords.
Defaults
Caller ID is disabled.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1.(2)XF
This command was introduced on the Cisco 800 series routers.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
This command is available on Cisco 800 series routers that have plain old telephone service (POTS)
ports. The command is effective only if you subscribe to caller ID service. If you enable caller ID on a
router without subscribing to the caller ID service, caller ID information does not appear on the
telephone display.
The configuration of caller ID must match the device connected to the POTS port. That is, if a telephone
supports the caller ID feature, use the command caller-id to enable the feature. If the telephone does not
support the caller ID feature, use the command default or disable the caller ID feature. Odd ringing
behavior might occur if the caller ID feature is disabled when it is a supported telephone feature or
enabled when it is not a supported telephone feature.
Note
Examples
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
The following example enables a router to use the caller ID feature:
dial-peer voice 1 pots
caller-id
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caller-id
Related Commands
Command
Description
block-caller
Configures call blocking on caller ID.
debug pots csm csm
Activates events from which an application can determine and display
the status and progress of calls to and from POTS ports.
isdn i-number
Configures several terminal devices to use one subscriber line.
pots call-waiting
Enables local call waiting on a router.
registered-caller ring
Configures the Nariwake service-registered caller ring cadence.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
caller-id alerting dsp-pre-alloc
caller-id alerting dsp-pre-alloc
To statically allocate a digital signal processor (DSP) resource for receiving caller ID information for
on-hook (Type 1) Caller ID at a receiving Foreign Exchange Office (FXO) voice port, use the caller-id
alerting dsp-pre-alloc command in voice-port configuration mode. To disable the command’s effect,
use the no form of this command.
caller-id alerting dsp-pre-alloc
no caller-id alerting dsp-pre-alloc
Syntax Description
This command contains no keywords or arguments.
Defaults
No pre-allocation of DSP resources
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for the Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
The caller-id alerting dsp-pre-alloc command may be required on an FXO port if the central office uses
line polarity reversal to signal the start of Caller-ID information transmission. Pre-allocating a DSP
allows the DSP to listen for Caller-ID information continuously without requiring an alerting signal from
the CO.
This command is the FXO counterpart to the caller-id alerting line-reversal command, which is applied
to the Foreign Exchange Station (sending) end of the Caller-ID call.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
Note
Examples
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
The following example configures a voice port on a Cisco 2600 or 3600 router where Caller-ID
information is received:
voice-port 1/0/1
cptone US
caller-id enable
caller-id alerting line-reversal
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caller-id alerting dsp-pre-alloc
caller-id alerting dsp-pre-alloc
The following example configures a voice port on a Cisco MC3810 multiservice concentrator where
Caller-ID information is received:
voice-port 1/0
cptone northamerica
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
Related Commands
Command
Description
caller-id alerting
line-reversal
Sets the line-reversal method of Caller-ID call alerting.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
caller-id alerting line-reversal
caller-id alerting line-reversal
To set the line-reversal alerting method for Caller-ID information for on-hook (Type 1) Caller ID at a
sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting line-reversal command
in voice-port configuration mode. To disable the command’s effect, use the no form of this command.
caller-id alerting line-reversal
no caller-id alerting line-reversal
Syntax Description
This command has no keywords or arguments.
Defaults
No line-reversal alert
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for the Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This command is only required when the telephone device attached to an FXS port requires the
line-reversal method to signal the start of a Caller-ID transmission. Use it on FXS voice ports that send
Caller-ID information.
This command is the FXS counterpart to the caller-id alerting dsp-pre-alloc command, which is
applied to the FXO (receiving) end of the Caller-ID call with the line-reversal alerting method.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
Note
Examples
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
The following example configures a voice port on a Cisco 2600 or 3600 series router from which
Caller-ID information is sent:
voice-port 1/0/1
cptone US
station name A. Person
station number 4085551111
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
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caller-id alerting line-reversal
The following example configures a voice port on a Cisco MC3810 multiservice concentrator from
which Caller-ID information is sent:
voice-port 1/0
cptone northamerica
station name A. Person
station number 4085551111
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
Related Commands
Command
Description
caller-id alerting
dsp-pre-alloc
At the receiving end of a line-reversal alerting Caller-ID call, pre-allocates
DSPs for caller ID calls.
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caller-id alerting pre-ring
caller-id alerting pre-ring
To set a 250-millisecond pre-ring alerting method for caller ID information for on-hook (Type 1)
Caller ID at a sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting pre-ring
command in voice-port configuration mode. To disable the command, use the no form of this command.
caller-id alerting pre-ring
no caller-id alerting pre-ring
Syntax Description
This command has no keywords or arguments.
Defaults
No pre-ring alert
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for the Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This command is required only when the telephone device attached to an FXS port requires the pre-ring
(immediate ring) method to signal the start of caller ID transmission. Use it on FXS voice ports that send
caller ID information. This command allows the FXS port to send a short pre-ring preceding the normal
ring cadence. On an FXO port, an incoming pre-ring (immediate ring) is simply counted as a normal ring
using the caller-id alerting ring command.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
Note
Examples
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
The following example configures a voice port on a Cisco 2600 or 3600 series router from which
caller ID information is sent:
voice-port 1/0/1
cptone US
station name A. Person
station number 4085551111
caller-id alerting pre-ring
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caller-id alerting pre-ring
The following example configures a voice port on a Cisco MC3810 multiservice concentrator from
which caller ID information is sent:
voice-port 1/0
cptone northamerica
station name A. Person
station number 4085551111
caller-id alerting pre-ring 1
Related Commands
Command
Description
caller-id alerting
line-reversal
Enables caller ID operation and sets the line-reversal alerting type at an
FXS port.
caller-id alerting ring
Enables caller ID operation and sets an alerting ring type at an FXO or FXS
port.
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caller-id alerting ring
caller-id alerting ring
To set the ring-cycle method for receiving caller ID information for on-hook (Type 1) Caller ID at a
receiving Foreign Exchange Office (FXO) or a sending Foreign Exchange Station (FXS) voice port, use
the caller-id alerting ring command in voice-port configuration mode. To set the command to the
default, use the no form of this command.
caller-id alerting ring {1 | 2}
no caller-id alerting ring
Syntax Description
1
Use this setting if your telephone service provider specifies it to provide
caller ID alerting (display) after the first ring at the receiving station. This
is the most common setting.
2
Use this setting if your telephone service provider specifies it to provide
caller ID alerting (display) after the second ring. This setting is used in
Australia, where the caller ID information is sent following two short rings
(double-pulse ring).
Defaults
The default value is 1.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for the Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This setting is determined by the Bellcore/Telcordia or ETSI standard that your telephone service
provider uses for caller ID. Use it on FXO loop-start and ground-start voice ports where caller ID
information arrives and on FXS voice ports from which caller ID information is sent.
This setting must match on the sending and receiving ends on both ends of the telephone line connection.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
Note
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
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caller-id alerting ring
Examples
The following example configures a Cisco 2600 or 3600 series router voice port where caller ID
information is received:
voice-port 1/0/1
cptone US
caller-id alerting ring 1
The following example configures a Cisco 2600 or 3600 series router voice port from which caller ID
information is sent:
voice-port 1/0/1
cptone northamerica
station name A. Person
station number 4085551111
caller-id alerting ring 1
The following example configures a Cisco MC3810 multiservice concentrator voice port where caller
ID information is received:
voice-port 1/0
cptone northamerica
caller-id alerting ring 1
The following example configures a Cisco MC3810 multiservice concentrator voice port from which
caller ID information is sent:
voice-port 1/0
cptone northamerica
station name A. Person
station number 4085551111
caller-id alerting ring 1
Related Commands
Command
Description
caller-id alerting
line-reversal
Enables caller ID operation and sets the line-reversal alerting type at an
FXS port.
caller-id alerting
pre-ring
Enables caller ID operation and sets the pre-ring alerting method at an FXS
port.
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caller-id attenuation
caller-id attenuation
To set the attenuation for caller ID at a receiving Foreign Exchange Office (FXO) voice port, use the
caller-id attenuation command in voice-port configuration mode. To set the command to the default,
use the no form of this command.
caller-id attenuation [attenuation]
no caller-id attenuation
Syntax Description
attenuation
Defaults
The default value is 14 decibels (dB), signal level of –14 dBm.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for the Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
Specifies the attenuation. Valid values are from 0 to 64.
Use this setting to specify the attenuation for a caller ID FXO port. If the setting is not used, the
attenuation is set to 14 decibels (dB), signal level of –14 dBm.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
Note
Examples
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
The following example configures a Cisco 2600 or 3600 series router voice port where caller ID
information is received:
voice-port 1/0/1
cptone US
caller-id attenuation 0
The following example configures a Cisco MC3810 multiservice concentrator voice port where caller ID
information is received:
voice-port 1/0
cptone northamerica
caller-id attenuation 0
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caller-id block
caller-id block
To request the blocking of the display of caller ID information at the far end of a call from calls
originated at a Foreign Exchange Station (FXS) port, use the caller-id block command in voice-port
configuration mode at the originating FXS voice port. To allow the display of caller ID information, use
the no form of this command.
caller-id block
no caller-id block
Syntax Description
This command has no keywords or arguments.
Defaults
No blocking of caller ID information
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This command is used on FXS voice ports that are used to originate on-net telephone calls. This
command affects all calls sent to a far-end FXS station from the configured originating FXS station.
Calling number and called number are provided in the H.225 setup message for VoIP, through the H.225
Octet 3A field. Calling name information is included in a display information element.
Note
Cisco-switched calls using Voice over Frame Relay (VoFR) and Voice over ATM (VoATM)
carry calling party information in the Cisco proprietary setup message. For standards-based,
point-to-point VoFR (FRF.11) trunks where transparent signaling is applied for FXS-to-FXO
calls, only pass-through of in-band Automatic Number Identification (ANI) is supported.
ANI information is always unblocked for these communications. Interface technology using
transparent channel associated signaling (CAS) can support only ANI through Feature Group
D (in-band MF signaling). The Caller ID feature cannot be used with fixed point-to-point
trunk connects created using the connection trunk command.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
Note
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
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caller-id block
Examples
The following example configures a Cisco 2600 or 3600 series router voice port from which caller ID
information is sent:
voice-port 1/0/1
cptone US
station name A. Person
station number 4085551111
caller-id block
The following example configures a Cisco MC3810 multiservice concentrator voice port from which
caller ID information is sent:
voice-port 1/0
cptone northamerica
station name A. Person
station number 4085551111
caller-id block
Related Commands
Command
Description
caller-id enable
Enables caller ID operation.
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caller-id enable
caller-id enable
To allow the sending or receiving of caller ID information, use the caller-id enable command in
voice-port configuration mode at the sending Foreign Exchange Station (FXS) voice port or the
receiving Foreign Exchange Office (FXO) voice port. To disable the sending or receiving of caller ID
information, use the no form of this command, which also clears all other caller ID configuration settings
for the voice port.
caller-id enable
no caller-id enable
Syntax Description
This command has no keywords or arguments.
Defaults
No sending or receiving of caller ID information
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(2)XH
This command was implemented for the Cisco MC3810 multiservice
concentrator and for Cisco 2600 and 3600 series routers.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This command applies to FXS voice ports that send caller ID information and to FXO ports that receive
caller ID information. Calling number and called number are provided in the H.225.0 setup message for
VoIP, through the H.225.0 Octet 3A field. Calling name information is included in a display information
element.
Note
Cisco-switched calls using Voice over Frame Relay (VoFR) and Voice over ATM (VoATM)
carry calling party information in the Cisco proprietary setup message. For standards-based,
point-to-point VoFR (FRF.11) trunks where transparent signaling is applied for FXS-to-FXO
calls, only pass-through of in-band Automatic Number Identification (ANI) is supported.
ANI information is always unblocked for these communications. Interface technology using
transparent channel associated signaling (CAS) can support only ANI through Feature Group
D (in-band MF signaling). The Caller ID feature cannot be used with fixed point-to-point
trunk connects created using the connection trunk command.
If the station name, station number, or a caller-id alerting command is configured on the voice port,
these automatically enable caller ID, and the caller-id enable command is not necessary.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600
series routers.
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caller-id enable
Note
Examples
Specific hardware is required to provide full support for the Caller ID features. To determine
support for these features in your configuration, review the appropriate hardware
documentation and data sheets. This information is available on Cisco.com.
The following example configures a Cisco 2600 or 3600 series router voice port where caller ID
information is received:
voice-port 1/0/1
cptone US
caller-id enable
The following example configures a Cisco 2600 or 3600 series router voice port from which caller ID
information is sent:
voice-port 1/0/1
cptone northamerica
station name A. Person
station number 4085551111
caller-id enable
The following example configures a Cisco MC3810 multiservice concentrator voice port where caller ID
information is received:
voice-port 1/0
cptone northamerica
caller-id enable
The following example configures a Cisco MC3810 multiservice concentrator voice port from which
caller ID information is sent:
voice-port 1/0
cptone northamerica
station name A. Person
station number 4085551111
caller-id enable
Related Commands
Command
Description
caller-id alerting line-reversal Enables caller ID operation and sets the line-reversal alerting type
at an FXS port.
caller-id alerting pre-ring
Enables caller ID operation and sets the pre-ring alerting method at
an FXS port.
caller-id alerting ring
Enables caller ID operation and sets an alerting ring type at an FXO
or FXS port.
caller-id block
Disables the sending of caller ID information from an FXS port.
station name
Enables caller ID operation and sets the name sent from an FXS
port.
station number
Enables caller ID operation and sets the number sent from an FXS
port.
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calling-number outbound
calling-number outbound
To specify automatic number identification (ANI) to be sent out when T1-channel associated signaling
(T1-CAS) Feature Group D-Exchange Access North American (FGD-EANA) is configured as the
signaling type, use the calling-number outbound command in dial peer or voice-port configuration
mode. To disable the calling-number outbound command, use no form of this command.
calling-number outbound {range string1 string2 | sequence string1... string5| null}
no calling-number outbound {range string1 string2 | sequence string1 ... string5| null}
Syntax Description
range
Generates the sequence of ANI by rotating through the specified range
(string1 to string2).
sequence
Configures a sequence of discrete strings (string1... string5) to be passed out
as ANI for successive calls using the peer.
null
Suppresses ANI. If used, no ANI will be passed when this dial peer is
selected.
string# ...
Valid E.164 telephone number strings. Strings must be of equal length and
cannot be more than 32-digits long.
Defaults
No outbound calling number is specified.
Command Modes
Dial peer configuration
Voice-port configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300 universal access server.
Usage Guidelines
This command is effective only for Feature Group D–Exchange Access North American (FGD-EANA)
signaling.
Examples
Use the calling-number outbound command to enable or disable the passing of ANI on T1-CAS
FGD-EANA configured T1 interface for outgoing calls. Syntax for this command is the same for both
voice-port mode and dial peer mode. Examples are given for both modes.
calling-number outbound Range
calling-number outbound range string1 string2
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calling-number outbound
The values string1 and string2 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 less than four digits in length, then
entire strings will be used.
ANI will be generated by using the 408555 prefix and by rotating through 6000 to 6001 for each call
using this peer.
Dial peer configuration mode:
dial-peer voice 1 pots
calling-number outbound range 4085556000 4085556001
calling Number Outbound is effective only for fgd_eana signaling
Voice-port configuration mode:
voice-port 1:D
calling-number outbound range 4085556000 4085556005
Calling Number Outbound is effective only for fgd_eana signaling
calling-number outbound Sequence
calling-number outbound sequence string1 string2 string3
string4 string5
This option configures a sequence of discrete strings (string1...string5) to be passed out as ANI for
successive calls using the peer. The limit is five strings. All strings must be valid E.164 numbers, up to
32 digits in length.
Dial peer configuration mode:
dial-peer voice 1 pots
calling-number outbound sequence 6000 6006 4000 5000 5025
Calling Number Outbound is effective only for fgd_eana signaling
Voice-port configuration mode:
voice-port 1:D
calling-number outbound sequence 6000 6006 4000 5000 5025
Calling Number Outbound is effective only for fgd_eana signaling
calling-number outbound Null
calling-number outbound null
This option suppresses ANI. If used, no ANI will be passed when this dial peer is selected.
Dial peer configuration mode:
dial-peer voice 1 pots
calling-number outbound null
Calling Number Outbound is effective only for fgd_eana signaling
Voice-port configuration mode:
voice-port 1:D
calling-number outbound null
Calling Number Outbound is effective only for fgd_eana signaling
Related Commands
Command
Description
info-digits string1
Configures two information digits to be prepended to the ANI string.
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cap-list vfc
cap-list vfc
To add a voice codec overlay file to the capability file list, use the cap-list vfc command in global
configuration mode. To disable a particular codec overlay file that has been added to the capability list,
use the no form of this command.
cap-list filename vfc slot-number
no cap-list filename vfc slot-number
Syntax Description
filename
Identifies the codec file stored in voice feature card (VFC)
Flash memory.
slot-number
Identifies the slot where the VFC is installed. Valid values are
0, 1, and 2.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
11.3 NA
This command was introduced on the Cisco AS5300 universal
access server.
Usage Guidelines
When VCWare is unbundled, it automatically adds DSPWare to Flash memory, creates both the
capability and default file lists, and populates these lists with the default files for the particular version
of VCWare. The capability list defines the available voice codecs for H.323 capability negotiation. Use
the cap-list vfc command to add the indicated voice codec overlay file (defined by filename) to the
capability file list in Flash memory.
Examples
The following example adds the following codec to the list included in Flash memory:
config terminal
cap-list cdc-g711-1.0.14.0.bin vfc 0
Related Commands
Command
Description
default-file vfc
Specifies an additional (or different) file from the ones in the
default file list and stored in VFC Flash memory.
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card type
card type
To configure the card type on the port adapter of the Cisco 7200 series routers router, use the card type
command in global configuration mode. To restore the default value, use the no form of this command.
card type {t1 | e1} slot [bay]
no card type
Syntax Description
t1
Specifies T1 connectivity of 1.544 Mbps through the telephone
switching network, using AMI or B8ZS coding.
e1
Specifies a wide-area digital transmission scheme used predominately
in Europe that carries data at a rate of 2.048 Mbps.
slot
Slot number of the interface.
bay
(Optional) Card interface bay number in a slot (route/switch processor
[RSP] platform only).
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)XE
This command was introduced.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
The following example configures T1 data transmission on port 1 on the Cisco 7200 series routers router:
card type t1 1/0
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ccm-manager application redundant-link port
ccm-manager application redundant-link port
To configure the port number for the redundant link application, use the ccm-manager application
redundant-link port command in global configuration mode. To disable the configuration, use the no
form of this command.
ccm-manager application redundant-link port number
no ccm-manager application
Syntax Description
number
Defaults
Port 2428
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced with Cisco CallManager Version 3.0 and the
Cisco Voice Gateway 200 (VG200).
12.2(2)XA
The command was implemented on Cisco 2600 series and Cisco 3600 series.
12.2(4)T
The command was integrated into Cisco IOS Release 12.2(4)T.
Port number for the transport protocol. The protocol may be the User Data
Protocol (UDP), Reliable User Datagram Protocol (RDUP), or Transmission
Control Protocol (TCP). Permitted values are from 0 to 65535, and it must not be
a well-known reserved port number.
Usage Guidelines
This command is optional. Use this command only when defining an application-specific port other than
the default.
Examples
In the following example, the port number of the redundant link application is 2429:
ccm-manager application redundant-link port 2429
Related Commands
Command
Description
ccm-manager
redundant-host
Configures the IP address or the DNS name of up to two backup
Cisco CallManagers.
ccm-manager
switchback
Configures the switchback mode that determines when the primary
Cisco CallManager will be used if it becomes available again while a backup
Cisco CallManager is being used.
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ccm-manager mgcp
ccm-manager mgcp
To allow a gateway to communicate with the Cisco CallManager by means of Media Gateway Control
Protocol (MGCP) and supply redundant services, use the ccc-manager mgcp command in global
configuration mode. To disable this command, use the no form of this command.
ccm-manager mgcp
no ccm-manager mgcp
Syntax Description
This command has no arguments or keywords.
Defaults
Cisco CallManager does not communicate with the gateway through MGCP
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced with Cisco CallManager Version 3.0 and the Cisco
Voice Gateway 200 (VG200).
12.2(2)XA
The command was implemented on Cisco 2600 series and Cisco 3600 series routers.
12.2(4)T
The command was integrated into Cisco IOS Release 12.2(4)T.
Usage Guidelines
This command sets the gateway to MGCP mode. In MGCP mode, the gateway can communicate with
the Cisco CallManager through MGCP, and it can enable redundancy when a backup Cisco CallManager
is available.
Examples
In the following example, support for Cisco CallManager and redundancy is enabled within MGCP:
ccm-manager mgcp
Related Commands
Command
Description
mgcp
Enables Media Gateway Control Protocol mode.
ccm-manager
redundant-host
Configures the IP address or the DNS name of up to two backup
Cisco CallManagers.
ccm-manager
switchback
Configures the switchback mode that determines when the primary
Cisco CallManager will be used if it becomes available again while a backup
Cisco CallManager is being used.
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ccm-manager redundant-host
ccm-manager redundant-host
To configure the IP address or the DNS name of up to two backup Cisco CallManagers, use the
ccm-manager redundant-host command in global configuration mode. To disable the configuration of
a backup Cisco CallManager, use the no form of this command.
ccm-manager redundant-host {ip-address | DNS-name} [ip-address | DNS-name]
no ccm-manager redundant-host {ip-address | DNS-name} [ip-address | DNS-name]
Syntax Description
ip-address
Internet protocol address of the backup Cisco CallManager.
DNS-name
Domain name system of the backup Cisco CallManager.
Defaults
If you do not configure a backup Cisco CallManager, the redundancy feature is considered to be off
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced with Cisco CallManager Version 3.0 and the
Cisco Voice Gateway 200 (VG200).
12.2(2)XA
The command was implemented on Cisco 2600 series and Cisco 3600 series.
The DNS-name argument was added.
12.2(4)T
The command was integrated into Cisco IOS Release 12.2(4)T.
Usage Guidelines
You must configure one backup Cisco CallManager, and you can configure a maximum of two backup
Cisco CallManagers. The list of IP addresses or DNS names is an ordered list. The Cisco CallManager
defined in the mgcp call-agent command has the highest priority (that is, it is the primary
Cisco CallManager). The gateway selects a Cisco CallManager based on the order in which it appears
in this list.
Examples
In the following example, the IP address of the backup Cisco CallManager is 10.0.0.50:
ccm-manager redundant-host 10.0.0.50
Related Commands
Command
Description
ccm-manager application Configures the port number for the redundant link application.
ccm-manager switchback Configures the switchback mode that determines when the primary
Cisco CallManager will be used if it becomes available again while a
backup Cisco CallManager is being used.
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ccm-manager switchback
ccm-manager switchback
To specify when to use the primary Cisco CallManager once it becomes available again, use the
ccm-manager switchback command in global configuration mode. To disable switchback, use the no
form of this command.
ccm-manager switchback {graceful | immediate | schedule-time hh:mm | uptime-delay minutes}
no ccm-manager switchback
Syntax Description
graceful
After the last active call ends (when there is no voice call in setup mode on
the gateway), control returns to the primary Cisco CallManager.
immediate
Regardless of the current conditions, when the TCP link to the primary
Cisco CallManager is established, control switches back immediately to the
primary Cisco CallManager.
schedule-time hh:mm
At a specified hour and minute, based on a 24-hour clock, control returns to
the primary Cisco CallManager. If the specified time is earlier than the
current time, the switchback occurs at the specified time on the following
day.
uptime-delay minutes
When the primary Cisco CallManager has run for a specified number of
minutes after a network connection has been reestablished to that
CallManager, control is returned to the primary Cisco CallManager.
Permitted values are from 1 to 1440 (from 1 minute to 24 hours).
Defaults
Graceful
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced with Cisco CallManager Version 3.0 and the Cisco
Voice Gateway 200 (VG200).
12.2(2)XA
The command was implemented on Cisco 2600 series and Cisco 3600 series routers.
12.2(4)T
The command was integrated into Cisco IOS Release 12.2(4)T.
Usage Guidelines
This command allows you to configure switchback to the higher priority Cisco CallManager when it
becomes available. Switchback allows call control to revert back to the original (primary)
Cisco CallManager once service has been restored.
Examples
In the following example, the primary Cisco CallManager will be used as soon as it becomes available:
ccm-manager switchback immediate
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ccs connect (interface)
ccs connect (interface)
Command
Description
ccm-manager application
Configures the port number for the redundant link application.
ccm-manager
redundant-host
Configures the IP address or the DNS name of up to two backup
Cisco CallManagers.
ccm-manager
Manually redirects a Cisco 2600 series or Cisco 3600 series router to the
switchover-to-backup
backup Cisco CallManager.
To configure a common channel signaling (CCS) connection on an interface configured to support CCS
frame forwarding, use the ccs connect command in interface configuration mode. To disable the CCS
connection on the interface, use the no form of this command.
ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name ] [ cidnumber]
no ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cidnumber]
Syntax Description
serial
Makes a serial CCS connection for Frame Relay.
atm
Makes an ATM CCS connection for ATM.
dlci
(Optional) Specifies the data link connection identifier (DLCI) number.
pvc vpi/vci
(Optional) Specifies the permanent virtual circuit (PVC) virtual path
identifier or virtual channel identifier. Acceptable values are from 0 to
255; the slash is required.
pvc name
(Optional) Specifies the PVC string that names the PVC for recognition.
cidnumber
(Optional) If you have executed the ccs encap frf11 command, the
cidnumber option allows you to specify any channel identification (CID)
number from 5 to 255.
Defaults
No CCS connection is made.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(2)T
This command was introduced for the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
The CID syntax was added; the dlci keyword and vcd options were
removed.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
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ccs connect (interface)
Usage Guidelines
Use this command to configure a CCS connection. If the CCS connection is over Frame Relay, specify
a serial interface and the DLCI. If the CCS connection is over ATM, specify atm, the interface number
(0), and the PVC. If you have executed the ccs encap frf11 command, the cidnumber option allows you
to specify any CID from 5 to 255. If you do not issue the ccs encap frf11 command, Cisco encapsulation
is used, and any CID value other than 254 is ignored.
Note
CDP and keepalives are disabled by default on a D-channel interface.
This configuration is applicable to only the MC3810 multiservice concentrator.
Examples
To configure a frame relay CCS frame-forwarding connection on DLCI 100 by using the default CID of
254, enter the following command:
ccs connect serial 1 100
or:
ccs connect serial 1 100 10
To configure a CCS frame-forwarding connection over an ATM PVC, enter the following command:
ccs connect atm0 pvc 100/10
or:
ccs connect atm0 pvc 10/100 21
or:
ccs connect atm0 pvc mypvc_10 21
To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 using a CID of 110, enter
the following command:
ccs connect serial 1 100 110
Related Commands
Command
Description
ccs encap frf11
Allows the specification of the standard Annex-C FRF.11 format.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ccs connect (controller)
ccs connect (controller)
To configure a common channel signaling (CCS) connection on an interface configured to support CCS
frame forwarding, use the ccs connect command in controller configuration mode. To disable the CCS
connection on the interface, use the no form of this command.
ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cidnumber]
no ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cidnumber]
Syntax Description
serial
Makes a serial CCS connection for Frame Relay.
atm
Makes an ATM CCS connection.
dlci
(Optional) Specifies the data link connection identifier (DLCI) number.
pvc vpi/vci
(Optional) Specifies the permanent virtual circuit (PVC) virtual path
identifier or virtual channel identifier. Acceptable values are from 0 to
255; the slash is required.
pvc name
(Optional) Specifies the PVC string that names the PVC for recognition.
cidnumber
(Optional) If you have executed the ccs encap frf11 command, the
cidnumber option allows you to specify any channel identification (CID)
number from 5 to 255.
Defaults
No CCS connection is made.
Command Modes
Controller configuration
Command History
Release
Modification
12.0(2)T
This command was introduced for the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
The CID syntax was added; the dlci keyword and vcd options were
removed.
12.1(2)T
The CID syntax addition and removal of the dlci keyword and vcd
options were integrated into Cisco IOS Release 12.1(2)T.
12.1(2)XH
This command was supported on the Cisco 2600 series routers, Cisco
3600 series routers, Cisco 7200 series routers, and Cisco 7500 series
routers.
12.1(3)T
Support on the Cisco 2600 series routers, Cisco 3600 series routers,
Cisco 7200 series routers, and Cisco 7500 series routers was integrated
into Cisco IOS Release 12.1(3)T.
Usage Guidelines
Use this command to configure a CCS connection. If the CCS connection is over Frame Relay, specify
a serial interface and the DLCI. If the CCS connection is over ATM, specify atm, the slot number (0 only
on the Cisco MC3810), and the PVC.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ccs connect (controller)
If you have executed the ccs encap frf11 command, the cidnumber option allows you to specify any CID
from 5 to 255. If you do not issue the ccs encap frf11 command, Cisco encapsulation is used, and any
CID value other than 254 is ignored.
Note
Examples
CDP and keepalives are disabled by default on a D-channel interface.
To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 by using the default CID
of 254, enter the following command:
ccs connect serial 1 100
or:
ccs connect serial 1 100 10
To configure a CCS frame-forwarding connection over an ATM PVC, enter the following command:
ccs connect atm0 pvc 100/10
or:
ccs connect atm0 pvc 10/100 21
or:
ccs connect atm0 pvc mypvc_10 21
To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 using a CID of 110, enter
the following command:
ccs connect serial 1 100 110
Related Commands
Command
Description
ccs encap frf11
Allows the specification of the standard Annex-C FRF.11 format.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ccs encap frf11
ccs encap frf11
To configure the common channel signaling (CCS) packet encapsulation format for FRF.11, use the ccs
encap frf11 command in interface configuration mode. To disable ccs encapsulation for FRF11, use the
no form of this command.
ccs encap frf11
no ccs encap frf11
Syntax Description
This command has no keywords or arguments.
Defaults
By default, the format is a Cisco packet format, using a channel ID (CID) of 254.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced for the Cisco MC3810 multiservice
concentrator.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(2)XH
This command was supported on the Cisco 2600 series routers, 3600
series, 7200 series, and 7500 series routers.
12.1(3)T
The Cisco 2600 series routers, 3600 series, 7200 series, and 7500
series router support was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This command allows the specification of the standard Annex-C format. Use this command to define the
packet format for the CCS packet; it places the FRF.11 Annex-C (Data Transfer Syntax) standard header
on the CCS packets only.
Once the ccs encap frf11 command is executed, you can use the ccs connect command to specify a CID
other than 254.
Examples
The following example shows how to configure a serial interface for Frame Relay:
interface Serial1:15
ccs encap frf11
ccs connect Serial0 990 100
Related Commands
Command
Description
mode ccs frame-forwarding
Set to forward frames on the controller.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces cell-loss-integration-period
ces cell-loss-integration-period
To set the circuit emulation service (CES) cell-loss integration period, use the ces
cell-loss-integration-period command in interface configuration mode. To delete the cell-loss
integration period, use the no form of this command.
ces cell-loss-integration-period period
no ces cell-loss-integration-period period
Syntax Description
period
Defaults
2500 milliseconds
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810
multiservice concentrator.
Usage Guidelines
Time, in milliseconds, for the cell-loss integration period.
Possible values are from 1 to 2,147,483,647.
This command applies to ATM configuration on the Cisco MC3810 multiservice concentrator.
This command is supported on serial ports 0 and 1 with encapsulation atm-ces.
Examples
The following example configures the CES cell-loss integration period on serial port 0 to 1056:
interface serial 0
ces cell-loss-integration-period 1056
Related Commands
Command
Description
cbr
Configures the CBR for the ATM CES for an ATM PVC on the
Cisco MC3810 multiservice concentrator.
ces clockmode synchronous
Configures the ATM CES synchronous clock mode on the
Cisco MC3810 multiservice concentrator.
ces connect
Maps the CES service to an ATM PVC on the Cisco MC3810
multiservice concentrator.
ces initial-delay
Configures the size of the receive buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
ces max-buf-size
Configures the send buffer of a CES circuit on the Cisco
MC3810 multiservice concentrator.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces cell-loss-integration-period
Command
Description
ces partial-fill
Configures the number of user octets per cell for the ATM CES
on the Cisco MC3810 multiservice concentrator.
ces service
Configures the ATM CES type on the Cisco MC3810
multiservice concentrator.
encapsulation atm-ces
Enables CES ATM encapsulation on the Cisco MC3810
multiservice concentrator.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces-clock
ces-clock
To configure the clock for the CES interface, use the ces-clock command in controller configuration
mode. To disable the ces clock, use the no form of this command.
ces-clock {adaptive | srts | synchronous}
no ces-clock {adaptive | srts | synchronous}
Syntax Description
adaptive
Adjusts output clock on a received ATM Adaptation Layer 1 (AAL1) on
first-in, first-out basis. Use in unstructured mode.
srts
Sets the clocking mode to synchronous residual time stamp.
synchronous
Configures the timing recovery to synchronous for structured mode.
Defaults
The default setting is synchronous.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(2)T
This command was introduced.
Usage Guidelines
This command is used on Cisco 3600 series routers that have OC-3/STM-1 ATM CES network modules.
Examples
The following example configures the CES clock mode for synchronous residual time stamp:
ces-clock srts
Related Commands
Command
Description
controller
Configures the T1 or E1 controller.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces clockmode synchronous
ces clockmode synchronous
To configure the ATM circuit emulation service (CES) synchronous clock mode, use the ces clockmode
synchronous command in interface configuration mode. To restore the default value, use the no form of
this command.
ces clockmode synchronous
no ces clockmode synchronous
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
This command applies to ATM configuration on the Cisco MC3810 multiservice concentrator.
This command maps into the transmit clock source of the constant bit rate (CBR) interface. This
command is supported on serial ports 0 and 1 when set for CES ATM encapsulation.
Examples
The following example sets the ATM CES clock to synchronous mode on serial port 0:
interface serial 0
ces clockmode synchronous
Related Commands
Command
Description
encapsulation atm-ces Enables CES ATM encapsulation on the Cisco MC3810 multiservice
concentrator.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces connect
ces connect
To map the circuit emulation service (CES) service to an ATM permanent virtual circuit (PVC) on the
Cisco MC3810 multiservice concentrator, use the ces connect command in interface configuration
mode. To delete the CES map to the ATM PVC, use the no form of this command.
ces connect atm-interface pvc {name | [vpi/] vci}
no ces connect atm-interface pvc {name | [vpi/] vci}
Syntax Description
atm-interface
Number of the ATM interface. The only valid option on the
Cisco MC3810 multiservice concentrator is ATM0.
pvc
Specifies that the connection is to an ATM PVC.
name
The name of the ATM PVC.
vpi/
(Optional) The virtual path identifier value.
vci
The virtual channel identifier value.
Defaults
No ATM interface is defined.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810
multiservice concentrator.
Usage Guidelines
This command applies to ATM configuration on the Cisco MC3810.
This command is supported on serial ports 0 and 1. The ATM interface must be configured to
encapsulation atm-ces, and the vpi/vci must be defined on the interface.
Examples
The following example maps the CES service to PVC 20 on ATM port 0:
ces connect atm0 pvc 20
Related Commands
Command
Description
cbr
Configures the CBR for the ATM CES for an ATM PVC on the
Cisco MC3810 multiservice concentrator.
ces cell-loss-integration-period
Sets the CES cell-loss integration period on the Cisco MC3810
multiservice concentrator.
ces clockmode synchronous
Configures the ATM CES synchronous clock mode on the
Cisco MC3810 multiservice concentrator.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces connect
Command
Description
ces initial-delay
Configures the size of the receive buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
ces max-buf-size
Configures the send buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
ces partial-fill
Configures the number of user octets per cell for the ATM CES
on the Cisco MC3810 multiservice concentrator.
ces service
Configures the ATM CES type on the Cisco MC3810
multiservice concentrator.
encapsulation atm-ces
Enables CES ATM encapsulation on the Cisco MC3810
multiservice concentrator.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces initial-delay
ces initial-delay
To configure the size of the receive buffer of a circuit emulation service (CES) circuit, use the
ces initial-delay command in interface configuration mode. To remove the initial-delay value, use the
no form of this command.
ces initial-delay bytes
no ces initial-delay bytes
Syntax Description
bytes
Defaults
4000 bytes
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810
multiservice concentrator.
The size of the receive buffer of the CES circuit. The valid
range is from 1 to 16,000 bytes. This command is used to
accommodate cell jitter on the network. Bytes received from
the ATM network are buffered by this amount before being sent
to the CES port.
Usage Guidelines
This command applies to ATM configuration on the Cisco MC3810 multiservice concentrator.
Examples
The following example shows configuration of the transmit buffer of the CES circuit to 8,000 bytes:
ces initial-delay 8000
Related Commands
Command
Description
ces cell-loss-integration-period
Sets the CES cell-loss integration period on the Cisco MC3810
multiservice concentrator.
ces clockmode synchronous
Configures the ATM CES synchronous clock mode on the
Cisco MC3810 multiservice concentrator.
ces connect
Maps the CES service to an ATM PVC on the Cisco MC3810
multiservice concentrator.
ces max-buf-size
Configures the send buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces initial-delay
Command
Description
ces partial-fill
Configures the number of user octets per cell for the ATM CES
on the Cisco MC3810 multiservice concentrator.
ces service
Configures the ATM CES type on the Cisco MC3810
multiservice concentrator.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces max-buf-size
ces max-buf-size
To configure the transmit buffer of a circuit emulation service (CES) circuit, use the ces max-buf-size
command in interface configuration mode. To delete the CES transmit buffer size, use the no form of
this command.
ces max-buf-size size
no ces max-buf-size size
Syntax Description
size
Defaults
256
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810
multiservice concentrator.
Usage Guidelines
Maximum size of the transmit buffer for the CES. Possible
values are from 80 to 1520.
This command applies to ATM configuration on the Cisco MC3810.
This command causes incoming bytes received on a CES port to be buffered by the amount configured
and sent to the ATM Adaptation Layer 1 (AAL1) process as a block of data.
This command is supported on serial ports 0 and 1 when the encapsulation atm-ces command is
enabled.
Examples
The following example shows configuration of the maximum CES reassembly buffer size to 1520:
ces max-buf-size 1520
Related Commands
Command
Description
ces cell-loss-integration-period
Sets the CES cell-loss integration period on the Cisco MC3810
multiservice concentrator.
ces clockmode synchronous
Configures the ATM CES synchronous clock mode on the
Cisco MC3810 multiservice concentrator.
ces connect
Maps the CES service to an ATM PVC on the Cisco MC3810
multiservice concentrator.
ces initial-delay
Configures the size of the receive buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces max-buf-size
Command
Description
ces partial-fill
Configures the number of user octets per cell for the ATM CES
on the Cisco MC3810 multiservice concentrator.
ces service
Configures the ATM CES type on the Cisco MC3810
multiservice concentrator.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces service
ces service
To configure the ATM circuit emulation service (CES) type, use the ces service command in interface
configuration mode. To disable the ATM CES service type, use the no form of this command.
ces service structured
no ces service structured
Syntax Description
structured
Defaults
Structured
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810
multiservice concentrator.
Usage Guidelines
Specifies that the ATM CES type is structured. Structured is the
only option supported in this release.
This command applies to ATM configuration on the Cisco MC3810.
This command is supported on serial ports 0 and 1 when the encapsulation atm-ces command is
enabled.
Examples
The following example sets the CES service to structured for serial port 0:
interface serial 0
ces service structured
Related Commands
Command
Description
ces cell-loss-integration-period
Sets the CES cell-loss integration period on the Cisco MC3810
multiservice concentrator.
ces clockmode synchronous
Configures the ATM CES synchronous clock mode on the
Cisco MC3810 multiservice concentrator.
ces connect
Maps the CES service to an ATM PVC on the Cisco MC3810
multiservice concentrator.
ces initial-delay
Configures the size of the receive buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
ces service
Command
Description
ces max-buf-size
Configures the send buffer of a CES circuit on the
Cisco MC3810 multiservice concentrator.
ces partial-fill
Configures the number of user octets per cell for the ATM CES
on the Cisco MC3810 multiservice concentrator.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
clear backhaul-session-manager group
clear backhaul-session-manager group
To reset the stastistics or traffic counters for a specified session-group, use the clear
backhaul-session-manager group command in privileged EXEC mode.
clear backhaul-session-manager group stats { all | name group-name }
Syntax Description
all
All available session-groups.
name group-name
A specified session-group.
Defaults
The statistical information accumulates.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced.
Usage Guidelines
A session is the connection between a client and a server, and a session-group is a collection of sessions
in a group to implement switchover in case of a session failure. This command clears all statistics.
Examples
To clear all statistics for all available session-groups, see the following example:
Router# clear backhaul-session-manager group stats all
Related Commands
Command
Description
show backhaul-session-manager
group
Displays status, statistics, or configuration of a
specified or all session-groups.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
clear call fallback cache
clear call fallback cache
To clear the current Impairment/Calculated Planning Impairment Factor (ICPIF) estimates for all IP
addresses or for a specific IP address in the cache, use the clear call fallback cache command in EXEC
mode.
clear call fallback cache [ip-address]
Syntax Description
ip-address
Defaults
None
Command Modes
EXEC
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600
series and on the MC3810 multiservice concentrator.
(Optional) Specifies the target IP address.
Usage Guidelines
If no IP address is specified, the command will clear the cache of all ICPIF estimates for all IP addresses.
There is no no form of this command.
Examples
The following example clears the cache of the ICPIF estimate for IP address 209.165.200.225:
clear call fallback cache 209.165.200.225
Related Commands
Command
Description
show call fallback cache
Displays the current ICPIF estimates for all IP addresses in the
cache.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
clear call fallback stats
clear call fallback stats
To clear the call fallback statistics, use the clear call fallback stats command in EXEC mode.
clear call fallback stats
Syntax Description
This command has no arguments or keywords.
Defaults
There are no defaults.
Command Modes
EXEC
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600
series and on the MC3810 multiservice concentrator.
Usage Guidelines
There is no no form of this command.
Examples
The following example clears the call fallback statistics.
clear call fallback stats
Related Commands
Command
Description
show call fallback stats
Displays the call fallback statistics.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
clear controllers call-counters
To clear the system’s DS0’s High Water Marks (HWM) and all individual controller statistics, enter the
clear controllers call-counters command in privileged EXEC mode.
clear controllers {t1 | e1} number call-counters [system-hwm | all]
Syntax Description
t1 | e1
Specifies the type of controller. If the t1 or e1 keyword is
specified, then the command acts on the individual controller
specified in number. In this case, the additional options of
system-hwm or all will not be available.
number
Clears an individual controller. Select the shelf (for Cisco
ASRouters) /slot/port number in the following format:
shelf/slot/port.
system-hwm
(Optional) Clears the system’s HWM only (not the individual
controllers).
all
(Optional) Clears all HWMs (individual DS0s and the system
total HWM). This keyword clears all controller call-counters,
including the individual controller time slots’ time in use and
the number of calls on those time slots since the last reset was
done using the clear controllers command on that controller.
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
Modification
12.0(1)T
This command was introduced.
12.1(1)T
This command was applied to voice/WAN interface cards
(VWICs) for Cisco 2600 and Cisco 3600 series routers.
12.1(2)T
This command was integrated into the Cisco AS5300 and
AS5800.
Usage Guidelines
The clear controllers call-counters all command clears the system’s DS0s HWM and all individual
controller statistics, including the controllers’ DS0s HWM, Total Calls, and Total Duration. The clear
controllers call-counters system-hwm command clears the system’s DS0s HWM and leaves all other
call-counter statistics untouched.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
Refer to the following comments for the meaning of call counters displayed before and after executing
clear controllers call-counters related commands.
•
The numbers displayed under TotalCalls for each time slot represent total calls that were connected
successfully. Note that if a first call comes in on time slot 10, then show controllers t1 call-counters
displays 1 under the TotalCalls column for time slot 10. A value of 20 displayed under TotalCalls
for time slot 10 indicates a total of 20 calls connected on time slot 10 since the last time call counters
were cleared.
TotalCalls for time slots are set to zero when clear controllers t1 1/0/0 call-counters timeslot 10
is executed or clear controllers t1 1/0/0 call-counters is executed at a controller level.
The TotalCalls figures for the time slots are not the number of active calls in the controller—they
just display the time slot usage as the number of calls connected since the last clear was done.
This is also true for the TotalDuration displayed for the time slots—it indicates the total time usage
on time slots for the calls that were completed.
•
DS0s Active: Indicates the number of active calls on the specified controller. This number indicates
the current number of calls on the controller at any given time.
•
DS0s Active High Water Mark: Indicates the peak number of calls on the controller since the last
clear controllers t1 1/0/0 call-counters command was entered. If the number of active calls “DS0s
Active” is less than DS0s HWM, then HWM remains untouched. If new calls come in and the active
DS0s are more than the HWM, then the HWM is incremented to reflect the new peak number of calls
on that controller.
This value is reset to current active DS0s when the command clear controllers t1 1/3/0
call-counters is entered. For example, in the beginning, the HWM is 0. A new call comes in. The
HWM is 1. When the next call comes in, the HWM is 2. If 20 calls come in, the HWM is 20 and
also-active DS0s are 20. If 5 calls get disconnected, the DS0 active is 15, but the HWM is 20. A
clear controllers command is input for the specified controller. The HWM is reset to 15, which is
current active DS0s. The HWM is 15 and active DS0s are also 15. Ten calls get disconnected. Active
DS0s is 5 and the HWM remains at 15 until another clear controllers command is input. If Active
DS0s exceed 15, then the HWM is updated.
•
Examples
System DS0s High Water Mark: Same point as number 3, but this value reflects the HWM at a
system level, including DS0s of all controllers.
The examples below show two controllers: 1/3/0:3 and 1/3/0:8. Note the semantics of output shown by
show controllers t1 call-counters and how clear controllers call-counters affects them.
The following example shows the system in the beginning state (with no calls):
Router# show users
Line
User
* 0 con 0
Interface
User
Host(s)
idle
Mode
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 0
DS0's Active High Water Mark: 0
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
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Idle
00:00:00
Location
Idle Peer Address
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
0
00:00:00
23
pri
0
00:00:00
T1 1/3/0:8:
DS0's Active: 0
DS0's Active High Water Mark: 0
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
0
00:00:00
23
pri
0
00:00:00
System's DS0's Active High Water Mark: 0
All the fields are zero—indicating that no calls have come in since system startup or since the last clear
was made on controllers by the clear controllers command.
The following example shows that four calls have been initiated on these controllers:
Router# show users
Line
User
* 0 con 0
tty 1/5/12
Router
tty 1/5/13
Router
tty 1/5/14
Router
tty 1/5/15
Router
Interface
User
Host(s)
idle
Async interface
Async interface
Async interface
Async interface
Mode
Idle
Location
00:00:00
00:01:05
PPP: 55.61.1.1
00:00:48
PPP: 55.62.1.1
00:00:33
PPP: 55.54.1.1
00:00:19
PPP: 55.52.1.1
Idle Peer Address
Cisco IOS Voice, Video, and Fax Command Reference
VR-161
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:01:58
23
pri
1
00:02:27
T1 1/3/0:8:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:02:14
23
pri
1
00:02:46
System's DS0's Active High Water Mark: 4
Controllers 1/3/0:3 and 1/3/0:8 both have DS0 Active 2, DS0s HWM 2. The TotalCalls field represents
the total number of calls on that time slot since the controller was cleared. This accumulates for every
call made on that time stamp and will be set to zero on individual controllers when the clear controllers
Cisco IOS Voice, Video, and Fax Command Reference
VR-162
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
t1 1/3/0:8 call-counters command is entered or at the system level when the clear controllers
call-counters all command is entered. TotalDuration is the same as TotalCalls except that it reflects the
accumulated connection time on that time slot.
This has nothing to do with active calls, If a clear controllers command is entered for a controller that
has active calls, which have been connected from the past 30 minutes, the TotalCalls and TotalDuration
are reset to zero. In the following example, the controller is 1/3/0:3, with time slots 22 and 23 connected
and active. When the clear controllers t1 1/3/0:3 call-counters command is entered, the corresponding
fields are set to zero.
The following is an example of output when the show controllers t1 call-counters command is used:
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:29:14
23
pri
1
00:29:47
Router# clear controllers t1 1/3/0:3 call-counters
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
VR-163
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
16
17
18
19
20
21
22
23
pri
pri
pri
pri
pri
pri
pri
pri
0
0
0
0
0
0
0
0
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:10
00:00:10
<<<<<<
<<<<<<
Now a call is brought down (on 1/5/12).
Router# clear line 1/5/12
[confirm]
[OK]
Router# show users
Line
User
Host(s)
* 0 con 0
idle
tty 1/5/13
Router Async interface
tty 1/5/14
Router Async interface
tty 1/5/15
Router Async interface
Interface
User
Idle
Location
00:00:00
00:03:04
PPP: 55.62.1.1
00:02:49
PPP: 55.54.1.1
00:02:35
PPP: 55.52.1.1
Mode
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:03:44
23
pri
1
00:04:14
T1 1/3/0:8:
DS0's Active: 1
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
VR-164
Idle Peer Address
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
10
11
12
13
14
15
16
17
18
19
20
21
22
23
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
0
0
0
0
0
0
0
0
0
0
0
0
1
1
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:04:00
00:03:34
System's DS0's Active High Water Mark: 4
After a call gets disconnected, only the DS0’s Active field will change to reflect the current active call
on the controller. In the above example, 1/3/0:8 Active DS0 is changed to 1.
Now, call counters are cleared for an individual controller (for 1/3/0:8):
Router# clear controllers t1 1/3/0:8 ?
call-counters T1 call counters
<cr>
Router# clear controllers t1 1/3/0:8 call-counters
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:07:46
23
pri
1
00:08:15
T1 1/3/0:8:
DS0's Active: 1
DS0's Active High Water Mark: 1
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
VR-165
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
pri
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:00
00:00:35
00:00:00
System's DS0's Active High Water Mark: 4
After clearing call counters for controller 1/3/0:8, as shown earlier, TotalCalls and TotalDuration reset.
In addition DS0’s High Water Mark is also cleared to the number of active DS0s. Whenever the DS0s’
HWM is cleared it will not reset to zero, but rather it will be set to Active DS0s for that instance. So for
1/3/0:8, the HWM is 1 after clearing since DS0 Active is 1 (1 active call). TotalDuration is 35 seconds
for time slot 22 and TotalCall is 0 since they got reset when the clear call-counters command was
entered. Total calls on this time slot will be incremented when a new call comes in on this time slot.
Use the system HWM option:
Router# clear controllers ?
T1
T1 controller
T3
T3 controller
call-counters system call counters
Router# clear controllers call-counters ?
all
all controllers
system-hwm system High Water Mark
Router# clear controllers call-counters system-hwm
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
VR-166
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:08:51
23
pri
1
00:09:21
T1 1/3/0:8:
DS0's Active: 1
DS0's Active High Water Mark: 1
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
0
00:01:39
23
pri
0
00:00:00
System's DS0's Active High Water Mark: 3
The system HWM is reset to Total number of active calls in the system, which is 3; earlier it was 4.
Bring down another two calls and check output from the command show controllers t1 call-counters:
Router# show users
Line
User
Host(s)
* 0 con 0
idle
tty 1/5/13
Router Async interface
tty 1/5/14
Router Async interface
tty 1/5/15
Router Async interface
Interface
User
Mode
Router# clear line 1/5/13
[confirm]
[OK]
Router# clear line 1/5/14
[confirm]
[OK]
Router# show users
Line
User
Host(s)
* 0 con 0
idle
tty 1/5/15
Router Async interface
Interface
User
Mode
Idle
Location
00:00:00
00:09:49
PPP: 55.62.1.1
00:09:35
PPP: 55.54.1.1
00:09:20
PPP: 55.52.1.1
Idle Peer Address
Idle
Location
00:00:00
00:09:46
PPP: 55.52.1.1
Idle Peer Address
Router# show controllers t1 call-counters
T1 1/3/0:3:
Cisco IOS Voice, Video, and Fax Command Reference
VR-167
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
DS0's Active: 1
DS0's Active High
TimeSlot
Type
1
pri
2
pri
3
pri
4
pri
5
pri
6
pri
7
pri
8
pri
9
pri
10
pri
11
pri
12
pri
13
pri
14
pri
15
pri
16
pri
17
pri
18
pri
19
pri
20
pri
21
pri
22
pri
23
pri
T1 1/3/0:8:
DS0's Active: 0
DS0's Active High
TimeSlot
Type
1
pri
2
pri
3
pri
4
pri
5
pri
6
pri
7
pri
8
pri
9
pri
10
pri
11
pri
12
pri
13
pri
14
pri
15
pri
16
pri
17
pri
18
pri
19
pri
20
pri
21
pri
22
pri
23
pri
Water Mark: 2
TotalCalls
TotalDuration
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
1
00:11:04
1
00:10:20
Water Mark: 1
TotalCalls
TotalDuration
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:00:00
0
00:02:50
0
00:00:00
System's DS0's Active High Water Mark: 3
Note
Whenever a call goes down, HWM values are untouched. Only the DS0 Active changes. Now there
is only one call on 1/3/0:3. Observe the HWM for individual controllers. Total number of active calls
is 1.
Cisco IOS Voice, Video, and Fax Command Reference
VR-168
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
Clear the system-hwm with the following command:
Router# clear controllers call-counters system-hwm
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 1
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:12:16
23
pri
1
00:10:20
T1 1/3/0:8:
DS0's Active: 0
DS0's Active High Water Mark: 1
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
0
00:02:50
23
pri
0
00:00:00
System's DS0's Active High Water Mark: 1
Cisco IOS Voice, Video, and Fax Command Reference
VR-169
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
Note
Only the system HWM is reset to Active calls; for example, 1. For controllers 1/3/0:3 and
1/3/0:8, the HWMs are untouched.
Bring down another call, and display what the output will be:
Router# clear line 1/5/15
[confirm]
[OK]
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 0
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:12:40
23
pri
1
00:10:20
T1 1/3/0:8:
DS0's Active: 0
DS0's Active High Water Mark: 1
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
VR-170
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
22
23
pri
pri
0
0
00:02:50
00:00:00
System's DS0's Active High Water Mark: 1
Now use the all option, clearing at the system level:
Router# clear controllers call-counters all
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 0
DS0's Active High Water Mark: 0
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
0
00:00:00
23
pri
0
00:00:00
T1 1/3/0:8:
DS0's Active: 0
DS0's Active High Water Mark: 0
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
0
00:00:00
23
pri
0
00:00:00
System's DS0's Active High Water Mark: 0
Cisco IOS Voice, Video, and Fax Command Reference
VR-171
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
Clearing at the system level using the clear controllers call-counters command clears all DS0
controllers in the system and also clears the system’s HWM.
Now bring four calls up and show the call-counters output:
Router# show users
Line
User
* 0 con 0
tty 1/5/16
Router
tty 1/5/17
Router
tty 1/5/18
Router
tty 1/5/19
Router
Interface
User
Host(s)
idle
Async interface
Async interface
Async interface
Async interface
Idle
Location
00:00:00
00:01:01
PPP: 55.1.1.1
00:00:47
PPP: 55.2.1.1
00:00:28
PPP: 55.3.1.1
00:00:14
PPP: 55.4.1.1
Mode
Router# show controllers t1 call-counters
T1 1/3/0:3:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
18
pri
0
00:00:00
19
pri
0
00:00:00
20
pri
0
00:00:00
21
pri
0
00:00:00
22
pri
1
00:00:57
23
pri
1
00:01:30
T1 1/3/0:8:
DS0's Active: 2
DS0's Active High Water Mark: 2
TimeSlot
Type
TotalCalls
TotalDuration
1
pri
0
00:00:00
2
pri
0
00:00:00
3
pri
0
00:00:00
4
pri
0
00:00:00
5
pri
0
00:00:00
6
pri
0
00:00:00
7
pri
0
00:00:00
8
pri
0
00:00:00
9
pri
0
00:00:00
10
pri
0
00:00:00
11
pri
0
00:00:00
12
pri
0
00:00:00
13
pri
0
00:00:00
14
pri
0
00:00:00
15
pri
0
00:00:00
16
pri
0
00:00:00
17
pri
0
00:00:00
Cisco IOS Voice, Video, and Fax Command Reference
VR-172
Idle Peer Address
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear controllers call-counters
18
19
20
21
22
23
pri
pri
pri
pri
pri
pri
0
0
0
0
1
1
00:00:00
00:00:00
00:00:00
00:00:00
00:01:12
00:01:45
System's DS0's Active High Water Mark: 4
Related Commands
Command
Description
controller
Sets the router in controller configuration mode from global
configuration mode.
Cisco IOS Voice, Video, and Fax Command Reference
VR-173
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear csm-statistics modem
clear csm-statistics modem
To clear the call switching module (CSM) statistics for a modem or group of modems, use the clear
csm-statistics modem command in privileged EXEC mode.
clear csm-statistics modem [slot/port | modem-group-number]
Syntax Description
slot/port
(Optional) Identifies the location (and thereby the identity) of a specific
modem.
modem-group-number
(Optional) Designates a defined modem group.
Defaults
There are no defaults.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3 NA
This command was introduced.
Usage Guidelines
Use the clear csm-statistics modem command to clear CSM statistics for a particular modem or group
of modems. If the slot/port argument is specified, the CSM call statistics for calls using the identified
modem will be cleared. If a modem group number is specified, then the CSM call statistics for calls using
the modems associated with that group will be cleared. If no argument is specified, all CSM call statistics
for all modems will be cleared.
Examples
The following example clears CSM call statistics for calls coming in on modems associated with modem
group 2:
Router# clear csm-statistics modem 2
Related Commands
Command
Description
clear csm-statistics voice
Clears the CSM statistics for a particular or for all DSP channels.
Cisco IOS Voice, Video, and Fax Command Reference
VR-174
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear csm-statistics voice
clear csm-statistics voice
To clear the call switching module (CSM) statistics for a particular or for all digital signal processor
(DSP) channels, use the clear csm-statistics voice command in privileged EXEC mode.
clear csm-statistics voice [slot/dspm/dsp/dsp-channel]
Syntax Description
slot/dspm/dsp/dsp-channel
Defaults
There are no defaults.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3 NA
This command was introduced.
(Optional) Identifies the location of a particular DSP channel.
Usage Guidelines
Use the clear csm-statistics voice command to clear CSM statistics for a particular DSP channel. If the
slot/dspm/dsp/dsp-channel argument is specified, the CSM call statistics for calls using the identified
DSP channel will be cleared. If no argument is specified, all CSM call statistics for all DSP channels
will be cleared.
Examples
The following example clears CSM call statistics for calls coming in on all DSP channels:
Router# clear csm-statistics voice
Related Commands
Command
Description
clear csm-statistics modem
Clears the CSM statistics for a modem or group of modems.
Cisco IOS Voice, Video, and Fax Command Reference
VR-175
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear h323 gatekeeper call
clear h323 gatekeeper call
To force a disconnect of a specific call or of all calls active on a particular gatekeeper, use the clear h323
gatekeeper call command in privileged EXEC mode.
clear h323 gatekeeper call {all | local-callID local-callID}
Syntax Description
all
Forces all active calls currently associated with this gatekeeper to be
disconnected.
local-callID
Forces a single active call associated with this gatekeeper to be
disconnected.
local-callID
Specifies the local call identification number (CallID) that identifies
the call to be disconnected.
Defaults
There are no defaults.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(5)T
This command was introduced on the Cisco 2600 series routers, 3600
series, and on the MC3810 multiservice concentrator.
Usage Guidelines
If you want to force a particular call to be disconnected (as opposed to all active calls on the gatekeeper),
use the CallID number to identify that specific call. You can find the local CallID number for a specific
call by using the show gatekeeper calls command; the ID number is displayed in the LocalCallID
column. The “show gatekeeper calls Command Output” example shows output from the show
gatekeeper calls command.
Examples
The following example forces an active call on the gatekeeper to be disconnected. The local ID number
of the active call is 12-3339.
enable
clear h323 gatekeeper call local-callID 12-3339
The following example forces all active calls on the gatekeeper to be disconnected:
enable
clear h323 gatekeeper call all
Cisco IOS Voice, Video, and Fax Command Reference
VR-176
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear h323 gatekeeper call
Router# show gatekeeper calls
Total number of active calls =1
Gatekeeper Call Info
====================
LocalCallID
Age (secs)
BW
12-3339
94
768
Endpt(s): Alias
E.164Addr
CallSignalAddr
src EP: epA
10.0.0.11
dst EP: epB2zoneB.com
src PX: pxA
10.0.0.1
dst PX: pxB
172.21.139.90
Related Commands
(Kbps)
Port
RASSignalAddr
1720
10.0.0.11
Port
1700
1720
1720
24999
24999
10.0.0.11
172.21.139.90
Command
Description
show gatekeeper calls
Displays the status of each ongoing call of which a gatekeeper is aware.
Cisco IOS Voice, Video, and Fax Command Reference
VR-177
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear ip sctp statistics
clear ip sctp statistics
To clear statistics counts for SCTP, enter the clear ip sctp statistics command in privileged EXEC mode.
clear ip sctp statistics
Syntax Description
There are no arguments or keywords for this command.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.2(4)T
This command was introduced.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T and
support was added for the Cisco AS5300.
Usage Guidelines
This command clears both individual and overall statistics.
Examples
No output is generated from this command.
Router# clear ip sctp statistics
Related Commands
Command
Description
show ip sctp
association list
Shows a list of all current SCTP associations.
show ip sctp
association
parameters
Shows the parameters configured for the association defined by the
association ID.
show ip sctp
association statistics
Shows the current statistics for the association defined by the association
ID.
show ip sctp errors
Shows error counts logged by SCTP.
show ip sctp instances
Shows all currently defined SCTP instances.
show ip sctp statistics
Shows overall statistics counts for SCTP.
show iua as
Shows information about the current condition of an AS.
show iua asp
Shows information about the current condition of an ASP.
Cisco IOS Voice, Video, and Fax Command Reference
VR-178
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear mgcp statistics
clear mgcp statistics
To reset the Media Gateway Control Protocol (MGCP) statistical counters, use the clear mgcp statistics
command in privileged EXEC mode.
clear mgcp statistics
Syntax Description
This command has no arguments or keywords.
Defaults
There are no defaults.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access
server.
12.1(3)T
Support for this command was extended to the Cisco 3660,
Cisco uBR924, and Cisco 2600 series routers platforms.
Usage Guidelines
None
Examples
The following is an example of of how to enter the command.
Router# clear mgcp statistics
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
show mgcp statistics
Displays statistics for received and transmitted packets.
Cisco IOS Voice, Video, and Fax Command Reference
VR-179
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear rlm group
clear rlm group
To clear all time stamps to zero, use the clear rlm group link command in privileged EXEC mode.
clear rlm group group-number link
Syntax Description
group-number RLM group number (0 to 255).
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(7)
This command was introduced.
Examples
The following example clears the time stamps on RLM group 1:
Router# clear rlm group 1 link
Router#
02:48:17:
10.1.4.1]
02:48:17:
02:48:17:
02:48:17:
02:48:17:
02:48:17:
10.1.4.1]
02:48:17:
02:48:17:
02:48:17:
10.1.4.2]
02:48:17:
02:48:17:
02:48:17:
10.1.5.1]
02:48:17:
02:48:17:
02:48:17:
10.1.5.2]
02:48:17:
02:48:17:
02:48:17:
02:48:17:
02:48:17:
02:48:17:
rlm 1: [State_Up, rx ACTIVE_LINK_BROKEN] over link [10.1.1.1(Loopback1),
rlm
rlm
rlm
rlm
rlm
for
rlm
rlm
rlm
for
rlm
rlm
rlm
for
rlm
rlm
rlm
for
rlm
rlm
rlm
rlm
rlm
rlm
1: link [10.1.1.2(Loopback2), 10.1.4.2] requests activation
1: link [10.1.1.1(Loopback1), 10.1.4.1] is deactivated
1: [State_Recover, rx LINK_BROKEN] over link [10.1.1.2(Loopback2), 10.1.4.2]
1: link [10.1.1.1(Loopback1), 10.1.4.1] = socket[10.1.1.1, 10.1.4.1]
1: [State_Recover, rx USER_SOCKET_OPENED] over link [10.1.1.1(Loopback1),
user RLM_MGR
1: link [10.1.1.1(Loopback1), 10.1.4.1] is opened
1: link [10.1.1.2(Loopback2), 10.1.4.2] = socket[10.1.1.2, 10.1.4.2]
1: [State_Recover, rx USER_SOCKET_OPENED] over link [10.1.1.2(Loopback2),
user RLM_MGR
1: link [10.1.1.2(Loopback2), 10.1.4.2] is opened
1: link [10.1.1.1(Loopback1), 10.1.5.1] = socket[10.1.1.1, 10.1.5.1]
1: [State_Recover, rx USER_SOCKET_OPENED] over link [10.1.1.1(Loopback1),
user RLM_MGR
1: link [10.1.1.1(Loopback1), 10.1.5.1] is opened
1: link [10.1.1.2(Loopback2), 10.1.5.2] = socket[10.1.1.2, 10.1.5.2]
1: [State_Recover, rx USER_SOCKET_OPENED] over link [10.1.1.2(Loopback2),
user RLM_MGR
1: link [10.1.1.2(Loopback2), 10.1.5.2] is opened
1: [State_Recover, rx LINK_OPENED] over link [10.1.1.1(Loopback1), 10.1.4.1]
1: link [10.1.1.1(Loopback1), 10.1.4.1] requests activation
1: [State_Recover, rx LINK_OPENED] over link [10.1.1.2(Loopback2), 10.1.4.2]
1: [State_Recover, rx START_ACK] over link [10.1.1.1(Loopback1), 10.1.4.1]
1: link [10.1.1.1(Loopback1), 10.1.4.1] is activated
Cisco IOS Voice, Video, and Fax Command Reference
VR-180
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear rlm group
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
interface
Defines the IP addresses of the server, configures an interface type, and
enters interface configuration mode.
link (RLM)
Specifies the link preference.
protocol rlm port
Reconfigures the port number for the basic RLM connection for the whole
rlm-group.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the
link is declared down.
server (RLM)
Defines the IP addresses of the server.
show rlm group statistics Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
show rlm group timer
Displays the current RLM group timer values.
timer
Overwrites the default setting of timeout values.
Cisco IOS Voice, Video, and Fax Command Reference
VR-181
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear rudpv0 statistics
clear rudpv0 statistics
To clear the counters that track RUDP statistics, enter the clear rudpv0 statistics command in privileged
EXEC mode.
clear rudpv0 statistics
Syntax Description
This command has no arguments or keywords.
Defaults
The statistical information accumulates.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS
Release 12.1(1)T.
Examples
The following example shows how to clear RUDP statistics on a Cisco 2611 (Cisco SLT):
clear rudpv0 statistics
Related Commands
Command
Description
show rudpv0 failures
Displays RUDP information about failed connections and the
reasons for them. clear rudpv0 statistics resets the counters
for these statistics to 0.
show rudpv0 statistics
Displays RUDP information about number of packets sent,
received, and so forth. clear rudpv0 statistics resets the
counters for these statistics to 0.
Cisco IOS Voice, Video, and Fax Command Reference
VR-182
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear rudpv1 statistics
clear rudpv1 statistics
To clear the counters that track Reliable User Datagram Protocol (RUDP) statistics, use the clear
rudpv1 statistics command in privileged EXEC mode.
clear rudpv1 statistics
Syntax Description
This command has no arguments or keywords.
Defaults
The statistical information accumulates.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced.
Usage Guidelines
This command clears all statistics.
Examples
To clear all RUDP statistics for all available session-groups, see this example:
Router# clear rudpv1 statistics
Related Commands
Command
Description
debug rudpv1
Displays debugging information for RUDP.
show rudpv1
Displays RUDP information.
Cisco IOS Voice, Video, and Fax Command Reference
VR-183
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear sgcp statistics
clear sgcp statistics
To clear all Simple Gateway Control Protocol (SGCP) statistics, use the clear sgcp statistics command
in privileged EXEC mode.
clear sgcp statistics
Syntax Description
This command has no arguments or keywords.
Defaults
There are no defaults.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the
Cisco AS5300 universal access server only and was not generally
available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810 and
the Cisco 3600 series routers (except for the Cisco 3620) in a private
release that was not generally available.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T on
the Cisco MC3810 multiservice concentrator and Cisco 3600 series
routers and was made generally available.
Usage Guidelines
None
Examples
The following example shows all SGCP statistics being cleared:
Router# clear sgcp statistics
Related Commands
Command
Description
show sgcp statistics
Displays global statistics for SGCP packet counts.
Cisco IOS Voice, Video, and Fax Command Reference
VR-184
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear ss7 sm stats
clear ss7 sm stats
To clear the counters that track Session Manager statistics, use the clear ss7 sm stats command in
privileged EXEC mode.
clear ss7 sm stats
Syntax Description
This command has no arguments or keywords.
Defaults
The statistical information accumulates.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS
Release 12.1(1)T.
Examples
The following example shows how to clear Session Manager statistics on a Cisco 2611:
clear ss7 sm stats
Related Commands
Command
Description
show ss7 sm stats
Displays Session Manager information about number of
packets queued, received, and so forth. clear ss7 sm stats
resets the counters for these statistics to 0.
Cisco IOS Voice, Video, and Fax Command Reference
VR-185
Cisco IOS Voice, Video, and Fax Commands: A Through C
clear voice port
clear voice port
To clear voice port calls in progress, use the clear voice port command in privileged EXEC mode.
clear voice port [slot/port]
Syntax Description
slot/port
Defaults
There are no defaults.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
(Optional) The voice port slot number and port number. If you do not specify a
voice port, all calls on all voice ports are cleared.
If you do not specify a voice port, all calls on all voice ports are cleared. A confirmation prompt is
displayed.
There is no no form of this command.
Examples
The following example clears all calls on voice port 1/2 on the Cisco MC3810:
Router# clear voice port 1/2
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clock-select
clock-select
To establish the sources and priorities of the requisite clocking signals for the OC-3/STM-1 ATM Circuit
Emulation Service network module, use the clock-select command in CES configuration mode.
clock-select priority-number interface slot/port
Syntax Description
priority-number
Priority of the clock source. Values are 1 (high priority) to 4 (low priority).
interface
Specifies the interface to supply the clock source.
slot/port
Backplane slot number and port number on the interface.
Defaults
There are no defaults.
Command Modes
CES configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
This command is used on Cisco 3600 series routers that have OC-3/STM-1 ATM CES network modules.
To support synchronous or synchronous residual time stamp (SRTS) clocking modes, you must specify
a primary reference source to synchronize the flow of constant bit rate (CBR) data from its source to its
destination.
You can specify up to four clock priorities. The highest priority active interface in the router supplies
primary reference source to all other interfaces that require network clock synchronization services. The
fifth priority is the local oscillator on the network module.
Use the show ces clock-select command to display the currently configured clock priorities on the
router.
Examples
The following example defines two clock priorities on the router:
clock-select 1 cbr 2/0
clock-select 2 atm 2/0
Related Commands
Command
Description
channel-group
Configures the timing recovery clock for the CES interface.
clock source
Configures a transmit clock source for the CES interface.
show ces clock
Displays which ports are designated as network clock sources.
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codec (dial-peer)
codec (dial-peer)
To specify the voice coder rate of speech for a dial peer, use the codec command in dial peer
configuration mode. To reset the default value, use the no form of this command.
Cisco 2600 and 3600 Series Routers and Cisco 7200 and 7500 Series Routers
codec codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 |
g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 |
g729r8 | gsmefr | gsmfr} [bytes payload_size]
no codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 |
g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 |
g729r8 | gsmefr | gsmfr} [bytes payload_size]
Cisco AS5300 and AS5800 Universal Access Servers
codec codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24
| g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr}
[bytes payload_size]
no codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 |
g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr}
[bytes payload_size]
Cisco MC3810 Multiservice Concentrators
codec codec {clear-channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 |
g726r16 | g726r24 | g726r32 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr}
[bytes payload_size]
no codec {clear-channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 |
g726r16 | g726r24 | g726r32 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr}
[bytes payload_size]
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codec (dial-peer)
Syntax Description
For the Cisco 2600 and 3600 Series Routers and Cisco 7200 and 7500 Series Routers:
codec
Sets the codec options that you can use when you execute this command.
codec
Codec options are as follows:
•
clear-channel—Clear channel at 64,000 bits per second (bps).
•
g711alaw—G.711 a-Law at 64,000 bits per second.
•
g711ulaw—G.711 u-Law at 64,000 bps.
•
g723ar53—G.723.1 Annex A at 5300 bps.
•
g723ar63—G.723.1 Annex A at 6300 bps.
•
g723r53—G.723.1 at 5300 bps.
•
g723r63—G.723.1 at 6300 bps.
•
g726r16—G.726 at 16,000 bps.
•
g726r24—G.726 at 24,000 bps.
•
g726r32—G.726 at 32,000 bps.
•
g728—G.728 at 16,000 bps.
•
g729abr8—G.729 Annex A and B at 8000 bps.
•
g729ar8—G729 Annex A at 8000 bps.
•
g729br8—G.729 Annex B at 8000 bps.
•
g729r8—G.729 at 8000 bps. This is the default codec.
•
gsmefr—Global System for Mobile Communications Enhanced
Rate Codecs (GSMEFR) at 12,200 bps.
•
gsmfr—Global System for Mobile Communications Full Rate
(GSMFR) at 13200 bps.
bytes
(Optional) Specifies the number of bytes in the voice payload of each
frame.
payload_size
(Optional) The number of bytes in the voice payload of each frame. See
Table 12 for valid entries and default values.
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codec (dial-peer)
For the Cisco AS5300 and AS5800 Universal Access Servers:
codec
Sets the codec options that you can use when you execute this command.
codec
Codec options are as follows:
•
g711alaw—G.711 a-Law at 64,000 bits per second.
•
g711ulaw—G.711 u-Law at 64,000 bps.
•
g723ar53—G.723.1 Annex A at 5300 bps.
•
g723ar63—G.723.1 Annex A at 6300 bps.
•
g723r53—G.723.1 at 5300 bps.
•
g723r63—G.723.1 at 6300 bps.
•
g726r16—G.726 at 16,000 bps.
•
g726r24—G.726 at 24,000 bps.
•
g726r32—G.726 at 32,000 bps.
•
g728—G.728 at 16,000 bps.
•
g729abr8—G.729 Annex A and B at 8000 bps.
•
g729ar8—G729 Annex A at 8000 bps.
•
g729br8—G.729 Annex B at 8000 bps.
•
g729r8—G.729 at 8000 bps. This is the default codec.
•
gsmefr—Global System for Mobile Communications Enhanced
Rate Codecs (GSMEFR) at 12,200 bps.
•
gsmfr—Global System for Mobile Communications Full Rate
(GSMFR) at 13200 bps.
bytes
(Optional) Specifies the number of bytes in the voice payload of each
frame.
payload_size
(Optional) The number of bytes in the voice payload of each frame. See
Table 12 for valid entries and default values.
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codec (dial-peer)
For the Cisco MC3810 Multiservice Concentrators:
Defaults
codec
Sets the codec options that you can use when you execute this command.
codec
Codec options are as follows:
•
clear-channel—Clear channel at 64,000 bits per second (bps).
•
g711alaw—G.711 a-Law at 64,000 bits per second.
•
g711ulaw—G.711 u-Law at 64,000 bps.
•
g723ar53—G.723.1 Annex A at 5300 bps.
•
g723ar63—G.723.1 Annex A at 6300 bps.
•
g723r53—G.723.1 at 5300 bps.
•
g723r63—G.723.1 at 6300 bps.
•
g726r16—G.726 at 16,000 bps.
•
g726r24—G.726 at 24,000 bps.
•
g726r32—G.726 at 32,000 bps.
•
g728—G.728 at 16,000 bps.
•
g729abr8—G.729 Annex A and B at 8000 bps.
•
g729ar8—G729 Annex A at 8000 bps.
•
g729br8—G.729 Annex B at 8000 bps.
•
g729r8—G.729 at 8000 bps. This is the default codec.
•
gsmefr—Global System for Mobile Communications Enhanced Rate Codecs
(GSMEFR) at 12,200 bps.
•
gsmfr—Global System for Mobile Communications Full Rate (GSMFR) at
13,200 bps.
bytes
(Optional) Specifies the number of bytes in the voice payload of each frame.
payload_size
(Optional) The number of bytes in the voice payload of each frame. See Table 12 for
valid entries and default values.
g729r8, 30-byte payload for VoFR and VoATM
g729r8, 20-byte payload for VoIP
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This VoIP dial peer configuration command was introduced on the
Cisco 3600 series routers.
11.3(3)T
Support for Cisco 2600 series routers was added.
12.0(3)T
Support for the Cisco AS5300 universal access server was added. The
clear-channel keyword is not supported.
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codec (dial-peer)
Release
Modification
12.0(4)T
Support was added for the Cisco 3600 and 7200 series routers and for the
MC3810 multiservice concentrator. The command was modified for Voice
over Frame Relay (VoFR) dial peers.
12.0(5)XE
Additional codec choices and other options were added.
12.0(5)XK
The g729br8 and pre-ietf codec choices were added for the Cisco 2600 and
3600 series routers.
12.0(7)T
Support for the Cisco AS5800 universal gateway was added. Additional
voice coder rates of speech were added. There is no support for the
clear-channel keyword.
12.0(7)XK
The g729abr8 and g729ar8 codec choices were added for the Cisco
MC3810 multiservice concentrator, and the keyword pre-ietf was deleted.
12.1(1)T
The codec keyword additions and the pre-ietf deletion in Cisco IOS Release
12.0(7)XK were integrated in Cisco IOS Release 12.1(1)T.
12.1(5)
The gsmefr and gsmfr codec keywords were added.
Usage Guidelines
Note
VoFR and VoATM do not support the gsmefr and gsmfr codecs.
Use this command to define a specific voice coder rate of speech and payload size for a VoIP dial peer
or for a VoFR dial peer. This command is also used for VoATM.
A specific codec type can be configured on the dial peer as long as it is supported by the setting used
with the codec complexity voice-card configuration command. The codec complexity command is
voice-card specific and platform specific. The codec complexity voice-card configuration command is
set to either high or medium.
If the codec complexity command is set to high, the following options are available: g711alaw,
g711ulaw, g723ar53, g723ar63, g723r53, g723r63, g726r16, g726r24, g726r32, g728, g729r8, or
g729br8.
If the codec complexity command is set to medium, the following options are available: g711alaw,
g711ulaw, g726r16, g726r24, g726r32, g729r8, or g729br8.
The codec dial peer configuration command is particularly useful when you must change to a
small-bandwidth codec. Large-bandwidth codecs, such as G.711, do not fit in a small-bandwidth link.
However, the g711alaw and g711ulaw codecs provide higher quality voice transmission than other
codecs. The g729r8 codec provides near-toll quality with considerable bandwidth savings.
If codec values for the dial peers of a connection do not match, the call fails.
You can change the payload of each VoIP frame by using the bytes setting; you can change the payload
of each VoFR frame by using the bytes keyword with the payload_size setting. However, increasing the
payload size can add processing delay for each voice packet.
Table 12 describes the voice payload options and default values for the codecs and packet voice
protocols.
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codec (dial-peer)
Table 12
Voice Payload-per-Frame Options and Defaults
Codec
Protocol
Voice Payload Options (in bytes)
Default Voice
Payload (in bytes)
g711alaw
g711ulaw
VoIP
VoFR
VoATM
80, 160
40 to 240 in multiples of 40
40 to 240 in multiples of 40
160
240
240
g723ar53
g723r53
VoIP
VoFR
VoATM
20 to 220 in multiples of 20
20 to 240 in multiples of 20
20 to 240 in multiples of 20
20
20
20
g723ar63
g723r63
VoIP
VoFR
VoATM
24 to 216 in multiples of 24
24 to 240 in multiples of 24
24 to 240 in multiples of 24
24
24
24
g726r16
VoIP
VoFR
VoATM
20 to 220 in multiples of 20
10 to 240 in multiples of 10
10 to 240 in multiples of 10
40
60
60
g726r24
VoIP
VoFR
VoATM
30 to 210 in multiples of 30
15 to 240 in multiples of 15
30 to 240 in multiples of 15
60
90
90
g726r32
VoIP
VoFR
VoATM
40 to 200 in multiples of 40
20 to 240 in multiples of 20
40 to 240 in multiples of 20
80
120
120
g728
VoIP
VoFR
VoATM
10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10
40
60
60
g729abr8
g729ar8
g729br8
g729r8
VoIP
VoFR
VoATM
10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10
20
30
30
For toll quality, use the g711alaw or g711ulaw keyword. These values provide high-quality voice
transmission but use a significant amount of bandwidth. For nearly toll quality (and a significant savings
in bandwidth), use the g729r8 keyword.
On the Cisco MC3810 multiservice concentrator, this command was first supported as a voice port
command. On the Cisco MC3810 multiservice concentrator, you can also assign codec values to the
voice port. If configuring calls to a Cisco MC3810 multiservice concentrator running software versions
prior to 12.0(4)T, configure the codec command on the voice port. If configuring Cisco trunk permanent
calls, configure the codec command on the dial peer. If you configure the codec command on the dial
peer for VoFR permanent calls on the Cisco MC3810, the dial peer codec command setting overrides the
codec setting configured on the voice port.
Note
For regular switched calls on the Cisco MC3810 multiservice concentrator, the codec value must be
configured on the voice port, and the voice payload size is not configurable.
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codec (dial-peer)
Note
Examples
The clear-channel keyword is not supported on Cisco AS5300 and AS5800 universal gateways.
The following example shows how to configure a voice coder rate that provides toll quality voice with a
payload of 120 bytes per voice frame on a Cisco 2600 series routers or 3600 series router acting as a
terminating node. The example configuration begins in global configuration mode and is for VoFR dial
peer 200.
dial-peer voice 200 vofr
codec g711alaw bytes 120
The following example configures a voice coder rate for VoIP dial peer 10 that provides toll quality but
uses a relatively high amount of bandwidth:
dial-peer voice 10 voip
codec g711alaw
Related Commands
Command
Description
codec complexity
Specifies call density and codec complexity based on the codec used.
codec (DSP interface
dsp farm)
Specifies call density and codec complexity.
codec (voice port)
Specifies voice compression on the Cisco MC3810 multiservice
concentrator voice port.
show dial peer voice
Displays the codec setting for dial peers.
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codec (dsp)
codec (dsp)
To specify call density and codec complexity based on a particular codec standard, use the codec
command in DSP interface dsp farm mode. To reset the card type to the default, use the no form of the
command.
codec {high | med}
no codec {high | med}
Syntax Description
high
Specifies high complexity: two channels of any mix of codec.
med
Specifies medium complexity: four channels of g711/g726/g729a/fax.
Defaults
Medium complexity
Command Modes
DSP interface dsp farm
Command History
Release
Modification
12.0(5)XE
This command was introduced on the Cisco 7200 series routers.
12.1(1)T
This command was integrated into Cisco Release 12.1(1)T.
12.1(3)T.
Support was added for the Cisco 7500 series routers.
Usage Guidelines
This command is supported on only the Cisco 7200 and 7500 series routers.
Codec complexity refers to the amount of processing required to perform compression. Codec
complexity affects the number of calls, referred to as call density, that can take place on the DSPfarm
interfaces. The greater the codec complexity, the fewer the calls that are handled. For example, G.711
requires less DSP processing than G.728, so as long as the bandwidth is available, more calls can be
handled simultaneously by using the G.711 standard than by using G.728.
The DSPinterface dspfarm codec complexity setting affects the options available for the codec dial peer
configuration command.
To change codec complexity, you must first remove any configured channel associated signaling (CAS)
or DS0 groups and then reinstate them after the change.
Note
Examples
On the Cisco 2600 series routers, 3600 series, and MC3810 multiservice concentrator, codec
complexity is configured using the codec complexity command in voice-card configuration mode.
The following example configures the DSPfarm interface 1/0 on the Cisco 7200 series routers to support
high compression:
dspint DSPFarm 1/0
codec high
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codec (dsp)
Related Commands
Command
Description
codec (dial peer)
Specifies the voice codec rate of speech for a dial peer.
codec complexity
Specifies call density and codec complexity based on the codec
standard you are using.
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codec (voice-port)
codec (voice-port)
The codec command in voice-port configuration mode on the Cisco MC3810 multiservice concentrator
that was first supported in Cisco IOS Release 11.3(1)MA is no longer supported, beginning with
Cisco IOS Release 12.2. Configure the codec value using the codec dial peer configuration command.
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codec aal2-profile
codec aal2-profile
To set the codec profile for a digital signal processor (DSP) on a per-call basis, enter the codec
aal2-profile command in dial peer configuration mode. To restore the default value, use the no form of
the command.
codec aal2-profile {itut | atmf | custom} profile-number codec
no codec aal2-profile
Syntax Description
itut
Specifies the profile-number as an ITU-T type.
atmf
Specifies the profile-number as an ATMF type.
custom
Specifies the profile-number as a custom type.
profile-number
The available profile-number selections depend on the profile type.
For ITU-T:
•
1 = G.711ulaw
•
2 = G.711ulaw with silence insertion descriptor (SID)
•
7 = G.711ulaw and G.729ar8
For ATMF: None. ATMF is not supported.
For custom:
codec
•
100 = G.711ulaw and G.726r32
•
110 = G.711ulaw, G.726r32, and G.729ar8
Enter one codec for the DSP. The possible codec entries depend on the
profile-number. The valid entries are as follows:
•
For ITU 1—g711ulaw
•
For ITU 2—g711ulaw
•
For ITU 7—g711ulaw or g729ar8
•
For custom 100—g711ulaw or g726r32
•
For custom 110—g711ulaw or g726r32 or g729ar8
Defaults
The default is ITU profile 1 (G.711ulaw).
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1(1)XA
The command was introduced for the Cisco MC3810 multiservice concentrator.
12.1(2)T
This command was integrated into the 12.1(2)T release.
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codec aal2-profile
Usage Guidelines
Use this command to configure the DSP to operate with a specified profile type and codec.
You must enter the session protocol aal2-trunk command before configuring the codec AAL2 profile.
This command is used instead of the codec command for ATM Adaptation Layer 2 (AAL2) trunk
applications.
Examples
The following example sets the codec AAL2 profile type to ITU and configures a profile number of 7,
enabling codec G.729ar8:
dial-peer voice 100 voatm
session protocol aal2-trunk
codec aal2-profile itu 7 g729ar8
The following example sets the codec AAL2 profile type to custom and configures a profile number of
100, enabling codec G.726r32:
dial-peer voice 200 voatm
session protocol aal2-trunk
codec aal2-profile custom 100 g726r32
Related Commands
Command
Description
session protocol (dial peer)
Establishes a session protocol for calls between the local and remote
routers via the packet network.
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codec complexity
codec complexity
To specify call density and codec complexity based on the codec standard you are using, use the codec
complexity command in voice-card configuration mode. To reset the voice card to the default, use the
no form of this command.
codec complexity {high | medium}
no codec complexity
Syntax Description
high
medium
Each digital signal processor (DSP) supports two voice channels encoded in any of the
following formats:
•
g711alaw—G.711 A-law 64,000 bps.
•
g711ulaw—G.711 U-law 64,000 bps.
•
g723ar53—G.723.1 Annex A 5300 bps.
•
g723ar63—G.723.1 Annex A 6300 bps.
•
g723r53—G.723.1 5300 bps.
•
g723r63—G.723.1 6300 bps.
•
g723r16—G.726 16,000 bps.
•
g726r24—G726 24,000 bps.
•
g726r32—G.726 32,000 bps.
•
g728—G.728 16,000 bps.
•
g729r8—G.729 8000 bps. (default)
•
g729br8—G.729 Annex B 8000 bps.
•
fax relay—2400 bps, 4800 bps, 7200 bps, 9600 bps, 12 kbps, and 14.4 kbps.
Each DSP supports four voice channels encoded in any of the following formats:
•
g711alaw—G.711 a-Law 64,000 bps.
•
g711ulaw—G.711 u-Law 64,000 bps.
•
g726r16—G.726 16,000 bps.
•
g726r24—G.726 24,000 bps.
•
g726r32—G.726 32,000 bps.
•
g729r8—G.729 Annex A 8000 bps.
•
G729br8—G.729 Annex B with Annex A 8000 bps.
•
fax relay—2400 bps, 4800 bps, 7200 bps, 9600 bps, 12 kbps, and 14.4 kbps.
Defaults
medium complexity
Command Modes
Voice-card configuration
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codec complexity
Command History
Usage Guidelines
Release
Modification
12.0(5)XK
This command was introduced for the Cisco 2600 and Cisco 3600 series
routers.
12.0(7)T
This command was integrated into the 12.0(7)T release.
12.0(7)XK
This command was first supported on the Cisco MC3810 multiservice
concentrator series for use with the high-performance compression
module (HCM).
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Codec complexity refers to the amount of processing required to perform voice compression. Codec
complexity affects the call density—the number of calls that can take place on the DSPs. With higher
codec complexity, fewer calls can be handled. Select a higher codec complexity if that is required to
support a particular codec or combination of codecs. Select a lower codec complexity to support the
greatest number of voice channels, provided that the lower complexity is compatible with the particular
codecs in use.
To change codec complexity, all the DSP voice channels must be in the idle state.
Note
Examples
In the Cisco MC3810 multiservice concentrator, this command is valid only with one or more
HCMs installed, and you must specify voice card 0 in the command mode. If two HCMs are
installed, the codec complexity command configures both HCMs at once.
The following example sets the codec complexity to high on a Cisco MC3810 multiservice concentrator
containing one or two HCMs:
voice-card 0
codec complexity high
The following example sets the codec complexity to high on voice card 1 in a Cisco 2600 or 3600 series
router:
voice-card 1
codec complexity high
Related Commands
Command
Description
ds0-group
Defines T1/E1 channels for compressed voice calls and the CAS
method by which the router connects to the PBX or PSTN.
show voice dsp
Shows the current status of all DSP voice channels.
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codec preference
codec preference
To specify a list of preferred codecs to use on a dial peer, use the codec preference command in
voice-class configuration mode. To disable this functionality, use the no form of this command.
codec preference value codec_type [bytes payload-size]
no codec preference value codec_type
Syntax Description
value
Specifies the order of preference, with 1 being the most preferred and 14
being the least preferred.
codec_type
Specifies the codec preferred.
•
clear-channel—Clear Channel 64,000 bps
•
g711alaw—G.711 A Law 64,000 bps
•
g711ulaw—G.711 u Law 64,000 bps
•
g723ar53—G.723.1 ANNEX-A 5,300 bps
•
g723ar63—G.723.1 ANNEX-A 6,300 bps
•
g723r53—G.723.1 5,300 bps
•
g723r63—G.723.1 6,300 bps
•
g726r16—G.726 16,000 bps
•
g726r24—G.726 24,000 bps
•
g726r32—G.726 32,000 bps
•
g728—G.728 16,000 bps
•
g729abr8—G.729 ANNEX-A & B 8,000 bps
•
g729br8—G.729 ANNEX-B 8,000 bps
•
g729r8—G.729 8000 bps
•
gsmefr—Global System for Mobile Communications Enhanced Full
Rate (GSMEFR) 12,200 bps
•
gsmfr—Global System for Mobile Communications (GSM) Full Rate
(GSMFR) 13,200 bps
bytes
(Optional) Specifies that the size of the voice frame is in bytes.
payload-size
(Optional) Number of bytes you specify as the voice payload of each frame.
Values depend on the codec type and the packet voice protocol.
Defaults
No default behavior or values.
Command Modes
Voice-class configuration
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codec preference
Command History
Usage Guidelines
Release
Modification
12.0(2)XH
This command was introduced on the Cisco AS5300 universal access
server.
12.0(7)T
This command was first supported on the Cisco 2600 and 3600 series
routers.
12.0(7)XK
This command was first supported on the Cisco MC3810 multiservice
concentrator.
12.1(2)T
Support for the Cisco MC3810 multiservice concentrator was integrated
into Cisco Release IOS 12.1(2)T.
12.1(5)T
The codecs gsmefr and gsmfr were added.
The routers at opposite ends of the WAN may have to negotiate the codec selection for the network dial
peers. The codec preference command specifies the order of preference for selecting a negotiated codec
for the connection. Table 13 describes the voice payload options and default values for the codecs and
packet voice protocols.
Table 13
Voice Payload-per-Frame Options and Defaults
Codec
Protocol
Voice Payload Options (in bytes)
Default Voice
Payload (in bytes)
g711alaw
g711ulaw
VoIP
VoFR
VoATM
80, 160
40 to 240 in multiples of 40
40 to 240 in multiples of 40
160
240
240
g723ar53
g723r53
VoIP
VoFR
VoATM
20 to 220 in multiples of 20
20 to 240 in multiples of 20
20 to 240 in multiples of 20
20
20
20
g723ar63
g723r63
VoIP
VoFR
VoATM
24 to 216 in multiples of 24
24 to 240 in multiples of 24
24 to 240 in multiples of 24
24
24
24
g726r16
VoIP
VoFR
VoATM
20 to 220 in multiples of 20
10 to 240 in multiples of 10
10 to 240 in multiples of 10
40
60
60
g726r24
VoIP
VoFR
VoATM
30 to 210 in multiples of 30
15 to 240 in multiples of 15
30 to 240 in multiples of 15
60
90
90
g726r32
VoIP
VoFR
VoATM
40 to 200 in multiples of 40
20 to 240 in multiples of 20
40 to 240 in multiples of 20
80
120
120
g728
VoIP
VoFR
VoATM
10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10
40
60
60
g729abr8
g729ar8
g729br8
g729r8
VoIP
VoFR
VoATM
10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10
20
30
30
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codec preference
Examples
The following example creates codec preference list 99 and applies it to dial peer 1919:
voice class codec 99
codec preference 1 g711alaw
codec preference 2 g711ulaw bytes 80
codec preference 3 g723ar53
codec preference 4 g723ar63 bytes 144
codec preference 5 g723r53
codec preference 6 g723r63 bytes 120
codec preference 7 g726r16
codec preference 8 g726r24
codec preference 9 g726r32 bytes 80
codec preference 10 g729br8
codec preference 11 g729r8 bytes 50
codec preference 12 gsmefr
end
dial-peer voice 1919 voip
voice-class codec 99
Related Commands
Command
Description
voice class codec
Enters voice-class configuration mode and assigns an
identification tag number to a codec voice class.
voice-class codec (dial peer)
Assigns a previously configured codec selection preference list to
a dial peer.
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comfort-noise
comfort-noise
To generate background noise to fill silent gaps during calls if voice activity detection (VAD) is
activated, use the comfort-noise command in voice-port configuration mode. To provide silence when
the remote party is not speaking and VAD is enabled at the remote end of the connection, use the no form
of this command.
comfort-noise
no comfort-noise
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
Use the comfort-noise command to generate background noise to fill silent gaps during calls if VAD is
activated. If the comfort-noise command is not enabled, and VAD is enabled at the remote end of the
connection, the user will hear dead silence when the remote party is not speaking.
The configuration of the comfort-noise command only affects the silence generated at the local
interface; it does not affect the use of VAD on either end of the connection or the silence generated at
the remote end of the connection.
Note
Examples
On the Cisco MC3810 multiservice concentrator, this command cannot be disabled.
The following example enables background noise on the Cisco 3600 series routers:
voice-port 1/0/0
comfort-noise
Related Commands
Command
Description
vad (dial peer
configuration)
Enables VAD for the calls using a particular dial peer.
vad (voice-port
configuration)
Enables VAD for the calls using a particular voice port on the Cisco
MC3810 multiservice concentrator.
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compand-type
compand-type
To specify the companding standard used to convert between analog and digital signals in pulse code
modulation (PCM) systems, use the compand-type command in voice-port configuration mode. To
disable the compand type, use the no form of this command.
compand-type {u-law | a-law}
no compand-type {u-law | a-law}
Syntax Description
Defaults
u-law
Specifies the North American U-law ITU-T PCM encoding standard.
a-law
Specifies the European a-law ITU-T PCM encoding standard.
u-law (T1 digital)
a-law (E1 digital)
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was first introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
If you have a Cisco MC3810 multiservice concentrator or 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 configuration of the compand-type a-law command, however, if you request a list of commands,
the compand-type a-law command will display.
Note
Examples
On the Cisco 3600 series routers router, the U-law and a-law settings are configured using
the codec dial peer configuration command.
The following example configures a-law encoding on voice port 1/1 on the Cisco MC3810 multiservice
concentrator:
voice-port 1/1
compand-type a-law
Related Commands
Command
Description
codec (voice-port
configuration)
Configures voice compression on the Cisco MC3810 multiservice
concentrator voice port.
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condition
condition
To manipulate the signaling format bit-pattern for all voice signaling types, use the condition command
in voice-port configuration mode. To turn off conditioning on the voice port, use the no form of this
command.
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}
no 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}
Syntax Description
tx-a-bit
Sends A bit.
tx-b-bit
Sends B bit.
tx-c-bit
Sends C bit.
tx-d-bit
Sends D bit.
rx-a-bit
Receives A bit.
rx-b-bit
Receives B bit.
rx-c-bit
Receives C bit.
rx-d-bit
Receives D bit.
on
Forces the bit state to be 1.
off
Forces the bit state to be 0.
invert
Inverts the bit state.
Defaults
The signaling format is not manipulated (for all sent or received A, B, C, and D bits).
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series
routers.
12.1(2)T
The modifications in Cisco IOS Release 12.0(7)XK were integrated into
Cisco IOS Release 12.1(2)T.
Usage Guidelines
Use the condition command to manipulate the sent or received bit patterns to match expected patterns
on a connected device. Be careful not to destroy the information content of the bit pattern. For example,
forcing the A-bit on or off will prevent Foreign Exchange Office (FXO) interfaces from being able to
generate both an on-hook and off-hook state.
The condition command is applicable to digital voice ports only.
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condition
Examples
The following example manipulates the signaling format bit pattern on digital voice port 0:5 on a
Cisco MC3810 multiservice concentrator:
voice-port 0:5
condition tx-a-bit invert
condition rx-a-bit invert
The following example manipulates the signaling format bit pattern on voice port 1/0:0 on a Cisco 2600
or Cisco 3600 router:
voice-port 1/0:0
condition tx-a-bit invert
condition rx-a-bit invert
Related Commands
Command
Description
define
Defines the transmit and receive bits for North American E&M and E&M
MELCAS voice signaling.
ignore
Configures the North American E&M or E&M MELCAS voice port to
ignore specific receive bits.
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connect (atm)
connect (atm)
To define connections between T1 or E1 controller ports and the ATM interface, enter the connect
command in global configuration mode. Use the no form of this command to restore the default values.
connect id atm slot/port-1 {name of PVC/SVC | vpi/vci} {T1 | E1} slot/port-2 TDM-group-number
no connect id atm slot/port-1 {name of PVC/SVC | vpi/vci} {T1 | E1} slot/port-2
TDM-group-number
Syntax Description
id
A name for this connection.
atm
Specifies the ATM interface.
slot/port-1
The location of the ATM controller to be connected.
name of PVC/SVC
Specifies the permanent or switched virtual circuit.
vpi/vci
Specifies a virtual path identifier (VPI) and virtual channel identifier (VCI).
T1
Specifies a T1 port.
E1
Specifies an E1 port.
slot/port-2
The location of the T1 or E1 controller to be connected.
TDM-group-number
The number identifier of the time-division multiplexing (TDM) group
associated with the T1 or E1 controller port and created by using the
tdm-group command. Valid values are from 0 to 23 for T1 and from 0 to 30
for E1.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(2)T
This command was introduced for ATM interfaces on the Cisco 2600 and
3600 series routers.
Usage Guidelines
This command is used on Cisco 2600 and 3600 series routers to provide connections between T1/E1 and
ATM interfaces. This command is used after all interfaces are configured.
Once TDM groups are created on two different physical ports, you can use the connect command to start
the passage of data between the ports. If a crosspoint switch is provided in the advanced integration
module (AIM) slot, the connections can extend between ports on different cards. Otherwise, the
connection will be restricted to ports on the same VWIC card.
The voice/WAN interface card (VWIC) can make a connection only if the number of time slots at the
source and destination are the same. For the connection to be error free, the two ports must be driven by
the same clock source; otherwise, slips will occur.
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connect (atm)
Examples
The following example shows how the ATM permanent virtual circuit (PVC) and T1 TDM group are set
up and then connected:
interface atm 1/0
pvc pvc1 0/100 ces
exit
controller T1 1/1
tdm-group 3 timeslots 13-24 type e&m
exit
connect tdm1 atm 1/0 pvc1 0/100 T1 1/1 3
Related Commands
Command
Description
tdm-group
Creates TDM groups that can be connected.
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connect (drop-and-insert)
connect (drop-and-insert)
To define connections among T1 or E1 controller ports for drop-and-insert (also called TDM
cross-connect), use the connect command in global configuration mode. To restore default values, use
the no form of this command.
connect id {t1 | e1} slot/port-1 tdm-group-no-1 {t1 | e1} slot/port-2 tdm-group-no-2
no connect id {t1 | e1} slot/port-1 tdm-group-no-1 {t1 | e1} slot/port-2 tdm-group-no-2
Syntax Description
id
A name for this connection.
t1
Specifies a T1 port.
e1
Specifies an E1 port.
slot/port-1
The location of the first T1 or E1 controller to be connected. Valid values
for slot and port are 0 and 1.
tdm-group-no-1
The number identifier of the time-division multiplexing (TDM) group
associated with the first T1 or E1 controller port and created by using the
tdm-group command. Valid values are from 0 to 23 for T1 and from 0 to
30 for E1.
slot/port-2
The location of the second T1 or E1 controller port to be connected. Valid
values for slot are from 0 to 5, depending on the platform. Valid values
for port are 0 to 3, depending on the platform and the presence of a
network module.
tdm-group-no-2
The number identifier of the time-division multiplexing (TDM) group
associated with the second T1 or E1 controller and created by using the
tdm-group command. Valid values are from 0 to 23 for T1 and from 0 to
30 for E1.
Defaults
There is no drop-and-insert connection between the ports.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)XK
The command was introduced on the Cisco 2600 and 3600 series.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
12.1(1)T
The command was modified to accommodate two channel groups on a port
for 1- and 2-port T1/E1 multiflex voice/WAN interface cards (VWICs) on
the Cisco 3600 series routers.
Usage Guidelines
The connect command creates a named connect between two TDM groups associated with
drop-and-insert ports on T1 or E1 interfaces where you have already defined the groups by using the
tdm-group command.
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connect (drop-and-insert)
Once TDM groups are created on two different physical ports, you can use the connect command to start
the passage of data between the ports. If a crosspoint switch is provided in the AIM slot, the connections
can extend between ports on different cards. Otherwise, the connection will be restricted to ports on the
same VWIC card.
The VWIC card can make a connection only if the number of time slots at the source and destination are
the same. For the connection to be error-free, the two ports must be driven by the same clock source;
otherwise, “slips” will occur.
Examples
The following example shows a fractional T1 terminated on port 0 using time slots 1 through 8; a
fractional T1 is terminated on port 1 using time slots 2 through 12; and time slots 13 through 20 from
port 0 are connected to time slots 14 through 21 on port 1 by using the connect command:
controller t1 0/0
channel-group 1 timeslots 1-8
tdm-group 1 timeslots 13-20
exit
controller t1 0/1
channel-group 1 timeslots 2-12
tdm-group 2 timeslot 14-21
exit
connect exampleconnection t1 0/0 1 t1 0/1 2
Related Commands
Command
Description
show connect
Displays configuration information about drop-and-insert connections that
have been configured on a router.
tdm-group
Configures a list of time slots for creating clear channel groups
(pass-through) for TDM cross-connect.
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connect (global)
connect (global)
This command, created for the Cisco MC3810-IGX Interworking feature in Cisco IOS Release 12.0(2)T,
is not supported in Cisco IOS Release 12.2.
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connect voice
connect voice
This command, created for the Cisco MC3810-IGX Interworking feature in Cisco IOS Release 12.0(2)T,
is not supported in Cisco IOS Release 12.2.
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connection
connection
To specify a connection mode for a voice port, use the connection command in voice-port configuration
mode. To disable the selected connection mode, use the no form of this command.
connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}
no connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}
Syntax Description
plar
Specifies a private line automatic ringdown (PLAR) connection. PLAR is an
autodialing mechanism that permanently associates a voice interface with a far-end
voice interface, allowing call completion to a specific telephone number or PBX
without dialing. When the calling telephone goes off-hook, a predefined network
dial peer is automatically matched, which sets up a call to the destination telephone
or PBX.
tie-line
Specifies a connection that emulates a temporary tie-line trunk to a private branch
exchange (PBX). A tie-line connection is automatically set up for each call and torn
down when the call ends.
plar-opx
Specifies a PLAR off-premises extension (OPX) connection. Using this option, the
local voice port provides a local response before the remote voice port receives an
answer. On Foreign Exchange Office (FXO) interfaces, the voice port will not
answer until the remote side has answered.
digits
Specifies the destination telephone number. Valid entries are any series of digits that
specify the E.164 telephone number.
trunk
Specifies a connection that emulates a permanent trunk connection to a PBX. A
trunk connection remains permanent in the absence of any active calls.
answer-mode
(Optional) Specifies that the router should not attempt to initiate a trunk connection
but should wait for an incoming call before establishing the trunk. Used only with
the trunk keyword.
Defaults
No connection mode is specified.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3(1)MA1
This command was first supported on the Cisco MC3810 multiservice concentrator,
and the tie-line keyword was first made available on the Cisco MC3810 multiservice
concentrator.
11.3(1)MA5
The plar-opx keyword was first made available on the Cisco MC3810 multiservice
concentrator as the plar-opx-ringrelay keyword. The keyword was shortened in a
subsequent release.
12.0(2)T
This command was integrated into Cisco IOS Release 12.0(2)T.
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connection
Usage Guidelines
Release
Modification
12.0(3)XG
The trunk keyword was made available on the Cisco MC3810 multiservice
concentrator. The trunk answer-mode option was added.
12.0(4)T
The trunk and trunk answer-mode additions were integrated in Cisco IOS Release
12.0(4)T.
12.0(7)XK
This command was unified across the Cisco 2600, 3600, and MC3810 multiservice
concentrator platforms.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Use the connection command to specify a connection mode for a specific interface. For example, use
the connection plar command to specify a PLAR interface. The string you configure for this command
is used as the called number for all incoming calls over this connection. The destination peer is
determined by the called number.
Use the connection trunk command to specify a permanent tie-line connection to a PBX. Voice over IP
simulates a trunk connection by creating virtual trunk tie lines between PBXs connected to Cisco devices
on each side of a VoIP connection (see Figure 4). In this example, two PBXs are connected using a
virtual trunk. PBX-A is connected to Router A via an E&M voice port; PBX-B is connected to Router
B via an E&M voice port. The Cisco routers spoof the connected PBXs into believing that a permanent
trunk tie line exists between them.
Figure 4
Virtual Trunk Connection
1(308)555-1000
PBX-A
172.19.10.10
172.20.10.10
Router B
Router A
IP cloud
PBX-B
E&M
23958
E&M
1(510)555-4000
Virtual trunk connection
In configuring virtual trunk connections in Voice over IP, the following restrictions apply:
•
You can use the following voice port combinations:
– Ear and mouth (E&M) to E&M (same type)
– Foreign Exchange Station (FXS) to Foreign Exchange Office (FXO)
– FXS to FXS (with no signaling)
•
Do not perform number expansion on the destination pattern telephone numbers configured for
trunk connection.
•
Configure both end routers for trunk connections.
•
The connected Cisco routers must be Cisco 2600 or Cisco 3600 series routers.
Note
Because virtual trunk connections do not support number expansion, the destination patterns
on each side of the trunk connection must match exactly.
To configure one of the devices in the trunk connection to act as slave and only receive calls, use the
answer-mode option with the connection trunk command when configuring that device.
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connection
Note
When using the connection trunk command, you must enter the shutdown command followed by
the no shutdown command on the voice port.
VoIP establishes the trunk connection immediately after it is configured. Both ports on either end of the
connection are dedicated until you disable trunking for that connection. If for some reason the link
between the two switching systems goes down, the virtual trunk reestablishes itself after the link comes
back up.
Use the connection tie-line command when the dial plan requires that additional digits be added in front
of any digits dialed by the PBX and that the combined set of digits be used to route the call onto the
network. The operation is similar to the connection plar command operation, but in this case the tie-line
port waits to collect digits from the PBX. The tie-line digits are automatically stripped by a terminating
port.
If the connection command is not configured, the standard session application outputs a dial tone when
the interface goes off-hook until enough digits are collected to match a dial peer and complete the call.
Examples
The following example shows PLAR selected as the connection mode on a Cisco 3600 series routers
router, with a destination telephone number of 555-9262:
voice-port 1/0/0
connection trunk 5559262
The following example shows the tie-line selected as the connection mode on a Cisco MC3810, with a
destination telephone number of 555-9262:
voice-port 1/1
connection tie-line 5559262
The following example specifies a PLAR off-premises extension connection on a Cisco 3600 series
routers router, with a destination telephone number of 555-9262:
voice-port 1/0/0
connection plar-opx 5559262
The following example shows configuration of a Cisco 3600 series routers router for a trunk connection
and specifies that it will establish the trunk only when it receives an incoming call:
voice-port 1/0/0
connection trunk 5559262 answer-mode
The following examples show configuration of the routers on both sides of a VoIP connection (as
illustrated in Figure 4) to support trunk connections.
Router A
voice-port 1/0/0
connection trunk +15105554000
dial-peer voice 10 pots
destination-pattern +13085551001
port 1/0/0
dial-peer voice 100 voip
session-target ipv4:172.20.10.10
destination-pattern +15105554000
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connection
Router B
voice-port 1/0/0
connection trunk +13085551000
dial-peer voice 20 pots
destination-pattern +15105554001
port 1/0/0
dial-peer voice 200 voip
session-target ipv4:172.19.10.10
destination-pattern +13085551000
Related Commands
Command
Description
destination-pattern
Specifies the prefix or the full E.164 telephone number for a dial peer.
dial peer voice
Enters dial peer configuration mode and specifies the voice encapsulation type.
session-protocol
Establishes a session protocol for calls between the local and remote routers
via the packet network.
session-target
Configures a network-specific address for a dial peer.
voice-port
Enters voice-port configuration mode.
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connection-timeout
connection-timeout
To configure the time in seconds for which a connection is maintained after completion of a
communication exchange, enter the connection-timeout command in settlement configuration mode. To
return to the default value of this command, use the no form of this command to reset.
connection-timeout number
no connection-timeout number
Syntax Description
number
Defaults
The default connection timeout is 3600 seconds (1 hour).
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 and Cisco 3600
series and on the Cisco AS5300 universal access server.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Time (in seconds) for which a connection is maintained after the
communication exchange is completed. Values can range from 0 to
86,400 seconds; 0 means that the connection does not timeout.
Usage Guidelines
The router maintains the connection for the configured period in anticipation of future communication
exchanges to the same server.
Examples
The following example shows a connection configured to be maintained for 3600 seconds after
completion of a communications exchange:
settlement 0
connection-timeout 3600
Related Commands
Command
Description
customer-id
Sets the customer identification.
device-id
Sets the device identification.
encryption
Specifies the encryption method.
max-connection
Sets the maximum simultaneous connections.
response-timeout
Sets the response timeout.
retry-delay
Sets the retry delay.
retry-limit
Sets the connection retry limit.
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connection-timeout
Command
Description
session-timeout
Sets the session timeout.
settlement
Enters the settlement configuration mode.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Brings up/shuts down the settlement provider.
type
Specifies the provider type.
url
Specifies the Internet service provider address.
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Cisco IOS Voice, Video, and Fax Commands: A Through C
copy flash vfc
copy flash vfc
To copy a new version of VCWare from the Cisco AS5300 universal access server motherboard to voice
feature card (VFC) Flash memory, use the copy flash vfc command in privileged EXEC mode.
copy flash vfc slot-number
Syntax Description
slot-number
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3 NA
This command was introduced on the Cisco AS5300 universal access
server.
Usage Guidelines
Slot on the Cisco AS5300 universal access server in which the VFC is installed.
Valid entries are from 0 to 2.
Use the copy flash vfc command to use the standard copy user interface to copy a new version of
VCWare from the Cisco AS5300 universal access server motherboard to VFC Flash memory. The VFC
is a plug-in feature card for the Cisco AS5300 universal access server and has its own Flash memory
storage for embedded firmware. For more information about VFCs, refer to Installing Voice over IP
Feature Cards in Cisco AS5300 Universal Gateways.
Once the VCWare file has been copied, use the unbundle vfc command to uncompress and install
VCWare.
Examples
The following example copies a new version of VCWare from the Cisco AS5300 universal access server
motherboard to VFC Flash memory:
Router# copy flash vfc 0
Related Commands
Command
Description
copy tftp vfc
Copies a new version of VCWare from a TFTP server to VFC Flash memory.
unbundle vfc
Unbundles the current running image of VCWare or DSPWare into separate
files.
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copy tftp vfc
copy tftp vfc
To copy a new version of VCWare from a TFTP server to voice feature card (VFC) Flash memory, use
the copy tftp vfc command in privileged EXEC mode.
copy tftp vfc slot-number
Syntax Description
slot-number
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3 NA
This command was introduced on the Cisco AS5300 universal access
server.
Usage Guidelines
Slot on the Cisco AS5300 universal access server in which the VFC is installed.
Valid entries are from 0 to 2.
Use the copy tftp vfc command to copy a new version of VCWare from a TFTP server to VFC Flash
memory. The VFC is a plug-in feature card for the Cisco AS5300 universal access server and has its own
Flash storage for embedded firmware. For more information about VFCs, refer to Installing Voice Over
IP Feature Cards in Cisco AS5300 Universal Gateways.
Once the VCWare file has been copied, use the unbundle vfc command to uncompress and install
VCWare.
Examples
The following example copies a file from the TFTP server to VFC Flash memory:
Router# copy tftp vfc 0
Related Commands
Command
Description
copy flash vfc
Copies a new version of VCWare from the Cisco AS5300 universal access
server motherboard to VFC Flash memory.
unbundle vfc
Unbundles the current running image of VCWare or DSPWare into separate
files.
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cptone
cptone
To specify a regional analog voice interface-related tone, ring, and cadence setting, use the cptone
command in voice-port configuration mode. To disable the selected tone, use the no form of this
command.
cptone {locale}
no cptone {locale}
Syntax Description
locale
Specifies an analog voice interface-related default tone, ring, and cadence
setting for a specified country (for ISDN PRI and E1 R2 signaling). Keywords
for the argument locale are contained in Table 14. Keywords for ISDN PRI
signaling are contained in
The Cisco 2600 and 3600 series routers and the Cisco MC3810 multiservice
concentrator comply with the ISO 3166 country name standards, which use a
two-letter code to represent a country.
Defaults
The northamerica keyword is for the Cisco MC3810 multiservice concentrator for versions prior to
Cisco IOS Release 12.0(4)T and for ISDN PRI.
The us keyword is for the Cisco MC3810 multiservice concentrator for Cisco IOS Release 12.0(4)T and
higher and for E1 R2 signaling.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3(1)MA
The full keyword names for the countries were first supported on the
Cisco MC3810 multiservice concentrator.
12.0(4)T
Support was added for the ISO 3166 two-letter country codes on the
Cisco MC3810 multiservice concentrator.
12.1(5)XM
The keywords for Eqypt, Ghana, Jordan, Kenya, Lebanon, Nepal, Nigeria,
Pakistan, Panama, Saudi Arabia, and Zimbabwe were added.
Usage Guidelines
Use the cptone command to specify a regional analog voice interface-related default tone, ring, and
cadence setting for a specified voice port.
This command affects only the tones generated at the local interface. It does not affect any information
passed to the remote end of a connection or any tones generated at the remote end of a connection.
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cptone
If your device is configured to support E1 R2 signaling, the E1 R2 signaling type (whether ITU, ITU
variant, or local variant as defined by the cas-custom command) must match the appropriate pulse code
modulation (PCM) encoding type as defined by the cptone command. For countries for which a cptone
value has not yet been defined, you can try the following:
•
If the country uses a-Law E1 R2 signaling, use the gb value for the cptone command.
•
If the country uses U-law E1 R2 signaling, use the us value for the cptone command.
Table 14 lists valid entries for the locale argument.
Table 14
cptone locale Argument Values
Argument Value
Country
ar
Argentina
au
Australia
at
Austria
be
Belgium
br
Brazil
ca
Canada
cn
China
co
Colombia
cz
Czech Republic
dk
Denmark
eg
Egypt
fi
Finland
fr
France
de
Germany
gb
Great Britain
gh
Ghana
gr
Greece
hk
Hong Kong
hu
Hungary
is
Iceland
in
India
id
Indonesia
ie
Ireland
il
Israel
it
Italy
jo
Jordan
jp
Japan
ke
Kenya
kr
Korea Republic
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cptone
Table 14
cptone locale Argument Values (continued)
Argument Value
Country
lb
Lebanon
lu
Luxembourg
my
Malaysia
mx
Mexico
ng
Nigeria
nl
Netherlands
np
Nepal
nz
New Zealand
no
Norway
pa
Panama
pe
Peru
ph
Philippines
pk
Pakistan
pl
Poland
pt
Portugal
ru
Russian Federation
sa
Saudi Arabia
sg
Singapore
sk
Slovakia
si
Slovenia
za
South Africa
es
Spain
se
Sweden
ch
Switzerland
tw
Taiwan
th
Thailand
tr
Turkey
gb
Great Britain
us
United States
ve
Venezuela
zw
Zimbabwe
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cptone
Table 15 describes the argument values for ISDN PRI signaling.
Table 15
Examples
cptone locale Argument Values for ISDN PRI Signaling
Argument Value
Description
australia
Specifies an analog voice interface-related default tone, ring, and cadence
setting for Australia.
brazil
Specifies an analog voice interface-related default tone, ring, and cadence
setting for Brazil.
china
Specifies an analog voice interface-related default tone, ring, and cadence
setting for China.
finland
Specifies an analog voice interface-related default tone, ring, and cadence
setting for Finland.
france
Specifies an analog voice interface-related default tone, ring, and cadence
setting for France.
germany
Specifies an analog voice interface-related default tone, ring, and cadence
setting for Germany.
japan
Specifies an analog voice interface-related default tone, ring, and cadence
setting for Japan.
northamerica
Specifies an analog voice interface-related default tone, ring, and cadence
setting for North America.
sweden
Specifies an analog voice interface-related default tone, ring, and cadence
setting for Sweden.
unitedkingdom
Specifies an analog voice interface-related default tone, ring, and cadence
setting for the United Kingdom.
The following example configures United States as the call progress tone locale on the Cisco 3600 series
routers, beginning in global configuration mode:
voice-port 1/0/0
cptone us
The following example configures Singapore as the call progress tone locale on the Cisco MC3810
multiservice concentrator, beginning in global configuration mode:
voice-port 1/1
cptone sg
The following example configures Japan as the call progress tone locale:
voice-port 0:D
cptone japan
The following example configures Brazil as the call progress tone locale on the Cisco AS5300 universal
access server:
voice-port 1:0
cptone BR
description Brasil Tone
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cptone
Related Commands
Command
Description
voice-port
Opens voice-port configuration mode.
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cross-connect
cross-connect
To cross-connect two groups of digital signal level 0s (DS0s) from two controllers on the Cisco MC3810
multiservice concentrator or to cross-connect the Universal I/O (UIO) serial port for pass-through traffic
to a trunk controller, use the cross-connect command in global configuration mode. To remove the
cross-connect function for the given controller, use the no form of this command.
Pass-Through Traffic Between Two Controllers
cross-connect id controller-1 tdm-group-no-1 controller-2 tdm-group-no-2
no cross-connect id controller-1 tdm-group-no-1 controller-2 tdm-group-no-2
Pass-Through Traffic from a UIO Serial Port to a Trunk Controller
cross-connect id interface-serial controller tdm-group-no
no cross-connect id interface-serial controller tdm-group-no
Note
Syntax Description
The UIO serial port is either serial port 0 or 1.
Pass-Through Traffic Between Two Controllers
id
Unique identification (ID) assigned to this cross-connection. The valid range is
from 0 to 31.
controller-1
Type of the first controller (T1 0, T1 1, or E1)
tdm-group-no-1
Time-division multiplexing (TDM) group number assigned to the first
controller.
controller-2
Type of the second controller (T1, E1 0, or E1 1).
tdm-group-no-2
TDM group number assigned to the second controller.
Pass-Through Traffic from a UIO Serial Port to a Trunk Controller
id
Unique ID assigned to this cross connection.
interface-serial
Number of the serial port, either 0 or 1.
controller
Type of the controller. Enter one of the following: T1 0, T1 1, E1 0, or E1 1.
tdm-group-no
TDM group number assigned to the controller.
Defaults
No default behavior or values.
Command Modes
Global configuration
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cross-connect
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
This command applies to Voice over Frame Relay and Voice over ATM on the Cisco MC3810.
Examples
The following example configures a pass-through cross-connect from serial port 0 to controller T1 1 on
TDM group 20:
cross-connect 10 serial0 T1 1 20
Related Commands
Command
Description
supervisory
disconnect
Configures a list of time slots for creating clear channel groups
(pass-through) for TDM cross-connect.
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customer-id
customer-id
To identify a carrier or Internet service provider (ISP) with a settlement provider, enter the customer-id
command in settlement configuration mode. To reset the default value of this command, use the no form
of this command.
customer-id number
no customer-id number
Syntax Description
number
Defaults
The default customer ID is 0.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 and 3600 series and
on the AS5300.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Customer ID number as provided by the settlement server. The value
range is from 0 to 2,147,483,647.
Usage Guidelines
It is optional to identify a carrier or ISP with a settlement provider.
Examples
The following example identifies a carrier or service provider with the ID number of 1000:
settlement 0
custom id 1000
Related Commands
Command
Description
device-id
Sets the device identification.
encryption
Specifies the encryption method.
max-connection
Sets the maximum number of simultaneous connections.
response-timeout
Sets the response timeout.
retry-delay
Sets the retry delay.
retry-limit
Sets the connection retry limit.
session-timeout
Sets the session timeout.
settlement
Enters the settlement configuration mode.
show settlement
Displays the configuration for all settlement server transactions.
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customer-id
Command
Description
shutdown
Shuts down the settlement provider.
type
Specifies the provider type.
url
Specifies the Internet service provider address.
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customer-id
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Cisco IOS Voice, Video, and Fax Commands:
D Through F
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax
applications. The commands are presented in alphabetical order beginning with the letter D. Some
commands required for configuring voice, video, and fax may be found in other Cisco IOS command
references. Use the command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS
Voice, Video, and Fax Configuration Guide.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
default
default
To reset the value of a command to its default, use the default command in SIP user-agent configuration
mode.
default {inband-alerting | max-forwards | retry {invite | response | bye | cancel} | sip-server |
timers {trying | connect | disconnect | expires} | transport}
Syntax Description
inband-alerting
Resets inband-alerting to its default of generating the header
“Require: com.cisco.inband-alerting” in outgoing INVITE
messages. Tones are fed from the terminating gateway.
max-forwards
Resets max-forwards to its default of 6.
retry {invite | response | bye |
cancel}
Resets the specified retry to its default (6 for invite and
response; 10 for bye and cancel).
sip-server
Resets the sip-server to a null value.
timers {trying | connect |
disconnect | expires}
Resets the specified timer to its default (500 for trying, connect,
and disconnect; 180,000 for expires).
transport
Resets transport to the default of both User Datagram Protocol
(UDP) and TCP enabled.
Defaults
No default behavior or values.
Command Modes
SIP user-agent configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 and 3600
series routers and on the CiscoAS5300 universal access server.
Examples
The following example sets inband-alerting to its default.
sip-ua
default inband-alerting
Related Commands
Command
Description
cap-list vfc
Adds a voice codec overlay file to the capability file list.
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default-file vfc
default-file vfc
To specify an additional (or different) file from the ones in the default file list and stored in voice feature
card (VFC) Flash memory, use the default-file vfc command in global configuration mode. To delete the
file from the default file list, use the no form of this command.
default-file filename vfc slot
no default-file filename vfc slot
Syntax Description
filename
Indicates the file to be retrieved from VFC Flash memory and used (as the
default file) to boot up the system.
slot
Indicates the slot on the Cisco AS5300 universal access server in which the
VFC is installed. Valid entries are from 0 to 2.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)NA
This command was introduced on the Cisco AS5300 universal access
server.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
When VCWare is unbundled, it automatically adds DSPWare to Flash memory, creates both the
capability and default file lists, and populates these lists with the default files for that version of VCWare.
The default file list includes the files that will be used to boot up the system.
Use the default-file vfc command to add a specified file to the default file list, replacing the existing
default for that extension type.
Examples
The following example specifies that the bas-vfc-1.0.14.0.bin file, which is stored in VFC Flash memory,
be added to the default file list:
default-file bas-vfc-1.0.14.0.bin vfc 0
Related Commands
Command
Description
cap-list vfc
Adds a voice codec overlay file to the capability file list.
delete vfc
Deletes a file from VFC Flash memory.
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define
define
To define the transmit and receive bits for North American ear and mouth (E&M) and E&M Mercury
Exchange Limited Channel-Associated Signaling (MELCAS) voice signaling, use the define command
in voice-port configuration mode. To restore the default value, use the no form of this command.
define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000
| 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
no define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 |
1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
Syntax Description
Defaults
tx-bits
The bit pattern applies to the transmit signaling bits.
rx-bits
The bit pattern applies to the receive signaling bits.
seize
The bit pattern defines the seized state.
idle
The bit pattern defines the idle state.
0000 through 1111
Specifies the bit pattern.
The default is to use the preset signaling patterns as defined in American National Standards Institute
(ANSI) and European Conference of Postal and Telecommunications Administrations (CEPT)
standards, as follows:
•
For North American E&M:
– tx-bits idle 0000 (0001 if on E1 trunk)
– tx-bits seize 1111
– rx-bits idle 0000
– rx-bits seize 1111
•
For E&M MELCAS:
– tx-bits idle 1101
– tx-bits seize 0101
– rx-bits idle 1101
– rx-bits seize 0101
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA3
This command was introduced on the Cisco MC3810 multiservice
concentrator.
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define
Usage Guidelines
Release
Modification
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series
routers.
12.1(2)T
The support for Cisco 2600 and 3600 series routers was integrated into
Cisco IOS Release 12.1(2)T.
The define command applies to E&M digital voice ports associated with T1/E1 controllers.
Use the define command to match the E&M bit patterns with the attached telephony device. Be careful
not to define invalid configurations, such as all 0000 on E1, or identical seized and idle states. Use this
command with the ignore command.
Examples
To configure a voice port on a Cisco 2600 or 3600 series router that is sending traffic in North American
E&M signaling format to convert the signaling to MELCAS format, enter the following commands:
voice-port 1/0/0
define rx-bits idle 1101
define rx-bits seize 0101
define tx-bits idle 1101
define tx-bits seize 0101
To configure a voice port on a Cisco MC3810 multiservice concentrator that is sending traffic in North
American E&M signaling format to convert the signaling to MELCAS format, enter the following
commands:
voice-port 0/8
define rx-bits
define rx-bits
define tx-bits
define tx-bits
Related Commands
idle 1101
seize 0101
idle 1101
seize 0101
Command
Description
condition
Manipulates the signaling bit-pattern for all voice signaling types.
ignore
Configures a North American E&M or E&M MELCAS voice port to ignore
specific receive bits.
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delete vfc
delete vfc
To delete a file from voice feature card (VFC) Flash memory, use the delete vfc command in privileged
EXEC mode.
delete filename vfc slot
Syntax Description
filename
Specifies the file in VFC Flash memory to be deleted.
slot
Specifies the slot on the Cisco AS5300 universal access server
in which the specified VFC resides. Valid entries are from 0 to
2.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(1)NA
This command was introduced on the Cisco AS5300 universal
access server.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
Note
Examples
Use the delete vfc command to delete a specific file from VFC Flash memory and to remove the file from
the default list or capability list if the specified file is included in those lists.
Deleting a file from VFC Flash memory does not free the VFC Flash memory space that the file
occupied. To free VFC Flash memory space, use the erase vfc command.
The following example deletes the bas-vfc-1.0.14.0.bin file, which is stored in VFC Flash memory of
the VFC located in slot 0:
Router# delete bas-vfc-1.0.14.0.bin vfc 0
Related Commands
Command
Description
default-file vfc
Specifies an additional (or different) file from the ones in the
default file list and stored in VFC Flash memory.
erase vfc
Erases the Flash memory of a specified VFC.
show vfc directory
Displays the list of all files that reside on this VFC.
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description
description
To include a specific description about the digital signal processor (DSP) interface, use the description
command in voice-port configuration mode. To disable this feature, use the no form of this command.
description string
no description
Syntax Description
string
Defaults
Enabled with a null string.
Command Modes
Voice-port configuration
Command History
Release
Character string from 1 to 80 characters.
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers router.
11.3(1)MA
This command was implemented on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
Use the description command to include descriptive text about the DSP interface connection. This
information is displayed when a show command is issued, and it does not affect the operation of the
interface in any way.
Examples
The following example identifies voice port 1/0/0 on the Cisco 3600 series routers as being connected
to the purchasing department:
voice-port 1/0/0
description purchasing_dept
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description (dspfarm)
description (dspfarm)
To include a specific description about the digital signal processor (DSP) interface, use the description
command in DSPfarm interface configuration mode. To disable this feature, use the no form of this
command.
description string
no description string
Syntax Description
string
Defaults
Enabled with a null string.
Command Modes
DSPfarm interface configuration
Command History
Release
Character string from 1 to 80 characters.
Modification
11.3(1)T
This command was introduced for the Cisco 7200 series routers.
12.0(5)XE
The command was modified to reduce the maximum number of allowable
characters in a text string from 255 to 80.
12.1(1)T
The 12.0(5)XE modifications were integrated into Cisco IOS
Release 12.1(1)T.
Usage Guidelines
Use the description command to include descriptive text about this DSP interface connection. This
information is displayed when you issue a show command and does not affect the operation of the
interface in any way.
Examples
The following example identifies DSPfarm interface 1/0 on the Cisco 7200 series routers router as being
connected to the marketing department:
dspint dspfarm 1/0
description marketing_dept
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destination-pattern
destination-pattern
To specify either the prefix or the full E.164 telephone number (depending on your dial plan) to be used
for a dial peer, use the destination-pattern command in dial-peer configuration mode. To disable the
configured prefix or telephone number, use the no form of this command.
destination-pattern [+] string [T]
no destination-pattern [+] string [T]
Syntax Description
+
(Optional) Character indicating an E.164 standard number.
string
Series of digits that specify the E.164 or private dialing plan telephone number. Valid
entries are the digits 0 through 9, the letters A through D, and the following special
characters:
•
The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads.
•
Comma (,), which inserts a pause between digits.
•
Period (.), which matches any entered digit (this character is used as a wildcard).
•
Percent sign (%), which indicates that the previous digit/pattern occurred zero or
multiple times, similar to the wildcard usage in the regular expression.
•
Plus sign (+), which matches a sequence of one or more matches of the
character/pattern.
Note
T
Defaults
The plus sign used as part of the digit string is different from the plus sign
that can be used in front of the digit string to indicate that the string is an
E.164 standard number.
•
Circumflex (^), which indicates a match to the beginning of the string.
•
Dollar sign ($), which matches the null string at the end of the input string.
•
Backslash symbol (\), which is followed by a single character matching that
character or used with a single character with no other significance (matching that
character).
•
Question mark (?), which indicates that the previous digit occurred zero or one time.
•
Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed
in the brackets; only numeric characters from 0 to 9 are allowed in the range. This is
similar to a regular expression rule.
•
Parentheses “( )”, which indicate a pattern and is the same as the regular expression
rule.
(Optional) Control character indicating that the destination-pattern value is a variable
length dial string.
Enabled with a null string.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
destination-pattern
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3(1)MA
This command was implemented on the Cisco MC3810 multiservice
concentrator.
12.0(4)XJ
This command was modified for store and forward fax.
12.1(1)
The command as modified for store and forward fax was integrated into
Cisco IOS Release 12.1(1).
12.0(7)XR
Support for the plus sign, percent sign, question mark, brackets, and
parentheses symbols in the dial string was added to the Cisco AS5300
universal access server.
12.0(7)XK
Support for the plus sign, percent sign, question mark, brackets, and
parentheses in the dial string was added to the Cisco 2600, Cisco 3600, and
Cisco MC3810 multiservice concentrator.
12.1(1)T
The modifications made in Cisco IOS Release 12.0(7)XR for the
Cisco AS5300 were first supported in Cisco IOS Release 12.1(1)T and were
first supported on the T train for the following additional platforms:
Cisco 1750, Cisco 2600 series routers, Cisco 3600 series routers, Cisco
7200, and Cisco 7500.
12.1(2)T
The modifications made in Cisco IOS Release 12.0(7)XK for the
Cisco MC3810 multiservice concentrator were first supported on the T train.
Usage Guidelines
Use the destination-pattern command to define the E.164 telephone number for a dial peer.
This pattern is used to match dialed digits to a dial peer. The dial peer is then used to complete the call.
When a router receives voice data, it compares the called number (the full E.164 telephone number) in
the packet header with the number configured as the destination pattern for the voice-telephony peer.
The router then strips out the left-justified numbers corresponding to the destination pattern. If you have
configured a prefix, the prefix is appended to the front of the remaining numbers, creating a dial string,
which the router then dials. If all numbers in the destination pattern are stripped out, the user receives a
dial tone.
There are certain areas in the world (for example, in certain European countries) where valid telephone
numbers can vary in length. Use the optional control character T to indicate that a particular
destination-pattern value is a variable-length dial string. In this case, the system does not match the
dialed numbers until the interdigit timeout value has expired.
Note
Examples
Cisco IOS software does not check the validity of the E.164 telephone number; it accepts any series
of digits as a valid number.
The following example shows configuration of the E.164 telephone number 555-7922 for a dial peer:
dial-peer voice 10 pots
destination-pattern +5557922
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Cisco IOS Voice, Video, and Fax Commands: D Through F
destination-pattern
The following example shows configuration of a destination pattern in which the pattern “43” is repeated
multiple times preceding the digits “555”:
dial-peer voice 1 voip
destination-pattern 555(43)+
The following example shows configuration of a destination pattern in which the preceding digit/pattern
was repeated multiple times:
dial-peer voice 2 voip
destination-pattern 555%
The following example shows configuration of a destination pattern in which the digit numbers range
between 5553409 and 5559499:
dial-peer voice 3 vofr
destination-pattern 555[3-9]4[0=9]9
The following example shows configuration of a destination pattern in which the digit numbers range
between 5551439, 5553439, 5555439, 5557439, and 5559439:
dial-peer voice 4 voatm
destination-pattern 555[13579]439
Related Commands
Command
Description
answer-address
Specifies the full E.164 telephone number to be used to identify the dial peer
of an incoming call.
prefix
Specifies the prefix of the dialed digits for this dial peer.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
destination-pattern (interface)
destination-pattern (interface)
To specify the ISDN directory number for the telephone interface, use the destination-pattern
command in interface configuration mode. To disable the specified ISDN directory number, use the no
form of this command.
destination-pattern isdn
no destination-pattern
Syntax Description
isdn
Defaults
A default ISDN directory number is not defined for this interface.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
Local ISDN directory number assigned by your telephone
service provider.
This command is applicable to the Cisco 800 series routers.
You must specify this command when creating a dial peer. This command does not work if it is not
specified within the context of a dial peer. For information on creating a dial peer, refer to the
Cisco 800 Series Routers Software Configuration Guide.
Do not specify an area code with the local ISDN directory number.
Examples
The following example specifies 555-1111 as the local ISDN directory number:
destination-pattern 5551111
Related Commands
Command
Description
dial-peer voice
Enters dial-peer configuration mode, defines the type of dial
peer, and defines the tag number associated with a dial peer.
no call-waiting
Disables call waiting.
port (dial-peer)
Enables an interface on a PA-4R-DTR port adapter to operate
as a concentrator port.
ring
Sets up a distinctive ring for telephones, fax machines, or
modems connected to a Cisco 800 series router.
show dial-peer voice
Displays configuration information and call statistics for dial
peers.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
detect v54 channel-group
detect v54 channel-group
To enable V.54 loopback detection for the command sent from the remote device, use the detect v54
channel-group command in controller configuration mode. To disable the V.54 loopback detection, use
the no form of this command.
detect v54 channel-group channel-number
no detect v54 channel-group channel-number
Syntax Description
channel-number
Defaults
V.54 loopback detection is disabled.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 and 3600 series routers.
Channel number from 1 to 24 (T1) or from 1 to 31 (E1).
Usage Guidelines
Use the detect v54 channel-group controller configuration command to enable V.54 loopback
detection. The remote device will send a loopup inband payload command sequence in fractional T1
(FT1).
Examples
The following example sets the loopback detection for channel-group 1; then the loopback detection is
disabled for channel-group 1.
detect v54 channel-group 1
no detect v54 channel-group 1
Related Commands
Command
Description
loopback remote v54
channel-group
Activates a remote V.54 loopback for the channel group on the far end.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
device-id
device-id
To identify a gateway associated with a settlement provider, use the device-id command in settlement
configuration mode. To reset to the default value, use the no form of this command.
device-id number
no device-id number
Syntax Description
number
Defaults
The default device ID is 0.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 and 3600 series
routers and on the AS5300 access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Device ID number as provided by the settlement server. Values range is
from 0 to 2,147,483,647.
Usage Guidelines
It is optional to identify a gateway associated with a settlement provider.
Examples
The following example sets the device-id to 1000:
settlement 0
device-id 1000
Related Commands
Command
Description
customer-id
Identifies a carrier or Internet service provider with the settlement
provider.
settlement
Enters settlement configuration mode.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-control-mib
dial-control-mib
To specify attributes for the call history table, use the dial-control-mib command in global
configuration mode. To restore the default maximum size or retention time of the call history table, use
the no form of this command.
dial-control-mib {max-size number | retain-timer number}
no dial-control-mib {max-size number | retain-timer number}
Syntax Description
max-size number
Specifies the maximum size of the call history table. Valid entries are from 0
to 500 table entries. A value of 0 prevents any history from being retained.
retain-timer number
Specifies the length of time, in minutes, for entries in the call history table.
Valid entries are from 0 to 2,147,483,647 minutes. A value of 0 prevents any
history from being retained.
Defaults
The default call history table length is 50 table entries. The default retain timer is 15 minutes.
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.0(1)XA
This command was first applied to the CDR feature on the Cisco MC3810
multiservice concentrator.
12.0(2)T
The 12.0(1)XA application on the Cisco MC3810 multiservice concentrator
was integrated into Cisco IOS Release 12.0(2)T.
Examples
The following example configures the call history table to hold 400 entries, with each entry remaining
in the table for 10 minutes:
dial-control-mib max-size 400
dial-control-mib retain-timer 10
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer hunt
dial-peer hunt
To specify a hunt selection order for dial peers, use the dial-peer hunt command in dial-peer
configuration mode. To restore the default selection order, use the no form of this command.
dial-peer hunt hunt-order-number
no dial-peer hunt
Syntax Description
hunt-order-number
A number from 0 to 7 that selects a predefined hunting selection
order:
0—Longest match in phone number, explicit preference,
random selection. This is the default hunt order number.
1—Longest match in phone number, explicit preference, least
recent use.
2—Explicit preference, longest match in phone number,
random selection.
3—Explicit preference, longest match in phone number, least
recent use.
4—Least recent use, longest match in phone number, explicit
preference.
5—Least recent use, explicit preference, longest match in
phone number.
6—Random selection.
7—Least recent use.
Defaults
The default is the longest match in the phone number, explicit preference, random selection (hunt order
number 0).
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced and supported on the Cisco
2600, 3600, and 7200 series routers, the Cisco MC3810
multiservice concentrator, and the Cisco AS5300 universal
access servers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer hunt
Usage Guidelines
Use the dial-peer hunt dial-peer configuration command if you have configured hunt groups. “Longest
match in phone number” refers to the destination pattern that matches the greatest number of the dialed
digits. “Explicit preference” refers to the preference setting in the dial-peer configuration. “Least recent
use” refers to the destination pattern that has waited the longest since being selected. “Random
selection” weights all of the destination patterns equally in a random selection mode.
This command applies to POTS, Voice over IP (VoIP), Voice over Frame Relay (VoFR), Voice over ATM
(VoATM), and Multimedia Mail over Internet Protocol (MMOIP) dial peers.
Examples
The following example configures the dial peers to hunt in the following order: (1) longest match in
phone number, (2) explicit preference, (3) random selection.
dial-peer hunt 0
Related Commands
Command
Description
destination-pattern
Specifies the prefix or the complete telephone number for a dial
peer.
preference
Specifies the preferred selection order of a dial peer within a
hunt group.
show dial-peer voice
Displays configuration information for dial peers.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer terminator
dial-peer terminator
To change the character used as a terminator for variable-length dialed numbers, use the dial-peer
terminator command in global configuration mode. To restore the default terminating character, use the
no form of this command.
dial-peer terminator character
no dial-peer terminator
Syntax Description
character
Defaults
The default terminating character is #.
Command Modes
Global configuration
Command History
Release
Designates the terminating character for a variable-length dialed
number. Valid numbers and characters are #, *, 0, 1, 2, 3, 4, 5, 6, 7,
8, 9, a, b, c, and d. The default is #.
Modification
12.0
This command was introduced.
12.0(7)XK
Usage was restricted to variable-length dialed numbers. The
command was implemented on the Cisco 2600 and 3600 series
routers and on the MC3810 multiservice concentrator.
12.1(2)T
The modifications made in Cisco IOS Release 12.0(7)XK were
integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
There are certain areas in the world (for example, in certain European countries) where telephone
numbers can vary in length. When a dialed-number string has been identified as a variable length
dialed-number, the system does not place a call until the configured value for the timeouts interdigits
command has expired or until the caller dials the terminating character. Use the dial-peer terminator
global configuration command to change the terminating character.
Examples
The following example shows that “9” has been specified as the terminating character for variable-length
dialed numbers:
dial-peer terminator 9
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer terminator
Related Commands
Command
Description
answer-address
Specifies the preferred selection order of a dial peer within a hunt
group.
destination-pattern
Specifies the prefix or the complete telephone number for a dial peer.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
show dial-peer voice
Displays configuration information for dial peers.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer video
dial-peer video
To define a video ATM dial peer for a local or remote video codec, to specify video-related
encapsulation, and to enter dial-peer configuration mode use the dial-peer video command in global
configuration mode. To remove the video dial peer, use the no form of this command.
dial-peer video tag {videocodec | videoatm}
no dial-peer video tag {videocodec | videoatm}
Syntax Description
tag
Digits that define a particular dial peer. Defines the dial peer and assigns the
protocol type to the peer. Valid entries are from 1 to 10,000. The tag must be
unique on the router.
videocodec
Specifies a local video codec connected to the router.
videoatm
Specifies a remote video codec on the ATM network.
Defaults
No video dial peer is configured.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM interface configuration on the
Cisco MC3810 multiservice concentrator.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Usage Guidelines
The tag value must be unique to the device.
Examples
On a Cisco MC3810 multiservice concentrator, the following example sets up a local video dial peer
designated as 10:
dial-peer video 10 videocodec
Related Commands
Command
Description
show dial-peer video
Displays dial-peer configuration.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer voice
dial-peer voice
To enter dial-peer configuration mode (and to specify the method of voice encapsulation), use the
dial-peer voice command in global configuration mode. To disable a defined dial peer, use the no form
of this command.
Cisco 2600 Series Routers
dial-peer voice tag {pots | vofr | voip}
no dial-peer voice tag {pots | vofr | voip}
Cisco 3600 Series Routers
dial-peer voice tag {pots | voatm | vofr | voip}
no dial-peer voice tag
Cisco 7200 Series Routers
dial-peer voice tag {vofr}
no dial-peer voice tag {vofr}
Cisco 7204 VXR and Cisco 7206 VXR Routers
dial-peer voice tag {pots | voatm | vofr | voip}
no dial-peer voice tag {pots | voatm | vofr | voip}
Cisco AS5300 Universal Access Server
dial-peer voice tag {mmoip | pots | vofr | voip}
no dial-peer voice tag {mmoip | pots | vofr | voip}
Cisco MC3810 Multiservice Concentrator
dial-peer voice tag {pots | voatm | vofr | voip}
no dial-peer voice tag {pots | voatm | vofr | voip}
Syntax Description
tag
Digits that define a particular dial peer. Valid entries are from 1 to
2,147,483,647.
mmoip
Indicates that this is a multimedia mail peer using IP encapsulation on the
IP backbone.
Note
pots
On the Cisco AS5300 universal access server, MMoIP is available
only if you have modem ISDN channel aggregation (MICA)
technologies modems.
Indicates that this is a plain old telephone service (POTS) peer using
Voice over IP encapsulation on the IP backbone.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer voice
voatm
(Cisco 3600 series routers, Cisco MC3810 multiservice concentrators,
Cisco 7204 VXR routers, and Cisco 7206 VXR routers only) Specifies
that this is a Voice over ATM dial peer using the real-time AAL5 voice
encapsulation on the ATM backbone network.
vofr
Specifies that this is a Voice over Frame Relay dial peer using FRF.11
encapsulation on the Frame Relay backbone network.
voip
Indicates that this is a VoIP peer using voice encapsulation on the POTS
network.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Usage Guidelines
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3(1)MA
This command was first supported on the Cisco MC3810 multiservice
concentrator, with support for the pots, voatm, vofr, and vohdlc keywords.
12.0(3)T
This command was first supported on the AS5300, with support for the pots
and voip keywords.
12.0(3)XG
The vofr keyword was added for the Cisco 2600 series routers and
Cisco 3600 series platforms.
12.0(4)T
The vofr keyword was integrated into Cisco IOS Release 12.0(4)T. The vofr
keyword was added to the Cisco 7200 series routers platform.
12.0(4)XJ
The mmoip keyword was added for the Cisco AS5300 universal access
server platform. Also, the dial-peer voice command was implemented for
store and forward fax.
12.0(7)XK
The voip keyword was added for the Cisco MC3810 multiservice
concentrator, and the voatm keyword was added for the Cisco 3600 series
routers router. Support for vohdlc on the Cisco MC3810 multiservice
concentrator was removed in this release.
12.1(1)
The mmoip keyword addition in Cisco IOS Release 12.0(4)XJ was
integrated into Cisco IOS Release 12.1(1). The dial-peer voice
implementation for store and forward fax was also integrated into this
mainline release.
12.1(2)T
The keyword changes in Cisco IOS Release 12.0(7)XK were integrated into
Cisco IOS Release 12.1(2)T.
Use the dial-peer voice global configuration command to switch to dial-peer configuration mode from
global configuration mode and to define a particular dial peer. Use the exit command to exit dial-peer
configuration mode and return to global configuration mode.
After you have created a dial peer, that dial peer remains defined and active until you delete that
particular dial peer. To delete a dial peer, use the no form of this command. To disable a dial peer, use
the shutdown command in dial-peer configuration mode.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-peer voice
In store and forward fax on the Cisco AS5300 universal access server, the POTS dial peer defines the
inbound faxing line characteristics from the sending fax device to the receiving Cisco AS5300 universal
access server and the outbound line characteristics from the sending Cisco AS5300 universal access
server to the receiving fax device. The Multimedia Mail over Internet Protocol (MMoIP) dial peer
defines the inbound faxing line characteristics from the Cisco AS5300 universal access server to the
receiving Simple Mail Transfer Protocol (SMTP) mail server. This command applies to both on-ramp
and off-ramp store and forward fax functions.
Note
Examples
On the Cisco AS5300 universal access server, MMoIP is available only if you have modem ISDN
channel aggregation (MICA) technologies modems.
The following example shows how to access dial-peer configuration mode and configure a POTS peer
identified as dial peer 10 and an MMoIP dial peer identified as dial peer 20:
dial-peer voice 10 pots
dial-peer voice 20 mmoip
The following example deletes the MMoIP peer identified as dial peer 20:
no dial-peer voice 20 mmoip
Related Commands
Command
Description
codec (dial-peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay dial
peer.
destination-pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory
number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over Enables the generation of FRF.11 Annex A frames for a dial peer.
Frame Relay)
preference
Indicates the preferred order of a dial peer within a rotary hunt group.
sequence-numbers
Enables the generation of sequence numbers in each frame generated by the
DSP for Voice over Frame Relay applications.
session protocol
Establishes a session protocol for calls between the local and remote routers
via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination
gatekeeper.
voice-port
Enters voice-port configuration mode.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-type
dial-type
To specify the type of out-dialing for voice port interfaces, use the dial-type command in voice-port
configuration mode. To disable the selected type of dialing, use the no form of this command.
dial-type {dtmf | pulse | mf}
no dial-type
Syntax Description
dtmf
Dual tone multifrequency (DTMF) touch-tone dialing.
pulse
Pulse (rotary) dialing.
mf
Multifrequency tone dialing.
Defaults
dtmf
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3(1)MA3
The pulse keyword was added. This command was
implemented on the Cisco MC3810 multiservice concentrator.
12.0(7)XK
The mf keyword was added.
12.1(2)T
The pulse and mf keyword additions and the MC3810
multiservice concentrator platform implementation were
integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
Use the dial-type command to specify an out-dialing type for a Foreign Exchange Office (FXO) or ear
and mouth (E&M) voice port interface. This command specifies the tone type for digit detection and
out-pulsing. This command is not applicable to Foreign Exchange Station (FXS) voice ports because the
ports do not generate out-dialing. This command also specifies the detection direction. MF is not
supported for FXS and FXO.
Voice ports can always detect DTMF and pulse signals. This command does not affect voice port dialing
detection.
The dial-type command affects out-dialing as configured for the dial peer.
The dial-type command is not supported on FXO voice port interfaces on the Cisco MC3810
multiservice concentrator. If you are using the dial-type command with E&M WinkStart, use the dtmf
or mf option.
SGCP 1.1+ does not support pulse dialing.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
dial-type
Examples
The following example shows a voice port configured on the Cisco MC3810 multiservice concentrator
to support a rotary (pulse tone) dialer:
voice-port 1/1
dial-type pulse
The following example shows a voice port configured on the Cisco MC3810 multiservice concentrator
to support a DTMF (touch-tone) dialer:
voice-port 1/1
dial-type dtmf
The following example shows a voice port configured on the Cisco MC3810 multiservice concentrator
to support a multifrequency tone dialer:
voice-port 1/1
dial-type mf
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
digit-strip
digit-strip
To enable digit stripping on a plain old telephone service (POTS) dial-peer call leg, use the digit-strip
command in dial-peer configuration mode. To disable digit stripping on the dial-peer call leg, use the no
form of this command.
digit-strip
no digit-strip
Syntax Description
This command has no arguments or keywords.
Defaults
Digit stripping is enabled.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(7)XR1
This command was introduced for Voice over IP (VoIP) on the
Cisco AS5300.
12.0(7)XK
This command was first supported for the following voice
technologies on the following platforms:
•
VoIP (Cisco 2600 series routers, Cisco 3600 series routers,
Cisco MC3810 multiservice concentrator)
•
Voice over Frame Relay (VoFR)—Cisco 2600 series routers,
Cisco 3600 series routers, Cisco MC3810 multiservice
concentrator)
•
Voice over ATM (VoATM)—Cisco 3600 series routers and
Cisco MC3810 multiservice concentrator
12.1(1)T
The modifications made in Cisco IOS Release 12.0(7)XK were first
supported in Cisco IOS Release 12.1(1)T:
12.1(2)T
This command was first implemented in Cisco IOS Release 12.1(2)T
for the following voice technologies on the following platforms:
•
VoIP (Cisco MC3810 multiservice concentrator)
•
VoFR (Cisco 2600 series routers, Cisco 3600 series routers, and
Cisco MC3810 multiservice concentrator)
•
VoATM (Cisco 3600 series routers, Cisco MC3810 multiservice
concentrator)
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
digit-strip
Usage Guidelines
The digit-strip command is supported on POTS dial peers only.
When a called number is received and matched to a POTS dial peer, the matched digits are stripped and
the remaining digits are forwarded to the voice interface.
Table 16 lists a series of dial peers configured with a specific destination pattern and shows the longest
matched number after the digit is stripped based on the dial string 408 555-3048.
Table 16
Dial Peer Configurations with Longest Matched Number
Dial Peer
Destination Pattern
Preference
Session Target
Longest Matched
Number
1
4085553048
0 (highest)
100-voip
10
2
408[0-9]553048
0
200-voip
9
3
408555
0
300-voip
6
4
408555
1(lower)
400-voip
6
5
408%
1
500-voip
3
6
..........
0
600-voip
0
7
..........
1
1:D (interface)
0
Table 17 lists a series of dial peers configured with a specific destination pattern and shows the number
after the digit strip based on the dial string 408 555-3048 and the different dial peer symbols applied.
Table 17
Examples
Dial Peer Configurations with Digits Stripped
Dial Peer
Destination Pattern
Number After the Digit Strip
1
408555....
3048
2
408555.%
3048
3
408525.+
3048
4
408555.?
3048
5
408555+
3048
6
408555%
53048
7
408555?
53048
8
408555[0-9].%
3048
9
408555(30).%
3048
10
408555(30)%
3048
11
408555..48
3048
The following example disables digit stripping on a POTS dial peer:
dial-peer voice 100 pots
no digit-strip
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
digit-strip
Related Commands
Command
Description
numbering-type
Specifies number type for the VoIP or POTS dial peer.
rule
Applies a translation rule to a calling party number or a called party
number for both incoming and outgoing calls.
show translation-rule
Displays the contents of all the rules that have been configured for a
specific translation name.
test translation-rule
Tests the execution of the translation rules on a specific name-tag.
translation-rule
Creates a translation name and enters translation-rule configuration
mode.
voip-incoming
translation-rule
Captures calls that originate from H.323-compatible clients.
Cisco IOS Voice, Video, and Fax Command Reference
VR-260
Cisco IOS Voice, Video, and Fax Commands: D Through F
direct-inward-dial
direct-inward-dial
To enable the direct inward dial (DID) call treatment for the incoming called number, use the
direct-inward-dial command in dial-peer configuration mode. To disable DID, use the no form of this
command.
direct-inward-dial
no direct-inward-dial
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)NA
This command was introduced on the Cisco AS5300 universal access server.
12.0(4)XJ
This command was implemented for store and forward fax.
12.1(1)
The implementation of the command for store and forward fax was
integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Use the direct-inward-dial command to enable the DID call treatment for the incoming called numbers.
When this feature is enabled, the incoming call is treated as if the digits were received from the DID
trunk. The called number is used to select the outgoing dial peer. No dial tone is presented to the caller.
Use the no form of this command to disable DID. When disabled, the called number is used to select the
outgoing dial peer. The caller will be prompted for a called number via dial tone.
This command is applicable only to plain old telephone service (POTS) dial peers. This command
applies to on-ramp store and forward fax functions.
Examples
The following example enables DID call treatment for incoming called numbers:
dial peer voice 10 pots
direct-inward-dial
Cisco IOS Voice, Video, and Fax Command Reference
VR-261
Cisco IOS Voice, Video, and Fax Commands: D Through F
disc_pi_off
disc_pi_off
To enable an H.323 gateway to disconnect a call when it receives a Disconnect message with a progress
indicator (PI) value, use the disc_pi_off command in voice-port configuration mode. To restore the
default state, use the no form of this command.
disc_pi_off
no disc_pi_off
Syntax Description
This command has no arguments or keywords.
Defaults
The gateway does not disconnect a call when it receives a Disconnect message with a PI value.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.1(5)T
This command was introduced on the Cisco 2600 series routers, 3600
series, 7200 series, 7500 series, AS5300, AS5800, and MC3810
multiservice concentrator.
Usage Guidelines
The disc_pi_off voice-port command is valid only if the Disconnect with PI is received on the inbound
call leg. For example, if this command is enabled on the voice port of the originating gateway, and a
Disconnect with PI is received from the terminating switch, the Disconnect with PI is converted to a
Disconnect. But if this command is enabled on the voice port of the terminating gateway, and a
Disconnect with PI is received from the terminating switch, the Disconnect message is not converted to
a standard Disconnect because the Disconnect message is received on the outbound call leg.
Note
Examples
The disc_pi_off voice-port configuration command is valid only for the default session
application; it does not work for interactive voice response (IVR) applications.
The following example handles a Disconnect message with a PI value the same as a standard Disconnect
message for voice port 0:23:
voice-port 0:D
disc_pi_off
Related Commands
Command
Description
isdn t306
Sets a timer for Disconnect messages.
Cisco IOS Voice, Video, and Fax Command Reference
VR-262
Cisco IOS Voice, Video, and Fax Commands: D Through F
disconnect-ack
disconnect-ack
To configure a Foreign Exchange Station (FXS) voice port to return an acknowledgment upon receipt of
a disconnect signal, use the disconnect-ack command in voice-port configuration mode. To disable the
acknowledgment, use the no form of this command.
disconnect-ack
no disconnect-ack
Syntax Description
This command has no arguments or keywords.
Defaults
FXS voice ports return an acknowledgment upon receipt of a disconnect signal.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series
routers.
12.1(2)T
The support for the Cisco 2600 and 3600 series routers was
integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The disconnect-ack command configures an FXS voice port to remove line power if the equipment on
an FXS loop-start trunk disconnects first.
Examples
The following example, which begins in global configuration mode, turns off the disconnect
acknowledgment signal on voice port 1/1 on a Cisco MC3810 multiservice concentrator:
voice-port 1/1
no disconnect-ack
The following example, which begins in global configuration mode, turns off the disconnect
acknowledgment signal on voice port 1/1/0 on a Cisco 2600 or 3600 series router:
voice-port 1/0/0
no disconnect-ack
Command History
Command
Description
show voice port
Displays voice port configuration information.
Cisco IOS Voice, Video, and Fax Command Reference
VR-263
Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0 busyout (voice)
ds0 busyout (voice)
To force a DS0 time slot on a controller into the busyout state, use the ds0 busyout command in
controller configuration mode. To remove the DS0 time slot from the busyout state, use the no form of
this command.
ds0 busyout ds0-time-slot
no ds0 busyout ds0-time-slot
Syntax Description
ds0-time-slot
Defaults
DS0 time slots are not in busyout state.
Command Modes
Controller configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on Cisco MC3810 multiservice
concentrator and Cisco 2600 and 3600 series routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
DS0 time slots to be forced into the busyout state. The range is
from 1 to 24 and can include any combination of time slots.
The ds0 busyout command affects only DS0 time slots that are configured into a DS0 group and that
function as part of a digital voice port. If multiple DS0 groups are configured on a controller, any
combination of DS0 time slots can be busied out, provided that each DS0 time slot to be busied out is
part of a DS0 group.
If a DS0 time slot is in the busyout state, only the no ds0 busyout command can restore the DS0 time
slot to service.
To avoid conflicting command-line interface (CLI) commands, do not use the ds0 busyout command
and the busyout forced command on the same controller.
Examples
The following example configures DS0 time slot 6 on controller T1 0 to be forced into the busyout state:
controller t1 0
ds0 busyout 6
The following example configures DS0 time slots 1, 3, 4, 5, 6, and 24 on controller E1 1 to be forced
into the busyout state:
controller e1 1
ds0 busyout 1,3-6,24
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0 busyout (voice)
Related Commands
Command
Description
busyout seize
Changes the busyout seize procedure for a voice port.
show running configuration
Determines which DS0 time slots have been forced into the
busyout state.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
ds0-group
To specify the DS0 time slots that make up a logical voice port on a T1 or E1 controller and to specify
the signaling type by which the router communicates with the PBX or Public Switched Telephone
Network (PSTN), use the ds0-group command in controller configuration mode. To remove the group
and signaling setting, use the no form of this command.
Cisco 2600 and 3600 Series and Cisco MC3810 Multiservice Concentrator—T1
ds0-group ds0-group-no 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}
no ds0-group ds0-group-no
Cisco 2600 and 3600 Series and Cisco MC3810 Multiservice Concentrator—E1
ds0-group ds0-group-no timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start |
e&m-melcas-delay | e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start | ext-sig |
fgd-eana | fxo-ground-start | fxo-loop-start | fxo-melcas | fxs-ground-start | fxs-loop-start |
fxs-melcas | r2-analog | r2-digital | r2-pulse}
no ds0-group ds0-group-no
Cisco MC3810 Multiservice Concentrator—E1
ds0-group ds0-group-no timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start |
e&m-melcas-delay | e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start | ext-sig |
fgd-eana | fxo-ground-start | fxo-loop-start | fxo-melcas | fxs-ground-start | fxs-loop-start |
fxs-melcas}
no ds0-group ds0-group-no
Cisco 7200 and 7500 Series T1 and E1 Voice Ports
ds0-group ds0-group-no 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}
no dso-group ds0-group-no
Cisco AS5300 universal access server—T1
ds0-group ds0-group-no timeslots timeslot-list [service service-type] [type {e&m-fgb | e&m-fgd
| e&m-immediate-start | fxs-ground-start | fxs-loop-start | fgd-eana | fgd-os | r1-itu |
sas-ground-start | sas-loop-start | none}] [tone type] [addr info]
no ds0-group ds0-group-no
Cisco AS5300 universal access server—E1
ds0-group ds0-group-no timeslots timeslot-list type {none | p7 | r2-analog | r2-digital |
r2-lsv181-digital | r2-pulse}
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
no ds0-group ds0-group-no
Cisco AS5800 Universal Access Server—T1
ds0-group ds0-group-no timeslots timeslot-list type {e&m-fgb | e&m-fgd | e&m-immediate-start
| fxs-ground-start | fxs-loop-start | fgd-eana | r1-itu | r1-modified | r1-turkey |
sas-ground-start | sas-loop-start | none}
no ds0-group ds0-group-no
Cisco AS5800 E1 Voice Ports
ds0-group ds0-group-no timeslots timeslot-list type {e&m-fgb | e&m-fgd | e&m-immediate-start
| fxs-ground-start | fxs-loop-start | p7 | r2-analog | r2-digital | r2-pulse | sas-ground-start |
sas-loop-start | none}
no ds0-group ds0-group-no
Syntax Description
For the Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multiservice Concentrators—T1:
ds0-group-no
A value from 0 to 23 that identifies the DS0 group.
timeslots timeslot-list
Time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas. For T1,
allowable values are from 1 to 24. Examples are as follows:
•
2
•
1-15,17-24
•
1-23
•
2,4,6-12
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
type
The signaling method selection for the type keyword depends on the
connection that you are making. The ear and mouth (E&M) interface
allows connection for PBX trunk lines (tie lines) and telephone
equipment. The Foreign Exchange Station (FXS) interface allows
connection of basic telephone equipment and PBX. The Foreign
Exchange Office (FXO) interface is for connecting the central office
(CO) to a standard PBX interface where permitted by local regulations;
it is often used for off-premise extensions (OPXs). Types are the
following:
•
e&m-delay-dial—The originating endpoint sends an off-hook
signal and then waits for an off-hook signal followed by an on-hook
signal from the destination.
•
e&m-fgd—E&M Type II Feature Group D.
•
e&m-immediate-start—E&M immediate start.
•
e&m-wink-start—E&M Mercury Exchange Limited
Channel-Associated Signaling (MELCAS) wink-start signaling
support.
•
ext-sig—An option available only when the mode CCS command
is enabled on the Cisco MC3810 multiservice concentrator for
FRF.11 transparent common channel signaling (CCS) support.
•
fgd-eana—Feature Group D exchange access North American.
•
fxo-ground-start—FXO ground-start signaling support.
•
fxo-loop-start—FXO loop-start signaling support.
•
fxs-ground-start—FXS ground-start signaling support.
•
fxs-loop-start—FXS loop-start signaling support.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
For the Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multiservice Concentrators—E1:
ds0-group-no
An identifying value from 0 to 14 and 16 to 30. 15 is reserved.
timeslots timeslot-list
Time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas. For E1,
allowable values are from 1 to 31. Examples are as follows:
type
•
2
•
1-15,17-31
•
1-31
•
2,4,6-12, 17-31
The signaling method selection for the type keyword depends on the
connection that you are making. The E&M interface allows connection
for PBX trunk lines (tie lines) and telephone equipment. The FXS
interface allows connection of basic telephone equipment and PBX. The
FXO interface is for connecting the CO to a standard PBX interface
where permitted by local regulations; it is often used for OPXs. Types are
the following:
•
e&m-delay-dial—The originating endpoint sends an off-hook signal
and then waits for an off-hook signal followed by an on-hook signal
from the destination.
•
e&m-immediate-start—E&M immediate start.
•
e&m-melcas-delay—E&M MELCAS delay-start signaling support.
•
e&m-melcas-immed—E&M MELCAS immediate-start signaling
support.
•
e&m-melcas-wink—E&M MELCAS wink-start signaling support.
•
e&m-wink-start—The originating endpoint sends an off-hook
signal and waits for a wink start from the destination.
•
ext-sig—An option available only when the mode CCS command is
enabled on the Cisco MC3810 multiservice concentrator for FRF.11
transparent CCS support.
•
fgd-eana—Feature Group D exchange access North American.
•
fxo-ground-start—Specifies FXO ground-start signaling.
•
fxo-loop-start—Specifies FXO loop-start signaling.
•
fxo-melcas—MELCAS FXO signaling.
•
fxs-ground-start—FXS ground-start signaling.
•
fxs-loop-start—FXS loop-start signaling.
•
fxs-melcas—MELCAS FXS signaling.
•
r2-analog—Specifies R2 analog line signaling.
•
r2-digital—Specifies R2 digital line signaling.
•
r2-pulse—Specifies 7-pulse line signaling, a transmitted pulse that
indicates a change in the line state.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
For the Cisco 7200 and 7500 Series Routers—T1 and E1:
ds0-group-no
For T1, a value from 0 to 23 that identifies the DS0 group. For E1, a
value from 0 to 14 and 16 to 30. 15 is reserved
timeslots timeslot-list
Time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas. For T1,
allowable values are from 1 to 24. For E1, allowable values are from 1
to 31. Examples are as follows:
type
•
2
•
1-15,17-24
•
1-31
•
2,4,6-12
The signaling method selection for the type keyword depends on the
connection that you are making. The E&M interface allows connection
for PBX trunk lines (tie lines) and telephone equipment. The FXS
interface allows connection of basic telephone equipment and PBX. The
FXO interface is for connecting the CO to a standard PBX interface
where permitted by local regulations; it is often used for OPXs. Types
are the following:
•
e&m-delay-dial—The originating endpoint sends an off-hook
signal and then waits for an off-hook signal followed by an on-hook
signal from the destination.
•
e&m-fgd—E&M Type II Feature Group D.
•
e&m-immediate-start—E&M immediate start.
•
e&m-wink-start—E&M MELCAS wink-start signaling support.
•
fxs-ground-start—FXO ground-start signaling support.
•
fxs-loop-start—FXS loop start.
•
fxo-ground-start—Specifies FXO ground-start signaling.
•
fxo-loop-start—FXO loop-start signaling support.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
For the Cisco AS5300 Universal Access Server—T1:
ds0-group-no
A value from 0 to 23 that identifies the DS0 group.
timeslots timeslot-list
time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas. Allowable
values are from 1 to 24. Examples are as follows:
•
2
•
1-15,17-24
•
1-23
•
2,4,6-12
service service-type
(Optional) Indicates the type of calls to be handled by this DS0
group—data, fax, voice, or mgcp.
type
(Optional) The signaling method selection for the type keyword depends
on the connection that you are making. The E&M interface allows
connection for PBX trunk lines (tie lines) and telephone equipment. The
FXS interface allows connection of basic telephone equipment and PBX.
Types are the following:
•
e&m-fgb—E&M Type II Feature Group B.
•
e&m-fgd—E&M Type II Feature Group D.
•
e&m-immediate-start—E&M immediate start.
•
fxs-ground-start—FXS ground start.
•
fxs-loop-start—FXS loop start.
•
fgd-eana—Feature Group D exchange access North American.
•
fgd-os—Feature Group D operator services.
•
r1-itu—Line signaling based on international signaling standards.
•
sas-ground-start—Single attachment station (SAS) ground start.
•
sas-loop-start—SAS loop start.
•
none—Null signaling for external call control.
tone type
(Optional) Specifies the tone as dtmf or mf.
addr info
(Optional) Specifies the calling/called party.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
For the Cisco AS5300 Universal Access Server—E1:
ds0-group-no
An identifying value from 0 to 14 and 16 to 30. 15 is reserved.
timeslots timeslot-list
time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas. Allowable
values are from 1 to 31. Examples are as follows:
type
•
2
•
1-15,17-31
•
1-31
•
2,4,6-12, 24
The signaling method selection for the type keyword depends on the
connection that you are making. Types are the following:
•
none—Null signaling for external call control.
•
p7—Specifies the p7 switch type.
•
r2-analog—Specifies R2 analog line signaling.
•
r2-digital—Specifies R2 digital line signaling.
•
r2-lsv181-digital—Specifies a specific R2 digital line.
•
r2-pulse—Specifies 7-pulse line signaling, a transmitted pulse that
indicates a change in the line state.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
For the Cisco AS5300 Universal Access Server—T1:
ds0-group-no
A value from 0 to 23 that identifies the DS0 group.
timeslots timeslot-list
time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas.
Allowable values are from 1 to 24. Examples are as follows:
type
•
2
•
1-15,17-24
•
1-23
•
2,4,6-12
The signaling method selection for the type keyword depends on the
connection that you are making. The E&M interface allows connection
for PBX trunk lines (tie lines) and telephone equipment. The FXS
interface allows connection of basic telephone equipment and PBX.
Types are the following:
•
e&m-fgb—E&M Type II Feature Group B.
•
e&m-fgd—E&M Type II Feature Group D.
•
e&m-immediate-start—E&M immediate start.
•
fxs-ground-start—FXS ground start.
•
fxs-loop-start—FXS loop start.
•
fgd-eana—Feature Group D exchange access North American.
•
r1-itu—A line signaling based on international signaling standards.
•
r1-modified—An international signaling standard that is common to
channelized T1/E1 networks.
•
r1-turkey—A signaling standard used in Turkey.
•
sas-ground-start—SAS ground start.
•
sas-loop-start—SAS loop start.
•
none—Null signaling for external call control.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
For the Cisco AS5800 Universal Access Server—E1:
ds0-group-no
An identifying value from 0 to 14 and 16 to 30. 15 is reserved.
timeslots timeslot-list
time slot timeslot-list is a single time-slot number, a single range of
numbers, or multiple ranges of numbers separated by commas.
Allowable values are from 1 to 31. Examples are as follows:
type
•
2
•
1-15,17-31
•
1-31
•
2,4,6-12, 18-31
The signaling method selection for the type keyword depends on the
connection that you are making. The E&M interface allows connection
for PBX trunk lines (tie lines) and telephone equipment. The FXS
interface allows connection of basic telephone equipment and PBX.
Types are the following:
•
e&m-fgb—E&M Type II Feature Group B.
•
e&m-fgd—E&M Type II Feature Group D.
•
e&m-immediate-start—E&M immediate start.
•
fxs-ground-start—FXS ground start.
•
fxs-loop-start—FXS loop start.
•
p7—Specifies the p7 switch type.
•
r2-analog—Specifies R2 analog line signaling.
•
r2-digital—Specifies R2 digital line signaling.
•
r2-pulse—Specifies 7-pulse line signaling, a transmitted pulse that
indicates a change in the line state.
•
sas-ground-start—SAS ground start.
•
sas-loop-start—SAS loop start.
•
none—Null signaling for external call control.
Defaults
No DS0 group is defined.
Command Modes
Controller configuration
Command History
Release
Modification
11.2
This command was introduced for the Cisco AS5300 universal access server
as the cas-group command.
11.3(1)MA
The command was introduced as the voice-group command for the
Cisco MC3810 multiservice concentrator.
12.0(1)T
The cas-group command was introduced for the Cisco 3600 series routers.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ds0-group
Usage Guidelines
Release
Modification
12.0(5)T
The command was renamed ds0-group on the Cisco AS5300 and
Cisco 2600 and 3600 series routers. Some keyword modifications were
implemented.
12.0(5)XE
This command was introduced for the Cisco 7200 series.
12.0(7)XK
Support for this command was extended to the Cisco MC3810 multiservice
concentrator. When the ds0-group command became available on the Cisco
MC3810 multiservice concentrator, the voice-group command was removed
and no longer supported. The ext-sig keyword replaced the ext-sig-master
and ext-sig-slave keywords that were available with the voice-group
command.
12.0(7)XR
The mgcp service type was added.
12.1(1)T
The ds0-group command was implemented for the Cisco 7200 series.
12.1(2)XH
The e&m-fgd and fgd-eana keywords were added for Feature Group D
signaling.
12.1(3)T
The command was modified for the Cisco 7500 series routers. The fgd-os
signaling type and the voice service type were added.
The ds0-group command automatically creates a logical voice port that is numbered as follows:
•
Cisco 2600, 3600, and 7200 series routers:
– slot/port:ds0-group-no
•
Cisco MC3810 multiservice concentrators:
– slot:ds0-group-no
On the Cisco MC3810 multiservice concentrator, the slot number is the controller number.
Although only one voice port is created for each group, applicable calls are routed to any channel in the
group.
Note
Examples
Channel groups, CAS voice groups, DS0 groups, and TDM groups all use group numbers. All group
numbers configured for channel groups, CAS voice groups, DS0 groups, and TDM groups must be
unique on the local router. For example, you cannot use the same group number for a channel group
and for a TDM group.
The following example shows ranges of T1 controller time slots configured for FXS ground-start and
FXO loop-start signaling on a Cisco 2600 or 3600 series router:
T1 1/0
framing esf
linecode b8zs
ds0-group 1 timeslots 1-10 type fxs-ground-start
ds0-group 2 timeslots 11-24 type fxo-loop-start
The following example shows DS0 groups 1 and 2 on controller T1 1 configured on the Cisco MC3810
multiservice concentrator to support Transparent CCS:
controller T1 1
mode ccs cross-connect
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ds0-group
ds0-group 1 timeslots 1-10 type ext-sig
ds0-group 2 timeslots 11-24 type ext-sig
Related Commands
Command
Description
codec
Specifies the voice coder rate of speech for a dial peer.
codec complexity
Specifies call density and codec complexity based on the codec standard you
are using.
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dsn
dsn
To specify that a delivery status notice be delivered to the sender, use the dsn command in dial-peer
configuration mode. To cancel a specific delay status notice option, use the no form of this command.
dsn {delay | failure | success}
no dsn {delay | failure | success}
Syntax Description
delay
Indicates that when the mail is sent, the next-hop mailer is requested to send a
message to the FROM address if the mail message is delayed. The definition of
delay is made by each mailer and is not controllable by the sender (the
Cisco AS5300 universal access server). Each mailer in the path to the recipient
that supports the delivery status notification (DSN) extension receives the same
request.
failure
Indicates that when the mail is sent, the next-hop mailer is requested to send a
message to the FROM address if the mail message failed to be delivered. Each
mailer in the path to the recipient that supports the DSN extension receives the
same request.
success
Indicates that when the mail is sent, the next-hop mailer is requested to send a
message to the FROM address if the mail message is successfully delivered to
the recipient. Each mailer in the path to the recipient that supports the DSN
extension receives the same request.
Note
In the absence of any other DSN settings (“no dsn” or a mailer in the path that does not
support the DSN extension), a failure to deliver always causes a nondelivery message to be
generated. This nondelivery message is colloquially termed a “bounce.”
Defaults
The default is success and failure.
Command Modes
Dial-peer configuration
Command History
Release
Usage Guidelines
Modification
12.0(4)XJ
This command was introduced.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
This command is applicable to Multimedia Mail over Internet Protocol (MMoIP) dial peers.
DSNs are messages or responses that are automatically generated and sent to the sender or originator of
an e-mail message by the Simple Mail Transfer Protocol (SMTP) server, notifying the sender of the
status of the e-mail message. Specifications for DSN are described in RFC 1891, RFC 1892, RFC 1893,
and RFC 1894.
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dsn
The on-ramp DSN request is included as part of the fax-mail message sent by the on-ramp gateway when
the matching MMoIP dial peer has been configured. The on-ramp DSN response is generated by the
SMTP server when the fax-mail message is accepted. The DSN is sent back to the user defined in the
mta send mail-from command. The off-ramp DSN is requested by the e-mail client. The DSN response
is generated by the SMTP server when it receives a request as part of the fax-mail message.
Note
DSNs can be generated only if the mail client on the SMTP server is capable of responding
to a DSN request.
Because the SMTP server generates the DSNs, configure both the mail from: and rcpt to: commands
for the DSN feature to be operational, for example:
mail from: <[email protected]>
rcpt to: <[email protected]> NOTIFY=SUCCESS,FAILURE,DELAY
There are three different states that can be reported back to the sender:
•
Delay—Indicates that, for some reason, the message was delayed while being delivered to the
recipient.
•
Success—Indicates that the message was successfully delivered to the recipient’s mailbox.
•
Failure—Indicates that, for some reason, the SMTP server was unable to deliver the message to the
recipient.
Because these delivery states are not mutually exclusive, store and forward fax can be configured to
generate these messages for all or any combination of these events.
DSN messages notify the sender of the status of a particular e-mail message containing a fax Tag Image
File Format (TIFF) image. Use the dsn command to specify which notification messages will be sent to
the user.
The dsn command allows you to select more than one notification option by reissuing the command,
specifying a different notification option each time. To discontinue a specific notification option, use the
no form of the command for that specific keyword.
Note
If the failure keyword is not included when configuring DSN, the sender will receive absolutely no
notification of message delivery failure. Because a failure is usually significant, care should be taken
to always include the failure keyword as part of the dsn command configuration.
This command applies to on-ramp store and forward fax functions.
Examples
The following example specifies that a DSN message be returned to the sender when the e-mail message
containing the fax has been successfully delivered to the recipient or if the message containing the fax
has failed, for whatever reason, to be delivered:
dial-peer voice 10 mmoip
dsn success
dsn failure
The following example specifies that a DSN message be returned to the sender either when the e-mail
message containing the fax has been successfully delivered to the recipient or when the message has
been delayed:
dial-peer voice 10 mmoip
dsn success
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dsn
dsn delayed
Related Commands
Command
Description
mta send mail-from
Specifies the mail-from address (also called the RFC 821 envelope-from or
the Return-Path address).
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dspint dspfarm
dspint dspfarm
To enable the digital signal processor (DSP) interface, use the dspint dspfarm command in global
configuration mode.
dspint dspfarm slot/port
Syntax Description
slot
Specifies the slot number of the interface.
port
Specifies the port number of the interface.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)XE
This command was introduced on the Cisco 7200 series routers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
To disable the DSPfarm interface, enter the no shutdown command.
Examples
The following example creates a DSPfarm interface with a slot number of 1 and a port number of 0.
dspint dspfarm 1/0
Related Commands
Command
Description
show interfaces
dspfarm dsp
Displays information about the DSP interface.
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dtmf-relay (Voice over IP)
dtmf-relay (Voice over IP)
To specify how an H.323 gateway relays dual tone multifrequency (DTMF) tones between telephony
interfaces and an IP network, use the dtmf-relay command in dial-peer configuration mode. To remove
all signaling options and to send the DTMF tones as part of the audio stream, use the no form of this
command.
dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal]
no dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal]
Syntax Description
cisco-rtp
(Optional) Forwards DTMF tones by using Real-Time
Transport Protocol (RTP) with a Cisco proprietary payload
type.
h245-alphanumeric
(Optional) Forwards DTMF tones by using the H.245
“alphanumeric” User Input Indication method. Supports tones
0-9, *, #, and A-D.
h245-signal
(Optional) Forwards DTMF tones by using the H.245 “signal”
User Input Indication method. Supports tones 0-9, *, #, and
A-D.
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco AS5300 universal
access server.
12.0(2)XH
The cisco-rtp, h245-alphanumeric, and h245-signal
keywords were added.
12.0(5)T
This command was integrated into Cisco IOS Release 12.0(5)T.
12.0(7)XK
This command was first supported for Voice over IP (VoIP) on
the MC3810 multiservice concentrator.
12.1(2)T
Changes made in Cisco IOS Release 12.0(7)XK were integrated
into Cisco IOS Release 12.1(2)T.
Usage Guidelines
DTMF is the tone generated when you press a digit on a touch-tone phone. This tone is compressed at
one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on
the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out
of band using a standard H.323 out-of-band method and a proprietary RTP-based mechanism.
The dtmf-relay command determines the outgoing format of relayed DTMF tones. The gateway
automatically accepts all formats.
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dtmf-relay (Voice over IP)
The gateway sends DTMF tones in the format that you specify only if the remote device supports it. If
the remote device supports multiple formats, the gateway chooses the format based on the following
priority:
1.
cisco-rtp (highest priority)
2.
h245-signal
3.
h245-alphanumeric
4.
None—DTMF sent in-band
The principal advantage of the dtmf-relay command is that it sends DTMF tones with greater fidelity
than is possible in-band for most low-bandwidth codecs, such as G.729 and G.723. Without the use of
DTMF relay, calls established with low-bandwidth codecs may have trouble accessing automated
DTMF-based systems, such as voice mail, menu-based ACD systems, and automated banking systems.
Examples
Note
The cisco-rtp option of the dtmf-relay command is a proprietary Cisco implementation and
operates only between two Cisco AS5800 universal access servers running
Cisco IOS Release 12.0(2)XH, or between Cisco AS5800 universal access servers or Cisco
2600 or 3600 modular access routers running Cisco IOS Release 12.0(2)XH or later releases.
Otherwise, the DTMF relay feature does not function, and the gateway sends DTMF tones
in-band.
Note
The h245-alphanumeric and h245-signal DTMF settings on an MC3810 multiservice
concentrator require a high-performance compression module (HCM) and are not supported
on an MC3810 multiservice concentrator with a non-HCM voice compression module
(VCM).
The following example configures DTMF relay with the cisco-rtp option when sending DTMF tones to
dial peer 103:
dial-peer voice 103 voip
dtmf-relay cisco-rtp
end
The next example configures DTMF relay with the cisco-rtp or h245-signal options when sending
DTMF tones to dial peer 103:
dial-peer voice 103 voip
dtmf-relay cisco-rtp h245-signal
end
The next example configures the gateway to send DTMF in-band (the default) when sending DTMF
tones to dial peer 103:
dial-peer voice 103 voip
no dtmf-relay
end
Related Commands
Command
Description
dial-peer voice
Specifies the method of voice-related encapsulation.
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dtmf-relay (Voice over Frame Relay)
dtmf-relay (Voice over Frame Relay)
To enable the generation of FRF.11 Annex A frames for a dial peer, use the dtmf-relay command in
dial-peer configuration mode. To disable the generation of FRF.11 Annex A frames and return to the
default handling of dial digits, use the no form of this command.
dtmf-relay
no dtmf-relay
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(3)XG
This command was introduced on the Cisco 2600 series routers, 3600 series,
and MC3810 multiservice concentrator.
12.0(4)T
This command was integrated into Cisco IOS Release 12.0(4)T, and support
for the Cisco 7200 series router was added.
Usage Guidelines
This command applies to all Voice over Frame Relay (VoFR) and Voice over ATM (VoATM)
applications on the Cisco MC3810 multiservice concentrator and to VoFR applications on the Cisco
2600 series routers and 3600 series routers.
Cisco recommends that this command be used with low bit-rate codecs.
When dtmf-relay (VoFR) is enabled, the digital signal processor (DSP) generates Annex A frames
instead of passing a dual tone multifrequency (DTMF) tone through the network as a voice sample. For
information about the payload format of FRF.11 Annex A frames, refer to the Cisco IOS Wide-Area
Networking Configuration Guide and Cisco IOS Wide-Area Networking Command Reference,
Release 12.2.
Examples
The following example shows how to enable FRF.11 Annex A frames on a Cisco 2600 series routers or
3600 series router or on an MC3810 multiservice concentrator for VoFR dial peer 200, starting from
global configuration mode:
dial-peer voice 200 vofr
dtmf-relay
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dtmf-relay (Voice over Frame Relay)
Related Commands
Command
Description
called-number
(dial-peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call
leg when using a static FRF.11 trunk connection.
codec (dial-peer)
Specifies the voice coder rate of speech for a VoFR dial peer.
connection
Specifies a connection mode for a voice port.
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence
setting.
destination-pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory
number (depending on the dial plan) to be used for a dial peer.
preference
Indicates the preferred order of a dial peer within a rotary hunt group.
session protocol
Establishes a session protocol for calls between the local and remote routers
via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination
gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
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dtmf timer inter-digit
dtmf timer inter-digit
To configure the dual tone multifrequency (DTMF) interdigit timer for a DS0 group, use the dtmf timer
inter-digit command in T1 controller configuration mode. To restore the timer to its default value, use
the no form of this command.
dtmf timer inter-digit milliseconds
no dtmf timer inter-digit milliseconds
Syntax Description
milliseconds
Defaults
3000 milliseconds
Command Modes
T1 controller configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300 universal access server.
DTMF interdigit timer duration, in milliseconds. The valid range is from
250 to 3000. The default is 3000 milliseconds.
Usage Guidelines
Use the dtmf timer inter-digit command to specify the duration in milliseconds the router waits to
detect the end of DTMF digits. After this period, the router expects no more digits to arrive and
establishes the call.
Examples
The following example, beginning in global configuration mode, sets the DTMF interdigit timer value
to 250 milliseconds:
controller T1 2
ds0-group 2 timeslots 4-10 type e&m-fgb dtmf dnis
cas-custom 2
dtmf timer inter-digit 250
Related Commands
Command
Description
cas-custom
Customizes E1 R2 signaling parameters for a particular E1 channel group
on a channelized E1 line.
ds0-group
Configures channelized T1 time slots, which enables a Cisco AS5300
universal access server modem to answer and send an analog call.
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echo-cancel comfort-noise
echo-cancel comfort-noise
To specify that background noise be generated, use the echo-cancel comfort-noise command in
controller configuration mode. To disable this feature, use the no form of this command.
echo-cancel comfort-noise
no echo-cancel comfort-noise
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
Use the echo-cancel comfort-noise command to generate background noise to fill silent gaps during
calls if voice activated dialing (VAD) is activated. If comfort noise is not enabled and VAD is enabled
at the remote end of the connection, the user hears nothing or silence when the remote party is not
speaking.
The configuration of comfort noise affects only the silence generated at the local interface; it does not
affect the use of VAD on either end of the connection or the silence generated at the remote end of the
connection.
For the OC-3/STM-1 ATM Circuit Emulation Service network module, echo cancellation must be
enabled.
Examples
The following example enables comfort noise on a T1 controller:
controller T1 0/0
echo-cancel enable
echo-cancel comfort-noise
Related Commands
Command
Description
echo-cancel enable
(controller)
Enables echo cancellation on a voice port.
voice port
Specifies which port is used for voice traffic.
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echo-cancel compensation
echo-cancel compensation
To set attenuation for loud signals, use the echo-cancel compensation command in controller
configuration. To disable this feature, use the no form of this command.
echo-cancel compensation
no echo-cancel compensation
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 2600 series routers.
Usage Guidelines
Use the echo-cancel compensation command to add attenuation control to the T1 or E1 controller.
When this command is enabled, 6 decibels of attenuation are inserted if the signal level from the receive
direction is loud. When loud signals are not received, the attenuation is removed.
For the OC-3/STM-1 ATM Circuit Emulation Service network module, echo cancellation must be
enabled.
Examples
The following example enables attenuation control on a T1 controller:
controller T1 0/0
echo-cancel enable
echo-cancel compensation
Related Commands
Command
Description
echo-cancel enable
(controller)
Enables echo cancellation on a voice port.
voice port
Specifies which port is used for voice traffic.
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echo-cancel coverage
echo-cancel coverage
To adjust the maximum duration to cancel the voice echo, use the echo-cancel coverage command in
voice-port configuration mode. To reset this command to the default value, use the no form of this
command.
echo-cancel coverage {8 | 16 | 24 | 32}
no echo-cancel coverage
Syntax Description
8
8 milliseconds.
16
16 milliseconds.
24
24 milliseconds.
32
24 milliseconds.
Defaults
16 milliseconds
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3(1)MA
This command was implemented on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
12.0(5)XK
The command was modified to add the 8-millisecond option.
12.0(5)XE
The command was modified to support the Cisco 7200 router platform.
12.1(1)T
The Cisco IOS Release 12.0(5)XK and 12.0(5)XE changes were integrated
into Cisco IOS Release 12.1(1)T.
Use the echo-cancel coverage command to adjust the coverage size of the echo canceller. This
command enables cancellation of voice that is sent out the interface and received back on the same
interface within the configured amount of time. If the local loop (the distance from the interface to the
connected equipment that is producing the echo) is longer, the configured value of this command should
be extended.
If you configure a longer value for this command, it takes the echo canceller longer to converge; in this
case, the user might hear a slight echo when the connection is initially set up. If the configured value for
this command is too short, the user might hear some echo for the duration of the call because the echo
canceller is not cancelling the longer delay echoes.
There is no echo or echo cancellation on the network side (for example, the non-POTS side) of the
connection.
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echo-cancel coverage
Note
Examples
This command is valid only if the echo cancel feature has been enabled. For more information, see
the echo-cancel enable command.
The following example adjusts the size of the echo canceller to 8 milliseconds on the Cisco 7200 series
routers:
voice-port 1/0:0
echo-cancel enable
echo-cancel coverage 8
Related Commands
Command
Description
echo-cancel enable
(controller)
Enables echo cancellation on a voice port.
echo-cancel enable
Enable echo cancellation on a voice port.
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echo-cancel enable
echo-cancel enable
To enable the cancellation of voice that is sent out the interface and is received back on the same
interface, use the echo-cancel enable command in voice-port configuration mode. To disable echo
cancellation, use the no form of this command.
echo-cancel enable
no echo-cancel enable
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled for all interface types.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced for the Cisco 3600 series routers.
Usage Guidelines
The echo-cancel enable command enables cancellation of voice that is sent out the interface and is
received back on the same interface; sound that is received back in this manner is perceived by the
listener as an echo. Disabling echo cancellation might cause the remote side of a connection to hear an
echo. Because echo cancellation is an invasive process that can minimally degrade voice quality, this
command should be disabled if it is not needed.
The echo-cancel enable command does not affect the echo heard by the user on the analog side of the
connection.
There is no echo path for a 4-wire ear and mouth (E&M) interface. The echo canceller should be disabled
for this interface type.
Note
Examples
This command is valid only if the echo-cancel coverage command has been configured. For more
information, refer to the echo-cancel coverage command.
The following example enables the echo cancellation feature and adjusts the size of the echo canceller
to 16 milliseconds on the Cisco 3600 series routers:
voice-port 1/0/0
echo-cancel enable
echo-cancel coverage 16
The following example enables the echo cancellation feature and adjusts the size of the echo canceller
to 16 milliseconds on the Cisco MC3810 multiservice concentrator:
voice-port 1/1
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echo-cancel enable
echo-cancel enable
echo-cancel coverage 16
Related Commands
Command
Description
echo-cancel coverage
Specifies the amount of coverage for echo cancellation.
echo-cancel enable
(controller)
Enables the echo cancellation on a voice port.
non-linear
Enables nonlinear processing in the echo canceler.
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echo-cancel enable (controller)
echo-cancel enable (controller)
To enable the echo cancel feature, use the echo-cancel enable command in controller configuration
mode. To disable this feature, use the no form of this command.
echo-cancel enable
no echo-cancel enable
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled for all interface types.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(2)T
This command was introduced for the OC-3/STM-1 ATM Circuit
Emulation Service network module on the Cisco 3600 series routers.
Usage Guidelines
The echo-cancel enable command enables cancellation of voice that is sent out of the interface and
received back on the same interface. Disabling echo cancellation might cause the remote side of a
connection to hear an echo. Because echo cancellation is an invasive process that can minimally degrade
voice quality, this command should be disabled if it is not needed.
The echo-cancel enable command does not affect the echo heard by the user on the analog side of the
connection.
Note
This command is valid only if the echo-cancel coverage command has been configured.
The following example enables the echo cancel feature on a T1 controller:
controller T1 0/0
echo-cancel enable
echo-cancel coverage 32
Related Commands
Command
Description
echo-cancel coverage
Specifies the amount of coverage for echo cancellation.
echo-cancel enable
Enables the echo cancellation on a voice port.
non-linear
Enables nonlinear processing in the echo canceler.
voice port
Configures the voice port.
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echo-cancel loopback
echo-cancel loopback
To place the echo cancellation processor in loopback mode, use the echo-cancel loopback command in
controller configuration mode. To disable loopback of the echo cancellation processor, use the no form
of this command.
echo-cancel loopback
no echo-cancel loopback
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
Use an echo-cancel loopback test on lines to detect and distinguish equipment malfunctions caused by
either the line or the interface. If correct echo cancellation is not possible when an interface is in
loopback mode, the interface is the source of the problem.
Examples
On a Cisco 3600 series routers router, the following example sets up echo cancellation loopback
diagnostics:
controller T1 0/0
echo-cancel enable
echo-cancel coverage 32
echo-cancel loopback
Related Commands
Command
Description
echo-cancel enable
(controller)
Enables echo cancellation on a voice port.
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encapsulation atm-ces
encapsulation atm-ces
To enable circuit emulation service (CES) ATM encapsulation on the Cisco MC3810 multiservice
concentrator, use the encapsulation atm-ces command in interface configuration mode. To disable CES
ATM encapsulation, use the no form of this command.
encapsulation atm-ces
no encapsulation atm-ces
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0
This command was integrated into Cisco IOS Release 12.0.
Usage Guidelines
This command applies to ATM configuration on the Cisco MC3810 multiservice concentrator. This
command is supported only on serial ports 0 and 1.
Examples
The following example enables CES ATM encapsulation on serial port 0 on the Cisco MC3810
multiservice concentrator:
interface serial 0
encapsulation atm-ces
Related Commands
Command
Description
ces
cell-loss-integration
-period
Sets the CES cell-loss integration period on the Cisco MC3810 multiservice
concentrator.
ces clockmode
synchronous
Configures the ATM CES synchronous clock mode on the Cisco MC3810
multiservice concentrator.
ces connect
Maps the CES service to an ATM PVC on the Cisco MC3810 multiservice
concentrator.
ces initial-delay
Configures the size of the receive buffer of a CES circuit on the Cisco MC3810
multiservice concentrator.
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encapsulation atm-ces
Command
Description
ces max-buf-size
Configures the send buffer of a CES circuit on the Cisco MC3810 multiservice
concentrator.
ces partial-fill
Configures the number of user octets per cell for the ATM CES on the Cisco
MC3810 multiservice concentrator.
ces service
Configures the ATM CES type on the Cisco MC3810 multiservice
concentrator.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
encapsulation ftc-trunk
encapsulation ftc-trunk
This command was removed in Cisco IOS Release 12.1(2)T and is no longer supported in Cisco IOS
Release 12.2.
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encryption
encryption
To set the algorithm to be negotiated with the provider, use the encryption command in settlement
configuration mode. To reset to the default encryption method, use the no form of this command.
encryption {des-cbc-sha | des40-cbc-sha | dh-des-cbc-sha | dh-des40-cbc-sha | null-md5 |
null-sha | all}
no encryption {des-cbc-sha | des40-cbc-sha | dh-des-cbc-sha | dh-des40-cbc-sha | null-md5 |
null-sha | all}
Syntax Description
des-cbc-sha
Encryption type ssl_rsa_with_des_cbc_sha cipher suite.
des40-cbc-sha
Encryption type ssl_rsa_export_with_des40_cbc_sha cipher suite.
dh-des-cbc-sha
Encryption type ssl_dh_rsa_with_des_cbc_sha cipher suite.
dh-des40-cbc-sha
Encryption type ssl_dh_rsa_export_with_des40_cbc_sha cipher
suite.
null-md5
Encryption type ssl_rsa_with_null_md5 cipher suite.
null-sha
Encryption type ssl_rsa_with_null_sha cipher suite.
all
All encryption methods are used in the Secure Socket Layer (SSL).
Defaults
The default encryption method is all. If none of the encryption methods is configured, the system uses
all of the encryption methods in the SSL session negotiation.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 and 3600 series and
on the Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
For Cisco IOS Release 12.0(4)XH1, only one encryption method is allowed for each provider.
Examples
The following example sets the algorithm to be negotiated with the provider, using the encryption
command:
settlement 0
encryption des-cbc-sha
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encryption
Related Commands
Command
Description
connection-timeout
Sets the connection timeout.
customer-id
Sets the customer identification.
device-id
Sets the device identification.
max-connection
Sets the maximum number of simultaneous connections.
response-timeout
Sets the response timeout.
retry-delay
Sets the retry delay.
retry-limit
Sets the connection retry limit.
session-timeout
Sets the session timeout.
settlement
Enters settlement configuration mode.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Disables the settlement provider.
type
Specifies the provider type.
url
Specifies the ISP address.
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erase vfc
erase vfc
To erase the Flash memory of a specified voice feature card (VFC), use the erase vfc command in
privileged EXEC mode.
erase vfc slot
Syntax Description
slot
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco AS5300 universal
access server.
Specifies the slot on the Cisco AS5300 universal access server
in which the specified VFC resides. Valid entries are from 0 to
2.
Usage Guidelines
Use the erase vfc command to erase the contents of Flash memory for a specified VFC (thereby freeing
space in VFC Flash memory) including the default file list and the capability file list.
Examples
The following example erases the Flash memory on the VFC located in slot 0:
Router# erase vfc 0
Related Commands
Command
Description
delete vfc
Deletes a file from VFC Flash memory.
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expect-factor
expect-factor
To specify when the router generates an alarm to the network manager, indicating that the expected
quality of voice has dropped, use the expect-factor command in dial-peer configuration mode. To reset
the default value, use the no form of this command.
expect-factor value
no expect-factor value
Syntax Description
value
Defaults
0
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
Integers that represent the International Telecommunication Union (ITU)
specification for quality of voice as described in G.113. Valid entries are from
0 to 20, with 0 representing toll quality.
This command applies to Voice over IP (VoIP) dial peers.
VoIP monitors the quality of voice received over the network. Use the expect-factor command to
specify when the router generates a Simple Network Management Protocol (SNMP) trap to the network
manager.
Examples
The following example configures toll quality of voice when connecting to a dial peer:
dial-peer voice 10 voip
expect-factor 0
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fax interface-type
fax interface-type
To specify the voice feature card (VFC) to be used for a fax call, use the fax interface-type command
in global configuration mode. To return to the default fax protocol, use the no form of this command.
fax interface-type {modem | vfc}
no fax interface-type {modem | vfc}
Syntax Description
modem
Specifies modem fax calls.
vfc
Specifies the VFC fax calls.
Defaults
None
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300 access server.
12.1(5)T
This command was integrated into the Cisco IOS Release 12.1(5)T.
Usage Guidelines
When using this command to change the interface type for fax calls, you must reload (reboot or reset)
the router.
On the Cisco AS5300 access server, the keyword vfc maps to the fax-mail keyword. If you enter the
show run command, the fax-mail keyword will display. The voice gateway defaults are as follows:
Examples
•
If the gateway has modem cards only, the default is the modem keyword.
•
If the gateway has voice cards only, the default is the fax-mail keyword. The modem keyword is
unavailable. This applies to all platforms except the Cisco AS5300 access server.
•
If the gateway has both modems and voice cards, the default is the modem keyword.
The following example specifies the use of a VFC interface for fax calls:
configure terminal
fax interface-type vfc
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fax protocol (dial-peer)
fax protocol (dial-peer)
To specify the fax protocol for a specific Voice over IP (VoIP) dial peer, use the fax protocol command
in dial-peer configuration mode. To return to the default fax protocol, use the system keyword. To
disable the T.38 fax protocol for a specific dial peer, use the no form of this command.
fax protocol {cisco | t38 [ls_redundancy value] [hs_redundancy value] | system}
no fax protocol
Syntax Description
cisco
Cisco proprietary fax protocol.
t38
ITU-T T.38 standard fax protocol.
ls_redundancy value
(Optional) Low-speed redundancy for the T.38 fax protocol. The
value can be from 0 to 5. The default is 0. The ls_redundancy
parameter refers to data redundancy in the low-speed V.21-based
T.30 fax machine protocol.
hs_redundancy value
(Optional) High-speed redundancy for the T.38 fax protocol. The
value can be from 0 to 2. The default is 0. The hs_redundancy
parameter refers to data redundancy in the high-speed V.17, V.27, and
V.29 T.4 or T.6 fax machine image data.
system
Global fax protocol when neither cisco nor t38 is specified. The value
is taken from the global configuration by default.
Defaults
The default protocol is system.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers,
Cisco 3600 series routers, and Cisco MC3810 multiservice
concentrators.
Usage Guidelines
Use the fax protocol t38 command to configure T.38 Fax Relay for a specific dial peer. The t38 keyword
enables the T.38 Fax Relay protocol. The cisco keyword selects the original Cisco proprietary fax
protocol. When the system keyword is selected in the dial peer, it specifies the global default fax
protocol used by a dial peer, set by the fax protocol t.38 command. The optional ls_redundancy and
hs_redundancy parameters are used to send redundant T.38 fax packets.
Note
The ls_redundancy and hs_redundancy parameters are applicable only to the
T.38 Fax Relay protocol.
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fax protocol (dial-peer)
The ls_redundancy parameter refers to data redundancy in the low-speed, V.21-based T.30 fax machine
protocol. For the ls_redundancy parameter, the value can be from 0 to 5. The default is 0 (no
redundancy). The parameter value sets the redundancy factor for T.38 Fax Relay.
The hs_redundancy parameter refers to data redundancy in the high-speed V.17, V.27, and V.29 T.4 or
T.6 fax machine image data. For the hs_redundancy parameter, the value can be from 0 to 2. The default
is 0 (no redundancy). The parameter value sets the redundancy factor for T.38 Fax Relay.
Note
Examples
Setting the hs_redundancy parameter to a value greater than 0 causes a significant increase
in the network bandwidth consumed by the fax call.
The following example configures T.38 Fax Relay for VoIP, beginning in global configuration mode:
dial-peer voice 99 voip
fax protocol t38
Related Commands
Command
Description
fax rate
Establishes the rate at which a fax is sent to the specified dial peer.
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fax protocol (voice-service)
fax protocol (voice-service)
To specify the global default fax protocol for all Voice over IP (VoIP) dial peers, use the fax protocol
command in voice-service configuration mode. To return to the default fax protocol, use the no form of
this command.
fax protocol {cisco | t38 [ls_redundancy value] [hs_redundancy value]}
no fax protocol
Syntax Description
cisco
Cisco proprietary fax protocol.
t38
ITU-T T.38 standard fax protocol.
ls_redundancy value
(Optional) Low-speed redundancy for the T.38 fax protocol. The
value can be from 0 to 5. The default is 0. The ls_redundancy
parameter refers to data redundancy in the low-speed V.21-based
T.30 fax machine protocol.
hs_redundancy value
(Optional) High-speed redundancy for the T.38 fax protocol. The
value can be from 0 to 2. The default is 0. The hs_redundancy
parameter refers to data redundancy in the high-speed V.17, V.27, and
V.29 T.4 or T.6 fax machine image data.
Defaults
Cisco fax protocol
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series routers,
Cisco 3600 series routers, and Cisco MC3810 multiservice
concentrators.
Usage Guidelines
Use the fax protocol t38 command to configure T.38 Fax Relay for VoIP. The t38 keyword enables the
T.38 Fax Relay protocol. The cisco keyword selects the original Cisco proprietary fax protocol. When
the system keyword is selected in the dial peer, it specifies the global default fax protocol used by a dial
peer, set by the fax protocol t.38 command. The optional ls_redundancy and hs_redundancy
parameters are used to send redundant T.38 fax packets.
Note
The ls_redundancy and hs_redundancy parameters are applicable only to the
T.38 Fax Relay protocol.
The ls_redundancy parameter refers to data redundancy in the low-speed V.21-based T.30 fax machine
protocol. For the ls_redundancy parameter, the value can be from 0 to 5. The default is 0 (no
redundancy). The parameter value sets the redundancy factor for T.38 Fax Relay.
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fax protocol (voice-service)
The hs_redundancy parameter refers to data redundancy in the high-speed V.17, V.27, and V.29 T.4 or
T.6 fax machine image data. For the hs_redundancy parameter, the value can be from 0 to 2. The default
is 0 (no redundancy). The parameter value sets the redundancy factor for T.38 Fax Relay.
Note
Examples
Setting the hs_redundancy parameter to a value greater than 0 causes a significant
increase in the network bandwidth consumed by the fax call.
The following example configures the T.38 fax protocol for VoIP, beginning in global configuration
mode:
voice service voip
fax protocol t38
Related Commands
Command
Description
fax protocol (dial-peer)
Specifies the fax protocol for a specific VoIP dial peer.
fax protocol t.38
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fax rate
fax rate
To establish the rate at which a fax is sent to the specified dial peer, use the fax rate command in
dial-peer configuration mode. To reset the dial peer for voice calls, use the no form of this command.
fax rate {2400 | 4800 | 7200 | 9600 | 12000 | 14400} {disable | voice} [bytes rate]
no fax rate
Syntax Description
2400
Specifies a fax transmission speed of 2400 bits per second (bps).
4800
Specifies a fax transmission speed of 4800 bps.
7200
Specifies a fax transmission speed of 7200 bps.
9600
Specifies a fax transmission speed of 9600 bps.
12000
Specifies a fax transmission speed of 12,000 bps.
14400
Specifies a fax transmission speed of 14,400 bps.
disable
Disables Fax Relay transmission capability.
voice
Specifies the highest possible transmission speed allowed by the voice rate.
For example, if the voice codec is G.711, fax transmission may occur up to
14,400 bps because 14,400 bps is less than the 64-k voice rate. If the voice
codec is G.729 (8k), the fax transmission speed will be 7200 bps.
bytes rate
(Optional) Specifies fax packetization rate, in milliseconds. Range is 20 to
48. Default is 20.
•
For Cisco fax relay, this keyword-argument pair is valid only on Cisco
2600 series, Cisco 3600 series, Cisco 5300, and Cisco 7200 series
routers.
•
For T.38 fax relay, this keyword-argument pair is valid only on
Cisco 5350, Cisco 5400, and Cisco 5850 routers. For other routers, the
packetization rate for T.38 fax relay is fixed at 40 ms and cannot be
changed.
Defaults
Voice calls
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced as the fax-rate command on the Cisco 3600
series routers.
12.0(2)XH
The fax transmission rate of 12000 was added.
12.0(4)T
This command was supported on the Cisco MC3810 multiservice
concentrator.
12.1(3)T
The command name changed from fax-rate to fax rate (nonhyphenated).
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fax rate
Usage Guidelines
Release
Modification
12.1(3)XI
This command was implemented on the Cisco AS5300 universal access
server.
12.1(5)T
The modifications made in Cisco IOS Release 12.1(3)XI were integrated
into Cisco IOS Release 12.1(5)T.
Use the fax rate command to specify the fax transmission rate to the specified dial peer.
The values for this command apply only to the fax transmission speed and do not affect the quality of
the fax itself. The higher transmission speed values (14,400 bps) provide a faster transmission speed but
monopolize a significantly large portion of the available bandwidth. The lower transmission speed
values (2400 bps) provide a slower transmission speed and use a relatively smaller portion of the
available bandwidth.
Note
The fax call will not get compressed using the ip rtp header-compression command
because User Datagram Protocol (UDP) is being used and not Real-Time Transport Protocol
(RTP). For example, a 9600 bps fax call will take approximately 24 kbps.
If the fax rate transmission speed is set higher than the codec rate in the same dial peer, the data sent
over the network for fax transmission will be above the bandwidth reserved for Resource Reservation
Protocol (RSVP).
Tips
Because a large portion of the available network bandwidth will be monopolized by the fax
transmission, Cisco does not recommend setting the fax rate value higher than the value of the
selected codec. If the fax rate value is set lower than the codec value, faxes will take longer to send
but will use less bandwidth.
The voice keyword specifies the highest possible transmission speed allowed by the voice rate. For
example, if the voice codec is G.711, the fax transmission may occur up to 14,400 bps because 14,400
bps is less than the 64-k voice rate. If the voice codec is G.729 (8k), the fax transmission speed will be
7200 bps.
Examples
The following example configures a fax rate transmission speed of 9600 bps for faxes sent using a dial
peer:
dial-peer voice 100 voip
fax rate 9600 voice
The following example sets a fax rate transmission speed at 12,000 bps and the packetization rate at
20 milliseconds:
fax rate 12000 bytes 20
Related Commands
Command
Description
codec (dial-peer)
Specifies the voice coder rate of speech for a dial peer.
fax protocol (dial-peer)
Specifies the fax protocol for a specific VoIP dial peer.
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fax receive called-subscriber
fax receive called-subscriber
To define the called subscriber identifier (CSI), use the fax receive called-subscriber command in
global configuration mode. To disable the configured number, use the no form of this command.
fax receive called-subscriber {$d$ | string}
no fax receive called-subscriber {$d$ | string}
Syntax Description
$d$
Wildcard that specifies that the information displayed is captured from the
configured destination pattern.
string
Destination telephone number. Valid entries are the plus sign (+), numbers 0
through 9, and the space character. This string can specify an E.164 telephone
number; if you choose to configure an E.164 telephone number, use the plus sign
as the first character.
Defaults
Enabled with a null string
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Use the fax receive called-subscriber command to define the number displayed in the liquid crystal
display (LCD) of the sending fax device when you are sending a fax to a recipient. Typically, with a
standard Group 3 fax device, this is the telephone number associated with the receiving fax device. The
command defines the CSI.
This command applies to on-ramp store and forward fax functions.
Examples
The following example configures 555-1234 as the called-subscriber number:
fax receive called-subscriber 5551234
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fax-relay ecm disable
fax-relay ecm disable
To disable fax-relay Error Correction Mode (ECM) on the Voice over IP (VoIP) dial peer, use the
fax-relay ecm disable command in dial-peer configuration mode. To enable ECM, use the no form of
this command.
fax-relay ecm disable
no fax-relay ecm disable
Syntax Description
This command has no arguments or keywords.
Defaults
Fax-relay ECM is enabled.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(3)T
This command was introduced.
Usage Guidelines
When this command is entered, the digital signal processor (DSP) fax-relay firmware disables ECM by
modifying the Digital Information Signal (DIS) T.30 message. This is performed on DIS signals in both
directions so that ECM is disabled in both directions even if only one gateway is configured with ECM
disabled.
This setting is provisioned when the DSP channel starts fax relay and cannot be changed during the fax
relay session.
Examples
The following dial-peer configuration disables ECM on the voice dial peer:
fax-relay ecm disable
The following dial-peer configuration enables ECM on the voice dial peer:
no fax-relay ecm disable
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fax send center-header
fax send center-header
To specify the data that will appear in the center position of the fax header information, use the fax send
center-header command in global configuration mode. To disable the selected options, use the no form
of this command.
fax send center-header {$a$ | $d$ | $p$ | $s$ | $t$ | string}
no fax send center-header {$a$ | $d$ | $p$ | $s$ | $t$ | string}
Syntax Description
$a$
Wildcard that inserts the date in the selected position.
$d$
Wildcard that inserts the destination address in the selected
position.
$p$
Wildcard that inserts the page count in the selected position.
$s$
Wildcard that inserts the sender address in the selected position.
$t$
Wildcard that inserts the transmission time in the selected
position.
string
Text string providing information (in addition to any preset
wildcard values) included in the text header of the fax message.
Valid characters include any text string by itself or a string of
text in combination with command-specific wildcards—for
example, a text string displaying the sender’s company name or
a combinational string like Time:$t$.
Defaults
No center header information is displayed by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal
access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Mail messages that contain only text or that contain text attachments (Multipurpose Internet Mail
Extensions [MIME] media-type text) can be converted by the off-ramp Cisco AS5300 into a format
understood by fax machines using the Cisco AS5300 universal access server text-to-fax converter. When
this conversion is performed, this fax send center-header command is used to indicate what header
information should be added to the center top of pages.
Mail messages with Tag Image File Format (TIFF) attachments (MIME media image and subtype of
TIFF) are expected to include their own per-page headers. Cisco AS5300 software does not modify TIFF
attachments.
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fax send center-header
Note
Because the Cisco AS5300 universal access server does not alter fax TIFF attachments, you cannot
configure faxed header information for faxes being converted from TIFF files to standard fax
transmissions.
This command lets you configure multiple options at once—meaning that you can combine one or more
wildcards with text string information to personalize your fax header information.
Note
If the information selected for the fax send center-header command exceeds the space allocated for
the center fax header, the information is truncated.
This command applies to off-ramp store and forward fax functions.
Examples
The following example selects the transmission time of the fax as the center fax header information:
fax send center-header $t$
The following example configures the company name abc and its address as the center fax header
information:
fax send center-header abc $s$
Related Commands
Command
Description
fax send left-header
Specifies the data that will appear on the left in the fax header
information.
fax send right-header
Specifies the data that will appear on the right in the fax header
information.
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fax send coverpage comment
fax send coverpage comment
To define personalized text for the title field of a fax cover sheet, use the fax send coverpage comment
command in global configuration mode. To disable the defined comment, use the no form of this
command.
fax send coverpage comment string
no fax send coverpage comment string
Syntax Description
string
Defaults
No coverpage comment is displayed by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal
access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Text string that adds personalized text in the title field of the fax
cover sheet. Valid characters are any ASCII characters.
The fax send coverpage comment command can be overridden by the fax send coverpage e-mail
controllable command.
This command applies to off-ramp store and forward fax functions.
Examples
The following example configures an individualized title comment of ABC Fax Services for generated
fax cover sheets:
fax send coverpage enable
fax send coverpage comment ABC Fax Services
Related Commands
Command
Description
fax send coverpage
e-mail-controllable
Defers to the cover page setting in the e-mail header to generate
a standard fax cover sheet.
fax send coverpage enable
Enables the Cisco AS5300 universal access server to generate
fax cover sheets for faxes that originate from e-mail messages.
fax send coverpage show-detail
Prints all of the e-mail header information as part of the fax
cover sheet.
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fax send coverpage e-mail-controllable
fax send coverpage e-mail-controllable
To defer to the cover page setting in the e-mail header to generate a standard fax cover sheet, use the fax
send coverpage e-mail-controllable command in global configuration mode. To disable standard fax
sheet generation, use the no form of this command.
fax send coverpage e-mail-controllable
no fax send coverpage e-mail-controllable
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Also use the destination address of an e-mail message to control the cover page generation on a
per-recipient basis. Use the fax send coverpage e-mail-controllable command to configure the router
to defer to the cover page setting in the e-mail header.
In essence, the off-ramp router defers to the setting configured in the e-mail address itself. For example,
if the address has a parameter set to cover=no, this parameter will override the setting for the fax send
coverpage enable command and the off-ramp gateway will not generate and send a fax cover page. If
the address has a parameter set to cover=yes, the off-ramp gateway will defer to this parameter and
generate and send a fax cover page.
This command applies to off-ramp store and forward fax functions.
Table 18 shows examples of what the user would enter in the To: field of the e-mail message.
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fax send coverpage e-mail-controllable
Table 18
Examples
Example Entries for the To: Field
To: Field Entry
Description
[email protected]
Fax sent to an E.164-compliant long distance
telephone number in the United States. If the fax send
coverpage enable command has been configured,
store and forward fax will generate a fax cover page.
FAX=+1-312-555-3260/[email protected]
Fax sent to an E.164-compliant long distance
telephone number in the United States. In this
example, the fax send coverpage enable command is
superseded by the cover=no statement. No cover page
will be generated.
FAX=+1-312-555-3260/[email protected]
Fax sent to an E.164-compliant long distance
telephone number in the United States. In this
example, the fax send coverpage enable command is
superseded by the cover=yes statement. Store and
forward fax will generate a fax coverpage.
The following example enables standard generated fax cover sheets:
fax send coverpage enable
fax send coverpage e-mail-controllable
Related Commands
Command
Description
fax send coverpage
comment
Defines personalized text for the title field of a fax cover sheet.
fax send coverpage
enable
Enables the Cisco AS5300 universal access server to generate fax cover
sheets for faxes that originate from e-mail messages.
fax send coverpage
show-detail
Prints all of the e-mail header information as part of the fax cover sheet.
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fax send coverpage enable
fax send coverpage enable
To enable the Cisco AS5300 universal access server to generate fax cover sheets for faxes that originate
from e-mail messages, use the fax send coverpage enable command in global configuration mode. To
disable the generation of fax cover sheets, use the no form of this command.
fax send coverpage enable
no fax send coverpage enable
Syntax Description
This command has no arguments or keywords.
Defaults
This feature is disabled by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
This command is applicable only for faxes that originate as e-mail messages. The Cisco AS5300 does
not alter fax Tag Image File Format (TIFF) attachments. Therefore, this command cannot be used to
enable the AS5300 to generate fax cover pages for faxes that are being converted from TIFF files to
standard fax transmissions.
This command applies to off-ramp store and forward fax functions.
Examples
The following example enables the Cisco AS5300 universal access server to generate fax cover sheets:
fax send coverpage enable
Related Commands
Command
Description
fax send coverpage
comment
Defines personalized text for the title field of a fax cover sheet.
fax send coverpage
e-mail-controllable
Defers to the cover page setting in the e-mail header to generate a standard
fax cover sheet.
fax send coverpage
show-detail
Prints all of the e-mail header information as part of the fax cover sheet.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send coverpage show-detail
fax send coverpage show-detail
To print all of the e-mail header information as part of the fax cover sheet, use the fax send coverpage
show-detail command in global configuration mode. To disable the e-mail header information being
displayed, use the no form of this command.
fax send coverpage show-detail
no fax send coverpage show-detail
Syntax Description
This command has no arguments or keywords.
Defaults
No coverpage details are displayed by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
This command applies to off-ramp store and forward fax functions.
This command is applicable only for faxes that originate as e-mail messages. The Cisco AS5300 does
not alter fax Tag Image File Format (TIFF) attachments. Therefore, this command cannot be used to
enable the AS5300 to display additional fax cover page information for faxes that are being converted
from TIFF files to standard fax transmissions.
Examples
The following example configures an individualized generated fax cover sheet that contains the e-mail
header text:
fax send coverpage enable
no fax send coverpage e-mail-controllable
fax send coverpage show-detail
Related Commands
Command
Description
fax send coverpage
comment
Defines personalized text for the title field of a fax cover sheet.
fax send coverpage
e-mail-controllable
Defers to the cover page setting in the e-mail header to generate a standard
fax cover sheet.
fax send coverpage
enable
Enables the Cisco AS5300 universal access server to generate fax cover
sheets for faxes that originate from e-mail messages.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send left-header
fax send left-header
To specify the data that will appear on the left in the fax header information, use the fax send left-header
command in global configuration mode. To disable the selected options, use the no form of this
command.
fax send left-header {$a$ | $d$ | $p$ | $s$ | $t$ | string}
no fax send left-header {$a$ | $d$ | $p$ | $s$ | $t$ | string}
Syntax Description
$a$
Wildcard that inserts the date in the selected position.
$d$
Wildcard that inserts the destination address in the selected
position.
$p$
Wildcard that inserts the page count in the selected position.
$s$
Wildcard that inserts the sender address in the selected position.
$t$
Wildcard that inserts the transmission time in the selected
position.
string
Text string providing information (in addition to any preset
wildcard values) included in the text header of the fax message.
Valid characters include any text string by itself or a string of
text in combination with command-specific wildcards—for
example, a text string displaying the sender’s company name or
a combinational string like Time:$t$.
Defaults
No left header information is displayed by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal
access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Mail messages that contain only text or that contain text attachments (Multipurpose Internet Mail
Extensions [MIME] media text) can be converted by the off-ramp Cisco AS5300 into a format
understood by fax machines using the Cisco AS5300 text-to-fax converter. When this conversion is
performed, this fax send left-header command is used to indicate what header information should be
added to the left top of pages.
Mail messages with Tag Image File Format (TIFF) attachments (MIME media image and subtype of
TIFF) are expected to include their own per-page headers. Cisco AS5300 software does not modify TIFF
attachments.
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send left-header
This command lets you configure multiple options at once—meaning that you can combine one or more
wildcards with text string information to personalize your fax header information.
Note
If the information selected for the fax send left-header command exceeds the space allocated for the
left fax header, the information is truncated.
This command applies to off-ramp store and forward fax functions.
Examples
The following example selects the transmission time of the fax as the left fax header information:
fax send left-header $t$
The following example configures the company name Widget and its address as the left fax header
information:
fax send left-header widget $s$
Related Commands
Command
Description
fax send center-header
Specifies the data that will appear in the center position of the
fax header information.
fax send right-header
Specifies the data that will appear on the right in the fax header
information.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send max-speed
fax send max-speed
To specify the maximum speed at which an outbound fax will be sent, use the fax send max-speed
command in global configuration mode. To disable the selected speed, use the no form of this command.
fax send max-speed {2400 | 4800 | 7200 | 9600 | 12000 | 14400}
no fax send max-speed {2400 | 4800 | 7200 | 9600 | 12000 | 14400}
Syntax Description
2400
Indicates a transmission speed of 2400 bits per second (bps).
4800
Indicates a transmission speed of 4800 bps.
7200
Indicates a transmission speed of 7200 bps.
9600
Indicates a transmission speed of 9600 bps.
12000
Indicates a transmission speed of 12,000 bps.
14400
Indicates a transmission speed of 14,400 bps.
Defaults
14400 bps
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal
access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
This command applies to off-ramp store and forward fax functions.
Examples
The following example sets the outbound fax transmission rate at 2400 bps:
fax send max-speed 2400
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send right-header
fax send right-header
To specify the data that will appear on the right in the fax header information, use the fax send
right-header command in global configuration mode. To disable the selected options, use the no form
of this command.
fax send right-header {$a$ | $d$ | $p$ | $s$ | $t$ | string}
no fax send right-header {$a$ | $d$ | $p$ | $s$ | $t$ | string}
Syntax Description
$a$
Wildcard that inserts the date in the selected position.
$d$
Wildcard that inserts the destination address in the selected
position.
$p$
Wildcard that inserts the page count in the selected position.
$s$
Wildcard that inserts the sender address in the selected position.
$t$
Wildcard that inserts the transmission time in the selected
position.
string
Text string providing information (in addition to any preset
wildcard values) included in the text header of the fax message.
Valid characters include any text string by itself or a string of
text in combination with command-specific wildcards—for
example, a text string displaying the sender’s company name or
a combinational string like Time:$t$.
Defaults
No right header information is displayed by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal
access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
Mail messages that contain only text or contain text attachments (Multipurpose Internet Mail Extensions
[MIME] media text) can be converted by the off-ramp Cisco AS5300 into a format understood by fax
machines using the Cisco AS5300 text-to-fax converter. When this conversion is performed, this fax
send right-header command is used to indicate what header information should be added to right top
of pages.
Mail messages with Tag Image File Format (TIFF) attachments (MIME media image and subtype of
TIFF) are expected to include their own per-page headers. Cisco AS5300 software does not modify TIFF
attachments.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send right-header
This command lets you configure multiple options at once—meaning that you can combine one or more
wildcards with text string information to personalize your fax header information.
Note
If the information selected for the fax send right-header command exceeds the space allocated for
the right fax header, the information is truncated.
This command applies to off-ramp store and forward fax functions.
Examples
The following example selects the date of the fax as the right fax header information:
fax send right-header $a$
The following example configures the company name Widget and its address as the right fax header
information:
fax send right-header widget $s$
Related Commands
Command
Description
fax send center-header
Specifies the data that will appear in the center position of the
fax header information.
fax send left-header
Specifies the data that will appear on the left in the fax header
information.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
fax send transmitting-subscriber
fax send transmitting-subscriber
To define the transmitting subscriber identifier (TSI), use the fax send transmitting-subscriber
command in global configuration mode. To disable the configured value, use the no form of this
command.
fax send transmitting-subscriber {$s$ | string}
no fax send transmitting-subscriber {$s$ | string}
Syntax Description
$s$
Wildcard that inserts the sender name from the RFC 822 header
(captured by the on-ramp from the sending fax machine) in the
selected position.
string
Originating telephone number. Valid entries are the plus
sign (+), numbers 0 through 9, and the space character. This
string can specify an E.164 telephone number; if you choose to
configure an E.164 telephone number, use the plus sign as the
first character.
Defaults
No TSI information is displayed by default.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal
access server.
12.1(1)
This command was integrated into Cisco IOS Release 12.1(1).
Usage Guidelines
The transmitting subscriber number is the number displayed in the liquid crystal display (LCD) of the
receiving fax device. Typically, with a standard Group 3 fax device, this is the telephone number
associated with the transmitting or sending fax device. This command defines the TSI.
This command applies to off-ramp store and forward fax functions.
Examples
The following example configures the company name (as captured by the on-ramp from the sending fax
machine as +18005551234) as the TSI:
fax send transmitting-subscriber +18005551234
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
forward-alarms
forward-alarms
To turn on alarm forwarding so that alarms arriving on one T1/E1 port are sent to the other port on
dual-mode multiflex trunk interface cards, use the forward-alarms command in controller
configuration mode on the one port. To restore the default value so that no alarms are forwarded, use the
no form of this command.
forward-alarms
no forward-alarms
Syntax Description
This command has no arguments or keywords.
Defaults
Alarm forwarding is disabled.
Command Modes
Controller configuration
Command History
Release
Modification
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS
Release 12.1(1)T.
Usage Guidelines
When you enter this command, physical-layer alarms on the configured port are forwarded to the other
port on dual-port cards, simulating a one-way repeater operation. The system forwards RAIs (remote
alarm indications, Yellow Alarms), alarm indication signals (AIS, or Blue Alarms), losses of frame
(LOF alarms or Red Alarms), and losses of signaling (LOS alarms or Red Alarms).
Examples
The following example shows how to turn on alarm forwarding on controller E1 0/0 of a Cisco 2600
series router:
controller e1 0/0
forward-alarms
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
forward-digits
forward-digits
To specify which digits to forward for voice calls, use the forward-digits command in dial-peer
configuration mode. To specify that any digits not matching the destination-pattern are not to be
forwarded, use the no form of this command. To restore the default state, use the default form of this
command.
forward-digits {num-digit | all | extra}
no forward-digits
default forward-digits
Syntax Description
num-digit
The number of digits to be forwarded. If the number of digits is
greater than the length of a destination phone number, the
length of the destination number is used. The valid range is
from 0 to 32. Setting the value to 0 is equivalent to entering the
no forward-digits command.
all
Forwards all digits. If all is entered, the full length of the
destination pattern is used.
extra
If the length of the dialed digit string is greater than the length
of the dial-peer destination pattern, the extra right-justified
digits are forwarded. However, if the dial-peer destination
pattern is variable length ending with the character “T” (for
example: T, 123T, 123...T), extra digits are not forwarded.
Defaults
Dialed digits not matching the destination pattern are forwarded.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810
multiservice concentrator.
12.0(2)T
The implicit option was added.
12.0(4)T
This command was modified to support ISDNBF PRI QSIG
signaling calls.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600
series routers, the implicit keyword was removed, and the
extra keyword was added.
12.1(2)T
The modifications made in Cisco IOS Release 12.0(7)XK were
integrated into Cisco IOS Release 12.1(2)T.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
forward-digits
Usage Guidelines
This command applies only to plain old telephone service (POTS) dial peers. Forwarded digits are
always right justified so that extra leading digits are stripped. The destination pattern includes both
explicit digits and wildcards if present. Use the default form of this command if a nondefault
digit-forwarding scheme was entered previously and you wish to restore the default.
For QSIG ISDN connections, entering the forward-digits all command implies that all the digits of the
called party number are sent to the ISDN connection. When the forward-digits num-digit command and
a number from 1 to 32 are entered, the number of digits of the called party number specified (right
justified) are sent to the ISDN connection.
Examples
The following example shows that all digits in the destination pattern of a POTS dial peer are forwarded:
dial-peer voice 1 pots
destination-pattern 8...
forward-digits all
The following example shows that four of the digits in the destination pattern of a POTS dial peer are
forwarded:
dial-peer voice 1 pots
destination-pattern 555....
forward-digits 4
The following example shows that the extra right-justified digits that exceed the length of the destination
pattern of a POTS dial peer are forwarded:
dial-peer voice 1 pots
destination-pattern 555....
forward-digits extra
Related Commands
Command
Description
destination-pattern
Defines the prefix or the full E.164 telephone number to be used
for a dial peer.
show dial-peer voice
Displays configuration information for dial peers.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
frame-relay voice bandwidth
frame-relay voice bandwidth
To specify how much bandwidth should be reserved for voice traffic on a specific data-link connection
identifier (DLCI), use the frame-relay voice bandwidth command in map-class configuration mode. To
release the bandwidth previously reserved for voice traffic, use the no form of this command.
frame-relay voice bandwidth bps-reserved
no frame-relay voice bandwidth bps-reserved
The bandwidth, in bits per second (bps), reserved for voice traffic
for the specified map class. The range is from 8000 to
45,000,000 bps; the default is 0, which disables voice calls.
Syntax Description
bps-reserved
Defaults
Disabled (zero)
Command Modes
Map-class configuration
Command History
Release
Modification
12.0(3)XG
This command was introduced on the Cisco 2600 series routers,
3600 series, 7200 series, and on the MC3810 multiservice
concentrator.
12.0(4)T
This command was integrated into Cisco IOS Release 12.0(4)T.
Usage Guidelines
12.0(5)T
The queue depth keyword and argument option was added.
12.2(1)
The queue depth keyword and argument option was deleted.
To use this command, you must first associate a Frame Relay map class with a specific DLCI and then
enter map-class configuration mode and set the amount of bandwidth to be reserved for voice traffic for
that map class.
If a call is attempted and there is not enough remaining bandwidth reserved for voice to handle the
additional call, the call will be rejected. For example, if 64 kbps is reserved for voice traffic and a codec
and payload size is being used that requires 10 kbps of bandwidth for each call, the first six calls
attempted will be accepted, but the seventh call will be rejected.
Reserve queues are not required for Voice over Frame Relay (VoFR).
Note
Cisco strongly recommends that you set voice bandwidth to a value less than the committed
information rate (CIR) if Frame Relay traffic shaping is configured. Cisco also strongly
recommends that you set the minimum CIR (using the frame-relay mincir command) to be
at least equal to or greater than the voice bandwidth.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
frame-relay voice bandwidth
Calculating Required Bandwidth
The bandwidth required for a voice call depends on the bandwidth of the codec, the voice packetization
overhead, and the voice frame payload size. The smaller the voice frame payload size, the higher the
bandwidth required for the call. To make the calculation, use the following formula:
required_bandwidth = codec_bandwidth x (1 + overhead / payload_size)
As an example, the overhead for a VoFR voice packet is between 6 and 8 bytes: a 2-byte Frame Relay
header, a 1- or 2-byte FRF.11 header (depending on the CID value), a 2-byte cyclic redundancy check
(CRC), and a 1-byte trailing flag. If voice sequence numbers are enabled in the voice packets, there is
an additional 1-byte sequence number. Table 19 shows the required voice bandwidth for the G.729
8000-bps speech coder for various payload sizes.
Table 19
Required Voice Bandwidth Calculations for G.729
Codec
Codec
Bandwidth
Voice Frame
Payload Size
Required Bandwidth per
Call (6-Byte OH)
Required Bandwidth per
Call (8-Byte OH)
G.729
8000 bps
120 bytes
8400 bps
8534 bps
G.729
8000 bps
80 bytes
8600 bps
8800 bps
G.729
8000 bps
40 bytes
9200 bps
9600 bps
G.729
8000 bps
30 bytes
9600 bps
10134 bps
G.729
8000 bps
20 bytes
10400 bps
11200 bps
To configure the payload size for the voice frames, use the codec command from dial-peer configuration
mode.
Examples
The following example shows how to reserve 64 kbps for voice traffic for the “vofr” Frame Relay map
class on a Cisco 2600 series routers, 3600 series, or 7200 series router or on an MC3810 multiservice
concentrator:
interface serial 1/1
frame-relay interface-dlci 100
class vofr
exit
map-class frame-relay vofr
frame-relay voice bandwidth 64000
Related Commands
Command
Description
codec (dial-peer)
Specifies the voice coder rate of speech for a VoFR dial peer.
frame-relay fair-queue
Enables weighted fair queueing for one or more Frame Relay
PVCs.
frame-relay fragment
Enables fragmentation for a Frame Relay map class.
frame-relay interface-dlci
Assigns a DLCI to a specified Frame Relay subinterface on the
router or access server.
frame-relay mincir
Assigns the minimum CIR for Frame Relay traffic shaping.
map-class frame-relay
Specifies a map class to define QoS values for an SVC.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
frag-pre-queuing
frag-pre-queuing
This command was removed in Cisco IOS Release 12.1(2)T and is no longer supported in this release.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-max-delay
freq-max-delay
This command is not supported in Cisco IOS Release 12.2. This command was added in Cisco IOS
Release 12.2(2)T. For information about this command, refer to the Cisco IOS Voice, Video, and Fax
Command Reference, Release 12.2 T, at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122tcr/122tvr/index.htm
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-max-deviation
freq-max-deviation
This command is not supported in Cisco IOS Release 12.2. This command was added in Cisco IOS
Release 12.2(2)T. For information about this command, refer to the Cisco IOS Voice, Video, and Fax
Command Reference, Release 12.2 T, at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122tcr/122tvr/index.htm
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-max-power
freq-max-power
This command is not supported in Cisco IOS Release 12.2. This command was added in Cisco IOS
Release 12.2(2)T. For information about this command, refer to the Cisco IOS Voice, Video, and Fax
Command Reference, Release 12.2 T, at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122tcr/122tvr/index.htm
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-min-power
freq-min-power
This command is not supported in Cisco IOS Release 12.2. This command was added in Cisco IOS
Release 12.2(2)T. For information about this command, refer to the Cisco IOS Voice, Video, and Fax
Command Reference, Release 12.2 T, at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122tcr/122tvr/index.htm
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-pair
freq-pair
To specify the frequency components of a tone to be detected, use the freq-pair command in voice-class
configuration mode. To cancel detection of a tone, use the no form of this command.
freq-pair tone-id frequency-1 frequency-2
no freq-pair tone-id
Syntax Description
tone-id
A tag identifier for a tone to be detected. The range is from 1 to 16. There is no
default.
frequency-1
One frequency component of the tone to be detected, in Hz. The range is from
300 to 3600. There is no default.
frequency-2
A second frequency component of the tone to be detected, in Hz. The range is
from 300 to 3600, or you can specify 0. There is no default.
Defaults
No tone is specified for detection.
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and 3600 series routers and on
the MC3810 multiservice concentrator.
Usage Guidelines
To detect a tone with two frequency components (a dualtone), configure frequencies for frequency-1 and
frequency-2.
To detect a tone with only one frequency component, configure a frequency for frequency-1 and enter 0
for frequency-2.
You can configure a router to detect up to 16 tones.
Examples
The following example configures tone number 1 (tone-id 1) with frequency components of 480 Hz and
2400 Hz:
voice class dualtone 100
freq-pair 1 480 2400
exit
The following example configures tone number 1 (tone-id 1) with frequency components of 480 Hz and
2400 Hz and tone number 2 (tone-id 2) with frequency components of 560 Hz and 880 Hz:
voice class dualtone 50
freq-pair 1 480 2400
freq-pair 2 560 880
exit
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-pair
Related Commands
Command
Description
voice class dualtone
Creates a voice class for FXO tone detection parameters.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
freq-power-twist
freq-power-twist
This command is not supported in Cisco IOS Release 12.2. This command was added in Cisco IOS
Release 12.2(2)T. For information about this command, refer to the Cisco IOS Voice, Video, and Fax
Command Reference, Release 12.2 T, at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122tcr/122tvr/index.htm
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ftc-trunk frame-relay-dlci
ftc-trunk frame-relay-dlci
This command was removed in Cisco IOS Release 12.1(2)T and is no longer supported in this release.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ftc-trunk management-dlci
ftc-trunk management-dlci
This command was removed in Cisco IOS Release 12.1(2)T and is no longer supported in this release.
Cisco IOS Voice, Video, and Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: D Through F
ftc-trunk management-protocol
ftc-trunk management-protocol
This command was removed in Cisco IOS Release 12.1(2)T and is no longer supported in this release.
Cisco IOS Voice, Video, and Fax Command Reference
VR-338
Cisco IOS Voice, Video, and Fax Commands:
G Through P
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax
applications. The commands are presented in alphabetical order beginning with G. Some commands
required for configuring voice, video, and fax may be found in other Cisco IOS command references.
Use the command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS
Voice, Video, and Fax Configuration Guide.
Cisco IOS Voice, Video, and Fax Command Reference
VR-339
Cisco IOS Voice, Video, and Fax Commands: G Through P
gatekeeper
gatekeeper
To enter gatekeeper configuration mode, use the gatekeeper command in global configuration mode.
gatekeeper
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series routers.
12.0(3)T
This command was integrated into the Cisco IOS Release 12.0(3)T and,
support for the Cisco MC3810 multiservice concentrator was added.
Usage Guidelines
Press Ctrl-Z or use the exit command to exit gatekeeper configuration mode.
Examples
The following example brings the gatekeeper online:
gatekeeper
no shutdown
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gateway
gateway
To enable the H.323 Voice over IP (VoIP) gateway, use the gateway command in global configuration
mode. To disable the gateway, use the no form of this command.
gateway
no gateway
Syntax Description
This command has no arguments or keywords.
Defaults
The gateway is unregistered.
Command Modes
Global configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 3600 series routers and
Cisco AS5300 and Cisco AS5800 universal access servers.
Usage Guidelines
Use the gateway command to enable H.323 VoIP gateway functionality. After you enable the gateway,
it will attempt to discover a gatekeeper by using the H.323 RAS GRQ message. If you enter no
gateway voip, the VoIP gateway will unregister with the gatekeeper via the H.323 RAS URQ message.
Examples
The following example enables the gateway:
gateway
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group
group
To create a session-group and associate it with a specified session-set, use the group command in
backhaul session manager configuration mode. To delete the group, use the no form of this command.
group group-name set set-name
no group group-name set set-name
Syntax Description
group-name
Session-group name.
set set-name
Session-set name.
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To associate the group named Group5 with the set named Set1, see the following example:
Router(config-bsm)# group group5 set set1
Related Commands
Command
Description
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments that are
received before an EACK is sent.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer retransmit
Configures retransmission timeout.
group timer transfer
Configures state transfer timeout.
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group auto-reset
group auto-reset
To configure the maximum auto-reset value, use the group auto-reset command in backhaul session
manager configuration mode. To set the value to the default value, use the no form of this command.
group group-name auto-reset count
no group group-name auto-reset count
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
auto-reset count
Maximum number of auto-resets. Range is 0 through 255.
Defaults
5
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To configure the maximum auto-reset value for the group named Group5 to 6, see the following example:
Router(config-bsm)# group group5 auto-reset 6
Related Commands
Command
Description
group cumulative-ack
Configures maximum cumulative acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments that are
received before an EACK is sent.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
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group cumulative-ack
group cumulative-ack
To configure maximum cumulative acknowledgments, use the group cumulative-ack command in
backhaul session manager configuration mode. Maximum cumulative acknowledgments are the
maximum number of segments that are received before an acknowledgment is sent. To set the value to
the default value, use the no form of this command.
group group-name cumulative ack count
no group group-name cumulative ack count
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
cumulative ack count
Maximum number of segments received before acknowledgment.
Range is 0 through 255.
Defaults
3
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To set the cumulative acknowledgment maximum for Group5 to 4, see the following example:
Router(config-bsm)# group group5 cumulative-ack 4
Related Commands
Command
Description
group auto-reset
Configures the maximum auto-reset value.
group out-of-sequence
Configures maximum out-of-sequence segments that are
received before an EACK is sent.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
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group out-of-sequence
group out-of-sequence
To configure maximum out-of-sequence segments that are received before an EACK is sent, use the
group out-of-sequence command in backhaul session manager configuration mode. To set the value to
the default value, use the no form of this command.
group group-name out-of-sequence count
no group group-name out-of-sequence count
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
out-of-sequence count
Maximum number of out-of-sequence segments. Range is 0
through 255.
Defaults
3
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To set the out-of-sequence maximum for Group5 to 4, see the following example:
Router(config-bsm)# group group5 out-of-sequence 4
Related Commands
Command
Description
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative acknowledgments.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
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group receive
group receive
To configure maximum receive segments, use the group receive command in backhaul session manager
configuration mode. To set the value to the default value, use the no form of this command.
group group-name receive count
no group group-name receive count
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
receive count
Maximum number of segments in our receive window. The
other side should send no more than this number of segments
before receiving an acknowledgment for the oldest
outstanding segment. Range is 1 through 64.
Defaults
32
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To set the receive maximum to 10 for Group5, see the following example:
Router(config-bsm)# group group5 receive 10
Related Commands
Command
Description
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments that are
received before an EACK is sent.
group retransmit
Configures maximum retransmits.
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group retransmit
group retransmit
To configure maximum retransmits, use the group retransmit command in backhaul session manager
configuration mode. To set the value to the default value, use the no form of this command.
group group-name retransmit count
no group group-name retransmit count
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
retransmit count
Maximum number of retransmits. Range is 0
through 255.
Defaults
2
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To set the retransmit maximum for Group5 to 3, see the following example:
Router(config-bsm)# group group5 retrans 3
Related Commands
Command
Description
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments that are
received before an EACK is sent.
group receive
Configures maximum receive segments.
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group timer cumulative-ack
group timer cumulative-ack
To configure cumulative acknowledgment timeout, use the group timer cumulative ack command in
backhaul session manager configuration mode. Cumulative acknowledgment timeout is the maximum
number of milliseconds RUDP will delay before sending an acknowledgment for a received segment. To
set the value to the default value, use the no form of this command.
group group-name timer cumulative ack time
no group group-name timer cumulative ack time
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
timer cumulative ack time
Number of milliseconds RUDP will delay. Range is 100
through 65535.
Defaults
100
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To set the cumulative acknowledgment timer for Group5 to 325, see the following example:
Router(config-bsm)# group group5 timer cumulative-ack 325
Related Commands
Command
Description
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer retransmit
Configures retransmission timeout.
group timer transfer
Configures state transfer timeout.
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group timer keepalive
group timer keepalive
To configure keepalive (or null segment) timeout, use the group timer keepalive command in backhaul
session manager configuration mode. Keepalive timeout is the number of milliseconds RUDP will wait
before sending a keepalive segment. To set the value to the default value, use the no form of this
command.
group group-name timer keepalive time
no group group-name timer keepalive time
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
timer keepalive time
Number of milliseconds before RUDP sends a keepalive
segment. Range is 100 through 65535.
Defaults
1000
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To configure the keepalive timer for Group5 to 2050 milliseconds, see the following example:
Router(config-bsm)# group group5 timer keepalive 2050
Related Commands
Command
Description
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer retransmit
Configures retransmission timeout.
group timer transfer
Configures state transfer timeout.
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group timer retransmit
group timer retransmit
To configure retransmission timeout, use the group timer retransmit command in backhaul session
manager configuration mode. Retransmission timeout is the number of milliseconds RUDP will wait to
receive an acknowledgment for a segment. To set the value to the default value, use the no form of this
command.
group group-name timer retransmit time
no group group-name timer retransmit time
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
timer retransmit time
Number of milliseconds RUDP will delay. Range is 100
through 65535.
Defaults
300
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Usage Guidelines
The retransmit timer must be greater than the cumulative-ack timer.
Examples
To set the retransmit timer for Group5 to 650, see the following example:
Router(config-bsm)# group group5 timer retransmit 650
Related Commands
Command
Description
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer transfer
Configures state transfer timeout.
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group timer transfer
group timer transfer
To configure state transfer timeout, use the group timer transfer command in backhaul session manager
configuration mode. To set the value to the default value, use the no form of this command.
group group-name timer transfer time
no group group-name timer transfer time
Caution
Syntax Description
Do not change this parameter unless instructed to do so by Cisco technical support. There are
relationships between group parameters that can cause sessions to fail if not set correctly.
group-name
Session-group name.
timer transfer time
Maximum number of milliseconds RUDP will wait for a
transfer request. The range is 0 to 65535 milliseconds.
Defaults
2000
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To set the state transfer timer for Group5 to 1800, see the following example:
Router(config-bsm)# group group5 timer transfer-state 1800
Related Commands
Command
Description
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer retransmit
Configures retransmission timeout.
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gw-accounting
gw-accounting
To enable Voice over IP (VoIP) gateway-specific accounting and define the accounting method, use the
gw-accounting command in global configuration mode. To disable gateway-specific accounting, use the
no form of this command.
gw-accounting {h323 [vsa] | syslog | voip}
no gw-accounting {h323 [vsa] | syslog | voip}
Syntax Description
h323
Enables standard H.323 accounting using Internet Engineering Task Force
(IETF) RADIUS attributes.
vsa
(Optional) Enables H.323 accounting using RADIUS vendor specific
attributes (VSAs).
syslog
Enables the system logging facility to output accounting information in the
form of a system log message.
voip
Enables generic gateway-specific accounting.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced for the Cisco 2500 and 3600 series routers
and the AS5300 universal access server.
12.0(7)T
The vsa keyword was added.
12.1(1)T
The voip keyword was added.
Usage Guidelines
To collect basic start-stop connection accounting data, the gateway must be configured to support
gateway-specific H.323 accounting functionality. The gw-accounting command enables you to send
accounting data to the RADIUS server in one of four ways:
•
Using standard IETF RADIUS accounting attribute/value (AV) pairs—This method is the basic
method of gathering accounting data (connection accounting) according to the specifications
defined by the IETF. Use the gw-accounting h323 command to configure the standard IETF
RADIUS method of applying H.323 gateway-specific accounting. Table 20 shows the IETF
RADIUS attributes supported.
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Table 20
Supported IETF RADIUS Accounting Attributes
Number
Attribute
Description
30
Called-Station-Id
Allows the network access server to send the telephone number
that the user called as part of the Access-Request packet (using
Dialed Number Identification [DNIS] or similar technology).
This attribute is supported only on ISDN, and modem calls on
the Cisco AS5200 and Cisco AS5300 universal access server if
used with ISDN PRI.
31
Calling-Station-Id
Allows the network access server to send the telephone number
that the call came from as part of the Access-Request packet
(using Automatic Number Identification or similar
technology). This attribute has the same value as “remote-addr”
from TACACS+. This attribute is supported only on ISDN, and
modem calls on the Cisco AS5200 and Cisco AS5300 universal
access server if used with ISDN PRI.
42
Acct-Input-Octets
Indicates how many octets have been received from the port
over the course of the accounting service being provided.
43
Acct-Output-Octets
Indicates how many octets have been sent to the port over the
course of delivering the accounting service.
44
Acct-Session-Id
Indicates a unique accounting identifier that makes it easy to
match start and stop records in a log file. Acct-Session-Id
numbers restart at 1 each time the router is power-cycled or the
software is reloaded.
47
Acct-Input-Packets
Indicates how many packets have been received from the port
over the course of this service being provided to a framed user.
48
Acct-Output-Packets
Indicates how many packets have been sent to the port in the
course of delivering this service to a framed user.
For more information about RADIUS and the use of IETF-defined attributes, see the
Cisco IOS Security Configuration Guide.
•
Overloading the Acct-Session-Id field—Attributes that cannot be mapped to standard RADIUS are
packed into the Acct-Session-Id attribute field as ASCII strings separated by the character “/”. The
Acct-Session-Id attribute is defined to contain the RADIUS account session ID, which is a unique
identifier that links accounting records associated with the same login session for a user. To support
additional fields, we have defined the following string format for this field:
<session id>/<call leg setup time>/<gateway id>/<connection id>/<call origin>/
<call type>/<connect time>/<disconnect time>/<disconnect cause>/<remote ip address>
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Table 21 shows the field attributes that you use with the overloaded session-ID method and a brief
description of each.
Table 21
Field Attributes in Overloaded Acct-Session ID
Field Attribute
Description
Session-Id
Specifies the standard RADIUS account session ID.
Setup-Time
Provides the Q.931 setup time for this connection in Network Time
Protocol (NTP) format. NTP time formats are displayed as%H: %M: %S
%k %Z %tw %tn %td %Y where:
%H is hour (00 to 23).
%M is minutes (00 to 59).
%S is seconds (00 to 59).
%k is milliseconds (000 to 999).
%Z is timezone string.
%tw is day of week (Saturday through Sunday).
%tn is month name (January through December).
%td is day of month (01 to 31).
%Y is year including century (for example, 1998).
Gateway-Id
Indicates the name of the underlying gateway in the form
“gateway.domain_name.”
Call-Origin
Indicates the origin of the call relative to the gateway. Possible values
are originate and answer.
Call-Type
Indicates the call leg type. Possible values are telephony and VoIP.
Connection-Id
Specifies the unique global identifier used to correlate call legs that
belong to the same end-to-end call. The field consists of 4 long words
(128 bits). Each long word is displayed as a hexadecimal value and is
separated by a space character.
Connect-Time
Provides the Q.931 connect time for this call leg, in NTP format.
Disconnect-Time
Provides the Q.931 disconnect time for this call leg, in NTP format.
Disconnect-Cause
Specifies the reason a call was taken offline as defined in the Q.931
specification.
Remote-Ip-Address
Indicates the address of the remote gateway port where the call is
connected.
Because of the limited size of the Acct-Session-Id string, it is not possible to embed very many
information elements in it. Therefore, this feature supports only a limited set of accounting
information elements.
Use the gw-accounting h323 command to configure the overloaded session ID method of applying
H.323 gateway-specific accounting.
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•
Using vendor-specific RADIUS attributes—The IETF draft standard specifies a method for
communicating vendor-specific information between the network access server and the RADIUS
server by using the vendor-specific attribute (Attribute 26). Vendor-specific attributes (VSAs) allow
vendors to support their own extended attributes not suitable for general use. The Cisco RADIUS
implementation supports one vendor-specific option using the format recommended in the
specification. The Cisco vendor-ID is 9, and the supported option has vendor-type 1, which is named
“cisco-avpair.” The value is a string of the format:
protocol: attribute sep value *
“Protocol” is a value of the Cisco “protocol” attribute for a particular type of authorization.
“Attribute” and “value” are an appropriate attribute/value (AV) pair defined in the Cisco TACACS+
specification, and “sep” is “=” for mandatory attributes and “*” for optional attributes. This allows
the full set of features available for TACACS+ authorization to also be used for RADIUS.
The VSA fields and their ASCII values are listed in Table 22.
Table 22
VSA Fields and Their ASCII Values
IETF
RADIUS
Attribute
VendorSpecific
Company
Code
Subtype
Number
Attribute Name
Description
26
9
23
h323-remote-address
Indicates the IP address of the remote
gateway.
26
9
24
h323-conf-id
Identifies the conference ID.
26
9
25
h323-setup-time
Indicates the setup time for this
connection in Coordinated Universal
Time (UTC) formerly known as
Greenwich Mean Time (GMT) and
Zulu time.
26
9
26
h323-call-origin
Indicates the origin of the call
relative to the gateway. Possible
values are originating and
terminating (answer).
26
9
27
h323-call-type
Indicates the call leg type. Possible
values are telephony and VoIP.
26
9
28
h323-connect-time
Indicates the connection time for this
call leg in UTC.
26
9
29
h323-disconnect-time
Indicates the time this call leg was
disconnected in UTC.
26
9
30
h323-disconnect-cause
Specifies the reason a connection was
taken offline per the Q.931
specification.
26
9
31
h323-voice-quality
Specifies the impairment factor
(ICPIF) affecting voice quality for a
call.
26
9
33
h323-gw-id
Indicates the name of the underlying
gateway.
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gw-accounting
Use the gw-accounting h323 vsa command to configure the VSA method of applying H.323
gateway-specific accounting.
•
Using syslog records—The syslog accounting option exports the information elements associated
with each call leg through a system log message, which can be captured by a syslog daemon on the
network. The syslog output consists of the following:
<server timestamp> <gateway id> <message number> : <message label> : <list of AV
pairs>
The syslog messages fields are listed in Table 23.
Table 23
Syslog Mesage Output Fields
Field
Description
server timestamp
The time stamp created by the server when it receives the message
to log.
gateway id
The name of the gateway that emits the message.
message number
The number assigned to the message by the gateway.
message label
A string used to identify the message category.
list of AV pairs
A string that consists of <attribute name> <attribute value> pairs
separated by commas.
Use the gw-accounting syslog command to configure the syslog record method of gathering H.323
accounting data.
Examples
The following example configures basic H.323 accounting using IETF RADIUS attributes:
gw-accounting h323
The following example configures H.323 accounting using VSA RADIUS attributes:
gw-accounting h323 vsa
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gw-type-prefix
gw-type-prefix
To configure a technology prefix in the gatekeeper, use the gw-type-prefix command in gatekeeper
configuration mode. To remove the technology prefix, use the no form of this command.
gw-type-prefix type-prefix [[hopoff gkid1] [hopoff gkid2] [hopoff gkidn] [seq | blast]]
[default-technology] [[gw ipaddr ipaddr [port]]]
no gw-type-prefix type-prefix [[hopoff gkid1] [hopoff gkid2] [hopoff gkidn] [seq | blast]]
[default-technology] [[gw ipaddr ipaddr [ port]]]
Syntax Description
type-prefix
A technology prefix is recognized and is stripped before checking for
the zone prefix. It is strongly recommended that you select
technology prefixes that do not lead to ambiguity with zone prefixes.
Do this by using the # character to terminate technology prefixes, for
example, 3#.
hopoff gkid
(Optional) Use this option to specify the gatekeeper where the call is
to hop off, regardless of the zone prefix in the destination address.
The gkid argument refers to a gatekeeper previously configured using
the zone local or zone remote comment. You can enter this keyword
and argument multiple times to configure redundant gatekeepers for
a given technology prefix.
seq | blast
(Optional) If you list multiple hopoffs, this indicates that the LRQs
should be sent sequentially or simultaneously (blast) to the
gatekeepers according to the order in which they were listed. The
default is to send them sequentially.
default-technology
(Optional) Gateways registering with this prefix option are used as
the default for routing any addresses that are otherwise unresolved.
gw ipaddr ipaddr [port]
(Optional) Use this option to indicate that the gateway is incapable of
registering technology prefixes. When it registers, it adds the
gateway to the group for this type prefix, just as if it had sent the
technology prefix in its registration. This parameter can be repeated
to associate more than one gateway with a technology prefix.
Defaults
By default, no technology prefix is defined, and LRQs are sent sequentially to all the gatekeepers listed.
Command Modes
Gatekeeper configuration
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gw-type-prefix
Command History
Usage Guidelines
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 and 3600 series
routers and the AS5300 universal access server.
12.1(1)T
This command was modified to allow the user to specify multiple
hopoffs.
12.1(2)T
This command was modified to allow the user to specify whether
LRQs should be sent simultaneously or sequentially to the
gatekeepers.
More than one gateway can register with the same technology prefix. In such cases, a random selection
is made of one of them.
You do not have to define a technology prefix to a gatekeeper if there are gateways configured to register
with that prefix and if there are no special flags (hopoff gkid or default-technology) that you want to
associate with that prefix.
You need to configure the gateway type prefix of all remote technology prefixes that will be routed
through this gatekeeper.
Examples
The following example defines two gatekeepers for technology zone 3:
gw-type-prefix 3#* hopoff c2600-1-gk hopoff c2514-1-gk
Related Commands
Command
Description
show gatekeeper gw-type-prefix Displays the list of currently defined technology zones and the
gatekeepers responsible for each.
zone prefix
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Configures the gatekeeper with knowledge of its own prefix and
the prefix of any remote zone.
Cisco IOS Voice, Video, and Fax Commands: G Through P
h225 timeout tcp establish
h225 timeout tcp establish
To set the H.225 TCP timeout value for Voice over IP (VoIP) dial peers, use the h225 timeout tcp
establish command in voice class configuration mode. To set the timeout value to its default, use the no
form of this command.
h225 timeout tcp establish seconds
no h225 timeout tcp establish
Syntax Description
seconds
Defaults
The default timeout value is 15 seconds.
Command Modes
Voice class configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 1700, 2500, 2600, 3600,
7200 series routers, AS5300 universal access server, uBR900 series,
and uBR924.
Examples
Specifies the number of seconds for the timeout. Possible values are
0 to 30. The default is 15. If you specify 0, the H.225 TCP timer is
disabled.
The following example configures a timeout of 10 seconds, which is associated with the H.323 voice
class labeled 1:
voice class h323 1
h225 timeout tcp establish 10
Related Commands
Command
Description
voice class h323
Establishes an H.323 voice class.
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h323 asr
h323 asr
To enable application-specific routing (ASR) and specify the maximum bandwidth for a proxy, use the
h323 asr command in interface configuration mode. To remove a bandwidth setting but keep ASR
enabled, use no form of this command.
h323 asr [bandwidth max-bandwidth]
no h323 asr [bandwidth max-bandwidth]
Syntax Description
bandwidth
max-bandwidth
Defaults
ASR is disabled.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series routers.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
(Optional) Maximum bandwidth on the interface. The value ranges are from
1 to 10,000,000 kbps. If you do not specify the max-bandwidth, the value
defaults to the bandwidth on the interface. If you specify max-bandwidth as
a value greater than the interface bandwidth, the bandwidth defaults to the
interface bandwidth.
This command is independent of the h323 interface command.
This command is not supported on Frame Relay or ATM interfaces for the Cisco MC3810 multiservice
concentrator.
Note
Examples
Specifying the no h323 asr bandwidth max-bandwidth command removes the bandwidth setting but
leaves ASR enabled. You must enter the no h323 asr command to disable ASR.
The following example enables ASR and specifies a maximum bandwidth of 10,000 kbps:
h323 asr bandwidth 10000
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h323 call start
h323 call start
To force the H.323 Version 2 gateway to use Fast Connect or Slow Connect procedures for all H.323
calls, use the h323 call start command in voice-service configuration mode. To restore the default
condition, use the no form of this command.
h323 call start {fast | slow}
no h323 call start
Syntax Description
fast
Gateway uses H.323 Version 2 (Fast Connect) procedures.
slow
Gateway uses H.323 Version 1 (Slow Connect) procedures.
Defaults
The default is fast.
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600, 3600, and 7200 series
routers, the AS5300 universal access server and AS5800 universal
gateways, and on the Cisco MC3810 multiservice concentrator.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Usage Guidelines
In Cisco IOS Release 12.1(3)XI and later releases, H.323 Voice over IP (VoIP) gateways by default use
H.323 Version 2 (Fast Connect) for all calls including those initiating RSVP. Previously, gateways used
only Slow Connect procedures for RSVP calls. To enable Cisco IOS Release 12.1(3)XI gateways to be
backward compatible with earlier releases of Cisco IOS Release 12.1 T, the h323 call start command
forces the originating gateway to initiate calls using Slow Connect.
This h323 call start command is configured as part of the global voice-service configuration for VoIP
services. It does not take effect unless the call start system voice-class configuration command is
configured in the VoIP dial peer.
Examples
The following example selects Slow Connect procedures for the gateway:
voice service voip
h323 call start slow
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h323 call start
Related Commands
Command
Description
call rsvp-sync
Enables synchronization between RSVP and the H.323 voice
signaling protocol.
call rsvp-sync resv-timer
Sets the timer for RSVP reservation setup.
call start
Selects whether the H.323 gateway uses Fast Connect or
Slow Connect procedures for the specific VoIP dial peer.
debug call rsvp-sync events
Displays the events that occur during RSVP synchronization.
show call rsvp-sync conf
Displays the RSVP synchronization configuration.
show call rsvp-sync stats
Displays statistics for calls that attempted RSVP reservation.
voice service
Enters voice-service configuration mode and specifies the voice
encapsulation type.
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h323 gatekeeper
h323 gatekeeper
To specify the gatekeeper associated with a proxy and to control how the gatekeeper is discovered, use
the h323 gatekeeper command in interface configuration mode. To disassociate the gatekeeper, use the
no form of this command.
h323 gatekeeper [id gatekeeper-id] {ipaddr ipaddr [port] | multicast}
no h323 gatekeeper [id gatekeeper-id] {ipaddr ipaddr [port] | multicast}
Syntax Description
id gatekeeper-id
(Optional) The gatekeeper-id argument specifies the gatekeeper name.
Typically, this is a Domain Name Server (DNS) name, but it can also be a raw
IP address in dotted form. If this parameter is specified, gatekeepers that have
either the default or explicit flags set for the subnet of the proxy will respond.
If this parameter is not specified, only those gatekeepers with the default
subnet flag will respond.
ipaddr ipaddr [port]
If this parameter is specified, the gatekeeper discovery message will be
unicast to this address and, optionally, the port specified.
multicast
If this parameter is specified, the gatekeeper discovery message will be
multicast to the well-known RAS multicast address and port.
Defaults
No gatekeeper is configured for the proxy.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series routers.
Usage Guidelines
You must enter the h323 interface and h323 h323-id commands before using this command. The h323
gatekeeper command must be specified on your Cisco IOS platform or the proxy will not go online. The
proxy will use the interface address as its RAS signaling address.
Examples
The following example sets up a unicast discovery to a gatekeeper whose name is unknown:
h323 gatekeeper ipaddr 192.168.5.2
The following example sets up a multicast discovery for a gatekeeper of a particular name:
h323 gatekeeper id gk.zone5.com multicast
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h323 gatekeeper
Related Commands
Command
Description
h323 h323-id
Registers an H.323 proxy alias with a gatekeeper.
h323 interface
Specifies the interface from which the proxy will take its IP address.
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h323-gateway voip bind srcaddr
h323-gateway voip bind srcaddr
To designate a source IP address for the voice gateway, use the h323-gateway voip bind srcaddr
command in interface configuration mode. To remove the source IP address, use the no form of the
command.
h323-gateway voip bind srcaddr ip-address
no h323-gateway voip bind srcaddr
Syntax Description
ip-address
Defaults
No default behaviors or values.
Command Modes
Interface configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 1700, 2500, 2600,
3600, and 7200 series routers, the AS5300 universal access server,
and the uBR924.
Specifies the source IP address in dotted-decimal notation.
Usage Guidelines
You can issue this command on any interface in the router. You do not have to issue it on the interface
that you defined as the voice gateway interface (although it may be more convenient to do so). Issuing
the command for one interface assigns the source IP address for the entire router.
Examples
The following example assigns a source IP address of 10.1.1.1:
h323-gateway voip bind srcaddr 10.1.1.1
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h323-gateway voip h323-id
h323-gateway voip h323-id
To configure the H.323 name of the gateway that identifies this gateway to its associated gatekeeper, use
the h323-gateway voip h323-id command in interface configuration mode. To disable this defined
gateway name, use the no form of this command.
h323-gateway voip h323-id interface-id
no h323-gateway voip h323-id interface-id
Syntax Description
interface-id
Defaults
No gateway identification is defined.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 and 3600 series routers
and the Cisco AS5300 universal access server.
Examples
H.323 name (ID) used by this gateway when this gateway communicates
with its associated gatekeeper. Usually, this ID is the name of the gateway
with the gatekeeper domain name appended to the end:
[email protected]
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the
gateway ID is [email protected]
interface Ethernet0/0
ip address 172.16.53.13 255.255.255.0
h323-gateway voip interface
h323-gateway voip id GK15.cisco.com ipaddr 172.16.53.15 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 13#
Related Commands
Command
Description
h323-gateway voip id
Defines the name and location of the gatekeeper for this gateway.
h323-gateway voip
interface
Configures an interface as an H.323 interface.
h323-gateway voip
tech-prefix
Defines the technology prefix that the gateway will register with the
gatekeeper.
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h323-gateway voip id
h323-gateway voip id
To define the name and location of the gatekeeper for a specific gateway, use the h323-gateway voip id
command in interface configuration mode. To disable this gatekeeper identification, use the no form of
this command.
h323-gateway voip id gatekeeper-id {ipaddr ip-address [port-number] | multicast}
[priority number]
no h323-gateway voip id gatekeeper-id {ipaddr ip-address [port-number] | multicast}
[priority number]
Syntax Description
gatekeeper-id
Indicates the H.323 identification of the gatekeeper. This value must exactly
match the gatekeeper ID in the gatekeeper configuration. The recommended
format is name.doman-name.
ipaddr
Indicates that the gateway will use an IP address to locate the gatekeeper.
ip-address
Defines the IP address used to identify the gatekeeper.
port-number
(Optional) Defines the port number used.
multicast
Indicates that the gateway will use multicast to locate the gatekeeper.
priority number
(Optional) The priority of this gatekeeper. The range is 1 through 127, and the
default value is 127.
Defaults
No gatekeeper identification is defined.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 and 3600 series routers and the
Cisco AS5300 universal access server.
12.0(7)T
The priority number keyword and argument option was added.
Usage Guidelines
This command tells the H.323 gateway associated with this interface which H.323 gatekeeper to talk to
and where to locate it. The gatekeeper ID configured here must exactly match the gatekeeper ID in the
gatekeeper configuration.
You can configure up to two alternate gatekeepers.
The IP address of the gatekeeper does not have to be explicit; you can also use the multicast option.
Multicasting saves bandwidth by forcing the network to replicate packets only when necessary. The
multicast option, shown below, notifies every gatekeeper in the LAN using a universal address,
224.0.1.41.
h323-gateway voip id GK1 multicast
h323-gateway voip id GK2 ipaddr 172.18.193.65 1719
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h323-gateway voip id
Examples
The following example configures Ethernet interface 0.0 as the gateway interface and defines a specific
gatekeeper for it. In this example, the gatekeeper ID is GK15.cisco.com and its IP address is
172.16.53.15 (using port 1719).
interface Ethernet0/0
ip address 172.16.53.13 255.255.255.0
h323-gateway voip interface
h323-gateway voip id GK15.cisco.com ipaddr 172.16.53.15 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 13#
Related Commands
Command
Description
h323-gateway voip
h323-id
Configures the H.323 name of the gateway that identifies this gateway to its
associated gatekeeper.
h323-gateway voip
interface
Configures an interface as an H.323 interface.
h323-gateway voip
tech-prefix
Defines the technology prefix that the gateway will register with the
gatekeeper.
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h323-gateway voip interface
h323-gateway voip interface
To configure an interface as an H.323 gateway interface, use the h323-gateway voip interface command
in interface configuration mode. To disable H.323 gateway functionality for an interface, use the no form
of this command.
h323-gateway voip interface
no h323-gateway voip interface
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Interface configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 and 3600 series routers
and the AS5300 universal access server.
Examples
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the
h323-gateway voip interface command configures this interface as an H.323 interface.
interface Ethernet0/0
ip address 172.16.53.13 255.255.255.0
h323-gateway voip interface
h323-gateway voip id GK15.cisco.com ipaddr 172.16.53.15 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 13#
Related Commands
Command
Description
h323-gateway voip
h323-id
Configures the H.323 name of the gateway that identifies this gateway to its
associated gatekeeper.
h323-gateway voip id
Defines the name and location of the gatekeeper for this gateway.
h323-gateway voip
tech-prefix
Defines the technology prefix that the gateway will register with the
gatekeeper.
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h323-gateway voip tech-prefix
h323-gateway voip tech-prefix
To define the technology prefix that the gateway will register with the gatekeeper, use the h323-gateway
voip tech-prefix command in interface configuration mode. To disable this defined technology prefix,
use the no form of this command.
h323-gateway voip tech-prefix prefix
no h323-gateway voip tech-prefix prefix
Syntax Description
prefix
Defaults
Disabled
Command Modes
Interface configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 and 3600 series routers
and the Cisco AS5300 universal access server.
Usage Guidelines
Note
Examples
Defines the numbers used as the technology prefixes. Each technology
prefix can contain up to 11 characters. Although not strictly necessary, a
pound symbol (#) is frequently used as the last digit in a technology prefix.
Valid characters are 0 though 9, the pound symbol (#), and the asterisk (*).
This command defines a technology prefix that the gateway will then register with the gatekeeper.
Technology prefixes can be used as a discriminator so that the gateway can tell the gatekeeper that a
certain technology is associated with a particular call (for example, 15# could mean a fax transmission),
or it can be used like an area code for more generic routing. No standard currently defines what the
numbers in a technology prefix mean. By convention, technology prefixes are designated by a pound
symbol (#) as the last character.
Cisco gatekeepers use the asterisk (*) as a reserved character. If you are using Cisco gatekeepers, do
not use the asterisk as part of the technology prefix.
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the
technology prefix is defined as 13#.
interface Ethernet0/0
ip address 172.16.53.13 255.255.255.0
h323-gateway voip interface
h323-gateway voip id GK15.cisco.com ipaddr 172.16.53.15 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 13#
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h323-gateway voip tech-prefix
Related Commands
Command
Description
h323-gateway voip
h323-id
Configures the H.323 name of the gateway that identifies this gateway to its
associated gatekeeper.
h323-gateway voip id
Defines the name and location of the gatekeeper for this gateway.
h323-gateway voip
interface
Configures an interface as an H.323 interface.
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h323 h323-id
h323 h323-id
To register an H.323 proxy alias with a gatekeeper, use the h323 h323-id command in interface
configuration mode. To remove an H.323 proxy alias, use the no form of this command.
h323 h323-id h323-id
no h323 h323-id h323-id
Syntax Description
h323-id
Defaults
No H.323 proxy alias is registered.
Command Modes
Interface configuration
Command History
Release
Usage Guidelines
Note
Examples
Specifies the name of the proxy. It is recommended that this name be a fully
qualified e-mail ID, with the domain name being the same as that of its
gatekeeper.
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series routers.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Each entry registers a specified H.323 ID proxy alias to a gatekeeper. Typically, these aliases are either
simple text strings or legitimate e-mail IDs.
You must enter the h323 interface command before using this command. The h323 h323-id
command must be entered on the same interface as the h323 gatekeeper command. The proxy will
not go online without the h323 interface command.
The following example registers an H.323 proxy alias called [email protected] with a gatekeeper:
h323 h323-id [email protected]
Related Commands
Command
Description
h323 gatekeeper
Specifies the gatekeeper associated with a proxy and controls how the
gatekeeper is discovered.
h323 interface
Specifies the interface from which the proxy will take its IP address.
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h323 interface
h323 interface
To select an interface whose IP address will be used by the proxy to register with the gatekeeper, use the
h323 interface command in interface configuration mode. To use the default port, use the no h323
interface command and then the h323 interface command.
h323 interface [port-number]
no h323 interface [port-number]
Syntax Description
port-number
Defaults
Default port number is image dependent as described in the Syntax Description.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series routers.
12.1(5)T
The ability to specify the proxy port number was added on the 2600, 3600,
and 7200 series routers and on the Cisco MC3810 multiservice
concentrator.
(Optional) The port number the proxy will listen on for incoming call setup
requests. Range is 1 to 65,356. The default port number for the proxy is
11,720 in -isx- or -jsx- Cisco IOS images. The default port number for the
proxy is 1720 in -ix- Cisco IOS images, which do not contain the VoIP
gateway.
Usage Guidelines
At proxy startup, Cisco IOS software checks for the presence of the VoIP gateway subsystem. If the
subsystem is found to be present, the proxy code opens and listens for call setup requests on the new
port. The proxy then registers this port with the gatekeeper.
Examples
The following example shows how to configure Ethernet interface 0 for incoming call setup requests:
interface ethernet0
h323 interface
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h323 interface
Related Commands
Command
Description
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic from a zone
to another zone, within a zone, or for a session in a zone.
bandwidth remote
Specifies the total bandwidth for H.323 traffic between this gatekeeper and
any other gatekeeper.
h323 qos
Enables QoS on the proxy.
h323 t120
Enables the T.120 capabilities on your router and specifies bypass or proxy
mode.
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h323 qos
h323 qos
To enable quality of service (QoS) on the proxy, use the h323 qos command in interface configuration
mode. To disable QoS, use the no form of this command.
h323 qos {ip-precedence value | rsvp {controlled-load | guaranteed-qos}}
no h323 qos {ip-precedence value | rsvp {controlled-load | guaranteed-qos}}
Syntax Description
ip-precedence value
Specifies that RTP streams should set their IP precedence bits to the specified
value.
rsvp controlled-load
Specifies controlled load class of service.
rsvp guaranteed-qos
Specifies guaranteed QoS class of service.
Defaults
No QoS is configured.
Command Modes
Interface configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 and 3600 series routers.
Usage Guidelines
You must execute the h323 interface command before using this command.
Both IP precedence and RSVP QoS can be configured by invoking this command twice with the two
different QoS forms.
Examples
The following example enables QoS on the proxy:
interface Ethernet0
ip address 172.21.127.38 255.255.255.192
no ip redirects
ip rsvp bandwidth 7000 7000
ip route-cache same-interface
fair-queue 64 256 1000
h323 interface
h323 qos rsvp controlled-load
h323 h323-id [email protected]
h323 gatekeeper ipaddr 172.21.127.39
Related Commands
Command
Description
h323 interface
Specifies the interface from which the proxy will take its IP address.
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h323 t120
h323 t120
To enable the T.120 capabilities on your router and to specify bypass or proxy mode, use the h323 t120
command in interface configuration mode.
h323 t120 {bypass | proxy}
Syntax Description
bypass
Bypass mode. In this mode, the H.245 Open Logical Channel messages for
T.120 data channels are passed unmodified through the proxy, and TCP
connections for T.120 are established directly between the two endpoints of
the H.323 call.
proxy
Proxy mode. In this mode, T.120 features function properly.
Defaults
Bypass mode
Command Modes
Interface configuration
Command History
Release
Modification
12.1(5)T
This command was introduced on on the Cisco 2600, 3600, and 7200 series
routers and the Cisco MC3810 multiservice concentrator.
Usage Guidelines
The no form of this command has no function—the only possible commands are h323 t120 bypass and
h323 t120 proxy.
Examples
The following example shows how to enable the T.120 capabilities:
proxy h323
interface ethernet0
h323 t120 proxy
Related Commands
Command
Description
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic from a zone
to another zone, within a zone, or for a session in a zone.
bandwidth remote
Specifies the total bandwidth for H.323 traffic between this gatekeeper and
any other gatekeeper.
h323 interface
Defines which port the proxy will listen on.
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huntstop
huntstop
To disable all dial-peer hunting if a call fails when using hunt groups, use the huntstop command in
dial-peer configuration mode. To reenable dial-peer hunting, use the no form of this command.
huntstop
no huntstop
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(5)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
Support for this command was extended to the Cisco 2600 and 3600 series
routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
Note
Examples
Once you enter this command, no further hunting is allowed if a call fails on the specified dial peer.
This command can be used with all types of dial peers.
The following example shows how to disable dial-peer hunting on a specific dial peer:
dial peer voice 100 vofr
huntstop
The following example shows how to reenable dial-peer hunting on a specific dial peer:
dial peer voice 100 vofr
no huntstop
Related Commands
Command
Description
dial-peer voice
Enters dial-peer configuration mode and specifies the method of
voice-related encapsulation.
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icpif
icpif
To specify the Impairment/Calculated Planning Impairment Factor (ICPIF) for calls sent by a dial peer,
use the icpif command in dial-peer configuration mode. To restore the default value, use the no form of
this command.
icpif integer
no icpif integer
Syntax Description
integer
Defaults
20
Command Modes
Dial-peer configuration
Command History
Release
Usage Guidelines
Integer, expressed in equipment impairment factor units, that specifies the
ICPIF value. Valid entries are 0 to 55. The default is 20.
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.0(7)XK
This command was first supported on the Cisco MC3810 multiservice
concentrator.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
This command is applicable only to Voice over IP (VoIP) dial peers.
Use the icpif command to specify the maximum acceptable impairment factor for the voice calls sent by
the selected dial peer.
Examples
The following example disables the icpif command:
dial-peer voice 10 voip
icpif 0
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idle-voltage
idle-voltage
To specify the idle voltage on an Foreign Exchange Station (FXS) voice port, use the idle-voltage
command in voice-port configuration mode. To restore the default idle voltage, use the no form of this
command.
idle-voltage {high | low}
no idle-voltage
Syntax Description
high
The talk-battery (tip-to-ring) voltage is high (–48V) when the FXS port is idle.
low
The talk-battery (tip-to-ring) voltage is low (–24V) when the FXS port is idle.
Defaults
The idle voltage is –24V.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(4)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
Some fax equipment and answering machines require a –48V idle voltage to be able to detect an off-hook
condition in a parallel phone.
If the idle voltage setting is high, the talk battery reverts to –24V whenever the voice port is active (off
hook).
The idle-voltage command applies only to FXS voice ports on Cisco MC3810 multiservice
concentrators.
Examples
The following example sets the idle voltage to –48V on voice port 1/1 on a Cisco MC3810 multiservice
concentrator:
voice-port 1/1
idle-voltage high
The following example restores the default idle voltage (–24V) on voice port 1/1 on a Cisco MC3810
multiservice concentrator:
voice-port 1/1
no idle-voltage
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idle-voltage
Related Commands
Command
Description
show voice port
Displays voice port configuration information.
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ignore
ignore
To configure the North American E&M or E&M MELCAS voice port to ignore specific receive bits, use
the ignore command in voice-port configuration mode. To restore the default value, use the no form of
this command.
ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
no ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
Syntax Description
Defaults
rx-a-bit
Ignores the receive A bit.
rx-b-bit
Ignores the receive B bit.
rx-c-bit
Ignores the receive C bit.
rx-d-bit
Ignores the receive D bit.
The default is mode-dependent:
•
North American E&M:
– The receive B, C, and D bits are ignored.
– The receive A bit is not ignored.
•
E&M MELCAS:
– The receive A bit is ignored.
– The receive B, C, and D bits are not ignored.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The ignore command applies to E&M digital voice ports associated with T1/E1 controllers. Repeat the
command for each receive bit to be configured. Use this command with the define command.
Examples
To configure voice port 1/1 on a Cisco MC3810 multiservice concentrator to ignore receive bits A, B,
and C and to monitor receive bit D, enter the following commands:
voice-port 1/1
ignore rx-a-bit
ignore rx-b-bit
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ignore
ignore rx-c-bit
no ignore rx-d-bit
To configure voice port 1/0/0 on a Cisco 3600 series router to ignore receive bits A, C, and D and to
monitor receive bit B, enter the following commands:
voice-port 1/0/0
ignore rx-a-bit
ignore rx-c-bit
ignore rx-d-bit
no ignore rx-b-bit
Related Commands
Command
Description
condition
Manipulates the signaling bit pattern for all voice signaling types.
define
Defines the transmit and receive bits for North American E&M and E&M
MELCAS voice signaling.
show voice port
Displays configuration information for voice ports.
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image encoding
image encoding
To select a specific encoding method for fax images associated with an MMoIP dial peer, use the image
encoding command in dial-peer configuration mode. To restore the default value, use the no form of this
command.
image encoding {mh | mr | mmr | passthrough}
no image encoding {mh | mr | mmr | passthrough}
Syntax Description
mh
Specifies Modified Huffman image encoding. This is the IETF standard.
mr
Specifies Modified Read image encoding.
mmr
Specifies Modified Modified Read image encoding.
passthrough
Specifies that the image will not be modified by an encoding method.
Defaults
passthrough
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Use the image encoding command to specify an encoding method for e-mail fax TIFF images for a
specific MMoIP dial peer. This command applies primarily to the on-ramp MMoIP dial peer. Although
you can optionally create an off-ramp dial peer and configure a particular image encoding value for that
off-ramp call leg, store and forward fax ignores the off-ramp MMoIP setting and sends the file using
Modified Huffman encoding.
There are four available encoding methods:
•
Modified Huffman (MH)—One-dimensional data compression scheme that compresses data in only
one direction (horizontal). Modified Huffman compression does not allow the transmission of
redundant data. This encoding method produces the largest image file size.
•
Modified Read (MR)—Two-dimensional data compression scheme (used by fax devices) that
handles the data compression of the vertical line and that concentrates on the space between lines
and within given characters.
•
Modified Modified Read (MMR)—Data compression scheme used by newer Group 3 fax devices.
This encoding method produces the smallest possible image file size and is slightly more efficient
than Modified Read.
•
Passthrough—No encoding method will be applied to the image—meaning that the image will be
encoded by whatever encoding method is used by the fax device.
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image encoding
The IETF standard for sending fax TIFF images is Modified Huffman encoding with fine or standard
resolution. RFC 2301 requires that compliant receivers support TIFF images with MH encoding and fine
or standard resolution. If a receiver supports features beyond this minimal requirement, you might want
to configure the Cisco AS5300 universal access server to send enhanced-quality documents to that
receiver.
The primary reason to use a different encoding scheme from MH is to save network bandwidth. MH
ensures interoperability with all Internet fax devices, but it is the least efficient of the encoding schemes
for sending fax TIFF images. For most images, MR is more efficient than MH, and MMR is more
efficient than MR. If you know that the recipient is capable of receiving more efficient encodings than
just MH, store and forward fax allows you to send the most efficient encoding that the recipient can
process. For end-to-end closed networks, you can choose any encoding scheme because the off-ramp
gateway can process MH, MR, and MMR.
Another factor to consider is the viewing software. Many viewing applications (for example, those that
come with Windows 95 or Windows NT) are able to display MH, MR, and MMR. Therefore you should
decide, based on the viewing application and the available bandwidth, which encoding scheme is right
for your network.
This command applies to both on-ramp and off-ramp store and forward fax functions.
Examples
The following example selects Modified Modified Read as the encoding method for fax TIFF images
sent by MMoIP dial peer 10:
dial-peer voice 10 mmoip
image encoding mmr
Related Commands
Command
Description
image resolution
Specifies a particular fax image resolution for a specific MMoIP dial peer.
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image resolution
image resolution
To specify a particular fax image resolution for a specific MMoIP dial peer, use the image resolution
command in dial-peer configuration mode. To restore the default value, use the no form of this
command.
image resolution {fine | standard | superfine | passthrough}
no image resolution {fine | standard | superfine | passthrough}
Syntax Description
fine
Configures the fax TIFF image resolution to be 204-by-196 pixels per inch.
standard
Configures the fax TIFF image resolution to be 204-by-98 pixels per inch.
superfine
Configures the fax TIFF image resolution to be 204-by-391 pixels per inch.
passthrough
Indicates that the resolution of the fax TIFF image will not be altered.
Defaults
passthrough
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Use the image resolution command to specify a specific resolution (in pixels per inch) for e-mail fax
TIFF images sent by the specified MMoIP dial peer. This command applies primarily to the on-ramp
MMoIP dial peer. Although you can optionally create an off-ramp dial peer and configure a particular
image resolution value for that off-ramp call leg, store and forward fax ignores the off-ramp MMoIP
setting and sends the file using fine resolution.
This command enables you to increase or decrease the resolution of a fax TIFF image, thereby changing
not only the resolution but also the size of the fax TIFF file. The IETF standard for sending fax TIFF
images is Modified Huffman encoding with fine or standard resolution. The primary reason to configure
a different resolution is to save network bandwidth.
This command applies to both on-ramp and off-ramp store and forward fax functions.
Examples
The following example selects the fine resolution (meaning 204-by-196 pixels per inch) for e-mail fax
TIFF images associated with MMoIP dial peer 10:
dial-peer voice 10 mmoip
image encoding mh
image resolution fine
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image resolution
Related Commands
Command
Description
image encoding
Selects a specific encoding method for fax images associated with an
MMoIP dial peer.
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impedance
impedance
To specify the terminating impedance of a voice-port interface, use the impedance command in
voice-port configuration mode. To restore the default value, use the no form of this command.
impedance {600c | 600r | 900c | complex1 | complex2}
no impedance {600c | 600r | 900c | complex1 | complex2}
Syntax Description
600c
Specifies 600 ohms (complex).
600r
Specifies 600 ohms (real).
900c
Specifies 900 ohms (complex).
complex1
Specifies complex 1.
complex2
Specifies complex 2.
Defaults
600r
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
Use the impedance command to specify the terminating impedance of an Foreign Exchange Office
(FXO) voice-port interface. The impedance value selected needs to match the specifications from the
specific telephony system to which it is connected. Different countries often have different standards for
impedance. CO switches in the United States are predominantly 600r. PBXs in the United States are
normally either 600r or 900c.
If the impedance is set incorrectly (if there is an impedance mismatch), there will be a significant amount
of echo generated (which could be masked if the echo-cancel command has been enabled). In addition,
gains might not work correctly if there is an impedance mismatch.
Configuring the impedance on a voice port will change the impedance on both voice ports of a VPM
card. This voice port must be shut down and then opened for the new value to take effect.
Examples
The following example configures an FXO voice port on the Cisco 3600 series router for a terminating
impedance of 600 ohms (real):
voice-port 1/0/0
impedance 600r
The following example configures an E&M voice port on the Cisco MC3810 multiservice concentrator
for a terminating impedance of 900 ohms (complex):
voice-port 1/1
impedance 900c
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incoming called-number
incoming called-number
To specify a digit string that can be matched by an incoming call to associate the call with a dial peer,
use the incoming called-number command in dial-peer configuration mode. To reset the default value,
use the no form of this command.
incoming called-number string
no incoming called-number string
Syntax Description
string
Defaults
No incoming called number is defined.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
11.3NA
This command was introduced on the Cisco AS5800 universal gateway.
12.0(4)XJ
This command was modified for store and forward fax.
12.0(7)XK
This command was first supported on the Cisco MC3810 multiservice
concentrator platform.
12.1(2)T
This command was integrated into the Cisco IOS Release 12.1(2)T.
Usage Guidelines
Specifies the incoming called telephone number. Valid entries are any series
of digits that specify the E.164 telephone number.
When a Cisco device (such as a Cisco AS5300 universal access server or Cisco AS5800 universal
gateway) is handling both modem and voice calls, it needs to be able to identify the service type of the
call—meaning whether the incoming call to the server is a modem or a voice call. When the access server
handles only modem calls, the service type identification is handled through modem pools. Modem pools
associate calls with modem resources based on the called number (DNIS). In a mixed environment, in
which the server receives both modem and voice calls, you need to identify the service type of a call by
using the incoming called-number command.
If you do not use the incoming called-number command, the server attempts to resolve whether an
incoming call is a modem or voice call based on the interface over which the call comes. If the call comes
in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes
in over a voice port associated with a dial peer, the call is assumed to be a voice call.
By default, there is no called number associated with the dial peer, which means that incoming calls will
be associated with dial peers based on matching calling number with answer address, call number with
destination pattern, or calling interface with configured interface.
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incoming called-number
Use the incoming called-number command to define the destination telephone number for a particular
dial peer. For the on-ramp POTS dial peer, this telephone number is the DNIS number of the incoming
fax call. For the off-ramp MMoIP dial peer, this telephone number is the destination fax machine
telephone number.
This command applies to both Voice over IP (VoIP) and POTS dial peers and applies to both on-ramp
and off-ramp store and forward fax functions.
This command is also used to provide a matching VoIP dial peer on the basis of called number when fax
or modem pass-through with named service events (NSEs) is defined globally on a terminating gateway.
You can ensure that all calls will match at least one dial peer by using the following commands:
Router(config)# dial-peer voice tag voip
Router(config-dial-peer)# incoming called-number .
Examples
The following example configures calls coming in to the server with a called number of 3799262 as
being voice calls:
dial peer voice 10 pots
incoming called-number 3799262
The following example configures the number (310) 555-9261 as the incoming called number for
MMoIP dial peer 10:
dial-peer voice 10 mmoip
incoming called-number 3105559261
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info-digits
info-digits
To automatically prepend two information digits to the beginning of a dialed number associated with the
given POTS dial peer, use the info-digits command in dial-peer configuration mode. To keep the router
from automatically prepending the two-digit information numbers to the beginning of the POTS dial
peer, use the no form of this command.
info-digits string
no info-digits
Syntax Description
string
Specifies the two-digit prefix that the router will automatically prepend to the
dialed number for the given POTS dial peer.
Note
This string cannot contain any more or any less than two digits.
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.2(1)T
This command was introduced on Cisco 2600 series, Cisco 3600 series, and
Cisco 3700 series routers and on Cisco AS5300 series universal access
servers.
Usage Guidelines
This command is designed to prepend a pair of information digits to the beginning of the dialed number
string for the POTS dial peer that will enable you to dynamically redirect the outgoing call. The
info-digits command is only available for POTS dial peers.
Examples
The following example prepends the information number string 91 to the beginning of the dialed number
for POTS dial peer 10:
dial-peer voice 10 pots
info-digits 91
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information-type
information-type
To select a particular information type for either an Mail Message over IP (MMoIP) or Plain Old
Telephone Service (POTS) dial peer, use the information-type command in dial-peer configuration
mode. To reset the default value for this command, use the no form of this command.
information-type {fax | voice}
no information-type {fax | voice}
Syntax Description
fax
Indicates that the information type has been set to store and forward fax.
voice
Indicates that the information type has been set to voice.
Defaults
Voice
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.0(4)XJ
This command was modified for store and forward fax.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This command applies to both on-ramp and off-ramp store and forward fax functions.
Examples
The following example sets the information type for MMoIP dial peer 10 to fax:
dial-peer voice 10 mmoip
information-type fax
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input gain
input gain
To configure a specific input gain value, use the input gain command in voice-port configuration mode.
To disable the selected amount of inserted gain, use the no form of this command.
input gain decibels
no input gain decibels
Syntax Description
decibels
Defaults
Zero (0) decibels
Command Modes
Voice-port configuration
Command History
Release
Usage Guidelines
Specifies, in decibels, the amount of gain to be inserted at the receiver side of
the interface. Acceptable values are integers from –6 to 14.
Modification
11.3(1)T
This command was introduced.
11.3(1)MA
This command was first supported on the Cisco MC3810 multiservice
concentrator.
A system-wide loss plan must be implemented using both the input gain and output attenuation
commands. Other equipment (including PBXs) in the system must be considered when creating a loss
plan. The default value for this command assumes that a standard transmission loss plan is in effect,
meaning that there must be an attenuation of –6 dB between phones. Connections are implemented to
provide –6 dB of attenuation when the input gain and output attenuation commands are configured
with the default value of 0 dB.
You cannot increase the gain of a signal to the Public Switched Telephone Network (PSTN), but you can
decrease it. If the voice level is too high, you can decrease the volume by either decreasing the input gain
or increasing the output attenuation.
You can increase the gain of a signal coming into the router. If the voice level is too low, you can increase
the input gain by using the input gain command.
Examples
The following example configures a 3-dB gain to be inserted at the receiver side of the interface in the
Cisco 3600 series router:
port 1/0/0
input gain 3
The following example configures a 3-dB gain to be inserted at the receiver side of the interface in the
Cisco MC3810 multiservice concentrator:
port 1/1
input gain 3
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input gain
Related Commands
Command
Description
output attenuation
Configures a specific output attenuation value for a voice port.
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interface (RLM server)
interface (RLM server)
To define the IP addresses of the Redundant Link Manager (RLM) server, use the interface command
in interface configuration mode. To disable this function, use the no form of this command.
interface name-tag
no interface name-tag
Syntax Description
name-tag
Defaults
Disabled
Command Modes
Interface configuration
Command History
Release
Modification
11.3(7)
This command was introduced.
Name to identify the server configuration so that multiple entries of server
configuration can be entered.
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example shows how to configure the access server interfaces for RLM servers named
Loopback1 and Loopback2:
interface Loopback1
ip address 10.1.1.1 255.255.255.255
interface Loopback2
ip address 10.1.1.2 255.255.255.255
rlm group 1
server r1-server
link address 10.1.4.1 source Loopback1 weight 4
link address 10.1.4.2 source Loopback2 weight 3
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interface (RLM server)
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
link (RLM)
Specifies the link preference.
protocol rlm port
Reconfigures the port number for the basic RLM connection for the whole
rlm-group.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the
link is declared down.
server (RLM)
Defines the IP addresses of the server.
show rlm group
statistics
Displays the network latency of the RLM group.
show rlm group
status
Displays the status of the RLM group.
show rlm group timer Displays the current RLM group timer values.
shutdown (RLM)
Shuts down all of the links under the RLM group.
timer
Overwrites the default setting of timeout values.
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ip precedence (dial-peer)
ip precedence (dial-peer)
To set IP precedence (priority) for packets sent by the dial peer, use the ip precedence command in
dial-peer configuration mode. To restore the default value, use the no form of this command.
ip precedence number
no ip precedence number
Syntax Description
number
Defaults
The default value for this command is zero (0).
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)NA
This command was introduced on the Cisco 2500, 3600 series routers and the
Cisco AS5300 universal access server.
Usage Guidelines
Integer specifying the IP precedence value. Valid entries are from 0 to 7. A
value of 0 means that no precedence (priority) has been set.
Use the ip precedence (dial-peer) command to configure the value set in the IP precedence field when
voice data packets are sent over the IP network. This command should be used if the IP link utilization
is high and the quality of service for voice packets needs to have a higher priority than other IP packets.
The ip precedence (dial-peer) command should also be used if RSVP is not enabled and the user would
like to give voice packets a higher priority than other IP data traffic.
This command applies to Voice over IP (VoIP) peers.
Examples
The following example sets the IP precedence to 5:
dial-peer voice 10 voip
ip precedence 5
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ip udp checksum
ip udp checksum
To calculate the UDP checksum for voice packets sent by the dial peer, use the ip udp checksum
command in dial-peer configuration mode. To disable this feature, use the no form of this command.
ip udp checksum
no ip udp checksum
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
Use the ip udp checksum command to enable UDP checksum calculation for each of the outbound voice
packets. This command is disabled by default to speed up the transmission of the voice packets. If you
suspect that the connection has a high error rate, you should enable the ip udp checksum command to
prevent corrupted voice packets forwarded to the digital signal processor (DSP).
This command applies to Voice over IP (VoIP) peers.
Examples
The following example calculates the UDP checksum for voice packets sent by dial peer 10:
dial-peer voice 10 voip
ip udp checksum
Related Commands
Command
Description
loop-detect
Enables loop detection for T1 for Voice over ATM, Voice over Frame Relay,
and Voice over HDLC.
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isdn bind-l3
isdn bind-l3
To configure the ISDN serial interface for backhaul, use the isdn bind-l3 command in interface
configuration mode. To disable backhaul on the interface, use the no form of this command.
isdn bind-l3 set-name
no isdn bind-l3 set-name
Syntax Description
set-name
Defaults
No default behavior or values.
Command Modes
Interface configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
Session-set name.
To configure the ISDN serial interface for backhaul for the set named Set1, see the following example:
Router(config-if)# isdn bind-l3 set1
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isdn contiguous-bchan
isdn contiguous-bchan
To configure contiguous bearer channel handling on an E1 PRI interface, use the isdn
contiguous-bchan command in interface configuration mode. To disable the contiguous B-channel
handling, use the no form of this command.
isdn contiguous-bchan
no isdn contiguous-bchan
Syntax Description
This command has no arguments or keywords.
Defaults
By default, contiguous B channel handling is disabled.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco 2500, 3600, and 7200 series
routers and on the Cisco MC3810 multiservice concentrator.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
Use the isdn contiguous-bchan command to specify contiguous bearer channel handling so that B
channels 1 through 30, skipping 16, map to time slots 1 through 31. This is available for E1 PRI
interfaces only, when the primary-qsig switch type option is configured by using the isdn switch-type
command.
Examples
The following example shows the command configuration on the E1 interface of a Cisco 3660 series
router E1 interface:
interface Serial5/0:15
no ip address
ip mroute-cache
no logging event link-status
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn continguous-bchan
Related Commands
Command
Description
isdn switch-type
primary-qsig
Configures the primary-qsig switch type for PRI support.
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isdn global-disconnect
isdn global-disconnect
To allow passage of “release” and “release complete” messages over the voice network, use the isdn
global-disconnect command in interface configuration mode. To disable the passage of these messages,
use the no form of this command.
isdn global-disconnect
no isdn global-disconnect
Syntax Description
This command has no arguments or keywords.
Defaults
Passage of messages is disabled by default; “release” and “release complete” messages terminate locally
by default.
Command Modes
Interface configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 2600, 3600, and 7200 series
routers and on the Cisco MC3810 multiservice concentrator.
Usage Guidelines
Enter this command under the isdn interface with switch type bri-qsig or pri-qsig. Use the isdn
global-disconnect command to allow passage of “release” and “release complete” messages end-to-end
across the network. This is required for certain types of QSIG PBXs whose software or features require
either Facility or User Info IEs in those messages to be passed end-to-end between the PBXs. All QSIG
interfaces that connect the PBXs to the routers must have this command enabled. This command is
available when using the BRI QSIG or PRI QSIG switch type in either master or slave mode.
Examples
The following example shows command configuration on the T1 PRI interface of a Cisco 3660 series
router:
interface Serial5/0:23
no ip address
ip mroute-cache
no logging event link-status
isdn switch-type primary-qsig
isdn global-disconnect
isdn overlap-receiving
isdn incoming-voice voice
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isdn global-disconnect
Related Commands
Command
Description
isdn protocol-emulate
Configures the interface to serve as either the QSIG slave or the
QSIG master.
isdn switch-type
Configures the switch type for BRI or PRI support.
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isdn i-number
isdn i-number
To configure several terminal devices to use one subscriber line, use the isdn i-number command in
interface configuration mode.
isdn i-number n ldn
Syntax Description
n
Subscriber line 1, 2, or 3, as specified in the NTT specification.
ldn
LDN assigned to the router plain old telephone service (POTS) port.
Defaults
Each terminal device uses one subscriber line.
Command Modes
Interface configuration
Command History
Release
Modification
12.1.(2)XF
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
Enter the interface bri command before entering the isdn i-number command.
Examples
The following example shows screen output for two LDNs configured under BRI interface 0:
interface bri0
isdn i-number 1 5551234
isdn i-number 2 5556789
exit
dial-peer voice 1 pots
destination-pattern 5551234
exit
dial-peer voice 2 pots
destination-pattern 5556789
exit
Related Commands
Command
Description
interface bri
Specifies a BRI interface and enters interface configuration mode.
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isdn network-failure-cause
isdn network-failure-cause
To specify the cause code to pass to the PBX when a call cannot be placed or completed because of
internal network failures, use the isdn network-failure-cause command in interface configuration
mode. To unconfigure the use of this cause code, use the no form of this command.
isdn network-failure-cause value
no isdn network-failure-cause value
Syntax Description
value
Defaults
No default behavior or values.
Command Modes
Interface configuration
Command History
Release
Modification
12.1(2)T
This command was introduced to the Cisco IOS 12.1(2)T on the Cisco 2600,
3600, and 7200 series routers and on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
Number from 1 to 127. See Table 24 for a list of failure cause code values.
The PBX can reroute calls based on the cause code returned by the router.
This command allows the original cause code to be changed to the value specified if the original cause
code is not one of the following:
•
NORMAL_CLEARING (16)
•
USER_BUSY (17)
•
NO_USER_RESPONDING (18)
•
NO_USER_ANSWER (19)
•
NUMBER_CHANGED (22)
•
INVALID_NUMBER_FORMAT (28)
•
UNSPECIFIED_CAUSE (31)
•
UNASSIGNED_NUMBER (1)
Table 24 describes the cause codes.
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isdn network-failure-cause
Table 24
ISDN Failure Cause Codes
Failure Cause Code
Meaning
1
Unallocated or unassigned number.
2
No route to specified transit network.
3
No route to destination.
6
Channel unacceptable.
7
Call awarded and being delivered in an established channel.
16
Normal call clearing.
17
User busy.
18
No user responding.
19
No answer from user (user alerted).
21
Call rejected.
22
Number changed.
26
Nonselected user clearing.
27
Destination out of order.
28
Invalid number format.
29
Facility rejected.
30
Response to status enquiry.
31
Normal, unspecified.
34
No circuit/channel available.
38
Network out of order.
41
Temporary failure.
42
Switch congestion.
43
Access information discarded.
44
Requested channel not available.
45
Preempted.
47
Resources unavailable, unspecified.
49
Quality of service unavailable.
50
Requested facility not subscribed.
52
Outgoing calls barred.
54
Incoming calls barred.
57
Bearer capability not authorized.
58
Bearer capability not available now.
63
Service or option not available, unspecified.
65
Bearer capability not implemented.
66
Channel type not implemented.
69
Requested facility not implemented.
70
Only restricted digital information bearer capability is available.
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isdn network-failure-cause
Table 24
Examples
ISDN Failure Cause Codes (continued)
Failure Cause Code
Meaning
79
Service or option not implemented, unspecified.
81
Invalid call reference value.
82
Identified channel does not exist.
83
Suspended call exists, but this call ID does not.
84
Call ID in use.
85
No call suspended.
86
Call with requested call ID is cleared.
88
Incompatible destination.
91
Invalid transit network selection.
95
Invalid message, unspecified.
96
Mandatory information element missing.
97
Message type nonexistent or not implemented.
98
Message not compatible with call state or message type nonexistent or not
implemented.
99
Information element nonexistent or not implemented.
100
Invalid information element contents.
101
Message not compatible with call state.
102
Recovery on timer expiry.
111
Protocol error, unspecified.
127
Interworking, unspecified.
The following is an example of specifying a cause code to pass to a PBX when a call cannot be placed
or completed of internal network failures:
isdn network-failure-cause 28
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ivr autoload
ivr autoload
To load files from a particular TFTP server (as indicated by a defined URL), use the ivr autoload
command in global configuration mode. To disable this function, use the no form of this command.
ivr autoload url location
no ivr autoload url location
Syntax Description
url
Indicates that a URL is used to locate the index file that contains a list of all
available audio files.
location
Specifies the URL of the index file.
Defaults
No URL is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 and 3600 series routers and
the Cisco AS5300 universal access server.
Usage Guidelines
The index file contains a list of audio files (URL) that can be downloaded from the TFTP server. Use
the ivr autoload command to download audio files from TFTP to memory. The command only starts up
a background process. The background process (loader) does the actual down loading of the files.
The background process first reads the index file from either Flash or TFTP. It parses the files line by
line looking for the URL. It ignores lines that start with # as comment lines. Once it has a correct URL,
it tries to read that .au file into memory and creates a media object. If there are any errors during the
reading of the file, it retries the configured number of times. If the mode is set to “verbose,” the loader
logs the transaction to console. Once parsing has reached the end of the index file, the background
process exits memory.
Perform the following checks before initiating the background process. If one of the checks fail, it
indicates the background process is not started, and instead you will see an error response to the
command.
•
Check if any prompt is being actively used (IVR is actively playing some prompts). If there are
active prompts, the command fails, displaying the following error message (.au files are also
referred to as prompts):
command is not allowed when prompts are active
•
Check if there is already a background process in progress. If there is a process, the command fails,
displaying the following error:
previous autoload command is still in progress
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ivr autoload
•
Check if an earlier ivr autoload command has already been configured. If an ivr autoload
command has already been configured, the user sees the following response when the command is
issued:
previous command is being replaced
•
When the no ivr autoload command is issued, if there was already an ivr autoload command in
progress, it will be aborted.
The audio files (prompts) loaded using the ivr autoload command are not dynamically swapped out of
memory. They are considered as autoloaded prompts as opposed to “dynamic” prompts. (See the
ivr prompt memory command for details on dynamic prompts.)
Examples
The following example loads audio files from the TFTP server (located at
//jurai/mgindi/tclware/index4):
ivr autoload url tftp://jurai/mgindi/tclware/index4
The index file for this example index4 is shown as follows:
more index4
tftp://jurai/mgindi/tclware/au/en/en_one.au
tftp://jurai/mgindi/tclware/au/ch/ch_one.au
tftp://jurai/mgindi/tclware/au/ch/ch_one.au
An example of an index file on Flash is shown as follows:
flash:index
Related Commands
Command
Description
ivr prompt memory
Configures the maximum amount of memory that the dynamic audio files
(prompts) occupy in memory.
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ivr autoload retry
ivr autoload retry
To specify the number of times that the system will try to load audio files from TFTP to memory when
there is an error, use the ivr autoload retry command in global configuration mode. To disable this
function, use the no form of this command.
ivr autoload retry number
no ivr autoload retry number
Syntax Description
number
Defaults
Three times
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 and 3600 series routers and
on the Cisco AS5300 universal access server.
Examples
Number from 1 to 5. The default value is 3.
The following example configures the system to try three times to load audio files:
ivr autoload retry 3
Related Commands
Command
Description
ivr prompt memory
Configures the maximum amount of memory that the dynamic audio files
(prompts) occupy in memory.
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ivr autoload mode
ivr autoload mode
To load files from TFTP to memory using either verbose or silent mode, use the ivr autoload mode
command in global configuration mode. To disable this function, use the no form of this command.
ivr autoload mode {verbose url location [retry number]} | {silent url location [retry number]}
no ivr autoload mode {verbose url location [retry number]} | {silent url location
[retry number]}
Syntax Description
verbose
Displays the file transfer activity to the console. This mode is recommended
while debugging.
url
Indicates that a URL is used to locate the index file that contains a list of all
available audio files.
location
Specifies the URL of the index file.
retry
(Optional) Specifies the number of times the system will try to transfer a file
when there are errors. This parameter applies to each file transfer.
number
(Optional) Number of times from 1 to 5. The default value is 3.
silent
Performs the file transfer in silent mode, meaning that no file transfer activity is
displayed to the console.
Defaults
Silent mode
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 and 3600 series routers and
on the Cisco AS5300 universal access server.
Usage Guidelines
The index file contains a list of audio files (URL) that can be downloaded from the TFTP server. Use
the ivr autoload command to download audio files from TFTP to memory. The command only starts up
a background process. The background process (loader) does the actual downloading of the files.
The background process first reads the index file from either Flash or TFTP. It parses the files line by
line looking for the URL. It ignores lines that start with # as comment lines. Once it has a correct URL,
it tries to read that.au file into memory and creates a media object. If there are any errors during the
reading of the file, it retries the configured number of times. If the mode is set to verbose, the loader
logs the transaction to console. Once parsing has reached the end of the index file, the background
process exits memory.
Perform the following checks before initiating the background process. If one of the checks fails, it
indicates the background process is not started, and instead you will see an error response to the
command.
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ivr autoload mode
•
Check if any prompt is being actively used (IVR is actively playing some prompts). If there are
active prompts, the command fails, displaying the following error message (.au files are also
referred to as prompts):
command is not allowed when prompts are active
•
Check if there is already a background process in progress. If there is a process, the command fails,
displaying the following error:
previous autoload command is still in progress
•
Check if an earlier ivr autoload command has already been configured. If an ivr autoload
command has already been configured, the user sees the following response when the command is
issued:
previous command is being replaced
•
When the no ivr autoload command is issued, if there was already an ivr autoload command in
progress, it will be aborted.
The audio files (prompts) loaded using the ivr autoload command are not dynamically swapped out of
memory. They are considered as autoloaded prompts as opposed to “dynamic” prompts. (See the
ivr prompt memory command for details on dynamic prompts.)
Examples
The following example configures verbose mode:
ivr autoload mode verbose url tftp://jurai/mgindi/tclware/index4 retry 3
The index file for the example index4 is shown as follows:
more index4
tftp://jurai/mgindi/tclware/au/en/en_one.au
tftp://jurai/mgindi/tclware/au/ch/ch_one.au
tftp://jurai/mgindi/tclware/au/ch/ch_one.au
The following is an example of index file on Flash:
flash:index
Related Commands
Command
Description
ivr prompt memory
Configures the maximum amount of memory that the dynamic audio files
(prompts) occupy in memory.
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ivr prompt memory
ivr prompt memory
To configure the maximum amount of memory that the dynamic audio files (prompts) occupy in
memory, use the ivr prompt memory command in global configuration mode. To disable the maximum
memory size, use the no form of this command.
ivr prompt memory size files number
no ivr prompt memory size files number
Syntax Description
size
Specifies the maximum memory to be used by the free dynamic prompts, in
kilobytes. Valid entries are from 128 to 16,384.
files number
Specifies the number of files that can stay in memory. Valid entries for the
number argument are 50 to 1000.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 and 3600 series routers and
on the Cisco AS5300 universal access server.
Usage Guidelines
When both the number and size parameters are specified, the minimum memory out of the two will be
used for memory calculations.
All the prompts that are not autoloaded or fixed are considered dynamic. Dynamic prompts are loaded
in to memory from TFTP or Flash, as and when they are needed. When they are actively used for playing
prompts, they are considered to be in “active” state. However, once the prompt playing is complete,
these prompts are no longer active and are considered to be in “free” state.
The free prompts either stay in memory or are removed from memory depending on the availability of
space in memory for these free prompts. The prompt-mem command essentially specifies a maximum
memory to be used for these free prompts.
The free prompts are saved in memory and are queued in a waitQ. When the waitQ is full (either because
the totally memory occupied by the free prompts exceeds the maximum configured value or the number
of files in the waitQ exceeds maximum configured), oldest free prompts are removed from memory.
Examples
The following example shows how to use the ivr prompt memory command:
ivr prompt memory 2048 files 500
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ivr prompt memory
Related Commands
Command
Description
ivr autoload
Loads files from a particular TFTP server.
show call
prompt-mem-usage
Displays the memory site use by prompts.
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line-power
line-power
To configure the BRI port to supply line power to the terminal equipment (TE), use the line-power
command in interface configuration mode. To disable the line power supply, use the no form of this
command.
line-power
no line-power
Syntax Description
This command has no arguments or keywords.
Defaults
The BRI port does not supply line power.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(3)XG
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(4)T
This command was integrated into the Cisco IOS Release 12.0(4)T.
12.1(3)XI
This command was added for the Cisco 2600 and Cisco 3600 series router.
Usage Guidelines
This command is supported only if an installed BRI voice module (BVM) or BRI VIC is equipped to
supply line power (phantom power).
This command is used only on a BRI port that is operating in NT mode. A BRI port that is operating in
TE mode is automatically disabled as a source of line power, and the line-power command is rejected.
When you use the line-power command, the line power provision is activated on a BRI port if the port
is equipped with the hardware to supply line power. When you enter the no line-power command, the
line power provision is deactivated on a BRI port.
Examples
The following example configures a BRI port to supply power to an attached TE device:
interface bri 1
line-power
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line-termination
line-termination
To set the line termination on an E1 controller, use the line-termination command in controller
configuration mode. To restore the default value, use the no form of this command.
line-termination { 75-ohm | 120-ohm }
no line-termination
Syntax Description
75-ohm
Matches the balanced BNC 75-ohm interface.
120-ohm
Matches the unbalanced twisted-pair 120-ohm interface.
Defaults
The default value is 120-ohm.
Command Modes
Controller configuration
Command History
Release
Modification
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS
Release 12.1(1)T.
Usage Guidelines
This command applies to E1 controllers only.
Examples
The following example shows how to set controller E1 0/0 to a line-termination of 75-ohm:
controller e1 0/0
line-termination 75-ohm
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link (RLM)
link (RLM)
To enable a Redundant Link Manager (RLM) link, use the link command in RLM configuration mode.
To disable this function, use the no form of this command.
link {hostname name | address ip-address} source loopback-source weight factor
no link {hostname name | address ip-address} source loopback-source weight factor
Syntax Description
hostname name
RLM host name. If host name is used, RLM will look up the DNS server
periodically for the host name configured until lookup is successful or the
configuration is removed.
address ip-address
IP address of the link.
source
loopback-source
Loopback interface source. We recommend you use the loopback interface
as the source, so that it is independent of the hardware condition. Also, the
source interface should be different in every link to avoid falling back to the
same routing path. If you intend to use the same routing path for the failover,
a single link is sufficient to implement it.
weight factor
An arbitrary number that sets link preference. The higher the weighting factor
number assigned, the higher priority it gets to become the active link. If all
entries have the same weighting factor assigned, all links will be treated
equally. There is no preference among servers according to the assumption
that only one server will accept the connection requests at any given time.
Otherwise, preferences are extended across all servers.
Defaults
Disabled
Command Modes
RLM configuration
Command History
Release
Modification
11.3(7)
This command was introduced.
Usage Guidelines
This command is a preference-weighted multiple entries command. Within the same server, the link
preference is specified in weighting.
Examples
The following example specifies the RLM group (network access server), the device name, and the link
addresses and their weighting preferences:
rlm group 1
server r1-server
link address 10.1.4.1 source Loopback1 weight 4
link address 10.1.4.2 source Loopback2 weight 3
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link (RLM)
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
interface
Defines the IP addresses of the server, configures an interface type, and
enters interface configuration mode.
protocol rlm port
Reconfigures the port number for the basic RLM connection for the
whole rlm-group.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the
link is declared down.
server (RLM)
Defines the IP addresses of the server.
show rlm group statistics
Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
show rlm group timer
Displays the current RLM group timer values.
shutdown (RLM)
Shuts down all of the links under the RLM group.
timer
Overwrites the default setting of timeout values.
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protocol rlm port
protocol rlm port
To configure the RLM port number, use the protocol rlm port RLM configuration command. To disable
this function, use the no form of this command.
protocol rlm port port-number
no protocol rlm port port-number
Syntax Description
port-number
Defaults
3000
Command Modes
RLM configuration
Command History
Release
Modification
11.3(7)
This command was introduced.
Usage Guidelines
RLM port number. See Table 87 for the port number choices.
The port number for the basic RLM connection can be reconfigured for the entire RLM group. Table 87
lists the default RLM port numbers.
Table 25
Default RLM Port Number
Protocol
Port Number
RLM
3000
ISDN
Port[RLM]+1
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protocol rlm port
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
interface
Defines the IP addresses of the server, configures an interface
type, and enters interface configuration mode.
link (RLM)
Specifies the link preference.
retry keepalive
Allows consecutive keepalive failures a certain amount of time
before the link is declared down.
server (RLM)
Defines the IP addresses of the server.
show rlm group statistics
Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
show rlm group timer
Displays the current RLM group timer values.
shutdown (RLM)
Shuts down all of the links under the RLM group.
timer
Overwrites the default setting of timeout values.
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loopback
loopback
To set the loopback method for testing a T1 or E1 interface, use the loopback command in controller
configuration mode. To restore the default value, use the no form of this command.
loopback {diagnostic | local {payload | line} | remote {v54 channel-group channel-number | iboc
| esf {payload | line}}}
no loopback
Syntax Description
diagnostic
Loops the outgoing transmit signal back to the receive signal.
local
Places the interface into local loopback mode.
payload
Places the interface into external loopback mode at the payload level.
line
Places the interface into external loopback mode at the line level.
remote
Keeps the local end of the connection in remote loopback mode.
v54 channel-group
Activates a V.54 channel-group loopback at the remote end. Available for
both T1 and E1 facilities.
channel-number
Specifies the channel number range (from 0 to 1) for the V.54 channel-group
loopback.
iboc
Sends an in band bit oriented code to the far end to cause it to go into line
loopback.
esf
Specifies Extended Super Frame (ESF) as the T1 or E1 frame type. Only
available under T1 or E1 controllers when ESF is configured on the
controller. The following are keywords:
•
payload—Activates remote payload loopback by sending Facility Data
Link (FDL) code. FDL is a 4-kbps out-of-band signaling channel in ESF.
•
line—Activates remote line loopback by sending FDL code.
Defaults
No loopback is configured.
Command Modes
Controller configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced as a controller configuration command for the
Cisco MC3810 multiservice concentrator.
12.0(5)T and
12.0(5)XK
The command was introduced as an ATM interface configuration command
for the Cisco 2600 and 3600 series router.
12.0(5)XE
The command was introduced as an ATM interface configuration command
for the Cisco 7200 and 7500 series.
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loopback
Release
Modification
12.0(5)XK and
12.0(7)T
The command was introduced as a controller configuration command for the
Cisco 2600 and 3600 series router.
12.1(1)T
The command was modified as a controller configuration command for the
Cisco 2600 series.
Usage Guidelines
You can use a loopback test on lines to detect and distinguish equipment malfunctions caused either by
the line and channel service unit/digital service unit (CSU/DSU) or by the interface. If correct data
transmission is not possible when an interface is in loopback mode, the interface is the source of the
problem.
Examples
The following example shows how to set the diagnostic loopback method on controller T1 0/0:
controller t1 0/0
loopback diagnostic
The following example shows how to set the payload loopback method on controller E1 0/0:
controller e1 0/0
loopback local payload
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loop-detect
loop-detect
To enable loop detection for T1, use the loop-detect command in controller configuration mode. To
cancel the loop detect operation, use the no form of this command.
loop-detect
no loop-detect
Syntax Description
This command has no arguments or keywords.
Defaults
Loop detection is disabled.
Command Modes
Controller configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
This command applies to Voice over Frame Relay and Voice over ATM on the Cisco MC3810
multiservice concentrator.
Examples
The following example configures loop detection for controller T1 0:
controller t1 0
loop-detect
Related Commands
Command
Description
loopback (interface)
Diagnoses equipment malfunctions between an interface and a device.
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loss-plan
loss-plan
To specify the analog-to-digital gain offset for an analog Foreign Exchange Office (FXO) or Foreign
Exchange Station (FXS) voice port, use the loss-plan command in voice-port configuration mode. To
restore the default value, use the no form of this command.
loss-plan {plan1 | plan2 | plan3 | plan4 | plan5 | plan6 | plan7 | plan8 | plan9}
no loss-plan
Syntax Description
plan1
FXO: A-D gain = 0 dB, D-A gain = 0 dB.
FXS: A-D gain = –3 dB, D-A gain = –3 dB.
plan2
FXO: A-D gain = 3 dB, D-A gain = 0 dB.
FXS: A-D gain = 0 dB, D-A gain = –3 dB.
plan3
FXO: A-D gain = –3 dB, D-A gain = 0 dB.
FXS: Not applicable.
plan4
FXO: A-D gain = –3 dB, D-A gain = –3 dB.
FXS: Not applicable.
plan5
FXO: Not applicable.
FXS: A-D gain = –3 dB, D-A gain = –10 dB.
plan6
FXO: Not applicable.
FXS: A-D gain = 0 dB, D-A gain = –7 dB.
plan7
FXO: A-D gain = 7 dB, D-A gain = 0 dB.
FXS: A-D gain = 0 dB, D-A gain = –6 dB.
plan8
FXO: A-D gain = 5 dB, D-A gain = –2 dB.
FXS: Not applicable.
plan9
FXO: A-D gain = 6 dB, D-A gain = 0 dB.
FXS: Not applicable.
Defaults
FXO: A-D gain = 0 dB, D-A gain = 0 dB (loss plan 1)
FXS: A-D gain = –3 dB, D-A gain = –3 dB (loss plan 1)
Command Modes
Voice-port configuration
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loss-plan
Command History
Usage Guidelines
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
The following additional signal level choices were added: plan 3, plan 4,
plan 8, and plan 9.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
The loss-plan command sets the analog signal level difference (offset) between the analog voice port
and the digital signal processor (DSP). Each loss plan specifies a level offset in both directions—from
the analog voice port to the DSP (A-D) and from the DSP to the analog voice port (D-A).
Use this command to obtain the required levels of analog voice signals to and from the DSP.
The loss-plan command is supported only on Cisco MC3810 multiservice concentrators, on FXO and
FXS analog voice ports.
Examples
The following example configures FXO voice port 1/6 for a –3 dB offset from the voice port to the DSP
and for a 0 dB offset from the DSP to the voice port:
voice-port 1/6
loss-plan plan3
The following example configures FXS voice port 1/1 for a 0 dB offset from the voice port to the DSP
and for a –7 dB offset from the DSP to the voice port:
voice-port 1/1
loss-plan plan6
Related Commands
Command
Description
impedance
Specifies the terminating impedance of a voice port interface.
input gain
Specifies the gain applied by a voice port to the input signal from the PBX
or other customer premises equipment.
output attenuation
Specifies the attenuation applied by a voice port to the output signal
toward the PBX or other customer premises equipment.
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lrq forward-queries
lrq forward-queries
To enable a gatekeeper to forward Location Requests (LRQs) that contain E.164 addresses that match
zone prefixes controlled by remote gatekeepers, use the lrq forward-queries command in gatekeeper
configuration mode. To disable this function, use the no form of this command.
lrq forward-queries
no lrq forward-queries
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 2500, 3600 series routers and on
the Cisco MC3810 multiservice concentrator.
Usage Guidelines
LRQ forwarding is dependent on a Cisco nonstandard field that first appeared in Cisco IOS
Release 12.0(3)T. This means that any LRQ received from a non-Cisco gatekeeper or any gatekeeper
running a Cisco IOS software image prior to Cisco IOS Release 12.0(3)T will not be forwarded.
The routing of E.164-addressed calls is dependent on the configuration of zone prefix tables (for
example, area code definitions) on each gatekeeper. Each gatekeeper is configured with a list of prefixes
controlled by itself and by other remote gatekeepers. Calls are routed to the zone that manages the
matching prefix. Thus, in the absence of a directory service for such prefix tables, you, the network
administrator, may have to define extensive lists of prefixes on all the gatekeepers in your administrative
domain.
To simplify this task, you can select one of your gatekeepers as the “directory” gatekeeper and configure
that gatekeeper with the complete list of prefixes and the lrq forward-queries command. You can then
simply configure all the other gatekeepers with their own prefixes and the wildcard prefix “*” for your
directory gatekeeper.
This command affects only the forwarding of LRQs for E.164 addresses. LRQs for H.323-ID addresses
are never forwarded.
Examples
The following example shows how this command is used to simplify configuration by selecting one
gatekeeper as the directory gatekeeper. Refer to Figure 5.
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lrq forward-queries
Figure 5
Example Scenario with Directory Gatekeeper and Two Remote Gatekeepers
gk-directory
gk-west
1. End point or gateway in
zone gk-west makes a
call to 12125551234
3. gk-directory knows that gk-east
manages 12125551234, so the
LRQ is forwarded to gk-east
gk-east
4. gk-east responds directly
to gk-west
18342
2. gk-west sends an LRQ
for 12125551234 to
gk-directory
Configuration on gk-directory
On the directory gatekeeper called gk-directory, identify all the prefixes for all the gatekeepers in your
administrative domain:
zone local gk-directory cisco.com
zone remote gk-west cisco.com 172.16.1.1
zone remote gk-east cisco.com 172.16.2.1
zone
zone
zone
zone
prefix
prefix
prefix
prefix
gk-west
gk-west
gk-west
gk-west
1408.......
1415.......
1213.......
1650.......
zone prefix gk-east 1212.......
zone prefix gk-east 1617.......
lrq forward-queries
Configuration on gk-west
On the gatekeeper called gk-west, configure all the locally managed prefixes for that gatekeeper:
zone local gk-west cisco.com
zone remote gk-directory cisco.com 172.16.2.3
zone
zone
zone
zone
zone
prefix
prefix
prefix
prefix
prefix
gk-west 1408.......
gk-west 1415.......
gk-west 1213.......
gk-west 1650.......
gk-directory *
Configuration on gk-east
On the gatekeeper called gk-east, configure all the locally managed prefixes for that gatekeeper:
zone local gk-east cisco.com
zone remote gk-directory cisco.com 172.16.2.3
zone prefix gk-east 1212.......
zone prefix gk-east 1617.......
zone prefix gk-directory *
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lrq forward-queries
Now when an endpoint or gateway in zone gk-west makes a call to 12125551234, gk-west will send an
LRQ for that E.164 address to gk-directory, which forwards the LRQ to gk-east. Gatekeeper gk-east
responds directly to gk-west.
Related Commands
Command
Description
lrq reject-unknown-prefix
Enables the gatekeeper to reject all LRQs for zone prefixes that are not
configured.
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lrq reject-unknown-prefix
lrq reject-unknown-prefix
To enable the gatekeeper to reject all Location Requests (LRQs) for zone prefixes that are not
configured, use the lrq reject-unknown-prefix command in gatekeeper configuration mode. To
reenable the gatekeeper to accept and process all incoming LRQs, use the no form of this command.
lrq reject-unknown-prefix
no lrq reject-unknown-prefix
Syntax Description
This command has no arguments or keywords.
Defaults
The gatekeeper accepts and processes all incoming LRQs.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 and 3600 series routers.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
Use the lrq reject-unknown-prefix command to configure the gatekeeper to reject any incoming LRQs
for a destination E.164 address that does not match any of the configured zone prefixes.
Whether or not you enable the lrq reject-unknown-prefix command, the following is true when the
E.164 address matches a zone prefix:
•
If the matching zone prefix is local (that is, controlled by this gatekeeper), the LRQ is serviced.
•
If the matching zone prefix is remote (that is, controlled by some other gatekeeper), the LRQ is
rejected.
If you do not enable the lrq reject-unknown-prefix command and the target address does not match any
known local or remote prefix, the default behavior is to attempt to service the call using one of the local
zones. If this default behavior is not suitable for your site, configure the lrq reject-unknown-prefix
command on your router to force the gatekeeper to reject such requests.
Examples
Consider the following gatekeeper configuration:
zone local gk408 cisco.com
zone local gk415 cisco.com
zone prefix gk408 1408.......
zone prefix gk415 1415.......
lrq reject-unknown-prefix
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lrq reject-unknown-prefix
In this sample configuration, the gatekeeper is configured to manage two zones. One zone contains
gateways with interfaces in the 408 area code, and the second zone contains gateways in the 415 area
code. Then using the zone prefix command, the gatekeeper is configured with the appropriate prefixes
so that calls to those area codes hop off in the optimal zone.
Now say some other zone has been erroneously configured to route calls to the 212 area code to this
gatekeeper. When the LRQ for a number in the 212 area code arrives at this gatekeeper, the gatekeeper
fails to match the area code, and the LRQ is rejected.
If this was your only site that had any gateways in it and you wanted your other sites to route all calls
that require gateways to this gatekeeper, you can undo the lrq reject-unknown-prefix command by
simply using the no lrq reject-unknown-prefix command.Now when the gatekeeper receives an LRQ
for the address 12125551234, it will attempt to find an appropriate gateway in either one of the zones
gk408 or gk415 to service the call.
Related Commands
Command
Description
lrq forward-queries
Enables a gatekeeper to forward LRQs that contain E.164 addresses
that match zone prefixes controlled by remote gatekeepers.
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lrq timeout blast window
lrq timeout blast window
To configure the timeout window for use when sending multiple Location Requests (LRQs) (either
sequentially or simultaneously), use the lrq timeout blast window command in gatekeeper
configuration mode. To return to the default value, use the no form of this command.
lrq timeout blast window seconds
no lrq timeout blast window
Syntax Description
seconds
Defaults
By default, the duration of the window is 6 seconds.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 2500, 2600, 3600, and
7200 series routers and on the Cisco MC3810 multiservice
concentrator.
Examples
The duration of the window, in seconds. Possible values are
1 through 10. The default is 6.
The following example sets the window to 3 seconds:
lrq timeout blast window 3
Related Commands
Command
Description
gatekeeper gw-type-prefix
Sets the gatekeepers responsible for each technology prefix.
zone prefix
Adds a prefix to a gatekeeper’s zone list.
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lrq timeout seq delay
lrq timeout seq delay
To configure the delay for use when sending Location Requests (LRQs) sequentially, use the lrq
timeout seq delay command in gatekeeper configuration mode. To return to the default value, use this
no form of the command.
lrq timeout seq delay value
no lrq timeout seq delay
Syntax Description
value
Defaults
By default, the duration of the window is five 100 millisecond units (500 ms or 0.5 seconds)
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(2)T
This command was introduced on the Cisco 2500, 2600, 3600, and
7200 series routers, and on the Cisco MC3810 multiservice
concentrator.
Examples
The duration of the delay, in 100 millisecond units. Possible values
are 1 through 10. The default is 5 (500 ms or 0.5 seconds).
The following example sets the window to 300 milliseconds:
lrq timeout seq delay 3
Related Commands
Command
Description
gatekeeper gw-type-prefix
Sets the gatekeepers responsible for each technology prefix.
zone prefix
Adds a prefix to a gatekeeper’s zone list.
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max-conn
max-conn
To specify the maximum number of allowed connections for a particular Voice over IP (VoIP) or POTS
dial peer, use the max-conn command in dial-peer configuration mode. To set an unlimited number of
connections for this dial peer, use the no form of this command.
max-conn number
no max-conn
Syntax Description
number
Defaults
The no form of this command is the default, meaning an unlimited number of connections.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series roter.
12.0(4)XJ
This command was modified for store and forward fax on the Cisco AS5300
universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Specifies the maximum number of connections for this dial peer. Valid values
for this field are 1 to 2,147,483,647.
This command applies to both VoIP and POTS dial peers. Use the max-conn command to define the
maximum number of connections used simultaneously on the Cisco AS5300 universal access server to
send fax-mail.
This command applies to off-ramp store and forward fax functions.
Examples
The following example configures the maximum number of connections for VoIP dial peer 10 as 5:
dial-peer voice 10 voip
max-conn 5
Related Commands
Command
Description
mta receive
maximum-recipients
Specifies the maximum recipients for all SMTP connections.
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max-connection
max-connection
To set the maximum number of simultaneous connections to be used for communication with a
settlement provider, use the max-connection command in settlement configuration mode. To reset to
the default value of this command, use the no form of this command.
max-connection number
no max-connection number
Syntax Description
number
Defaults
The default is 10 connections.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 and Cisco 3600 series
routers and on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Examples
Specifies the maximum number of HTTP connections to a settlement
provider.
The following command sets the maximum number of simultaneous connections to 10:
settlement 0
max-connection 10
Related Commands
Command
Description
connection-timeout
Configures the time that a connection is maintained after completing a
communication exchange.
customer-id
Sets the customer identification.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
response-timeout
Configures the maximum time to wait for a response from a server.
retry-delay
Sets the time between attempts to connect with the settlement provider.
retry-limit
Sets the maximum number of connection attempts to the provider.
session-timeout
Sets the interval for closing the connection when there is no input or output
traffic.
settlement
Enters settlement configuration mode and specifies the attributes specific to
a settlement provider.
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max-connection
Command
Description
shutdown
Brings up the settlement provider.
type
Configures an SAA-RTR operation type.
url
Configures the ISP address.
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max-forwards
max-forwards
To set the maximum number of proxy or redirect servers that can forward the request, use the
max-forwards command in the SIP user agent configuration mode. To reset the default value, use the
no form of this command.
max-forwards number
no max-forwards
Syntax Description
number
Defaults
The default number of hops is 6.
Command Modes
SIP user agent configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and Cisco 3600 series routers
and on the Cisco AS5300 universal access server.
Number of hops. Possible values are 1 through 15. The default is 6.
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
The following is an example of forwarding requests to proxy or redirect servers:
sip-ua
max-forwards 2
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max-redirects
max-redirects
To set the maximum number of redirect servers that the user agent allows, use the max-redirects
command in dial-peer configuration mode. To reset the default value, use the no form of this command.
max-redirects number
no max-redirects
Syntax Description
number
Defaults
The default number of redirects is 1.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 and Cisco 3600 series
routers and on the Cisco AS5300 universal access server.
Examples
Maximum number of redirect servers that a call can traverse. Possible values
are 1 through 10.
The following is an example of setting the maximum number of redirect servers that the user agent
allows:
dial-peer voice 102 voip
max-redirects 2
Related Commands
Command
Description
dial-peer voice
Enters dial-peer configuration mode and specifies the method of
voice-related encapsulation.
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mdn
mdn
To request that a message disposition notice (MDN) be generated when the message is processed
(“opened”), use the mdn command in dial-peer configuration mode. To restore the default value, use the
no form of this command.
mdn
no mdn
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access router.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Message disposition notification is an e-mail message that is generated and sent to the sender when the
message is opened by the receiver. Use the mdn command to request that an e-mail response message
be sent to the sender when the e-mail that contains the fax TIFF image has been opened.
This command applies to on-ramp store and forward fax functions.
Examples
The following example requests that a message disposition notice be generated by the recipient:
dial-peer voice 10 mmoip
mdn
Related Commands
Command
Description
mta receive
generate-mdn
Specifies that the off-ramp gateway process a response MDN from an SMTP
server.
mta send
return-receipt-to
Specifies the address where MDNs are sent.
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mgcp
mgcp
To allocate resources for the media gateway control protocol (MGCP) and start the MGCP daemon, use
the mgcp command in global configuration mode. To terminate all calls, release all allocated resources,
and stop the MGCP daemon, use the no form of this command.
mgcp [port]
no mgcp
Syntax Description
port
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access
server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
Usage Guidelines
(Optional) Specifies a UDP port for the MGCP gateway. Valid values are
1025 through 65,535. If no port is specified, the command defaults to UDP
port 2427.
Once you start the MGCP daemon using the mgcp command, you can suspend it (for example, for
maintenance) by using the mgcp block-newcalls command. When you are ready to resume normal
MGCP operations, use the no mgcp block-newcalls command. Use the no mgcp command only if you
intend to terminate all MGCP applications and protocols.
When the MGCP daemon is not active, all MGCP messages are ignored.
Examples
The following example shows how to initiate the MGCP daemon:
mgcp
Related Commands
Command
Description
debug mgcp
Enables debugging on MGCP.
mgcp ip-tos
Terminates all MGCP activity in an orderly manner.
mgcp request retries
Specifies the number of times to retry sending the mgcp command.
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mgcp block-newcalls
mgcp block-newcalls
To block new calls while maintaining existing calls, use the mgcp block-newcalls command in global
configuration mode. To resume media gateway control protocol (MGCP) operation, use the no form of
this command.
mgcp block-newcalls
no mgcp block-newcalls
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
Usage Guidelines
This command is valid only if the mgcp command is enabled.
Once you issue this command, all requests for new connections (CreateConnection requests) are denied.
All existing calls will be maintained until participants terminate them or you use the no mgcp command.
When the last active call is terminated, the MGCP daemon will be terminated and all resources allocated
to it will be released. The no mgcp block-newcalls command returns the router to normal MGCP
operations.
Examples
The following example shows how to prevent the gateway from receiving new calls:
mgcp block-newcalls
Related Commands
Command
Description
mgcp
Allocates resources for the MGCP and starts the daemon.
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mgcp call-agent
mgcp call-agent
To configure the call agent (media gateway controller) address, use the mgcp call-agent command in
global configuration mode. To unconfigure the call agent address, use the no form of this command.
mgcp call-agent {ip-address | host-name} [port] [service-type type]
no mgcp call-agent
Syntax Description
ip-address | host-name
Specifies the IP address or domain name of the call agent.
port
(Optional) Specifies the User Datagram Protocol (UDP) port for the call
agent to use. Valid values are 1025 through 65,535. If a port is not specified,
the default is UDP 2427.
service-type type
(Optional) Specifies the type of gateway control service to be supported by
the call agent. Valid values are mgcp and sgcp.
Defaults
MGCP service-type
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
The service-type type option was added.
Usage Guidelines
Use this command on any platform and media gateway.
If you do not specify a UDP port from the command line, media gateway control protocol (MGCP) will
use 2427 as the default call agent UDP port.
When service-type is set to mgcp, the call agent processes the Restart in Progress (RSIP) error messages
sent by the gateway. When service-type is sent to sgcp, the call agent ignores the RSIP messages.
Examples
The following examples illustrate several formats for specifying the call agent (use any one of these
formats):
mgcp
mgcp
mgcp
mgcp
Related Commands
call-agent
call-agent
call-agent
call-agent
209.165.200.225 service-type sgcp
209.165.200.225 5530 service-type mgcp
igloo service-type sgcp
igloo 2009 service-type mgcp
Command
Description
mgcp
Starts the MGCP daemon.
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mgcp codec
mgcp codec
To select the default codec type and its optional packetization period value, use the mgcp codec
command in global configuration mode. To set the parameters to their default values, use the no form
of this command.
mgcp codec type [packetization-period value]
no mgcp codec
Syntax Description
type
Specifies the types of codec supported. Valid codecs are G711alaw,
G711ulaw, G723ar53, G723ar63, G723r53, G723r63, G729ar8, G729br8,
and G729r8.
packetization-period
value
(Optional) This parameter is useful when the preferred compression
algorithm and packetization period parameter is not provided by the Media
Gateway Controller. The value range depends on the type of codec selected.
For example, the range for G729r8 is 10 to 220 in increments of 10. For
G711ulaw, the range is 10 to 20 in increments of 10. For G723ur53, the range
is 30 to 330 in increments of 10.
Defaults
G711ulaw
Command Modes
Global configuration
Command History
Release
Examples
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
This example shows how to specify the default codec type:
mgcp codec g711alaw
The following example specifies the codec type and sets the packetization period:
mgcp codec g729r8 packetization-period 150
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
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mgcp default-package
mgcp default-package
To configure the default package capability type for the media gateway, use the mgcp default-package
command in global configuration mode. This command does not support a no form. To change the
default package, use the mgcp default-package command with a different, actively supported package.
Residential Gateways
mgcp default-package {line-package | dtmf-package | gm-package}
Trunking Gateways
mgcp default-package {as-package | dtmf-package | gm-package | rtp-package |
trunk-package}
Syntax Description
Defaults
as-package
Announcement server package.
dtmf-package
DTMF package.
gm-package
Generic media package.
line-package
Line package.
rtp-package
RTP package.
trunk-package
Trunk package.
For residential gateways (RGWs): line-package
For trunking gateways (TGWs): trunk-package
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
The line-package keyword and a distinction between residential and
trunking gateways were added.
Usage Guidelines
This command is helpful when the Media Gateway Controller does not provide the package capability
to be used for the given connection.
Before selecting a package as the default, use the show mgcp command to ensure that the package is
actively supported. If the package you want does not appear in the display, use the
mgcp package-capability command to add the package to the supported list.
If only one package is actively supported, it becomes the default package.
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mgcp default-package
Examples
The following example shows how to set the default package:
mgcp default-package as-package
! The announcement server package type will be the new default package type.
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
mgcp package-capability Includes a specific MGCP package that is supported by the gateway.
Cisco IOS Voice, Video, and Fax Command Reference
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mgcp dtmf-relay
mgcp dtmf-relay
To ensure accurate forwarding of digits on compressed codecs, use the mgcp dtmf-relay command in
controller configuration mode. To disable this process for noncompressed codecs, use the no form of
this command.
mgcp dtmf-relay {codec | low-bit-rate} mode {cisco | out-of-band}
no mgcp dtmf-relay
Syntax Description
codec
Specifies use of either a G.711 or a G.726 codec.
low-bit-rate
Specifies a low-bit-rate codec other than G.711 and G.726.
mode
Specifies the mode.
cisco
This mode removes the DTMF tone from the voice stream and sends FRF.11
with a special payload 121 for the DTMF digits.
out-of-band
This mode removes the DTMF tone from the voice stream and does not send
FRF.11.
Defaults
Non compressed codecs are disabled.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(3)T
This command was first supported by media gateway control protocol
(MGCP) on the Cisco 2600 and Cisco 3600 series routers, the
Cisco AS5300 universal access server, and the Cisco uBR924 cable access
router.
Usage Guidelines
Use this command to access an announcement server or a voice mail server that does not have the
capability to decode RTP packets containing DTMF digits. When the mgcp dtmf-relay command is
active, the DTMF digits are removed from the voice stream and carried by FRF.11 so that the server can
decode the digits.
Examples
The following example shows how to remove the DTMF tone from the voice stream and send FRF.11
with a special payload for the DTMF digits:
mgcp dtmf-relay codec mode cisco
Related Commands
Command
Description
mgcp
Allocates resources of the MGCP and starts the daemon.
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mgcp ip-tos
mgcp ip-tos
To enable or disable the IP type of service (ToS) for MGCP-controlled connections, use the mgcp ip-tos
command in global configuration mode. To set the parameters to their default values, use the no form
of this command.
mgcp ip-tos {high-reliability | high-throughput | low-cost | low-delay | precedence value}
no mgcp ip-tos {high-reliability | high-throughput | low-cost | low-delay | precedence value}
Syntax Description
high-reliability
Specifies high-reliability ToS.
high-throughput
Specifies high-throughput ToS.
low-cost
Specifies low-cost ToS.
low-delay
Specifies low-delay ToS.
precedence value
Specifies the value of the IP precedence bit. Valid values are from 0 to 7.
The default IP precedence value is 3.
Defaults
Services disabled; precedence is 3.
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
Usage Guidelines
Only one of the high-reliability, high-throughput, low-cost, or low-delay parameters can be enabled at
any given time. Enabling one parameter disables any other that was active. Enabling one parameter has
no effect on the precedence parameter. When you configure a new value for the precedence parameter,
the old value is erased.
The no form of the mgcp ip-tos command disables the first four parameters and sets the precedence
parameter back to 3.
Examples
In the following example, activating the low-delay keyword disables the other three parameters.
mgcp
mgcp
mgcp
mgcp
mgcp
ip-tos
ip-tos
ip-tos
ip-tos
ip-tos
high-reliability
high-throughput
low-cost
low-delay
precedence 4
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mgcp ip-tos
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
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Cisco IOS Voice, Video, and Fax Commands: G Through P
mgcp max-waiting-delay
mgcp max-waiting-delay
To specify the media gateway control protocol (MGCP) maximum waiting delay (MWD), use the mgcp
max-waiting-delay command in global configuration mode. To restore the default value, use the no
form of this command.
mgcp max-waiting-delay milliseconds
no mgcp max-waiting-delay
Syntax Description
milliseconds
Defaults
3000 milliseconds
Command Modes
Global configuration
Command History
Release
The number of milliseconds to wait after restart. The valid range is 0 to
600,000 milliseconds (600 seconds).
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
Usage Guidelines
Use the maximum waiting delay to send out an RSIP message to the call agent with the restart method.
This command helps prevent traffic bottlenecks caused by MGCP gateways all trying to connect at the
same time after a restart.
Examples
The following example shows how to set the MGCP maximum waiting delay to 600 milliseconds:
mgcp max-waiting-delay 600
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
mgcp restart-delay
Configures the graceful tear down method sent in the RSIP message.
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mgcp modem passthru
mgcp modem passthru
To enable the gateway to send and receive modem and fax data, use the mgcp modem passthru
command in controller configuration mode. To disable support for modem and fax data, use the no form
of this command.
mgcp modem passthru {cisco | ca}
no mgcp modem passthru
Syntax Description
cisco
When the gateway detects a modem/fax tone, the gateway switches the
codec to G.711 to allow the analog data to pass through.
ca
When the gateway detects a modem/fax tone, the gateway alerts the call
agent to switch the codec to G.711 to allow the analog data to pass through.
Defaults
ca
Command Modes
Controller configuration
Command History
Release
Modification
12.1(3)T
This command was added to MGCP.
Usage Guidelines
When the cisco keyword is activated and the gateway detects a modem/fax tone, the gateway switches
the codec to G.711 then sends the analog data to a remote gateway. The remote gateway also switches
the codec on its side of the call to G.711 to allow the analog data to pass through.
When the ca keyword is activated and the gateway detects a modem/fax tone, the gateway alerts the call
agent to switch the codec to G.711 to allow the analog data to pass through. The call agent must send an
MDCX signal to the G.711 codec for successful data pass-through.
Examples
The following example configures a gateway to send and receive modem or fax data:
mgcp modem passthru cisco
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
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mgcp package-capability
mgcp package-capability
To specify a media gateway control protocol (MGCP) package capability for a gateway, use the mgcp
package-capability command in global configuration mode. To remove a specific MGCP package
capability from the list of capabilities, use the no form of this command.
All Residential Gateways
mgcp package-capability {line-package | dtmf-package | gm-package | rtp-package}
no mgcp package-capability {line-package | dtmf-package | gm-package | rtp-package}
Cisco AS5300 Universal Access Server
mgcp package-capability {trunk-package | dtmf-package | gm-package | rtp-package |
as-package | script-package}
no mgcp package-capability {trunk-package | dtmf-package | gm-package | rtp-package |
as-package | script-package}
All Other Trunking Gateways
mgcp package-capability {trunk-package | dtmf-package | gm-package | rtp-package |
as-package}
no mgcp package-capability {trunk-package | dtmf-package | gm-package | rtp-package |
as-package}
Syntax Description
Defaults
line-package
Line package.
trunk-package
Trunk package.
dtmf-package
DTMF package.
gm-package
Generic media package.
rtp-package
RTP package.
as-package
Announcement server package.
script-package
Script package.
For all residential gateways (RGWs): line-package
For all trunking gateways (TGWs): trunk-package
Command Modes
Global configuration
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mgcp package-capability
Command History
Usage Guidelines
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
The command was expanded to the Cisco uBR924, Cisco 2600 series router
router, and Cisco 3660 platforms. The line-package, rtp-package, and
script-package keywords were added.
Select packages that are supported by your call agent. Events specified in the MGCP messages from the
call agent must belong to one of the supported packages. Otherwise, the connection requests are refused
by the media gateway.
Use this command before specifying a default package using the mgcp default-package command.
Specify at least one package to have a default.
Enter each package as a separate command.
Examples
The following example shows how to specify an MGCP package capability for a gateway:
mgcp
mgcp
mgcp
mgcp
Related Commands
package-capability trunk-package
package-capability dtmf-package
package-capability script-package
default-package trunk-package
Command
Description
mgcp
Starts the MGCP daemon.
mgcp default-package Configures the default package capability type for the media gateway.
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mgcp playout
mgcp playout
To tune the jitter buffer packet size attempted for MGCP-controlled connections, use the mgcp playout
command in global configuration mode. To restore the default value, use the no form of this command.
mgcp playout {adaptive init-value min-value max-value | fixed init-value}
no mgcp playout {adaptive | fixed}
Syntax Description
Defaults
adaptive init-value
min-value max-value
Specifies a user-defined variable range (in milliseconds) for the jitter buffer
packet size. The range for each value is 4 to 250. The default values are:
init-value 60, min-value 4, and max-value 200. Note that init-value must be
between min-value and max-value.
fixed init-value
Specifies a fixed size (in milliseconds) for the jitter buffer packet size. Valid
values are from 4 to 250.
adaptive 60 4 200
No default for fixed.
Command Modes
Global configuration
Command History
Release
Examples
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
The following example illustrates a jitter buffer configuration with an initial playout of 100, a minimum
buffer size of 50, and a maximum buffer size of 150:
mgcp playout adaptive 100 50 150
The following example illustrates setting the jitter buffer to a fixed playout of 120:
mgcp playout fixed 120
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
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mgcp quality-threshold
mgcp quality-threshold
To set the jitter buffer size threshold, latency threshold, and packet-loss threshold parameters, use the
mgcp quality-threshold command in global configuration mode. To restore the default parameter
values, use the no form of this command.
mgcp quality-threshold {hwm-jitter-buffer value | hwm-latency value | hwm-packet-loss value
| lwm-jitter-buffer value | lwm-latency value | lwm-packet-loss value}
no mgcp quality-threshold {hwm-jitter-buffer | hwm-latency | hwm-packet-loss |
lwm-jitter-buffer | lwm-latency | lwm-packet-loss}
Syntax Description
hwm-jitter-buffer value
Specifies the high-water-mark jitter buffer size. Valid range is from 100 to
200 the default value is 150 milliseconds.
hwm-latency value
Specifies the high-water-mark latency value. Valid range is from 250 to
400 milliseconds, and the default value is 300 milliseconds.
hwm-packet-loss value
Specifies the high-water-mark packet-loss value. Valid range is from 5000
to 25,000 milliseconds, and the default value is 10000 milliseconds.
lwm-jitter-buffer value
Specifies the low-water-mark jitter buffer size. Valid range is from 4 to 60
milliseconds, and the default value is 30 milliseconds.
lwm-latency value
Specifies the low-water-mark latency value. Valid range is from 125 to
200 milliseconds, and the default value is 150 milliseconds.
lwm-packet-loss value
Specifies the low-water-mark packet-loss value. Valid range is from 1 to
3000 milliseconds, and the default value is 1000 milliseconds.
Defaults
The voice quality parameter defaults are (in milliseconds): hwm-jitter-buffer 150, hwm-latency 300,
hwm-packet-loss 10,000, lwm-jitter-buffer 30, lwm-latency 150, and lwm-packet-loss 1000.
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access
server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco
uBR924, and Cisco 2600 series router router platforms.
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mgcp quality-threshold
Usage Guidelines
The following parameters impact the quality of voice calls:
•
Jitter buffer (storage area containing active call voice packets that have been received from the
network and are waiting to be decoded and played)
•
Latency (network delay in sending/receiving packets)
•
Packet loss (number of packets lost per 100,000 packets for a given call)
For good voice quality, the system should perform below the lwm values. As the values go higher, voice
quality degrades. The system generates a report when the values go above the hwm levels. Set the hwm
and lwm values sufficiently apart so that you receive reports on poor performance, but not so close
together that you receive too much feedback.
Enter each parameter as a separate command.
Examples
The following example shows how the different parameters can be set to new values:
mgcp
mgcp
mgcp
mgcp
mgcp
mgcp
Related Commands
quality-threshold
quality-threshold
quality-threshold
quality-threshold
quality-threshold
quality-threshold
hwm-jitter-buffer 100
hwm-latency 250
hwm-packet-loss 5000
lwm-jitter-buffer 50
lwm-latency 200
lwm-packet-loss 20
Command
Description
mgcp
Starts the MGCP daemon.
mgcp playout
Tunes the jitter buffer packet size.
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mgcp request retries
mgcp request retries
To specify the number of times to retry sending the mgcp command, use the mgcp request retries
command in global configuration mode. To restore the default value, use the no form of this command.
mgcp request retries count
no mgcp request retries
Syntax Description
count
Defaults
Three times
Command Modes
Global configuration
Command History
Release
Specifies the number of times a Notify message is resent to the Call Agent
before the request is dropped. The valid range is 1 to 10.
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
Usage Guidelines
This command applies to a trunking gateway.
Examples
The following example shows that the system will try to send the mgcp command 10 times before
dropping the request:
mgcp request retries 10
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
mgcp request timeout Specifies how long the system waits for a response to a request.
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mgcp request timeout
mgcp request timeout
To specify how long the system waits for a response to a request, use the mgcp request timeout
command in global configuration mode. To restore the default value, use the no form of this command.
mgcp request timeout timeout
no mgcp request timeout
Syntax Description
timeout
Defaults
500 milliseconds
Command Modes
Global configuration
Command History
Release
Examples
Specifies the number of milliseconds to wait for a response to a request. The
valid range is 1 to 10,000 (10 seconds).
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco uBR924,
and Cisco 2600 series router router platforms.
The following example configures the system to wait 40 milliseconds for a reply to a request:
mgcp request timeout 40
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
mgcp request retries
Specifies the number of times to retry sending the mgcp command.
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mgcp restart-delay
mgcp restart-delay
To select the delay value sent in the Restart in Progress (RSIP) graceful tear down, use the mgcp
restart-delay command in global configuration mode. To restore the default value, use the no form of
this command.
mgcp restart-delay seconds
no mgcp restart-delay
Syntax Description
seconds
Defaults
Zero (0) seconds
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access
server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco
uBR924, and Cisco 2600 series router router platforms.
Specifies the restart delay value in seconds. The valid range is from 0
to 600.
Usage Guidelines
Use the restart value to send a RSIP message that indicates when the connection in the gateway will be
torn down.
Examples
The following example sets the restart delay to 30 seconds:
mgcp restart-delay 30
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
mgcp max-waiting-delay
Specifies the MGCP MWD after a restart.
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mgcp sdp simple
mgcp sdp simple
To initiate a subset of the SDP protocol, use the mgcp sdp simple command in controller configuration
mode. To return to the full set of SDP protocol fields, use the no form of this command.
mgcp sdp simple
no mgcp sdp simple
Syntax Description
This command has no arguments or keywords.
Defaults
no mgcp sdp simple
Command Modes
Controller configuration
Command History
Release
Modification
12.1(3)T
This command was added to the media gateway control protocol (MGCP).
Usage Guidelines
When the mgcp sdp simple command is enabled, the gateway will not generate three SDP fields: time,
session name, and other (username, session id, sdp version, network type, address type, or address).
Certain call agents require this modified SDP protocol to send data through the network.
Examples
The following example configures the modified SDP protocol:
mgcp sdp simple
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
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mgcp vad
mgcp vad
To set the default VAD parameter for the media gateway control protocol (MGCP), use the mgcp vad
command in global configuration mode. To disable the VAD parameter, use the no form of this
command.
mgcp vad
no mgcp vad
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced for the Cisco AS5300 universal access
server.
12.1(3)T
Support for this command was extended to the Cisco 3660, Cisco
uBR924, and Cisco 2600 series router router platforms.
Usage Guidelines
Use the MGCP VAD parameter to tell the MGCP gateway to turn silence suppression on or off.
Examples
The following example turns silence suppression on:
mgcp vad
Related Commands
Command
Description
mgcp
Starts the MGCP daemon.
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microcode reload controller
microcode reload controller
To reload the firmware and FPGA from the command-line interface (CLI) without reloading the
Cisco IOS image, use the microcode reload controller command in privileged EXEC mode.
microcode reload controller {t1 | e1} {x/y}
Syntax Description
t1
Specifies T1.
e1
Specifies E1.
x/y
Controller slot and unit numbers.
Defaults
No microcode reload activity is initiated.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(2)XH
This command was introduced on the Cisco 2600 and Cisco 3600 series
routers.
Usage Guidelines
Configurations such as channel groups, TDM connections, and loopbacks and so on in the running
configuration are restored after this command is issued. If either of the controller ports on the WIC is in
a looped state before this command is issued, the looped condition is dropped. You have to reinitiate the
loopbacks from the remote end by doing “no loop” and “loop remote” from the controller configuration.
If the BERT test is running, that test will be aborted for the microcode reload.
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mmoip aaa global-password
mmoip aaa global-password
To define a password to be used with CiscoSecure for Windows NT when using store and forward fax,
use the mmoip aaa global-password command in global configuration mode. To restore the default
state, use the no form of this command.
mmoip aaa global-password password
no mmoip aaa global-password password
Syntax Description
password
Defaults
No password is defined.
Command Modes
Global configuration
Command History
Release
Usage Guidelines
Character string used to define the password for CiscoSecure for Windows
NT to be used with store and forward fax. The maximum length is 64
alphanumeric characters.
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
CiscoSecure for Windows NT might require a separate password in order to complete authentication, no
matter what security protocol you use. This command defines the password to be used with CiscoSecure
for Windows NT. All records on the Windows NT server use this defined password.
This command applies to on-ramp store and forward fax functions when using a modem card. It is not
used with voice feature cards.
Examples
The following example defines a password (password) when CiscoSecure for Windows NT is used with
store and forward fax:
mmoip aaa global-password password
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mmoip aaa method fax accounting
mmoip aaa method fax accounting
To define the name of the method list to be used for authentication, authorization, and accounting (AAA)
accounting with store and forward fax, use the mmoip aaa method fax accounting command in global
configuration mode. To restore the default value, use the no form of this command.
mmoip aaa method fax accounting method-list-name
no mmoip aaa method fax accounting method-list-name
Syntax Description
method-list-name
Defaults
No AAA accounting method list is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Character string used to name a list of accounting methods to be used with store
and forward fax.
This command defines the name of the AAA accounting method list to be used with store and forward
fax. The method list itself, which defines the type of accounting services provided for store and forward
fax, is defined using the aaa accounting global configuration command. Unlike standard AAA (where
each defined method list can be applied to specific interfaces and lines), the AAA accounting method
lists used in store and forward fax are applied globally on the Cisco AS5300 universal access server.
After the accounting method lists have been defined, they are enabled by using the mmoip aaa
receive-accounting enable command.
This command applies to both on-ramp and off-ramp store and forward fax functions when using a
modem card. It is not used with voice feature cards.
Examples
The following example defines a AAA accounting method list (called sherman) to be used with store
and forward fax:
aaa new-model
mmoip aaa method fax accounting sherman
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mmoip aaa method fax accounting
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security
purposes when you use RADIUS or TACACS+.
mmoip aaa
receive-accounting
enable
Enables on-ramp store and forward fax AAA accounting services.
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mmoip aaa method fax authentication
mmoip aaa method fax authentication
To define the name of the method list to be used for authentication, authorization, and accounting (AAA)
authentication with store and forward fax, use the mmoip aaa method fax authentication command in
global configuration mode. To restore the default state, use the no form of this command.
mmoip aaa method fax authentication method-list-name
no mmoip aaa method fax authentication method-list-name
Syntax Description
method-list-name
Defaults
No AAA authentication method list is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Character string used to name a list of authentication methods to be used with
store and forward fax.
This command defines the name of the AAA authentication method list to be used with store and forward
fax. The method list itself, which defines the type of authentication services provided for store and
forward fax, is defined using the aaa authentication global configuration command. Unlike standard
AAA (where each defined method list can be applied to specific interfaces and lines), AAA
authentication method lists used with store and forward fax are applied globally on the Cisco AS5300
universal access server.
After the authentication method lists have been defined, they are enabled by using the mmoip aaa
receive-authentication enable command.
This command applies to both on-ramp and off-ramp store and forward fax functions.
Examples
The following example defines a AAA authentication method list (called xyz) to be used with store and
forward fax:
aaa new-model
mmoip aaa method fax authentication xyz
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mmoip aaa method fax authentication
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security
purposes when you use RADIUS or TACACS+.
mmoip aaa
Enables on-ramp store and forward fax AAA authentication services.
receive-authentication
enable
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mmoip aaa receive-accounting enable
mmoip aaa receive-accounting enable
To enable on-ramp authentication, authorization, and accounting (AAA) accounting services, use the
mmoip aaa receive-accounting enable command in global configuration mode. To restore the default
state, use the no form of this command.
mmoip aaa receive-accounting enable
no mmoip aaa receive-accounting enable
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1T)
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This command enables AAA accounting services if a AAA accounting method list has been defined
using both the aaa accounting command and the mmoip aaa method fax accounting command.
This command applies to on-ramp store and forward fax functions.
Examples
The following example enables a AAA accounting method list (called xyz) to be used with inbound store
and forward fax. In this example, store and forward fax is being configured to track start and stop
connection accounting records.
aaa new-model
mmoip aaa method fax accounting xyz
aaa accounting connection sherman stop-only radius
mmoip aaa receive-accounting enable
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security
purposes when you use RADIUS or TACACS+.
mmoip aaa method
fax accounting
Defines the name of the method list to be used for AAA accounting with
store and forward fax.
Cisco IOS Voice, Video, and Fax Command Reference
VR-464
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa receive-authentication enable
mmoip aaa receive-authentication enable
To enable on-ramp authentication, authorization, and accounting (AAA) authentication services, use the
mmoip aaa receive-authentication enable command in global configuration mode. To restore the
default state, use the no form of this command.
mmoip aaa receive-authentication enable
no mmoip aaa receive-authentication enable
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This command enables AAA authentication services if an AAA authentication method list has been
defined using both the aaa authentication command and the mmoip aaa method fax authentication
command.
This command applies to on-ramp store and forward fax functions.
Examples
The following example enables a AAA authentication method list (called xyz) to be used with inbound
store and forward fax. In this example, RADIUS authentication (and if the RADIUS server fails, then
local authentication) is being configured for store and forward fax.
aaa new-model
mmoip aaa method fax authentication xyz
aaa authentication login peabody radius local
mmoip aaa receive-authentication enable
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security
purposes when you use RADIUS or TACACS+.
mmoip aaa method
fax authentication
Defines the name of the method list to be used for AAA authentication with
store and forward fax.
Cisco IOS Voice, Video, and Fax Command Reference
VR-465
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa receive-id primary
mmoip aaa receive-id primary
To specify the primary location where the authentication, authorization, and accounting (AAA) protocol
retrieves its account identification information for on-ramp faxing, use the mmoip aaa receive-id
primary command in global configuration mode. To restore the default state, which means that the
account identification source is undefined, use the no form of this command.
mmoip aaa receive-id primary {ani | dnis | gateway | redialer-id | redialer-dnis}
no mmoip aaa receive-id primary {ani | dnis | gateway | redialer-id | redialer-dnis}
Syntax Description
ani
Indicates that AAA uses the calling party telephone number (automatic number
identification or ANI) as the AAA account identifier.
dnis
Indicates that AAA uses the called party telephone number (dialed number
identification service or DNIS) as the AAA account identifier.
gateway
Indicates that AAA uses the router-specific name derived from the host name
and domain name as the AAA account identifier, displayed in the following
format: router-name.domain-name.
redialer-id
Indicates that AAA uses the account string returned by the external redialer
device as the AAA account identifier. In this case, the redialer ID is either the
redialer serial number or the redialer account number.
redialer-dnis
Indicates that AAA uses the called party telephone number (dialed number
identification service or DNIS) as the AAA account identifier captured by the
redialer if a redialer device is present.
Defaults
No account identification source is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Normally, when AAA is being used for simple user authentication, AAA uses the username information
defined in the user profile for authentication. With store and forward fax, you can specify that the ANI,
DNIS, gateway ID, redialer ID, or redialer DNIS be used to identify the user for authentication. This
command defines what AAA uses for the primary identifier for inbound or on-ramp user authentication
with store and forward fax.
Store and forward fax allows you to define either a primary or a secondary identifier. (You configure the
secondary identifier using the mmoip aaa receive-id secondary command.)
Cisco IOS Voice, Video, and Fax Command Reference
VR-466
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa receive-id primary
AAA does not use these methods sequentially—meaning that if the primary identifier is defined and
AAA cannot authenticate the primary identifier information, it will not use the secondary identifier for
authentication. Authentication simply fails.
Defining only the secondary identifier enables you to service two different scenarios
simultaneously—for example, if you are offering fax services to two different companies, one of which
uses redialers and the other does not. In this case, configure the mmoip aaa receive-id primary
command to use the redialer DNIS, and configure the mmoip aaa receive-id secondary command to
use ANI. With this configuration, when a user dials in and the redialer DNIS is not null, the redialer
DNIS is used as the authentication identifier. If a user dials in and the redialer DNIS is null, ANI is used
as the authentication identifier.
This command applies to on-ramp store and forward fax functions.
Examples
The following example defines the DNIS captured by the redialer as the AAA authentication identifier
for store and forward fax:
aaa new-model
mmoip aaa receive-id primary redialer-dnis
Related Commands
Command
Description
mmoip aaa receive-id
secondary
Specifies the secondary location where AAA retrieves its account
identification information for on-ramp faxing if the primary identifier has
not been defined.
Cisco IOS Voice, Video, and Fax Command Reference
VR-467
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa receive-id secondary
mmoip aaa receive-id secondary
To specify the secondary location where the authentication, authorization, and accounting (AAA)
protocol retrieves its account identification information for on-ramp faxing if the primary identifier has
not been defined, use the mmoip aaa receive-id secondary command in global configuration mode. To
restore the default state, which means that the account identification source is undefined, use the no form
of this command.
mmoip aaa receive-id secondary {ani | dnis | gateway | redialer-id | redialer-dnis}
no mmoip aaa receive-id secondary {ani | dnis | gateway | redialer-id | redialer-dnis}
Syntax Description
ani
Indicates that AAA uses the calling party telephone number (automatic number
identification or ANI) as the AAA account identifier.
dnis
Indicates that AAA uses the called party telephone number (dialed number
identification service or DNIS) as the AAA account identifier.
gateway
Indicates that AAA uses the router-specific name derived from the host name
and domain name as the AAA account identifier, displayed in the following
format: router-name.domain-name.
redialer-id
Indicates that AAA uses the account string returned by the external redialer
device as the AAA account identifier. In this case, the redialer ID is either the
redialer serial number or the redialer account number.
redialer-dnis
Indicates that AAA uses the called party telephone number (dialed number
identification service or DNIS) as the AAA account identifier captured by the
redialer if a redialer device is present.
Defaults
No account identification source is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Normally, when AAA is being used for simple user authentication, AAA uses the username information
defined in the user profile for authentication. With store and forward fax, you can specify that the ANI,
DNIS, gateway ID, redialer DNIS, or redialer ID be used to identify the user for authentication. This
command defines what AAA uses for the secondary identifier for inbound or on-ramp user
authentication with store and forward fax if the primary identifier has not been defined.
Store and forward fax allows you to define either a primary or a secondary identifier. (You configure the
primary identifier using the mmoip aaa receive-id primary command.)
Cisco IOS Voice, Video, and Fax Command Reference
VR-468
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa receive-id secondary
AAA does not use these methods sequentially—meaning that if the primary identifier is defined and
AAA cannot match the primary identifier information, it will not use the secondary identifier for
authentication. Authentication simply fails.
Defining only the secondary identifier enables you to service two different scenarios
simultaneously—for example, if you are offering fax services to two different companies, one of which
uses redialers and the other does not. In this case, configure the mmoip aaa receive-id primary
command to use the redialer DNIS, and configure the mmoip aaa receive-id secondary command to
use ANI. With this configuration, when a user dials in and the redialer DNIS is not null, the redialer
DNIS is used as the authentication identifier. If a user dials in and the redialer DNIS is null, ANI is used
as the authentication identifier.
This command applies to on-ramp store and forward fax functions.
Examples
The following example defines the DNIS captured by the redialer as the secondary AAA authentication
identifier for store and forward fax:
aaa new-model
mmoip aaa receive-id secondary redialer-dnis
Related Commands
Command
Description
mmoip aaa receive-id
primary
Specifies the primary location where AAA retrieves its account
identification information for on-ramp faxing.
Cisco IOS Voice, Video, and Fax Command Reference
VR-469
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa send-accounting enable
mmoip aaa send-accounting enable
To enable off-ramp authentication, authorization, and accounting (AAA) accounting services, use the
mmoip aaa send-accounting enable command in global configuration mode. To restore the default
state, use the no form of this command.
mmoip aaa send-accounting enable
no mmoip aaa send-accounting enable
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This command enables AAA accounting services if an AAA accounting method list has been defined
using both the aaa accounting command and the mmoip aaa method fax accounting command.
This command applies to off-ramp store and forward fax functions when using a modem card. It is not
used with voice feature cards.
Examples
The following example enables an AAA accounting method list (called xyz) to be used with outbound
store and forward fax. In this example, store and forward fax is being configured to track start and stop
connection accounting records.
aaa new-model
mmoip aaa method fax accounting xyz
aaa accounting connection sherman stop-only radius
mmoip aaa send-accounting enable
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security
purposes when you use RADIUS or TACACS+.
mmoip aaa method
fax accounting
Defines the name of the method list to be used for AAA accounting with
store and forward fax.
Cisco IOS Voice, Video, and Fax Command Reference
VR-470
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa send-authentication enable
mmoip aaa send-authentication enable
To enable off-ramp authentication, authorization, and accounting (AAA) authentication services, use the
mmoip aaa send-authentication enable command in global configuration mode. To restore the default
value, use the no form of this command.
mmoip aaa send-authentication enable
no mmoip aaa send-authentication enable
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This command enables AAA authentication services if an AAA authentication method list has been
defined using both the aaa authentication command and the mmoip aaa method fax authentication
command.
This command applies to off-ramp store and forward fax functions.
Examples
The following example enables an AAA authentication method list (called xyz) to be used with outbound
store and forward fax. In this example, RADIUS authentication (and if the RADIUS server fails, then
local authentication) is being configured for store and forward fax.
aaa new-model
mmoip aaa method fax authentication xyz
aaa authentication login peabody radius local
mmoip aaa send-authentication enable
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security
purposes when you use RADIUS or TACACS+.
mmoip aaa method
fax authentication
Defines the name of the method list to be used for AAA authentication with
store and forward fax.
Cisco IOS Voice, Video, and Fax Command Reference
VR-471
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa send-id primary
mmoip aaa send-id primary
To specify the primary location where the authentication, authorization, and accounting (AAA) protocol
retrieves its account identification information for off-ramp faxing, use the mmoip aaa send-id
primary command in global configuration mode. To restore the default state, which means that account
identification source is undefined, use the no form of this command.
mmoip aaa send-id primary {account-id | envelope-from | envelope-to | gateway}
no mmoip aaa send-id primary {account-id | envelope-from | envelope-to | gateway}
Syntax Description
account-id
Indicates that AAA uses the account username from the originating fax-mail
system as the AAA account identifier. This means that the off-ramp gateway
uses the account identifier in the X-account ID field of the e-mail header.
Using this attribute offers end-to-end authentication and accounting tracking.
envelope-from
Indicates that AAA uses the account username from the fax-mail header as the
AAA account identifier.
envelope-to
Indicates that AAA uses the recipient derived from the fax-mail header as the
AAA account identifier.
gateway
Indicates that AAA uses the router-specific name derived from the host name
and domain name as the AAA account identifier, displayed in the following
format: router-name.domain-name.
Defaults
No account identification source is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Normally, when AAA is being used for simple user authentication, AAA uses the username information
defined in the user profile for authentication. With store and forward fax, you can specify that the
account ID, username, or recipient name from the e-mail header information be used to identify the user
for authentication. This command defines what AAA uses for the primary identifier for outbound or
off-ramp user authentication with store and forward fax.
Store and forward fax allows you to define either a primary or a secondary identifier. (You configure the
secondary identifier using the mmoip aaa send-id secondary command.) AAA extracts the
authentication identifier information from the defined sources. If the field is blank (meaning undefined),
AAA will use the secondary identifier source if configured. The secondary identifier is used only when
the primary identifier is null. In this case, when AAA sees that the primary identifier is null, it will check
to see if a secondary identifier has been defined and use that value for user authentication.
Cisco IOS Voice, Video, and Fax Command Reference
VR-472
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa send-id primary
AAA does not use these methods sequentially—meaning that if the primary identifier is defined and
AAA cannot authenticate the primary identifier information, it will not use the secondary identifier for
authentication. Authentication simply fails.
When you enable authentication, the on-ramp gateway inserts whatever value you configure for the
mmoip aaa receive-id primary command in the X-account ID field of the e-mail header. This
X-account ID field contains the value that is used for authentication and accounting by the on-ramp
gateway. For example, if the mmoip aaa receive-id primary command is set to gateway, the on-ramp
gateway name (for example, hostname.domain-name) is inserted in the X-account ID field of the e-mail
header of the fax-mail message.
If you want to use this configured gateway value in the X-account ID field, you must configure the
mmoip aaa send-id primary command with the account-id keyword. This particular keyword enables
store and forward fax to generate end-to-end authentication and accounting tracking records. If you do
not enable authentication on the on-ramp gateway, the X-account ID field is left blank.
This command applies to off-ramp store and forward fax functions.
Examples
The following example defines the recipient name as defined in the envelope-to field of the e-mail
header to be used as the AAA authentication identifier for store and forward fax:
aaa new-model
mmoip aaa send-id primary envelope-to
Related Commands
Command
Description
mmoip aaa send-id
primary
Specifies the primary location where AAA retrieves its account
identification information for off-ramp faxing.
mmoip aaa send-id
secondary
Specifies the secondary location where AAA retrieves its account
identification information for off-ramp faxing.
Cisco IOS Voice, Video, and Fax Command Reference
VR-473
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa send-id secondary
mmoip aaa send-id secondary
To specify the secondary location where the authentication, authorization, and accounting (AAA)
protocol retrieves its account identification information for off-ramp faxing, use the mmoip aaa send-id
secondary command in global configuration mode. To restore the default state, which means that
account identification source is undefined, use the no form of this command.
mmoip aaa send-id secondary {account-id | envelope-from | envelope-to | gateway}
no mmoip aaa send-id secondary {account-id | envelope-from | envelope-to | gateway}
Syntax Description
account-id
Indicates that AAA uses the account username from the originating fax-mail
system as the AAA account identifier. This means that the off-ramp gateway
uses the account identifier in the x-account ID field of the e-mail header. Using
this attribute offers end-to-end authentication and accounting tracking.
envelope-from
Indicates that AAA uses the account username from the fax-mail header as the
AAA account identifier.
envelope-to
Indicates that AAA uses the recipient derived from the fax-mail header as the
AAA account identifier.
gateway
Indicates that AAA uses the router-specific name derived from the host name
and domain name as the AAA account identifier, displayed in the following
format: router-name.domain-name.
Defaults
No account identification source is defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Normally, when AAA is being used for simple user authentication, AAA uses the username information
defined in the user profile for authentication. With store and forward fax, you can specify that the
account ID, username, or recipient name from the e-mail header information be used to identify the user
for authentication. This command defines what AAA uses for the secondary identifier for outbound or
off-ramp user authentication with store and forward fax.
Store and forward fax allows you to define either a primary or a secondary identifier. (You configure the
primary identifier using the mmoip aaa send-id primary command.) AAA extracts the authentication
identifier information from the defined sources. If the field is blank (meaning undefined), AAA will use
the secondary identifier source if configured. The secondary identifier is used only when the primary
identifier is null. In this case, when AAA sees that the primary identifier is null, it will check to see if a
secondary identifier has been defined and use that value for user authentication.
Cisco IOS Voice, Video, and Fax Command Reference
VR-474
Cisco IOS Voice, Video, and Fax Commands: G Through P
mmoip aaa send-id secondary
AAA does not use these methods sequentially—meaning that if the primary identifier is defined and
AAA cannot match the primary identifier information, it will not use the secondary identifier for
authentication. Authentication simply fails.
When you enable authentication, the on-ramp gateway inserts whatever value you configure for the
mmoip aaa receive-id secondary command in the X-account ID field of the e-mail header (if
store-and-forward uses the defined secondary identifier). This X-account ID field contains the value that
is used for authentication and accounting by the on-ramp gateway. For example, if the
mmoip aaa receive-id secondary command is set to gateway, the on-ramp gateway name (for example,
hostname.domain-name) is inserted in the X-account ID field of the e-mail header of the fax-mail
message.
If you want to use this configured gateway value in the X-account ID field, you must configure the
mmoip aaa send-id secondary command with the account-id keyword. This particular keyword
enables store and forward fax to generate end-to-end authentication and accounting tracking records. If
you do not enable authentication on the on-ramp gateway, the X-account ID field is left blank.
This command applies to off-ramp store and forward fax functions.
Examples
The following example defines the recipient name as defined in the envelope-to field of the e-mail
header to be used as the AAA authentication identifier for store and forward fax:
aaa new-model
mmoip aaa send-id secondary envelope-to
Related Commands
Command
Description
mmoip aaa send-id
primary
Specifies the primary location where AAA retrieves its account
identification information for off-ramp faxing.
mmoip aaa send-id
secondary
Specifies the secondary location where AAA retrieves its account
identification information for off-ramp faxing.
Cisco IOS Voice, Video, and Fax Command Reference
VR-475
Cisco IOS Voice, Video, and Fax Commands: G Through P
mode
mode
To set the mode of the T1/E1 controller and enter specific configuration commands for each mode type,
use the mode command in controller configuration mode. To restore the default mode of the controller,
use the no form of this command.
mode {atm | cas}
no mode {atm | cas}
Syntax Description
atm
Places the controller into ATM mode and creates an ATM interface (ATM 0)
on the Cisco MC3810 multiservice concentrator. When ATM mode is
enabled, no channel groups, channel-associated signaling (CAS) groups,
CCS groups, or clear channels are allowed because ATM occupies all the
DS0s on the T1/E1 trunk.
When you set the controller to ATM mode, the controller framing is
automatically set to ESF for T1 or CRC4 for E1. The linecode is
automatically set to B8ZS for T1 or HDBC for E1. When you remove ATM
mode by entering the no mode atm command, ATM interface 0 is deleted.
ATM mode is supported only on controller 0 (T1 or E1 0).
cas
Places the controller into CAS mode, which allows you to create channel
groups, CAS groups, and clear channels (both data and CAS modes).
CAS mode is supported on both controller 0 and controller 1.
Defaults
No mode is configured.
Command Modes
Controller configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
This command applies to the Cisco MC3810 multiservice concentrator with the digital voice module
(DVM) installed.
When no mode is selected, channel groups and clear channels (data mode) can be created using the
channel group and tdm-group commands, respectively.
Cisco IOS Voice, Video, and Fax Command Reference
VR-476
Cisco IOS Voice, Video, and Fax Commands: G Through P
mode
On the Cisco MC3810 multiservice concentrator, some DS0s are used exclusively for different signaling
modes. The DS0 channels have the following limitations when mixing different applications (such as
voice and data) on the same network trunk:
Examples
•
On E1 controllers, DS0 16 is used exclusively for either CAS or CCS, depending on which mode is
configured.
•
On T1 controllers, DS0 24 is used exclusively for CCS.
The following example configures ATM mode on controller T1 0. This is required for Voice over ATM.
controller T1 0
mode atm
The following example configures CAS mode on controller T1 1:
controller T1 1
mode cas
Related Commands
Command
Description
channel-group
Defines the time slots that belong to each T1 or E1 circuit.
tdm-group
Configures a list of time slots for creating clear channel groups
(pass-through) for TDM cross-connect.
Cisco IOS Voice, Video, and Fax Command Reference
VR-477
Cisco IOS Voice, Video, and Fax Commands: G Through P
mode ccs
mode ccs
To configure the T1/E1 controller to support common channel signaling (CCS) cross-connect or CCS
frame forwarding, use the mode ccs command in controller configuration mode. To disable support for
CCS cross-connect or CCS frame forwarding on the controller, use the no form of this command.
mode ccs {cross-connect | frame-forwarding}
no mode ccs {cross-connect | frame-forwarding}
Syntax Description
cross-connect
Enables CCS cross-connect on the controller.
frame-forwarding
Enables CCS frame forwarding on the controller.
Defaults
No CCS mode is configured.
Command Modes
Controller configuration
Command History
Release
Modification
12.0(2)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.1(2)XH
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers.
12.1(3)T
The modifications in 12.1(2)XH were integrated into 12.1(3)T.
Examples
To enable CCS cross-connect on controller T1 1, enter the following commands:
controller T1 1
mode ccs cross-connect
To enable CCS frame forwarding on controller T1 1, enter the following commands:
controller T1 1
mode ccs frame-forwarding
Related Commands
Command
Description
ccs connect
Configures a CCS connection on an interface configured to support CCS
frame forwarding.
Cisco IOS Voice, Video, and Fax Command Reference
VR-478
Cisco IOS Voice, Video, and Fax Commands: G Through P
modem passthrough (dial-peer)
modem passthrough (dial-peer)
To configure modem passthrough over VoIP for a specific dial peer, use the modem passthrough
command in dial-peer configuration mode. To disable modem passthrough for a specific dial peer, use
the no modem passthrough command.
modem passthrough {system | nse [payload-type number] codec {g711ulaw | g711alaw}
[redundancy]}
no modem passthrough
Syntax Description
system
Defaults to the global configuration.
nse
Named signaling event.
payload-type
(Optional) NSE payload type.
number
(Optional) The value of the payload type (96–119).
codec
Voice compression for speech or audio signals. Codec selections for
upspeed. The upspeed method is the method used to dynamically change the
codec type and speed to meet network conditions. This means that you
might move to a faster codec when you have both voice and data calls, and
then slow down when there is only voice traffic.
g711ulaw
Codec G.711 u-Law 64000 bits per second for T1.
g711alaw
Codec G.711 A-Law 64000 bits per second for E1.
redundancy
(Optional) Packet redundancy (RFC 2198) for modem traffic.
Defaults
Defining system as the method in dial peer points to the voice service Voice over IP (VoIP)
configuration default and is intended to simplify gateway configuration. The default is modem
passthrough system. As required, the gateway defaults to no modem passthrough.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(3)T
This command was introduced for the Cisco AS5300 universal access
server.
Usage Guidelines
Use the modem passthrough dial-peer configuration command to configure modem passthrough over
VoIP for a specific dial peer. The payload type is an optional parameter for the nse keyword. Use the
same payload-type number for both the originating gateway and the terminating gateway. The
payload-type number can be set from 96 to 119. If you do not specify the payload-type number, the
number defaults to 100.
Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec
is required for T1, and g711alaw codec is required for E1.
The redundancy keyword is an optional parameter for sending redundant packets for modem traffic.
Cisco IOS Voice, Video, and Fax Command Reference
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modem passthrough (dial-peer)
When the system keyword is enabled, the following parameters are not available: nse, payload-type,
codec, and redundancy. The system keyword overrides the configuration for the dial peer, and the
values from the global configuration are used.
Examples
The following example shows how modem passthrough over VoIP is configured for a specific dial peer
in dial-peer configuration mode:
modem passthrough nse codec g711ulaw redundancy
Related Commands
Command
Description
dial-peer voice
Enters dial-peer configuration mode.
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modem passthrough (voice-service)
modem passthrough (voice-service)
To configure modem passthrough over VoIP for the Cisco AS5300 universal access server, use the
modem passthrough command in voice-service configuration mode. To disable modem passthrough,
use the no form of this command.
modem passthrough nse [payload-type number] codec {g711ulaw | g711alaw}
[redundancy] [maximum-sessions value]
no modem passthrough
Syntax Description
nse
Named signaling event.
payload-type
(Optional) NSE payload type.
number
(Optional) The value of the payload type. The number can be from 96 to
119.
codec
Codec selections for upspeed. The upspeed method is the method used to
dynamically change the codec type and speed to meet network conditions.
This means that you might move to a faster codec when you have both voice
and data calls and then slow down when there is only voice traffic.
g711ulaw
Codec G.711 u-Law 64000 bits per second for T1.
g711alaw
Codec G.711 A-Law 64000 bits per second for E1.
redundancy
(Optional) Packet redundancy (RFC 2198) for modem traffic.
maximum-sessions
(Optional) Maximum number of simultaneous modem passthrough
sessions.
value
(Optional) The number of simultaneous modem pass through sessions. The
minimum value is 1, and the maximum value is 26. The default is 16
sessions.
Defaults
Disabled
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(3)T
This command was introduced for the Cisco AS5300 universal access
server.
Usage Guidelines
Use the modem passthrough command to configure modem passthrough over Voice over IP (VoIP) for
the Cisco AS5300 universal access server. The default behavior for the voice service voip command is
no modem passthrough.
The payload type is an optional parameter for the nse keyword. Use the same payload-type number for
both the originating gateway and the terminating gateway. The payload-type number can be set from
96 to 119. If you do not specify the payload-type number, the number defaults to 100.
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modem passthrough (voice-service)
Use the same codec type for both the originating gateway and the terminating gateway. g711ulaw codec
is required for T1, and g711alaw codec is required for E1.
The redundancy keyword is an optional parameter for sending redundant packets for modem traffic.
The maximum-sessions keyword is an optional parameter for the modem passthrough command. This
parameter determines the maximum number of simultaneous modem passthrough sessions. The
recommended value for the maximum-sessions keyword is 16. The value can be set from 1 to 26.
When using the voice service voip and modem passthrough nse commands on a terminating gateway
to globally set up fax or modem pass-through with NSEs, you must also ensure that each incoming call
will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate
calls with dial peers by using the incoming called-number command to specify a sequence of digits that
incoming calls can match. You can ensure that all calls will match at least one dial peer by using the
following commands:
Router(config)# dial-peer voice tag voip
Router(config-dial-peer)# incoming called-number .
Examples
The following example shows modem pass-though configuration in voice-service configuration mode
for NSE payload type 101 using codec G.711:
modem passthrough nse payload-type 101 codec g711ulaw redundancy maximum-sessions 1
Related Commands
Command
Description
incoming
called-number
Defines an incoming called number to match a specific dial peer.
voice service voip
Enters voice-service configuration mode and specifies the voice
encapsulation type.
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Cisco IOS Voice, Video, and Fax Commands: G Through P
mta receive aliases
mta receive aliases
To specify a host name accepted as a Simple Mail Transfer Protocol (SMTP) alias for off-ramp faxing,
use the mta receive aliases command in global configuration mode. To disable this alias, use the no
form of this command.
mta receive aliases string
no mta receive aliases string
Syntax Description
string
Defaults
Enabled with an empty string
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Note
Specifies the host name or IP address to be used as an alias for the SMTP server.
If you specify an IP address to be used as an alias, you must enclose the IP
address in brackets as follows: [xxx.xxx.xxx.xxx].
This command creates an accept or reject alias list. The first alias is used by the mailer to identify itself
in SMTP banners and when generating its own RFC 822 Received: header.
This command does not automatically include reception for a domain IP address—it must be
explicitly added. To explicitly add a domain IP address, use the following format: mta receive
aliases [ip-address]. Use the IP address of the Ethernet or the FastEthernet interface of the off-ramp
gateway.
This command applies to on-ramp store and forward fax functions.
Examples
The following example specifies the host name seattle-fax-offramp.example.com as the alias for the
SMTP server:
mta receive aliases seattle-fax-offramp.example.com
The following example specifies the IP address 172.166.0.0 as the alias for the SMTP server:
mta receive aliases [172.16.0.0]
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mta receive aliases
Related Commands
Command
Description
mta receive
generate-mdn
Specifies that the off-ramp gateway process a response MDN from an SMTP
server.
mta receive
maximum-recipients
Specifies the maximum number of recipients for all SMTP connections.
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mta receive generate-mdn
mta receive generate-mdn
To specify that the off-ramp gateway process a response message disposition notice (MDN) from a
Simple Mail Transfer Protocol (SMTP) server, use the mta receive generate-mdn command in global
configuration mode. To disable message delivery notice generation, use the no form of this command.
mta receive generate-mdn
no mta receive generate-mdn
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
When message delivery notification is enabled on a sending Cisco AS5300 universal access server, the
device inserts a flag in the off-ramp message e-mail header, requesting that the receiving Cisco AS5300
universal access server generate the message delivery notification and return that message to the sender
when the e-mail message that contains the fax image is opened. Use the mta receive generate-mdn
command to enable the receiving device—the off-ramp gateway—to process the response message
delivery notification.
Depending on the configuration, usage, and features of the mailers used at a site, it might be desirable
to enable or disable MDN generation. (DSN generation cannot be disabled.)
Specifications for MDN are described in RFC 2298.
This command applies to off-ramp store and forward fax functions.
Examples
The following example enables the receiving device to generate message delivery notices:
mta receive generate-mdn
Related Commands
Command
Description
mdn
Requests that a message disposition notice be generated when the fax-mail
message is processed (opened).
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mta receive generate-mdn
Command
Description
mta receive aliases
Specifies a host name accepted as an SMTP alias for off-ramp faxing.
mta receive
maximum-recipients
Specifies the maximum number of recipients for all SMTP connections.
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mta receive maximum-recipients
mta receive maximum-recipients
To specify the maximum number of recipients for all Simple Mail Transfer Protocol (SMTP)
connections, use the mta receive maximum-recipients command in global configuration mode. To
restore the default value, use the no form of this command.
mta receive maximum-recipients number
no mta receive maximum-recipients
Syntax Description
number
Defaults
The default is 0 recipients, meaning that incoming mail messages will not be accepted; therefore, no
faxes are sent by the off-ramp gateway.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Specifies the maximum number of recipients for all SMTP connections.
Valid entries are from 0 to 1024.
This command applies to off-ramp store and forward fax functions.
Note
Unless the sending mailer supports the X-SESSION SMTP service extension, each incoming
SMTP connection will be allowed to send only to one recipient and thus consume only one
outgoing modem.
Use the mta receive maximum-recipients command to configure the maximum number of modems that
you want to allocate for fax usage at any one time. You can use this command to limit the resource usage
on the gateway. When the value for the number argument is set to 0, no new connections can be
established. This is particularly useful when preparing to shut down the system.
Examples
The following example defines 10 as the maximum number of recipients for all SMTP connections:
mta receive maximum-recipients 10
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mta receive maximum-recipients
Related Commands
Command
Description
mta receive aliases
Specifies a host name accepted as an SMTP alias for off-ramp faxing.
mta receive
generate-mdn
Specifies that the off-ramp gateway process a response MDN from an SMTP
server.
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mta send mail-from
mta send mail-from
To specify the mail-from address (also called the RFC 821 envelope-from address or the Return-Path
address), use the mta send mail-from command in global configuration mode. To disable this return
path information, use the no form of this command.
mta send mail-from {hostname string | username string | username $s$}
no mta send mail-from {hostname string | username string | username $s$}
Syntax Description
hostname string
Text string that specifies the Simple Mail Transfer Protocol (SMTP) host name
or IP address. If you specify an IP address, you must enclose the IP address in
brackets as follows: [xxx.xxx.xxx.xxx].
username string
Text string that specifies the sender username.
username $s$
Wildcard that specifies that the username will be derived from the calling
number.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Use the mta send mail-from command to designate the sender of the fax TIFF attachment. This value
is equivalent to the return path information in an e-mail message.
The postmaster address, configured with the mta send postmaster command, is used if the mail-from
address is blank.
This command applies to on-ramp store and forward fax functions.
Examples
The following example specifies that the mail-from username information will be derived from the
calling number of the sender:
mta send mail-from username $s$
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mta send mail-from
Related Commands
Command
Description
mta send origin-prefix Adds information to the e-mail prefix header.
mta send postmaster
Defines where an e-mail message should be delivered (the mail server
postmaster account) if it cannot be delivered to the defined destination.
mta send
return-receipt-to
Specifies the address where MDNs are sent.
mta send server
Specifies a destination mail server or servers.
mta send subject
Specifies the subject header of the e-mail message.
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mta send origin-prefix
mta send origin-prefix
To add information to the e-mail prefix header, use the mta send origin-prefix command in global
configuration mode. To disable the defined string, use the no form of this command.
mta send origin-prefix string
no mta send origin-prefix string
Syntax Description
string
Defaults
Null string
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Text string that adds comments to the e-mail prefix header. If this string
contains more than one word, the string value should be contained within
quotation marks (“abc xyz”).
Normally, the store and forward fax feature provides the slot and port number from which the e-mail
came in the e-mail prefix header information. Use this command to append the defined text string to the
front of the e-mail prefix header information. This test string is a prefix string that is appended with the
modem port and slot number and passed in the originator_comment field of the
esmtp_client_engine_open() call. Eventually, this ends up in the received header field of the fax-mail
message, for example:
Received (test onramp Santa Cruz slot1 port15) by router-5300.cisco.com for
<[email protected]> (with Cisco NetWorks); Fri, 25 Dec 1998 001500 -0800
In other words, using the command mta send origin-prefix dog causes the Received header to contain
the following information:
Received (dog, slot 3 modem 8) by as5300-sj.example.com....
This command applies to on-ramp store and forward fax functions.
Examples
The following example shows how to add information to the e-mail prefix header:
mta send origin-prefix “Cisco-Powered Fax System”
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mta send origin-prefix
Related Commands
Command
Description
mta send mail-from
Specifies the mail-from address (also called the RFC 821 envelope-from
address or the Return-Path address).
mta send postmaster
Defines where an e-mail message should be delivered (the mail server
postmaster account) if it cannot be delivered to the defined destination.
mta send
return-receipt-to
Specifies the address where MDNs are sent.
mta send server
Specifies a destination mail server or servers.
mta send subject
Specifies the subject header of the e-mail message.
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mta send postmaster
mta send postmaster
To define where an e-mail message should be delivered (the mail server postmaster account) if it cannot
be delivered to the defined destination, use the mta send postmaster command in global configuration
mode. To disable this defined postmaster, use the no form of this command.
mta send postmaster e-mail-address
no mta send postmaster e-mail-address
Syntax Description
e-mail-address
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
E-mail address that defines where this e-mail message should be delivered
(the mail server postmaster account) if it cannot be delivered to the defined
destination.
If you have configured the Cisco AS5300 universal access server to generate DSNs and MDNs but you
have not configured the sender information (using the mta send mail-from command) or the Simple
Mail Transfer Protocol (SMTP) server, DSNs and MDNs are delivered to the e-mail address determined
by this command.
The address defined by this command is used as the mta send mail-from address if the evaluated string
is blank. An address, such as [email protected], is recommended (where example.com is
replaced with your domain name, and fax-administrator is aliased to the person responsible for the
operation of the AS5300 universal access server fax functions). At some sites, this may be the same
person as the e-mail postmaster, but at most sites this is likely to be a different person and thus should
be a different e-mail address.
This command applies to on-ramp store and forward fax functions.
Examples
The following example configures the e-mail address [email protected] as the sender for all
incoming faxes. Thus, any returned DSNs will be delivered to [email protected] if the mail-from
field is otherwise blank.
mta send postmaster [email protected]
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mta send postmaster
Related Commands
Command
Description
mta send mail-from
Specifies the mail-from address (also called the RFC 821 envelope-from
address or the Return-Path address).
mta send origin-prefix Adds information to the e-mail prefix header.
mta send
return-receipt-to
Specifies the address where MDNs are sent.
mta send server
Specifies a destination mail server or servers.
mta send subject
Specifies the subject header of the e-mail message.
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mta send return-receipt-to
mta send return-receipt-to
To specify the address where message disposition notices (MDNs) are sent, use the mta send
return-receipt-to command in global configuration mode. To restore the default value, use the no form
of this command.
mta send return-receipt-to {hostname string | username string | username $s$}
no mta send return-receipt-to {hostname string | username string | username $s$}
Syntax Description
hostname string
Text string that specifies the Simple Mail Transfer Protocol (SMTP) host name
or IP address where MDNs are sent. If you specify an IP address, you must
enclose the IP address in brackets as follows: [xxx.xxx.xxx.xxx].
username string
Text string that specifies the sender username where MDNs are sent.
username $s$
Wildcard that specifies that the username are derived from the calling number.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Use the mta send return-receipt-to command to define where you want MDNs to be sent after the
fax-mail is opened.
Note
Store and forward fax supports the Eudora proprietary format, meaning that the header that store and
forward fax generates is in compliance with RFC 2298 (MDN).
Note
Multimedia Mail over Internet Protocol (MMoIP) dial peers must have MDN enabled to generate
return receipts in off-ramp fax-mail messages.
This command applies to on-ramp store and forward fax functions.
Examples
The following example configures scoobee as the SMTP mail server to which DSNs are sent:
mta send return-receipt-to hostname server.com
mta send return-receipt-to username scoobee
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mta send return-receipt-to
Related Commands
Command
Description
mta send mail-from
Specifies the mail-from address (also called the RFC 821 envelope-from
address or the Return-Path address).
mta send origin-prefix Adds information to the e-mail prefix header.
mta send postmaster
Defines where an e-mail message should be delivered (the mail server
postmaster account) if it cannot be delivered to the defined destination.
mta send server
Specifies a destination mail server or servers.
mta send subject
Specifies the subject header of the e-mail message.
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mta send server
mta send server
To specify a destination mail server or servers, use the mta send server command in global
configuration mode. To disable the specified destination mail server, use the no form of this command.
mta send server {host-name | IP-address}
no mta send server {host-name | IP-address}
Syntax Description
host-name
Defines the host name of the destination mail server.
IP-address
Defines the IP address of the destination mail server.
Defaults
IP address defined as 0.0.0.0
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Use the mta send server command to provide a backup destination server in case the first configured
mail server is unavailable. (This command is not intended to be used for load distribution.)
You can configure up to ten different destination mail servers using the mta send server command. If
you configure more than one destination mail server, the Cisco AS5300 universal access server attempts
to contact the first mail server configured. If that mail server is unavailable, it will contact the next
configured destination mail server.
DNS MX records are not used to look up host names provided to this command.
Note
When you use the mta send server command, you should configure the Cisco AS5300 universal
access server to perform name lookups using the ip name-server command.
This command applies to on-ramp store and forward fax functions.
Examples
The following example defines the mail servers scoobee.example.com and doogie.example.com as the
destination mail servers:
mta send server scoobee.example.com
mta send server doogie.example.com
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mta send server
Related Commands
Command
Description
ip name-server
Specifies the address of one or more name servers to use for name and
address resolution.
mta send mail-from
Specifies the mail-from address (also called the RFC 821 envelope-from
address or the Return-Path address).
mta send origin-prefix Adds information to the e-mail prefix header.
mta send postmaster
Defines where an e-mail message should be delivered (the mail server
postmaster account) if it cannot be delivered to the defined destination.
mta send
return-receipt-to
Specifies the address where MDNs are sent.
mta send subject
Specifies the subject header of the e-mail message.
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mta send subject
mta send subject
To specify the subject header of the e-mail message, use the mta send subject command in global
configuration mode. To disable this string, use the no form of this command.
mta send subject string
no mta send subject string
Syntax Description
string
Defaults
Null string
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XJ
This command was introduced on the Cisco AS5300 and Cisco AS5800
universal access servers.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
Note
Examples
Text string that specifies the subject header of an e-mail message.
This command applies to on-ramp store and forward fax functions.
The string does not need to be enclosed in quotation marks.
The following example defines the subject header of an e-mail message as “fax attachment”:
mta send subject fax attachment
Related Commands
Command
Description
mta send mail-from
Specifies the mail-from address (also called the RFC 821 envelope-from
address or the Return-Path address).
mta send origin-prefix Adds information to the e-mail prefix header.
mta send postmaster
Defines where an e-mail message should be delivered (the mail server
postmaster account) if it cannot be delivered to the defined destination.
mta send
return-receipt-to
Specifies the address where MDNs are sent.
mta send server
Specifies a destination mail server or servers.
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music-threshold
music-threshold
To specify the threshold for on-hold music for a specified voice port, use the music-threshold command
in voice-port configuration mode. To disable this feature, use the no form of this command.
music-threshold number
no music-threshold number
Syntax Description
number
Defaults
–38 dB
Command Modes
Voice-port configuration
Command History
Release
Usage Guidelines
The on-hold music threshold in decibels (dB). Valid entries are any integer from
–70 to –30.
Modification
11.3(1)T
This command was introduced for the Cisco 3600 series router.
12.0(4)T
Support was added for the Cisco MC3810 multiservice concentrator.
Use this command to specify the decibel level of music played when calls are put on hold. This command
tells the firmware to pass steady data above the specified level. It only affects the operation of voice
activity detection (VAD) when receiving voice.
If the value for this command is set too high, VAD interprets music-on-hold as silence, and the remote
end does not hear the music. If the value for this command is set too low, VAD compresses and passes
silence when the background is noisy, creating unnecessary voice traffic.
Examples
The following example sets the decibel threshold to –35 for the music played when calls are put on hold:
voice port 0:D
music-threshold -35
The following example sets the decibel threshold to –35 for the music played when calls are put on hold
on the Cisco 3600 series router:
voice-port 1/0/0
music-threshold -35
The following example sets the decibel threshold to –35 for the music played when calls are put on hold
on the Cisco MC3810 multiservice concentrator:
voice-port 1/1
music-threshold -35
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music-threshold
Related Commands
Command
Description
voice-port
Enters voice-port configuration mode.
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network-clock base-rate
network-clock base-rate
To configure the network clock base rate for universal I/O serial ports 0 and 1 on the Cisco MC3810
multiservice concentrator, use the network-clock base-rate command in global configuration mode. To
disable the current network clock base rate, use the no form of this command.
network-clock base-rate {56k | 64k}
no network-clock base-rate {56k | 64k}
Syntax Description
56k
Sets the network clock base rate to 56 kbps.
64k
Sets the network clock base rate to 64 kbps.
Defaults
56 kbps
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
This command applies to Voice over Frame Relay and Voice over ATM on the Cisco MC3810
multiservice concentrator.
Examples
The following example sets the network clock base rate to 64 kbps:
network-clock base-rate 64k
Related Commands
Command
Description
network-clock-select
Uses the network clock source to provide timing to the system backplane
PCM bus.
network-clock-switch
Configures the switch delay time to the next priority network clock source
when the current network clock source fails.
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network-clock-switch
network-clock-switch
To configure the switch delay time to the next priority network clock source when the current network
clock source fails, use the network-clock-switch command in global configuration mode. To cancel the
network clock delay time selection, use the no form of this command.
network-clock-switch {switch-delay | never} {restore-delay | never}
no network-clock-switch
Syntax Description
switch-delay
The delay time, in seconds, before the next priority network clock source is
used when the current network clock source fails. The range is from 0 to 99
seconds. The default is 10 seconds.
never
Indicates no delay time before the current network clock source recovers.
restore-delay
The delay time before the current network clock source recovers. The range
is from 0 to 99 seconds.
never
Indicates no delay time, in seconds, before the next priority network clock
source is used when the current network clock source fails.
Defaults
10 seconds
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Examples
The following command switches the network clock source after 20 seconds and sets the delay time
before the current network clock source recovers to 20 seconds:
network-clock-switch 20 20
Related Commands
Command
Description
network-clock-select
Uses the network clock source to provide timing to the system backplane
PCM bus.
Cisco IOS Voice, Video, and Fax Command Reference
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non-linear
non-linear
To enable nonlinear processing in the echo canceller, use the non-linear command in voice-port
configuration mode. To disable nonlinear processing, use the no form of this command.
non-linear
no non-linear
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series router.
Usage Guidelines
The function enabled by the non-linear command is also generally known as residual echo suppression.
This command is associated with the echo canceller operation. The echo-cancel enable command must
be enabled for the non-linear command to take effect. Use the non-linear command to shut off any
signal if no near-end speech is detected.
Enabling the non-linear command normally improves performance, although some users might perceive
truncation of consonants at the end of sentences when this command is enabled.
Examples
The following example enables nonlinear call processing on the Cisco 3600 series router:
voice-port 1/0/0
non-linear
The following example enables nonlinear call processing on the Cisco MC3810 multiservice
concentrator:
voice-port 1/1
non-linear
Related Commands
Command
Description
echo-cancel enable
Enables the cancellation of voice that is sent out the interface and is received
on the same interface.
Cisco IOS Voice, Video, and Fax Command Reference
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nsap
nsap
To specify the network service access point (NSAP) address for a local video dial peer, use the nsap
command in dial-peer configuration mode. To remove any configured NSAP address from the dial peer,
use the no form of this command.
nsap nsap-address
no nsap
Syntax Description
nsap-address
Defaults
No NSAP address for a video dial peer is configured.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM video dial-peer configuration on the
Cisco MC3810 multiservice concentrator.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(9)T.
A 40-digit hexadecimal number; the number must be unique on the device.
Usage Guidelines
The address must be unique on the router.
Examples
On a Cisco MC3810 multiservice concentrator, the following example sets up an NSAP address for the
local video dial peer designated as 10:
dial-peer video 10 videocodec
nsap 47.0091810000000002F26D4901.333333333332.02
Related Commands
Command
Description
dial-peer video
Defines a video ATM dial peer for a local or remote video codec, specifies
video-related encapsulation, and enters dial-peer configuration mode.
show dial-peer video
Displays dial-peer configuration.
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num-exp
num-exp
To define how to expand a telephone extension number into a particular destination pattern, use the
num-exp command in global configuration mode. To cancel the configured number expansion, use the
no form of this command.
num-exp extension-number expanded-number
no num-exp extension-number
Syntax Description
extension-number
Digit or digits that define an extension number for a particular dial peer.
expanded-number
Digit or digits that define the expanded telephone number or destination
pattern for the extension number listed.
Defaults
No number expansion is defined.
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series router.
12.0(3)T
This command was supported on the Cisco AS5300 universal access server.
12.0(4)XL
This command was supported on the Cisco AS5800 universal gateway.
12.0(7)T
The modifications in 12.0(4)XL were integrated in 12.0(7)T.
12.0(7)XK
This command was supported on the Cisco MC3810 multiservice
concentrator platform.
12.1(2)T
The modifications in 12.0(7)XK were integrated in 12.1(2)T.
Usage Guidelines
Use the num-exp global configuration command to define how to expand a particular set of numbers
(for example, a telephone extension number) into a particular destination pattern. With this command,
you can bind specific extensions and expanded numbers together by explicitly defining each number, or
you can define extensions and expanded numbers using variables. You can also use this command to
convert seven-digit numbers to numbers containing less than seven digits.
Use a period (.) as a variable or wildcard, representing a single number. Use a separate period for each
number that you want to represent with a wildcard—for example, if you want to replace four numbers
in an extension with wildcards, type in four periods.
Examples
The following example expands the extension number 55541 to the number 14085555541:
num-exp 55541 14085555541
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num-exp
The following example expands all five-digit extensions beginning with 5 such that the 5 is replaced
with the digits 1408555 at the beginning of the extension number:
num-exp 5.... 1408555....
Related Commands
Command
Description
dial-peer terminator
Designates a special character to be used as a terminator for variable
length dialed numbers.
forward-digits
Specifies which digits to forward for voice calls.
prefix
Specifies a prefix for a dial peer.
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numbering-type
numbering-type
To match on a number type for a dial-peer call leg, use the numbering-type command in dial-peer
configuration mode. To remove the numbering type for a dial-peer call leg, use the no form of this
command.
numbering-type {international | abbreviated | national | network | reserved | subscriber |
unknown}
no numbering-type {international | abbreviated | national | network | reserved | subscriber |
unknown}
Syntax Description
international
Specifies international numbering type.
abbreviated
Specifies abbreviated numbering type.
national
Specifies national numbering type.
network
Specifies network numbering type.
reserved
Specifies reserved numbering type.
subscriber
Specifies subscriber numbering type.
unknown
Specifies if the numbering type is unknown.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(7)XR1
This command was introduced for Voice over IP on the Cisco AS5300
universal access server.
12.0(7)XK
This command was first supported for the following voice
technologies on the following platforms:
•
Voice over IP (Cisco 2600 series router router, Cisco 3600 series
router, Cisco MC3810 multiservice concentrator)
•
Voice over Frame Relay (Cisco 2600 series router router, Cisco
3600 series router, Cisco MC3810 multiservice concentrator)
•
Voice over ATM (Cisco 3600 series router, Cisco MC3810
multiservice concentrator)
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numbering-type
Release
Modification
12.1(1)T
This command was first supported on the T train for the following
voice technology on the following platforms:
•
12.1(2)T
Voice over IP (Cisco 1750, Cisco 2600 series router, Cisco 3600
series router, Cisco AS5300 universal access server, Cisco 7200
series router, and Cisco 7500 series)
This command was first supported on the T train for the following
voice technologies on the following platforms:
•
Voice over IP (Cisco MC3810 multiservice concentrator)
•
Voice over Frame Relay (Cisco 2600 series router, Cisco 3600
series router, Cisco MC3810 multiservice concentrator)
•
Voice over ATM (Cisco 3600 series router, Cisco MC3810
multiservice concentrator)
Usage Guidelines
The numbering-type command is supported for plain old telephone service (POTS), VoIP, VoFR, and
VoATM dial peers. The numbering type options are implemented as defined by the ITU Q.931
specification.
Examples
The following example shows how to configure a POTS dial peer for network usage:
dial-peer voice 100 pots
numbering-type network
The following example shows how to configure a VoIP dial peer for subscriber usage:
dial-peer voice 200 voip
numbering-type subscriber
Related Commands
Command
Description
rule
Applies a translation rule to a calling party number or a called party
number for both incoming and outgoing calls.
show translation-rule
Displays the contents of all the rules that have been configured for a
specific translation name.
test translation-rule
Tests the execution of the translation rules on a specific name-tag.
translate
Applies a translation rule to a calling party number or a called party
number for incoming calls.
translate-outgoing
Applies a translation rule to a calling party number or a called party
number for outgoing calls.
translation-rule
Creates a translation name and enters translation-rule configuration
mode.
voip-incoming
translation-rule
Captures calls that originate from H.323-compatible clients.
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operation
operation
To select a specific cabling scheme for E&M ports, use the operation command in voice-port
configuration mode. To restore the default, use the no form of this command.
operation {2-wire | 4-wire}
no operation {2-wire | 4-wire}
Syntax Description
2-wire
Specifies a 2-wire E&M cabling scheme.
4-wire
Specifies a 4-wire E&M cabling scheme.
Defaults
2-wire operation
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series..
11.3(1)MA
Support was added for the Cisco MC3810 multiservice concentrator.
Usage Guidelines
This command applies to both the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator.
The operation command affects only voice traffic. Signaling is independent of 2-wire versus 4-wire
settings. If the wrong cable scheme is specified, the user might get voice traffic in only one direction.
Configuring the operation 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.
This command is not applicable to FXS or FXO interfaces because they are, by definition, 2-wire
interfaces.
On the Cisco MC3810 multiservice concentrator, this command applies only to the analog voice module
(AVM).
Examples
The following example specifies that an E&M port on the Cisco 3600 series router uses a 4-wire cabling
scheme:
voice-port 1/0/0
operation 4-wire
The following example specifies that an E&M port on the Cisco MC3810 multiservice concentrator uses
a 2-wire cabling scheme:
voice-port 1/1
operation 2-wire
Cisco IOS Voice, Video, and Fax Command Reference
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output attenuation
output attenuation
To configure a specific output attenuation value, use the output attenuation command in voice-port
configuration mode. To disable the selected output attenuation value, use the no form of this command.
output attenuation decibels
no output attenuation
Syntax Description
decibels
Defaults
The default value for FXO, FXS, and E&M ports is 0.
Command Modes
Voice-port configuration
Command History
Release
Usage Guidelines
The amount of attenuation, in decibels, at the transmit side of the interface. An
acceptable value is any integer from –6 to 14.
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series router.
11.3(1)MA
Support was added for the Cisco MC3810 multiservice concentrator.
A system-wide loss plan must be implemented using both the input gain and output attenuation
commands. Other equipment (including PBXs) in the system must be considered when creating a loss
plan. The default value for this command assumes that a standard transmission loss plan is in effect,
meaning that there must be an attenuation of –6 dB between phones. Connections are implemented to
provide –6 dB of attenuation when the input gain and output attenuation commands are configured
with the default value of 0 dB.
You cannot increase the gain of a signal to the Public Switched Telephone Network (PSTN), but you can
decrease it. If the voice level is too high, you can decrease the volume by either decreasing the input gain
or increasing the output attenuation.
You can increase the gain of a signal coming into the router. If the voice level is too low, you can
increase the input gain by using the input gain command.
Examples
On the Cisco 3600 series router, the following example configures a 3-dB gain to be inserted at the
transmit side of the interface:
voice-port 1/0/0
output attenuation 3
On the Cisco AS5300 universal access server, the following example configures a 3-dB gain to be
inserted at the transmit side of the interface:
voice-port 0:D
output attenuation 3
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output attenuation
On the Cisco MC3810 multiservice concentrator, the following example configures a 6-dB gain to be
inserted at the transmit side of the interface:
voice-port 1/1
output attenuation 6
Related Commands
Command
Description
input gain
Configures a specific input gain value for a voice port.
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Cisco IOS Voice, Video, and Fax Commands: G Through P
playout-delay (dial-peer)
playout-delay (dial-peer)
To tune the playout buffer on digital signal processors (DSPs) to accommodate packet jitter caused by
switches in the WAN, use the playout-delay command in dial-peer configuration mode. To restore the
default value, use the no form of this command.
playout-delay {nominal milliseconds | maximum milliseconds | minimum {default | low | high}}
no playout-delay {nominal milliseconds | maximum milliseconds | minimum {default | low |
high}}
Syntax Description
nominal milliseconds
The nominal keyword represents the amount of playout delay applied at the
beginning of a call by the jitter buffer in the gateway. In fixed mode, this is
also the maximum size of the jitter buffer throughout the call.
The milliseconds argument is the number of milliseconds (ms) of delay; the
range accepted is from 0 to 1500, although this value depends on the type
of DSP and how the voice card is configured for codec complexity. (See the
codec complexity command.)
If the voice card is configured for high codec complexity, the highest value
that can be configured for nominal for compressed codecs is 250 ms. For
medium-complexity codec configurations, the highest nominal value is
150 ms.
Voice hardware that does not support the voice card complexity
configuration (such as analog voice modules for the Cisco 3600 series
router) has an upper limit of 250 ms.
maximum milliseconds (Adaptive mode only)
The maximum keyword represents the upper limit of the jitter buffer, or the
highest value to which the adaptive delay will be set.
The milliseconds argument is the number of milliseconds of delay; the
range accepted is from 40 to 1700, although this value depends on the type
of DSP and how the voice card is configured for codec complexity. (See the
codec complexity command.)
If the voice card is configured for high codec complexity, the highest value
that can be configured for maximum for compressed codecs is 250 ms. For
medium-complexity codec configurations, the highest maximum value is
150 ms.
Voice hardware that does not support the voice card complexity
configuration (such as analog voice modules for the Cisco 3600 series
router) has an upper limit of 250 ms.
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playout-delay (dial-peer)
minimum
(Adaptive mode only)
The minimum keyword represents the lower limit of the jitter buffer, or the
lowest value to which adaptive delay will be set.
The low keyword represents 10 milliseconds. Use this keyword when there
are low jitter conditions in the network.
The high keyword represents 80 milliseconds. Use this keyword when there
are high jitter conditions in the network.
The default keyword represents 40 milliseconds and is appropriate when
there are normal jitter conditions in the network. This is the default value
used in adaptive mode when a minimum value is not configured.
Defaults
The default for nominal is 200 milliseconds.
The default for maximum is 200 milliseconds.
The default for minimum is 40 milliseconds.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator in the voice-port configuration mode.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series router
routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)XI
This command was first supported on the Cisco ICS 7750.
12.1(5)T
The minimum keyword was introduced, and support for dial-peer
configuration mode was added on the Cisco 2600 series, Cisco 3600 series,
and Cisco 7200 series routers, Cisco MC3810 multiservice concentrator,
and Cisco AS5200, Cisco AS5300, Cisco AS5400, and Cisco AS5800
universal access servers.
Usage Guidelines
Before Cisco IOS Release 12.1(5)T, the playout-delay command was configured in voice-port
configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay
should be configured in dial-peer configuration mode on the Voice over IP (VoIP) dial peer that is on
the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and
relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are
configured on the gateway, playout delay should be configured in dial-peer configuration mode. When
there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice
port. If conflicting playout delay values have been configured on a voice port and on a dial peer, the
dial-peer configuration takes precedence.
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playout-delay (dial-peer)
Playout delay is the amount of time that elapses between the time that a voice packet is received at the
jitter buffer on the DSP and the time that it is played out to the codec. In most networks with normal
jitter conditions, the defaults are adequate and you will not need to configure the playout-delay
command.
In situations in which you want to improve voice quality by reducing jitter or you want to reduce network
delay, you can configure playout-delay parameters. The parameters are slightly different for each of the
two playout delay modes, adaptive and fixed (see the playout-delay mode command).
In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival
variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for
jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have
been configured. The maximum limit establishes the highest value to which the adaptive delay will be
set. The minimum limit is the low-end threshold for the delay of incoming packets by the adaptive jitter
buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted
toward the improvement of voice quality at the expense of network delay: jitter buffer size increases
rapidly in response to spikes in network transmissions and decreases slowly in response to reduced
congestion.
In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the
jitter buffer in the gateway and is also the maximum size of the jitter buffer throughout the call.
As a general rule, if there is excessive breakup of voice due to jitter with the default playout delay
settings, increase playout delay times. If your network is small and jitter is minimal, decrease playout
delay times for a smaller overall delay.
When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is
actually adjusting the playout delay correctly. The constant readjustment of playout delay to erratic
network conditions causes voice quality problems that are usually alleviated by increasing the minimum
playout delay value in adaptive mode or by increasing the nominal delay for fixed mode.
Use the show call active voice command to display the current delay, as well as high- and low-water
marks for delay during a call. Other fields that can help determine the size of a jitter problem are
ReceiveDelay, GapFillWith..., LostPackets, EarlyPackets, and LatePackets. The following is an
example of the output from the show call active voice command:
VOIP:
ConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]
IncomingConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]
RemoteIPAddress=192.168.100.101
RemoteUDPPort=18834
RoundTripDelay=26 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=TRUE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=cisco
SessionTarget=
OnTimeRvPlayout=417000
GapFillWithSilence=850 ms
GapFillWithPrediction=2590 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=70 ms
LoWaterPlayoutDelay=29 ms
ReceiveDelay=39 ms
LostPackets=0
EarlyPackets=0
LatePackets=86
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playout-delay (dial-peer)
Examples
The following example uses the default adaptive mode for the playout-delay command with a
minimum playout delay of 10 milliseconds and a maximum playout delay of 60 milliseconds, on the
VoIP dial peer tagged 80. The size of the jitter buffer will be adjusted up and down on the basis of the
amount of jitter that the DSP finds, but will never be smaller than 10 milliseconds, and never larger than
60 milliseconds.
dial-peer 80 voip
playout-delay minimum low
playout-delay maximum 60
Related Commands
Command
Description
playout-delay mode
Selects fixed or adaptive mode for the jitter buffer on DSPs.
show call active voice
Displays active call information for voice calls.
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playout-delay (voice-port)
playout-delay (voice-port)
To tune the playout buffer to accommodate packet jitter caused by switches in the WAN, use the
playout-delay command in voice-port configuration mode. To restore the default value, use the no form
of this command.
playout-delay {maximum | nominal} milliseconds
no playout-delay {maximum | nominal}
Syntax Description
maximum
The delay time that the digital signal processor (DSP) allows before
starting to discard voice packets. The default is 160 milliseconds.
nominal
The initial (and minimum allowed) delay time that the DSP inserts before
playing out voice packets. The default is 80 milliseconds.
milliseconds
Playout-delay value, in milliseconds. The range for maximum playout
delay is 40 to 320, and the range for nominal playout delay is 40 to 240.
Defaults
The default for maximum delay is 160 milliseconds. The default for nominal delay is 80 milliseconds.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on Cisco 2600 and 3600 series routers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
If there is excessive breakup of voice due to jitter with the default playout delay settings, increase the
delay times. If your network is small and jitter is minimal, decrease the delay times to reduce delay.
Examples
The following example configures a nominal playout delay of 80 milliseconds and a maximum playout
delay of 160 milliseconds on voice port 1/1 on a Cisco MC3810 multiservice concentrator:
voice-port 1/1
playout-delay nominal 80
playout-delay maximum 160
The following example configures a nominal playout delay of 80 milliseconds and a maximum playout
delay of 160 milliseconds on voice port 1/0/0 on the Cisco 2600 or 3600:
voice-port 1/0/0
playout-delay nominal 80
playout-delay maximum 160
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playout-delay (voice-port)
Related Commands
Command
Description
vad
Enables voice activity detection.
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playout-delay mode (dial-peer)
playout-delay mode (dial-peer)
To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors
(DSPs), use the playout-delay mode command in dial-peer configuration mode. To restore the default
value, use the no form of this command.
playout-delay mode {adaptive | fixed}
no playout-delay mode {adaptive | fixed}
Syntax Description
adaptive
Jitter buffer size and amount of playout delay are adjusted during a call, on
the basis of current network conditions.
fixed
Jitter buffer size does not adjust during a call; a constant playout delay is
added.
Defaults
The default is the adaptive keyword.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(5)T
Support for this command was added on the Cisco 2600 and Cisco 3600
series routers, Cisco MC3810 multiservice concentrator, and the Cisco ICS
7750. In addition, the no-timestamps keyword was removed from the
syntax.
Usage Guidelines
Tips
Before Cisco IOS Release 12.1(5)T, the playout-delay command was configured only in voice-port
configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay
should be configured in dial-peer configuration mode on the VoIP dial peer that is on the receiving end
of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the
DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the
gateway, playout delay should be configured in dial-peer configuration mode.
When there are numerous dial peers to configure, it might be simpler to configure playout delay on
a voice port. If conflicting playout delay values have been configured on a voice port and on a dial
peer, the dial-peer configuration takes precedence.
In most networks with normal jitter conditions, the default is adequate and you do not need to configure
the playout-delay mode command.
The default is adaptive mode, in which the average delay for voice packets varies depending on the
amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and
shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum
and minimum limits that have been configured.
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playout-delay mode (dial-peer)
Select fixed mode only when you understand your network conditions well, and when you have a
network with very poor quality of service (QoS) or when you are interworking with a media server or
similar transmission source that tends to create a lot of jitter at the transmission source. In most
situations it is better to configure adaptive mode and let the DSP size the jitter buffer according to current
conditions.
Examples
The following example configures adaptive playout-delay mode with the minimum delay set at high
(80 milliseconds), on a VoIP dial peer that has a tag of 80:
dial-peer 80 voip
playout-delay mode adaptive
playout-delay minimum high
Related Commands
Command
Description
playout-delay
Tunes the jitter buffer on DSPs for playout delay of voice packets.
show call active voice
Displays active call information for voice calls.
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playout-delay mode (voice-port)
playout-delay mode (voice-port)
To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors
(DSPs), use the playout-delay mode command in voice port configuration mode. To restore the default
value, use the no form of this command.
playout-delay mode {adaptive | fixed}
no playout-delay mode {adaptive | fixed}
Syntax Description
adaptive
Jitter buffer size and amount of playout delay are adjusted during a call, on
the basis of current network conditions.
fixed
Jitter buffer size does not adjust during a call; a constant playout delay is
added.
Defaults
The default is the adaptive keyword.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
The playout-delay command was introduced on the Cisco MC3810
multiservice concentrator.
12.0(7)XK
The playout-delay command was first supported on the Cisco 2600 and
Cisco 3600 series routers.
12.1(2)T
The playout-delay command was integrated into Cisco IOS Release
12.1(2)T.
12.1(3)XI
The keyword mode was introduced, and the playout-delay mode command
was first supported on the Cisco ICS 7750.
12.1(5)T
The no-timestamps keyword was removed from the syntax.
Usage Guidelines
Tips
Before Cisco IOS Release 12.1(5)T, the playout-delay command was configured only in voice-port
configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay
should be configured in dial-peer configuration mode on the VoIP dial peer that is on the receiving end
of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the
DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the
gateway, playout delay should be configured in dial-peer configuration mode.
When there are numerous dial peers to configure, it might be simpler to configure playout delay on
a voice port. If conflicting playout delay values have been configured on a voice port and on a dial
peer, the dial-peer configuration takes precedence.
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playout-delay mode (voice-port)
In most networks with normal jitter conditions, the default is adequate and you do not need to configure
the playout-delay mode command.
The default is adaptive mode, in which the average delay for voice packets varies depending on the
amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and
shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum
and minimum limits that have been configured.
Select fixed mode only when you understand your network conditions well, and when you have a
network with very poor quality of service (QoS) or when you are interworking with a media server or
similar transmission source that tends to create a lot of jitter at the transmission source. In most
situations it is better to configure adaptive mode and let the DSP size the jitter buffer according to current
conditions.
Examples
The following example configures fixed mode on a Cisco 3640 voice port, with a nominal delay of
80 milliseconds.
voice-port 1/1/0
playout-delay mode fixed
playout-delay nominal 80
Related Commands
Command
Description
playout-delay
Tunes the jitter buffer on DSPs for playout delay of voice packets.
show call active voice
Displays active call information for voice calls.
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port (dial peer)
port (dial peer)
To associate a dial peer with a specific voice port, use the port command in dial-peer configuration
mode. To cancel this association, use the no form of this command.
Cisco 1750 Router
port slot-number/port
no port slot-number/port
Cisco 2600 and 3600 Series Routers
port {slot-number/subunit-number/port} | {slot/port:ds0-group-no}
no port {slot-number/subunit-number/port} | {slot/port:ds0-group-no}
Cisco MC3810 Multiservice Concentrator
port slot/port
no port slot/port
Cisco AS5300 Universal Access Server
port controller number:D
no port controller number:D
Cisco AS5800 Universal Gateway
port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
no port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
Cisco 7200 Series Routers
port {slot/port:ds0-group-no} | {slot-number/subunit-number/port}
no port {slot/port:ds0-group-no} | {slot-number/subunit-number/port}
Syntax Description
For the Cisco 1750 Router:
slot-number
Slot number in the router in which the VIC is installed. Valid entries are from 0
to 2, depending on the slot in which it has been installed.
port
Voice port number. Valid entries are 0 or 1.
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port (dial peer)
For the Cisco 2600 and 3600 Series Routers:
slot-number
Slot number in the Cisco router where the voice interface card is installed. Valid
entries are from 0 to 3, depending on the slot where it has been installed.
subunit-number
Subunit on the voice interface card where the voice port is located. Valid entries
are 0 or 1.
port
Voice port number. Valid entries are 0 or 1.
slot
Router location where the voice port adapter is installed. Valid entries are from
0 to 3.
port
Voice interface card location. Valid entries are 0 or 3.
dso-group-no
Indicates the defines 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.
For the Cisco MC3810 Multiservice Concentrator:
slot/port
The slot variable specifies the slot number in the Cisco router where the voice
interface card is installed. The only valid entry is 1.
The port variable specifies the voice port number. Valid ranges are as follows:
Analog voice ports: from 1 to 6.
Digital T1: from 1 to 24.
Digital E1: from 1 to 15, and from 17 to 31.
For the Cisco AS5300 Universal Access Server::
controller number
Specifies the T1 or E1 controller.
:D
Indicates the D channel associated with ISDN PRI.
For the Cisco AS5800 Universal Gateway:
shelf/slot/port
Specifies the T1 or E1 controller on the T1 card. Valid entries for the shelf
variable is 0 to 9999. Valid entries for the slot value is 0 to 11. Valid entries for
the port variable is 0 to 11.
shelf/slot/parent:por Specifies the T1 controller on the T3 card. Valid entries for the shelf variable is
t
0 to 9999. Valid entries for the slot variable is 0 to 11. Valid entries for the port
variable is 1 to 28. The value for the parent variable is always 0.
:D
Indicates the D channel associated with ISDN PRI.
For the Cisco 7200 Series Routers:
slot
Router location where the voice port adapter is installed. Valid entries are from
0 to 3.
port
Voice interface card location. Valid entries are 0 or 1.
dso-group-no
Indicates the defines 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.
slot-number
Indicates the slot number in the Cisco router where the voice interface card is
installed. Valid entries are from 0 to 3, depending on the slot where it has been
installed.
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port (dial peer)
subunit-number
Indicates the subunit on the voice interface card where the voice port is located.
Valid entries are 0 or 1.
port
Indicates the voice port number. Valid entries are 0 or 1.
Defaults
No port is configured.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced (Cisco 3600 series router).
Usage Guidelines
11.3(3)T
Port-specific values for the Cisco 2600 were added.
11.3(1)MA
Port-specific values for the Cisco MC3810 multiservice concentrator were
added.
12.0(3)T
Port-specific values for the Cisco AS5300 universal access server were
added.
12.0(4)T
Support was added for the Cisco uBR924 platform.
12.0(7)T
Port-specific values for the Cisco AS5800 universal gateway were added.
This command is used for calls incoming from a telephony interface to select an incoming dial peer and
for calls coming from the VoIP network to match a port with the selected outgoing dial peer.
This command applies only to POTS peers.
Examples
The following example associates a Cisco 3600 series router POTS dial peer 10 with voice port 1, which
is located on subunit 0, and accessed through port 0:
dial-peer voice 10 pots
port 1/0/0
The following example associates a Cisco MC3810 multiservice concentrator POTS dial peer 10 with
voice port 0, which is located in slot 1:
dial-peer voice 10 pots
port 1/0
The following example associates a Cisco AS5300 universal access server POTS dial peer 10 with voice
port 0:D:
dial-peer voice 10 pots
port 0:D
The following example associates a Cisco AS5800 universal gateway POTS dial peer 10 with voice port
1/0/0:D (T1 card):
dial-peer voice 10 pots
port 1/0/0:D
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port media
port media
To specify the serial interface to which the local video codec is connected for a local video dial peer, use
the port media command in video dial-peer configuration mode. To remove any configured locations
from the dial peer, use the no form of this command.
port media interface
no port media
Syntax Description
interface
Defaults
No interface is specified.
Command Modes
Video dial-peer configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM video dial-peer configuration on the
Cisco MC3810 multiservice concentrator.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
Serial interface to which the local codec is connected. Valid entries are the
numbers 0 or 1.
On a Cisco MC3810 multiservice concentrator local video dial peer designated as 10, the following
example shows serial interface 0 as the specified interface for the codec:
dial-peer video 10 videocodec
port media Serial0
Related Commands
Command
Description
port signal
Specifies the slot location of the VDM and the port location of the
EIA/TIA-366 interface for signaling.
show dial-peer video
Displays dial-peer configuration.
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port signal
port signal
To specify the slot location of the video dialing module (VDM) and the port location of the EIA/TIA-366
interface for signaling for a local video dial peer, use the port signal command in video dial-peer
configuration mode. To remove any configured locations from the dial peer, use the no form of this
command.
port signal slot/port
no port signal
Syntax Description
slot/
Slot location of the VDM. Valid values are 1 and 2.
port
Port location of the EIA/TIA-366 interface. The Cisco MC3810
multiservice concentrator VDM has only one port, so the port value is
always 0.
Defaults
No locations are specified.
Command Modes
Video dial-peer configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM video dial-peer configuration on the
Cisco MC3810 multiservice concentrator.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
On a Cisco MC3810 multiservice concentrator, the following example shows how to set up the VDM
and EIA/TIA-366 interface locations for the local video dial peer designated as 10:
dial-peer video 10 videocodec
port signal 1/0
Related Commands
Command
Description
port media
Specifies the serial interface to which the local video codec is connected.
show dial-peer video
Displays dial-peer configuration.
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pots call-waiting
pots call-waiting
To enable the local call waiting feature on a Cisco 800 series router, use the global configuration
pots call-waiting command in global configuration mode. To disable the local call waiting feature, use
the no form of this command.
pots call-waiting {local | remote}
no pots call-waiting {local | remote}
Syntax Description
local
Enable call waiting on a local basis for the routers.
remote
Rely on the network provider service instead of the router to hold calls.
Defaults
The call waiting default is remote if the Call Waiting feature is not configured. In that case, the call
holding pattern follows the settings of the service provider rather than those of the router.
Command Modes
Global configuration
Command History
Release
Modification
12.1.(2)XF
The command pots call-waiting was introduced on the Cisco 800 series
routers.
Usage Guidelines
To display the call waiting setting, use the show running-config or show pots status command. The
ISDN call waiting service is used if it is available on the ISDN line connected to the router even if local
call waiting is configured on the router. That is, if the ISDN line supports call waiting, the local call
waiting configuration on the router is ignored.
Examples
The following example enables local call waiting on a router:
pots call-waiting local
Related Commands
Command
Description
call-waiting
Configure Call Waiting for a specific dial peer.
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pots country
pots country
To configure your connected telephones, fax machines, or modems to use country-specific default
settings for each physical characteristic, use the pots country command in global configuration mode.
To disable the use of country-specific default settings for each physical characteristic, use the no form
of this command.
pots country country
no pots country country
Syntax Description
country
Defaults
A default country is not defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
Specifies the country in which your router is located.
This command applies to the Cisco 800 series routers.
If you need to change a country-specific default setting of a physical characteristic, you can use the
associated command listed in the “Related Commands” section. Enter the pots country ? command to
get a list of supported countries and the code you must enter to indicate a particular country.
Examples
The following example specifies that the devices connected to the telephone ports use default settings
specific to Germany for the physical characteristics:
pots country de
Related Commands
Command
Description
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series
router fail to detect that a calling party has disconnected.
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
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pots country
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems
connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and
other information on the telephone interfaces on a Cisco 800 series router.
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pots dialing-method
pots dialing-method
To specify how the router collects and sends digits dialed on your connected telephones, fax machines,
or modems, use the pots dialing-method command in global configuration mode. To disable the
specified dialing method, use the no form of this command.
pots dialing-method {overlap | enblock}
no pots dialing-method {overlap | enblock}
Syntax Description
overlap
The router sends each digit dialed in a separate message.
enblock
The router collects all digits dialed and sends the digits in one message.
Defaults
The default depends on the setting of the pots country command. For more information, see to the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
This command applies to Cisco 800 series routers.
To interrupt the collection and transmission of dialed digits, enter a pound sign (#), or stop dialing digits
until the interdigit timer runs out (10 seconds).
Examples
The following example specifies that the router uses the enblock dialing method:
pots dialing-method enblock
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series router
fail to detect that a calling party has disconnected.
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pots dialing-method
Command
Description
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected
to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and
other information on the telephone interfaces on a Cisco 800 series router.
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pots disconnect-supervision
pots disconnect-supervision
To specify how a router notifies the connected telephones, fax machines, or modems when the calling
party has disconnected, use the pots disconnect-supervision command in global configuration mode.
To disable the specified disconnect method, use the no form of this command.
pots disconnect-supervision {osi | reversal}
no pots disconnect-supervision {osi | reversal}
Syntax Description
osi
Open switching interval (OSI) is the duration for which DC voltage applied
between tip and ring conductors of a telephone port is removed.
reversal
Polarity reversal of tip and ring conductors of a telephone port.
Defaults
The default depends on the setting of the pots country command. For more information, see to the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
This command applies to Cisco 800 series routers.
Most countries except Japan typically use the osi option. Japan typically uses the reversal option.
Examples
The following example specifies that the router uses the OSI disconnect method:
pots disconnect-supervision osi
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots
dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on
your connected telephones, fax machines, or modems.
pots
disconnect-time
Specifies the interval in which the disconnect method is applied if telephones,
fax machines, or modems connected to a Cisco 800 series router fail to detect
that a calling party has disconnected.
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pots disconnect-supervision
Command
Description
pots
distinctive-ring-gu
ard-time
Specifies the delay in which a telephone port can be rung after a previous call
is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems
connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to
a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other
information on the telephone interfaces on a Cisco 800 series router.
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pots disconnect-time
pots disconnect-time
To specify the interval in which the disconnect method is applied if your connected telephones, fax
machines, or modems fail to detect that a calling party has disconnected, use the pots disconnect-time
command in global configuration mode. To disable the specified disconnect interval, use the no form of
this command.
pots disconnect-time interval
no pots disconnect-time interval
Syntax Description
interval
Defaults
The default depends on the setting of the pots country command. For more information, see the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
Number from 50 to 2000 (milliseconds).
This command applies to Cisco 800 series routers.
The pots disconnect-supervision command configures the disconnect method.
Examples
The following example specifies that the connected devices apply the configured disconnect method for
100 milliseconds after a calling party disconnects:
pots disconnect-time 100
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
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pots disconnect-time
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems
connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and
other information on the telephone interfaces on a Cisco 800 series router.
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pots distinctive-ring-guard-time
pots distinctive-ring-guard-time
To specify the delay in which a telephone port can be rung after a previous call is disconnected, use the
pots distinctive-ring-guard-time command in global configuration mode. To disable the specified
delay, use the no form of this command.
pots distinctive-ring-guard-time milliseconds
no pots distinctive-ring-guard-time milliseconds
Syntax Description
milliseconds
Defaults
The default depends on the setting of the pots country command. For more information, see the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Number from 0 to 1000 (milliseconds).
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example specifies that a telephone port can be rung 100 milliseconds after a previous call
is disconnected:
pots distinctive-ring-guard-time 100
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series
router fail to detect that a calling party has disconnected.
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
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pots distinctive-ring-guard-time
Command
Description
pots line-type
Specifies the impedance of telephones, fax machines, or modems
connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
ring
Sets up a distinctive ring for telephones, fax machines, or modems
connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and
other information on the telephone interfaces on a Cisco 800 series router.
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pots encoding
pots encoding
To specify the pulse code modulation (PCM) encoding scheme for your connected telephones, fax
machines, or modems, use the pots encoding command in global configuration mode. To disable the
specified PCM encoding scheme, use the no form of this command.
pots encoding {alaw | ulaw}
no pots encoding {alaw | ulaw}
Syntax Description
alaw
International Telecommunication Union Telecommunication
Standardization Section (ITU-T) PCM encoding scheme used to represent
analog voice samples as digital values.
ulaw
North American PCM encoding scheme used to represent analog voice
samples as digital values.
Defaults
The default depends on the setting of the pots country command. For more information, see the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
This command applies to Cisco 800 series routers.
Europe typically uses the alaw option. North America typically uses the ulaw option.
Examples
The following example specifies alaw as the PCM encoding scheme:
pots encoding alaw
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
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pots encoding
Command
Description
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series
router fail to detect that a calling party has disconnected.
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
pots line-type
Specifies the impedance of telephones, fax machines, or modems
connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and
other information on the telephone interfaces on a Cisco 800 series router.
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pots line-type
pots line-type
To specify the impedance of your connected telephones, fax machines, or modems, use the pots
line-type command in global configuration mode. To disable the specified line type, use the no form of
this command.
pots line-type {type1 | type2 | type3}
no pots line-type {type1 | type2 | type3}
Syntax Description
type1
Runs at 600 ohms.
type2
Runs at 900 ohms.
type3
Runs at 300 or 400 ohms.
Defaults
The default depends on the setting of the pots country command. For more information, see the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example specifies type1 as the line type:
pots line-type type1
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series router
fail to detect that a calling party has disconnected.
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pots line-type
Command
Description
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other
information on the telephone interfaces on a Cisco 800 series router.
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pots ringing-freq
pots ringing-freq
To specify the frequency on the Cisco 800 series router at which your connected telephones, fax
machines, or modems ring, use the pots ringing-freq command in global configuration mode. To
disable the specified ringing frequency, use the no form of this command.
pots ringing-freq {20Hz | 25Hz | 50Hz}
no pots ringing-freq {20Hz | 25Hz | 50Hz}
Syntax Description
20Hz
Connected devices ring at 20 Hz.
25Hz
Connected devices ring at 25 Hz.
50Hz
Connected devices ring at 50 Hz.
Defaults
The default depends on the setting of the pots country command. For more information, see the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example specifies a ringing frequency of 50 Hz:
pots ringing-freq 50Hz
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco
800 series router to use country-specific default settings for each
physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits
dialed on your connected telephones, fax machines, or modems.
pots
disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected
telephones, fax machines, or modems when the calling party has
disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series
router fail to detect that a calling party has disconnected.
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pots ringing-freq
Command
Description
pots
Specifies the delay in which a telephone port can be rung after a
distinctive-ring-guard-time previous call is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems
connected to a Cisco 800 series router.
pots silence-time
Specifies the interval of silence after a calling party disconnects
(Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones,
fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and
other information on the telephone interfaces on a Cisco 800 series
router.
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pots silence-time
pots silence-time
To specify the interval of silence after a calling party disconnects, use the pots silence-time command
in global configuration mode. To disable the specified silence time, use the no form of this command.
pots silence-time interval
no pots silence-time interval
Syntax Description
interval
Defaults
The default depends on the setting of the pots country command. For more information, see the pots
country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Number from 0 to 10 (seconds).
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example specifies 10 seconds as the interval of silence:
pots silence-time 10
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series router
fail to detect that a calling party has disconnected.
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
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pots silence-time
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected
to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax
machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other
information on the telephone interfaces on a Cisco 800 series router.
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pots tone-source
pots tone-source
To specify the source of dial, ringback, and busy tones for your connected telephones, fax machines, or
modems, use the pots tone-source command in global configuration mode. To disable the specified tone
source, use the no form of this command.
pots tone-source {local | remote}
no pots tone-source {local | remote}
Syntax Description
local
Router supplies the tones.
remote
Telephone switch supplies the tones.
Defaults
The default setting is local.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series routers.
Usage Guidelines
This command applies to Cisco 800 series routers.
This command applies only to ISDN lines connected to a EURO-ISDN (NET3) switch.
Examples
The following example specifies remote as the tone source:
pots tone-source remote
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800
series router to use country-specific default settings for each physical
characteristic
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed
on your connected telephones, fax machines, or modems.
pots
Specifies how a Cisco 800 series router notifies the connected telephones,
disconnect-supervision fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if
telephones, fax machines, or modems connected to a Cisco 800 series router
fail to detect that a calling party has disconnected.
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pots tone-source
Command
Description
pots
Specifies the delay in which a telephone port can be rung after a previous
distinctive-ring-guard- call is disconnected (Cisco 800 series routers).
time
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or
modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected
to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems
connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800
series router).
show pots status
Displays the settings of the telephone port physical characteristics and other
information on the telephone interfaces on a Cisco 800 series router.
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pre-dial delay
pre-dial delay
To configure a delay on an Foreign Exchange Office (FXO) interface between the beginning of the
off-hook state and the initiation of dual-tone multifrequency (DTMF) signaling, use the pre-dial delay
command in voice-port configuration mode. To restore the default value, use the no form of the
command.
pre-dial delay seconds
no pre-dial delay
Syntax Description
seconds
Defaults
The default setting is 1 second.
Command Modes
Voice-port configuration
Command History
Release
Usage Guidelines
Delay, in seconds, before signaling begins. Valid values are from 0 to 10.
Modification
11.(7)T
This command was introduced on the Cisco 3600 series routers.
12.0(2)T
This command was integrated into 12.0(2)T.
This command applies to Cisco 3600 series routers.
To disable the command, set the delay to 0. When an FXO interface begins to draw loop current
(off-hook state), a delay is required between the initial flow of loop current and the beginning of
signaling. Some devices initiate signaling too quickly, resulting in redial attempts. The pre-dial delay
command allows a signaling delay.
Examples
The following example sets a predial delay value of 3 seconds on the FXO port of a Cisco 3600 series
router:
voice-port 1/0/0
pre-dial delay 3
Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timing delay-duration
Configures delay dial signal duration for a specified voice port.
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preference
preference
To indicate the preferred order of a dial peer within a hunt group, use the preference command in
dial-peer configuration mode. To remove the preference value on the voice port, use the no form of this
command.
preference value
no preference value
Syntax Description
value
Defaults
0 (highest preference)
Command Modes
Dial-peer configuration
Command History
Release
Usage Guidelines
An integer from 0 to 10, where the lower the number, the higher the preference.
The default value is 0 (highest preference).
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice concentrator.
12.0(3)T
This command was supported on the Cisco 2600 series router and 3600 series
routers.
12.0(4)T
Support was added for VoFR dial peers on the Cisco 2600 series router and
3600 series routers.
This command applies to plain old telephone service (POTS) dial peers, Voice over IP (VoIP) dial peers,
Voice over Frame Relay (VoFR) dial peers, and Voice over ATM (VoATM) dial peers on the Cisco
MC3810 multiservice concentrator.
Use the preference command to indicate the preference order for matching dial peers in a rotary group.
Setting the preference enables the desired dial peer to be selected when multiple dial peers within a hunt
group are matched for a dial string.
Note
If POTS and voice-network peers are mixed in the same hunt group, the POTS dial peers must have
priority over the voice-network dial peers.
Use this command with the Rotary Calling Pattern feature.
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preference
The hunting algorithm precedence is configurable. For example, if you wish a call processing sequence
to go to destination A first, to destination B second, and to destination C third, you would assign
preference (0 being the highest priority) to the destinations in the following order:
Examples
•
Preference 0 to A
•
Preference 1 to B
•
Preference 2 to C
The following example configures POTS dial peer 10 with a preference of 1, POTS dial peer 20 with a
preference of 2, and VoFR dial peer 30 to a preference of 3:
dial-peer voice 10 pots
destination pattern 5552150
preference 1
exit
dial-peer voice 20 pots
destination pattern 5552150
preference 2
exit
dial-peer voice 30 vofr
destination pattern 5552150
preference 3
exit
The following examples show different dial peer configurations using the preference command:
Dialpeer
1
2
3
4
5
destpat
4085551048
408555
408555
..........
..........
preference
0 (highest)
0
1 (lower)
1
0
session-target
jmmurphy-voip
sj-voip
backup-sj-voip
0:D
(interface)
anywhere-voip
If the destination number is 4085551048, the order of attempts will be 1, 2, 3, 5, 4:
Dialpeer
destpat
preference
1
408555
0
2
4085551048
1
3
4085551
0
4 ..............4085551.........0
If the number dialed is 4085551048, the order will be 2, 3, 4, 1.
Note
The default behavior is that the longest matching dial peer supersedes the preference value.
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preference
Related Commands
Command
Description
called-number
(dial-peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call
leg when using a static FRF.11 trunk connection.
codec (dial-peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay dial
peer.
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence
setting.
destination-pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory
number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over Enables the generation of FRF.11 Annex A frames for a dial peer.
Frame Relay)
session protocol
Establishes a session protocol for calls between the local and remote routers
via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination
gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
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prefix
prefix
To specify the prefix of the dialed digits for a dial peer, use the prefix command in dial-peer
configuration mode. To disable this feature, use the no form of this command.
prefix string
no prefix
Syntax Description
string
Defaults
Null string
Command Modes
Dial-peer configuration
Command History
Release
Usage Guidelines
Integers that represent the prefix of the telephone number associated with the
specified dial peer. Valid numbers are 0 through 9, and a comma (,). Use a
comma to include a pause in the prefix.
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.0(4)XJ
This command was modified for store and forward fax and to add support on the
Cisco AS5300 universal access server.
12.1(1)T
Modifications in Cisco IOS Release 12.0(4)XJ were integrated into
Cisco IOS Release 12.1(1)T.
Use the prefix command to specify a prefix for a specific dial peer. When an outgoing call is initiated
to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone
number associated with the dial peer.
If you want to configure different prefixes for dialed numbers on the same interface, you need to
configure different dial peers.
This command is applicable only to plain old telephone service (POTS) dial peers. This command
applies to off-ramp store and forward fax functions.
Examples
The following example specifies a prefix of 9 and then a pause:
dial-peer voice 10 pots
prefix 9,
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prefix
Related Commands
Command
Description
answer-address
Specifies the full E.164 telephone number to be used to identify the dial peer of
an incoming call.
destination-pattern Specifies either the prefix or the full E.164 telephone number (depending on
your dial plan) to be used for a dial peer.
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pri-group
pri-group
To specify an ISDN PRI on a channelized T1 or E1 controller, use the pri-group command in controller
configuration mode. To remove the ISDN PRI configuration, use the no form of this command.
pri-group timeslots timeslot-range
no pri-group
Syntax Description
timeslots timeslot-range
Defaults
There is no ISDN-PRI group configured.
Command Modes
Controller configuration
Command History
Release
Usage Guidelines
Specifies a single range of values. For T1, the allowable range is from 1 to
23. For E1, the allowable range is from 1 to 15.
Modification
11.0
This command was introduced.
12.0(2)T
This command was introduced for the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was introduced for the Cisco 2600 and Cisco 3600 series
routers.
12.1(2)T
The modifications in Cisco IOS Release 12.0(7)XK were integrated into
Cisco IOS Release 12.1(2)T.
The pri-group command applies to the configuration of Voice over Frame Relay and Voice over ATM
on the Cisco MC3810 multiservice concentrator and the Cisco 2600 and Cisco 3600 series routers.
Before you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1
controller.
Note
Examples
Only one PRI group can be configured on a controller.
The following example configures ISDN-PRI on all time slots of controller E1 on a Cisco 2600 series
router router:
controller E1 4/1
pri-group timeslots 1-7,16
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pri-group
Related Commands
Command
Description
isdn switch-type
Configures the Cisco 2600 series router router PRI interface
to support QSIG signaling.
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pri-group nec-fusion
pri-group nec-fusion
To configure your NEC PBX to support Fusion Call Control Signaling (FCCS), use the pri-group
nec-fusion command in controller configuration mode. To disable FCCS, use the no form of this
command.
pri-group nec-fusion {pbx-ip-address | pbx-ip-host-name} pbx-port number
no pri-group nec-fusion {pbx-ip-address | pbx-ip-host-name} pbx-port number
Syntax Description
pbx-ip-address
The IP address of the NEC PBX.
pbx-ip-host-name
The host name of the NEC PBX.
pbx-port number
Choose a port number for the PBX. The range for the PBX port is 49152 to
65535. If you do not specify a port number, the default value of 55000 will
be used. If this value is already in use, the next greater value will be used.
Defaults
Port number 55000
Command Modes
Controller configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco AS5300 universal access server.
12.2(1)
This command was modified to add support for Setup messages from a POTS
dial peer.
Usage Guidelines
This command is used only if the PBX in your configuration is an NEC PBX, and if you are configuring
it to run FCCS and not QSIG signaling.
Examples
The following example shows how to configure this NEC PBX to use FCCS:
pri-group nec-fusion 172.31.255.255 pbx-port 60000
Related Commands
Command
Description
isdn protocol-emulate Configures the Layer 2 and Layer 3 port protocol of a BRI voice port or a
PRI interface to emulate NT (network) or TE (user) functionality.
isdn switch type
Configures the Cisco AS5300 universal access server PRI interface to
support QSIG signaling.
show cdapi
Displays the CDAPI.
show rawmsg
Displays the raw messages owned by the required component.
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progress_ind
progress_ind
To set a specific progress indicator (PI) in call Setup, Progress, or Connect messages from an H.323
Voice over IP (VoIP) gateway, use the progress_ind command in dial-peer configuration mode. To
restore the default condition, use the no or disable forms of this command.
progress_ind {setup | connect | progress | alert} {enable pi-number | disable}
no progress_ind {setup | connect | progress | alert}
Note
Syntax Description
This command is not supported on VoIP gateways using session initiation protocol (SIP).
setup
Sets the progress indicator for Setup messages.
connect
Sets the progress indicator for Connect messages.
progress
Sets the progress indicator for Progress messages.
alert
Sets the progress indicator for Alert messages.
enable
Enables the configuration of the progress indicator.
pi-number
The progress indicator that is sent in all messages of the specified type from
the outbound dial peer. For Setup messages from plain old telephone service
(POTS) or VoIP dial peers, values are 0, 1, or 3. For Progress, Connect, or
Alert messages from a POTS dial peer, values are 1, 2, or 8.
disable
Disables the user configuration of the progress indicator.
Defaults
The default progress indicator from the switch is not intercepted or modified.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600 series, Cisco 3600 series,
Cisco 7200 series, and Cisco 7500 series routers, Cisco MC3810
multiservice concentrator, and Cisco AS5300 and Cisco AS5800 universal
access servers.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(1)
This command was modified to add support for Setup messages from a
POTS dial peer.
Usage Guidelines
The progress_ind command overrides the default progress indicator that is sent by the switch. This
enables you to set the progress indicator at the H.323 gateway, if necessary, to ensure the proper
end-to-end signaling for VoIP calls. This command sets the progress indicator only in messages from
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progress_ind
outbound dial peers that have a set destination pattern, configured by using the destination-pattern
command. If a message contains multiple progress indicators, the progress_ind command overrides
only the first progress indicator in the message.
The disable and no forms of the progress_ind command have the same result: The call messages are
not intercepted by the session application, and the default progress indicator, if any, is forwarded
unmodified.
Note
Examples
A progress indicator that is configured by using the progress_ind command will not override
the default progress indicator in a Progress message, if the Progress message is sent after
backward cut-through has occurred (for example, because an Alert message with a progress
indicator of 8 was sent before the Progress message).
The following example sets the progress indicator to 1 in Progress and Connect messages from the
number 3 POTS dial peer:
dial-peer voice 3 pots
destination-pattern 55275
progress_ind progress enable 1
progress_ind connect enable 1
Related Commands
Command
Description
dial-peer voice
Enters dial-peer configuration mode and configures a VoIP or POTS
dial peer.
destination-pattern
Specifies the telephone number that is used to identify the outbound
dial peer for the call.
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proxy h323
proxy h323
To enable the proxy feature on your router, use the proxy h323 command in global configuration mode.
To disable the proxy feature, use the no form of this command.
proxy h323
no proxy h323
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 series and Cisco 3600
series routers.
Usage Guidelines
If the multimedia interface is not enabled using the proxy h323 command, or if no gatekeeper is
available, starting the proxy allows it to attempt to locate these resources. No calls will be accepted until
the multimedia interface and the gatekeeper are found.
Examples
The following example turns on the proxy feature:
proxy h323
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R Through Sh
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax
applications. The commands are presented in alphabetical order beginning with R. Some commands
required for configuring voice, video, and fax may be found in other Cisco IOS command references.
Use the command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS
Voice, Video, and Fax Configuration Guide.
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register e164
register e164
To configure a gateway to register or deregister (remove the registration for) a fully qualified plain old
telephone service (POTS) dial-peer E.164 address with a gatekeeper, use the register e164 command in
dial-peer configuration mode. To deregister an E.164 address, use the no form of this command.
register e164
no register e164
Syntax Description
This command has no keywords or arguments.
Defaults
No E.164 addresses are registered until you enter this command.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(5)T
This command was introduced on the Cisco AS5300 universal access server.
Usage Guidelines
Use this command to register the E.164 address of an analog telephone line attached to a Foreign
Exchange Station (FXS) port on a router. The gateway automatically registers fully qualified E164
addresses. Use the no register e164 command to deregister an address. Use the register e164 command
to register a deregistered address.
Before you automatically or manually register an E.164 address with a gatekeeper, you must create a dial
peer (using the dial-peer command), assign an FXS port to the peer (using the port command), and
assign an E.164 address (using the destination-pattern command). The E.164 address must be a fully
qualified address. For example, +5551212, 5551212, and 4085551212 are fully qualified addresses;
408555.... is not a fully qualified address. E.164 addresses are registered only for active
interfaces—those that are not shut down. If an FXS port or its interface is shut down, the corresponding
E.164 address is deregistered.
Tips
Examples
You can use the show gateway command to find out if the gateway is connected to a gatekeeper and
if a fully qualified E.164 address is assigned to the gateway. Use the zone-prefix command at the
gatekeeper to define prefix patterns, such as 408555...., that apply to one or more gateways.
The following command sequence places the gateway in dial-peer configuration mode, assigns an E.164
address to the interface, and registers that address with the gatekeeper:
dial-peer voice 111 pots
port 1/0/0
destination-pattern 5551212
register e164
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register e164
The following commands deregister an address with the gatekeeper:
dial-peer voice 111 pots
no register e164
The following example shows that you must have a connection to a gatekeeper and define a unique E.164
address before you can register an address:
dial-peer voice 222 pots
port 1/0/0
destination 919555....
register e164
ERROR-register-e164:Dial-peer destination-pattern is not a full E.164 number
no gateway
dial-peer voice 111 pots
register e164
ERROR-register-e164:No gatekeeper
Related Commands
Command
Description
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN
directory number (depending on the dial plan) to be used for a dial peer.
dial-peer
Enters dial-peer configuration mode, defines the type of dial peer, and
defines the tag number associated with a dial peer.
port
Enables an interface on a PA-4R-DTR to operate as a concentrator port.
show gateway
Displays the current gateway status.
zone prefix
Configures the gatekeeper with knowledge of its own prefix and the prefix
of any remote zone.
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registered-caller ring
registered-caller ring
To configure the Nariwake service registered caller ring cadence, use the registered-caller ring
command in dial-peer configuration mode.
registered-caller ring cadence
Syntax Description
cadence
Defaults
The default Nariwake service registered caller ring cadence is ring 1.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1.(2)XF
The command registered-caller ring was introduced on the Cisco 800
series routers.
Usage Guidelines
A value of 0, 1, or 2. The default ring cadence for registered callers is 1
and for unregistered callers is 0. The on and off periods of ring 0 (normal
ringing signals) and ring 1 (ringing signals for the Nariwake service) are
defined in the NTT user manual.
If your ISDN line is provisioned for the I Number or dial-in services, you must also configure a dial peer
by using the destination-pattern not-provided command. Either port 1 or port 2 can be configured
under this dial peer. The router then forwards the incoming call to voice port 1. (See the “Examples”
section below.
If more than one dial peer is configured with the destination-pattern not-provided command, the router
uses the first configured dial peer for the incoming calls. To display the Nariwake ring cadence setting,
use the show run command.
Examples
The following example sets the ring cadence for registered callers to 2.
pots country jp
dial-peer voice 1 pots
registered-caller ring 2
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req-qos
req-qos
To specify the desired quality of service to be used in reaching a specified dial peer, use the req-qos
command in dial-peer configuration mode. To restore the default value for this command, use the no
form of this command.
req-qos {best-effort | controlled-load | guaranteed-delay}
no req-qos
Syntax Description
best-effort
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth
reservation.
controlled-load
Indicates that RSVP guarantees a single level of preferential service,
presumed to correlate to a delay boundary. The controlled load service uses
admission (or capacity) control to assure that preferential service is received
even when the bandwidth is overloaded.
guaranteed-delay
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate
and preferential queueing if the bandwidth reserved is not exceeded.
Defaults
best-effort
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
Usage Guidelines
This command is applicable only to VoIP dial peers.
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like
acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth
so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation
Protocol (RSVP) to request quality of service guarantees from the network.
Examples
The following example configures guaranteed-delay as the desired (requested) quality of service to a
dial peer:
dial-peer voice 10 voip
req-qos guaranteed-delay
Related Commands
Command
Description
acc-qos
Defines the acceptable QoS for any inbound and outbound call on a VoIP dial
peer.
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reset
reset
To reset a set of digital signal processors (DSPs), use the reset command in global configuration mode.
reset number
Syntax Description
number
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
12.0(5)XE
This command was introduced on the Cisco 7200 series routers.
12.0(7)T
This command was integrated into the Cisco IOS Release 12.0(7)T.
Examples
Specifies the number of DSPs to be reset. The number of DSPs ranges
from 0 to 30.
The following example displays the reset command configuration for DSP 1:
reset 1
01:24:54:%DSPRM-5-UPDOWN: DSP 1 in slot 1, changed state to up
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resource threshold
resource threshold
To configure a gateway to report H.323 resource availability to the its gatekeeper, use the resource
threshold command in gateway configuration mode. To disable gateway resource-level reporting, use
the no form of this command.
resource threshold [all] [high percentage-value] [low percentage-value]
no resource threshold
Syntax Description
all
(Optional) Applies the high- and low- parameter settings to all monitored
H.323 resources. This is the default condition.
high percentage-value
(Optional) A resource utilization level that triggers a Resource Availability
Indicator (RAI) message indicating that H.323 resource use is high. Enter a
number between 1 and 100 that represents the high-resource utilization
percentage. A value of 100 specifies high-resource usage when any H.323
resource is unavailable. The default is 90 percent.
low percentage-value
(Optional) Resource utilization level that triggers an RAI message indicating
that H.323 resource usage has dropped below the high-usage level. Enter a
number between 1 and 100 that represents the acceptable resource utilization
percentage. After the gateway sends a high-utilization message, it waits to
send the resource recovery message until the resource use drops below the
value defined by the low parameter. The default is 90 percent.
Defaults
Reports low resources when 90 percent of resources are in use, and reports resource availability when
resource use drops below 90 percent.
Command Modes
Gateway configuration
Command History
Release
Modification
12.0(5)T
This command was introduced on the Cisco AS5300 universal access server.
Usage Guidelines
The resource threshold command defines the resource load levels that trigger Resource Availability
Indicator (RAI) messages. To view the monitored resources, enter the show gateway command.
The monitored H.323 resources include digital signal processor (DSP) channels and DS0s. Use the show
call resource voice stats command to see the total amount of resources available for H.323 calls.
Note
The DS0 resources that are monitored for H.323 calls are limited to the ones that are associated with
a voice POTS dial peer.
See the dial-peer configuration commands for details on how to associate a dial peer with a PRI or CAS
group.
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resource threshold
When any monitored H.323 resources exceed the threshold level defined by the high parameter, the
gateway sends an RAI message to the gatekeeper with the AlmostOutOfResources field flagged. This
message reports high resource usage.
When all gateway H.323 resources drop below the level defined by the low parameter, the gateway sends
the RAI message to the gatekeeper with the AlmostOutOfResources field cleared.
When a gatekeeper can choose between multiple gateways for call completion, the gatekeeper uses
internal priority settings and gateway resource statistics to determine which gateway to use. When all
other factors are equal, a gateway that has available resources will be chosen over a gateway that has
reported limited resources.
Examples
The following command defines the H.323 resource limits for a gateway:
resource threshold high 70 low 60
Related Commands
Command
Description
show call resource voice
stats
Displays resource statistics for an H.323 gateway.
show call resource voice
threshold
Displays the threshold configuration settings and status for an H.323
gateway.
show gateway
Displays the current gateway status.
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response-timeout
response-timeout
To configure the maximum time to wait for a response from a server, use the response-timeout
command in settlement configuration mode. To restore the default value of this command, use the no
form of this command.
response-timeout number
no response-timeout number
Syntax Description
number
Defaults
The default response timeout is one (1) second.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Response waiting time in seconds.
Usage Guidelines
If no response is received within the response-timeout time limit, the current connection ends, and the
router attempts to contact the next service point.
Examples
The following example illustrates a response-timeout set to 1 second.
settlement 0
response-timeout 1
Related Commands
Command
Description
connection-timeout
Configures the time for which a connection is maintained after completion
of a communication exchange.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for
communication with a settlement provider.
retry-delay
Sets the time between attempts to connect with the settlement provider.
retry-limit
Sets the maximum number of attempts to connect to the provider.
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response-timeout
Command
Description
session-timeout
Sets the interval for closing the connection when there is no input or output
traffic.
settlement
Enters settlement mode and specifies the attributes specific to a settlement
provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no
shutdown
Deactivates the settlement provider/activates the settlement provider.
type
Configures an SAA-RTR operation type.
url
Specifies the Internet service provider address.
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retry-delay
retry-delay
To set the time between attempts to connect with the settlement provider, use the retry-delay command
in settlement configuration mode. To restore the default value, use the no form of this command.
retry-delay number
no retry-delay
Syntax Description
number
Defaults
The default retry delay is two seconds.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 and 3600 series routers and
the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Length of time (in seconds) between attempts to connect with the settlement
provider. The valid range for retry delay is from 1 to 600 seconds.
Usage Guidelines
After exhausting all service points for the provider, the router is delayed for the specified length of time
before resuming connection attempts.
Examples
The following example sets a retry value of 15 seconds:
settlement 0
relay-delay 15
Related Commands
Command
Description
connection-timeout
Configures the time for which a connection is maintained after completion
of a communication exchange.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for
communication with a settlement provider.
response-timeout
Configures the maximum time to wait for a response from a server.
retry-limit
Sets the maximum number of attempts to connect to the provider.
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retry-delay
Command
Description
session-timeout
Sets the interval for closing the connection when there is no input or output
traffic.
settlement
Enters settlement configuration mode and specifies the attributes specific to
a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no
shutdown
Deactivates the settlement provider/activates the settlement provider.
type
Configures an SAA-RTR operation type.
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retry-limit
retry-limit
To set the maximum number of attempts to connect to the provider, use the retry-limit command in
settlement configuration mode. To restore the default value, use the no form of this command.
retry-limit number
no retry-limit number
Syntax Description
number
Defaults
The default retry limit is one (1) retry.
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Maximum number of connection attempts in addition to the first attempt.
Usage Guidelines
If no connection is established after the configured retries, the router ceases connection attempts. The
retry limit number does not count the initial connection attempt. A retry limit of one (default) results in
a total of two connection attempts to every service point.
Examples
The following example sets the number of retries to 1:
settlement 0
retry-limit 1
Related Commands
Command
Description
connection-timeout
Configures the time for which a connection is maintained after a
communication exchange is complete.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for
communication with a settlement provider.
response-timeout
Configures the maximum time to wait for a response from a server.
retry-delay
Sets the time between attempts to connect with the settlement provider.
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retry-limit
Command
Description
session-timeout
Sets the length of interval for closing the connection when there is no input
or output traffic.
settlement
Enters settlement mode and specifies the attributes specific to a settlement
provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Brings up the settlement provider.
type
Configures an SAA-RTR operation type.
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retry (SIP user-agent)
retry (SIP user-agent)
To configure the number of retry attempts for Session Initiation Protocol (SIP) messages, use the retry
command in SIP user-agent configuration mode. To reset this command to the default value, use the no
form of this command.
retry {invite number | response number | bye number | cancel number}
no retry {invite number | response number | bye number | cancel number}
Syntax Description
invite number
Number of INVITE retries: 1 through 10 are valid inputs; default = 6.
response number
Number of RESPONSE retries: 1 through 10 are valid inputs; default = 6.
bye number
Number of BYE retries: 1 through 10 are valid inputs; default = 10.
cancel number
Number of CANCEL retries: 1 through 10 are valid inputs; default = 10.
Defaults
invite: 6
response: 6
bye: 10
cancel: 10
Command Modes
SIP user-agent configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
In the following example, the number of invite retries has been set to 5.
sip-ua
retry invite 5
Related Commands
Command
Description
sip-ua
Enables the sip-ua configuration commands, with which you configure the
user agent.
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ring
ring
To set up a distinctive ring for your connected telephones, fax machines, or modems, use the ring
command in interface configuration mode. To disable the specified distinctive ring, use the no form of
this command.
ring cadence-number
no ring cadence-number
Syntax Description
cadence-number
Number from 0 through 2:
•
Type 0 is a primary ringing cadence—default ringing cadence
for the country your router is in.
•
Type 1 is a distinctive ring—0.8 seconds on, 0.4 seconds off,
0.8 seconds on, 0.4 seconds off.
•
Type 2 is a distinctive ring—0.4 seconds on, 0.2 seconds off,
0.4 seconds on, 0.2 seconds off, 0.8 seconds on, 4 seconds
off.
Defaults
The default is 0.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series router.
Usage Guidelines
This command applies to Cisco 800 series routers.
You can specify this command when creating a dial peer. This command will not work if it is not
specified within the context of a dial peer. For information on creating a dial peer, refer to the
Cisco 800 Series Routers Software Configuration Guide.
Examples
The following example specifies the type 1 distinctive ring:
ring 1
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ring
Related Commands
Command
Description
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an
ISDN directory number (depending on the dial plan) to be used for a
dial peer.
dial-peer voice
Enters dial-peer configuration mode, defines the type of dial peer, and
defines the tag number associated with a dial peer.
no call-waiting
Disables call waiting.
port (dial-peer)
Enables an interface on a PA-4R-DTR port adapter to operate as a
concentrator port.
pots
Specifies a delay in which a telephone port can be rung after a previous
distinctive-ring-guard-time call is disconnected (for Cisco 800 series routers).
ring
Sets up a distinctive ring for telephones, fax machines, or modems
connected to a Cisco 800 series router.
show dial-peer voice
Displays configuration information and call statistics for dial peers.
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ring cadence
ring cadence
To specify the ring cadence for a Foreign Exchange Station (FXS) voice port, use the ring cadence
command in voice-port configuration mode. To restore the default value, use the no form of this
command.
ring cadence {pattern-number | define pulse interval}
no ring cadence
Syntax Description
pattern-number
Predefined ring cadence patterns. Each pattern specifies a ring-pulse
time and a ring-interval time.
•
pattern01—2 seconds on, 4 seconds off
•
pattern02—1 second on, 4 seconds off
•
pattern03—1.5 seconds on, 3.5 seconds off
•
pattern04—1 second on, 2 seconds off
•
pattern05—1 second on, 5 seconds off
•
pattern06—1 second on, 3 seconds off
•
pattern07—0.8 second on, 3.2 seconds off
•
pattern08—1.5 seconds on, 3 seconds off
•
pattern09—1.2 seconds on, 3.7 seconds off
•
pattern09—1.2 seconds on, 4.7 seconds off
•
pattern11—0.4 second on, 0.2 second off, 0.4 second on, 2 seconds
off
•
pattern12—0.4 second on, 0.2 second off, 0.4 second on, 2.6
seconds off
define
User-definable ring cadence pattern. Each number pair specifies one
ring-pulse time and one ring-interval time. You must enter numbers in
pairs, and you can enter from 1 to 6 pairs. The second number in the last
pair that you enter specifies the interval between rings.
pulse
A number (1 or 2 digits) specifying ring pulse (on) time in hundreds of
milliseconds.
The range is from 1 to 50, for pulses of 100 ms to 5000 ms.
For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
interval
A number (1 or 2 digits) specifying ring interval (off) time in hundreds
of milliseconds.
The range is from 1 to 50, for pulses of 100 to 5000 ms.
For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
Defaults
Ring cadence defaults to the pattern you specify with the cptone command.
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ring cadence
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.0(7)XK
This command was first supported on the Cisco 2600 and 3600 series routers,
and the patternXX keyword was introduced.
12.1(2)T
This command was integrated into the 12.1(2)T release.
Usage Guidelines
The patternXX keyword provides preset ring cadence patterns for use on any platform. The define
keyword allows you to create a custom ring cadence. On the Cisco 2600 and 3600 series routers, only
one or two pairs of digits can be entered under the define keyword.
Examples
The following example configures the ring cadence for 1 second on and 4 seconds off on voice port 1/1
on a Cisco MC3810 multiservice concentrator:
voice-port 1/1
ring cadence pattern02
The following example configures the ring cadence for 1 second on, 1 second off, 1 second on, and
5 seconds off on voice port 1/2 on a Cisco MC3810 multiservice concentrator:
voice-port 1/2
ring cadence define 10 10 10 50
The following example configures the ring cadence for 1 second on and 2 seconds off on voice port 1/0/0
on a Cisco 2600 or 3600 series router:
voice-port 1/0/0
ring cadence pattern04
Related Commands
Command
Description
ring frequency
Specifies the ring frequency for a specified FXS voice port.
cptone
Specifies the default tone, ring, and cadence settings according to country.
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ring frequency
ring frequency
To specify the ring frequency for a specified Foreign Exchange Station (FXS) voice port, use the ring
frequency command in voice-port configuration mode. To restore the default value, use the no form of
this command.
ring frequency number
no ring frequency number
Syntax Description
number
Defaults
25 Hz on the Cisco 3600 series routers and 20 Hz on the Cisco MC3810 multiservice concentrators.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco MC3810 multiservice
concentrator.
Usage Guidelines
Ring frequency (hertz) used in the FXS interface. Valid entries on the
Cisco 3600 series are 25 and 50. Valid entries on the Cisco MC3810
multiservice concentrator are 20 and 30.
Use the ring frequency command to select a specific ring frequency for an FXS voice port. Use the no
form of this command to reset the default value. The ring frequency you select must match the connected
equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring
frequency is usually country-dependent. You should take into account the appropriate ring frequency for
your area before configuring this command.
This command does not affect ringback, which is the ringing a user hears when placing a remote call.
Examples
The following example configures the ring frequency on the Cisco 3600 series for 25 Hz:
voice-port 1/0/0
ring frequency 25
The following example configures the ring frequency on the Cisco MC3810 multiservice concentrator
for 20 Hz:
voice-port 1/1
ring frequency 20
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ring frequency
Related Commands
Command
Description
ring cadence
Specifies the ring cadence for an FXS voice port on the Cisco MC3810
multiservice concentrator.
ring number
Specifies the number of rings for a specified FXO voice port.
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ring number
ring number
To specify the number of rings for a specified Foreign Exchange Office (FXO) voice port, use the ring
number command in voice-port configuration mode. To restore the default value, use the no form of this
command.
ring number number
no ring number number
Syntax Description
number
Defaults
One ring
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series router.
Usage Guidelines
Number of rings detected before answering the call. Valid entries are
numbers from 1 to 10. The default is 1.
Use the ring number command to set the maximum number of rings to be detected before answering a
call over an FXO voice port. Use the no form of this command to reset the default value, which is one
ring.
Normally, this command should be set to the default so that incoming calls are answered quickly. If you
have other equipment available on the line to answer incoming calls, you might want to set the value
higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if
the equipment online did not answer the incoming call in the configured number of rings.
This command is not applicable to Foreign Exchange Station (FXS) or E&M interfaces because they do
not receive ringing on incoming calls.
Examples
The following example on the Cisco 3600 series sets five rings as the maximum number of rings to be
detected before closing a connection over this voice port:
voice-port 1/0/0
ring number 5
The following example on the Cisco MC3810 multiservice concentrator sets five rings as the maximum
number of rings to be detected before closing a connection over this voice port:
voice-port 1/1
ring number 5
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ring number
Related Commands
Command
Description
ring frequency
Specifies the ring frequency for a specified FXS voice port.
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roaming (dial-peer)
roaming (dial-peer)
To enable the roaming capability for the dial peer, use the roaming command in dial-peer configuration
mode. To disable the roaming capability, use the no form of this command.
roaming
no roaming
Syntax Description
This command has no arguments or keywords.
Defaults
No roaming
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server‘.
Usage Guidelines
Enable the roaming capability of a dial peer if that dial peer can terminate roaming calls. If a dial peer
is dedicated to local calls only, disable the roaming capability.
The roaming dial peer must work with a roaming service provider. If the dial peer allows a roaming user
to go through, and the service provider is not roaming-enabled, the call fails.
Examples
The following example enables the roaming capability for the dial peer:
dial-peer voice 10 voip
roaming
Related Commands
Command
Description
roaming (settlement)
Enables the roaming capability for a settlement provider.
settle-call
Limits the dial peer to using only the specific clearinghouse identified
by the specified provider-number.
settlement roam-pattern
Configures a pattern to match against when determining roaming.
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roaming (settlement)
roaming (settlement)
To enable the roaming capability for a settlement provider, use the roaming command in settlement
configuration mode. To disable the roaming capability, use the no form of this command.
roaming
no roaming
Syntax Description
This command has no arguments or keywords.
Defaults
No roaming
Command Modes
Settlement configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
Usage Guidelines
Enable roaming capability of a settlement provider if that provider can authenticate a roaming user and
route roaming calls.
A roaming call is successful only if both the settlement provider and the outbound dial peer for that call
are roaming-enabled.
Examples
The following example enables the roaming capability for the settlement provider:
settlement 0
roaming
Related Commands
Command
Description
roaming (dial-peer mode) Enables the roaming capability for the dial peer.
settle-call
Limits the dial peer to using only the specific clearinghouse identified
by the specified provider-number.
settlement roam-pattern
Configures a pattern to match against when determining roaming.
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rtsp client session history duration
rtsp client session history duration
To specify how long to keep Real Time Streaming Protocol (RTSP) session history records in memory,
use the rtsp client session history duration command in global configuration mode. To set the value to
the default, use the no form of this command.
rtsp client session history duration number
no rtsp client session history duration number
Syntax Description
number
Defaults
10 minutes
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300 universal
access server.
Examples
Specifies how long, in minutes, to keep the record.
The following example sets the RTSP session history to 500 minutes:
rtsp client session history duration 500
Related Commands
Command
Description
call application voice load
Allows reload of an aplication that was loaded via the MGCP
scripting package.
rtsp client session history
records
Specifies the number of RTSP client session history records kept
during the session.
show call application voice
Displays all TCL or MGCP scripts that are loaded.
show rtsp client session
Displays cumulative information about the RTSP session records.
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rtsp client session history records
rtsp client session history records
To configure the number of records to keep in the RTSP client session history, use the rtsp client session
history records command in global configuration mode. To set the value to the default, use the no form
of this command.
rtsp client session history records number
no rtsp client session history records number
Syntax Description
number
Defaults
50 records
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300 universal access
server.
Examples
Specifies the number of records to retain in a session history. Values
range from 1 to 100000.
The following example sets the RTSP client history to 500 records:
rtsp client session history records 500
Related Commands
Command
Description
call application voice load
Allows reload of an aplication that was loaded via the MGCP
scripting package.
rtsp client session history
duration
Specifies the how long the RTSP is kept during the session.
show call application voice
Displays all TCL or MGCP scripts that are loaded.
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rule
rule
To apply a translation rule to a calling party number or a called party number for both incoming and
outgoing calls, use the rule command in translation-rule configuration mode. To remove the translation
rule, use the no form of this command.
rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]
no rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]
Syntax Description
Note
name-tag
The tag number by which the rule set will be referenced. This is an
arbitrarily chosen number. Range is from 1 through 2147483647.
input-matched-pattern
The input string of digits for which pattern matching is performed.
substituted-pattern
The replacement digit string that results after pattern matching is
performed. Regular expressions are used to carry out this process.
match-type
(Optional) The choices for this field are international, national,
subscriber, abbreviated, unknown, and any, as defined by the
International Telecommunication Union Telecommunication
Standardization Sector (ITU-T) Q.931 specification. If you enter the
match-type value, then you must also enter the substituted-type value.
substituted-type
(Optional) The choices for this field are international, national,
subscriber, abbreviated and unknown, as defined by the ITU Q.931
specification.
In the syntax description above, the square brackets indicate optional values. When using this
command, do not include these square brackets as part of the syntax. They are not valid parameters
in the rule command. The square brackets can only be used in actual syntax for such commands as
the destination-pattern and incoming called-number commands, where the syntax specifically
allows this delimiter.
Defaults
No default behavior or values.
Command Modes
Translation-rule configuration
Command History
Release
Modification
12.0(7)XR1
This command was introduced for Voice over IP on the Cisco AS5300
universal access server.
12.0(7)XKs
This command was first supported for Voice over IP on the following
platforms: Cisco 2600 and 3600 series routers and Cisco MC3810
multiservice concentrators.
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rule
Release
Modification
12.1(1)T
This command was first supported on the T train for Voice over IP on the
following platforms: Cisco 1750 routers, Cisco 2600 and 3600 series
routers, Cisco AS5300 universal access servers, Cisco 7200 series
routers, and Cisco 7500 series routers.
12.1(2)T
This command was first supported for Voice over IP on the
Cisco MC3810 multiservice concentrator.
Usage Guidelines
When configuring your dial peers, you are provided with an option called the translation rule. This
option applies a translation rule to a calling party number (Automatic Number Identification [ANI]) or
a called party number (Dial Number Information Service [DNIS]) for both incoming and outgoing calls
within Cisco H.323 voice-enabled gateways. Also, the rule allows translation of the type of number.
Examples
The following example applies a translation rule. If a called number starts with 5552205 or 52205, then
translation rule 21 will use the rule command to forward the number to 14085552205 instead.
translation-rule 21
rule 1 555.% 1408555 subscriber international
rule 2 7.% 1408555 abbreviated international
In the next example, if a called number is either 14085552205 or 014085552205, then after the execution
of the translation rule 345, the forwarding digits will be 52205. If the match type is configured and the
type is not “unknown,” then the dial peer matching will be required to match input string numbering
type.
translation-rule 345
rule 1 .%555.% 7 any abbreviated
Related Commands
Command
Description
numbering-type
Specifies number type for the VoIP or POTS dial peer.
test translation-rule
Tests the execution of the translation rules on a specific name tag.
translate
Applies a translation rule to a calling party number or a called party number
for incoming calls
translate-outgoing
Applies a translation rule to a calling party number or a called party number
for outgoing calls
translation-rule
Creates a translation name and enters translation-rule configuration mode.
voip-incoming
translation-rule
Captures calls that originate from H.323-compatible clients.
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security
security
To enable authentication and authorization on a gatekeeper, use the security command in gatekeeper
configuration mode. To disable security, use the no form of this command.
security {any | h323-id | e164} {password default password | password separator character}
no security {any | h323-id | e164} {password default password | password separator character}
Syntax Description
any
Uses the first alias of an incoming registration, admission, and status
(RAS) protocol registration, regardless of its type, as the means of
identifying the user to RADIUS/TACACS+.
h323-id
Uses the first H.323 ID type alias as the means of identifying the user
to RADIUS/TACACS+.
e164
Uses the first E.164 address type alias as the means of identifying the
user to RADIUS/TACACS+.
password default password
Specifies the default password that the gatekeeper associates with
endpoints when authenticating them with an authentication server. The
password must be identical to the password on the authentication
server.
password separator
character
Specifies the character that endpoints use to separate the H.323-ID
from the piggybacked password in the registration. Specifying this
character allows each endpoint to supply a user-specific password.
The separator character and password will be stripped from the string
before it is treated as an H.323-ID alias to be registered.
Note that passwords may only be piggybacked in the H.323-ID, not the
E.164 address, because the E.164 address allows a limited set of
mostly numeric characters. If the endpoint does not wish to register an
H.323-ID, it can still supply an H.323-ID consisting of just the
separator character and password. This H.323-ID consisting of just the
separator character and password will be understood to be a password
mechanism and no H.323-ID will be registered.
Defaults
No default
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2600 series and
Cisco 3600 series routers.
Usage Guidelines
Use the security command to enable identification of registered aliases by RADIUS/TACACS+. If the
alias does not exist in RADIUS/TACACS+, the endpoint will not be allowed to register.
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security
A RADIUS/TACACS+ server and encryption key must have been configured in Cisco IOS software for
security to work.
Only the first alias of the proper type will be identified. If no alias of the proper type is found, the
registration will be rejected.
This command does not allow you to define the password mechanism unless the security type (h323-id
or e164 or any) has been defined. Although the no security password command undefines the password
mechanism, it leaves the security type unchanged, so security is still enabled. However, the no security
command disables security entirely, including removing any existing password definitions.
Examples
The following example enables identification of registrations using the first H.323 ID found in any
registration:
security h323id
The following example enables security, authenticating all users by using their H.323-IDs and a
password of qwerty2x:
security h323-id
security password qwerty2x
The next example enables security, authenticating all users by using their H.323-IDs and the password
entered by the user in the H.323-ID alias he or she registers:
security h323-id
security password separator !
Now if a user registers with an H.323-ID of joe!024aqx, the gatekeeper authenticates user joe with
password 024aqx, and if that is successful, registers the user with the H.323-ID of joe. If the exclamation
point is not found, the user is authenticated with the default password, or a null password if no default
has been configured.
The following example enables security, authenticating all users by using their E.164 IDs and the
password entered by the user in the H.323-ID alias he or she registers:
security e164
security password separator !
Now if a user registers with an E.164 address of 5551212 and an H.323-ID of !hs8473q6, the gatekeeper
authenticates user 5551212 and password hs8473q6. Because the H.323-ID string supplied by the user
begins with the separator character, no H.323-ID is registered, and the user is known only by the E.164
address.
Related Commands
Command
Description
accounting (gatekeeper) Enables the accounting security feature on the gatekeeper.
radius-server host
Specifies a RADIUS server host.
radius-server key
Sets the authentication and encryption key for all RADIUS
communications between the router and the RADIUS daemon.
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sequence-numbers
sequence-numbers
To enable the generation of sequence numbers in each frame generated by the digital signal processor
(DSP) for Voice over Frame Relay applications, use the sequence-numbers command in dial-peer
configuration mode. To disable the generation of sequence numbers, use the no form of this command.
sequence-numbers
no sequence-numbers
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.0(3)XG
This command was introduced on the Cisco 2600 and 3600 series
routers and the Cisco MC3810 multiservice concentrator.
12.0(4)T
This command was integrated into the Cisco IOS Release 12.0(4)T.
Usage Guidelines
Sequence numbers on voice packets allow the digital signal processor (DSP) at the playout side to detect
lost packets, duplicate packets, or out-of-sequence packets. This helps the DSP to mask out occasional
drop-outs in voice transmission at the cost of one extra byte per packet. The benefit of using sequence
numbers versus the cost in bandwidth of adding an extra byte to each voice packet on the Frame Relay
network must be weighed to determine whether to disable this function for your application.
Another factor to consider is that this command does not affect codecs that require a sequence number,
such as G.726. If you are using a codec that requires a sequence number, the DSP will generate one
regardless of the configuration of this command.
Examples
The following example shows how to disable the generation of sequence numbers for VoFR frames on a
Cisco 2600 series or 3600 series router or on a Cisco MC3810 multiservice concentrator for VoFR dial
peer 200, starting from global configuration mode:
dial-peer voice 200 vofr
no sequence-numbers
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sequence-numbers
Related Commands
Command
Description
called-number
(dial-peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call
leg when using a static FRF.11 trunk connection.
codec (dial-peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay dial
peer.
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence
setting.
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN
directory number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over Enables the generation of FRF.11 Annex A frames for a dial peer.
Frame Relay)
session protocol
(Voice over Frame
Relay)
Establishes a session protocol for calls between the local and remote routers
via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination
gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
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server (RLM)
server (RLM)
To identify an RLM server, use the server RLM configuration command. To remove the identification,
use the no form of this command
server name-tag
no server name-tag
Syntax Description
name-tag
Defaults
Disabled
Command Modes
RLM configuration
Command History
Release
Modification
11.3(7)
This command was introduced.
Name to identify the server configuration so that multiple entries of server
configuration can be entered.
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example identifies the RLM server and defines the associated IP addresses:
rlm group 1
server r1-server
link address 10.1.4.1 source Loopback1 weight 4
link address 10.1.4.2 source Loopback2 weight 3
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server (RLM)
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
interface
Defines the IP addresses of the server, configures an interface type, and
enters interface configuration mode.
link (RLM)
Specifies the link preference.
protocol rlm port
Reconfigures the port number for the basic RLM connection for the
whole rlm-group.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the
link is declared down.
show rlm group statistics Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
show rlm group timer
Displays the current RLM group timer values.
shutdown (RLM)
Shuts down all of the links under the RLM group.
timer
Overwrites the default setting of timeout values.
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server registration-port
server registration-port
To configure the listener port for the server to establish a connection with the gatekeeper, use the server
registration-port command in gatekeeper configuration mode. To force the gatekeeper to close the
listening socket so that no more new registration takes place, use the no form of this command.
server registration-port port number
no server registration-port port number
Syntax Description
port number
Specifies a single range of values from 1 through 65535 for the port
number on which the gatekeeper listens for external server connections.
Defaults
The registration port of the gatekeeper is not configured, so no external server can register with this
gatekeeper.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2500 series, Cisco 2600
series, Cisco 3600 series, and Cisco 7200 series routers and on the
Cisco MC3810 multiservice concentrator.
Usage Guidelines
Note
Examples
Use this command to configure a server registration port to poll for servers that want to establish
connections with the gatekeeper on this router.
The no form of this command forces the gatekeeper on this router to close the listen socket, so it
cannot accept more registrations. However, existing connections between the gatekeeper and servers
are left open.
The following example shows how a listener port for a server is established for connection with a
gatekeeper:
server registration-port 20000
Related Commands
Command
Description
server trigger
Configure static server triggers for specific RAS messages to be forwarded
to a specified server.
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server trigger
server trigger
To configure a static server trigger for external applications, use the server trigger command in
gatekeeper configuration mode. To remove a single statically configured trigger entry, use the no form
of this command. To remove every static trigger you configured if you want to delete them all, use the
all keyword.
server trigger {arq | lcf | lrj | lrq | rrq | urq} gkid priority server-id server-ipaddress server-port
no server trigger {arq | lcf | lrj | lrq | rrq | urq} gkid priority
no server trigger all
Syntax Description
all
Specified to delete all command-line interface configured triggers.
arq, lcf, lrj, lrq, rrq, urq
Registration, admission, and status (RAS) protocol message types. Use
these message types to specify a submode in the gatekeeper
configuration mode in which you configure a trigger for the gatekeeper
to act upon. Specify only one message type per server trigger command.
There is a different trigger submode for each message type. Each trigger
submode has its own set of applicable commands.
gkid
The local gatekeeper identifier.
priority
The priority for each trigger. The range is from 1 through 20, with 1
being the highest priority.
server-id
The ID number of the external application.
server-ipaddress
The IP address of the server.
server-port
The port on which the Cisco IOS gatekeeper listens for messages from
the external server connection.
Defaults
No server triggers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2500 series, Cisco 2600
series, Cisco 3600 series, and Cisco 7200 series routers and on the
Cisco MC3810 multiservice concentrator.
Usage Guidelines
Use this command to configure a static server trigger. There are six different server triggers—one for
each of the RAS messages. To configure a trigger, go to its submode where a set of subcommands are
used to trigger a condition. See the following examples.
In ARQ submode, enter the following syntax:
server trigger arq gkid priority server-id server-ipaddress server-port
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server trigger
In LCF submode, enter the following syntax:
server trigger lcf gkid priority server-id server-ipaddress server-port
In LRJ submode, enter the following syntax:
server trigger lrj gkid priority server-id server-ipaddress server-port
In LRQ submode, enter the following syntax:
server trigger lrq gkid priority server-id server-ipaddress server-port
In RRQ submode, enter the following syntax:
server trigger rrq gkid priority server-id server-ipaddress server-port
In URQ submode, enter the following syntax:
server trigger urq gkid priority server-id server-ipaddress server-port
The following options are available in all submodes:
info-only
shutdown
Information only—no need to wait for acknowledgment.
Enter this subcommand to temporarily disable a trigger. The gatekeeper does
not consult triggers in a shutdown state when determining what message to
forward.
The destination-info argument is under the ARQ, LRQ, LCF, and LRJ submode and has the following
options:
destination-info
e164
email-id
h323-id
word
Configure destination-info to trigger one of the following conditions:
Configure an E.164 pattern.
Configure an email ID.
Configure an H.323 ID.
When configuring the e164 address option, the email-id option, or the h323-id
option above, the E.164 address can end in a trailing ‘., ‘s, or ‘*’.
The redirect-reason argument is under the ARQ and LRQ submodes and has the following options:
redirect-reason
0
1
2
4
9
10
Configure a redirect-reason to trigger on (range of 0 through 65535) with the
following reserved values:
Unknown reason.
Call forwarding busy or called DTE busy.
Call forwarded no reply.
Call deflection.
Called DTE out of order.
Call forwarding by the call DTE.
Call forwarding unconditionally.
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server trigger
The remote-ext-address argument is under the LCF trigger submode and has the following options:
remote-ext-address
e164
word
Configure remote extension addresses, with the following options:
Configure an E.164 pattern.
When configuring the e164 address option, the email-id option, or the
h323-id option above, the E.164 address can end in a trailing ‘., ‘s, or ‘*’.
The endpoint-type argument is under the RRQ and URQ trigger submodes and has the following options:
endpoint-type
Configure the type of endpoint to trigger, with the following options:
gatekeeper
The endpoint is an H.323 gatekeeper.
h320-gateway
The endpoint is an H.320 gateway.
mcu
The endpoint is a multipoint control unit (MCU).
other-gateway
The endpoint is another type of gateway not specified on this list.
proxy
The endpoint is a H.323 proxy.
terminal
The endpoint is an H.323 proxy.
voice-gateway
The endpoint is a voice gateway.
The supported-prefix keyword is under the RRQ and URQ submodes and has the following options:
supported-prefix
Configure the gateway technology prefix to trigger on.
word
Enter a word within the set of “0123456789#*” when configuring the
E.164 pattern for a gateway technology prefix.
Entering the no form of the server trigger command removes the trigger definition from the Cisco IOS
gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a server trigger on gatekeeper sj.xyz.com to notify external server
“Server-123” of any call to an E.164 number that starts with 1800 followed by any 7 digits (1800551212,
for example):
Gatekeeper
server trigger arq sj.xyz.com 1 Server-123 1.14.93.130 1751
destination-info e164 1800.......
exit
Related Commands
Command
Description
server registration port
Configure a gatekeeper listening port to listen for external server
connections.
show gatekeeper servers
Show a list of currently registered and statically configured triggers on
this gatekeeper router.
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session
session
To associate a transport session with a specified session-group, use the session group command in
backhaul session manager configuration mode. It is assumed that the server is located on a remote
machine. To delete the session, use the no form of this command.
session group group-name remote_ip remote_port local_ip local_port priority
no session group group-name remote_ip remote_port local_ip local_port priority
Syntax Description
group
Specifies the session-group name.
group-name
Session-group name.
remote_ip
Remote IP address.
remote_port
Remote port number. Range is 1024 through 9999.
local_ip
Local IP address.
local_port
Local port number. Range is 1024 through 9999.
priority
Priority of the session-group. Range is 0 through 9999 and 0
is the highest priority.
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850
platform.
Examples
To associate a transport session with the session-group Group5 and specify the parameters described
above, see the following example:
Router(config-bsm)# session group group5 161.44.2.72 5555 172.18.72.198 5555 1
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session protocol
session protocol
To specify a session protocol for calls between the local and remote routers using the packet network,
use the session protocol command in dial-peer configuration mode. To reset the default value for this
command, use the no form of this command.
session protocol {cisco | sipv2 | aal2-trunk | smtp}
no session protocol
Syntax Description
cisco
Configure the dial peer to use proprietary Cisco VoIP session protocol.
sipv2
SIP users should use this option. This option configures the VoIP dial peer
to use IETF SIP.
aal2-trunk
AAL2 nonswitched trunk session protocol.
smtp
Specifies Simple Mail Transfer Protocol (SMTP) session protocol.
Defaults
No default behaviors or values.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series router.
12.0(4)XJ
This command was modified for store-and-forward fax on the Cisco AS5300
universal access server.
Usage Guidelines
12.1(1)T
The sipv2 option was added.
12.1(1)XA
Support was added for VoATM dial peers on the Cisco MC3810 multiservice
concentrator with the aal2-trunk keyword.
12.1(2)T
Modifications to this command in Cisco IOS Release 12.1(1)XA were
integrated into Cisco IOS Release 12.1(2)T.
The keyword cisco is applicable only to VoIP on the Cisco 3600 series routers. The keyword aal2-trunk
is applicable only to VoATM on the Cisco MC3810 multiservice concentrator.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following is an example of configuring a VoIP dial peer for H.323 or SIP as the session protocol for
VoIP call signaling:
dial-peer voice 102 voip
session protocol sipv2
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session protocol
The following example selects AAL2 trunking as the session protocol on a Cisco MC3810 multiservice
concentrator:
dial-peer voice 10 voatm
session protocol aal2-trunk
The following example selects Cisco Session Protocol as the session protocol on a Cisco 3600 series
router:
dial-peer voice 20 voip
session protocol cisco
The following example selects SMTP as the session protocol:
dial-peer voice 10 mmoip
session protocol smtp
Related Commands
Command
Description
dial-peer voice
Enters dial-peer configuration mode and specifies the method of
voice-related encapsulation.
session target (VoIP)
Configures a network-specific address for a dial peer.
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session protocol (Voice over Frame Relay)
session protocol (Voice over Frame Relay)
To establish a Voice over Frame Relay protocol for calls between the local and remote routers via the
packet network, use the session protocol command in dial-peer configuration mode. To reset the default
value, use the no form of this command.
session protocol {cisco-switched | frf11-trunk}
no session protocol
Syntax Description
cisco-switched
Specifies proprietary Cisco VoFR session protocol. (This is the only valid
session protocol for the Cisco 7200 series.)
frf11-trunk
Specifies FRF.11 session protocol.
Defaults
cisco-switched
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced for VoIP.
12.0(3)XG
This command was modified to support VoFR on the Cisco 2600, 3600, and
7200 series routers and the Cisco MC3810 multiservice concentrator.
12.0(4)T
The cisco-switched and frf11-trunk keywords were added for VoFR dial
peers.
Usage Guidelines
Note
Examples
For Cisco-to-Cisco dial peer connections, Cisco recommends that you use the default session protocol
because of the advantages it offers over a pure FRF.11 implementation. When connecting to
FRF.11-compliant equipment from other vendors, use the FRF.11session protocol.
When using the FRF.11 session protocol on Cisco 2600 series and 3600 series routers, you must also
use the called-number command.
The following example shows how to configure the FRF.11 session protocol on a Cisco 2600 series or
3600 series router for VoFR dial peer 200:
dial-peer voice 200 vofr
session protocol frf11-trunk
called-number 5552150
The following example shows how to configure the FRF.11 session protocol on a Cisco MC3810
multiservice concentrator for VoFR dial peer 200:
dial-peer voice 200 vofr
session protocol frf11-trunk
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session protocol (Voice over Frame Relay)
Related Commands
Command
Description
called-number (dial-peer) Enables an incoming VoFR call leg to get bridged to the correct POTS
call leg when using a static FRF.11 trunk connection.
codec (dial-peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay
dial peer.
cptone
Specifies a regional analog voice interface-related tone, ring, and
cadence setting.
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN
directory number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over
Frame Relay)
Enables the generation of FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred order of a dial peer within a rotary hunt group.
session target
Specifies a network-specific address for a specified dial peer or
destination gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
Cisco IOS Voice, Video, Fax Command Reference
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session protocol aal2
session protocol aal2
To enter the voice-service-session configuration mode and specify AAL2 trunking on a Cisco MC3810
multiservice concentrator, use the session protocol aal2 command in voice-service configuration mode.
session protocol aal2
Syntax Description
This command has no keywords or arguments.
Defaults
There is no default setting for this command.
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(1)XA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.1(2)T
This command was integrated into the 12.1(2)T release.
Usage Guidelines
This command applies to VoATM on the MC3810 multiservice concentrator.
In the voice-service-session configuration mode for AAL2, you can configure only AAL2 features, such
as call admission control and subcell multiplexing.
Examples
The following example shows how to access the voice-service-session configuration mode, beginning in
global configuration mode:
voice service voatm
session protocol aal2
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session protocol multicast
session protocol multicast
To set the session protocol as multicast, use the session protocol multicast command dial-peer
configuration mode. To negate this command and return to the Cisco default session protocol, use the no
version of this command.
session protocol multicast
no session protocol multicast
Syntax Description
There are no keywords or arguments.
Defaults
When this command is not implemented, the default session protocol is cisco.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(2)XH
This command was introduced on Cisco 2600 and Cisco 3600 series
routers for the Cisco Hoot and Holler over IP application.
12.1(3)T
This command was integrated into the Cisco IOS Release 12.1(3)T.
Usage Guidelines
Use the session protocol multicast dial-peer configuration command for voice conferencing in a
Hoot and Holler networking implementation. This command allows more than two ports to join the same
session simultaneously. It is supported on Cisco 2600 and Cisco 3600 series routers.
Examples
The following example shows the use of the session protocol multicast dial-peer configuration
command in context with its accompanying commands:
dial-peer voice 111 voip
destination-pattern 111
session protocol multicast
session target ipv4:237.111.0.111:22222
ip precedence 5
codec g711ulaw
Related Commands
Command
Description
session target ipv4
Assigns the session target for voice-multicasting dial peers.
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session target (VoATM)
session target (VoATM)
To specify a network-specific address for a specified VoATM dial peer, use the session target command
in dial-peer configuration mode. To restore default values for this parameter, use the no form of this
command.
Cisco 3600 Series Routers Voice over ATM Dial Peers
session target interface pvc {name | vpi/vci | vci}
no session target
Cisco MC3810 Multiservice Concentrator Voice over ATM Dial Peers
session target {serial | atm} interface pvc {word | vpi/vci | vci} cid
no session target
Syntax Description
serial
Specifies the serial interface for the dial-peer address.
atm
Specifies the ATM interface. The only valid number is 0.
interface
Interface type and interface number on the router.
pvc
The specific ATM permanent virtual circuit (PVC) for this dial peer.
word
(Optional) A name that identifies the PVC. The argument can identify
the PVC if a word identifier was assigned when the PVC was created.
name
The PVC name.
vpi/vci
ATM network virtual path identifier (VPI) and virtual channel
identifier (VCI) of this PVC.
On the Cisco 3600, if you have the Multiport T1/E1 ATM network
module with IMA installed, the valid range for vpi is from 0 to 5, and
the valid range for vci is from 1 to 255.
If you have the OC3 ATM Network Module installed, the valid range
for vpi is from 0 to 15, and the valid range for vci is from 1 to 1023.
vci
ATM network virtual channel identifier (VCI) of this PVC.
cid
ATM network channel identifier (CID) of this PVC. The valid range
is from 8 to 255.
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
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session target (VoATM)
Command History
Usage Guidelines
Release
Modification
11.3(1)T
This command was introduced.
11.3(1)MA
Support was added for VoATM, VoHDLC, and POTS dial peers on the
Cisco MC3810 multiservice concentrator.
12.0(7)XK
Support was added for VoATM dial peers on the Cisco 3600 series
routers. Support for VoHDLC on the Cisco MC3810 multiservice
concentrator was removed.
12.1(2)T
Support was added for VoATM on Cisco MC3810 multiservice
concentrators.
Use the session target command to specify a network-specific address or domain name for a dial peer.
Whether you select a network-specific address or a domain name depends on the session protocol you
select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The
loopback point will depend on the call origin and the loopback type selected.
This command applies to on-ramp store-and-forward fax functions.
You must enter the session protocol aal2-trunk dial-peer configuration command before you can
specify a cid for a dial peer for VoATM on the Cisco MC3810 multiservice concentrator.
Note
Examples
This command does not apply to plain old telephone service (POTS) dial peers.
The following example configures a session target for Voice over ATM on a Cisco MC3810 multiservice
concentrator. The session target is sent to ATM interface 0, and for a PVC with a VCI of 20.
dial-peer voice 12 voatm
destination-pattern 13102221111
session target atm0 pvc 20
The following example delivers fax-mail to multiple recipients:
dial-peer voice 10 mmoip
session target [email protected]mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its
/etc/aliases file:
marketing-information:
[email protected],
[email protected]
The following example displays configuring a session target for Voice over ATM on the Cisco 3600
series. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.
dial-peer voice 12 voatm
destination-pattern 13102221111
session target atm1/0 pvc 1/100
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session target (VoATM)
Related Commands
Command
Description
called-number
Enables an incoming VoFR call leg to be bridged to the correct POTS
call leg.
codec (dial-peer)
Specifies the voice coder rate of speech for a dial peer.
cptone
Specifies a regional tone, ring, and cadence setting for an analog voice
port.
destination-pattern
Specifies either the prefix or the full E.164 telephone number
(depending on your dial plan) to be used for a dial peer.
dtmf-relay
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred selection order of a dial peer within a hunt
group.
session protocol
Establishes a VoFR protocol for calls between the local and the remote
routers via the packet network.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
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session target (VoFR)
session target (VoFR)
To specify a network-specific address for a specified VoFR dial peer, use the session target command
in dial-peer configuration mode. To restore default values for this parameter, use the no form of this
command.
Cisco 2600 and 3600 Series Routers Voice over Frame Relay Dial Peers
session target interface dlci [cid]
no session target
Cisco MC3810 Multiservice Concentrator Voice over Frame Relay Dial Peers
session target interface dlci [cid]
no session target
Cisco 7200 Series Routers Voice over Frame Relay Dial Peers
session target interface dlci
no session target
Syntax Description
interface
Specifies the serial interface and interface number (slot number and
port number) associated with this dial peer. For the range of valid
interface numbers for the selected interface type, enter a ? character
after the interface type.
dlci
Specifies the data link connection identifier for this dial peer. The
valid range is from 16 to 1007.
cid
(Optional) Specifies the DLCI subchannel to be used for data on
FRF.11 calls. A CID must be specified only when the session protocol
is frf11-trunk. When the session protocol is cisco-switched, the CID
is dynamically allocated. The valid range is from 4 to 255.
Note
By default, CID 4 is used for data; CID 5 is used for
call-control. We recommend that you select CID values
between 6 and 63 for voice traffic. If the CID is greater than
63, the FRF.11 header will contain an extra byte of data.
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
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session target (VoFR)
Command History
Usage Guidelines
Release
Modification
11.3(1)T
This command was introduced.
11.3(1)MA
Support was added for VoFR, VoHDLC, and POTS dial peers on the
Cisco MC3810 multiservice concentrator.
12.0(3)XG
Support was added for VoFR dial peers on the Cisco 2600 series and
3600 series routers. The cid option was added.
12.0(4)T
Support was added for VoFR and POTS dial peers on the Cisco
7200 series routers and the support added in Cisco IOS Release
12.0(3)XG was integrated into Cisco IOS Release 12.0(4)T.
Use the session target command to specify a network-specific address or domain name for a dial peer.
Whether you select a network-specific address or a domain name depends on the session protocol you
select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The
loopback point will depend on the call origin and the loopback type selected.
For VoFR dial peers, the cid option is not allowed when the cisco-switched option for the
session protocol command is used.
Examples
The following example configures a session target for Voice over Frame Relay on a Cisco MC3810
multiservice concentrator with a session target on serial port1 and a DLCI of 200:
dial-peer voice 11 vofr
destination-pattern 13102221111
session target serial1 200
The following example shows how to configure serial interface 1/0, DLCI 100 as the session target for
VoFR dial peer 200 (an FRF.11 dial peer) on a Cisco 2600 series or 3600 series router, starting from
global configuration mode and using the FRF.11 session protocol:
dial-peer voice 200 vofr
destination-pattern 13102221111
called-number 5552150
session protocol frf11-trunk
session target serial 1/0 100 20
The following example delivers fax-mail to multiple recipients:
dial-peer voice 10 mmoip
session target [email protected]
Assuming that mailer.example.com is running sendmail, you can put the following information into its
/etc/aliases file:
marketing-information:
[email protected],
[email protected]
Cisco IOS Voice, Video, Fax Command Reference
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session target (VoFR)
Related Commands
Command
Description
called-number
Enables an incoming VoFR call leg to be bridged to the correct POTS
call leg.
codec (dial-peer)
Specifies the voice coder rate of speech for a dial peer.
cptone
Specifies a regional tone, ring, and cadence setting for an analog voice
port.
destination-pattern
Specifies either the prefix or the full E.164 telephone number
(depending on your dial plan) to be used for a dial peer.
dtmf-relay
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred selection order of a dial peer within a hunt
group.
session protocol
Establishes a VoFR protocol for calls between the local and the remote
routers via the packet network.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
Cisco IOS Voice, Video, Fax Command Reference
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session target (VoIP)
session target (VoIP)
To specify a network-specific address for a specified VoIP dial peer, use the session target command in
dial-peer configuration mode. To restore default values for this parameter, use the no form of this
command.
Cisco 2600 and Cisco 3600 Series Routers and Cisco MC8310 Multiservice Concentrator Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed | ras | settlement}
no session target
Cisco AS5300 Universal Access Server Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed | mailto: | {name | $d$}@domain-name |
ipv4:destination-address | dns:[$s$. | $d$. | $u$. | $e$.] host-name}
no session target
Cisco AS5800 Universal Access Server Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed}
no session target
Syntax Description
ipv4:destination-address IP address of the dial peer.
dns:[$s$...] host-name
Indicates that the domain name server will be used to resolve the name of
the IP address. Valid entries for this parameter are characters representing
the name of the host device.
(Optional) Use one of the following three wildcards with this keyword
when defining the session target for Voice over IP (VoIP) peers:
$s$.—Indicates that the source destination pattern will be used as part of
the domain name.
$d$.—Indicates that the destination number will be used as part of the
domain name.
$e$.—Indicates that the digits in the called number will be reversed,
periods will be added between the digits of the called number, and this
string will be used as part of the domain name.
$u$.—Indicates that the unmatched portion of the destination pattern (such
as a defined extension number) will be used as part of the domain name.
loopback:rtp
Indicates that all voice data will be looped back to the source. This is
applicable for VoIP peers.
loopback:compressed
Indicates that all voice data will be looped back in compressed mode to the
source. This is applicable for POTS peers.
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session target (VoIP)
loopback:uncompresse
d
Indicates that all voice data will be looped-back in uncompressed mode to
the source. This is applicable for POTS peers.
ras
Indicates that the registration, admission, and status (RAS) signaling
function protocol is being used, meaning that a gatekeeper will be
consulted to translate the E.164 address into an IP address.
settlement
provider-number
Indicates that the settlement server is the target to resolve the terminating
gateway address. Enter the provider IP address for provider number.
Defaults
The default state for this command is enabled, with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 2600 series and
Cisco 3600 series routers.
12.0(3)T
Support was added for VoIP and POTS dial peers on the Cisco AS5300
universal access server. The parameter was added for RAS.
12.0(4)XJ
Support was added for store-and-forward fax on the Cisco AS5300
universal access server platform.
12.1(1)T
Support was added for session target type of settlement.
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer.
Whether you select a network-specific address or a domain name depends on the session protocol you
select.
The session target loopback command is used for testing the voice transmission path of a call. The
loopback point will depend on the call origin and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional
wildcards can reduce the number of VoIP dial peer session targets you must configure if you have groups
of numbers associated with a particular router.
Use the session target ras command to specify that the RAS protocol is being used to determine the IP
address of the session target.
In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of
RAS with the settle-call command. When configuring the VoIP dial peers for a settlement server, if
session target type is settlement, the provider-number parameter in the session target and settle-call
commands should be identical.
When the VoIP dial peers are configured for a settlement server, if the session target type is settlement,
the provider-number parameter in the session target and settle-call commands should be identical.
Examples
The following example configures a session target using DNS for a host, “voice_router,” in the domain
cisco.com:
dial-peer voice 10 voip
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session target (VoIP)
session target dns:voice_router.cisco.com
The following example configures a session target using DNS, with the optional $u$. wildcard. In this
example, the destination pattern has been configured to allow for any four-digit extension, beginning
with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched
portion of the dialed number—in this case, the four-digit extension—to identify the dial peer. As in the
preceding example, the domain is “cisco.com.”
dial-peer voice 10 voip
destination-pattern 1310222....
session target dns:$u$.cisco.com
The following example configures a session target using DNS, with the optional $d$. wildcard. In this
example, the destination pattern has been configured for 13102221111. The optional wildcard $d$.
indicates that the router will use the destination pattern to identify the dial peer in the “cisco.com”
domain.
dial-peer voice 10 voip
destination-pattern 13102221111
session target dns:$d$.cisco.com
The following example configures a session target using DNS, with the optional $e$. wildcard. In this
example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that
the router will reverse the digits in the destination pattern, add periods between the digits, and then use
this reverse-exploded destination pattern to identify the dial peer in the “cisco.com” domain.
dial-peer voice 10 voip
destination-pattern 12345
session target dns:$e$.cisco.com
The following example configures a session target using RAS:
dial-peer voice 11 voip
destination-pattern 13102221111
session target ras
The following example configures a session target using settlement:
session target settlement:0
Related Commands
Command
Description
called-number
Enables an incoming VoFR call leg to be bridged to the correct POTS call
leg.
codec (dial-peer)
Specifies the voice coder rate of speech for a dial peer.
cptone
Specifies a regional tone, ring, and cadence setting for an analog voice
port.
dtmf-relay
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred selection order of a dial peer within a hunt group.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
destination-pattern
Specifies either the prefix or the full E.164 telephone number (depending
on your dial plan) to be used for a dial peer.
session protocol
Establishes a session protocol for calls between the local and remote
routers through the packet network in Voice over IP.
settle-call
Specifies that settlement is to be used for this dial peer, regardless of
session target type.
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session transport
session transport
To configure the VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport
layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in
dial-peer configuration mode. To reset the value to the default, use the no form of this command.
session transport {udp | tcp }
Syntax Description
Defaults
udp
Configure the SIP dial peer to use the UDP transport layer protocol. This
is the default.
tcp
Configure the SIP dial peer to use the TCP transport layer protocol.
The SIP dial peer uses UDP.
Note
The transport protocol specified with the transport command and the one specified with the session
transport command must be the same.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
Usage Guidelines
Use show sip-ua status to ensure that the transport protocol that you set using the session transport
command matches the protocol set using the transport command.
Examples
The following example shows a VoIP dial peer configured to use UDP as the underlying transport
layer protocol for SIP messages:
dial-peer voice 102 voip
session transport udp
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set
set
To create a fault-tolerant or non-fault-tolerant session-set with the client or server option, use the set
command in backhaul session manager configuration mode. To delete the set, use the no form of this
command.
set set-name { client | server } { ft | nft }
no set set-name { client | server } { ft | nft }
Syntax Description
set-name
Session-set name.
client
Client option. The session-set should only be configured as
client for backhaul.
server
Server option.
ft
Fault-tolerant. Fault-tolerance is the level of ability within a
system to operate properly even if a group in the set fails.
nft
Non-fault-tolerant. Only one group is allowed in a
non-fault-tolerant set.
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
Usage Guidelines
There can be multiple groups associated with a session-set.
The session-set should only be configured for the client for backhaul (not the server).
A set cannot be deleted unless the groups associated with the set are deleted first.
Examples
To specify the client set named Set1 to fault-tolerant, see the following example:
Router(config-bsm)# set set1 client ft
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settle-call
settle-call
To force a call to be authorized with a settlement server that uses the address resolution method specified
in the session target type command, use the settle-call command in dial-peer configuration mode. To
make sure that no authorization will be performed by a settlement server, use the no form of this
command.
settle-call provider-number
no settle-call provider-number
Syntax Description
provider-number
Digit defining the ID of a particular settlement server. The only valid entry is 0.
Note
If session target type is settlement, the provider-number argument
in the session target and settle-call commands should be identical.
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
Usage Guidelines
Using the session target command, a dial peer can determine the address of the terminating gateway
through the ipv4, dns, ras, and settlement keywords.
If the session target is not settlement, and the settle-call provider-number argument is set, the gateway
resolves the terminating gateway’s address using the specified method and then requests the settlement
server to authorize that address and create a settlement token for that particular address. If the server
cannot authorize the terminating gateway address suggested by the gateway, the call fails.
Do not combine the session target types ras and settle-call. Combination of session target types is not
supported in Cisco IOS Release 12.1(1)T.
Examples
The following example sets a call to be authorized with a settlement server that uses the address
resolution method specified in the session target:
dial-peer voice 10 voip
destination-pattern 1408.......
session target ipv4:172.22.95.14
settle-call 0
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settle-call
Related Commands
Command
Description
session target
Specifies a network-specific address for a specified dial peer.
Cisco IOS Voice, Video, Fax Command Reference
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settlement
settlement
To enter settlement configuration mode and specify the attributes specific to a settlement provider, use
the settlement command in global configuration mode. To disable the settlement provider, use the no
form of this command.
settlement provider-number
no settlement provider-number
Syntax Description
provider-number
Defaults
0
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Specifies a digit that defines a particular settlement server. The only valid
entry is 0.
Usage Guidelines
The variable provider-number defines a particular settlement provider. For Cisco IOS Release 12.1, only
one clearinghouse per system is allowed, and the only valid value for provider-number is 0.
Examples
This example shows how to enter settlement configuration mode:
settlement 0
Related Commands
Command
Description
connection-timeout
Configures the length of time for which a connection is maintained after a
communication exchange is completed.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for
communication with a settlement provider.
response-timeout
Configures the maximum time to wait for a response from a server.
retry-delay
Sets the time between attempts to connect with the settlement provider.
retry-limit
Sets the connection retry limit.
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settlement
Command
Description
session-timeout
Sets the interval for closing the connection when there is no input or output
traffic.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Brings up the settlement provider.
type
Configures an SAA-RTR operation type.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
settlement roam-pattern
settlement roam-pattern
To configure a pattern that must be matched to determine if a user is roaming, use the settlement
roam-pattern command in global configuration mode. To delete a particular pattern, use the no form of
this command.
settlement provider-number roam-pattern pattern {roaming | no roaming}
no settlement provider-number roam-pattern pattern {roaming | no roaming}
Syntax Description
provider-number
Digit defining the ID of particular settlement server. The only valid entry is 0.
pattern
Specifies a user account pattern.
roaming | no roaming Determines whether a user is roaming.
Defaults
No default pattern
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco AS5300 universal access server.
Usage Guidelines
Multiple “roam patterns” could be entered on one gateway.
Examples
The following example will configure a pattern that determines if a user is roaming:
settlement
settlement
settlement
settlement
Related Commands
0 roam-pattern 1222 roam
0 roam-pattern 1333 noroam
roam-pattern 1444 roam
roam-pattern 1555 noroam
Command
Description
roaming (settlement)
Enables the roaming capability for a settlement provider.
settlement
Enters settlement configuration mode.
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sgcp
sgcp
To start and allocate resources for the Simple Gateway Control Protocol (SGCP) daemon, use the sgcp
command in global configuration mode. To terminate all calls, release all allocated resources, and kill
the SGCP daemon, use the no form of this command.
sgcp
no sgcp
Syntax Description
This command has no arguments or keywords.
Defaults
The SGCP daemon is not enabled.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the
Cisco AS5300 universal access server only and was not generally
available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers (except
for the Cisco 3620) in a private release that was not generally
available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally
available on the Cisco 3600 series router and the Cisco MC3810
multiservice concentrator
Usage Guidelines
When the SGCP daemon is not active, all SGCP messages are ignored.
When you enter the no sgcp command, the SGCP process is removed.
Note
Examples
After you enter the no sgcp command, you must save the configuration and reboot the router for the
disabling of SGCP to take effect.
The following example shows the SGCP daemon being enabled:
sgcp
The following example shows the SGCP daemon being disabled:
no sgcp
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sgcp
Related Commands
Command
Description
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown Gracefully terminates all SGCP activity.
sgcp max-waiting-delay Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
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sgcp call-agent
sgcp call-agent
To define the IP address of the default Simple Gateway Control Protocol (SGCP) call agent in the router
configuration file, use the sgcp call-agent command in global configuration mode. To remove the IP
address of the default SGCP call agent from the router configuration, use the no form of this command.
sgcp call-agent ipaddress [:udp port]
no sgcp call-agent ipaddress
Syntax Description
ipaddress
Specifies the IP address or hostname of the call agent.
:udp port
(Optional) Specifies the UDP port of the call agent.
Defaults
No IP address is configured.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the
Cisco AS5300 universal access server only and was not generally
available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers (except
for the Cisco 3620) in a private release that was not generally
available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally
available on the Cisco 3600 series router and the Cisco MC3810
multiservice concentrator
Usage Guidelines
Setting this command defines the IP address of the default SGCP call agent to which the router sends an
initial RSIP (Restart In Progress) packet when the router boots up. This is used for initial boot-up only
before the SGCP call agent contacts the router acting as the gateway.
When you enter the no sgcp call-agent command, only the IP address of the default SGCP call agent is
removed.
Examples
The following example shows SGCP being enabled and the IP address of the call agent being specified:
sgcp
sgcp call-agent 209.165.200.225
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sgcp call-agent
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp graceful-shutdown Gracefully terminates all SGCP activity.
sgcp max-waiting-delay Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
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sgcp graceful-shutdown
sgcp graceful-shutdown
To block all new calls and gracefully terminate all existing calls (wait for the caller to end the call), use
the sgcp graceful-shutdown command in global configuration mode. To unblock all calls and allow new
calls to go through, use the no form of this command.
sgcp graceful-shutdown
no sgcp graceful-shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300
universal access server only and was not generally available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers (except for
the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally available
on the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator
Usage Guidelines
Once you issue this command, all requests for new connections (CreateConnection requests) are denied.
All existing calls are maintained until users terminate them, or until you enter the no sgcp command.
When the last active call is terminated, the SGCP daemon is terminated, and all resources allocated to it
are released.
Examples
The following example shows all new calls being blocked and existing calls being terminated:
sgcp graceful-shutdown
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
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sgcp graceful-shutdown
Command
Description
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
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sgcp max-waiting-delay
sgcp max-waiting-delay
To set the Simple Gateway Control Protocol (SGCP) maximum waiting delay to prevent restart
avalanches, use the sgcp max-waiting-delay command in global configuration mode. To restore the
default value, use the no form of this command.
sgcp max-waiting-delay delay
no sgcp max-waiting-delay delay
Syntax Description
delay
Defaults
3,000 milliseconds
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300
universal access server only, and was not generally available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers (except for
the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally available
on the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator
Examples
Sets the maximum waiting delay (MWD) value in milliseconds. The
valid range is from 0 to 600,000. The default is 3000.
The following example shows the maximum wait delay value set to 40 milliseconds:
sgcp max-waiting-delay 40
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
sgcp max-waiting-delay
Command
Description
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp modem passthru
sgcp modem passthru
To enable Simple Gateway Control Protocol (SGCP) modem or fax pass-through, use the sgcp modem
passthru command in global configuration mode. To disable SGCP modem or fax pass-through, use the
no form of this command.
sgcp modem passthru {ca | cisco | nse}
no sgcp modem passthru {ca | cisco | nse}
Syntax Description
ca
Uses the call agent controlled modem upspeed method violation
message.
cisco
Uses a Cisco-proprietary upspeed method based on the protocol.
nse
Uses the NSE-based modem upspeed method.
Defaults
SGCP modem or fax pass-through is disabled by default.
Command Modes
Global configuration.
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 multiservice
concentrator and Cisco 3600 series routers (except the Cisco 3620) in
a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally
available on the Cisco 3600 series router and the Cisco MC3810
multiservice concentrator
Usage Guidelines
You can use this command for fax pass-through because the answer tone can come from either modem
or fax transmissions. The upspeed method is the method used to dynamically change the codec type and
speed to meet network conditions.
If you use the nse option, you must also configure the sgcp tse payload command.
Examples
The following example shows SGCP modem pass-through configured using the call agent upspeed
method:
sgcp modem passthru ca
The following example shows SGCP modem pass-through configured using the proprietary Cisco
upspeed method:
sgcp modem passthru cisco
The following example shows SGCP modem pass-through configured using the NSE-based modem
upspeed:
Cisco IOS Voice, Video, Fax Command Reference
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sgcp modem passthru
sgcp modem passthru nse
sgcp tse payload 110
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown Gracefully terminates all SGCP activity.
sgcp max-waiting-delay Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp quarantine-buffer disable
sgcp quarantine-buffer disable
To disable the Simple Gateway Control Protocol (SGCP) quarantine buffer, use the sgcp
quarantine-buffer disable command in global configuration mode. To reenable the SGCP quarantine
buffer, use the no form of this command.
sgcp quarantine-buffer disable
no sgcp quarantine-buffer disable
Syntax Description
This command has no arguments or keywords.
Defaults
The SGCP quarantine buffer is enabled.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 multiservice
concentrator and the Cisco 3600 series routers (except for the Cisco
3620) in a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally available
on the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator
Usage Guidelines
The SGCP quarantine buffer is the mechanism for buffering the SGCP events between two RQNT
messages.
Examples
The following example shows the SGCP quarantine buffer being disabled:
sgcp quarantine-buffer disable
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp quarantine-buffer disable
Command
Description
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
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sgcp request retries
sgcp request retries
To specify the number of times to retry sending “notify” and “delete” messages to the Simple Gateway
Control Protocol (SGCP) call agent, use the sgcp request retries command in global configuration
mode. To restore the default value, use the no form of this command.
sgcp request retries count
no sgcp request retries
Syntax Description
count
Defaults
The default for the number of times a “notify” and “delete” message is retransmitted to the SGCP call
agent before it is dropped is 3
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the
Cisco AS5300 universal access server only and was not
generally available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers
(except for the Cisco 3620) in a private release that was not
generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally
available on the Cisco 3600 series router and the
Cisco MC3810 multiservice concentrator
Usage Guidelines
Specifies the number of times a “notify” and “delete” message
is retransmitted to the SGCP call agent before it is dropped. The
valid range is from 1 to 100. The default is 3.
The actual retry count may be different from the value you enter for this command. The retry count is
also limited by the call agent. If there is no response from the call agent after 30 seconds, the gateway
will not retry anymore, even though the number set using the sgcp request retries command has not
been reached.
The router will stop sending retries after 30 seconds, regardless of the setting for this command.
Examples
The following example shows the system configured to send the sgcp command 10 times before
dropping the request:
sgcp request retries 10
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sgcp request retries
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart
avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request timeout
Specifies how long the system should wait for a response to a
request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent
can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random
algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp request timeout
sgcp request timeout
To specify how long the system should wait for a response to a request, use the sgcp request timeout
command in global configuration mode. To restore the default value, use the no form of this command.
sgcp request timeout timeout
no sgcp request timeout
Syntax Description
timeout
Defaults
500 milliseconds
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300
universal access server only and was not generally available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers (except for
the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally available
on the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator
Specifies the number of milliseconds to wait for a response to a request.
Valid range is from 1 to 10,000.
Usage Guidelines
This command is used for “notify” and “delete” messages, which are sent to the SGCP call agent.
Examples
The following example shows the system configured to wait 40 milliseconds for a reply to a request:
sgcp request timeout 40
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp request timeout
Command
Description
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp restart
sgcp restart
To trigger the router to send a Restart in Progress (RSIP) message to the Simple Gateway Control
Protocol (SGCP) call agent indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller, use the sgcp restart command in global configuration mode. To
restore the default value, use the no form of this command.
sgcp restart {delay delay | notify}
no sgcp restart {delay delay | notify}
Syntax Description
delay delay
Specifies the restart delay timer value in milliseconds. The valid
range is from 0 to 600, and the default value is 0.
notify
Enables the restart notification upon the SGCP/digital interface
state transition.
Defaults
Zero (0)
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 multiservice
concentrator and Cisco 3600 series routers (except the Cisco 3620)
in a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally
available on the Cisco 3600 series router and the Cisco MC3810
multiservice concentrator
Usage Guidelines
This command is used to send RSIP messages from the router to the SGCP call agent. The RSIP
messages are used to synchronize the router and the call agent. RSIP messages are also sent when the
sgcp command is entered to enable the SGCP daemon.
You must enter the notify option to enable RSIP messages to be sent.
Examples
The following example shows the system configured to wait 40 milliseconds before restarting SGCP:
sgcp restart delay 40
The following example shows the system configured to send an RSIP notification to the SGCP call agent
when the T1 controller state changes:
sgcp restart notify
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sgcp restart
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart
avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a
request.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random
algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
Cisco IOS Voice, Video, Fax Command Reference
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sgcp retransmit timer
sgcp retransmit timer
To configure the Simple Gateway Control Protocol (SGCP) retransmission timer to use a random
algorithm, use the sgcp retransmit timer command in global configuration mode. To restore the default
value, use the no form of this command.
sgcp retransmit timer {random}
no sgcp retransmit timer {random}
Syntax Description
random
Defaults
The SGCP retransmission timer does not use the random algorithm.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco 3600 and
Cisco MC3810 multiservice concentrator in a private release that
was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally
available on the Cisco 3600 series router and the Cisco MC3810
multiservice concentrator
Usage Guidelines
Enables the SGCP retransmission timer to use a random algorithm.
Use this command to enable the random algorithm component of the retransmission timer. For example,
if the retransmission timer is set to 200 milliseconds, the first retransmission timer is 200 milliseconds,
but the second retransmission timer picks up a timer value randomly between either 200 or 400. The third
retransmission timer picks up a timer value randomly of 200, 400, or 800 as shown below:
•
First retransmission timer: 200
•
Second retransmission timer: 200 or 400
•
Third retransmission timer: 200, 400, or 800
•
Fourth retransmission timer: 200, 400, 800, or 1600
•
Fifth retransmission timer: 200, 400, 800, 1600, or 3200 and so on.
After 30 seconds, the retransmission timer no longer retries.
Examples
The following example shows the retransmission timer set to use the random algorithm:
sgcp retransmit timer random
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
sgcp retransmit timer
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband Tse for fax/modem operation.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
sgcp timer
sgcp timer
To configure how the gateway detects the Real-Time Transport Protocol (RTP) stream lost, use the sgcp
timer command in global configuration mode. To restore the default value, use the no form of this
command.
sgcp timer {receive-rtcp timer | rtp-nse timer}
no sgcp timer {receive-rtcp timer | rtp-nse timer}
Syntax Description
Defaults
receive-rtcp timer
Sets the multiples of the RTP Control Protocol (RTCP) transmission
interval in milliseconds. The valid range is from 1 to 100, and the default
is 5.
rtp-nse timer
Sets the multiples of the RTP named signaling event (NSE) timeout in
milliseconds. The valid range is from 100 to 3000, and the default is
200.
Default for receive-rtcp timer is 5.
Default for rtp-nse timer is 200.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300
universal access server only and was not generally available.
12.0(7)XK
Support for this command was extended to the Cisco MC3810
multiservice concentrator and the Cisco 3600 series routers (except for
the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally available
on the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator
Usage Guidelines
The RTP NSE timer is used for proxy ringing (the ringback tone is provided at the originating gateway).
Examples
The following example shows the receive-rtcp timer set to 100 milliseconds:
sgcp timer receive-rtcp 100
The following example shows the rtp-nse timer set to 1000 milliseconds:
sgcp timer rtp-nse 1000
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
sgcp timer
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp
graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp
max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru Enables SGCP modem or fax pass-through.
sgcp
quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer Configures the SGCP retransmission timer to use a random algorithm
method.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
sgcp tse payload
sgcp tse payload
To enable Inband Telephony Signaling Events (TSE) for fax and modem operation, use the sgcp tse
payload command in global configuration mode. To restore the default value, use the no form of this
command.
sgcp tse payload type
no sgcp tse payload type
Syntax Description
type
Defaults
Zero (0)
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 multiservice
concentrator and Cisco 3600 series routers (except the Cisco 3620) in a
private release that was not generally available.
12.1(2)T
This command was integrated into 12.1(2)T and was generally available
on the Cisco 3600 series router and the Cisco MC3810 multiservice
concentrator
Usage Guidelines
Sets the TSE payload type. The valid range is from 96 to 119. The
default is 0, meaning that the command is disabled.
Because this command is disabled by default, you must specify a TSE payload type.
If you configure the sgcp modem passthru command to the nse value, then you must configure this
command.
Examples
The following example shows the Simple Gateway Control Protocol (SGCP) modem pass-through set
using the NSE-based modem upspeed and the Inband Telephony Signaling Events payload value set to
110:
sgcp modem passthru nse
sgcp tse payload 110
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
sgcp tse payload
Command
Description
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer
disable
Disables the SGCP quarantine buffer.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete”
messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent
indicating that the T1 controller is up or down so that the call agent can
synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm
method.up or down so that the call agent can synchronize
sgcp timer
Configures how the gateway detects the RTP stream host.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show aal2 profile
show aal2 profile
To display the ATM adaptation layer 2 (AAL2) profiles configured on the system, use the show aal2
profile command in privileged EXEC mode.
show aal2 profile all | {itut profile-number | custom profile-number | atmf profile-number}
Syntax Description
all
Displays International Telecommunication Union Telecommunication
Standardization Sector (ITU-T), ATM Forum, and custom AAL2
profiles configured on the system.
itut
Displays ITU-T profiles configured on the system.
profile-number
Specifies the profile number of the AAL2 profile to display. The
available choices are as follows:
For ITU-T:
•
1 = G.711 u-law
•
2 = G.711 u-law with silence insertion descriptor (SID)
•
7 = G.711 u-law and G.729ar8
For ATMF: None. ATMF is not supported.
For custom:
•
100 = G.711 u-law and G.726r32
•
110 = G.711 u-law, G.726r32, and G.729ar8
custom
Displays custom profiles configured on the system.
atmf
Displays ATM Forum profiles configured on the system.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)XA
This command was introduced on the Cisco MC3810 multiservice
concentrator.
12.1(2)T
This command was integrated into the 12.1(2)T release.
Usage Guidelines
This command applies to AAL2 Voice over ATM (VoATM) applications on the Cisco MC3810
multiservice concentrator.
Use the show aal2 profile EXEC command to display the AAL2 profiles configured in the system.
Examples
The following is sample output from the show aal2 profile command for displaying all the profiles
configured in the system:
Router# show aal2 profile all
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show aal2 profile
Printing all the Profiles in the system
Profile Type: ITUT Profile Number: 1 SID Support: 0
Red enable: 1 Num entries: 1
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Profile Type: ITUT Profile Number: 2 SID Support: 1
Red enable: 1 Num entries: 1
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Profile Type: custom Profile Number: 100 SID Support: 1
Red enable: 1 Num entries: 2
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7
Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15
Profile Type: ITUT Profile Number: 7 SID Support: 1
Red enable: 1 Num entries: 2
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Coding type: g729ar8 Packet length: 10 UUI min: 0 UUI max: 15
Profile Type: custom Profile Number: 110 SID Support: 1
Red enable: 1 Num entries: 3
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7
Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15
Coding type: g729ar8 Packet length: 30 UUI min: 8 UUI max: 15
Table 26 provides an alphabetical listing of the fields in this output and a description of each field.
Table 26
Related Commands
show aal2 profile Field Descriptions
Field
Description
Profile Type
Category of codec types configured on DSP. Possible types are ITU-T,
ATMF, and custom.
ITUT Profile Number
Predefined combination of one or more codec types configured for a digital
signal processor (DSP).
SID Support
Silence insertion descriptor.
Red enable
Redundancy enable for type3 packets.
Num entries
Number of profile elements.
Coding type
Voice compression algorithm.
Packet length
Sample size.
UUI min
Minimum sequence number on the voice packets.
UUI max
Maximum sequence number on the voice packets.
Command
Description
codec aal2-profile
Sets the codec profile for a DSP on a per-call basis.
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show atm video-voice address
show atm video-voice address
To display the network service access point (NSAP) address for the ATM interface, enter the show atm
video-voice address command in privileged EXEC mode.
show atm video-voice address
Syntax Description
This command has no keywords or arguments.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(5)XK
This command was introduced for the Cisco MC3810 multiservice
concentrator.
12.0(7)T
Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release
12.0(7)T.
Usage Guidelines
Enter this command to review ATM interface NSAP addresses that have been assigned with the atm
video aesa command and to ensure that ATM management is confirmed for those addresses.
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays ATM interface NSAP
addresses:
Router# show atm video-voice address
nsap address
47.0091810000000002F26D4901.00107B4832E1.FE
47.0091810000000002F26D4901.00107B4832E1.C8
Related Commands
type
VOICE_AAL5
VIDEO_AAL1
ilmi status
Confirmed
Confirmed
Command
Description
codec aal2-profile
Sets the codec profile for a DSP on a per-call basis.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show backhaul-session-manager group
show backhaul-session-manager group
To display status, statistics, or configuration information for all available session-groups, use the
show backhaul-session-manager group command in privileged EXEC mode.
show backhaul-session-manager group { status | stats | cfg } { all | name group-name }
Syntax Description
status
Displays status information for session-groups.
stats
Displays statistics for session-groups.
cfg
Displays configuration information for session-groups.
all
Displays information for all available session-groups.
name group-name
Displays information for a specific session-group. The
group-name argument specifies the name of the
session-group.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
The following displays statistics for all session-groups:
Router# show backhaul-session-manager group stats all
Session-Group grp1 statistics
Successful Fail-Overs
:0
Un-Successful Fail-Over attempts:0
Active Pkts receive count :0
Standby Pkts receive count :0
Total PDUs dispatch err
:0
The following displays the current configuration for all session-groups:
Router# show backhaul-session-manager group cfg all
Session-Group
Group Name :grp1
Set Name
:set1
Sessions
:3
Dest:10.5.0.3 8304 Local:10.1.2.15 8304 Priority:0
Dest:10.5.0.3 8300 Local:10.1.2.15 8300 Priority:2
Dest:10.5.0.3 8303 Local:10.1.2.15 8303 Priority:2
RUDP Options
timer cumulative ack :100
timer keepalive
:1000
timer retransmit
:300
timer transfer state :2000
receive max
:32
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show backhaul-session-manager group
cumulative ack max
retrans max
out-of-sequence max
auto-reset max
:3
:2
:3
:5
The following displays the current status of all session-groups. This group named grp1 belongs to the
set named set1.
The Status will be either Group-OutOfService (no session in the group has been established) or
Group-Inservice (at least one session in the group has been established).
The Status(use) will be either Group-Standby (the VSC connected to the other end of this group will go
into standby mode), Group-Active (the VSC connected to the other end of this group will be the active
VSC), or Group-None (the VSC has not declared its intent yet).
Router# show backhaul-session-manager group status all
Session-Group
Group Name
:grp1
Set Name
:set1
Status
:Group-OutOfService
Status (use) :Group-None
Related Commands
Command
Description
show backhaul-session-manager
session
Displays status, statistics, or configuration of sessions.
show backhaul-session-manager
set
Displays session-groups associated with a specific or all
session-sets.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show backhaul-session-manager session
show backhaul-session-manager session
To display various information for about a session or sessions, use the show backhaul-session-manager
session command in privileged EXEC mode.
show backhaul-session-manager session { all | ip ip_address }
Syntax Description
all
All available sessions.
ip ip_address
The IP address of the local or remote session.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To display information for all available sessions, see the following example.
The State will be OPEN (the connection is established), OPEN_WAIT (the connection is awaiting
establishment), OPEN_XFER (session failover is in progress for this session, which is a transient state),
or CLOSE (this session is down, also a transient state). The session will move to OPEN_WAIT after
waiting a fixed amount of time.
The Use-status field indicates whether PRI signaling traffic is currently being transported over this
session . The field will be either OOS (this session is not being used to transport signaling traffic) or IS
(this session is being used currently to transport all PRI signaling traffic). OOS does not indicate if the
connection is established and IS indicates that the connection is established.
Router# show backhaul-session-manager session all
Session information -Session-id:35
Group:grp1 /*this session belongs to the group named 'grp1' */
Configuration:
Local:10.1.2.15
, port:8303
Remote:10.5.0.3
, port:8303
Priority:2
RUDP Option:Client, Conn Id:0x2
State:
Status:OPEN_WAIT, Use-status:OOS, /*see explanation below */
Statistics:
# of resets:0
# of auto_resets 0
# of unexpected RUDP transitions (total) 0
# of unexpected RUDP transitions (since last reset) 0
Receive pkts - Total:0 , Since Last Reset:0
Recieve failures - Total:0 ,Since Last Reset:0
Transmit pkts - Total:0, Since Last Reset:0
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show backhaul-session-manager session
Transmit Failures (PDU Only)
Due to Blocking (Not an Error) - Total:0, Since Last Reset:0
Due to causes other than Blocking - Total:0, Since Last
Reset:0
Transmit Failures (NON-PDU Only)
Due to Blocking(Not an Error) - Total:0, Since Last Reset:0
Due to causes other than Blocking - Total:0, Since Last
Reset:0
RUDP statistics
Open failures:0
Not ready failures:0
Conn Not Open failures:0
Send window full failures:0
Resource unavailble failures:0
Enqueue failures:0
Related Commands
Command
Description
show backhaul-session-manager
group
Displays status, statistics, or configuration of a specified or
all session-groups.
show backhaul-session-manager
set
Displays session-groups associated with a specified or all
session-sets.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show backhaul-session-manager set
show backhaul-session-manager set
To display session-groups associated with a specified session-set or all session-sets, use the
show backhaul-session-manager set command in privileged EXEC mode.
show backhaul-session-manager set { all | name session-set-name }
Syntax Description
all
All available session-sets.
name session-set-name
A specified session-set.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced.
Examples
To show session groups associated with all session-sets, see the following example:
Router# show backhaul-session-manager set all
Related Commands
Command
Description
show backhaul-session-manager
group
Displays status, statistics, or configuration of a specified or
all session-groups.
show backhaul-session-manager
session
Displays status, statistics, or configuration of a session or all
sessions.
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call active
show call active
To display active call information for voice calls or fax transmissions in progress, use the show call
active command in user EXEC or privileged EXEC mode.
show call active {voice | fax}[brief]
Syntax Description
voice
Specifies that information be displayed for all active voice calls.
fax
Specifies that information be displayed for all active fax calls.
brief
(Optional) Displays a truncated version of the active call information.
Defaults
No default behavior or values.
Command Modes
User EXEC or
Privileged EXEC
Command History
Release
Usage Guidelines
Modification
11.3(1)T
This command was introduced on the Cisco 2600 series and 3600 series.
12.0(3)XG
Support for VoFR was added.
12.0(4)XJ
This command was modified for store-and-forward fax on the Cisco AS5300
universal access server.
12.0(4)T
This command was first supported on the Cisco 7200 series.
12.0(7)XK
This command was first supported on the Cisco MC3810 multiservice
concentrator.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
This command was modified for Modem Passthrough over VoIP on the
Cisco AS5300 universal access server.
Use the show call active command to display the contents of the active call table. This command
displays information about call times, dial peers, connections, quality of service, and other status and
statistical information. If you use the voice keyword, information is displayed about all voice calls
currently connected through the router or access server. If you use the fax keyword, information is
displayed about all fax calls currently connected.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
See Table 19 for a listing of the information types associated with this command.
Cisco IOS Voice, Video, Fax Command Reference
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show call active
Examples
The following is sample output from the show call active voice command:
Router# show call active voice
GENERIC:
SetupTime=104443 ms
Index=1
PeerAddress=50110
PeerSubAddress=
PeerId=100
PeerIfIndex=104
LogicalIfIndex=10
ConnectTime=104964
CallDuration=00:02:43
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=2
TransmitPackets=15720
TransmitBytes=2362904
ReceivePackets=15670
ReceiveBytes=2737904
TELE:
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
TxDuration=155310 ms
VoiceTxDuration=155310 ms
FaxTxDuration=0 ms
CoderTypeRate=g711ulaw
NoiseLevel=-75
ACOMLevel=11
OutSignalLevel=-13
InSignalLevel=-22
InfoActivity=2
ERLLevel=27
SessionTarget=
ImgPages=0
GENERIC:
SetupTime=104648 ms
Index=1
PeerAddress=55240
PeerSubAddress=
PeerId=2
PeerIfIndex=105
LogicalIfIndex=0
ConnectTime=104964
CallDuration=00:02:47
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=16026
TransmitBytes=2608248
ReceivePackets=16075
ReceiveBytes=2609164
VOIP:
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
RemoteUDPPort=18202
RoundTripDelay=2 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=TRUE
SessionProtocol=cisco
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show call active
SessionTarget=ipv4:1.14.82.14
OnTimeRvPlayout=40
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
ReceiveDelay=67 ms
LostPackets=0 ms
EarlyPackets=0 ms
LatePackets=0 ms
VAD = enabled
CoderTypeRate=g729r8
CodecBytes=20
SignalingType=cas
Modem passthrough signaling method is nse:
Buffer Fill Events = 0
Buffer Drain Events = 0
Percent Packet Loss = 0
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 157sec
Time between Buffer Drain/Fills = Min 0sec Max 0sec
The following is sample output from the show call active voice brief command:
Router# show call active voice brief
<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
3
3
: 104443hs.1 +521 pid:100 Answer 50110 active
dur 00:03:28 tx:20151/3036404 rx:20102/3517936
Tele 0:D:1: tx:199630/199630/0ms g711ulaw noise:-75 acom:11
i/0:-22/-13 dBm
: 104648hs.1 +316 pid:2 Originate 55240 active
dur 00:03:28 tx:20102/3276712 rx:20151/3277628
IP 1.14.82.14:18202 rtt:3ms pl:40/0ms lost:0/0/0 delay:67/67/67ms g729r8
MODEMPASS nse buf:0/0 loss 0% 0/0 last 195s dur:0/0s
The following is sample output from the show call active fax command:
Router# show call active fax
GENERIC:
SetupTime=22021 ms
Index=1
PeerAddress=wook song
PeerSubAddress=
PeerId=0
PeerIfIndex=0
LogicalIfIndex=0
ConnectTime=24284
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show call active
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=10
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=41190
MMOIP:
ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]
RemoteIPAddress=0.0.0.0
SessionProtocol=SMTP
SessionTarget=
MessageId=
AccountId=
ImgEncodingType=MH
ImgResolution=fine
AcceptedMimeTypes=2
DiscardedMimeTypes=1
Notification=None
GENERIC:
SetupTime=23193 ms
Index=1
PeerAddress=527....
PeerSubAddress=
PeerId=3469
PeerIfIndex=157
LogicalIfIndex=30
ConnectTime=24284
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=10
TransmitPackets=5
TransmitBytes=6513
ReceivePackets=0
ReceiveBytes=0
TELE:
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
TxDuration=24010 ms
FaxTxDuration=10910 ms
FaxRate=14400
NoiseLevel=-1
ACOMLevel=-1
OutSignalLevel=0
InSignalLevel=0
InfoActivity=0
ERLLevel=-1
SessionTarget=
ImgPages=0
The following is sample output from the show call active fax brief command:
Router# show call active fax brief
<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state> \
tx:<packets>/<bytes> rx:<packets>/<bytes> <state>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call active
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
1
: 22021hs.1 +2263 pid:0 Answer wook song active
tx:0/0 rx:0/41190
IP 0.0.0.0 AcceptedMime:2 DiscardedMime:1
1
: 23193hs.1 +1091 pid:3469 Originate 527.... active
tx:10/13838 rx:0/0
Tele : tx:31200/10910/20290ms noise:-1 acom:-1 i/0:0/0 dBm
Table 27 provides an alphabetical listing of the fields displayed in the output from the show call active
command and a description of each field.
Table 27
show call active Field Descriptions
Field
Description
ACOM Level
Current ACOM level for this call. ACOM is the combined loss achieved
by the echo canceler, which is the sum of the Echo Return Loss, Echo
Return Loss Enhancement, and nonlinear processing loss for the call.
Buffer Drain Events
Total number of jitter buffer drain events.
Buffer Fill Events
Total number of jitter buffer fill events.
CallDuration
Length of the call in hours, minutes, and seconds, hh:mm:ss.
CallOrigin
Call origin: answer or originate.
CallState
Current state of the call.
ChargedUnits
Total number of charging units that apply to this peer since system
startup. The unit of measure for this field is hundredths of second.
CodecBytes
Payload size in bytes for the codec used.
CoderTypeRate
Negotiated coder rate. This value specifies the send rate of voice or fax
compression to its associated call leg for this call.
ConnectionId
Global call identifier for this gateway call.
ConnectTime
Time at which the call was connected.
Consecutive-packets-lost
Events
Total number of consecutive (two or more) packet-loss events.
Corrected packet-loss
Events
Total number of packet loss events that were corrected using the
RFC 2198 method.
Dial-Peer
Tag of the dial peer sending this call.
ERLLevel
Current Echo Return Loss (ERL) level for this call.
FaxTxDuration
Duration of fax transmission from this peer to the voice gateway for this
call. You can derive the Fax Utilization Rate by dividing the
FaxTxDuration value by the TxDuration value.
GapFillWithInterpolation
Duration of a voice signal played out with a signal synthesized from
parameters, or samples of data preceding and following in time because
voice data was lost or not received in time from the voice gateway for this
call.
GapFillWithRedundancy
Duration of a voice signal played out with a signal synthesized from
available redundancy parameters because voice data was lost or not
received in time from the voice gateway for this call.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call active
Table 27
show call active Field Descriptions (continued)
Field
Description
GapFillWithPrediction
Duration of the voice signal played out with signal synthesized from
parameters, or samples of data preceding in time, because voice data was
lost or not received in time from the voice gateway for this call. Examples
of such pullout are frame-eraser or frame-concealment strategies in
G.729 and G.723.1 compression algorithms.
GapFillWithSilence
Duration of a voice signal replaced with silence because voice data was
lost or not received in time for this call.
HiWaterPlayoutDelay
High-water mark Voice Playout FIFO Delay during this call.
Index
Dial peer identification number.
InfoActivity
Active information transfer activity state for this call.
InfoType
Information type for this call, for example, voice or fax.
InSignalLevel
Active input signal level from the telephony interface used by this call.
Last Buffer Drain/Fill
Event
Time since the last jitter buffer drain or fill event, in seconds.
LogicalIfIndex
Index number of the logical interface for this call.
LoWaterPlayoutDelay
Low water mark Voice Playout FIFO Delay during this call.
Modem passthrough
signaling method in use
Indicates that this is a modem pass-through call and that named signaling
events (NSEs)—also called telephone-events in RFC 2833—are used for
signaling codec upspeed. The upspeed method is the method used to
dynamically change the codec type and speed to meet network
conditions. This means that you might move to a faster codec when you
have both voice and data calls and then slow down when there is only
voice traffic.
NoiseLevel
Active noise level for this call.
OnTimeRvPlayout
Duration of voice playout from data received on time for this call. Derive
the Total Voice Playout Duration for Active Voice by adding the
OnTimeRvPlayout value to the GapFill values.
OutSignalLevel
Active output signal level to the telephony interface used by this call.
PeerAddress
Destination pattern or number associated with this peer.
PeerId
ID value of the peer table entry to which this call was made.
PeerIfIndex
Voice port index number for this peer. For ISDN media, this would be the
index number of the B channel used for this call.
PeerSubAddress
Subaddress when this call is connected.
Percent Packet Loss
Total percent packet loss.
ReceiveBytes
Number of bytes received by the peer during this call.
ReceiveDelay
Average Playout FIFO Delay plus the Decoder Delay during this voice
call.
ReceivePackets
Number of packets received by this peer during this call.
RemoteIPAddress
Remote system IP address for the VoIP call.
RemoteUDPPort
Remote system UDP listener port to which voice packets are sent.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call active
Table 27
Related Commands
show call active Field Descriptions (continued)
Field
Description
RoundTripDelay
Voice packet round trip delay between the local and remote systems on
the IP backbone for this call.
SelectedQoS
Selected RSVP quality of service (QoS) for this call.
SessionProtocol
Session protocol used for an Internet call between the local and remote
routers through the IP backbone.
SessionTarget
Session target of the peer used for this call.
SetupTime
Value of the system UpTime when the call associated with this entry was
started.
SignalingType
Signaling type for this call; for example, channel-associated signaling
(CAS) or common-channel signaling (CCS).
Time between Buffer
Drain/Fills
Minimum and maximum durations between jitter buffer drain or fill
events, in seconds.
TransmitBytes
Number of bytes sent by this peer during this call.
TransmitPackets
Number of packets sent by this peer during this call.
TxDuration
Duration of transmit path open from this peer to the voice gateway for
this call.
VAD
Whether voice activation detection (VAD) was enabled for this call.
VoiceTxDuration
Duration of voice transmission from this peer to the voice gateway for
this call. Derive the Voice Utilization Rate by dividing the
VoiceTxDuration value by the TxDuration value.
Command
Description
show call history
Displays the call history table.
show dial-peer voice
Displays configuration information for dial peers.
show num-exp
Displays how the number expansions are configured in Voice over IP.
show voice port
Displays configuration information about a specific voice port.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call application voice
show call application voice
To define the names of the audio files that the interactive voice response (IVR) script will play, the
operation of the abort keys, the prompts that are used, and caller interaction, use the show call
application voice command in EXEC mode.
show call application voice [name | summary]
Syntax Description
name
(Optional) The name of the desired IVR application.
summary
(Optional) Displays a one-line summary. If the command is entered
without the summary keyword, a complete detailed description is
displayed of the application.
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 series and
Cisco 3600 series routers and the Cisco AS5300 universal access
server.
Usage Guidelines
If the name of a specific application is entered, it will give information about that application.
If the summary keyword is entered, a one-line summary will be displayed about each application.
If the command is entered without the summary, a detailed description of the entered IVR application
is displayed.
Examples
This example shows the output for the clid_authen_collect IVR script:
Router# show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
and goto state start
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
Do Action IVR_ACT_END.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call application voice
and do nothing
State get_account has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
patName=account
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_pin.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
patName=pin
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_destination.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
destination= called=
calling=
account=
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call application voice
State authenticate_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY.
URL: flash:auth_failed.au
allowInt=0, pContent=0x0
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
Router# show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
and goto state start
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
Do Action IVR_ACT_END.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
and do nothing
State get_account has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
patName=account
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_pin.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
patName=pin
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call application voice
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_destination.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
destination= called=
calling=
account=
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY.
URL: flash:auth_failed.au
allowInt=0, pContent=0x0
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
Related Commands
Command
Description
call application voice
Defines the name to be used for an application and indicates the location
of the appropriate IVR script to be used with this application.
call application voice load Reloads the designated TCL script.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call fallback cache
show call fallback cache
To see the current Calculated Planning Impairment Factor (ICPIF) estimates for all IP addresses in cache,
use the show call fallback cache command in EXEC mode.
show call fallback cache [ip-address]
Syntax Description
ip-address
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series and
Cisco 3600 series routers and on the Cisco MC3810 multiservice
concentrator.
(Optional) Specifies a specific IP address.
Usage Guidelines
To clear all entries in the cache, use the clear call fallback cache command.
Examples
The following example displays output from the show call fallback cache command:
Router# show call fallback cache
Probe
----1
2
IP Address
---------1.1.1.4
122.24.56.25
Codec
----g729r8
g729r8
Delay
----40
14810
Loss
---00
5
ICPIF
----0
1
Reject
-----9
4
Accept
------
2 active probes
Field
------Probe
IP Address
Codec
Delay
Loss
ICPIF
Reject
Accept
active probes
Cisco IOS Voice, Video, Fax Command Reference
VR-666
Description
-----------Probe number
IP Address to which the probe is sent
Codec Type of the probe
Delay in milliseconds that the probe incurred
Loss in % that the probe incurred
Computed ICPIF value for the probe
Number of times that calls of Codec Type <Codec>
were rejected to the IP Address
Number of times that calls of Codec Type <Codec>
were accepted to the IP Address
Number of destinations being probed
Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call fallback cache
Router# show call fallback cache 10.14.115.53
Probe
----1
IP Address
---------10.14.115.53
Codec
----g729r8
ICPIF
----0
Reject
-----0
Accept
-----2
1 active probes
Related Commands
Command
Description
show call fallback stats
Displays the call fallback statistics.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call fallback config
show call fallback config
To display the call fallback configuration, use the show call fallback config command in EXEC mode.
show call fallback config
Syntax Description
This command has no arguments or keywords.
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series and
Cisco 3600 series routers and on the Cisco MC3810 multiservice
concentrator.
Examples
The following example displays output from the show call fallback config command:
Router# show call fallback config
VoIP fallback config:
Fallback is ON
Using ICPIF threshold:
ICPIF value timeout:20 seconds
ICPIF threshold:20
Number of packets in a probe:20
IP precedence of probe packets:2
Fallback cache size:2 entries
Fallback cache timeout:240 seconds
Instantaneous value weight:65
MD5 Keychain:secret
Related Commands
Command
Description
call fallback monitor
Enables the monitoring of destinations without fallback to alternate
dial peers.
show voice trunk-conditioning Enables fallback to alternate dial peers in case of network
signaling
congestion.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call fallback stats
show call fallback stats
To display the call fallback statistics, use the show call fallback stats command in EXEC mode.
show call fallback stats
Syntax Description
This command has no arguments or keywords.
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series and Cisco 3600
series routers and on the Cisco MC3810 multiservice concentrator.
Usage Guidelines
To remove all values, use the clear call fallback stats command.
Examples
The following example displays output from the show call fallback stats command:
Router# show call fallback stats
VOIP Fallback Stats:
Total accepted calls:3
Total rejected calls:1
Total cache overflows:1
Field
------Total accepted calls
Total rejected calls
Total cache overflows
pruning.
Related Commands
Description
-----------Number of times that calls were successful over IP.
Number of times that calls were rejected over IP.
Number of times that the fallback cache overflowed and requied
Command
Description
clear call fallback stats
Clears the call fallback statistics.
show call fallback cache
Displays the current ICPIF estimates for all IP addresses in the cache.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call history
show call history
To display the call history table for voice calls or fax transmissions, use the show call history command
in user EXEC or privileged EXEC mode.
show call history {voice | fax}[last number | brief]
Syntax Description
voice
Specifies that call history information be displayed for voice calls.
fax
Specifies that call history information be displayed for fax calls.
last number
(Optional) Displays the last calls connected, where the number of
calls that appear is defined by the number argument. Valid values are
from 1 to 100.
brief
(Optional) Displays a truncated version of the call history table.
Defaults
No default behavior or values.
Command Modes
User EXEC
Privileged EXEC
Command History
Usage Guidelines
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
12.0(3)XG
Support for Voice over Frame Relay (VoFR) was added on the
Cisco 2600 and Cisco 3600 series.
12.0(4)XJ
This command was modified for store-and-forward fax.
12.0(4)T
The brief keyword was added and the command was first supported
on the Cisco 7200 series.
12.0(7)XK
Support for the brief keyword was added on the Cisco MC3810
multiservice concentrator.
12.1(2)T
This command was integrated into Cisco IOS 12.1(2)T.
The show call history command displays a call history table containing a list of voice or fax calls
connected through the router in descending time order. The maximum number of calls contained in the
table can be set to a number between 0 and 500 using the dial-control-mib command in global
configuration mode. The default maximum number of table entries is 50. Each call record is aged out of
the table after a configurable number of minutes has elapsed, also specified by the dial-control-mib
command. The default timer value is 15 minutes.
You can display subsets of the call history table by using specific keywords. To display the last calls
connected through this router, use the keyword last, and define the number of calls to be displayed with
the number argument.
Cisco IOS Voice, Video, Fax Command Reference
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Cisco IOS Voice, Video, and Fax Commands: R Through Sh
show call
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