12 Voice Configuration Guide Book

H3C MSR Series Routers
Voice Configuration Guide(V7)
Hangzhou H3C Technologies Co., Ltd.

http://www.h3c.com
Software version: MSR-CMW710-R0007
Document version: 6W100-20140320
Copyright 2014, Hangzhou H3C Technologies Co., Ltd. and its licensors
All rights reserved

No part of this manual may be reproduced or transmitted in any form or by any means without prior
written consent of Hangzhou H3C Technologies Co., Ltd.
Trademarks
, IRF, NetPilot, Netflow,
H3C,
, H3CS, H3CIE, H3CNE, Aolynk,
, H3Care,
SecEngine, SecPath, SecCenter, SecBlade, Comware, ITCMM and HUASAN are trademarks of
Hangzhou H3C Technologies Co., Ltd.
All other trademarks that may be mentioned in this manual are the property of their respective owners
Notice
The information in this document is subject to change without notice. Every effort has been made in the
preparation of this document to ensure accuracy of the contents, but all statements, information, and
recommendations in this document do not constitute the warranty of any kind, express or implied.
Preface
The H3C MSR documentation set includes 14 configuration guides, which describe the software features
for the H3C MSR Series Routers and guide you through the software configuration procedures. These
configuration guides also provide configuration examples to help you apply software features to different
network scenarios.
The Voice Configuration Guide(V7) describes fundamentals and configuration of Voice Entity, Voice
Subscriber, Dial Plan, SIP, H.323, Call Services, and so on.
This preface includes:
Audience
Conventions
About the H3C MSR documentation set
Obtaining documentation
Technical support
Documentation feedback
These configuration guides apply to the following models of the H3C MSR series routers:
Model
MSR 2600
MSR 26-30
MSR 3600
MSR 5600
MSR 36-10
MSR 36-20
MSR 36-40
MSR 36-60
MSR3600-28
MSR3600-51
MSR 56-60
MSR 56-80
Audience
This documentation is intended for:
Network planners
Field technical support and servicing engineers
Network administrators working with the routers
Conventions
This section describes the conventions used in this documentation set.
Command conventions
Convention
Description
Boldface
Bold text represents commands and keywords that you enter literally as shown.
Italic
Italic text represents arguments that you replace with actual values.
[]
Square brackets enclose syntax choices (keywords or arguments) that are optional.
{ x | y | ... }
Braces enclose a set of required syntax choices separated by vertical bars, from which
you select one.
[ x | y | ... ]
Square brackets enclose a set of optional syntax choices separated by vertical bars, from
which you select one or none.
{ x | y | ... } *
Asterisk marked braces enclose a set of required syntax choices separated by vertical
bars, from which you select at least one.
[ x | y | ... ] *
Asterisk marked square brackets enclose optional syntax choices separated by vertical
bars, from which you select one choice, multiple choices, or none.
&<1-n>
The argument or keyword and argument combination before the ampersand (&) sign can
be entered 1 to n times.
A line that starts with a pound (#) sign is comments.
Convention
Description
Symbols
WARNING
An alert that calls attention to important information that if not understood or followed can
result in personal injury.
CAUTION
An alert that calls attention to important information that if not understood or followed can
result in data loss, data corruption, or damage to hardware or software.
IMPORTANT
An alert that calls attention to essential information.

An alert that contains additional or supplementary information.
NOTE
TIP
An alert that provides helpful information.
Network topology icons

Represents a generic network device, such as a router, switch, or firewall.
Represents a routing-capable device, such as a router or Layer 3 switch.
Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that supports
Layer 2 forwarding and other Layer 2 features.
Port numbering in examples

The port numbers in this document are for illustration only and might be unavailable on your device.
About the H3C MSR documentation set

The H3C MSR documentation set includes:
Category
Documents
Purposes
Product description and

specifications
Marketing brochures
Describe product specifications and benefits.
Installation guide
Provides a complete guide to hardware installation

and hardware specifications.
MSR Series Routers

Interface Module Manual
Provide the hardware specifications of cards.
MSR Series Routers

Configuration guides(v7)
Describe software features and configuration

procedures.
MSR Series Routers

Command references(v7)
Provide a quick reference to all available

commands.
Hardware specifications
and installation
Software configuration
Operations and
maintenance
Release notes
Provide information about the product release,

including the version history, hardware and software
compatibility matrix, version upgrade information,
technical support information, and software
upgrading.
Obtaining documentation
You can access the most up-to-date H3C product documentation on the World Wide Web
at http://www.h3c.com.
Click the links on the top navigation bar to obtain different categories of product documentation:
[Technical Support & Documents > Technical Documents]Provides hardware installation, software
upgrading, and software feature configuration and maintenance documentation.
[Products & Solutions]Provides information about products and technologies, as well as solutions.
[Technical Support & Documents > Software Download]Provides the documentation released with the
software version.
Technical support
service@h3c.com
http://www.h3c.com
Documentation feedback
You can e-mail your comments about product documentation to info@h3c.com.
We appreciate your comments.
Contents
Configuring analog voice interfaces 1
FXS interface 1
FXO interface 1
E&M interface 1
Configuration task list 1
Configuring basic functions 2
Configuring call progress tones 2
Configuring an FXS interface 3
Configuring CID 3
Setting the electrical impedance 4
Configuring the packet loss compensation mode 5
Configuring an FXS interface to send LCFO signals 5
Configuring an FXO interface 6
Configuring CID 6
Configuring busy tone detection 6
Configuring an on-hook delay 9
Configuring the off-hook mode 10
Configuring ring detection parameters 10
Setting the electrical impedance 11
Configuring the packet loss compensation mode 11
Binding an FXS interface to an FXO interface 11
Configuring an E&M interface 12
Configuring the cable type 12
Configuring the signal type 12
Configuring a start mode for E&M signaling 12
Configuring E&M non-signaling mode 15
Enabling E&M control signals pass-through 16
Configuring the output gain of SLIC chip 16
Configuring DTMF 16
Configuring DTMF tone sending 17
Configuring DTMF tone detection 17
Adjusting parameters for voice interfaces 18
Adjusting gains 18
Adjusting timing parameters 18
Configuring the comfortable noise function 19
Configuring echo cancellation 19
Displaying and maintaining analog voice interfaces 21
Analog voice interface configuration examples 21
Two-dial configuration example for the FXO interface 21
FXO interface PLAR configuration example 22
E&M interface configuration example 24
E&M non-signaling mode configuration example 25
FXS&FXO 1:1 binding configuration example 27
Configuring digital voice interfaces 29
E1 and T1 interfaces 29
BSV interfaces 29
Configuring basic parameters for an E1 interface 30
i
Configuring a TDM clock source 30

Configuring other parameters 31
Configuring basic parameters for a T1 interface 31
Configuring a TDM clock source 31
Configuring other parameters 32
Configuring a BSV interface 32
Configuring a BRI interface 33
Configuring a digital voice interface 33
Configuring a timeslot set 33
Creating a timeslot set 33
Configuring a logical digital voice interface 34
Configuring R2 signaling 34
Configuring basic R2 signaling parameters 39
Configuring R2 digital line signaling 42
Configuring R2 interregister signaling 43
Configuring the ISDN protocol 44
Binding a digital voice interface to a POTS entity 45
Displaying and maintaining digital voice interfaces 45
Digital voice interface configuration examples 45
R2 signaling configuration example 45
E1 DSS1 signaling configuration example 47
BSV DSS1 signaling configuration example 49
Configuring voice entities 52

Overview 52
Configuring a POTS entity 52
Creating a POTS entity and configuring basic parameters 53
Configuring codecs for a POTS entity 53
Configuring a POTS entity to register with the registrar 54
Configuring DTMF for a POTS entity 55
Enabling VAD for a POTS entity 55
Configuring options related to dial program 56
Configuring a VoIP entity 56
Creating a VoIP entity and configuring basic parameters 57
Configuring codecs for a VoIP entity 57
Configuring DTMF for a VoIP entity 58
Enabling VAD for a VoIP entity 59
Configuring options related to dial program 59
Displaying and maintaining voice entities 60
Configuring dial programs 61
Configuring caller control 61
Procedure 61
Examples 62
Configuring caller group control 63
Configuring a subscriber group 63
Configuring a caller group on a voice entity 63
Enabling private line auto ring-down 64
Configuring a number match mode 64
Procedure 64
Examples 65
Configuring the maximum number of total calls allowed by a voice entity 66
ii
Configuring number substitution 66

Number substitution on the calling router 67
Number substitution on the called router 67
Configuring global number substitution 68
Configuring number substitution for a voice entity 69
Examples 70
Configuring a priority for a voice entity 71
Configuring a number sending mode 71
Procedure 71
Examples 71
Configuring a dial prefix 73
Dial program configuration examples 73
Configuring number substitution 74
Configuring caller group control 76
Configuring SIP 79
Overview 79
Terminology 79
SIP functions 79
SIP messages 80
SIP configuration task list 80
Configuring SIP UA registration 81
Configuration prerequisites 81
Configuring SIP credentials 82
Enabling a POTS entity to register with the registrar 84
Configuring registrar information 84
Displaying SIP UA registration status 85
Configuring the call destination address for a VoIP entity 85
Configuring the call destination IP address for a VoIP entity 85
Configuring a VoIP entity to obtain the call destination address from a proxy server 85
Configuring the destination domain name and port number for a VoIP entity 86
Configuring extended SIP functions 86
Configuring out-of-band DTMF 86
Configuring periodic refresh of SIP sessions 87
Configuring PSTN cause-to-SIP status mappings 87
Configuring caller privacy 88
Setting the P-Asserted-Identity or P-Preferred-Identity header field 88
Displaying and maintaining SIP 89
SIP UA configuration examples 89
Configuring direct SIP calling 89
Configuring SIP calling through a SIP server 90
Configuring SIP calling through DNS 92
Configuring out-of-band DTMF 93
Configuring SIP trunk 96
Background 96
Features 97
Typical applications 97
Protocols and standards 98
SIP trunk configuration task list 98
Enabling SIP-to-SIP calling 98
Configuring a SIP trunk account 99
Enabling codec transparent transmission 99
Enabling media flow-around 99
Enabling DO-EO conversion 100
iii
Displaying and maintaining SIP trunk 100

SIP trunk configuration examples 101
Configuring call services 103

Call waiting 103
Call hold 103
Call forwarding 103
Call transfer 103
Call backup 104
Call services configuration task list 104
Configuring the call hold mode 104
Configuring call forwarding 105
Call services configuration examples 105
Call waiting configuration example 105
Call forwarding configuration example 107
Call transfer configuration example 109
Call backup configuration example 110
Index 112
iv
Configuring analog voice interfaces

Analog voice interfaces include FXS, FXO, and E&M interfaces.
FXS interface
A Foreign Exchange Station (FXS) interface connects to a standard telephone, fax machine, or a Private
Branch Exchange (PBX) through an RJ-11 connector and a telephone cable. It provides ring, voltage, and
dial tone based on level changes on the Tip/Ring line. An FXS interface can only connect to an FXO
interface.
FXO interface
A foreign exchange office (FXO) interface connects to a PBX through an RJ-11 connector and a telephone
cable. It provides ring, voltage, and dial tone based on level changes on the Tip/Ring line. An FXO
interface can only connect to an FXS interface.
E&M interface
An ear & mouth or receive & transmit (E&M) interface is a common trunk line that connects to a PBX
through an RJ-48 connector. The E&M interface supports E&M signaling. The E signaling receives signals
from the peer, and the M signaling sends signals to the peer. An E&M interface can only connect to an
E&M interface.
Configuration task list

Tasks at a glance
(Optional.) Configuring basic functions
(Optional.) Configuring call progress tones
(Optional.) Configuring an FXS interface
Configuring CID
Setting the electrical impedance
Configuring the packet loss compensation mode
Configuring an FXS interface to send LCFO signals
Tasks at a glance
(Optional.) Configuring an FXO interface
Configuring CID
Configuring busy tone detection
Configuring an on-hook delay
Configuring the off-hook mode
Configuring ring detection parameters
(Optional.) Binding an FXS interface to an FXO interface

(Optional.) Configuring an E&M interface
Configuring the cable type

Configuring the signal type
Configuring a start mode for E&M signaling
Configuring E&M non-signaling mode
Enabling E&M control signals pass-through
(Optional.) Configuring DTMF
Configuring DTMF tone sending

Configuring DTMF tone detection
(Optional.) Adjusting parameters for voice interfaces
Adjusting gains
Adjusting timing parameters
Configuring the comfortable noise function
Configuring echo cancellation
Configuring basic functions

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice interface view.
subscriber-line line-number
N/A
3.
Configure a description for

the voice interface.
description text
By default, the description of a

voice interface is interface name
Interface.
4.
Restore the default settings for

the voice interface.
default
N/A
5.
Bring up the voice interface.
undo shutdown
By default, a voice interface is up.
Configuring call progress tones

Perform this task to configure call progress tones for all voice interfaces on the device.
To configure call progress tones:
2
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
Specify a country:
cptone country-type locale
Customize call progress tone

3.
4.
Configure call progress tones.
Configure the amplitude value

for call progress tones.
parameters:
cptone custom { busy-tone |
congestion-tone | dial-tone |
ringback-tone |
special-dial-tone |
waiting-tone } comb freq1
freq2 time1 time2 time3 time4
cptone tone-type { all | busy-tone

| congestion-tone | dial-tone |
ringback-tone | special-dial-tone |
waiting-tone } amplitude value
By default, the call progress tones

of China are used.
The default is 1000 for busy tone

and congestion tone, 400 for dial
tone and special dial tone, and
600 for ringback tone and waiting
tone.
Configuring an FXS interface

This section covers the procedures for configuring an FXS interface.
Configuring CID
Caller identification (CID) enables called terminals to display the calling information, including the
calling number, calling name, date, and time.
CID supports two data formats:
Single-data-message formatContains date, time, and calling number.
Multiple-data-message formatContains date, time, calling number, and calling name.
The called telephone displays the calling number differently as follows:
The called telephone displays the calling number if the terminating device is enabled with CID and
can obtain the calling number.
The called telephone displays the character "O" if the terminating device is enabled with CID but
fails to obtain the calling number (for example, the originating device does not send the calling
number).
The called telephone displays the character "P" if CID is disabled on the terminating device.
The FXS interface sends the CID to the called telephone through frequency shift keying (FSK) modulation
between the first and second rings. For the CID to appear on the called telephone, the called user should
pick up the handset after the second ring.
The CID function requires configurations on both FXS and FXO interfaces. For configuration on the FXO
interface, see "Configuring CID."
Configuration guidelines
The PBX and called telephones must support CID.

3
To ensure correct call time, make sure the router system time transmitted in data-message format
stays synchronous with the local standard time.
For the CID function to operate correctly, keep the cid send command enabled.
Configuration procedure
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXS interface view.
N/A
3.
(Optional.) Configure the

calling name for the FXS
interface.
By default, no calling name is configured.
4.
Enable the FXS interface

to send CID to the remote
end.
calling-name text
The calling name can be sent only in the

multiple-data-message format.
Use this command on the originating
device.
Optional.
By default, this function is enabled.
cid send
Use this command on the originating

device.
By default, the bellcore mode is used.
5.
Configure the standard

mode for sending CID.
cid standard-type { bellcore |

brazil }
This command is for use on the

terminating device, which encapsulates
the CID by using the configured standard
mode and sends it to the called
telephone.
The message format configured by using
the cid type command takes effect only
when the bellcore mode is used.
The default is complex.
6.
Configure the data

message format.
7.
Enable CID on the FXS

interface.
Use this command on the terminating

device.
cid type { complex | simple }
When the originating device supports

only one message format, you must use
that message format on the terminating
device.
By default, CID is enabled.
cid display
Use this command on the terminating

device.

The electrical impedance setting must be subject to country specifications. Each country corresponds to
an impedance value. You specify an impedance value while specifying a country.
To set the electrical impedance:
Step
1.
Enter system view.
Command
Remarks
system-view
N/A
Step
2.
Set the electrical impedance

of a country.
3.
Command
Remarks
N/A
impedance { country-name | r550

| r600 | r650 | r700 | r750 |
r800 | r850 | r900 | r950 }
The default is the electrical

impedance of China.
You must configure the same
electrical impedance value on the
originating and terminating
devices.

You can configure the packet loss compensation mode to alleviate the impact of packet loss on voice
quality:
For discrete packet loss, you can use the general compensation mode to reconstruct lost packets.
For continuous packet loss, you can use the voice gateway-specific compensation mode to
compensate for lost packets.
To configure the packet loss compensation mode:

Step
Command
Remarks
1.
Enter system view.
System-view
N/A
2.
N/A
3.
Configure the packet loss

compensation mode for the
FXS interface.
plc-mode { general | specific }
By default, the specific algorithm

provided by the voice gateway is
used for an FXS interface.
Configuring an FXS interface to send LCFO signals

You can configure an FXS interface to send a loop current feed open (LCFO) signal when the peer goes
on-hook. This function is used mainly in North America.
To configure an FXS interface to send LCFO signals:
Step
Command
Remarks
1.
Enter system view.
System-view
N/A
2.
N/A
3.
Configure the FXS interface to

send an LCFO signal when the
peer goes on-hook.
disconnect lcfo
By default, no LCFO signal is sent

(a busy tone is played to the peer
end).
Set the duration of the LCFO

signal.
timer disconnect-pulse value
The default is 750 milliseconds.
4.
Configuring an FXO interface

This section covers the procedures for configuring an FXO interface.
Configuring CID
The CID function must be configured on both the FXS and FXO interfaces. For information about
configuring this function on the FXS interface, see "Configuring CID."
For the CID function to work correctly, enable both CID receiving and CID sending.
Enabling CID receiving for an FXO interface

The FXO interface receives the CID. By default, it detects the CID from the PBX between the first and
second rings.
To enable CID receiving:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter FXO interface view.
N/A
3.
Configure the time for CID detection

and after the CID detection, the
number of rings the FXO interface
receives before going off-hook.
cid ring { 0 | 1 | 2 } [ times ]
By default, CID detection is

performed between the first and
second rings, and the FXO
interface goes off-hook as soon
as the detection is complete (cid
ring 1 0).
Enable CID receiving for the FXO

interface.
cid receive
By default, this function is

enabled.
Command
Remarks
4.
Enabling CID sending for an FXO interface

Step
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Enable CID sending for the FXO

interface.
cid send
By default, this function is

enabled.
Configuring busy tone detection

PBX switches might use different busy tone standards. If a user on the PBX side hangs up, the router knows
the on-hook operation by detecting a busy tone.
As shown in Figure 1, Telephone A establishes a call to Telephone B, and then Telephone A goes on-hook.
The PBX plays the busy tone to Router A after detecting the on-hook condition of Telephone A. If Router
A cannot identify the played busy tone, the FXO interface on Router A will remain seized or fault
detection occurs. To solve this problem, you can configure busy tone detection for the FXO interface.
Figure 1 Busy tone detection
You can configure busy tone detection by customizing busy tone parameters or configuring automatic
busy tone detection. If the tone that the router receives from the PBX matches the busy tone parameters,
the router considers the tone as a busy tone and shuts down the FXO interface.
Configuring a busy tone standard

Two busy tone standards are available: European standard and North American standard. After you
specify a standard, the device uses a set of parameters compliant with that standard for busy tone
detection. If the actual busy tone data do not completely match the set of parameters, the device can use
customized busy tone parameters or automatic busy tone detection to accurately detect busy tones.
To configure a busy tone standard:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Specify a busy tone standard.
area { europe |
north-america }
By default, the European

standard is used.
Customizing busy tone parameters

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Customize busy tone

parameters.
busytone-detect custom
area-number index argu f1 f2 p1
p2 p3 p4 p5 p6 p7
N/A
By default, the European standard
is used.
4.
Enable using customized busy

tone parameters.
area custom
This command takes effect on all

FXO interfaces on the device.
The customized busy tone
parameters take effect only after
you configure this command.
Configuring automatic busy tone detection

Figure 2 Automatic busy tone detection
As shown in Figure 2, the process of automatic busy tone detection is as follows:

1.
Telephone A establishes a call to Telephone B.

7
2.
Telephone A first goes on-hook. The PBX plays busy tones to Router A after detecting the on-hook
condition.
3.
Execute the busytone-detect auto command on Router A to detect the busy tone. To make sure the
FXO interface can capture the busy tone sent by the PBX, H3C recommends that you execute this
command two seconds after Telephone A goes on-hook.
4.
The console prompts that busy tone detection is in progress and prompts detection success when
the detection is complete.
5.
Check whether the detected busy tone parameters are valid by repeating steps 1 and 2.
After Telephone A goes on-hook, the PBX plays a busy tone to Router A. If Router A detects the busy
tone, it shuts down the FXO interface.
To configure automatic busy tone detection:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure automatic busy

tone detection.
busytone-detect auto index

line-number
N/A
4.
Return to system view.
quit
N/A
5.
N/A
6.
Configure the number of busy

tone detection periods.
The default is 2.
busytone-detect period value
Test multiple values to select the

value that can ensure normal
on-hook.
Configuring busy tone sending

If the PBX does not play busy tones, configure the FXO interface to send busy tones.
To configure busy tone sending:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Enable busy tone sending.
send-busytone enable
By default, this function is disabled.
4.
Set the busy tone duration.
send-busytone time seconds
The default is 3 seconds.
Configuring silence detection-based automatic on-hook

If the device fails to detect busy tones or the PBX does not play busy tones, you can configure this feature
to implement automatic on-hook.
When the duration of silence (whose volume is smaller than the configured threshold) exceeds the
configured silence duration, the FXO interface automatically disconnects the call.
Improper configuration of this function can lead to false on-hook. A good practice is to configure multiple
sets of parameters and select the set of parameters that can quickly release the FXO interface and does
not causes false on-hook.
8
To configure silence detection-based automatic on-hook:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure silence
detection-based automatic
on-hook.
silence-detect threshold threshold

time time-length
By default, the silence threshold is

20, and the silence duration f is
7200 seconds (2 hours).
Configuring forced on-hook

In some countries, PBXs do not play busy tones, or the busy tones only last for a short period of time.
When noise is present on a link, even the silence detection-based automatic on-hook function
(configured with silence-detect threshold) cannot detect the busy tones and fails to release a call after
on-hook. To solve this problem, configure the forced on-hook function. Forced on-hook disconnects a call
when the specified time expires, even if the call is ongoing.
To configure forced on-hook:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
By default, forced on-hook is disabled.
3.
Configure forced on-hook.
hookoff-time time
If you configure this command on an FXO

interface of a card, the configuration
takes effect on all FXO subscriber lines of
the card.
Configuring an on-hook delay

An FXO interface goes on-hook when detecting a busy tone. This will cause the user of an IP phone
connected to the FXO interface to mistake the on-hook as a line problem because the user cannot hear
the busy tones.
To solve this problem, you can configure a delay time. During the delay time, the FXO interface continues
sending the busy tones to the IP phone.
To configure an on-hook delay:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure the delay time from

when the FXO interface detects
a busy tone to when the
interface goes on-hook.
busytone-hookon delay-timer
value
The delay time is 0 seconds (the

FXO interface goes on-hook
immediately after detecting a busy
tone).
Configuring the off-hook mode

The FXO interface supports two off-hook modes:
Immediate modeUpon receiving a call, the FXO interface goes off-hook and sends a dial tone to
the calling party. Then, the calling party dials the destination number.
Delay modeUpon receiving a call, the FXO interface places a call to the specified private line
number. When the called party picks up the phone, the FXO goes off-hook. This mode needs to
work with the private line auto ring-down (PLAR) function. For more information about PLAR, see
"Configuring dial programs."
To configure the off-hook mode:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure the off-hook mode.
hookoff-mode { delay |
immediate }
By default, the immediate mode is

used.
Configuring ring detection parameters

PBXs from different vendors use different types of ring signals. By setting ring detection parameters, you
can enable the FXO interface to detect ring signals of different frequencies and waveforms.
To configure ring detection parameters:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
Do not set the debounce time

during a conversation.
Do not set too short a debounce

3.
Set the debounce time for ring

detection on the FXO
interface.
ring-detect debounce value
time, because false ring tone

recognition might occur under
power line interference.
If you configure this command
on an FXO interface of a card,

the configuration takes effect on
all FXO subscriber lines of the
card.
4.
Set the frequency value for

ring detection.
ring-detect frequency value
10
The default is 40 Hz.

The electrical impedance setting must be subject to country specifications. Each country corresponds to
an impedance value. You specify an impedance value while specifying a country.
To set the electrical impedance:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
Set the electrical impedance

of a country.
3.
impedance { country-name | r550

| r600 | r650 | r700 | r750 |
r800 | r850 | r900 | r950 }
The default is the electrical

impedance of China.
You must configure the same
electrical impedance value on the
devices.

You can configure the packet loss compensation mode to alleviate the impact on voice quality:
For discrete packet loss, you can use the general compensation mode to reconstruct lost packets.
For continuous packet loss, you can use the voice gateway-specific compensation mode to
reconstruct lost packets.
To configure the packet loss compensation mode:

Step
Command
Remarks
1.
Enter system view.
System-view
N/A
2.
N/A
3.
Configure the packet loss

compensation mode for the
FXO interface.
plc-mode { general | specific }
By default, the voice

gateway-specific compensation
mode is used for an FXO interface.
Binding an FXS interface to an FXO interface

The one-to-one binding between FXS and FXO interfaces enhances the reliability of voice
communication. By working with the PLAR function, this feature enables the telephone connected to the
FXS interface to exclusively use the bound FXO interface to place and receive calls.
The on-hook/off-hook state of the bound FXS and FXO interfaces is consistent. If the FXS interface
receives a call when the bound FXS interface goes off-hook, the calling party will hear busy tones.
To bind an FXS interface to an FXO interface:
Step
1.
Enter system view.
Command
Remarks
system-view
N/A
11
Step
Command
Remarks
2.
N/A
3.
Bind an FXS interface to the

FXO interface.
hookoff-mode delay bind

fxs_subscriber_line [ ring-immediately ]
By default, no FXS interface is

bound.
By default, the PLAR function is
disabled.
4.
Enable the PLAR function.
private-line string
For more information about this

command, see Voice Command
Reference.
5.
Configure the interval

between on-hook and
off-hook.
timer hookoff-interval milliseconds
Configuring an E&M interface

Configuring the cable type
You must configure the same cable type for the E&M interfaces on the originating and terminating
devices. Otherwise, only one-way voice communication can be implemented.
To configure the cable type for an E&M interface:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E&M interface view.
N/A
3.
Configure the cable type for

the E&M interface.
cable { 2-wire | 4-wire }
The default is 4-wire.
Configuring the signal type

You must configure the same signal type on the originating and terminating devices.
To configure the signal type for an E&M interface:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure a signal type for the

E&M interface.
type { 1 | 2 | 3 | 5 }
The default is 5 (corresponding V

type signal).
Configuring a start mode for E&M signaling

Three start modes are available for E&M signaling:
12
Immediate startAfter off-hook, the originating side waits for a specified period of time to send the
called number to the terminating side. During this period of time, the originating side does not
check whether the terminating side is ready to receive the called number. The terminating side
enters off-hook state after receiving the called number.
Figure 3 Immediate start
Delay startThe originating side goes off-hook and seizes the trunk. After detecting the seizure
signal from the originating side, the terminating side enters the off-hook state and remains in this
state until it is ready to receive the called number. Then, the terminating side enters the on-hook state
and sends a signal to indicate that the line is idle. After receiving the idle signal, the originating
side sends the called number to the terminating side, which relays the call to the user phone.
Figure 4 Delay start
Wink startThe terminating side remains in on-hook state until it receives the seizure signal from
the originating side and then sends a wink to the originating side. After receiving the wink, the
originating side sends the called number to the terminating side, which relays the call to the user
phone.
13
Figure 5 Wink start
To configure the immediate start mode:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure the immediate start

mode for the E&M interface.
signal immediate
The default is the immediate start

mode.
4.
Configure a delay the

originating side waits to send
DTMF tones in immediate start
mode.
delay send-dtmf milliseconds

Use this command on the
originating device.
To configure the delay start mode:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure the delay start mode for the

E&M interface.
signal delay
The default is the immediate

start mode.
4.
Configure the seizure signal duration in

delay start mode.
delay hold milliseconds
5.
Configure the delay time from when the

terminating side detects a seizure
signal to when it sends the seizure
signal in delay start mode.
delay rising milliseconds

terminating device.
terminating device.
To configure the wink start mode:

Step
1.
Enter system view.
Command
Remarks
system-view
N/A
14
Step
Command
Remarks
2.
N/A
3.
Configure the wink start mode for

the E&M interface.
signal wink

mode.
4.
Configure the delay time from when

the terminating side receives a
seizure signal to when it sends a
wink signal in the wink start mode.
delay send-wink milliseconds
5.
6.
Configure the duration of a wink

signal sent by the terminating side
in wink start mode.
Configure the timeout time for the
originating side to wait for a wink
signal after sending a seizure
signal in the wink start mode.

terminating device.
delay wink-hold milliseconds
delay wink-rising milliseconds

terminating device.
The default is 2000
milliseconds.
originating device.
Configuring E&M non-signaling mode

The E&M non-signaling mode is applied when the E&M interface of the peer device does not provide the
M line and E line. In this mode, the E&M interface communicates with the peer device without signaling.
You can configure the PLAR function by using the private-line command to form a three-segment E&M
virtual private line (E&M-VoIP-E&M). When a subscriber picks up the phone, the originating device
directly dials the number specified by using the private-line command.
To configure E&M non-signaling mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Configure the immediate start

mode for the E&M interface.
signal immediate

mode.
By default, E&M non-signaling
mode is disabled.
4.
Enable E&M non-signaling mode.
open-trunk { caller [ monitor

interval ] | called }
Configure the open-trunk caller

[ monitor interval ] command on
the originating device, and
configure the open-trunk called
command on the terminating
device.
originating device.
5.
Enable the PLAR function.
private-line string
15
For more information about the

PLAR function, see "Configuring
dial programs."
Enabling E&M control signals pass-through

This feature operates only when the E&M non-signaling mode is enabled. As shown in Figure 6, an E&M
virtual private line is set up between the tone generator and the radio. Enable E&M control signals
pass-through for Router A and Router B so that they can send seize and idle signals for the E&M virtual
line over the IP network.
Figure 6 E&M analog control signals pass-through
To enable E&M control signals pass-through:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Enable E&M control signals

pass-through.
By default, this function is disabled.

passthrough
Configure this command on both

the originating and terminating
devices.
Configuring the output gain of SLIC chip

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
Optional.
3.
Configure the output gain of

the SLIC chip.
slic-gain { 0 | 1 }
The default is 0 (0.8 dB).

This command controls signal
amplification.
Configuring DTMF
Dual tone multi-frequency (DTMF) uses a mixture of a high frequency tone and a lower frequency tone to
represent a key on a keypad. Each column of keys is represented by a high frequency tone and each row
of keys is represented by a low frequency tone. For example, as shown in Figure 7, the digit 1 is
represented by the combination of a pure 697 Hz signal and a pure 1209 Hz signal. Such DTMF tones
have good immunity to interference.
16
Figure 7 DTMF keypad frequencies

Column Frequency Group
1209Hz
1336Hz
1477Hz
1633Hz
697Hz
770Hz
852Hz
941Hz
A DTMF tone must last at least 45 milliseconds. A minimum interval of 23 milliseconds is required
between two DTMF tones to make sure DTMF tones are recognizable. Such requirements are roughly the
same in all countries. For more information, see the ITU-T Recommendation Q.24.
Configuring DTMF tone sending

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Configure the duration of

DTMF tones and the interval
between DTMF tones.
dtmf time {interval | persist }

milliseconds
The default duration and interval

are both 120 milliseconds.
4.
Configure the amplitude of

DTMF tones.
dtmf amplitude value
The default is 9.0 dBm.
Configuring DTMF tone detection

Use the following ways to detect DTMF tones:
Energy detectionCalculates the spectrum shape of input DTMF signals and matches the spectrum
shape with the threshold parameters. A DTMF tone is considered valid if it matches all threshold
parameters.
Sensitivity detectionA higher DTMF detection sensitivity reduces the possibility of missing a true
DTMF tone but increases the possibility of false detection. A lower DTMF detection sensitivity
reduces the possibility of false detection but increases the possibility of missing a true DTMF tone.
To configure DTMF tone detection:

17
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
4.
Configure the threshold

parameters for DTMF
detection.
dtmf threshold analog index

value
Configure the DTMF

detection sensitivity level.
dtmf sensitivity-level { high |

low | medium
[ frequency-tolerance value ] }
By default, indexes 0 to 12 correspond to

1400, 458, -9, -9, -9, -9, -3, -12, -12, 30,
300, 3200, and 375, respectively.
This command is used by professional
personnel to adjust the device in the case
of DTMF detection failure. Usually, the
default value is adopted.
By default, the DTMF detection sensitivity
is low.
This command applies to only FXS and
FXO interfaces
Adjusting parameters for voice interfaces

The configuration tasks in this section are all optional.
Adjusting gains
You can adjust gains to control the amount of volume in the input or output direction.
IMPORTANT:
Gain adjustment might lead to call failures. H3C recommends not adjusting the gain. If necessary, do it
under the guidance of technical personnel.
To adjust gains:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Set the input gain.
receive gain value
The default is 0 dB.
4.
Set the output gain.
transmit gain value
The default is 0 dB.
Adjusting timing parameters

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
18
Step
3.
4.
5.
Command
Remarks
Configure the interval

between off-hook and dialing
the first digit.
timer first-dial seconds
This command applies only to FXO

and FXS interfaces.
Configure the maximum

interval for dialing the next
digit.
timer dial-interval seconds

duration for playing ringback
tones.
timer ring-back seconds

The default is 1 second.
6.
7.
Configure the dial delay time.
Configure the hookflash time

range.
delay start-dial seconds
timer hookflash-detect
hookflash-range
This command applies to only FXO

and FXS interfaces.
By default, the hookflash time
range is 50 to 180 milliseconds;
that is, if an on-hook condition lasts
for a period that falls within the
hookflash time range, it is
considered a hookflash.
This command applies to only FXS
interfaces.
8.

duration the terminating
device waits for the first digit.
timer wait-digit { seconds |

infinity }

This command applies to only E&M
interfaces.
Configuring the comfortable noise function

You can configure this feature to generate comfortable background noise to replace the silent gaps
during a conversation.
To configure the comfortable noise function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Enable the comfortable noise

function.
cng-on
Configuring echo cancellation

An echo is the audible leak-through of your own voice into your own receive path. When the voice of a
user leaks into the receive path, it is an echo. The echo cancellation function can alleviate the echo
problem.
The time when an echo occurs and the size of the echo are relatively fixed. As shown in Figure 8, there
is a voice signal at time 0, and an echo occurs after about 40 milliseconds (ms). To cancel this echo, set
19
the echo cancellation delay (the time between when an interface sends out a signal and when to the
interface receives an echo) to 33 ms and the echo cancellation coverage to 16 ms.
Figure 8 Echo
Configuring the echo cancellation function

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Enable the echo cancellation

function.
echo-canceler enable
4.
Configure the echo

cancellation delay.
echo-canceler delay milliseconds
5.
Configure the echo

cancellation coverage.
echo-canceler tail-length
milliseconds
Adjusting echo cancellation parameters

Table 1 describes how to adjust echo cancellation parameters for different symptoms.
Table 1 Adjusting echo cancellation parameters
Symptom
Parameters adjusted
Effect
A user hears echoes or loud

background noises from the peer
when speaking.
Speed up the convergence of

comfortable noise amplitudes.
Too fast convergence might make

noises uncomfortable.
There are loud environment noises.
Increase the maximum amplitude

of comfortable noises.
Too large amplitude might make

noises uncomfortable.
A user hears echoes when

speaking.
Enlarge the mixture proportion

control factor of comfortable
noises.
Too high a control factor leads to

audio discontinuity.
There are echoes when both

parties speak at the same time.
Enlarge the two-way judgment

threshold.
Too high a judgment threshold

slows down the convergence of the
filter factor.
To adjust echo cancellation parameters:

Step
1.
Enter system view.
Command
Remarks
system-view
N/A
20
Step
2.
3.
Enter voice view.
Adjust echo cancellation

parameters.
Command
Remarks
voice-setup
N/A
echo-canceler { convergence-rate
value | max-amplitude value |
mix-proportion-ratio value |
talk-threshold value }
By default, the convergence rate of

comfort noise amplitude is 0, the
maximum amplitude of comfort
noise is 256, the comfort noise
mixture proportion control factor is
100, and the two-way judgment
threshold is 1.
The echo-canceler enable
command must be configured for
the configured parameters to take
effect.
Enabling the nonlinear function of echo cancellation

After echo cancellation is enabled, nonlinear parts in the line can cause residual echo. The nonlinear
function, also called residual echo suppression, can remove the residual echo.
To enable the nonlinear function of echo cancellation:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.
Enable the nonlinear function

of echo cancellation.

This command is available only
after you configure the
echo-canceler enable command.
nlp-on
Displaying and maintaining analog voice

interfaces
Execute display commands in any view.
Task
Command
Display information about analog voice

interfaces.
display voice subscriber-line
Analog voice interface configuration examples

Two-dial configuration example for the FXO interface
As shown in Figure 9, Router A and Router B are connected through an IP network and can reach each
other. The PBX is configured with a trunk number 07552003.
21
Configure the two routers to enable Telephone B to establish a call with Telephone A through two dials.
Figure 9 Network diagram
Eth2/1
1.1.1.1/24
FXS 1/0
Telephone A
0101001
IP network
Eth2/1
2.2.2.2/24
FXO 1/0
Router B
Router A
PBX
Telephone B
07552001
1.
On Router A, configure the local number as 0101001 for POTS entity 1001, and bind FXS
interface line1/0 to the POTS entity.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/0
2.
Configure Router B:
# Configure the called number as 010 for VoIP entity 010, and configure the destination IP
address as 1.1.1.1.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 0101001.
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Configure the local number as 07552001 for POTS entity 2001, and bind FXO interface
line1/0 to POTS entity 2001.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/0
# Configure the number sending mode as all.

[RouterB-voice-dial-entity2001] send-number all
Verifying the configuration

After the user of Telephone B dials 07552003, the user hears a dial tone. Then, the user of Telephone B
dials 0101001, and Telephone A rings. After the user of Telephone A picks up the handset, the two users
can establish a conversation.
FXO interface PLAR configuration example

Network requirements
other. The PBX is configured with a trunk number 07552003.
22
Configure PLAR for the FXO interface of Router B. When the user of Telephone B dials 07552003, the
FXO interface automatically calls Telephone A.
Eth2/1
1.1.1.1/24
FXS 1/0
Telephone A
0101001
IP network
Eth2/1
2.2.2.2/24
FXO 1/0
Router B
Router A
PBX
Telephone B
07552001
1.
On Router A, configure the local number as 0101001 for POTS entity 1001, and bind FXS
interface line1/0 to the POTS entity.
2.
Configure Router B:
address as 1.1.1.1.
[RouterB-voice-dial-entity10] match-template 0101001.
# Configure the local number as 07552001 for POTS entity 2001, and bind FXO interface
line1/0 to the POTS entity.
# Configure the number sending mode as all.

# Enable the PLAR function and configure the delay off-hook mode for the FXO interface.
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] private-line 0101001
[RouterB-subscriber-line1/0] hookoff-mode delay

After the user of Telephone B dials 07552003, the FXO interface automatically calls Telephone A, and
Telephone A rings. After the user of Telephone A picks up the handset, the two users can establish a
conversation.
23
E&M interface configuration example

other.
Configure the two routers to enable Telephone A and Telephone B to establish calls.
Eth2/1
1.1.1.1/24
FXS 1/0
Telephone A
0101001
IP network
Eth2/1
2.2.2.2/24
Router A
E&M 5/0
Router B
PBX
Telephone B
1.
Configure Router A:
address as 2.2.2.2.
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
[RouterA-voice-dial-entity755] quit
# Configure the local number as 0101001 for POTS entity 1001, and bind FXS interface line1/0
to the POTS entity.
2.
Configure Router B:
address as 1.1.1.1.
# Configure the local number as 07552001 for POTS entity 2001, and bind E&M interface line
5/0 to the POTS entity.
[RouterB-voice-dial-entity2001] return
24
# Enter the view of E&M interface 5/0. The E&M interface must have the same configuration as the
connected PBX.
# Configure the wink start mode.

[RouterB-subscriber-line5/0] signal wink
# Configure the 4-wire cable type (optional, because the default is the 4-wire cable type).
[RouterB-subscriber-line5/0] cable 4-wire
# Configure the signal type as 5 (optional, because the default is 5).

[RouterB-subscriber-line5/0] type 5

After the user of Telephone A dials 07552001, Telephone B rings. When the user of Telephone B picks
up the handset, the two users can establish a conversation.
After the user of Telephone B dials 0101001, Telephone A rings. When the user of Telephone A picks up
the handset, the two users can establish a conversation.
E&M non-signaling mode configuration example

As shown in Figure 12, configure the PLAR mode for the E&M interface on Router A. When the user of
Telephone A dials picks up the handset, the E&M interface automatically calls Telephone B.
1.
Configure Router A:
address as 2.2.2.2.
# Configure the local number as 1000 for POTS entity 1000, and bind E&M interface line 5/0 to
the POTS entity.
25

[RouterA-voice-dial-entity1000] return
# Enter the view of E&M interface 5/0.

[RouterA] subscriber-line 5/0
# Configure the immediate start mode (optional, because the default is the immediate start mode).
[RouterA-subscriber-line5/0] signal immediate
# Enable the PLAR function.

[RouterA-subscriber-line5/0] private-line 2000
# Enable the E&M non-signaling mode.

[RouterA-subscriber-line5/0] open-trunk caller
# Enable E&M analog control signals pass-through.

[RouterA-subscriber-line5/0] passthrough
# Disable the echo cancellation function.

[RouterA-subscriber-line5/0] undo echo-canceler enable
2.
Configure Router B:
address as 1.1.1.1.
# Configure the local number as 2000 for POTS entity 2000, and bind E&M interface line 5/0 to
the POTS entity.
[RouterB-voice-dial-entity2000] return
# Enter the view of E&M interface 5/0.

# Configure the immediate start mode (optional, because the default is the immediate start mode).
[RouterB-subscriber-line5/0] signal immediate
# Enable the E&M non-signaling mode.

[RouterB-subscriber-line5/0] open-trunk called
# Enable E&M control signals pass-through.

[RouterB-subscriber-line5/0] passthrough
# Disable the echo cancellation function.

[RouterB-subscriber-line5/0] undo echo-canceler enable
26

After the user of Telephone A picks up the handset, Router A automatically calls Telephone B. After the
user of Telephone B picks up the handset, the two users can establish a conversation.
FXS&FXO 1:1 binding configuration example

As shown in Figure 12, Telephone A calls Telephone B through the IP network. When the IP network is
unavailable, Telephone A uses the bound FXO interface to call Telephone B through the PSTN network.
192.168.0.71/24
FXS 1/0
Telephone A
0101001
IP
192.168.0.76/24
Router B
Router A
FXS 1/0
Telephone B
2101002
FXO 2/0
PSTN
FXO 2/0
1.
Configure Router A:
# Configure the called number template as 210. for VoIP entity 210, and configure the
destination IP address as 192.168.0.76.
[RouterA-voice-dial-entity210] match-template 210....
# Configure the local number as 0101001 for POTS entity 0101001, and bind the FXS interface
# Configure the called number template as .T for POTS entity 211 used for call backup on FXO
interface line 2/0, and configure the permitted calling number as 0101001.
[RouterA-voice-dial-entity211] match-template .T
[RouterA-voice-dial-entity211] send-number all
[RouterA-voice-dial-entity211] caller-permit 0101001
27
[RouterA-voice-dial] quit
[RouterA-voice] quit
# Enable the PLAR function for the FXO interface line 2/0, and bind the FXS interface line 1/0 to
the FXO interface line 2/0.
[RouterA] subscriber-line 2/0
[RouterA-subscriber-line2/0] private-line 0101001
[RouterA-subscriber-line2/0] hookoff-mode delay bind 1/0
2.
Configure Router B:
# Configure the called number template as 010. for VoIP entity 010, and configure the
[RouterB-voice-dial-entity10] match-template 010....
# Configure the local number as 2101002 for POTS entity 2101002, and bind the FXS interface
# Configure the called number template as .T for POTS entity 011 used for call backup on FXO
interface line 2/0, and configure the permitted calling number as 2101002.
[RouterB-voice-dial-entity11] match-template .T
[RouterB-voice-dial-entity11] caller-permit 2101002
[RouterB-voice-dial] quit
[RouterB-voice] quit
# Enable the PLAR function for the FXO interface line2/0, and bind the FXS interface line1/0 to the
FXO interface line2/0.
[RouterB-subscriber-line2/0] private-line 2101002
[RouterB-subscriber-line2/0] hookoff-mode delay bind 1/0
When the IP network operates correctly, Telephone A calls Telephone B through the IP network.
When the IP network is unavailable, Telephone A uses the bound FXO interface to call Telephone
B through the PSTN network.
28
Configuring digital voice interfaces

Digital voice interfaces include E1, T1, and BSV interfaces.
E1 and T1 interfaces
E1 interfaces (also called VE1 interfaces) and T1 interfaces (also called VT1 interfaces) can connect to
the PSTN as trunk interfaces. An E1 interface provides 32 timeslots and 2.048 Mbps bandwidth. A T1
interface provides 24 timeslots and 1.544 Mbps bandwidth. ITU-T E1 is used mainly in Europe and
China. ANSI T1 is used mainly in America, Canada, and Japan.
E1/T1 voice transmission allows a router to provide more communication channels, greatly improving
router utilization.
Figure 14 shows a typical network for E1/T1 interfaces.
BSV interfaces
BRI S/T voice (BSV) interfaces can compress, send, receive, and decompress voice packets. BSV
interfaces connect to PBXs as trunk interfaces. A BSV interface provides two B channels and one D
channel (2B+D) and supports only DSS1 signaling.

To configure E1 and T1 interfaces, perform the following tasks:
Tasks at a glance
(Required.) Perform one of the following tasks:
Configuring basic parameters for an E1 interface

{
Configuring a TDM clock source
Configuring other parameters
Configuring basic parameters for a T1 interface

{
(Required.) Configuring a timeslot set
Creating a timeslot set

Configuring a logical digital voice interface
29
Tasks at a glance
(Required.) Perform one of the following tasks:
Configuring R2 signaling
{
Configuring basic R2 signaling parameters
Configuring R2 digital line signaling
Configuring R2 interregister signaling
Configuring the ISDN protocol

(Required.) Binding a digital voice interface to a POTS entity
To configure BSV interfaces, perform the following tasks:

Tasks at a glance
(Required.) Configuring a BSV interface
Configuring a BRI interface

Configuring a digital voice interface
(Required.) Configuring the ISDN protocol
(Required.) Binding a digital voice interface to a POTS entity
Configuring basic parameters for an E1 interface

This section describes basic E1 interface settings.

E1 interfaces must use a common TDM clock source during timeslot interchange to prevent frame slips
and bit errors.
When both E1 and T1 cards are present on a device, all the E1 or T1 SIC cards are a subsystem, and
each E1 or T1 HMIM card is a subsystem. Each subsystem determines the clock source according to the
following rules:
If the line keyword is specified for all interfaces, the clock on the interface with the lowest number
is used. If the interface goes down, the clock on the interface with the second lowest number is used.
If the line and primary keywords are specified for one interface and the line or internal keyword is
specified for all other interfaces, the clock on that one interface is used.
If the line keyword is specified for one interface and the internal keyword is specified for all other
interfaces, the clock on that one interface is used.
The clock source of only one interface can be set to line primary.
The TDM clock sources on the local and peer devices must match. For example, if you set the clock source
to line for a subsystem on the local device, you must set the clock source to internal on the peer device;
and vice versa.
To configure a TDM clock source for an E1 interface:
30
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 number
N/A
3.
Configure a TDM clock source

for the E1 interface.
tdm-clock { internal | line

[ primary ] }
The default is internal.

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 interface view.
controller e1 number
N/A
3.
Configure a description.
description text
The default is interface name

Interface.
4.
Configure the framing format.
frame-format { crc4 | no-crc4 }
The default is no-crc4.
5.
Set the line coding format.
code { ami | hdb3 }
The default is hdb3.
6.
Set the cable type.
cable { long | short }
The default is long.
7.
Configure the link idle code

type.
idle-code { 7e | ff }
The default is 0x7E.
8.
Set the interframe filling tag

type.
itf type { 7e | ff }
9.
(Optional.) Enable loopback

and set the loopback mode.
loopback { local | payload |

remote }
By default, loopback is disabled.
10. Restore the default settings for

the E1 interface.
default
N/A
11. Shut down the E1 interface.
undo shutdown
By default, an E1 interface is up.
Configuring basic parameters for a T1 interface

This section describes the basic T1 interface settings.

A T1 interface has the same TDM clock source configuration as an E1 interface. For more information,
see "Configuring a TDM clock source."
To configure a TDM clock source:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller t1 number
N/A
31
Step
Configure a TDM clock source
for the T1 interface.
3.
Command
Remarks
tdm-clock { internal | line

[ primary ] }
The default is internal.

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter T1 interface view.
controller t1 number
N/A
3.
Configure a description.
description text

Interface.
4.
Configure the framing format.
frame-format { esf | sf }
The default is esf.
5.
Set the line coding format.
code { ami | b8zs }
The default is b8zs.
6.
Set the cable type.
cable { long | 0db | -7.5db |

-15db | -22.5db | short | 133ft |
266ft | 399ft | 533ft | 655ft }
The default is long 0db.
7.
Set the interframe filling tag

type.
itf type { 7e | ff }

By default:
For LOS alarms, the threshold of
8.
Set alarm thresholds.
alarm-threshold { ais { level-1 |

level-2 } | lfa { level-1 | level-2 |
level-3 | level-4 } | los
{ pulse-detection | pulse-recovery }
value }
pulse-detection is 176 and the

threshold of pulse-recovery is
22. If the number of pulses
detected during the total length
of 176 pulse detection intervals
is smaller than 22 (the
pulse-recovery threshold), a
LOS alarm occurs.
Both AIS alarm threshold and
LFA alarm threshold are level-1.
9.
Enable loopback and set the

loopback mode.
loopback { local | payload |

remote }
By default, loopback is disabled.
10. Restore the default settings for

the T1 interface.
default
N/A
11. Bring up the T1 interface.
undo shutdown
By default, a T1 interface is up.
Configuring a BSV interface

A BSV interface can generate basic rate interfaces (BRIs) and digital voice interfaces.
BRIHas the same logical characteristics as an ISDN BRI but can be used for only voice
transmission. You can configure ISDN parameters on a BRI interface.
Digital voice interfaceCan be configured with voice commands. A digital voice interface has two
subinterfaces, which correspond to the two B channels. You cannot configure commands on the
32
subinterfaces, but you can use the display voice subscriber-line line-number.subnumber command
to view the call state of a subinterface.
Configuring a BRI interface

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter BRI interface view.
interface bri interface-number
N/A
3.
(Optional.) Configure a
description for the interface.
description text

Interface.
4.
(Optional.) Enable loopback

detection for B channels.
loopback { b1 | b2 | both }
By default, loopback detection is

disabled.
5.

expected bandwidth of the
interface.
bandwidth bandwidth-value
The default is 0 kbps.
6.
Set the MTU of the interface.
mtu size
The default is 1500 bytes.
7.
Set the polling interval for the

interface.
timer-hold seconds
8.
(Optional.) Restore the default

settings for the interface.
default
N/A
9.
Bring up the interface.
undo shutdown
By default, a BRI interface is up.
activate
By default, a BRI interface is not

activated.
10. (Optional.) Activate the

interface.
For more information about these commands, see Interface Command Reference.
Configuring a digital voice interface

The description, shutdown, receive gain, transmit gain, and cng-on commands have the same functions
as those for analog voice interfaces. For more information about those commands, see related sections
in "Configuring analog voice interfaces."
Configuring a timeslot set

Creating a timeslot set
To use an E1 interface, you need to create a timeslot set and configure the signaling type, Then, the
system automatically creates a logical digital voice interface. Timeslot 0 is used for frame synchronization,
timeslot 16 is used to transmit signaling, and the other 30 timeslots are used to transmit voice packets.
A T1 interface is used in a similar way to an E1 interface, except that it has 24 timeslots and timeslot 24
is used to transmit signaling.
Only RS signaling can be configured for a timeslot set.
To create a timeslot set:
33
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 or T1 interface
view.
controller { e1 | t1 } number
N/A
3.
Create a timeslot set and

enable R2 signaling for it.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
By default, no timeslot set is created.
Configuring a logical digital voice interface

The system automatically creates a logical voice interface in the form of E1/T1 interface number:timeslot
set number for a timeslot set.
The description, shutdown, receive gain, transmit gain, and cng-on commands for a logical digital voice
interface have the same functions as those for analog voice interfaces. For more information about those
commands, see related sections in "Configuring analog voice interfaces."
To configure a logical digital voice interface:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter logical digital voice

interface view.
subscriber-line
number:ts-set-number
N/A
pcm { a-law | -law }
The default is a-law for logical

digital voice interfaces of E1
interfaces and -law for logical
digital voice interfaces of T1
interfaces.
3.
Configure a companding law

for PCM.
Configuring R2 signaling
ITU-T recommendations Q.400 through Q.490 define the R2 signaling standards. However, the R2
signaling standards implemented in different countries and regions are ITU variants.
In R2 signaling, the calling side serves as the originating PBX, and the called side serves as the
terminating PBX. Signaling sent by the originating PBX is called forward signaling, and signaling sent by
the terminating PBX is called backward signaling, as shown in Figure 15.
Figure 15 R2 signaling elements
34
R2 signaling include two categories: digital line signaling and interregister signaling. Digital line
signaling conveys status information about E1 trunks to describe whether the trunks are seized, released,
or blocked. Interregister signaling transmits and requests calling and called numbers.
ITU-T digital line signaling

Digital line signaling monitors the state of a trunk and controls calls. It can identify the following state
changes:
The calling party goes off-hook and seizes the line.
The called party goes off-hook and answers the call.
The calling party releases the call.
The called party releases the call.
Digital line signaling sets the line to be idle or seized according to the state of the trunk. This signaling
is transmitted through timeslot 16. The two transmission directions of each line have four bits (A, B, C and
D) as flag bits, with C and D bits fixed to 01. The forward line signaling adopts af and bf bits, and the
backward line signaling adopts ab and bb bits, as shown in Table 2:
Table 2 Line signaling bit description
Bit
Description
Value = 0
Value = 1
af
Identifies the state of the originating device and

indicates the state of the calling line.
Off-hook, seized
On-hook (idle)
bf
Indicates a fault state from the originating side to

the terminating side.
Normal
Faulty
ab
Indicates the state of the called line (on-hook or

off-hook).
Off-hook by called
party
On-hook by called party
bb
Indicates the state of the terminating device (idle

or seized).
Idle
Seized or blocked
Table 3 State code of line signaling

Signaling code
State of the circuit
Forward
Backward
af
bf
ab
bb
Idle or release
Seized
Seizure-ack
Answer
Clear-back
Clear-forward
0/1
Blocked
Unblocked
R2 digital line signaling exchange includes the following processes:
Call establishment.
35
When the circuit is idle, the originating side sends a forward seizure signal (00) to the terminating
side, and the terminating side sends back a seizure acknowledgement signal (11). Then the circuit
is seized, and interregister signaling exchange begins. When the called party picks up the phone,
the terminating side sends a backward answer signal (01). After the originating side recognizes
the received signal, it establishes the call.
Figure 16 Call establishment
Call release at the originating side.

The originating side sends a clear-forward signal 10. When the terminating side recognizes the
clear-forward signal, it sends a backward signal 10. When the originating side recognizes the
backward signal 10, it releases the call.
Figure 17 Call release at originating side
Call release at the terminating side.

The terminating side sends a clear-back signal 11 to the originating side, and the originating side
sends back a clear-forward signal 10. When the terminating side recognizes the forward signal
10, it sends a release-guard signal 10. When the originating side recognizes the release-guard
signal 10, it releases the call.
Figure 18 Call release at terminating side
36
Blocking in idle state or during conversation.
After the originating side receives a blocking signal 11 from the terminating side when the circuit
is idle or during conversation, the circuit is blocked. In this case, the originating side still sends the
forward signal 10 to indicate that the line is idle. When the terminating side unblocks the circuit,
it sends a backward signal 10 to indicate that the line is idle. The originating side maintains the
forward signal 10 and unblocks the local-end circuit for the next call.
ITU-T interregister signaling

Interregister signaling transmits and requests calling and called numbers. It adopts the multi-frequency
compelled (MFC) mode and includes forward signaling and backward signaling. Forward signaling
exchange includes Group I and Group II, and backward signaling exchange includes Group A and
Group B. When the originating side recognizes the seizure acknowledgement signal, interregister
signaling begins to send the first digit of the called number, and waits for the response of Group A
signaling from the terminating side.
Group I forward signalsInclude connection control signals and digit signals.
Table 4 Forward Group I signals

Designation
Basic meaning
I-1 through I-10
Digits 1, 2, 3, 4, 5, 6, 7, 8, 9, and 0, responsible for sending number

information to the terminating side
I-11
Spare for national use
I-12
Request refused
I-13
Connected to tested device
I-14
I-15
Address identification terminator and pulse terminator (used in

international calls)
Group A backward signalsControl signals used for controlling and acknowledging Group I
forward signals.
Table 5 Group A backward signals

Designation
Basic Meaning
A-1
Send next digit
A-2
Send last but one digit
A-3
Address-complete; changeover to reception of Group B signals
A-4
Congestion in the national network
A-5
Send calling party's category
A-6
Address-complete, charge, and set up speech conditions
A-7
Send last but two digits
A-8
Send last but three digits
A-9
A-10
A-11
Send country code indicator
37
Designation
Basic Meaning
A-12
Send language or discrimination digit
A-13
Send nature of circuit
A-14
Request for information on use of an echo canceller (is an incoming half-echo

suppressor required?)
A-15
Congestion in an international exchange; terminate interregister signaling

interaction or at its output
Group II forward signalsIdentify the calling party category. The system looks at the calling party
category to decide whether the calling party can perform forced release or break-in.
Table 6 Group II forward signals

Designation
Basic Meaning
II-1
Subscriber without priority
II-2
Subscriber with priority
II-3
Maintenance equipment
II-4
II-5
Operator
II-6
Data transmission
II-7
Subscriber (or operator without forward transfer facility), for international use
II-8
Data transmission (for international use)
II-9
Subscriber with priority (for international use)
II-10
Operator with forward transfer facility (for international aid use)
II-11 through II-15
Group B backward signalsAcknowledge Group II signals and indicate the status of the called
party.
Table 7 Group B backward signals

Designation
Basic Meaning
B-1
B-2
Send special information tone
B-3
Subscriber line busy
B-4
Congestion
B-5
Unallocated number
B-6
Subscriber line free, charge
B-7
Subscriber line free, no charge
B-8
Subscriber line out of order
B-9 through B-15
38
Figure 19 shows the exchange process requesting calling party information, which is typical of R2
interregister signaling.
Figure 19 ITU-T R2 interregister signaling exchange process
Originating side
Calling number: 123
Terminating side
Called number 789
Line signaling exchange

Send called number digit 7 (I-7)
Request next digit (A-1)
Request calling party information (A-5)
Send calling accounting category 2 (II-7)
Send calling number digit 1 (I-1)
Send number terminator (I-15)
Request next digit (A-1)
All digits are received completely (A-3)
Send calling service category 2 (II-2)
Subscriber idle and charging starts (B-6)
Line signaling exchange
Configuring basic R2 signaling parameters

Configuring the R2 signaling standard
For the device to perform R2 signaling exchange with devices in a specific country or region, you need
to specify the R2 signaling standard of that country or region.
To configure the R2 signaling standard:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter E1 or T1 interface view.
N/A
3.

By default, no timeslot set is

created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Specify the R2 signaling

standard.
mode zone-name
[ default-standard ]
By default, the ITU-T standard is

adopted.
39
Configuring the trunk direction and routing mode

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.


created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
The default is dual.
5.
Configure the trunk direction.
trunk-direction timeslots
timeslots-list { dual | in | out }
6.
Configure the routing mode.
select-mode { max | maxpoll | min

| minpoll }
You cannot configure the in or out

direction on both ends of a trunk. If
you configure the dual direction on
both ends, you need to configure
different routing modes on the two
ends by using the select-mode
command to prevent timeslot
contention.
The default is min.
Configuring the terminating side to send busy tones to the originating side
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.


created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
6.
Configure the terminating side

to send busy tones to the
originating side.
Configure the duration for

playing busy or ringback
tones.
By default, the terminating side

sends busy tones to the originating
side.
send ringbusy enable
timer { ringback | ringbusy } time
If you want the originating side to

play busy tones, execute the undo
send ringbusy enable command on
By default, the duration for playing
a ringback tone is 60000
milliseconds, and the duration for
playing a busy tone is 30000
milliseconds.
This command takes effect only
after you configure the send
ringbusy enable command.
This command takes effect only on
40
Enabling DTMF to receive and send numbers

R2 signaling has two modes to send and receive numbers:
MFCThe originating and terminating sides use interregister signaling to transmit and request
number information, including the calling number, line information, and billing. In the exchange
process, the terminating side sends responses to the originating side.
DTMFThe originating side transmits the called number to the terminating side digit by digit. The
terminating side does not send any responses.
Compared with the MFC mode, the DTMF mode has a faster connection speed but transmits a smaller
amount of information.
To enable DTMF to receive and send numbers:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.


created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
By default, MFC mode is used.
5.
Enable DTMF to receive and

send numbers.
6.
Configure the delay before

sending DTMF tones.
dtmf enable
Configure the same mode on the

devices. Otherwise, call
establishment will fail.
timer dtmf-delay time
This command takes effect only

when the dtmf enable command
has been configured.
Configuring the connection mode

There are two connection modes:
terminalAfter receiving the called number, the terminating side must wait for the real state (busy
or idle) of the called party before returning the corresponding interregister information to the
originating side.
segmentAfter receiving the called number, the terminating side directly returns the "called party
idle" interregister signaling, without waiting for the real state of the terminating side. If the called
party is busy, the terminating side plays busy tones to the originating side.
To configure the connection mode:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.

timeslot-set ts-set-number timeslot-list

timeslots-list signal r2

created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
41
Step
5.
Configure the connection

mode.
Command
Remarks
callmode { terminal | segment }
By default, the terminal

mode is used.
Maintaining specified timeslots

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.

timeslot-set ts-set-number timeslot-list

timeslots-list signal r2
By default, no timeslot
set is created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Maintain specified timeslots.
ts { block | open | query | reset } timeslots

timeslots-list
N/A
Configuring R2 digital line signaling

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.


created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.
Configure the originating side

to require the terminating side
to send answer signals.
answer enable
By default, the originating side

requires the terminating side to
send answer signals.
Enable re-answer signal

processing on the originating
side.
re-answer enable
By default, re-answer signal

processing is disabled on the
originating side.

to send clear-back signals
when the originating side first
disconnects the line.
clear-forward-ack enable

does not send clear-back signals
when the originating side first
disconnects the line.
8.
Enable metering signal

processing.
metering enable
By default, metering signal

processing is disabled.
9.

to require the terminating side
to send seizure
acknowledgement signals.
seizure-ack enable

requires the terminating side to
send seizure acknowledgement
signals.
6.
7.
10. Configure the ABCD bit

pattern for line signals.
dl-bits { answer | blocking |

clear-back | clear-forward | idle |
release-guard | seizing |
seizing-ack } { receive | transmit }
ABCD
42
The defaults are the values of the

ITU-T standard.
For specific default values, see this
command in Voice Command
Reference.
Step
Command
Remarks
11. Configure the C and D signal

bits.
renew ABCD
The default depends on the R2

signaling standard (configured by
using the mode command).
12. Configure line signal

inversion mode.
reverse ABCD
The default is 0000, that is, line

signal inversion is disabled.
By default, the timeout time is:
60000 milliseconds for answer

13. Set the timeout time of line
signals.
timer dl { answer | clear-back |

clear-forward | | re-answer |
release-guard | seizing } time
signals.
10000 milliseconds for
clear-back, clear-forward, and

release-guard signals.
1000 milliseconds for seizing

and re-answer signals.
Configuring R2 interregister signaling

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
N/A
3.


created.
4.
Enter R2 CAS view.
cas ts-set-number
N/A
5.

to request calling party
information (calling category
and calling number) from the
originating side.
ani { all | ka }

does not request calling party
information.
Configure the number of

dialed digits that the
terminating side collects
before requesting calling
party information.
ani-digit number
The default is 1.
Use Group B signals to

complete interregister
signaling exchange.
group-b enable
By default, Group B signals are

used.
final-callednum enable

does not send a number terminator
to the terminating side after
sending the called number.
6.
7.
8.
9.

to send a number terminator
to the terminating side after
sending the called number.
Configure a signal code for a

special character.
special-character character
number
43
By default, no signal code is

configured for a special character.
To configure signal codes for more
special characters, repeat this
command.
Step
Command
Remarks
10. Configure interregister signal

values.
register-value { billingcategory |
callcreate-in-groupa |
callingcategory | congestion |
demand-refused | digit-end |
null-number | req-billingcategory
|
req-callednum-and-switchgroupa
| req-callingcategory |
req-currentcallednum-in-groupc |
req-currentdigit |
req-firstcallednum-in-groupc |
req-firstcallingnum | req-firstdigit
| req-lastfirstdigit |
req-lastseconddigit |
req-lastthirddigit |
req-nextcallednum |
req-nextcallingnum |
req-switch-groupb |
subscriber-abnormal
|subscriber-busy |
subscriber-charge
|subscriber-idle } value
The defaults depend on the R2

signaling standard (configured by
using the mode command).
11. Configure the duration of

register pulse signals.
timer register-pulse time
12. Configure the maximum time

to wait for a Group B signal.
timer group-b time
Configuring the ISDN protocol

To use an E1 interface as an ISDN PRI, you need to create a PRI set by bundling timeslots other than
timeslots 0 and 16. Timeslot 0 is used to transmit synchronization information, and timeslot 16 is used as
the D channel for signaling transmission. For each PRI set, the system generates a serial interface and a
logical digital voice interface. The serial interface is named serial interface-number:15, where
interface-number is the number of the E1 interface. The serial interface has the same logical
characteristics as synchronous serial interfaces and can be configured with ISDN settings.
A T1 interface is used in the same way as an E1 interface, except that it has 24 timeslots and timeslot 24
is used as the D channel for signaling transmission.
For more information about the ISDN protocol, see Layer 2WAN Configuration Guide.
To configure the ISDN protocol:
Step
Command
1.
Enter system view.
system-view
2.
3.
Bundle timeslots into a PRI set.
pri-set [ timeslots-list range ]
44
Binding a digital voice interface to a POTS entity

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter voice dial program

view.
dial-program
N/A
4.
Create a POTS entity and

enter POTS entity view.
entity entity-number pots
N/A
For E1 and T1 interfaces:

5.
Bind a digital voice interface

to the POTS entity.
line line-number:{ ts-set-number

| 15 | 23 }
For BSV interfaces:
By default, no digital voice

interface is bound to the POTS
entity.
line line-number:
Displaying and maintaining digital voice interfaces

Task
Command
Display digital voice interfaces (for E1

or T1 interfaces).
display voice subscriber line line-number:{ ts-set-number |

ts-set-number.sub-timeslot | 15 | 23 }
Display digital voice interfaces (for

BSV interfaces).
display voice subscriber line { line-number |

ts-set-number.subnumber }
Digital voice interface configuration examples

This section provides configuration examples for digital voice interfaces.
R2 signaling configuration example

As shown in Figure 20, configure R2 signaling on the routers so the telephones can call each other.
45
1.
Configure Router A:
# Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 0/0.
[RouterA] interface GigabitEthernet 0/0
[RouterA-GigabitEthernet0/0] ip address 1.1.1.1 255.255.255.0
[RouterA-GigabitEthernet0/0] quit
# Create a timeslot set on interface E1 5/0.

[RouterA] controller e1 5/0
[RouterA-E1 5/0] timeslot-set 1 timeslot-list 1-31 signal r2
[RouterA-E1 5/0] quit
# Configure the local number 0101001 for POTS entity 1001, and bind the digital voice interface
line 5/0:1 to the POTS entity.
[RouterA-voice-dial-entity1001] line 5/0:1
# Configure the local number 0101002 for PORTS entity 1002, and bind the digital voice
interface line 5/0:1 to the POTS entity.
# Configure the called number template 0755. for VoIP entity 0755, and configure the
2.
Configure Router B:
[RouterB] interface GigabitEthernet 0/0
[RouterB-GigabitEthernet0/0] ip address 2.2.2.2 255.255.255.0
46
[RouterB-GigabitEthernet0/0] quit
# Create a timeslot set on interface E1 5/0.

[RouterB] controller e1 5/0
[RouterB-E1 5/0] timeslot-set 1 timeslot-list 1-31 signal r2
[RouterB-E1 5/0] quit
# Configure the local number 07552001 for POTS entity 2001, and bind the digital voice
interfaces line 5/0:1 to the POTS entity.
[RouterB-voice-dial-entity2001] line 5/0:1
# Configure the called number template 010. for VoIP entity 010, and configure the destination
IP address as 1.1.1.1.
[RouterB-voice-dial-entity10] address ip 1.1.1.1

The telephones can call each other through R2 signaling. You can use the display voice subscriber-line
command to view information about digital voice interfaces.
E1 DSS1 signaling configuration example

As shown in Figure 21, the routers connect to the PBXs through E1 interfaces.
Configure DSS1 signaling on the routers so the telephones can call each other.
47
1.
Configure Router A:
# Bundle the timeslots on interface E1 5/0 into a PRI set.

[RouterA-E1 5/0] pri-set
interfaces line 5/0:15 to the POTS voice entity.
[RouterA-voice-dial-entity755] address ip 2.2.2.2
2.
Configure Router B:
48

# Bundle the timeslots on interface E1 5/0 into a PRI set.

[RouterB-E1 5/0] pri-set

The telephones can call each other through DSS1 signaling. You can use the display voice
subscriber-line command to view information about digital voice interfaces.
BSV DSS1 signaling configuration example

As shown in Figure 21, the routers connect to the PBXs through BSV interfaces.
Configure DSS1 signaling on the routers so the telephones can call each other.
49
1.
Configure Router A:
interfaces line 1/1 to the POTS entity.
[RouterA-voice-dial-entity755] address sip 2.2.2.2
2.
Configure Router B:
50

The telephones can call each other through DSS1 signaling. You can use the display voice
subscriber-line command to view information about digital voice interfaces.
51
Configuring voice entities

Overview
Voice entities include POTS and VoIP entities:
POTS entityA local POTS entity connects to a local telephone and maintains local number
information. A trunk POTS entity connects to the PSTN and maintains call destination information.
VoIP entityConnects to the IP side and maintains the called information such as the called
number and call destination.. A VoIP entity can use SIP to make VoIP calls.
Figure 23 Voice entities

PBX
Router A
Router B
PSTN
Local POTS
entity
IP
Trunk POTS entity
POTS entity
VoIP entity
Configuring a POTS entity

This section covers the procedures for creating and configuring a POTS entity.

To configure a POTS entity, perform the following tasks:
Tasks at a glance
(Required.) Creating a POTS entity and configuring basic parameters
(Optional.) Configuring codecs for a POTS entity
(Optional.) Configuring a POTS entity to register with the registrar
(Optional.) Configuring DTMF for a POTS entity
(Optional.) Enabling VAD for a POTS entity
(Optional.) Configuring options related to dial program
52
Creating a POTS entity and configuring basic parameters

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter dial program view.
dial-program
N/A
4.
Create a POTS entity and enter

POTS entity view.
By default, no POTS voice

entities exist.
5.
(Optional.) Configure a description

for the POTS entity.
description string
By default, no description is
configured.
By default, no number template
is configured for the POTS
entity.
The POTS entity uses the number
template to match phone
numbers. For example, the
match-template 20 command
matches all numbers beginning
with 20.
6.
Configure a number template for

the POTS entity.
match-template match-string
7.
Bind a voice interface to the POTS

entity.
line line-number
By default, no voice interface is

bound to the POTS entity.
8.
Enable the POTS entity.
undo shutdown
By default, a POTS entity is

enabled.
Configuring codecs for a POTS entity

By default, a POTS entity has four codecs g729r8, g711alaw, g711ulaw, and g723r53 in descending
order of priority.
You can use one of the following methods to configure codecs for a POTS entity:
Method 1: Configure codecs for a POTS entity.
Method 2: Create a codec template, assign priorities to codecs in the codec template, and bind the
codec template to a POTS entity.
Two parties must have the same codecs to communicate with each other. You can use the display voice
sip call command (see Voice Command Reference) to view the codec used by two parties.
To configure codecs for a POTS entity (method 1):
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

N/A
53
Step
5.
Configure a codec for the

POTS entity.
Command
Remarks
codec { g711alaw | g711ulaw |

g723r53 | g723r63 | g726r16 |
g726r24 | g726r32 | g726r40 |
g729a | g729br8 | g729r8 }
[ bytes payload-size ]
By default, no codecs are

configured.
To configure codecs for a POTS entity (method 2):

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Create a codec template.
voice class codec tag
By default, no codec templates

exist.
Assign a priority to a codec in

the codec template.
codec preference priority

{ g711alaw | g711ulaw |
g723r53 | g723r63 | g726r16 |
g726r24 | g726r32 | g726r40 |
g729a | g729br8 | g729r8 }
5.
Return to voice view.
quit
N/A
6.
dial-program
N/A
7.

N/A
8.
Bind the codec template to the

POTS entity.
voice-class codec tag
By default, the binding does not

exist.
4.
By default, no codecs exist in a

template.
Repeat this command to assign
priorities to more codecs.
Configuring a POTS entity to register with the registrar

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

N/A
5.
Configure the POTS entity to

register with the registrar
server.
By default, a POTS entity initiates a

registration to the registrar server
after the SIP registration
configuration is complete.
register number
54
If you do not want to register a

number or you want to deregister a
number from the registrar server,
you can use the undo register
number command.
Configuring DTMF for a POTS entity

There are two ways to transmit DTMF tones: in-band signaling and out-of-band signaling.
In-band signaling sends DTMF tones in RTP packets, and out-of-band signaling sends DTMF tones in SIP
messages, or RFC 2883-compliant RTP messages (NTE mode).
To use NTE mode, configure the outband nte command and the same payload type value on the
originating and terminating sides. Otherwise, DTMF tones might fail to be transmitted. For more
information about out-of-band DTMF, see "Configuring SIP."
To configure DTMF for a POTS entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

N/A
(Method 1) Configure NTE
Use either method.
mode:
{
5.
Configure out-of-band DTMF

transmission.
Enable NTE mode:

outband nte
Configure the value of the
payload type used by NTE:
rtp payload-type nte value
(Method 2) Configure SIP

mode:
outband sip
By default, in-band transmission is

adopted, and the value of the
payload type used by NTE is 101.
When the device is connected to a
device from another vendor, you
cannot set the payload type field to
any value forbidden by that
device. Otherwise, an NTE
negotiation failure might occur.
Enabling VAD for a POTS entity

The voice activity detection (VAD) discriminates between silence and speech on a voice connection
according to their energies. With VAD, a POTS entity does not generate traffic during periods of silence.
To enable VAD for a POTS entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

N/A
By default, VAD is disabled.
5.
Enable VAD.
vad-on [ g723r53 | g723r63 |

g729a | g729r8 ] *
55
The G.711 and G.726 codecs do

not support VAD. The G.729br8
codec always supports VAD.
Configuring options related to dial program

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

N/A
5.
Configure permitted calling

numbers.
caller-permit calling-string
Optional.
By default, the voice entity permits all calling
numbers.
Optional.
Configure the priority of the

POTS entity.
priority priority-order
7.
Bind a number substitution

rule list to the voice entity.
substitute { called |
calling } list-number
By default, no number substitution rule list is

bound to a voice entity (no number
substitution is performed).
8.
Bind a subscriber group to

the voice entity.
caller-group { deny |
permit } group-id
By default, no subscriber group is bound to

the voice entity (any calling number is
allowed to originate calls).
9.
Configure the number

sending mode.
send-number
{ digit-number | all |
truncate }
By default, the truncate mode is used.
6.
By default, the priority level is 0.

The smaller the number, the higher the
priority.
For more information about the above commands, see "Configuring dial programs."
Configuring a VoIP entity

This section covers the procedures for creating and configuring a VoIP entity.

To configure a VoIP entity, perform the following tasks:
Tasks at a glance
(Required.) Creating a VoIP entity and configuring basic parameters
(Optional.) Configuring codecs for a VoIP entity
(Optional.) Configuring DTMF for a VoIP entity
(Optional.) Enabling VAD for a VoIP entity
(Optional.) Configuring options related to dial program
56
Creating a VoIP entity and configuring basic parameters

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a VoIP entity and enter

VoIP entity view.
entity entity-number voip
By default, no VoIP voice entities

exist.
5.
description for the VoIP entity.
description string
configured.
By default, no number template is
configured for the POTS entity.
Configure a number template

for the POTS entity.
6.
7.
Configure the destination IP

address.
address sip ip ip-address [ port

port-number ]
8.
Enable the VoIP entity.
undo shutdown
The POTS entity uses the number

template to match phone numbers.
For example, the match-template
20 command matches all numbers
beginning with 20.
By default, no destination IP
address is configured.
For more information about this
command, see Voice Command
Reference.
By default, a VoIP entity is enabled.
Configuring codecs for a VoIP entity

By default, a VoIP entity has four codecs g729r8, g711alaw, g711ulaw, and g723r53 in descending
order of priority.
You can use one of the following methods to configure codecs for a VoIP entity:
Method 1: Configure codecs directly for a VoIP entity.
Method 2: Create a codec template, assign priorities to codecs in the codec template, and bind the
codec template to a VoIP entity.
Two parties must have the same codecs to communicate with each other. You can use the display voice
sip call command (see Voice Command Reference) to view the codec used by two parties.
To configure codecs for a VoIP entity (method 1):
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

VoIP entity view.
N/A
57
Step
5.
Specify a codec for the VoIP

entity.
Command
Remarks
codec { g711alaw | g711ulaw |

g723r53 | g723r63 | g726r16 |
g726r24 | g726r32 | g726r40 |
g729a | g729br8 | g729r8 }
By default, no codecs are

configured.
To configure codecs for a VoIP entity (method 2):

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Create a codec template.
voice class codec tag
By default, no codec templates

exist.
Assign a priority to a codec in

the codec template.
codec preference priority

{ g711alaw | g711ulaw |
g723r53 | g723r63 | g726r16 |
g726r24 | g726r32 | g726r40 |
g729a | g729br8 | g729r8 }
5.
quit
N/A
6.
dial-program
N/A
7.

VoIP entity view.
N/A
8.
Bind the codec template to the

VoIP entity.
voice-class codec tag
By default, the binding does not

exist.
4.
By default, no codecs exist in a

template.
Repeat this command to assign
priorities to more codecs.
Configuring DTMF for a VoIP entity

There are two ways to transmit DTMF tones: in-band signaling and out-of-band signaling.
In-band signaling sends DTMF tones in RTP packets, and out-of-band signaling sends DTMF tones in SIP
messages, or RFC 2883-compliant RTP messages (NTE mode).
To use the NTE mode, configure command outband nte and the same payload type value on both the
originating and terminating sides. Otherwise, DTMF tones might fail to be transmitted. For more
information about out-of-band DTMF, see "Configuring SIP."
To configure DTMF for a VoIP entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

N/A
58
Step
Command
Remarks
(Method 1) Configure NTE
Use either method.
mode:
{
5.
Configure out-of-band DTMF .
Enable NTE mode:

outband nte
Configure the value of the
payload type used by NTE:
rtp payload-type nte value
(Method 2) Configure SIP

mode:
outband sip
By default, inband transmission is

adopted, and the value of the
payload type used by NTE is 101.
When the device is connected to a
device from another vendor, you
cannot set the payload type field to
any value forbidden by that
device. Otherwise, an NTE
negotiation failure might occur.
Enabling VAD for a VoIP entity

The voice activity detection (VAD) discriminates between silence and speech on a voice connection
according to their energies. With VAD, a POTS entity does not generate traffic during periods of silence
in an active voice connection.
To enable VAD:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.

VoIP entity view.
N/A
By default, VAD is disabled.
5.
Enable VAD.
vad-on [ g723r53 | g723r63 |

g729a | g729r8 ] *
The G.711 and G.726 codecs do

not support VAD. The G.729br8
codec always supports VAD.
Configuring options related to dial program

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a VoIP entity and

enter VoIP entity view.
N/A
5.

permitted calling numbers.
Optional.
59
By default, the voice entity permits all calling

numbers.
Step
Command
Remarks
Optional.
Configure the priority of the

VoIP entity.
7.
Bind a number substitution

rule list to the VoIP entity.
substitute { called |
calling } list-number
By default, no number substitution rule list is

bound to a VoIP entity (no number
substitution is performed).
8.
Bind a subscriber group to

the VoIP entity.
caller-group { deny |
permit } group-id
By default, no subscriber group is bound to a

VoIP entity (any calling number is allowed to
originate calls).
6.
By default, the priority level is 0.

The smaller the number, the higher the
priority.
For more information about the above commands, see "Configuring dial programs."
Displaying and maintaining voice entities

Task
Command
Display voice call information.
display voice call-info { tag | all }
Display the configuration of voice entities.
display voice entity { entity-tag | all | pots | voip }
60
Configuring dial programs

This chapter describes using dial programs to define call control policies such as caller control, number
substitution, call sending, and number match policies.

Tasks at a glance
(Optional.) Configuring caller control
(Optional.) Configuring caller group control
(Optional.) Enabling private line auto ring-down
(Optional.) Configuring a number match mode
(Optional.) Configuring the maximum number of total calls allowed by a voice entity
(Optional.) Configuring number substitution
(Optional.) Configuring a priority for a voice entity
(Optional.) Configuring a number sending mode
(Optional.) Configuring a dial prefix
Configuring caller control

Caller control allows you to filter outgoing calls on the calling side or filter incoming calls on the called
side based on calling numbers.
Procedure
To configure caller control:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a voice entity and

enter voice entity view.
entity entity-number { pots | voip }
N/A
61
Step
Command
Remarks
By default, the voice entity permits
all calling numbers.
5.
Configure permitted calling

numbers.
The calling-string argument is in the

format of { [ + ] string [ $ ] }| $. For
details, see Voice Command
Reference.
Examples
As shown in Figure 24, configure caller control to permit only the number 1000 to call the number 2000.
Figure 24 Caller control diagram
You can configure caller control either on the calling side or on the called side to meet the requirement.
Method 1: Configure the calling side to permit only the calling number 1000 to call the number
2000.
[RouterA-voice-dial-entity2000] caller-permit 1000$
Method 2: Configure the called side to permit only the calling number 1000 to call the number
2000.
[RouterB-voice-dial-entity2000] caller-permit 1000$
62
Configuring caller group control

Caller group control allows you to filter outgoing calls on the calling side or filter incoming calls on the
called side based on caller groups. Caller group control is often used on the calling side.
A caller group references a subscriber group that defines the matching calling numbers. After you bind
a caller group to a voice entity on the calling side, the matching calling numbers can or cannot call the
called numbers on the voice entity.
Configuring a subscriber group

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a subscriber group and

enter subscriber group view.
subscriber-group group-id
description for the subscriber
group.
description text
5.
6.
Configure a match template for

the subscriber group.
By default, no subscriber group

exists.
You can configure a maximum of
10 subscriber groups.
configured for the subscriber
group.
By default, no match template is
configured for the subscriber
group.
You can configure multiple match
templates for a subscriber group.
Configuring a caller group on a voice entity

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a voice entity and enter

voice entity view.
entity entity-number { pots |

voip }
N/A
5.
Configure the voice entity to

permit or deny the calling
numbers in the specified
subscriber group.
caller-group { deny | permit }

group-id
By default, all calling numbers are

permitted.
63
Enabling private line auto ring-down

The private line auto ring-down (PLAR) function enables a subscriber line to automatically call the
specified called number when the phone goes off-hook.
To configure the PLAR function:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter subscriber line view.
N/A
3.
Configure PLAR.
private-line string
By default, PLAR is not configured.
Configuring a number match mode

Two number match modes are available: longest match and shortest match.
Suppose you have configured match-template 0106688 on a voice entity and match-template
01066880011 on another voice entity.
When a subscriber dials 01066880011:
If shortest match is used, the router matches and calls the number 0106688.
If longest match is used, the router matches and calls the number 01066880011.
When a subscriber dials 0106688:
If shortest match is used, the router immediately matches and calls the number 0106688.
If longest match is used, the router waits until the dial timer expires. Then the router matches the
received number against match templates. If a match is found, the router calls the matching number.
If no match is found, the router releases the call.
You can configure a terminator that identifies the end of a number. Suppose the terminator is the pound
sign #. When a subscriber dials 0106688#, the router immediately matches the number 0106688 with
match templates regardless of whether longest match is used. If a match is found, the router calls the
matching number.
Procedure
To configure a number match mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Configure a number match

mode.
number-match { longest |
shortest }
By default, shortest match is used.
5.
Configure a terminator.
terminator character
By default, no terminator is
configured.
64
Examples
As shown in Figure 25, configure number match modes for calls from Telephone A to Telephones B and
C.
Using shortest match

1.
Configure Router A:
# Configure POTS entity 1000.
[RouterA-voice-dial-entity1000] match-template 10001234$
# Configure VoIP entity 2000 and VoIP entity 2001.

[RouterA-voice-dial] entity2000 voip
[RouterA-voice-dial] entity2001 voip
2.
Configure Router B:
# Configure POTS entity 2000 and POTS entity 2001.
[RouterB-voice-dial-entity2000] match-template 20001234$
65

[RouterB-voice-dial] entity2001 pots
[RouterB-voice-dial-entity2001] match-template 200012341234$
After you dial number 20001234 at Telephone A, the number matches VoIP entity 2000 and
Telephone B rings because the device uses shortest match mode by default.
Configuring longest match

# Configure the longest match mode on Router A.
[RouterA-voice-dial] number-match longest
After you dial number 20001234 at Telephone A and waits for a period of time, the number matches
VoIP entity 2000 and Telephone B rings. If you dial 1234 during the waiting period (the dialed number
is 200012341234), the number matches VoIP entity 2001 and Telephone C rings.
Configuring a terminator
# Configure the terminator # on Router A.
[RouterA-voice-dial] terminator #
After you dial 20001234# at Telephone A, the number immediately matches VoIP entity 2000 and
Telephone B rings.
Configuring the maximum number of total calls

allowed by a voice entity
You can limit the maximum number of total calls (both incoming and outgoing) allowed by a voice entity.
The device stops establishing new calls when the limit is reached.
To configure the maximum number of total calls allowed:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Enter voice entity view.
N/A
5.

number of total calls allowed
by the voice entity.
max-conn max-number
By default, no maximum number of

total calls is configured.
Configuring number substitution

A number substitution rule replaces a matching number with another number. You can configure number
substation rules and then apply these rules globally or to voice entities.
A number substitution process is as follows:
66
1.
The router first matches a number against the preferred number substitution rule. If they match, the
router replaces the number based on the preferred rule.
2.
If the match fails, the router matches the number against other number substitution rules in
sequence. Once a rule is matched, the router replaces the number based on the matching rule and
stops matching other number substitution rules.
Number substitution on the calling router

As shown Figure 26, the calling router performs number substitution for a number as follows:
1.
The router matches the number against global number substitution rules. If a match is found, the
router replaces the number based on the matching rule.
2.
If no match is found, the router selects a matching voice entity, and replaces the number based on
the matching rule on the voice entity.
3.
The router initiates a call to the called side.
Figure 26 Flow chart for number substitution on the calling router

Voice interface
PSTN
Yes
Global number
substitution rule matched?
Performs number
substitution
No
Voice entity matched?
No
Yes
Number
substitution rule on the voice
entity matched?
No
Yes
Performs number substitution
Calls the called party
Call failure
Number substitution on the called router

As shown Figure 26, the called router performs number substitution for a number as follows:
1.
The router matches the number against global number substitution rules. If a match is found, the
router replaces the number based on the matching rule.
67
2.
If no match is found, the router selects a matching voice entity, and matches the number against
rules on the voice entity. If a match is found, the router replaces the number based on the matching
rule.
3.
If no match is found, the router calls the called number if the callee is a local subscribe line, or
initiates a call to the PSTN if the callee is in the PSTN.
Figure 27 Flow chart for number substitution on the called router

Receives the called
number
Global number substitution

rule matched?
Yes
Performs number
substitution
No
No
Voice entity matched?
Yes
Number substitution rule on

the voice entity matched?
No
Yes
Performs number substitution
PSTN
Call failure
Voice interface
Configuring global number substitution

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a number substitution

rule list and enter
number-substitute view.
number-substitute list-number
By default, no number substitution

rule list exists.
Configure a dot-match rule.
dot-match { end-only | left-right |

right-left }
By default, the dot match rule is

end-only.
5.
68
Step
Command
Remarks

rule is configured.
6.
Configure a number
substitution rule.
rule id input-template
output-template [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *
7.

preferred number substitution
rule.
first-rule id
By default, the preferred number

substitution rule is not configured.
8.
Return to dial program view.
quit
N/A
9.
Apply the number substitution

rule list to the calling/called
number of
incoming/outgoing calls.
substitute { incoming-call |
outgoing-call } { called | calling }
list-number

rule list is applied.
Configuring number substitution for a voice entity

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Create a number substitution

rule list and enter
number-substitute view.
number-substitute list-number

rule list exists.
Configure a dot match rule.
dot-match { end-only | left-right |

right-left }
By default, the dot match rule is

end-only.
6.
Configure a number
substitution rule.
rule id input-template
output-template [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *

rule is configured.
7.

preferred number substitution
rule.
first-rule id
By default, the preferred number

substitution rule is not configured.
8.
Return to dial program view.
quit
N/A
9.
entity entity-number { pots |voip }
N/A
substitute { called | calling }

list-number

rule list is applied to a voice entity.
5.
10. Apply the number substitution

rule list to the voice entity.
69
Examples
This section provides examples for global number substitution and voice entity number substitution.
Configuring global number substitution

# Create number substitution rule list 1 and add a rule to the list.
[RouterA-voice-dial] number-substitute 1
[RouterA-voice-dial-substitute101] rule 0 ^010....$ ....
[RouterA-voice-dial-substitute101] quit
# Configure the destination address as 1.1.1.2 and the called number as 1001 for VoIP entity 1001.
# Apply number substitution rule list 1 to the called number of outgoing calls.
[RouterA-voice-dial] substitute outgoing-call called 1
When you use Telephone A to call number 0101001, Router A replaces 0101001 with 1001 and sends
the call to the destination address 1.1.1.2. Then Telephone B rings.
Configuring voice entity number substitution

# Create number substitution rule list 1 and add a rule to the list.
[RouterA-voice-dial-substitute101] rule 0 ^010....$ ....
# Configure the destination address as 1.1.1.2 and the called number as 0101001 for VoIP entity 1001.
# Apply number substitution rule list 1 to the called number on VoIP entity 1001.
[RouterA-voice-dial-entity1001] substitute called 1
When you use Telephone A to call number 0101001, VoIP entity 1001 replaces 0101001 with 1001 and
sends the call to the destination address 1.1.1.2. Then Telephone B rings.
70
Configuring a priority for a voice entity

If a number matches multiple voice entities, the router selects the voice entity with the highest priority.
To configure a priority for a voice entity:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Configure a priority for the

voice entity.
The default priority is 0.
Configuring a number sending mode

The number sending mode controls how the originating router sends a called number. The router
supports the following number sending modes:
Sends the least significant digits of a called number (the number of the least significant digits is
specified with the send-number digit-number command).
Sends all the digits of a called number.
Sends a truncated called number. When the match-template command configured for a voice
entity contains dots ".", only the digits that match the dots are sent.
Procedure
To configure a number sending mode:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Enter POTS voice entity view.
N/A
5.
Configure a number sending

mode.
send-number { digit-number | all |

truncate }
By default, the truncate mode is

used.
Examples
As shown in Figure 29, Router A and Router B connect to each other through an IP link and an E1 link.
The telephones at two sides call each other through the E1 link. This section provides examples for
configuring number sending modes to achieve different call policies.
71

FXS 3/0
FXS 3/0
Telephone A
2000
Router A
Eth2/1
1.1.1.1/24
Eth2/1
1.1.1.2/24
E1 1/1
E1 1/1
Telephone B
01001
FXS 3/1
Telephone C
0101001
FXS 3/2
Telephone D
1001
Router B
Configuring basic settings

1.
Configure Router A:
# Configure a timeslot set using R2 signaling for controller E1 1/1.
[RouterA-E1 1/1] timeslot-set 0 timeslot-list 1-31 signal r2
# Configure a match template to match 010.$ for POTS entity 1001 and bind line 1/1:0 to the
entity.
[RouterA-voice-dial-entity1001] match-template 010....$
# Configure a local number 2000 for POTS entity 2000, and bind line3/0 to the entity.
2.
Configure Router B:
# Configure a timeslot set using R2 signaling for controller E1 1/1.
[RouterB-E1 1/1] timeslot-set 0 timeslot-list 1-31 signal r2
# Configure a local number 1001 for POTS entity 1002, and bind line 3/0 to the entity.
72

Configuring number sending modes on Router A

# Configure Router A to send the five least significant digits of a called number.
[RouterA-voice-dial-entity1001] send-number 5
If you use Telephone A to call 0101001, Router A sends the called number 01001, which are the five least
significant digits. Then Telephone B rings.
# Configure Router A to send all the digits of a called number.
If you use Telephone A to call 0101001, Router A sends the complete called number 0101001. Then
Telephone C rings.
# Configure Router A to send a truncated called number.
[RouterA-voice-dial-entity1001] send-number truncate
If you use Telephone A to call 0101001, Router A sends the called number 1001, which match the dots in
"010.$". Then Telephone D rings.
Configuring a dial prefix

After you configure a dial prefix, the router adds the prefix before each called number. If the called
number exceeds 31 digits, the router sends only the first 31 digits.
To configure a dial prefix:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Enter POTS entity view.
N/A
5.
Configure a dial prefix.
dial-prefix string
By default, no dial prefix is

configured.
Dial program configuration examples

This section provides dial program configuration examples.
73
Configuring number substitution

As shown in Figure 30, configure number substitution on the routers so departments in different areas can
call each other by using the area code of the destination area plus a four-digit local number. For example,
to use phone 1688 of the F department in area B to call phone 6788 of the M department in area A, dial
the number 0103366, in which 3366 is the number of the M department in area B. On phone 6788, the
calling number 1688 is displayed as 0211234, which is the area code of area A plus the number of the
F department in area A.
B area code: 021
A area code: 010
M Dept. 3366
M Dept. 6788
Eth2/1
2.2.2.2/24
FXO 1/0
F Dept. 1688
PBX
WAN
Eth2/1
1.1.1.1/24
Router B
FXO 1/0
Router A
PBX
F Dept 1234
S Dept. 6565
S Dept. 2323
The following only provides the number substitution configuration for calls from area B to area A.
1.
Configure Router B:
# Create number substitution rule list 21101 and add rules to the list.
[RouterB-voice-dial] number-substitute 21101
[RouterB-voice-dial-substitute21101] rule 1 ^0101688$ 0001
[RouterB-voice-dial-substitute21101] quit
[RouterB-voice-dial] number-substitute 21102
[RouterB-voice-dial-substitute21102] quit
# Configure the match template "010...." to match called numbers, and set the destination IP
address as 1.1.1.1 for VoIP entity 10.
74
# Apply the number substitution rule list 21101 to called numbers on VoIP entity 10, which will
change the called numbers 0101688, 0103366, and 0102323 into 0001, 0002, and 0003
respectively.
[RouterB-voice-dial-entity10] substitute called 21101
# Apply the number substitution rule list 21102 to calling numbers on VoIP entity 10, which will
change the calling numbers 1688, 3366, and 2323 into 0210001, 0210002, and 0210003
respectively.
[RouterB-voice-dial-entity10] substitute calling 21102
2.
Configure Router A:
[RouterA-voice-dial-substitute101] rule 1 ^0001$ 1234
[RouterA-voice-dial-substitute102] dot-match left-right
[RouterA-voice-dial-substitute102] rule 1 ^...0001$ ...1234
# Apply the number substitution rule list 101 to the called numbers of incoming calls, which will
change the called numbers 0001, 0002, and 0003 into 1234, 6788, and 6565 respectively.
[RouterA-voice-dial] substitute incoming-call called 101
# Apply the number substitution rule list 102 to the calling numbers of incoming calls, which will
change the called numbers 0210001, 0210002, and 0210003 into 0211234, 0216788, and
0216565 respectively.
[RouterA-voice-dial] substitute incoming-call calling 102
# Configure the match template "." for POTS entity 1010, bind line 1/0 to the entity, and
configure the entity to send all the digits of a called number.
[RouterA-voice-dial-entity1010] match-template ....

Use phone 1688 to dial the number 0103366. Then phone 6788 rings and the displayed calling number
is 0211234.
During the calling process, Router B changes the calling number 1688 into 0210001 and changes the
called number 0103366 into 0002, and then Router A changes 0210001 into 0211234 and changes
0002 into 6788.
75
Configuring caller group control

As shown in Figure 31, areas A, B, and C call each other through the SIP server. Configure caller group
control on Router A so when the SIP server fails, the areas call each other though the public PBXs in PSTN
as follows:
The phone numbers starting with 1100 in area A can only call the phone numbers in area B.
The phone numbers starting with 1200 in area A can call the phone numbers both in area B and
area C.

Area A
Area B
110000
Router B
2100
E1 1/0
1100..
Internal
PBX
110099
Router A
IP
120000
Internal
PBX
2200
Public PBX
Area C
Router C
1200..
3100
E1 1/0
120099
Public PBX
Internal
PBX
SIP server
3200
Public PBX
This example does not provide SIP server and digital voice interface configurations. For related
information, see "Configuring SIP" and "Configuring voice interfaces."
1.
Configure Router A:
# Configure subscriber group 1 that matches calling numbers starting with 1100 and configure
subscriber group 2 that matches calling numbers starting with 1200.
[RouterA-voice-dial] subscriber-group 1
[RouterA-voice-dial-group1] match-template 1100..
[RouterA-voice-dial-group1] quit
[RouterA-voice-dial] subscriber-group 2
[RouterA-voice-dial-group2] match-template 1200..
[RouterA-voice-dial-group2] quit
# Configure VoIP entity 2000 to use the SIP proxy server to find call destination addresses and to
match called numbers in area B.
76
[RouterA-voice-dial-entity2000] address sip proxy

[RouterA-voice-dial-entity2000] match-template 2...
# Configure VoIP entity 3000 to use the SIP proxy server to find call destination addresses and to
match called numbers in area C.
# Configure a caller group to permit subscriber groups 1 and 2 on POTS entity 2100 so the phone
numbers starting with 1100 or 1200 in area A can call the phone numbers in area B.
[RouterA-voice-dial-entity2100] caller-group permit 1
# Configure a caller group to permit subscriber group 2 on POTS entity 3100 so the phone
numbers starting with 1200 in area A can call the phone numbers in area C.
[RouterA-voice-dial] quit
# Configure POTS entity 2100 to match local numbers and bind line 1/0:15 to the entity.
[RouterA-voice-dial-entity2100] match-template 1.....
2.
Configure Router B:
[RouterB-voice-dial-entity2100] match-template 2...
3.
Configure Router C:
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3100 pots
[RouterC-voice-dial-entity3100] line 1/0:15
77
[RouterC-voice-dial-entity3100] send-number all

[RouterC-voice-dial-entity3100] match-template 3...

The phone numbers starting with 1100 in area A can only call the phone numbers in area B.
The phone numbers starting with 1200 in area A can call the phone numbers both in area B and area
C.
78
Configuring SIP
Overview
The Session Initiation Protocol (SIP) is an application layer control protocol that can create, modify, and
terminate multimedia sessions such as voice and video calls over IP networks.
Terminology
User agent
A user agent (UA) is a SIP endpoint such as a phone, a gateway, or a router.
There are two types of UAs: user agent client (UAC) and user agent server (UAS). A UAC sends SIP
requests, and a User Agent Server (UAS) receives SIP requests and returns SIP responses. These roles of
UAC and UAS only last for the duration of a SIP transaction.
Proxy server
A proxy server primarily forwards session requests and responses. It can also provide call control,
accounting, and authorization functions.
Redirect server
A redirect server sends new addresses to UACs so the UAC sends session requests to the new addresses.
Location server
A location server provides UA information to proxy and redirect servers.
Registrar
A registrar receives registrations from UAs and generates UA information. The UA information is stored
on the location server.
SIP functions
SIP supports the following facets of establishing and terminating multimedia communications:
User locationDetermines the end system to be used for communication. SIP can use UA
information on the registrar, or use information provided by DNS or LDAP to locate end systems.
User availabilityDetermines the willingness of the called party to engage in communications.
User capabilitiesDetermines the media type and media parameters to be used. In a message
exchange process, each SIP endpoint advertises media information so that all other participants
can learn about its capabilities.
Session setupEstablishes session parameters at both called and calling parties.
Session managementTransfers and terminates sessions, modifies session parameters, and

invokes services.
79
SIP messages
SIP is a text-based protocol. A SIP message is either a request from a client to a server, or a response from
a server to a client.
SIP requests include INVITE, ACK, OPTIONS, BYE, CANCEL, and REGISTER.
INVITEInvites a user to join a call.
ACKAcknowledges the response to a request.
OPTIONSQueries for the capabilities.
BYEReleases an established call.
CANCELGives up a call attempt.
REGISTERRegisters with the SIP registrar.
SIP responses indicate the status of a call or registration. Responses are distinguished by status codes. As
shown in Table 8, each status code is a 3-digit integer, where the first digit defines the class of the
response, and the last two digits describe the response message in more detail.
Table 8 Status codes of responses
Code
Description
Class
100199
The request was received and is being processed.
Provisional
200299
The request was successfully received, understood, and accepted.
Success
300399
A further action needs to be taken to complete the request.
Redirection
400499
The request contains bad syntax or cannot be fulfilled at this server.
Client error
500599
The server failed to fulfill an apparently valid request.
Server error
600699
The request cannot be fulfilled at any server.
Global failure
SIP configuration task list

Tasks at a glance
Configuring SIP UA registration
(Optional.) Configuring SIP credentials

(Required.) Enabling a POTS entity to register with the registrar
(Required.) Configuring registrar information
(Required.) Use one of the following methods to specify a call destination address for a VoIP entity
Configuring the call destination IP address for a VoIP entity

Configuring a VoIP entity to obtain the call destination address from a proxy server
Configuring the destination domain name and port number for a VoIP entity
(Optional.) Configuring the call destination address for a VoIP entity
80
Tasks at a glance
(Optional.) Configuring extended SIP functions
Configuring out-of-band DTMF

Configuring periodic refresh of SIP sessions
Configuring PSTN cause-to-SIP status mappings
Configuring caller privacy
Setting the P-Asserted-Identity or P-Preferred-Identity header field
Configuring SIP UA registration

A SIP UA registers with a SIP registrar as shown in Figure 32:
1.
The SIP UA sends a REGISTER request to the registrar.
2.
The registrar returns a 200 OK response to the UA if the registration is accepted.
Figure 32 Registration process

Registrar
UA
IP network
REGISTER
200 OK
If the registrar needs to authenticate the UA, the UA registers with the registrar as shown in Figure 33:
3.
The UA sends a REGISTER request to the registrar.
4.
The registrar returns a 401/407 response, challenging the originator to provide credentials.
5.
The UA sends a REGISTER request that includes credentials to the registrar.
6.
The registrar returns a 200 OK response to the UA if the authentication succeeds.
Figure 33 Registration process with authentication
Configuration prerequisites
Complete the following tasks before you configure SIP credentials information:
81
Configure a number template on each voice entity with the match-template command and bind
each voice entity to a voice subscriber line with the line command.
Enable the voice entities and voice subscriber lines (in undo shutdown state).
Configuring SIP credentials

If the registrar provides UA authentication, configure SIP credentials on the SIP UA in the following ways:
Use the user command in SIP view to configure global SIP credentials.
Use the credentials command to configure SIP credentials for a SIP trunk account. For more
information about SIP trunk, see Configuring SIP trunk.
Use the user command in voice entity view to configure SIP credentials for a voice entity.
Registration with multiple registrars

A SIP UA can register with up to six registrars, and it uses the domain name in the 401/407 response
from a registrar to identify the credentials to be sent to the registrar.
You can configure only one username with the user command in SIP view or voice entity view. The
username can contain 12 credentials bindings. A binding that does not include a domain name can be
used to respond to a 401/407 response that does not match any domain name-included binding. The
following example configures four credentials bindings:
[Sysname-voice-dial-entity100] user 1000 password simple 1000 realm server1
[Sysname-voice-dial-entity100] user 1000 password simple 3000
The first three bindings each contain a domain name, and the last binding contains no domain name. If
the SIP UA receives a 401/407 response that includes a domain name server2, the SIP UA responds with
the username 1000 and password 1000. If the SIP UA receives a 401/407 response that includes a
domain name server4, the SIP UA responds with the username 1000 and password 3000 because no
credentials binding contains the domain name server4.
Credentials selection for a phone number that exists on multiple voice entities
Upon receiving a 401/407 response for a phone number that exists on multiple voice entities, the SIP UA
considers the phone number belongs to the voice entity with the smallest ID and selects the credentials for
the phone number in the following order:
1.
Credentials on the voice entity with the smallest ID.
2.
Credentials configured with the credentials command.
3.
Credentials configured in SIP view.
The SIP UA always uses the matching credentials for the phone number even if a voice entity that has a
higher match priority is added. If no matching credentials are found, the SIP UA fails to register the
phone number.
For example, the registrar maintains the username abcd, password 1234, and domain name abc for the
phone number 1000. The SIP UA has the following settings for the phone number 1000:
POTS entity 1 maintains the username abcd, password 1234, and domain name aaa for the phone
number 1000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
82
[Sysname-voice-dial] entity 1 pots

[Sysname-voice-dial-entity1] match-template 1000
[Sysname-voice-dial-entity1] user abcd password simple 1234 realm aaa
POTS entity 2 maintains the username abcd, password 1234, and domain name abc for the phone
number 1000.
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 2 pots
[Sysname-voice-dial-entity2] match-template 1000
[Sysname-voice-dial-entity2] user abcd password simple 1234 realm abc
The SIP trunk account maintains the username abcd, password 1234, and domain name abc for the
phone number 1000.
[Sysname-voice] sip
[Sysname-voice-sip] credentials number 1000 username abcd password simple 1234 realm
abc
The global SIP credentials information includes the username abcd, password 1234, and domain
name abc.
[Sysname-voice] sip
[Sysname-voice-sip] user abcd password simple 1234 realm abc
Upon receiving a 401/407 response for the phone number 1000, the SIP UA considers the phone
number belongs to POTS entity 1 and selects the credentials on POTS entity 1 because POTS entity 1 has
a smaller ID, but the credentials fail the authentication. Then the SIP UA selects the credentials configured
with the credentials command and the credentials pass the authentication. The output from the display
voice sip register-status command shows that the phone number 1000 belongs to voice entity 1.
<Sysname> display voice sip register-status
Number
Entity
Registrar Server
Expires Status
-------------------------------------------------------------------------------1000
192.168.4.240:5060
2877
Online
To configure global SIP credentials:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
4.
Configure global SIP

credentials.
user username password { cipher |

simple } password [ realm realm ]
By default, no global SIP credentials are

configured.
To configure SIP credentials for a POTS entity:
83
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Configure SIP credentials for

the POTS entity.
user username password
{ cipher | simple } password

[ cnonce cnonce | realm
realm ] *
By default, no SIP credentials are

configured for a POTS entity.
Enabling a POTS entity to register with the registrar

Perform this task to enable a POTS entity to register with the registrar. You can also disable a POTS entity
to register with the registrar by using the undo register-number command if registration is not needed.
To enable a POTS entity to register with the registrar:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Enable the POTS entity to

register with the registrar.
register-number
By default, a POTS entity registers

with the registrar after UA
registration is configured.
Configuring registrar information

Perform this task to specify a registrar. The expires keyword sets the registration expiry time, and the
refresh-ratio keyword sets the refresh percentage. When the registration time reaches the registration
expiry time multiplied by the refresh percentage, a voice entity or SIP trunk re-registers the number with
the registrar to avoid expiration.
To configure registrar information:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
4.
Configure the registrar

information for the SIP
UA.
registrar registrar-index ip ip-address

[ port port-number ] [ expires seconds ]
[ refresh-ratio ratio-percentage ]
By default, no registrar information

is configured.
84
Displaying SIP UA registration status

Task
Command
Display SIP UA registration status.
display voice sip register-status
Configuring the call destination address for a VoIP

entity
You can configure the call destination address for a VoIP entity in one of the following ways:
Configure the call destination IP address.
Configure the VoIP entity to obtain the call destination address from a proxy server.
Configure the destination domain name and port number. Only DNS A records are supported. For
more information about DNS, see Layer 3IP Services Configuration Guide.
Configuring the call destination IP address for a VoIP entity

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
Enter VoIP entity view.
N/A
5.
Configure the call destination

IP address.
address sip ip ip-address [ port

port-number ]
By default, no call destination

address is configured.
Configuring a VoIP entity to obtain the call destination address

from a proxy server
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
4.
Specify the proxy server.
proxy { dns domain-name port

port-number | ip ip-address [ port
port-number ] }
5.
quit
By default, no proxy server is

specified.
The port number must be 5060 for
the domain name.
N/A
85
Step
Command
Remarks
6.
dial-program
N/A
7.
N/A
8.
Enable the VoIP entity to

obtain the call destination
address from the proxy server.
address sip proxy
By default, a VoIP entity does not

obtain the call destination address
from a proxy server.
Configuring the destination domain name and port number for

a VoIP entity
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Configure the destination

domain name and port
number.
address sip dns domain-name port

port-number
By default, no destination domain

name or port number is
configured.
The port number must be 5060.
Configuring extended SIP functions

This section describes extended SIP configurations.

There are two ways to transmit DTMF tones: in-band signaling and out-of-band signaling. In-band
signaling sends DTMF tones in RTP packets, and out-of-band signaling sends DTMF tones in SIP
messages.
To use out-of-band signaling, configure the outband sip command on both the calling and called
devices.
To configure out-of-band DTMF:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Enable out-of-band DTMF.
outband sip
By default, inband DTMF is

enabled.
86
Configuring periodic refresh of SIP sessions

If a proxy server fails to receive a BYE message, it will always retain the state for the session. To solve this
problem, RFC 4082 defines a session refresh mechanism, which periodically sends re-INVITE or UPDATE
requests (referred to as refresh requests) to notify the proxy server of the current session state. The refresh
interval is determined through negotiation between the SIP UA and the SIP proxy server.
Session refresh uses two header fields Session-Expires and Min-SE, and a 422 response:
Session-ExpiresConveys the maximum session expiration time. If no refresh request is received

within this time, the session is considered ended.
Min-SEConveys the minimum session expiration time, which is used to avoid frequent refresh
requests from occupying excessive network bandwidth.
422 responseWhen a UAS or SIP proxy server receives a request in which the Session-Expires
field conveys a value smaller than the local Min-SE value, the UAS or SIP proxy server sends a 422
response that contains the local Min-SE value to notify the requesting party of the minimum session
expiration time.
To configure periodic refresh of SIP sessions:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
session refresh
By default, session refresh is

disabled if the device acts as a
UAC, and is enabled if the device
acts as a UAS.
4.
Enable session refresh.
Configure this command on the

UAC.
5.
Set the maximum session

expiration time and minimum
session expiration time.
By default, both the maximum

session expiration time and
minimum session expiration time
are 1800 seconds.
min-se time [ session-expires

interval ]
Configuring PSTN cause-to-SIP status mappings

The default PSTN cause-to-SIP status mappings are used for communication between a SIP network and
a PSTN. For information about these default mappings, see Voice Command Reference. You can also
change the default mappings to meet specific needs.
To configure PSTN cause-to-SIP status mappings:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
87
Step
Command
Remarks
4.
Configure a PSTN cause

code-to-SIP status
mapping.
set pstn-cause pstn-cause

sip-status sip-status
For information about default PSTN cause

code-to-SIP status mappings, see Voice
Command Reference.
5.
Configure a SIP
status-to-PSTN cause
mapping.
set sip-status sip-status

pstn-cause pstn-cause
For information about default SIP

status-to-PSTN cause mappings, see
Voice Command Reference.
6.
Display PSTN cause-to-SIP

status mappings.
display voice sip map { pstn-sip

| sip-pstn }
N/A
Configuring caller privacy

SIP uses the following header fields to provide caller privacy:
PrivacyIf the Privacy header field carries "Privacy: none", the caller ID is displayed. If the Privacy
header field carries "Privacy: id", the caller ID is hidden.
Remote-Party-IDIf the Remote-Party-ID header field carries "privacy=off", the caller ID is

displayed. If the Remote-Party-ID header field carries "privacy=full", the caller ID is hidden. If the
Remote-Party-ID header field is set, the device adds this field to an INVITE request. A device that
receives the INVITE request preferably obtains caller information from this field regardless of
whether the local Remote-Party-ID header field is set.
NOTE:
The Remote-Party-ID header field cannot coexist with the P-Preferred-Identity or P-Asserted-Identity
header field.
To configure caller privacy:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
4.
Add the Privacy header field

into INVITE requests.
privacy
By default, INVITE requests do not

carry the Privacy header field.
5.
Add the Remote-Party-ID

header field into INVITE
requests.
remote-party-id
By default, INVITE requests do not

carry the Remote-Party-ID header
field.
Setting the P-Asserted-Identity or P-Preferred-Identity header

field
Any two of the P-Asserted-Identity, P-Preferred-Identity, and Remote-Party-ID header fields cannot coexist.
To set the P-Asserted-Identity or P-Preferred-Identity header field:
88
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
4.
Add the P-Asserted-Identity or

P-Preferred-Identity header
field into SIP messages.
asserted-id { pai | ppi }
By default, SIP messages do not

carry the P-Asserted-Identity or
P-Preferred-Identity header field.
Displaying and maintaining SIP

Use display commands in any view.
Task
Command
Display SIP UA registration status information.
Display PSTN cause-to-SIP status mappings .
display voice sip map { pstn-sip | sip-pstn }
Display SIP calling information.
display voice sip call
SIP UA configuration examples

Configuring direct SIP calling
As shown in Figure 34, configure direct SIP calling on Router A and Router B so the phones 1111 and
2222 can call each other.
1.
Configure Router A:
# Configure an IP address for Ethernet 2/1.
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure call destination address 192.168.2.2 and number 2222 for VoIP entity 2222.
89

# Configure local number 1111 for POTS entity 1111 and bind FXS voice subscriber line 1/0 to
the POTS entity.
2.
Configure Router B:
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
the POTS entity.
[RouterB-voice-dial]entity 1111 voip

# Phone 1111 and phone 2222 can call each other.
# Executing the display voice sip call command can display the SIP call information.
Configuring SIP calling through a SIP server

As shown in Figure 35, configure SIP calling through the SIP server on Router A and Router B so the
phones 1111 and 2222 can call each other.
On Router A, all phone numbers use the same username routerA and password 1234.
On Router B, phone number 2222 uses the username routerb, password 7890, and domain name
server1.
90
1.
Configure Router A:
# Specify the SIP registrar and proxy server.

[RouterA-voice] sip
[RouterA-voice-sip] registrar 1 ip 192.168.2.3
[RouterA-voice-sip] proxy ip 192.168.2.3
# Configure user name routerA and plaintext password 1234.

[RouterA-voice-sip] user routerA password simple 1234
[RouterA-voice-sip] quit
the POTS entity.
# Configure VoIP entity 2222 to get the call destination address from the proxy server and to
match number 2222.
2.
Configure Router B:
# Specify the SIP registrar and proxy server.

91
[RouterB-voice] sip
[RouterB-voice-sip] registrar 1 ip 192.168.2.3
[RouterB-voice-sip] proxy ip 192.168.2.3
[RouterB-voice-sip] quit
the POTS entity.
# Configure user name routerb, plaintext password 7890, and domain name server1.
[RouterB-voice-dial-entity2222] user routerB password simple 7890 realm server1
# Configure VoIP entity 2222 to get the call destination address from the proxy server and to
match number 1111.
[RouterB-voice-dial-entity1111] address sip proxy

# After the routers register phone numbers on the registrar, phone 1111 and phone 2222 can call each
other through the SIP proxy server.
# Executing the display voice sip register-status command can display the registration state for phones
1111 and 2222.
Configuring SIP calling through DNS

As shown in Figure 36, configure Router A to get the call destination address through DNS and configure
Router B to use a static call destination address so the phones 1111 and 2222 can call each other.
1.
Configure Router A:
92
# Map the IP address 192.168.2.2 to the host name cc.news.com.

[Sysname] ip host cc.news.com 192.168.2.2
# Configure the VoIP entity 2222 to call the destination through domain name cc.news.com and
to match number 2222.
[RouterA-voice-dial-entity2222] address sip dns cc.news.com port 5060
the POTS entity.
2.
Configure Router B:
the POTS entity.
# Configure the destination address 192.168.2.1 and number 1111 for the VoIP entity 1111.
[RouterB-voice-dial]entity 1111 voip

# Phone 1111 and phone 2222 can call each other.

As shown in Figure 37, configure Router A and Router B to use out-of-band signaling to transmit DTMF
tones so the phones 1111 and 2222 can call each other.
93
1.
Configure Router A:
# Configure out-of-band DTMF for VoIP entity 2222.

[RouterA-voice-dial-entity2222] outband sip
the POTS entity.
# Configure out-of-band DTMF for POTS entity 1111.

[RouterA-voice-dial-entity1111] outband sip
2.
Configure Router B:
# Configure out-of-band DTMF for VoIP entity 1111.

[RouterB-voice-dial-entity1111] outband sip
the POTS entity.
94

# Configure out-of-band DTMF for POTS entity 2222.

[RouterB-voice-dial-entity2222] outband sip
3.
Verify the configuration.

Phones 1111 and 2222 can call each other and the routers use out-of-band signaling to transmit
DTMF tones.
95
Configuring SIP trunk

This chapter describes how to configure SIP trunk.
Background
As shown in Figure 38, in a typical telephone network, a PBX forwards internal calls among enterprise
phones and forwards outbound calls over a PSTN trunk.
Figure 38 Typical telephone network
With the development of IP technology, many enterprises deploy SIP-based IP-PBX networks as shown
in Figure 39. All internal calls are placed using SIP, and external calls are still placed over a PSTN trunk.
The problem is that the enterprises have to maintain both the SIP network and PSTN trunk, which
increases the difficulty of network management.
Figure 39 SIP+PSTN network
As more enterprise IP-PBX networks run SIP and more Internet Telephone Service Providers (ITSPs) use SIP
to provide basic voice communication structures, enterprises urgently need a technology that can connect
the enterprise IP-PBX network to the ITSP over SIP. This technology is called SIP trunk. A typical network
diagram of SIP trunk is shown in Figure 40.
The SIP trunk function can be embedded into the voice gateway or the firewall deployed at the edge of
an enterprise private network. The device providing the SIP trunk function is called the SIP trunk device,
or the SIP trunk gateway.
96
Figure 40 All IP-based network
Features
SIP trunk has the following features:
The SIP trunk device and the ITSP only establish one secure and QoS guaranteed SIP trunk link. The
SIP trunk link can carry multiple concurrent calls, and the ITSP only authenticates the link instead of
each SIP call carried on this link.
The enterprise IP-PBX forwards internal calls. The SIP trunk device forwards outbound calls to the
ITSP, and then the devices in the ITSP forward the calls to the PSTN. Enterprises do not need to
maintain the PSTN trunk, which reduces the costs of hardware and maintenance.
The enterprise can connect to multiple ITSP by setting destination addresses.
With the SIP trunk device deployed, the entire network can use SIP to better support IP
communication services, such as voice, video conferencing, and instant messaging.
The SIP trunk device initiates a new call request to the ITSP on behalf of the user after receiving a call
request from the user, and both the user and the ITSP communicate only with the SIP trunk device.
The SIP trunk device forwards both SIP signaling messages and RTP media messages.
Typical applications
The SIP trunk device is deployed between the enterprise IP-PBX and the ITSP. All internal calls are placed
by the enterprise IP-PBX. All outbound calls are forwarded by the SIP trunk device to the ITSP through the
SIP trunk link. Figure 41 shows a typical SIP trunk technology.
97
Figure 41 SIP trunk network diagram
Protocols and standards
RFC 3261
RFC 3515
SIP connect Technical Recommendation v1.1
SIP trunk configuration task list

Tasks at a glance
(Required.) Enabling SIP-to-SIP calling
(Required.) Configuring a SIP trunk account
(Optional.) Enabling codec transparent transmission
(Optional.) Enabling media flow-around
(Optional.) Enabling DO-EO conversion
Enabling SIP-to-SIP calling

After you enable SIP-to-SIP calling, the device works as a SIP trunk device. HP does not recommend using
a SIP trunk device as a SIP UA.
To enable SIP-to-SIP calling:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enable SIP-to-SIP
calling.
allow-connections sip to sip
By default, SIP-to-SIP calling is disabled.
98
Configuring a SIP trunk account

A SIP trunk account contains a phone number, credentials, and realms assigned by the service provider.
SIP can send a REGISTER request for the phone number to up to six registrars specified with the registrar
command, and it uses the realm value in 401/407 responses from the registrars to identify the matching
credentials. You can configure up to 12 realm values for a phone number, and up to 128 SIP trunk
accounts on the device.
To configure a SIP trunk account:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
Enter SIP view.
sip
N/A
4.
Create an SIP trunk

account.
credentials number number

username username password
{ cipher | simple } password realm
realm
By default, no SIP trunk account is

created.
Specify a registrar.
registrar registrar-index { ip
ip-address | dns domain-name }
[ port port-number ] [ expires
seconds ] [ refresh-ratio
ratio-percentage ]
By default, no registrar is specified.
5.
Enabling codec transparent transmission

If the SIP trunk device does not support any codecs on the calling and called parties, you can enable
codec transparent transmission. The SIP trunk device transparently forwards codec capability sets
between the two parties without intervening codec negotiation.
To enable codec transparent transmission:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Enable codec transparent

transmission.
codec transparent
By default, codec
transparent transmission is
disabled.
Enabling media flow-around

This feature enables the SIP trunk device to directly forward media packets between SIP endpoints,
without changing the media address for the media packets.
99
To enable media flow-around:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
media flow-around
By default, the SIP trunk

device changes the media
address of a media packet
to its own address before
forwarding the media
packet.
5.
Enable media flow-around.
Enabling DO-EO conversion

An INVITE request with SDP Offer is an early offer, and an INVITE request without SDP Offer is a delayed
offer. Some ITSPs mandate early offer calls for charge security. To meet this requirement, enable delayed
offer to early offer (DO-EO) conversion on the SIP trunk device.
To enable DO-EO conversion:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
By default, DO-EO
conversion is disabled.
5.
Enable DO-EO conversion.
voice-class sip early-offer forced
Displaying and maintaining SIP trunk

Task
Command
Display SIP trunk account

registration status information.
100
This command does not

takes effect if codec
transparent transmission or
media flow-around is
enabled.
SIP trunk configuration examples

As shown in Figure 42, configure the SIP trunk device to forward calls between the private network and
public network.
1.
Configure Router A:
# Specify local number 2000 on POTS entity 2000.
# Configure the called number as 1000, and the destination IP address as 1.1.1.2 (the address of
the interface on the SIP trunk device) on VoIP entity 1000.
2.
Configure the SIP trunk device:

# Enable SIP-to-SIP calling.
<TG> system-view
[TG] voice-setup
[TG-voice] allow-connections sip to sip
[TG-voice] sip
# Create a SIP trunk account that contains number 2000, username 2000, password 2000, and
realm abc.
[TG-voice-sip] credentials number 2000 username 2000 password simple 2000 realm abc
# Specify the registrar 10.1.1.2 for the SIP trunk account.

101
[TG-voice-sip] registrar 1 ip 10.1.1.2

[TG-voice-sip] quit
# Configure the destination address as 2.1.1.2 for outbound calls from private phone 2000 to
public phone 1000.
[TG-voice] dial-program
[TG-voice-dial] entity 1 voip
[TG-voice-dial-entity1] address sip ip 2.1.1.2
[TG-voice-dial-entity1] match-template 1000
[TG-voice-dial-entity1] quit
# Configure the destination address as 1.1.1.1 for inbound calls from public phone 1000 to
private phone 2000.
[TG-voice-dial] entity 2 voip
[TG-voice-dial-entity2] address sip ip 1.1.1.1
[TG-voice-dial-entity2] match-template 2000
3.
Configure Router B:
# Specify local number 1000 on POTS entity 1000.
# Configure the called number as 2000, and the destination IP address as 2.1.1.1 (the address of
the interface on the SIP trunk device) on VoIP entity 2000.
4.
Verify the configuration:

a. Display SIP trunk account information on the SIP trunk device.
[TG-voice-dial-entity2] display voice sip register-status
Number
Entity
Registrar Server
Expires Status
------------------------------------------------------------------------------2000
10.1.1.2:5060
1802
Online
The output shows that the private phone number 2000 has registered with the server at
10.1.1.2.
b. Execute the display voice statistics call-active command to verify that all calls between the
private network and public network pass through the SIP trunk device.
c. On the SIP server of the ITSP, you can view only the interface address of the SIP trunk device,
which means that the SIP trunk device can hide the information of private users.
102
Configuring call services

Call services include call waiting, call hold, call forwarding, call transfer, and call backup.
Call waiting
When subscriber C calls subscriber A who is in a conversation with subscriber B, the call is not rejected.
Like a normal call, subscriber C hears ringback tones, and subscriber A hears call waiting tones. This is
call waiting.
Subscriber A can answer the new call by pressing the flash hook or hanging up to end the call with
subscriber B. In the former case, subscriber B is held. In the latter case, subscriber A is immediately
alerted and can pick up the phone to answer the call originated by subscriber C.
Call hold
When subscriber A in a conversation with subscriber B presses the flash hook, subscriber B is held (in
silent state or listening to waiting tones). The system plays dial tones to subscriber A, waiting for
subscriber A to initiate a new call. If subscriber A fails to dial within a period of time, the system stops
playing dial tones and subscriber A cannot initiate a new call. Subscriber A can resume the call with
subscriber B by pressing the flash hook again. This is call hold
Call forwarding
When the called party cannot answer a call, the called party notifies the calling party of the
forwarded-to number. The calling party re-initiates a call request to the new destination. This is call
forwarding. For example, subscriber A calls subscriber B who is busy, and subscriber A forwards the call
to subscriber C. Then, subscriber A and subscriber C establishes a call. Subscriber A is the initiator of call
forwarding, subscriber B is the recipient, and subscriber C is the final recipient.
The system supports the following types of call forwarding:
Call forwarding unconditionalForwards an incoming call to the predetermined destination,

regardless of the state of the called party.
Call forwarding busyForwards an incoming call to the predetermined destination when the
called party is busy.
Call forwarding no replyForwards an incoming call to the predetermined destination when the
called party provides no answer within a configurable period of time.
Call forwarding unavailableForwards an incoming call to the predetermined destination when

the voice interface of the called party is shut down by executing the shutdown command.
Call transfer
Subscriber A (the originator) and subscriber B (the recipient) are in a conversation. Subscriber A presses
the flash hook to place the call on hold. Subscriber A dials another number to originate a call to
103
subscriber C (the final recipient). After Subscriber A hangs up, the call between subscriber B and
subscriber C is established. This is call transfer.
There are three types of call transfer:
Call transfer without notificationThe originator hangs up before receiving ringback tones from
the final recipient.
Call transfer with early notificationThe originator hangs up after receiving ringback tones from
the final recipient and before speaking with the final recipient.
Call transfer with notificationThe originator hangs up after speaking with the final recipient.
During the call transfer process, if the recipient does not support call transfer or the final recipient is busy
or provides no reply, the initiator can re-establish the original call.
Call backup
The calling party might fail to receive a response after initiating a call to the called party through the IP
or PSTN network. In this case, if there is a backup link (PSTN link or VoIP link) to the called party, the
calling party can re-initiate a call to the called party over the backup link. This is call backup.
Call services configuration task list

Call waiting, call backup, call transfer are enabled on the device by default. No configurations are
required for them.
To configure call services, perform the following tasks:
Tasks at a glance
(Optional.) Configuring the call hold mode
(Optional.) Configuring call forwarding
Configuring the call hold mode

There are two call hold modes:
Silent mode (inactive)During call hold, the held party hears silence. This mode is configured on
the holding party (the initiator of call hold).
Unidirectional playing mode (sendonly)During call hold, the held party hears tones or music
played by a third-party music server. To play tones, you need to configure the sendonly keyword on
the holding party. To play music on hold by the third-party music server, you must configure the
sendonly moh-number string option on the SIP trunk device and configure the inactive keyword on
the holding party. For information about configuring a SIP trunk device, see "Configuring SIP trunk."
To configure the call hold mode:

Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice view.
voice-setup
N/A
104
Step
3.
Configure the call hold mode.
Command
Remarks
call-hold-format { inactive |
sendonly [ moh-number string ] }
The default is inactive.
Configuring call forwarding

Before configuring call forwarding, make sure there is a route from the initiator to the final recipient.
You need to correctly plan forwarded-to numbers to avoid incorrect numbers and circular calls.
A call can be forwarded five times at most.
To configure call forwarding:
Step
Command
Remarks
1.
Enter system view.
system-view
N/A
2.
Enter voice dial program view.
dial-program
N/A
3.

N/A
By default, call forwarding is
disabled.
4.
Configure call forwarding.
call-forwarding { no-reply |
on-busy | unavailable |
unconditional } number number
The unconditional, unavailable,

on-busy, and no-reply call
forwarding types can be
configured at the same time and
have a descending order of
priority.
Call services configuration examples

This section provides call services configuration examples.
Call waiting configuration example

As shown in Figure 43, configure call waiting so when a user places a call from Telephone C to
Telephone A that is in a conversation with Telephone B, the subscriber at Telephone C hears ringback
tones, and the subscriber at Telephone A hears call waiting tones.
105

Router A
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Before performing the following configuration, make sure Router A, Router B and Router C can reach
each other.
1.
Configure Router A:
# Create VoIP entity 2000, configure the destination IP address as 10.1.1.2, and configure the
called number as 2000.
# Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 1/0 to
the POTS entity.
2.
Configure Router B:
106
the POTS entity.
3.
Configure Router C:
the POTS entity.
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
[RouterC-voice-dial-entity3000] quit

Operation 1: When the subscriber at Telephone C dials 1000 to call Telephone A that is in a
conversation with Telephone B, the subscriber at Telephone C hears ringback tones, and the subscriber
at Telephone A hears call waiting tones. If then the subscriber at Telephone A hangs up, Telephone A
rings, and the subscriber at Telephone A can pick up the phone to start a conversation with Telephone
C.
Operation 2: When the subscriber at Telephone C dials 1000 to call Telephone A that is in a
conversation with Telephone B, the subscriber at Telephone A can press the flash hook to start a
conversation with Telephone C, and Telephone B is held. The subscriber at Telephone A can press the
flash hook again to continue the talk with Telephone B, and then Telephone C is held.
Call forwarding configuration example

As shown in Figure 44, configure call forwarding on Router B so when a user places a call from
Telephone A to Telephone B, Router B forwards the call to Telephone C when Telephone B is busy, and
Telephone A and Telephone C start a conversation.
107

Router A
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
each other.
1.
Configure Router A:
the POTS entity.
2.
Configure Router B:
number template as 3000.
the POTS entity.
# Configure call forwarding.

108

[RouterB-subscriber-line1/0] call-forwarding on-busy number 3000
3.
On Router C, configure the local number as 3000 for POTS entity 3000, and bind FXS interface
line 1/0 to the POTS entity.
Call transfer configuration example

As shown in Figure 45, configure call transfer to achieve the following purpose:
When Telephone A and Telephone B are in a conversation, the subscriber at Telephone A can press the
flash hook to put the call on hold, and then dials 3000 to call Telephone C. After the subscriber at
Telephone A hangs up, Telephone B and Telephone C start a conversation.
Router A
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
each other.
1.
Configure Router A:
109

the POTS entity.
2.
On Router B, configure the local number as 2000 for POTS entity 2000, and bind FXS interface
3.
On Router C, configure the local number as 3000 for POTS entity 3000, and bind FXS interface
Call backup configuration example

As shown in Figure 46, Router A and Router B are connected through two IP links. One of the two links
has a higher priority and serves as the primary link. Configure call backup so when the primary link goes
down, the backup link can take over.
1.
Configure Router A:
# Create VoIP entity 2000 with a priority of 1, configure the destination IP address as 10.1.1.2,
and configure the called number as 2000.
110

[RouterA-voice-dial-entity2000] priority 1
# Create VoIP entity 3000 with a priority of 2, configure the destination IP address as 20.1.1.2,
and configure the called number as 2000.
[RouterA-voice-dial-entity2000] priority 2
the POTS entity.
2.
On Router B, configure the local number as 2000 for POTS entity 2000, and bind FXS interface
111
Index
ABCDEFOPST
Configuring SIP UA registration,81
Configuring the call destination address for a VoIP

entity,85
Adjusting parameters for voice interfaces,18

Analog voice interface configuration examples,21
Configuring the call hold mode,104
Configuring the ISDN protocol,44
Background,96
Binding a digital voice interface to a POTS entity,45
Configuring the maximum number of total calls

allowed by a voice entity,66
Binding an FXS interface to an FXO interface,11
BSV interfaces,29
Dial program configuration examples,73
Digital voice interface configuration examples,45
Call services configuration examples,105
Displaying and maintaining analog voice

interfaces,21
Call services configuration task list,104
Displaying and maintaining digital voice interfaces,45
Configuration task list,61
Displaying and maintaining SIP,89
Displaying and maintaining SIP trunk,100
Displaying and maintaining voice entities,60
Configuring a BSV interface,32

Configuring a dial prefix,73
Configuring a number match mode,64
E&M interface,1
Configuring a number sending mode,71
E1 and T1 interfaces,29
Configuring a POTS entity,52
Enabling codec transparent transmission,99
Configuring a priority for a voice entity,71
Enabling DO-EO conversion,100
Configuring a SIP trunk account,99
Enabling media flow-around,99
Configuring a timeslot set,33
Enabling private line auto ring-down,64
Configuring a VoIP entity,56
Enabling SIP-to-SIP calling,98
Configuring an E&M interface,12
Configuring an FXO interface,6
Features,97
Configuring an FXS interface,3
FXO interface,1
Configuring basic functions,2
FXS interface,1
Configuring basic parameters for a T1 interface,31

Configuring basic parameters for an E1 interface,30
Configuring call forwarding,105
Overview,79
Configuring call progress tones,2
Overview,52
Configuring caller control,61
Configuring caller group control,63
Protocols and standards,98
Configuring DTMF,16
Configuring extended SIP functions,86
Configuring number substitution,66
SIP configuration task list,80
Configuring R2 signaling,34
112
SIP trunk configuration examples,101
SIP trunk configuration task list,98
Typical applications,97
SIP UA configuration examples,89
113

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H3C MSR Series Routers

Voice Configuration Guide(V7)

Hangzhou H3C Technologies Co., Ltd.

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About the H3C MSR documentation set

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About the H3C MSR documentation set

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Configuring a TDM clock source 30

Configuring voice entities 52

Configuring number substitution 66

Displaying and maintaining SIP trunk 100

Configuring call services 103

Configuring analog voice interfaces

Configuration task list

(Optional.) Binding an FXS interface to an FXO interface

Configuring the cable type

(Optional.) Configuring DTMF

Configuring DTMF tone sending

Configuring basic functions

Enter system view.

Enter voice interface view.

Configure a description for

By default, the description of a

Restore the default settings for

Bring up the voice interface.

By default, a voice interface is up.

Configuring call progress tones

Enter system view.

Enter voice view.

cptone country-type locale

Customize call progress tone

Configure call progress tones.

Configure the amplitude value

cptone tone-type { all | busy-tone

By default, the call progress tones

The default is 1000 for busy tone

Configuring an FXS interface

Single-data-message formatContains date, time, and calling number.

Multiple-data-message formatContains date, time, calling number, and calling name.

The called telephone displays the calling number differently as follows:

The PBX and called telephones must support CID.

Enter system view.

Enter FXS interface view.

(Optional.) Configure the

By default, no calling name is configured.

Enable the FXS interface

The calling name can be sent only in the