Configuraticonfig Voipon Guide - Voice (V200R002C01 - 01)

Huawei AR2200 Series Enterprise Routers
V200R002C01
Configuration Guide - Voice

Issue
01
Date
2012-04-20
HUAWEI TECHNOLOGIES CO., LTD.
Copyright Huawei Technologies Co., Ltd. 2012. All rights reserved.

No part of this document may be reproduced or transmitted in any form or by any means without prior written
consent of Huawei Technologies Co., Ltd.
Trademarks and Permissions

and other Huawei trademarks are trademarks of Huawei Technologies Co., Ltd.
All other trademarks and trade names mentioned in this document are the property of their respective holders.
Notice
The purchased products, services and features are stipulated by the contract made between Huawei and the
customer. All or part of the products, services and features described in this document may not be within the
purchase scope or the usage scope. Unless otherwise specified in the contract, all statements, information,
and recommendations in this document are provided "AS IS" without warranties, guarantees or representations
of any kind, either express or implied.
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.
Huawei Technologies Co., Ltd.

Address:
Huawei Industrial Base

Bantian, Longgang
Shenzhen 518129
People's Republic of China
Website:
http://www.huawei.com
Email:
support@huawei.com
Issue 01 (2012-04-20)
Huawei Proprietary and Confidential

Copyright Huawei Technologies Co., Ltd.

About This Document
About This Document

Intended Audience
This document provides the basic concepts, configuration procedures, and configuration
examples in different application scenarios of the voice feature.
This document describes how to configure the voice feature.
This document is intended for:
l
Data configuration engineers
Commissioning engineers
Network monitoring engineers
System maintenance engineers
Symbol Conventions
The symbols that may be found in this document are defined as follows.
Symbol
Description
DANGER
WARNING
CAUTION
Issue 01 (2012-04-20)
Indicates a hazard with a high level of risk, which if not

avoided, will result in death or serious injury.
Indicates a hazard with a medium or low level of risk, which
if not avoided, could result in minor or moderate injury.
Indicates a potentially hazardous situation, which if not
avoided, could result in equipment damage, data loss,
performance degradation, or unexpected results.
TIP
Indicates a tip that may help you solve a problem or save

time.
NOTE
Provides additional information to emphasize or supplement

important points of the main text.

ii

About This Document
Command Conventions
The command conventions that may be found in this document are defined as follows.
Convention
Description
Boldface
The keywords of a command line are in boldface.
Italic
Command arguments are in italics.
[]
Items (keywords or arguments) in brackets [ ] are optional.
{ x | y | ... }
Optional items are grouped in braces and separated by vertical

bars. One item is selected.
[ x | y | ... ]
Optional items are grouped in brackets and separated by vertical

bars. One item is selected or no item is selected.
{ x | y | ... }*
Optional items are grouped in braces and separated by vertical

bars. A minimum of one item or a maximum of all items can be
selected.
[ x | y | ... ]*
Optional items are grouped in brackets and separated by vertical

bars. Several items or no item can be selected.
&<1-n>
The parameter before the & sign can be repeated 1 to n times.
A line starting with the # sign is comments.
Interface Numbering Conventions

Interface numbers used in this manual are examples. In device configuration, use the existing
interface numbers on devices.
Change History
Changes between document issues are cumulative. Therefore, the latest document issue contains
all updates made in previous issues.
Changes in Issue 01 (2012-04-20)

Initial commercial release.
Issue 01 (2012-04-20)

iii

Contents
Contents
About This Document.....................................................................................................................ii
1 SIPAG Configuration...................................................................................................................1
1.1 SIP AG Overview...............................................................................................................................................2
1.2 SIP AG Features Supported by the AR2200......................................................................................................2
1.3 Configuring the AR2200 to Work in SIP AG Mode..........................................................................................6
1.4 Configuring a SIP AG Interface.........................................................................................................................7
1.4.1 Establishing the Configuration Task.........................................................................................................7
1.4.2 Configuring Media and Signaling IP Address Pools.................................................................................8
1.4.3 Setting Parameters on the SIP AG Interface.............................................................................................8
1.4.4 Checking the Configuration.....................................................................................................................15
1.5 (Optional) Setting Parameters for Voice Interfaces..........................................................................................16
1.5.1 Setting Parameters for an FXS Interface.................................................................................................16
1.5.2 Setting parameters for a BRA Interface...................................................................................................18
1.5.3 Setting Parameters for a PRA Interface...................................................................................................19
1.6 (Optional) Setting Voice Parameters on a SIP AG...........................................................................................21
1.6.1 Setting System Parameters......................................................................................................................21
1.6.2 Setting SIP Stack Parameters..................................................................................................................30
1.6.3 Setting DSP Parameters...........................................................................................................................32
1.7 Configuring a SIPAG User...............................................................................................................................43
1.7.1 Establishing the Configuration Task.......................................................................................................44
1.7.2 Setting Parameters for a SIP AG User.....................................................................................................44
1.7.3 Configuring a SIP AG User Group..........................................................................................................46
1.7.4 (Optional) Configuring Functions or Services in a SIP Service Data Profile.........................................47
1.7.5 Checking the Configuration.....................................................................................................................49
1.8 (Optional) Configuring the BEST Service.......................................................................................................50
1.9 Resetting a SIP AG...........................................................................................................................................52
1.10 Maintaining a SIP AG....................................................................................................................................53
1.10.1 Clearing SIP AG Statistics....................................................................................................................53
1.11 Configuration Examples.................................................................................................................................53
1.11.1 Example for Configuring a SIP AG......................................................................................................53
1.11.2 Using a PRA Trunk to Connect the SIP AG to a PBX..........................................................................57
1.11.3 Example for Configuring the BEST Service.........................................................................................61
2 PBX Configuration......................................................................................................................66
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iv

Contents
2.1 PBX Overview..................................................................................................................................................68

2.2 PBX Features Supported by the AR2200.........................................................................................................69
2.3 PBX Configuration Procedures........................................................................................................................74
2.4 (Optional) Setting Parameters for Voice Interfaces..........................................................................................78
2.4.1 Setting Parameters for an FXS Interface.................................................................................................78
2.4.2 Setting Parameters for an FXO Interface.................................................................................................80
2.4.3 Setting parameters for a BRA Interface...................................................................................................81
2.4.4 Setting Parameters for a PRA Interface...................................................................................................83
2.5 Setting PBX Parameters...................................................................................................................................84
2.5.1 Configuring the AR2200 to Work in PBX Mode....................................................................................84
2.5.2 Configuring Media and Signaling IP Address Pools...............................................................................85
2.5.3 (Optional) Setting System Parameters.....................................................................................................86
2.5.4 (Optional) Setting SIP Stack Parameters.................................................................................................95
2.5.5 (Optional) Setting DSP Parameters.........................................................................................................98
2.6 Configuring an Enterprise..............................................................................................................................108
2.7 Configuring a Call Prefix...............................................................................................................................110
2.8 Configuring a PBX User.................................................................................................................................112
2.9 (Optional) Configuring a SIP Server..............................................................................................................113
2.10 (Optional)Configuring SBC Proxy Function................................................................................................115
2.10.1 Configuring the SBC Proxy Function (Public IP Address for the SIP Server and Private IP Addresses
for SIPUEs).....................................................................................................................................................116
2.10.2 Configuring the SBC Proxy Function (Private IP Address for the SIP Server and Public IP Addresses
for SIP UEs)....................................................................................................................................................118
2.10.3 Configuring the SBC Proxy Function (Private IP Address for the SIP Server and Private IP Addresses
for SIP UEs)....................................................................................................................................................121
2.11 (Optional)Configuring a CDR Server...........................................................................................................124
2.12 Configuring a Trunk Group..........................................................................................................................126
2.12.1 Establishing the Configuration Task...................................................................................................126
2.12.2 Configuring a Trunk Group.................................................................................................................127
2.12.3 Binding Trunks in a Trunk Group.......................................................................................................139
2.12.4 Resetting a SIP Trunk Group...............................................................................................................141
2.12.5 Checking the Configuration.................................................................................................................142
2.13 Configuring a Call Route..............................................................................................................................142
2.14 (Optional)Configuring Individualized PBX Services...................................................................................144
2.14.1 Configuring the Abbreviated Dialing Service.....................................................................................144
2.14.2 Configuring the Call-out Restriction Service......................................................................................145
2.14.3 Configuring the Call Forwarding Service...........................................................................................147
2.14.4 Configuring the Number Barring Service...........................................................................................148
2.14.5 Configuring the DND Service.............................................................................................................149
2.14.6 Configuring the RAC Service..............................................................................................................151
2.14.7 Configuring the Remote Office Service..............................................................................................152
2.14.8 Configuring the Secretary Service.......................................................................................................153
2.14.9 Configuring the Wake-up Service.......................................................................................................154
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Contents
2.14.10 Configuring the RBT Service............................................................................................................155

2.14.11 Configuring the SCR Service............................................................................................................157
2.14.12 Configuring the SCA Service............................................................................................................158
2.15 (Optional) Configuring Group Services on the PBX....................................................................................159
2.15.1 Configuring the Call Interception Service...........................................................................................159
2.15.2 Configuring the Distinctive Ringing Service......................................................................................161
2.15.3 Configuring the Enterprise RBT Service.............................................................................................162
2.15.4 Configure the IVR Service..................................................................................................................164
2.15.5 Configuring the Number Change Service...........................................................................................170
2.15.6 Configuring the Pre-routing Number Change Service........................................................................171
2.15.7 Configuring the Post-routing Number Change Service.......................................................................173
2.15.8 Configuring the PBX Line Selection Service......................................................................................175
2.15.9 Configuring the Co-Group Pickup Service.........................................................................................176
2.15.10 Configuring Simultaneous Ringing...................................................................................................177
2.15.11 Configuring Sequential Ringing........................................................................................................179
2.15.12 Configuring the One Number Link You Service...............................................................................180
2.15.13 Configuring the Blacklist or Whitelist Function...............................................................................181
2.15.14 Configuring the DISA Service..........................................................................................................183
2.15.15 Configuring the Immediate Conference Service...............................................................................186
2.16 Configuration Examples...............................................................................................................................188
2.16.1 Example for Configuring Voice Services for a Small- or Medium-sized Enterprise..........................188
2.16.2 Example for Configuring the AR to Implement Communication for Different Enterprises...............193
2.16.3 Example for Using an H.323 Trunk Group in Peer Mode to Configure Calls Between the Headquarters
and Branch......................................................................................................................................................199
2.16.4 Example for Using an H.323 Trunk Group in Registration Mode to Configure Calls Between the
Headquarters and Branch................................................................................................................................208
2.16.5 Example for Using the SIP Trunk Group to Configure Calls Between the Headquarters and Branch
........................................................................................................................................................................213
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vi

1 SIPAG Configuration
SIPAG Configuration
About This Chapter

This chapter describes the concepts and configuration of SIPAG, and provides a configuration
example.
1.1 SIP AG Overview
This section describes the SIP AG in VoIP applications.
1.2 SIP AG Features Supported by the AR2200
This section describes SIP AG features supported by the AR2200.
1.3 Configuring the AR2200 to Work in SIP AG Mode
Before configuring SIP AG service features, configure the AR2200 to work in SIP AG mode.
1.4 Configuring a SIP AG Interface
This section describes how to configure a SIP AG interface.
1.5 (Optional) Setting Parameters for Voice Interfaces
This section describes how to set parameters for voice interfaces.
1.6 (Optional) Setting Voice Parameters on a SIP AG
This section describes how to set voice parameters on a SIP AG.
1.7 Configuring a SIPAG User
This section describes how to configure a SIPAG user.
1.8 (Optional) Configuring the BEST Service
The branch exchange for survivable telephony (BEST) service enables SIP AGs in enterprise
branches to manage local calls when communication with the headquarters voice server fails.
1.9 Resetting a SIP AG
This section describes how to reset a SIP AG.
1.10 Maintaining a SIP AG
This section describes how to maintain a SIP AG.
1.11 Configuration Examples
This section provides a SIPAG configuration example.
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1.1 SIP AG Overview

This section describes the SIP AG in VoIP applications.
VoIP
On the Public Switched Telephone (PSTN), telephone lines are occupied exclusively. The fees
for toll calls are high.
To reduce data and voice fees and meet service requirements, voice over IP (VoIP) is used.
VoIP transmits voice services over the IP network. In VoIP, a voice gateway encapsulates voice
signals into frames and transmits them as IP packets. Currently, IP phone gateways implement
communication between the PSTN network and IP network. As PC-to-phone, phone-to-PC, and
phone-to-phone technologies have developed, the voice quality is improved greatly. VoIP can
meet commercial requirements.
Voice Gateway
The IP network is a packet switched network. The transmission costs on the IP network are lower
than those on the PSTN; therefore, VoIP will take place traditional voice phones gradually.
Replacing all phone networks with VoIP will take high costs. The step-by-step migration
solution is recommended.
In most enterprise voice solutions, phone networks of branches are reserved and the IP network
is used between branches.
In the step-by-step migration solution, a voice gateway is required to connect the two types of
networks. A router is often used as the voice gateway.
IMS
The IP Multimedia Core Network Subsystem (IMS) is an architectural framework for providing
IP multimedia services, including audio, video, text, and instant messages. It was designed by
the wireless standards body 3rd Generation Partnership Project (3GPP) in Release 5.
SIP
The Session Initiation Protocol (SIP) is a text-based signaling protocol. SIP messages are
classified into request and response messages. SIP can be used for creating, modifying, and
terminating two-party or multiparty sessions. SIP can be used for multimedia conferences,
remote education, and Internet calls.
SIPAG
A SIP access gateway (SIP AG) is a voice gateway that exchanges SIP signals with other devices
between the PSTN/ISDN and IP network. It can implement VoIP functions.
1.2 SIP AG Features Supported by the AR2200

This section describes SIP AG features supported by the AR2200.
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Using a SIP AG as a Voice Gateway

As shown in Figure 1-1, a SIP AG functions as an IP voice gateway to implement
communication between the IP network and the PSTN or ISDN. The usage scenarios are as
follows:
l
A user connects to the SIP AG through the PSTN. The SIP AG converts analog signals into
digital signals and compresses and packetizes the digital signals so that the voice packets
can be transmitted on the IP network.
Voice packets are transmitted to the SIP AG through the IP network. The IP voice gateway
of the called party converts the voice packets into analog voice signals and transmits them
to the terminal of the called party through the PSTN.
NOTE
l The AR2200 series routers support the voice features only after a DSP module is installed.
l To provide voice services for POTS users, 4FXS1FXO board is required.
l To provide voice services for ISDN users, 2BST board is required.
Figure 1-1 SIP AG as a voice gateway
IMS/IP
SIPAG
POTS
Eth1/0/0
Modem
FAX
SIP AG call flow

The SIP AG call flow is as follows:
1.
A user picks up the phone and the SIP AG detects offhook signals.
2.
The DSP module of the SIP AG generates a dial tone.
3.
The user hears the dial tone played by the session application of the SIP AG and starts to
dial up.
4.
The session application of the SIP AG collects the number dialed by the user.
5.
The session application of the SIP AG matches the number with preconfigured called
number profiles.
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6.
When the voice gateway of the calling party successfully matches the number with a
preconfigured called number profile, the number is mapped to the voice gateway of the
called party.
7.
The SIP AG initiates a call over the IMS and establishes a logical channel for each call.
Then the channel is used to send and receive voice data.
8.
The IMS searches for the destination phone and initiates a call.
9.
After the called party picks up the phone, a call is set up. After the calling party or called
party hangs up the phone, the call is ended.
Voice Services Supported by the SIP AG

The SIP AG supports the basic voice service and developed services such as the three-party
service and call waiting service. Among these services, some voice services need to be
configured on the SIP AG.
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Service Type
Introduction
Configure
d on the
SIP AG or
Not
Basic voice service
The basic voice service is the basic call connection

function, including intra-office calls, local calls,
national toll calls, international toll calls, and transit
calls.
Yes
Three-party service
The third-party service allows a calling party or called

party in a conversation to call a third party without
ending the current conversation. Then the calling party
or original called party can implement a three-party
conversation or talk to the other two parties.
Yes
Call waiting
service
When user A is talking with user B over the phone and

at this moment user C is calling user A, user A hears a
call waiting tone, indicating that there is a call waiting
for user A.
Yes
MWI service
The message waiting indicator (MWI) service allows a

user to read unread messages or leave messages. When
the called user is busy, the MWI is on, indicating that
there are leave messages.
Yes
Malicious call
identification
(MCID) service
The user that registers the MCID service with the carrier
can query the phone number of the attacker that initiates
malicious calls after performing relevant operations.
Yes
Call transfer
service
The call transfer service allows the called party to

transfer an incoming call to a third party by pressing the
hookflash so that the calling party establishes a
connection with a new called party.
Yes
Call conference
service
The call conference service allows more than three

parties to communicate together.
Yes


Service Type
Introduction
Configure
d on the
SIP AG or
Not
Calling line
identification
presentation
(CLIP) service
The CLIP service displays the calling number in onhook

state or offhook state (for call waiting). The displayed
information includes the phone number, name, date, and
time.
No
Calling line
identification
restriction (CLIR)
service
The CLIR service displays the phone number of a

calling party on the terminal of a called party.
No
Distinctive ringing
service
The distinctive ringing service plays different ring tones

for incoming calls.
No
Differentiated
ringback tone
service
The differentiated ringback tone service enables the SIP

AG to play different ring tones for ringback tones.
No
Advice of charge
(AoC) service
The AoC service enables the SIP AG to display the

charge rate, fee notification during a call, and the total
fee of the call.
No
Polarity reversal
charging service
The polarity reversal charging service notifies the

accounting terminal of the starting point and stop point
of accounting using polarity reversal when users are in
a conversation or end the conversation.
No
Polarity reversal
pulse charging
service
When users are in a conversation or end the

conversation, a polarity reversal pulse is generated on
an interface, notifying the accounting terminal of the
starting point and stop point of accounting.
No
Urgent call process
If the SIP AG finds an urgent call, the SIP AG inserts

the urgent call flag into the SIP message.
No
Completion of
Calls to Busy
Subscriber (CCBS)
service
When the called number is busy, the CCBS service

enables the SIP AG to monitor the called party status.
When the called party is idle, the SIP AG notifies the
calling party and determines whether to make a call
according to the status of the calling and called parties.
No
Multiple MSN
numbers on a
POTS interface
Multiple MSN numbers can be configured on a POTS

interface.
No
Hotline service
l Instant hotline service: After a user picks up a phone,

the SIP AG dials the hotline number for the user.
No
l No dialing within a long time after picking up the

phone: If a user does not dial any number within the
specified period of time after picking up the phone,
the SIP AG dials the hotline number for the user.
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Service Type
Introduction
Configure
d on the
SIP AG or
Not
Anonymous call
service
The anonymous call service enables the called party not

to view information about incoming calls.
No
License Support
The SIPAG function is used with a license. To use the SIPAG function, apply for and purchase
the following license from the Huawei local office:
AR2200 value-added service package for voice services
To use the BEST function, you must purchase the CM&BEST License in addition to the precedng
licenses.
1.3 Configuring the AR2200 to Work in SIP AG Mode

Before configuring SIP AG service features, configure the AR2200 to work in SIP AG mode.
Applicable Environment
You can configure the AR2200 to work in SIP AG or PBX mode. Before configuring SIP AG
service features, configure the AR2200 to work in SIP AG mode. You can run the display voice
service-mode command to view the working mode of the AR2200. If the AR2200 works in
PBX mode, delete the PBX configurations and configure the AR2200 to work in SIP AG mode.
If the AR2200 works in SIP AG mode, skip this configuration.
Pre-configuration Tasks
Before configuring the AR2200 to work in SIP AG mode, complete the following task:
l
Configuring IP addresses for interfaces and routing protocols to ensure connectivity
Procedure
Step 1 Run:
system-view
The system view is displayed.

Step 2 Run:
voice
The voice view is displayed.

Step 3 Run:
service-mode sipag
The AR2200 is configured to work in SIP AG mode.

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Step 4 Run:
quit
Return to the system view.

Step 5 Run:
quit
Return to the user view.

Step 6 Run:
save
The configuration is saved.

Step 7 Run:
reboot
The AR2200 is restarted.

NOTE
After the AR2200 is configured to work in SIP AG mode, restart the AR2200 to make the configuration
take effect.
----End
Checking the Configuration

After the configuration is complete, the AR2200 works in SIP AG mode.
Run the display voice service-mode command to check the working mode of the AR2200.
1.4 Configuring a SIP AG Interface

This section describes how to configure a SIP AG interface.
1.4.1 Establishing the Configuration Task

Before configuring a SIP AG interface, familiarize yourself with the applicable environment,
complete the pre-configuration tasks, and obtain the data required for the configuration. This
will help you complete the configuration task quickly and accurately.
To allow the SIP AG and IMS network to exchange media and signaling streams, the media and
signaling IP addresses, signaling port number, and transmission protocol need to be configured.
Before configuring a SIP AG interface, complete the following task:
l
Configuring the AR2200 to work in SIP AG mode
Data Preparation
To configure a SIP AG interface, you need the following data.
Issue 01 (2012-04-20)


No.
Data
Media and signaling IP addresses
SIP AG interface number, media and signaling IP addresses, signaling port number,
IP addresses and port numbers of primary and secondary proxy servers, transmission
protocol, and home domain name
1.4.2 Configuring Media and Signaling IP Address Pools

Media and signaling IP address pools provide media and signaling IP addresses for SIP AG
interfaces.
Context
A SIP AG interface must obtain media and signaling IP addresses from media and signaling IP
address pools respectively. The signaling IP address pool stores IP addresses of SIP AG
interfaces and the media IP address pool stores IP addresses of media streams. Media and
signaling IP address pools can contain the same IP addresses.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
voip-address media interface interface-type interface-number { ip-address |
dynamic }
A media IP address pool is configured.

Step 4 Run:
voip-address signalling interface interface-type interface-number { ip-address |
dynamic }
A signaling IP address pool is configured.

When configuring media and signaling IP address pools:
l To specify the ip-address parameter, ensure that an IP address has been configured for the
specified interface.
l To specify the dynamic parameter, ensure that an IP address has been dynamically assigned
to the specified interface.
----End
1.4.3 Setting Parameters on the SIP AG Interface

This section describes how to set parameters on a SIP AG interface.
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Context
To allow the SIP AG and IMS network to exchange media and signaling streams, set the media
and signaling IP addresses, signaling port number, and transmission protocol.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 (Optional) Run:
local-digitmap name-value { append | type { normal | emergency | scc | directcentrex | second-centrex } }body-value
The digitmap is configured.

Step 4 Run:
sipag sipag-interface-id
A SIP AG interface is created and the SIP AG interface view is displayed.

Step 5 Set mandatory parameters on the SIP AG interface.
Parameter
Command
Remarks
Media IP address
media-addr { media-ip | addr-name

addr-name }
The media IP address

of the SIP AG interface
must be obtained from
the media IP address
pool.
Signaling IP address and

signaling port number
signalling-addr { signal-ip | addrname signal-addr-name } signal-port
The signaling IP
address of the SIP AG
interface must be
obtained from the
signaling IP address
pool.
By default, no
signaling port number
is configured for a SIP
AG interface.
Paramet
ers for
the
primary
proxy
server
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Static IP
address
primary-proxy-addr static static-ipaddress static-ip-address-port

By default, no static IP
address is configured
for the primary proxy
server.

Parameter
Command
Remarks
DNS-A
domain name
and port
number
primary-proxy-addr dns-a dns-adomain-name dns-a-port-number
By default, no DNS-A
domain name or port
number is configured
for the primary proxy
server.
DNS-NAPTR
domain name
primary-proxy-addr dns-naptr dnsnaptr-domain-name
By default, no DNSNAPTR domain name

is configured for the
primary proxy server.
DNS-SRV
domain name
primary-proxy-addr dns-srv dnssrv-domain-name
By default, no DNSSRV domain name is

configured for the
primary proxy server.
Transmission protocol
transfer transfer-protocol
By default, a SIP AG
interface uses the User
Datagram Protocol
(UDP) protocol as the
transmission protocol.
Home domain name
home-domain home-domain-value
By default, no home
domain name is
configured for a SIP
AG interface.
Step 6 Set optional parameters on the SIP AG interface.
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Parameter
Command
Remarks
Description
description description
By default, no
description is
AG interface.
Profile index
profile profile-index
By default, the profile

index of a SIP AG
interface is 1.
Domain name
ag-domain ag-domain-name
By default, no domain
name is configured for
a SIP AG interface.
Conference factory URI
conference-factory-uri uri
By default, no
conference factory
URI is configured for a
SIP AG interface.
Authent
ication
auth mode mode-value
By default, the
authentication mode is
user.
Authenticatio
n mode

10

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Parameter
Command
Remarks
paramet
ers
auth username auth-username

password { cipher | ha1 { cipher |
simple password-value1 } | simple
password value2 }
By default, no
authentication user
name or password is
AG user.
Timer value
digitmap-timer { long long-timervalue | short short-timer-value | start

start-timer-value }
By default, the value of

a digitmap long timer is
20s, the value of a
digitmap short timer is
4s, and the value of a
digitmap start timer is
20s.
DTMF parameters
dtmf-transmission-mode
{ thoroughly | erase }
By default, the DTMF

transmission mode is
transparent
transmission.
Fax/
Modem
paramet
ers
Fax/Modem
codec
negotiation
mode
fax-modem common negotiationmode { negotiate | self-switch }
By default, the fax/

modem codec
negotiation mode is
negotiation.
Packetization
interval in
transparent
transmission
mode
fax-modem common rtp-interval

{ 5ms | 10ms | 20ms | 30ms }
By default, the
packetization interval
in transparent
transmission mode is
10 ms.
VBD attribute
type
fax-modem common vbd-attributetype { chinatelecom |

extchinatelecom | ietf | v152 }
By default, the VBD

attribute type is V.152.
VBD codec
mode
fax-modem common vbd-codec

{ g711a | g711u }
By default, the VBD

codec mode is A-law
algorithm in G.711.
VBD payload
type
fax-modem common vbd-pt-type

{ dynamic | static }
By default, the VBD

payload type is static.
Fax
transmission
mode
fax-modem fax transmission-mode

{ thoroughly | t38 }
By default, the fax

transmission mode of a
SIP AG interface is
transparent
transmission.
Modem
transmission
mode
fax-modem modem transmissionmode { thoroughly | relay }
By default, the modem

transmission mode of a
SIP AG interface is
transparent
transmission.
Authenticatio
n user name
and password

11

Issue 01 (2012-04-20)
Parameter
Command
Remarks
Paramet
ers of
the
seconda
ry proxy
server
Static IP
address and
port number
secondary-proxy-addr static staticip-address static-ip-address-port
By default, no static IP
address or port number
secondary proxy
server.
DNS-A
domain name
and port
number
secondary-proxy-addr dns-a dns-adomain-name dns-a-port-number
By default, no DNS-A
domain name or port
number is configured
for the secondary proxy
server.
DNS-NAPTR
domain name
secondary-proxy-addr dns-naptr
dns-naptr-domain-name
By default, no DNSNAPTR domain name

secondary proxy
server.
DNS-SRV
domain name
secondary-proxy-addr dns-srv dnssrv-domain-name
By default, no DNSSRV domain name is

configured for the
secondary proxy
server.
Mode used for sending

DHCP Request messages
proxy-dhcp-option option-value
By default, the mode

used for sending DHCP
Request messages is
none.
Software parameters
ag number-parameter name value
Different software
parameters may use
different value ranges.
For details, see ag
number-parameter.
String parameters
ag string-parameter strpara-name
strpara-value
For details about string

parameters, see ag
string-parameter.
RFC
2833
transmis
sion
RFC 2833
DTMF
transmission
nte-dtmf { enable | disable }
By default, RFC 2833

DTMF transmission is
enabled.
RFC 2833 fax

and modem
transmission
nte-fax-modem { enable | disable }

fax and modem
transmission is
enabled.
RFC 2833
hookflash
transmission
nte-flash-hook { enable | disable }

hookflash transmission
is disabled.

12

Parameter
Mode in
which RFC
2833
transmission
is started
based on
negotiation
Proxy
function
s
Issue 01 (2012-04-20)
Command
Remarks
nte-negotiation-mode mode-value

redundancy
transmission is started
based on negotiation in
passive-nego mode.
Mode used by proxy-detect-mode { option | probe |

a SIP AG to
register }
detect a proxy
server
uses the probe mode to
detect a proxy server.
Mode in
which a SIP
AG is dual
homed to
proxy servers
proxy-dual-home { disable | manualswitchover | auto-switchover }
supports dual homing
but does not support
automatic switchover.
Mode used by
a SIP AG to
update the
proxy server
address
proxy-refresh-mode { no-switch |
defer | immediate }
updates the proxy
server IP address after
a delay (defer mode).
Registration of a SIP AG
with the specified proxy
server
proxy-switchover { primary |
secondary }
To maintain or
troubleshoot the
current proxy server,
run this command.
RFC
2198
redunda
ncy
transmis
sion
Mode in
which RFC
2198
redundancy
transmission
is started
redundancy-negotiation-mode
negotiation-mode

redundancy
in no-initiative-start
mode.
Transmission
of RFC 2833
audio data
using RFC
2198
redundancy
transmission
redundancy-nte { enable | disable }
By default, a SIP AG is
disabled from using
RFC 2198 redundancy
transmission to
transmit the RFC 2833
audio data.
Mode in
which RFC
2198
redundancy
transmission
is started
redundancy-start-mode mode-value

redundancy
in ordinary mode.

13

Parameter
Issue 01 (2012-04-20)
Command
Remarks
VBD
transmission
using RFC
2198
redundancy
transmission
redundancy-vbd { enable | disable }
By default, a SIP AG is
enabled to transmit
VBD using RFC 2198
redundancy
transmission.
Voice
transmission
using RFC
2198
redundancy
transmission
redundancy-voice { enable |
disable }
does not use RFC 2198
redundancy
transmission to
transmit voice.
Registrar URI
register-server-uri uri
By default, no registrar
URI is configured for a
SIP AG.
Ringing parameters
ring-mode index ringmode-name

[ cadence-ring cadencering-value |
initial-ring initialring-value ] *
By default, ringing
parameters of a SIP AG
is empty.
SCTP stack parameters
sctp { assoc-maximumretransmission assoc-max-retrans |

checksum-arithmetic checksum-arith
| heartbeat-interval hb-interval |
high-congestion-level highcongestion-level-value | in-streamnumber in-stream-num | lowcongestion-level low-congestionlevel-value | maximum-initretransmission max-init-retrans | nocongestion-level no-congestion-levelvalue | out-stream-number outstream-num | path-maximumretransmission path-max-retrans |
rto-alpha rto-alpha-value | rto-beta
rto-beta-value | rto-init rto-init-value |
rto-maximum rto-max | valid-cookielife valid-cookie-life-value | }
After SCTP stack

parameters are set,
reset the SIP AG to
make SCTP stack
parameters take effect.
SDP negotiation mode
sdp-negotiation-mode { local |
remote }
By default, the remote

priority mode is used
for SDP negotiation.
Index of the service logic

used by a SIP AG
service-logic service-logic-index
By default, the service

logic index of a SIP AG
is 0.

14

Parameter
Command
Remarks
Mapping between a call

scenario and a status code
status-code-mapping scene statuscode
After configuring the

status code of a call
scenario, reset the SIP
AG to make the
configuration take
effect.
MWI
subscrip
tion
function
MWI
subscription
function
subscribe mwi { enable | disable }
By default, the MWI

subscription function is
disabled.
Register
status
subscription
function
subscribe reg { enable | disable }
By default, the register

status subscription
function is disabled.
UA profile
subscription
function
subscribe ua-profile { enable |

disable }
By default, the UA
profile subscription
function is enabled on a
SIP AG.
Codec priority, codec

value, and packetization
interval
user-defined-profile codec-priority
pri-value codec-value pt-value
If you change the

priority of a codec
value, priorities of
other codec values are
also changed.
Software parameter with

a specified sequence
number
user-defined-profile numberparameter strpara-no strpara-value
Priority of a service type
user-defined-profile service-priority
pri-value srvid-value
If you change the

priority of a service
type, priorities of other
service types are also
changed.
----End
1.4.4 Checking the Configuration

Procedure
l
Run the display voice sipag [ sipag-interface-id { running | config } ] command to check
the configuration of SIP AG parameters.
Run the display voice voip-address command to check the signaling and media IP
addresses pool.
----End
Issue 01 (2012-04-20)

15


1.5.1 Setting Parameters for an FXS Interface

This section describes how to set parameters for an FXS interface.
An FXS interface connects to a POTS phone. To achieve high transmission efficiency on an
FXS interface, properly set parameters for the FXS interface on the AR2200, including physical
attributes, electrical attributes, and KC attributes.
Before setting parameters for an FXS interface, complete the following task:
l
Configuring the AR2200 to work in SIPAG mode
Ensuring that the voice 4FXS1FXO board is working properly
Data Preparation
To set parameters for an FXS interface, you need the following data.
No.
Data
Polarity reversal pulse level width, polarity reversal mode, and dialing mode
Current, impedance, send gain, and receive gain
High-level pulse width, low-level pulse width, KC accounting mode, and voltage
operating
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
port fxs slotid/subcardid/portid
The FXS interface view is displayed.

Issue 01 (2012-04-20)

16

Step 4 Set physical attributes for the FXS interface.

l Run:
reverse-pole-pulse enable
The polarity reversal pulse function is enabled.

l Run:
reverse-pole-pulse level level-value
The polarity reversal pulse level width is set.

l Run:
polarity-reverse-mode { hard
| soft }
The polarity reversal mode is set.

l Run:
clip-reverse-pole-pulse enable
The CLIP polarity reversal pulse function is enabled.

l Run:
dial-mode { dtmf | pulse | both }
The dialing mode is set.

l Run:
line-lock enable
Port locking is enabled.

l Run:
nlp-mode { disable | normal | ec-gain-adjust | enhance | reduce }
The non-linear processing (NLP) mode is set.

Step 5 Set electrical attributes for the FXS interface.
l Run:
electric current current-value
The current is set.

l Run:
electric impedance impedance-value
The impedance is set.

l Run:
electric send-gain sendgain-value
The send gain is set.

l Run:
electric receive-gain recvgain-value
The receive gain is set.

Step 6 Set KC attributes for the FXS interface.
l Run:
kc high-level highlevel
The high-level pulse width is set.

l Run:
kc low-level lowlevel
The low-level pulse width is set.

l Run:
kc type { 16kc | 12kc }
The KC accounting mode is set.

l Run:
Issue 01 (2012-04-20)

17

kc voltage voltage
The voltage operating is set.

----End

Run the display voice port fxs [ state ] [ slotid/subcardid/portid ] command to check the FXS
interface configuration.
1.5.2 Setting parameters for a BRA Interface

This section describes how to set parameters for a BRA interface.
A basic rate access (BRA) interface connects to an ISDN phone. On the AR2200, you can enable
the BRA interface Layer 2 monitoring, remote power supply, automatic deactivation, and alarm
functions, and set the working mode and Layer 1 activation mode on a BRA interface.
Before setting parameters for a BRA interface, complete the following task:
l
Ensuring that the 2BST board is working properly
Data Preparation
To set parameters for a BRA interface, you need the following data.
No.
Data
Interface working mode, automatic deactivation delay, and Layer 1 activation mode
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
port bra slotid/subcardid/portid
The BRA interface view is displayed.

Step 4 Run:
l2-monitor enable
Issue 01 (2012-04-20)

18

BRA interface Layer 2 monitoring is enabled.

Step 5 Run:
work-mode { p2p | p2mp }
The working mode is set.

Step 6 Run:
remote-power enable
Remote power supply is enabled.

Step 7 Run:
auto-deactive enable
Automatic deactivation is enabled.

Step 8 Run:
auto-deactive delay delay
The automatic deactivation delay is set.

Step 9 Run:
active-mode { unstable | stable }
The Layer 1 activation mode is set.

Step 10 Run:
alarm enable
The alarm function is enabled.

----End

Run the display voice port bra [ state ] [ slotid/subcardid/portid ] command to check the BRA
1.5.3 Setting Parameters for a PRA Interface

This section describes how to set parameters for a PRA interface.
A PRA interface connects to a PBX or PSTN network. On the AR2200, you can enable the
CRC4 check, E1 interface Layer 2 monitoring, and E1 interface pulse code modulation (PCM)
alarm functions, and set the CRC alarm threshold and E1 interface signaling mode on a PRA
interface.
Before setting parameters for a PRA interface, complete the following task:
l
Ensuring that the 1E1T1-M/2E1T1-M board is working properly
Issue 01 (2012-04-20)

19

Data Preparation
To set parameters for a PRA interface, you need the following data.
No.
Data
Access digital section mode
CRC alarm threshold and signaling mode of the E1 interface
Procedure
Step 1 Run:
system-view

Step 2 Run:
set workmode slot slot-id
e1t1 e1-voice
The E1 interface board is configured as an E1 voice card.

Step 3 Run:
voice

Step 4 Run:
port ve1 slotid/subcardid/portid
The PRA interface view is displayed.

Step 5 Run:
access-mode { digital | direct }
The access digital section mode is set.

Step 6 Run:
crc4 enable
CRC4 check is enabled.

Step 7 Run:
crc-alarm-threshold { es es-threshold | cses cses-threshold | dm dm- threshold }
The CRC alarm threshold is set.

Step 8 Run:
l2-monitor enable
E1 interface Layer 2 monitoring is enabled.

Step 9 Run:
pcm-alarm
The PCM alarm function is enabled.

Step 10 Run:
signal { CCS | CAS }
Issue 01 (2012-04-20)

20

The signaling mode is set.

----End

Run the display voice port ve1 [ state ] [ slotid/subcardid/portid ] command to check the PRA
1.6 (Optional) Setting Voice Parameters on a SIP AG

This section describes how to set voice parameters on a SIP AG.
1.6.1 Setting System Parameters

This section describes how to set system parameters.
Establishing the Configuration Task

Before setting system parameters, familiarize yourself with the applicable environment,
The AR2200 working in SIP AG mode can exchange information with a softswitch device using
SIP. Different countries and regions use different voice parameter standards; therefore, set voice
parameters on the SIP AG in accordance with local standards.
Before setting system parameters, complete the following task:
l
Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Data Preparation
To set system parameters, you need the following data.
Issue 01 (2012-04-20)
No.
Data
Country/Region identifier
Lower threshold and upper threshold for hookflash pressing
MWI mode
Codec mode for voice services
AC amplitude of the ringing current, frequency of the ringing current, and cadence
ratio
21

No.
Data
CLIP parameters in offhook state:
l Interval between the time when the ACK message is received and the time when
the FSK is transmitted
l Maximum duration between the time when the DT-AS is transmitted and the time
when the ACK message is received
l Duration of the DT-AS
l Level of the DT-AS
l Level of the FSK
l Number of bits of the FSK synchronization mask
7
CLIP parameters in onhook state:

l Number of bits seized by a channel
l Interval between the time when the DT-AS is transmitted and the time when the
FSK is transmitted
l Level of the FSK
Maximum uplink bandwidth and reserved bandwidth for emergency calls
Configuring a Country/Region Identifier

This section describes how to configure a country/region identifier.
A country/region identifier is configured on a SIP AG so that user terminals connect to the SIP
AG can comply with the local standard.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
country { brazil | britain-msfuk | britain-etsi | bulgaria | china-hongkong
china-mainland | egypt | france | singapore | thailand }
A country/region identifier is configured.

Issue 01 (2012-04-20)

22

By default, a country/region identifier is China mainland.

----End
Setting Upper and Lower Thresholds for Hookflash Pressing

The upper and lower thresholds for hookflash pressing determine the hookflash duration.
Context
Hookflash or flash is a button on a telephone that simulates quickly hanging up and then picking
up again (a quick off-hook/on-hook/off-hook cycle). The hookflash can be pressed by a calling
party or a called party:
l
Hookflash pressed by a called party: If the called party user A wants to transfer an incoming
call to user B, user A can press the hookflash and dial the number of user B.
Hookflash pressed by a calling party: User A calls user B. User B answers the call and talks
with user A. User A can press the hookflash and dial the number of user C after hearing a
special dial tone.
After user A presses the hookflash:

l
If user C is busy, user A can press the hookflash and talk with user B.
If user C does not respond for a long period of time, user A can press the hookflash and
talk with user B.
If the phone of user C rings, user A hangs up and user B hears the ringback tone. User C
picks up the phone and talks with user B.
Whether a called party can be transferred to a toll call is restricted by the outgoing right of
the called party.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
flash-hook lower lower-value
The lower threshold for hookflash pressing is set.

By default, the lower threshold for hookflash pressing is 100 ms.
Step 4 Run:
flash-hook upper upper-value
The upper threshold for hookflash pressing is set.

By default, the upper threshold for hookflash pressing is 350 ms.
Issue 01 (2012-04-20)

23

NOTE
The lower threshold for hookflash pressing must be 50 ms less than the upper threshold for hookflash
pressing.
----End
Setting the MWI Mode

You need to set the MWI mode according to user habits.
Context
If there are leave messages, the user device configured with the MWI function makes the
indicator on or plays a tone, indicating that there are leave messages. You can set the MWI mode
according to user habits.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
mwi-mode { fsk-with-ring | fsk-without-ring | high-voltage }
The MWI mode is set.

----End
Configuring the Codec Mode for Voice Services

You need to configure the codec mode for voice services so that the user device can work in
compliance with the local standard.
Context
G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.
711 defines two main compression algorithms, the -law algorithm (used in North America &
Japan) and A-law algorithm (used in Europe and China). A-law encoding takes a 13-bit signed
linear audio sample as input. -law encoding takes a 15-bit signed linear audio sample as input.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
Issue 01 (2012-04-20)

24


Step 3 Run:
pcm { a-law | u-law }
The codec mode is configured for voice services.

----End
Enabling the AR2200 to Check the Checksum in Packets

You can enable the AR2200 to check the checksum in received messages or encapsulate the
checksum in messages to be sent.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run the following commands as required.
l
Run:
sctp checksum receive enable
The AR2200 is enabled to check the checksum in received messages.

l
Run:
sctp checksum send enable
The AR2200 is enabled to encapsulate the checksum in messages to be sent.

----End
Setting Ringing Parameters

You can set the AC amplitude of the ringing current, frequency of the ringing current, cadence
ratio, and initial ringing function on the AR2200 to meet local standards.
Context
Different countries and regions use different ringing standards. You can set the AC amplitude
of the ringing current to adjust the ringing tone volume, voice pitch, cadence ratio, and initial
ringing function on the AR2200 to meet local standards.
Procedure
Step 1 Run:
system-view

Issue 01 (2012-04-20)

25

Step 2 Run:
voice

Step 3 Run the following command according to user requirements.
l
Run:
ring frequency { 16hz | 25hz | 50hz }
The frequency of the ringing current is set.

l
Run:
user-defined-ring ring-index { first-ring first-ring-period | first-interval
first-interval | second-ring second-ring-period | second-interval secondinterval | third-ring third-ring-period | third-interval third-interval }*
The cadence ratio of a specified user-defined ring type is set.

l
Run:
stop-initial-ring { enable | disable }
The initial ringing function is enabled.

----End
Enabling the Function That Reduces the Feed on Locked Ports

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
park-feed enable
The function that reduces the feed on locked ports is enabled.

This function takes effect for only FXS ports.
----End
Setting CLIP Parameters in Offhook State

You can set CLIP parameters in offhook state on the AR2200 so that the AR2200 can work with
the phone terminal.
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be configured
on the AR2200 so that the AR2200 can work with the phone terminal. Generally, default
parameter settings are used. If CLIP parameters are not set properly, change relevant CLIP
parameters.
Issue 01 (2012-04-20)

26

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

l
Run:
clip offhook ack-fsk-interval ack-fsk-interval
The interval between the time when the ACK message is received and the time when the
frequency-shift keying (FSK) is transmitted in offhook state is set.
l
Run:
clip offhook dtas-ack-interval dtas-ack-interval
The maximum duration between the time when the dual tone-alerting signal (DT-AS) is
transmitted and the time when the ACK message is received in offhook state is set.
l
Run:
clip offhook dtas-duration dtas-dur-value
The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.
l
Run:
clip offhook dtas-level dtas-level
The level of the DT-AS in offhook state is configured.

l
Run:
clip offhook fsk-level fsk-level
The level of the FSK in offhook state is configured.

l
Run:
clip offhook mark-signal-bit mark-signal-bit
The number of bits of the FSK synchronization mask in offhook state is set.
----End
Setting CLIP Parameters in Onhook State

You can set CLIP parameters in onhook state on the AR2200 so that the AR2200 can work with
the phone terminal.
Context
When the CLIP service is registered, CLIP parameters in onhook state need to be configured on
the AR2200 so that the AR2200 can work with the phone terminal. Generally, default parameter
settings are used. If CLIP parameters are not set properly, change relevant CLIP parameters.
Procedure
Step 1 Run:
system-view
Issue 01 (2012-04-20)

27


Step 2 Run:
voice

l
Run:
clip onhook channel-seize-bit channel-seize-bit
The number of bits seized by a channel in onhook state is set.

l
Run:
clip onhook dtas-duration dtas-dur-value
The duration of the DT-AS in onhook state is set.

l
Run:
clip onhook dtas-fsk-interval dtas-fsk-interval
The interval between the time when the DT-AS is transmitted and the time when the FSK
is transmitted in onhook state is set.
l
Run:
clip onhook dtas-level dtas-level
The level of the DT-AS in onhook state is configured.

l
Run:
clip onhook fsk-level fsk-level
The level of the FSK in onhook state is configured.

l
Run:
clip onhook mark-signal-bit marksignal-bit
The number of bits of the FSK synchronization mask in onhook state is set.
----End
Setting Uplink Bandwidth Parameters

This section describes how to set uplink bandwidth parameters.
Context
The AR2200 provides uplink bandwidth control. When the system detects that the uplink
bandwidth usage reaches the configured upper threshold, it restricts calls and generates an alarm.
If the uplink bandwidth is insufficient, the system processes calls based on user levels. Common
users may not obtain services.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
Issue 01 (2012-04-20)

28


l
Run:
media-bandwidth-control enable
The AR2200 is enabled to restrict calls when the uplink bandwidth is enabled.
l
Run:
media-bandwidth-control maximum max-bandwidth
The maximum uplink bandwidth is set.

l
Run:
media-bandwidth-control reserved-for-emergency reserved-bandwidth
The reserved bandwidth is set for emergency calls.

NOTE
The reserved bandwidth for emergency calls must be smaller than the maximum uplink bandwidth
configured by the media-bandwidth-control maximum command.
----End
Setting the Number of SIP Register Messages Sent Every Second

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sip-reg-count-per-second reg-count-value
The number of SIP register messages sent every second is set.

----End

After the configuration of system parameters is complete, you can view the system parameters
such as the SIP interface's IP address, upper and lower thresholds for hookflash pressing, and
CLIP parameters.
Procedure
l
Run the display voice configuration command to check the voice configuration.
Run the display voice user-defined-ring [ring-index ] command to check user-defined

ring information.
Run the display voice clip command to check CLIP parameters.
Issue 01 (2012-04-20)

29

Run the display voice sip-reg-count-per-second command to check the number of SIP
Register messages initiated per second.
----End
1.6.2 Setting SIP Stack Parameters

This section describes how to set SIP stack parameters.

Before setting SIP stack parameters, familiarize yourself with the applicable environment,
SIP is an IETF-defined signaling protocol widely used for controlling communication sessions
such as voice and video calls over Internet Protocol (IP). SIP, RTP, RTCP, RTSP, and other
protocols constitute a SIP protocol stack.
Before setting SIP protocol stack parameters, complete the following task:
l
Data Preparation
To set SIP protocol stack parameters, you need the following data.
No.
Data
Minimum value of the session timer, value of the unreliable provisional

response retransmission timer, value of the reliable provisional response
retransmission timer, value of the T1 timer, value of the T2 timer, value of the
T4 timer, and value of the TD timer
Values of the Max-forwards field, Organization header field, Server header

field, and User-Agent header field
Setting the Values of Timers

This section describes how to set values of timers of the SIP stack.
Procedure
Step 1 Run:
system-view
Issue 01 (2012-04-20)

30


Step 2 Run:
voice

Step 3 Run:
sip
The SIP view is displayed.

l
Run:
entity-based-sessions-timer enable
The session timer defined by RFC 4028 based on the UA is defined.

l
Run:
min-se min-seperiod
The minimum value of the session timer is set.

l
Run:
session-progress session-progressperiod
The value of the unreliable provisional response retransmission timer is set.

l
Run:
session-rel-progress session-rel-progressperiod
The value of the reliable provisional response retransmission timer is set.

l
Run:
t1 t1period
The value of the T1 timer is set.

l
Run:
t2 t2period

l
Run:
t4 t4period

l
Run:
td tdperiod
The value of the TD timer is set.

----End
Setting Values of Header Fields

This section describes how to set values of header fields.
Procedure
Step 1 Run:
system-view
Issue 01 (2012-04-20)

31


Step 2 Run:
voice

Step 3 Run:
sip

l
Run:
header-folding enable
Header folding in the authentication header field is enabled.

l
Run:
field-header max-forwards max-forwards-value
The value of the Max-forwards header field is set.

l
Run:
field-header organization organization-head
The value of the Organization header field is set.

l
Run:
field-header server server-head
The value of the Server header field is set.

l
Run:
field-header user-agent user-agent-head
The value of the User-agent header field is set.

----End

After the configuration is complete, you can view the configured SIP stack parameters.
Procedure
l
Run the display voice sip command to check SIP parameters.
----End
1.6.3 Setting DSP Parameters

This section describes how to set DSP parameters.

Before setting DSP parameters, familiarize yourself with the applicable environment, complete
the pre-configuration tasks, and obtain the data required for the configuration. This will help
you complete the configuration task quickly and accurately.
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The digital signal processing (DSP) collects, converts, filters, measures, enhances, compresses,
or identifies signals and coverts the signal from an analog to a digital form.
The DSP module converts analog voice signals into digital signals and stores a certain number
of digital signals into packets for transmission. To improve the voice communication quality,
the DSP needs to further process voice signals.
Before setting DSP parameters, complete the following task:
l
Data Preparation
To set DSP parameters, you need the following data.
No.
Data
VBD switching mode
Default DSP channel code type and default interval at which the DSP channel
packetizes RTP packets
T.30 redundancy parameter value of the T.38 fax, T.4 redundancy parameter value
of the T.38 fax, fax training mode, and maximum fax training rate
Alarm threshold of the dynamic jitter buffer, initial value of the dynamic jitter buffer,
maximum value of the dynamic jitter buffer, maximum value of the static jitter buffer,
minimum value of the dynamic jitter buffer, minimum value of the static jitter buffer,
and initial value of the static jitter buffer of a DSP channel
RTP payload type value, G.726-16k payload type value, G.726-24k payload type
value, G.726-32k payload type value, G.726-40k payload type value, NTE payload
type value, redundancy payload type value, and VBD payload value of a DSP channel
Interval at which a DSP channel sends RTCP packets and threshold for the number
of severe degrade seconds
Resource threshold in hierarchical control mode
Data event transmission mode, special process, DTMF transmission mode, echo
cancellation function, input gain, output gain, jitter buffer mode, NLP mode, and DSP
working mode in a DSP template
DSP loopback type and DSP channel parameters
Enabling Echo Cancellation on a DSP Channel

This section describes how to enable echo cancellation on a DSP channel.
Issue 01 (2012-04-20)

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Context
A user may hear the user's echo in the phone receiver in a conversation. If a proper delay in the
transmitted or received signal is set, the echo can be removed. If the delay exceeds 25 ms, the
voice quality deteriorates and the conversation ends. You can enable echo cancellation on a DSP
channel to remove echoes.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute
The DSP view is displayed.

Step 4 Run:
echo enable
Echo cancellation is enabled on the DSP channel.

----End
Enabling the PLC Function on a DSP Channel

This section describes how to enable the packet loss concealment (PLC) function on a DSP
channel.
Context
PLC is a technique that masks the effects of packet loss in VoIP communications. PLC is
effective only when the packet loss ratio is low. During communication, the average packet loss
ratio may be low, but a high burst packet loss ratio results in severe voice quality deterioration.
PLC can insert a static frame in the place where a packet is lost, regenerate a packet received
prior to the lost one, or generate an analog voice packet. If packets are lost during communication
and PLC is not used, the voice communication is interrupted. You can use a proper PLC
algorithm to minimize effects of packet losses.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
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Step 3 Run:
dsp-attribute

Step 4 Run:
plc enable
The PLC function is enabled on the DSP channel.

----End
Enabling Silence Compression on a DSP Channel

You can enable silence compression on a DSP channel to reduce the number of sent voice
packets.
Context
To save network bandwidth, enable silence compression on a DSP channel. When no voice is
detected, the encoder generates short silence codes, but does not generate voice compression
codes. In addition, the encoder notifies the receiver of silence start until the voice is restored.
The silence compression function reduces the number of sent voice packets.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
silence enable
Silence compression is enabled on the DSP channel.

----End
Setting the VBD Switching Mode

Procedure
Step 1 Run:
system-view
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Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
autovbd { auto | host-controlled }
The VBD switching mode is configured.

NOTE
By default, the VBD switching mode is host-controlled.
----End
Setting the Default DSP Channel Code Type and the Default Interval at Which a
DSP Channel Packetizes RTP Packets
This section describes how to set the default DSP channel code type and the default interval at
which a DSP channel packetizes RTP packets.
Context
The voice encoding technique encodes pulse-code modulation (PCM) samples into bits (frames).
This technique ensures robustness of voice services when the error code, network jitter, or burst
traffic occurs on a link. At the receiver side, voice frames are encoded into PCM samples, and
then are converted into voice waveforms. Different voice encoding techniques provide different
voice quality and a good voice quality requires high bandwidth.
In VoIP, before voice data are transmitted as UDP packets, the Real-time Transport Protocol
(RTP) processes the voice data. RTP is used to transmit real-time data and can transmit audio
and video data.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
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codec code { g711a | g711u | g723_1 | g729 | g726-16k | g726-24k | g726-32k |

g726-40k } [ rtp-interval { 5ms | 10ms | 20ms | 30ms } ]
The default DSP channel code type and the default interval at which a DSP channel packetizes
RTP packets are set.
----End
Setting Fax Parameters

This section describes how to set fax parameters.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Set fax parameters.
l
Run:
fax redundancy-t4 redundancy-t4value
The T.4 redundancy parameter value of the T.38 fax is set.

l
Run:

l
Run:
fax training-mode { e2e | local }
The fax training mode is set.

l
Run:
fax training-rate { v17 | v29 | v27 }
The maximum fax training rate is set.

l
Run:
fax v8negotiate enable
V8 negotiation is enabled.
----End
Setting Jitter Buffer Parameters

This section describes how to set jitter buffer parameters to improve the voice quality.
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Context
Delay variations in voice packet arrival time can occur because of network congestion or route
changes. To reduce sound distortion caused by the delay jitter and packet loss, a jitter buffer is
used. You can set proper jitter buffer parameters to minimize delay variations so that packets
can be processed in a timely manner and smooth voice communication can be provided as much
as possible.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Set parameters for the Jitter Buffer.
l
Run:
jitter-buffer adapt-jb-threshold adapt-jb-threshold
The alarm threshold of the dynamic jitter buffer is set.

l
Run:
jitter-buffer init-adapt-jb init-adapt-jb-value
The initial value of the dynamic jitter buffer is set.

l
Run:
jitter-buffer init-fixed-jb normal-fixed-jb-value
The initial value of the static jitter buffer is set.

l
Run:
jitter-buffer max-adapt-jb max-adapt-jb-value
The maximum value of the dynamic jitter buffer is set.

l
Run:
jitter-buffer min-adapt-jb min-adapt-jb-value
The minimum value of the dynamic jitter buffer is set.

l
Run:
jitter-buffer max-fixed-jb max-fixed-jb-value
The maximum value of the static jitter buffer is set.

l
Run:
jitter-buffer min-fixed-jb min-fixed-jb-value
The minimum value of the static jitter buffer is set.

----End
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Setting the Payload Type Value

This section describes how to set the payload type value.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Set Payload Type parameters.
NOTE
The following payload type values must be different.
Run:
payload-type clear-mode clearmode-value
The RTP payload type value of a DSP channel is set.

l
Run:
payload-type g726-16k g726-16k-value
The G.726-16k payload type value of a DSP channel is set.

l
Run:

l
Run:

l
Run:

l
Run:
payload-type nte nte-value
The NTE payload type value of a DSP channel is set.

l
Run:
payload-type redundancy redundancy-value
The redundancy payload type value of a DSP channel is set.

l
Run:
payload-type vbd vbd-value
The VBD payload type value of a DSP channel is set.

----End
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Setting RTCP Parameters

This section describes how to set RTP control protocol (RTCP) parameters.
Context
RTCP monitors the quality of service and conveys information about participants in an on-going
session. RTCP periodically sends packets to all the participants in the session to monitor the
quality of service and obtain identity information about the participants.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

l
Run:
rtcp rtcp-interval rtcp-interval
The interval at which a DSP channel sends RTCP packets is set.

l
Run:
rtcp rtcpxr enable
The RTP Control Protocol Extended Reports (RTCP XR) function is enabled.
l
Run:
rtcp sev-degradethreshold sev-degradethresholdval
The threshold for the number of severe degrade seconds is set.

l
Run:
rtcp vqm enable
The Voice Quality Monitor (VQM) function is enabled.

----End
Setting DSP Resource Parameters

This section describes how to set DSP resource parameters.
Context
DSP resources are limited and users have different requirements for DSP resources. To control
and allocate DSP resources properly, set the DSP resource control mode and the resource
threshold in hierarchical control mode.
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

l
Run:
resource-threshold mode { priority | normal }
The DSP resource control mode is configured.

l
Run:
resource-threshold { threshold1 threshold1 | threshold2 threshold2 |
threshold3 threshold3 }
The resource threshold in hierarchical control mode is set.

The resource threshold in hierarchical control mode can be set only when the DSP resource
control mode is priority.
----End
Configuring a DSP Template

This section describes how to configure a DSP template.
Context
To customize DSP parameters for data services, configure a DSP template. After a DSP template
is configured, specify the template for users according to the port and phone number. The DSP
template improves the call connection rate. After a template is specified successfully, parameters
in the DSP template take effect immediately.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute
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Step 4 Run:
template name
A DSP template is created.

Step 5 Set parameters in the DSP template.
l
Run:
data-event data-event-value
The data event transmission mode is configured.

l
Run:
dsp-special-flow dsp-special-flow-value
A special process is configured.

l
Run:
dtmf dtmf-value
The Dual-Tone Multi-frequency (DTMF) transmission mode is configured.

l
Run:
echo-cancel { enable | disable }
The echo cancellation function is enabled or disabled.

l
Run:
input-gain input-gain-value
The input gain is set.

l
Run:
output-gain output-gain-value
The output gain is set.

l
Run:
jitter-buffer { dynamic | static }
The jitter buffer mode is configured.

l
Run:
nlp nlp-value
The normal link pulse (NLP) transmission mode is configured.

l
Run:
work-mode work-mode-value
The DSP working mode is configured.

l
Run:
vbd redundancy
Voice band data (VBD) redundancy transmission is enabled.

----End
Configuring a DSP Channel

This section describes how to configure a DSP channel.
Context
You can enable a digital signal processor (DSP) channel to work in loopback mode, and set the
loopback mode (PCM-side loopback test and IP-side loopback test). When the DSP channel
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between the calling party and called party cannot transmit signals or can transmit signals only
in one direction, run the loop-back command to locate the fault. If the calling party hears the
echo in a PCM-side loopback test, the speech channel between the calling phone and the calling
DSP channel is functioning properly. If the called party hears the echo in an IP-side loopback
test, the speech channel between the called phone and the calling DSP channel is functioning
properly.
To control resources of DSP channels, prohibit the DSP channels. The prohibited DSP channels
cannot participate in resource allocation.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp slot/dsp-index
The DSP resource management view is displayed.

l
Run:
loop-back loopback-type channel
The loopback function is configured on a DSP channel.

l
Run:
prohibit channel [ count ]
DSP channels are prohibited.

----End

After the configuration of DSP parameters is complete, you can view the DSP parameters
including fax parameters and jitter buffer parameters.
Procedure
l
Run the display voice dsp-attribute command to check the DSP configuration.
Run the display voice dsp state { slot/dsp-index | channel slot/dsp-index/channel }

command to check the status of a DSP or DSP channel.
Run the display voice dsp-template command to check the DSP template configuration.
----End
1.7 Configuring a SIPAG User

This section describes how to configure a SIPAG user.
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Before configuring a SIP AG user, familiarize yourself with the applicable environment,
On the IMS, a SIP AG is directly connected to a user terminal. You need to set parameters for
users on the SIP AG so that the users can use services on the IMS.
Before configuring a SIP AG user, complete the following tasks:
l
Setting the operation mode to SIP AG
Configuring a SIP AG interface
Data Preparation
To configure a SIP AG user, you need the following data.
No.
Data
SIP AG user port number, SIP AG ID for the SIP AG user, and phone number of the
SIP AG user
(Optional) SIP AG user in the SIP service data profile, and phone number of the SIP
AG user in the SIP service data profile
1.7.2 Setting Parameters for a SIP AG User

This section describes how to set parameters for a SIP AG user.
On the IMS, a SIP AG is directly connected to a user terminal. You need to set parameters for
users on the SIP AG so that the users can use services on the IMS.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
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sipaguser sipaguser-name [ port interface-number ]
A SIP AG user is created and the SIP AG user view is displayed.

After creating a SIP AG user, set parameters for the SIP AG user, including the interface number
and basic phone number associated to the SIP AG User.
Step 4 Run:
agid sipag-interface-ID
A SIP AG ID is specified for the SIP AG user.

Step 5 Run:
base-telno telno-value [ sipagusergroup usergroup-id ]
The basic phone number is configured for the SIP AG user.

Step 6 Set optional parameters for the SIP AG user.
Issue 01 (2012-04-20)
Parameter
Command
Default Setting
Timeslot status
active-mask time-slot time-slot-value

config-status config-status-value
NOTE
This command is valid only on
VE1 interfaces.
Maximum
number of
automatic
recovery
attempts from
deterioration
auto-resume-limit auto-resume-limit
20
SIP AG's
capability to
support the
BELL ANS
flag
bell-ans-flag { enable | disable }
Calling
number format
clip-format { sdmffsk | mdmffsk | dtmf |

r15 | etsi }
FSK simple data format
Sequence in
which the
calling number
of a SIP AG
user is
displayed
clip-transmission-sequence { after-ring |
before-ring }
after-ring
Power-off
interval
dc-time dc-time
10 ms
Single-tone
ANSbar signal
detection
detect-ansbar-by-single-tone { enable |
disable }
disable
DSP chip input

gain
dsp-input-gain dsp-input-gain
0 dB
NOTE
This command is valid only on VE1 and BRA
interfaces.
disable
NOTE
This command is valid only on
FXS interfaces.

45

Parameter
Command
Default Setting
DSP chip
output gain
dsp-output-gain dsp-output-gain
0 dB
DSP template
name
dsp-template dsp-template
No DSP template name

configured
Service
termination of
a specified SIP
AG user
endservice { defer [ maxcobtime

maxcobtime-value ] | immediate }
By default, services of a SIP

AG user are not terminated.
Extended
phone number
extend-telno telno-value
[ sipagusergroup usergroup-id ]
No extended phone number

configured
FSK call
display mode
fsk-mode { BELL_202 | CCITT_V23 |

NTT }
BELL_202
FSK delay
fsk-time fsk-time
10 ms
User type
line-type { DEL | ECPBX | LCPBX |

PayPhone }
DEL
SIP AG user
priority
priority { cat1 | cat2 | cat3 }
cat3
TAS mode of
CLIP
tas-pattern { NO-TAS | DT-AS }
NO-TAS
UNI fault
reporting
function
uni-report { enable | disable }
disable
NOTE
This command is valid only on VE1 and BRA
interfaces.
Voice quality
enhancement
(VQE)
function
vqe-agc { enable | disable }
disable
Target VQE
level
vqe-agc-level vqe-agc-level
13
VQE noise
suppression
vqe-sns { enable | disable }
disable
Noise
suppression
level
vqe-sns-level vqe-sns-level
12 dB
----End
1.7.3 Configuring a SIP AG User Group

This section describes how to configure a SIP AG user group.
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sipagusergroup sipag-interface-id usergroup-id
A SIP AG user group is created.

Step 4 Set parameters for the SIP AG user group.
l
Run:
group-name usergroup-name
The description is configured.

l
Run:
auth username auth-username password { cipher | ha1 { cipher | simple authpassword1 } | simple auth-password2 }
The authentication user name and password are configured.

l
Run:
precinct-mode { local | remote }
The mode used to manage users in the SIP AP user group is configured.
l
Run:
register-uri-mode { inneruser | alone }
The registrar URI mode is configured.

l
Run:
subscribe ua-profile { enable
| disable
The mode used to initiate a UA profile subscription request is configured.

l
Run:
uri uri
A URI is configured.
l
Run:
endservice
Services of a specified user group are terminated.

----End
1.7.4 (Optional) Configuring Functions or Services in a SIP Service

Data Profile
This section describes how to configure functions or services in a SIP AG service data profile.
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After a SIP AG user is configured, communication can be implemented. You can enable other
services for the SIP AG user according to user requirements.
Before configuring other services for a SIP AG user, run the service-right conf disable
command in the SIP AG service data profile to disable the call conference service.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sipservicedata sipaguser-name telephone-number
The SIP service data profile view is displayed.

Step 4 Configure the following functions or services in the SIP service data profile.
Issue 01 (2012-04-20)
Function/Service
Command
Default Setting
Authentication user
name and password
auth username authusername password { cipher |

ha1 { cipher | simple authpassword1 } | simple authpassword2 }
No authentication user name

and password configured
Dial tone type
dial-tone { normal | special |

mwi }
Normal dial tone type
Dial-out mode for

Centrex users
centrex-dial { directly |
secondary }
Direct dialing
Centrex number
centrex-number centrexnumber
Empty
Centrex prefix
centrex-prefix centrex-prefix
No Centrex prefix configured
CFB forward-to number
service cfb cfbnum-value
No CFB forward-to number

configured
CFNR forward-to
number and timeout
interval
service cfnr cfnrnumtimevalue cfnrnum-value
No CFNR forward-to number

configured
CFO forward-to number
service cfo cfbnum-value
No CFO forward-to number

configured
CFU forward-to number
service cfu cfunum-value
No CFU forward-to number

configured

48

Function/Service
Command
Default Setting
Hotline number
service hotline hotline-time

hotline-number
No hotline number configured
Anonymous call service
service-right anonymous-call
enable
disable
Call forwarding service
service-right call-diversion
enable
disable
Call hold service
service-right call-hold enable
enable
Call transfer service
service-right call-transfer
enable
disable
Call conference service
service-right conf enable
enable
Call waiting service
service-right cw enable
enable
Hotline service
service-right hotline enable
disable
Malicious call
identification (MCID)
service
service-right malicious-callidentification enable
disable
MWI service
service-right mwi enable
disable
Terminal portability
service
service-right terminalportability enable
disable
Three-party service
service-right three-party
enable
enable
MWI service mode
mwi-mode { deferred |
immediate | combine }
Deferred MWI service mode
UUS1 service
service-right uus1 enable
disable
UUS2 service
disable
UUS3 service
disable
----End

Procedure
l
Run the display voice sipaguser [ sipaguser-name ] command to check the configuration
of the SIP AG user.
Run the display voice sipagusergroup sipag-interface-id [ usergroup-id ] command to

check information about user group.
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Run the display voice sipservicedata sipag-user-name telephone-number command to

check service data in the SIP service data profile.
----End
1.8 (Optional) Configuring the BEST Service

The branch exchange for survivable telephony (BEST) service enables SIP AGs in enterprise
branches to manage local calls when communication with the headquarters voice server fails.
Many enterprises or organizations deploy IP phones in their branches and use SIP servers in
headquarters to control calls of remote branches in a centralized manner. When communication
between a branch and the headquarters fails, the call service and other voice services in the
branch are interrupted. The BEST service can be configured to solve this problem. When
communication between a branch SIP AG and the headquarters SIP server fails, the SIP AG in
the branch starts to manage local calls to ensure uninterrupted voice services in the branch. When
communication between the branch and headquarters is restored, the headquarters SIP server
controls all calls.
As shown in Figure 1-2, an AR router functions as a SIP AG and connects to an IMS network.
When communication with the IMS network fails, the router manages local calls.
Figure 1-2 BEST networking
IMS
Network
Eth1/0/0
port 2/0/0
SIPAG
port 2/0/2
port 2/0/1
POTS
IP Phone
FAX
CAUTION
After enabling the BEST service, you must restart the router for the service to take effect, which
will interrupt all services on the router. Therefore, confirm your operation before enabling the
BEST service.
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Before configuring the BEST service, complete the following tasks:
l
Configuring the router to work in SIP AG mode
Configuring a SIP AG
NOTE
By default, a SIP AG uses the probe mode to detect a proxy server. The BEST service cannot be used
in probe mode; therefore, after creating a SIP AG, run the proxy-detect-mode command to set the
proxy detection mode to option or register.
Configuring SIP AG users

NOTE
The SIP AG bound to the BEST service must have user phone numbers configured.
Procedure
1.
Run:
system-view

2.
Run:
voice

3.
Run:
best enable
The BEST service is enabled.

4.
Run:
quit

5.
Run:
quit

6.
Run:
save
The current configuration is saved.

7.
Run:
reboot
The router is restarted.

NOTE
The BEST service takes effect only after you save the configuration and restart the router.
8.
Configure a call prefix.

NOTE
l When configuring the BEST service, the call prefix is bound to the enterprise default and DN
set defaultdialplan configured by using the enterprise default dn-set defaultdialplan
command.
9.
Configure a SIP server in the PBX view.
10. In the voice view, run:

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51

best probe mgid
A SIP AG is bound to the BEST service.

l
Run the display voice best config command to check the BEST service configuration.
Run the display voice best probe command to check the SIP AG bound to the BEST service.
Run the display voice best status command to check the status of the BEST service.
1.9 Resetting a SIP AG

This section describes how to reset a SIP AG.
To make modified SIP AG parameters take effect, reset the SIP AG.
Before resetting a SIP AG, complete the following task:
l
Configuring a SIP AG interface and a SIP AG user
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sipag sipag-interface-id
The SIP AG view is displayed.

Step 4 Run:
reset
The SIP AG view is reset.
CAUTION
Exercise caution when you run this command because resetting a SIP AG interrupts running
services.
----End
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1.10 Maintaining a SIP AG

This section describes how to maintain a SIP AG.
1.10.1 Clearing SIP AG Statistics

Context
CAUTION
The cleared SIP AG statistics cannot be restored. Exercise caution when you run reset
commands.
Procedure
Step 1 Run the reset sctp-association-statistics command in the SIP AG view to clear statistics about
SCTP associations on a SIP AG.
Step 2 Run the reset sctp-global-statistics command in the user view to clear global SCTP statistics.
----End

This section provides a SIPAG configuration example.
1.11.1 Example for Configuring a SIP AG

Networking Requirements
As shown in Figure 1-3, the SIP AG is directly connected to user terminals and exchanges media
and signaling streams with the IMS network. The SIP AG connects to POTS users through the
4FXS1FXO board and connects to ISDN users through the 2BST board. Users connect to the
IMS through the SIP AG to use voice, data, and multimedia services.
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53

Figure 1-3 SIP AG networking
IMS
Network
GE1/0/0
port 2/0/0
SIPAG
port 1/0/0
port 2/0/1
POTS
POTS
ISDN
Configuration Roadmap
The configuration roadmap is as follows:
1.
Configure the AR2200 to work in SIP AG mode.
2.
Create SIP AG interface and set parameters for the SIP AG interface.
3.
Create SIP AG users and set parameters of SIP AG users.
Data Preparation
To complete the configuration, you need the following data:
l
Interface through which the SIP AG exchanges media and signaling streams with the IMS
network: GigabitEthernet1/0/0
IP address of GigabitEthernet1/0/0: 1.1.1.1/24
IP address and port number of the primary proxy server: 2.2.2.2/5060
SIP AG's ports directly connected to user terminals: port 2/0/0, port 2/0/1, and port 1/0/0
Phone numbers of user terminals connected to port 2/0/0, port 2/0/1, and port 1/0/0:
11111111, 11112222, and 11113333
Procedure
Step 1 Configure the AR2200 to work in SIP AG mode and reboot the device.
<Huawei> system-view
[Huawei] voice
[Huawei-voice] service-mode sipag
The modification takes effect only after you save the data and reboot the devic
e. Are you sure to change the protocol configuration?(y/n)[n]:
y
[Huawei-voice] quit
[Huawei] quit
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54

<Huawei> save
The current configuration will be written to the device.
Are you sure to continue? (y/n)[n]:y
It will take several minutes to save configuration file, please wait..........
..............
Configuration file had been saved successfully
Note: The configuration file will take effect after being activated
<Huawei> reboot
Info: The system is comparing the configuration, please wait.
System will reboot! Continue ? [y/n]:y
Info: system is rebooting ,please wait...
Step 2 Configure a SIP AG interface.

# Configure an IP address pool.
[Huawei] interface gigabitethernet 1/0/0
[Huawei-GigabitEthernet1/0/0] ip address 1.1.1.1 24
[Huawei-GigabitEthernet1/0/0] quit
# Set the IP address pool for exchanging media and signaling streams.
[Huawei] voice
[Huawei-voice] voip-address signalling interface gigabitethernet 1/0/0 1.1.1.1
[Huawei-voice] voip-address media interface gigabitethernet 1/0/0 1.1.1.1
# Set parameters on the SIP AG interface.

[Huawei-voice] sipag 1
[Huawei-voice-sipag-1]
signalling-addr 1.1.1.1 5060

media-addr 1.1.1.1
transfer udp
primary-proxy-addr static 2.2.2.2 5060
home-domain huawei.com
profile 1
quit
Step 3 Configure SIP AG users.

# Configure SIP AG users.
[Huawei-voice] sipaguser 1
[Huawei-voice-sipaguser-1]
port 2/0/0
agid 1
base-telno 11111111
quit
port 2/0/1
agid 1
base-telno 11112222
quit
port 1/0/0
agid 1
base-telno 11113333
quit
# Reset the SIP AG.

[Huawei-voice-sipag-1] reset
Note: Reset Sipag interface succeeds.
Step 4 Verify the configuration.

Then user terminals on the SIP AG can implement voice communication.
Run the display voice sipag [ sipag-interface-id { running | config } ] command to view detailed
information about the SIP AG.
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<Huawei> display voice sipag 1 config
AGID
Dynamic signalling IP address name
Signalling IP
Signalling port
Dynamic media IP address name
Media IP
Transfer mode
Primary proxy IP 1
Primary proxy IP 2
Secondary proxy IP 1
Primary proxy port
Secondary proxy port
Primary proxy domain name
Secondary proxy domain name
Proxy address mode
Home domain name
SIP profile index
Service logic index
Server Address DHCP option
Description
AG domain name
Phone context
Register URI
Subscribe to UA profile
Subscribe to reg state
Subscribe to MWI
Mode of supporting proxy dual-homing
Proxy detection mode
Proxy refresh mode
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
1
1.1.1.1
5060
1.1.1.1
UDP
2.2.2.2
255.255.255.255
255.255.255.255
255.255.255.255
5060
65535
IP
huawei.com
1: Default
0: Default
0: None
Enable
Disable
Disable
Remote
Manual switch over
Probe
Run the display voice sipaguser [ sipaguser-name ] command to view detailed information
about the SIP AG users.
<Huawei> display voice sipaguser 1
Slotid/Subcard/Portid
: 2/0/0
AGID
:
Base telno
: 11111111
GroupID
:
----End
Configuration Files
Configuration file of the Router
#
voice
voip-address signalling interface GigabitEthernet 1/0/0
1.1.1.1
voip-address media interface GigabitEthernet 1/0/0
1.1.1.1
#
sipag 1
signalling-addr 1.1.1.1
5060
media-addr 1.1.1.1
primary-proxy-addr static 2.2.2.2
5060
#
sipaguser 1 port 2/0/0
base-telno 11111111
agid 1
#
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sipaguser 2
base-telno
agid 1
#
sipaguser 3
base-telno
agid 1
port 2/0/1
11112222
port 1/0/0
11113333
#
interface GigabitEthernet1/0/0
ip address 1.1.1.1 255.255.255.0
#
return
1.11.2 Using a PRA Trunk to Connect the SIP AG to a PBX

As shown in Figure 1-4, the Router functions as the SIP AG and uses a PRA trunk to connect
to user terminals connected to the PBX to transmit voice, data, and multimedia services. The
PBX can be an IP PBX (the AR router can work in PBX mode) or a digital stored program
control switch.
NOTE
The Router uses the 1E1T1-M board to connect to the PBX.
Figure 1-4 Using a PRA trunk to connect the SIP AG to a PBX
IMS
Network
GE1/0/0
Router
port 1/0/0
PRA trunk
PBX
POTS A
POTS B
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57

1.
Configure the AR2200 to work in SIPAG mode on the Router.
2.
Configure the 1E1T1-M board to work in e1-voice mode.
3.
Create SIP AG interface and set parameters for the SIP AG interface on the Router.
4.
Create SIP AG users and set parameters of SIP AG users on the Router.
5.
Configure the trunk type on the PBX as PBX and use the PBX as the user-side device.
Data Preparation
l
Interface through which the Router exchanges media and signaling streams with the IMS
network: GigabitEthernet1/0/0
IP address of GigabitEthernet1/0/0: 1.1.1.1/24
IP address and port number of the primary proxy server: 2.2.2.2/5060
Phone number of POTSA: 11111111
Phone number of POTSB: 11112222
Procedure
Step 1 Configure the AR2200 to work in SIP AG mode on the Router and reboot the device.
[Huawei] sysname Router
[Router] voice
[Router-voice] service-mode sipag
The modification takes effect only after you save the data and reboot the devic
e. Are you sure to change the protocol configuration?(y/n)[n]:
y
[Router-voice] quit
[Router] quit
<Huawei> save
..............
<Router> reboot
Step 2 Configure the 1E1T1-M board to work in e1-voice mode.

<Router> system-view
[Router] set workmode slot 1 e1t1 e1-voice
Changing the working mode will reset the board in slot 1. Continue? [y/n]:y
INFO: Resetting board[1] succeeded.
Step 3 Configure a SIP AG interface.

[Router] interface gigabitethernet 1/0/0
[Router-GigabitEthernet1/0/0] ip address 1.1.1.1 24
[Router-GigabitEthernet1/0/0] quit
# Set the IP address pool for exchanging media and signaling streams.
[Router] voice
[Router-voice] voip-address signalling interface gigabitethernet 1/0/0 1.1.1.1
[Router-voice] voip-address media interface gigabitethernet 1/0/0 1.1.1.1
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58

# Set parameters on the SIP AG interface.

[Router-voice] sipag 1
[Router-voice-sipag-1]

media-addr 1.1.1.1
transfer udp
profile 1
quit

[Router-voice] sipaguser 1
[Router-voice-sipaguser-1]
port 1/0/0
agid 1
extend-telno 11111111
quit
# Reset the SIP AG.

[Router-voice] sipag 1
[Router-voice-sipag-1] reset
Note: Reset Sipag interface succeeds.
Step 5 Configure the trunk type on the PBX as PBX and use the PBX as the user-side device.
The configurations on different types of PBXs are different, so the configuration procedure is
not mentioned here. If the AR is used as the PBX, see 2.12 Configuring a Trunk Group.
Then user terminals on the SIP AG can implement voice communication.
Run the display voice sipag [ sipag-interface-id { running | config } ] command to view detailed
information about the SIP AG.
<Router> display voice sipag 1 config
AGID
Dynamic signalling IP address name
Signalling IP
Signalling port
Dynamic media IP address name
Media IP
Transfer mode
Primary proxy IP 1
Primary proxy IP 2
Primary proxy port
Secondary proxy port
Primary proxy domain name
Secondary proxy domain name
Proxy address mode
Home domain name
SIP profile index
Service logic index
Server Address DHCP option
Description
AG domain name
Phone context
Register URI
Subscribe to UA profile
Subscribe to reg state
Subscribe to MWI
Issue 01 (2012-04-20)
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
1
1.1.1.1
5060
1.1.1.1
UDP
2.2.2.2
255.255.255.255
255.255.255.255
255.255.255.255
5060
65535
IP
huawei.com
1: Default
0: Default
0: None
Enable
Disable
Disable

59


: Remote
Mode of supporting proxy dual-homing : Manual switch over
Proxy detection mode
: Probe
Proxy refresh mode
:
Run the display voice sipaguser [ sipaguser-name ] command to view detailed information
about the SIP AG users.
<Router> display voice sipaguser 1
Slotid/Subcard/Portid
: 1/0/0
AGID
: 1
Base telno
:
GroupID
:
Extend telno
: 11111111
GroupID
:
Extend telno
: 11112222
GroupID
:
Priority
: cat3
Uri report
: Disable
Auto limit
: 20
B channel 0
: normal
B channel 1
: normal
B channel 2
: normal
B channel 3
: normal
B channel 4
: normal
B channel 5
: normal
B channel 6
: normal
B channel 7
: normal
B channel 8
: normal
B channel 9
: normal
B channel 10
: normal
B channel 11
: normal
B channel 12
: normal
B channel 13
: normal
B channel 14
: normal
B channel 15
: normal
B channel 16
: normal
B channel 17
: normal
B channel 18
: normal
B channel 19
: normal
B channel 20
: normal
B channel 21
: normal
B channel 22
: normal
B channel 23
: normal
B channel 24
: normal
B channel 25
: normal
B channel 26
: normal
B channel 27
: normal
B channel 28
: normal
B channel 29
: normal
B channel 30
: normal
B channel 31
: normal
----End
Configuration Files
#
board add 0/1 1VE1-MFT
#
set workmode slot 2 e1t1 e1-voice
#
interface GigabitEthernet1/0/0
ip address 1.1.1.1 255.255.255.0
#
voice
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60

voip-address signalling interface GigabitEthernet 1/0/0

1.1.1.1
voip-address media interface GigabitEthernet 1/0/0
1.1.1.1
#
sipag 1
signalling-addr 1.1.1.1
5060
media-addr 1.1.1.1
primary-proxy-addr static 2.2.2.2
5060
#
agid 1
#
return
1.11.3 Example for Configuring the BEST Service

The BEST service enables voice gateways in enterprise branches to manage local calls when
communication with the headquarters voice server fails.
As shown in Figure 1-5, the Router functions as a voice gateway and connects to an IMS
network. To ensure that the Router can manage local calls when communication with the IMS
network fails, configure the BEST service on the Router.
Figure 1-5 BEST networking
IMS
Network
GE1/0/0
port 2/0/0
SIPAG
port 2/0/2
Router
POTS
1000
POTS
1001
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61

1.
Configure the Router to work in SIP AG mode.
2.
Create a SIP AG and set parameters for the SIP AG interface.
3.
Set parameters of SIP AG users.
4.
Enable the BEST service.
5.
Configure a call prefix.
6.
Configure a SIP server.
7.
Bind the SIP AG to the BEST service.
Data Preparation
l
Interface through which the SIP AG exchanges media and signaling streams with the IMS
network: Ethernet1/0/0
IP address of Ethernet1/0/0: 10.10.1.171/24
IP address and port number of the primary proxy server for the SIP AG: 10.10.1.205, 5066
SIP AG's ports directly connected to user terminals: port2/0/0 and port2/0/1
Phone numbers of user terminals directly connected to port2/0/0 and port2/0/1 of the SIP
AG: 1000 and 1001
Media IP address of the SIP server: 192.168.1.3/24
Procedure
Step 1 Configure the AR2200 to work in SIP AG mode.
[Huawei] voice
[Huawei-voice] service-mode sipag
The modification takes effect only after you save the data and reboot the device.
Are you sure to change the protocol configuration?(y/n)[n]:
y
[Huawei-voice] quit
[Huawei] quit
<Huawei> save
..............
<Huawei> reboot
Step 2 Configure a SIP AG.

# Assign an IP addresses to the interface connected to the IMS network.
[Huawei] interface ethernet 1/0/0
[Huawei-Ethernet1/0/0] ip address 10.10.1.171 24
[Huawei-Ethernet1/0/0] quit
# Configure the signaling and media IP addresses.

[Huawei] voice
[Huawei-voice] voip-address signalling interface ethernet 1/0/0 10.10.1.171
[Huawei-voice] voip-address media interface ethernet 1/0/0 10.10.1.171
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62

# Create a SIP AG and configure parameters for the SIP AG.


media-addr 10.10.1.171
proxy-detect-mode option
quit

port 2/0/0
agid 1
base-telno 1000
quit
port 2/0/1
agid 1
base-telno 1001
quit
# Reset the SIP AG.

[Huawei-voice-sipag-1] reset
Are you sure to reset MG interface?(y/n)[n]: y
Reset MG interface succeeds!
Step 4 Enable the BEST service.

[Huawei-voice] best enable
The modification takes effect only after you save the data and reboot the
device.
Are you sure to change the BEST configuration?(y/n)[n]:y
[Huawei-voice] quit
[Huawei] quit
<Huawei> save
........
<Huawei> reboot
Step 5 Configure a call prefix.

[Huawei] voice
[Huawei-voice] callprefix 1
[Huawei-voice-callprefix-1]
enterprise default dn-set defaultdialplan centrex prefix 1

call-type category basic-service attribute 0
digit-length 4 6
quit
Step 6 Configure a SIP server.

[Huawei-voice] sipserver
[Huawei-voice-sipserver]
Issue 01 (2012-04-20)
signalling-address ip 10.10.1.171 port 5060

media-ip 10.10.1.171
register-uri huawei.com
reset

63

Are you sure to reset the SIP server?(y/n)[n]y

[Huawei-voice-sipserver] quit
Step 7 Bind the SIP AG to the BEST service.

[Huawei-voice] best probe 1
[Huawei] quit

Users connected to the SIP AG can communicate with each other.
Run the display voice best config command to check the BEST configuration. The following
information is displayed:
<Huawei> display voice best config
The BEST mode is enable.
Run the display voice best probe command to check the SIP AG bound to the BEST service.
The following information is displayed:
<Huawei> display voice best probe
The BEST probe mgid 1.
Run the display voice best status command to view the status of the BEST service. If the link
between the Router and the IMS network fails, the following information is displayed:
<Huawei> display voice best status
The BEST is running.
----End
Configuration Files
#
voice
voip-address signalling interface Ethernet 1/0/0 10.10.1.171
voip-address media interface Ethernet 1/0/0 10.10.1.171
best probe 1
#
sipserver
media-ip 10.10.1.171
#
callprefix 1
enterprise default dn-set defaultdialplan
prefix 1
digit-length 4 6
#
sipag 1
media-addr 10.10.1.171
proxy-detect-mode option
#
#
base-telno 1001
agid 1
#
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64

base-telno 1000
agid 1
#
return
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65

2 PBX Configuration
PBX Configuration
About This Chapter

This chapter describes the concepts and configuration of the Private Branch Exchange (PBX),
and provides configuration examples.
2.1 PBX Overview
This section describes the concept of PBX and PBX-related terms.
2.2 PBX Features Supported by the AR2200
The AR2200 provides the following PBX features: SIP server, CDR server, trunk/trunk group,
call routing, individual services, and group services.
2.3 PBX Configuration Procedures
This section provides PBX configuration procedures.
2.5 Setting PBX Parameters
This section describes how to set PBX parameters.
2.6 Configuring an Enterprise
After an enterprise is created, you can configure the Centrex group, prefix, and DN set.
2.7 Configuring a Call Prefix
A call prefix is located at the beginning of a called number.
2.8 Configuring a PBX User
A user identifier includes the user number and DN set.
2.9 (Optional) Configuring a SIP Server
A SIP server is the main component of the IP PBX and establishes SIP sessions.
2.10 (Optional)Configuring SBC Proxy Function
This section describes how to configure the SBC proxy function to implement voice services
for SIPUEs.
2.11 (Optional)Configuring a CDR Server
The call detail record (CDR) of users can be queried in real time, and the CDR data can be
analyzed by using a third-party tool so that users can quickly know call fees.
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2 PBX Configuration
2.12 Configuring a Trunk Group

A trunk group is associated with a call route to enable users from different exchange offices to
call each other.
2.13 Configuring a Call Route
A call route is associated with a trunk group to enable users from different offices to call each
other.
2.14 (Optional)Configuring Individualized PBX Services
The AR2200 supports multiple individualized PBX services.
2.15 (Optional) Configuring Group Services on the PBX
When functioning as a PBX, the AR2200 supports various individual services. You can
configure services for individual users as required.
This section provides a PBX configuration example.
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2 PBX Configuration
2.1 PBX Overview

This section describes the concept of PBX and PBX-related terms.
Introduction
A traditional private branch exchange (PBX) manages incoming and outgoing calls of an
enterprise. It connects the enterprise to the Public Switched Telephone Network (PSTN) and
provides services for devices such as telephones, fax machines, and modems. It allows users in
the enterprise to call each other using extension phones and routes inter-office calls to the PSTN
through a trunk line.
Traditional PBXs cannot meet requirements for computer telephony integration (CTI) and voice
over IP (VoIP). In addition, these PBXs are expensive and do not use standard and open
platforms, bringing difficulties in interconnection between PBXs of different vendors. IP PBXs
overcome the limitations of traditional PBXs. IP PBXs are based on the IP protocol and provide
the local exchange and IP user access functions. IP PBXs integrate the voice communications
system of an enterprise into the enterprise's data network so that the enterprise can build a uniform
voice and data network connecting branches offices and staff around the world.
The AR2200 can function as a PBX to provide traditional PBX functions and IP PBX functions.
NOTE
l The AR2200 series routers support the voice features only after a DSP module is installed.
l To provide voice services for POTS users, 4FXS/1FXO board is required.
l To provide voice services for ISDN users, 2BST board is required.
Terms
l
DN set
A dial number (DN) set defines a group of numbers that are processed in the same way.
A DN set, a country code, and an area code identify the home area of a user; a DN set and
a call prefix determine the dialing plan for a user. DN sets divide a physical network or a
device into multiple logical networks.
Centrex and enterprise

A central office exchange service (Centrex) group is a group of users on a PBX or users
connected to PBXs in different areas. The same service is provided for users in the same
Centrex group are provided the same service.
Each user in a Centrex group has two numbers, a private number and a long number. Users
in a Centrex group use private numbers to call each other by default. A user in the Centrex
group must dial the call prefix when calling a user out of the Centrex group. When
configuring or modifying attributes of Centrex users, you must specify the Centrex number.
An enterprise has one or more Centrex groups. Each Centrex group belongs to only one
enterprise. Users in an enterprise may belong to a Centrex group or not belong to any
Centrex group.
Centrex groups and enterprises identify users at the service layer, whereas DN sets identify
users at the switching layer.
Call prefix
A call prefix, an important attribute of the call service, defines a call number rule and
describes the call number distribution and routing plans in an exchange office. A call prefix
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2 PBX Configuration
identifies the service attribute (basic service or supplementary service; intra-office call,
national toll call, or international toll call) of a dialing plan and determines the range of
dialed number length. Call prefixes can also be used to control call permissions.
A PBX checks validity of a dialed number and connects or rejects the call based on the call
prefix. A call can be connected correctly only if the call matches the correct call prefix.
Therefore, proper call prefix configuration is the key to the call service. A call must match
at least one call prefix.
l
Trunk and trunk group

A trunk is a logical link between two exchange offices. Inter-office calls must be transmitted
by trunks. The AR2200 supports the following trunks: AT0, PRA, SIP, H323, and E1R2.
A trunk group is a set of trunks for inter-office calls with same attributes.
Call route
A call route binds an inter-office call prefix to a trunk group so that calls with the specified
call prefix are transmitted on the specified trunk line.
IMS
The IP Multimedia Core Network Subsystem (IMS) is an architectural framework for
providing IP multimedia services, including audio, video, text, and instant messages. It was
designed by the wireless standards body 3rd Generation Partnership Project (3GPP) in
Release 5.
SIP
SIP is an IETF-defined signaling protocol and runs at the control layer of the IMS. As an
application layer protocol, SIP establishes, modifies, or terminates multimedia sessions and
works with protocols such as the Real-time Transport Protocol (RTP), Real-Time Transport
Control Protocol (RTCP), Session Description Protocol (SDP), Real-time Stream Protocol
(RTSP), Domain Name System (DNS), Stream Control Transmission Protocol (SCTP),
and Transmission Control Protocol (TCP) to complete session setup and media negotiation.
H323
H.323 is a recommendation from the ITU Telecommunication Standardization Sector
(ITU-T) that defines the protocols to provide audio-visual communication sessions on
computer networks. H.323 initiates and terminates multimedia sessions, and can
dynamically change session attributes, including the required bandwidth, media type,
media encoding format, and support for broadcast.
PSTN user
PSTN users are Plain Old Telephone Service (POTS) users.
SIP UE
SIP user equipment (SIP UE) is a user device that connects to a PBX using the SIP protocol,
for example, an IP phone or a software terminal. SIP UEs connect to a PBX through the IP
network and obtain services from the PBX after registering on the PBX.
2.2 PBX Features Supported by the AR2200

The AR2200 provides the following PBX features: SIP server, CDR server, trunk/trunk group,
call routing, individual services, and group services.
SIP Server
If SIP users need to communicate with each other through the AR2200, configure a SIP server
on the AR2200.
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2 PBX Configuration
SBC Proxy
If employees on a business trip use private network IP addresses, the SIPUEs cannot properly
connect to the PBX at the enterprise's headquarters. If employees within the enterprise use private
network IP addresses, they cannot call an external number through the SIP trunk. To solve this
problem, the SBC proxy function needs to be configured on AR2200.
CDR Server
Call detail records (CDRs) are details about calls recorded in real time. You can use a third-party
tool to analyze CDRs of calls and obtain call fees of users. To record and view CDRs of users,
specify a CDR server on the AR2200.
Trunk and Trunk Group

Inter-office calls are transmitted by trunk lines. The AR2200 supports the following types of
trunks:
l
PRA trunk
A PRA trunk is connected to a VE1 interface and is usually used to connect two PBXs.
AT0 trunk
An AT0 trunk is connected to a foreign exchange office (FXO) interface and is used only
to connect to the upstream PSTN network.
SIP trunk
A SIP trunk is established on an IP link and connects a PBX and an IMS. There are two
types of SIP trunks: SIP AT0 trunk and common SIP trunk.
E1R2 trunk
The E1R2 trunk is bound to VE1 interfaces and is usually used to connect two PBXs. The
E1R2 trunk can be connected to a peer device using R2 signaling.
H323 trunk
An H323 trunk is established on an IP link. It connects a PBX and an IMS or connects
PBXs in different areas.
Trunks for inter-office calls with the same attributes are added to a trunk group. A trunk group
is associated with a call route to establish call sessions between exchange offices.
Call Route
A call route is bound to a trunk group to determine the trunk line for a specified call prefix. To
allow users to make inter-office calls, configure call routes.
Individual services
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Service Type
Description
Abbreviated dialing
Allows a user to dial an abbreviated code of

one or two digits instead of the original called
number.

70

Issue 01 (2012-04-20)
2 PBX Configuration
Service Type
Description
Outgoing call barring
Restricts call-out permissions of some

phones. For example, the outgoing call
barring service can prevent a phone from
making long-distance calls.
Call forwarding
Allows a called user to transfer a call to a

specified third party if the called user has
activated the call forwarding service and the
call meets the call forwarding conditions.
Number barring
Prevents a user from dialing a specified

number.
Do-not-disturb
Rejects calls to a user. When other users call

a user that has subscribed to the do-notdisturb service, they hear a do-not-disturb
tone or busy tone.
Reject anonymous call
Allows a user to reject a call when the calling

number is not displayed and plays a voice
prompt to the calling user.
Remote office
Allows a user to access the office telephone

network from any terminal and use original
services such as abbreviated dialing and call
transfer.
Secretary
Allows a user to designate another phone

number (for example, the secretary's phone
number) to process all incoming calls. That
is, all incoming calls of the user are
transferred to the secretary's phone number
first, and only the secretary can call the user
directly.
Wake-up service
Allows a phone to ring automatically at the

preset time.
Personal ring back tone (RBT)
Uses user-defined music or sound effects to

replace the common ringback tone, and plays
the music or sound effects to calling users.
Selective call rejection
Filters every incoming call against a list of

rejected calling numbers created by a user and
rejects calls made by numbers on the list.
Selective call acceptance
Filters every incoming call against a list of

calling numbers created by a user and accepts
calls from only the numbers on the list.

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Group services
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Service Type
Description
Call interception
Provides a voice prompt when a call fails to

be connected. The user can learn about the
failure cause and determine whether to redial
the called number according to the voice
prompt.
Distinctive ringing
Plays different ring tones to users based on

the prefix analysis result (local calls, national
calls, international calls, and intra-Centrex
calls).
Enterprise RBT
Uses music or sound effects made by an

enterprise to replace the common ringback
tone, and plays the music or sound effects to
calling users.
Interactive voice response (IVR)
Provides the automated attendant function

and allows enterprises to make their own IVR
menus and voice prompts, improving user
experience.
Number change
Hides calling numbers or displays the same

calling number for all outgoing calls.
Pre-routing number change
Provides various dialing modes and changes

the calling number displayed on the called
party's phone.
Post-routing number change
Provides various dialing modes and changes

the calling number displayed on the called
party's phone. A post-routing number change
plan can change a called number to a long
number to ensure that it complies with the
required number format.
PBX line selection
When a calling party dials the access number

of a PBX group, the system uses the
configured line selection mode to connect the
call to a user in the group.
Co-group pickup
Allows users in the same group to answer

calls for each other on their own phones. For
example, when the phone of user A rings, user
B in the same group can dial the service
access code plus user A's phone number to
answer the call.

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Service Type
Description
Simultaneous ringing

of a simultaneous ringing group, all member
phones in the group ring simultaneously, and
the called party can answer the call using any
ringing phone.
Sequential ringing

of a sequential ringing group, member phones
in the group ring in the configured sequence.
One number link you (ONLY)
When the calling party calls ONLY number

of the called party, multiple terminals of the
called party ring according to the configured
rules, and the called party can select one
terminal to answer the incoming call.
License Support
The PBX function is used with a license. To use the PBX function, apply for and purchase the
following license from Huawei local office:
Table 2-1 PBX licenses
License Name
Depends on
Remarks
This license has the following functions:
CM&BEST License
Value-added
service package
for voice services
l Implements intra-office calls.

l Works with a trunk to implement inter-office
call; however, either the calling or called
party must be a local user.
l Provides all services except IVR navigation.
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CT (Call Trunk)
License
Value-added
service package
for voice services
This license implements inter-office trunk

connections. All the interfaces on a device are
trunk interfaces, which cannot connect to IP
phones or analog phones.
IVR (Interactive
Voice Response)
License
CM&BEST
License or CT
(Call Trunk)
license
This license implements the IVR navigation

function.

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NOTE
l The PBX function is controlled by the CM&BEST, CT, and IVR licenses. For the functions provided
by each license, see the Remarks column in the preceding table.
l For the dependent licenses of the PBX function, see the Depends on column in the preceding table.
For example, the CM&BEST License depends on the value-added service package for voice services;
therefore, to use the CM function, load the value-added service package for voice services first.
l If a function has multiple capacity licenses, select one or multiple licenses. Multiple licenses can be
used together.
2.3 PBX Configuration Procedures

This section provides PBX configuration procedures.
The PBX configuration procedure differs in intra-office call and inter-office call scenarios.
PBX Configuration for Intra-Office Calls

As shown in Figure 2-1, the PBX only needs to process calls between users connected to the
PBX.
Figure 2-1 Networking of intra-office calls
PBX
FXS1/0/0
User A
FXS1/0/1
SIPUE
User B
Figure 2-2 shows the configuration flowchart for intra-office calls. After mandatory
configurations are complete, users can call each other.
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Figure 2-2 Configuration flowchart for intra-office calls

Start
Set voice interface

parameters
(Optional) Set parameters of FXS,

FXO, BRA, and VE1 ports
Configure global
PBX parameters
Configure global PBX parameters
Configure an
enterprise
Configure enterprise, Centrex group,

and DN set.
Configure a call
prefix
Configure a call prefix
Configure PBX
users
Configure the device identifier and

user identifier for each PBX user
Configure a SIP
server
(Optional) Skip this step when there

is no SIP user
Configure a CDR
server
(Optional) Skip this step when CDRs

do not need to be recorded
Configure
individual services
(Optional) Configure individual

services as required
Configure group
services
(Optional) Configure group services

as required
End
PBX Configuration for Inter-Office Calls

As shown in Figure 2-3, user A in the headquarters needs to call user B in the branch. The
outgoing call needs to be connected through a trunk line.
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Figure 2-3 Networking of inter-office calls
Branch
User B
AR
Enterprise A
SIP
PSTN
IP
SIP
AT0
AR
Headquarters
PBX
Enterprise A
User A
Figure 2-4 shows the configuration flowchart for inter-office calls. After mandatory
configurations are complete, users in the headquarters and branch can call each other.
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Figure 2-4 Configuration flowchart for inter-office calls

Start
Set voice interface

parameters
(Optional) Set parameters of FXS,

FXO, BRA, and VE1 interfaces
Configure global
PBX parameters
Configure global PBX parameters
Configure an
enterprise
Configure enterprise, Centrex group,

and DN set
Configure a call
prefix
Configure a call prefix
Configure PBX
users
Configure the device identifier and

user identifier for each PBX user
Configure a trunk
group
Create a trunk group and add trunks

to the group
Configure call
routes
Configure call attributes, validity

period, and trunk circuit selection
mode for each call route
Configure a SIP
server
(Optional) Skip this step when there

is no SIP user
Configure SBC
proxy
(Optional) Configure the SBC proxy

function
Configure a CDR
server
(Optional) Skip this step when CDRs

do not need to be recorded
Configure
individual services
(Optional) Configure individual

services as required
Configure group
services
(Optional) Configure group services

as required
End
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2.4.1 Setting Parameters for an FXS Interface

This section describes how to set parameters for an FXS interface.
An FXS interface connects to a POTS phone. To achieve high transmission efficiency on an
FXS interface, properly set parameters for the FXS interface on the AR2200, including physical
attributes, electrical attributes, and KC attributes.
Before setting parameters for an FXS interface, complete the following task:
l
Configuring the AR2200 to work in PBX mode
Data Preparation
To set parameters for an FXS interface, you need the following data.
No.
Data
Polarity reversal pulse level width, polarity reversal mode, and dialing mode
Current, impedance, send gain, and receive gain
High-level pulse width, low-level pulse width, KC accounting mode, and voltage
operating
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
port fxs slotid/subcardid/portid
The FXS interface view is displayed.

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Step 4 Set physical attributes for the FXS interface.

l Run:
reverse-pole-pulse enable
The polarity reversal pulse function is enabled.

l Run:
reverse-pole-pulse level level-value
The polarity reversal pulse level width is set.

l Run:
polarity-reverse-mode { hard
| soft }
The polarity reversal mode is set.

l Run:
clip-reverse-pole-pulse enable
The CLIP polarity reversal pulse function is enabled.

l Run:
dial-mode { dtmf | pulse | both }
The dialing mode is set.

l Run:
line-lock enable
Port locking is enabled.

l Run:
nlp-mode { disable | normal | ec-gain-adjust | enhance | reduce }
The non-linear processing (NLP) mode is set.

Step 5 Set electrical attributes for the FXS interface.
l Run:
electric current current-value
The current is set.

l Run:
electric impedance impedance-value

l Run:
electric send-gain sendgain-value
The send gain is set.

l Run:
electric receive-gain recvgain-value
The receive gain is set.

Step 6 Set KC attributes for the FXS interface.
l Run:
kc high-level highlevel
The high-level pulse width is set.

l Run:
kc low-level lowlevel
The low-level pulse width is set.

l Run:
kc type { 16kc | 12kc }
The KC accounting mode is set.

l Run:
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kc voltage voltage
The voltage operating is set.

----End

Run the display voice port fxs [ state ] [ slotid/subcardid/portid ] command to check the FXS
2.4.2 Setting Parameters for an FXO Interface

This section describes how to set parameters for an FXO interface.
An FXO interface connects to a PSTN. To achieve high transmission efficiency on an FXO
interface, properly set parameters for the FXO interface on the AR2200, including the gain,
impedance, ring current, and feed.
Before setting parameters for an FXO interface, complete the following task:
l
Data Preparation
To set parameters for an FXO interface, you need the following data.
No.
Data
Gain, impedance, ring current, and feed
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
port fxo slotid/subcardid/portid
The FXO interface view is displayed.

Step 4 Run:
gain { send sendgain-value | receive recvgain-value }
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The gain is set.

NOTE
If the RBT played for the caller through the FXO port is too loud, the FXO port may fail to detect the busy
tone from the caller; therefore, if the caller hangs up before the call is answered, the called user still hears
the ring tone for a period. This problem can be solved by adjusting the send gain on the FXO interface.
Step 5 Run:
impedance { DC value
| AC value }

Step 6 Run:
ring { minimum-duration min-duration-value | maximum-interval max-interval-value |
frequence frequence-value | AC-amplitude AC-amplitude | offset offset-value }
The ring current is set.

Step 7 Run:
feed { maximum-voltage max-vol-value | minimum-voltage min-vol-value | maximumcurrent max-current-value | minimum-current min-current-value }
The feed is set.

----End

Run the display voice port fxo [ state ] [ slotid/subcardid/portid ] command to check the FXO
2.4.3 Setting parameters for a BRA Interface

This section describes how to set parameters for a BRA interface.
A basic rate access (BRA) interface connects to an ISDN phone. On the AR2200, you can enable
the BRA interface Layer 2 monitoring, remote power supply, automatic deactivation, and alarm
functions, and set the working mode and Layer 1 activation mode on a BRA interface.
Before setting parameters for a BRA interface, complete the following task:
l
Ensuring that the 2BST board is working properly
Data Preparation
To set parameters for a BRA interface, you need the following data.
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No.
Data
Interface working mode, automatic deactivation delay, and Layer 1 activation mode

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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
port bra slotid/subcardid/portid
The BRA interface view is displayed.

Step 4 Run:
l2-monitor enable
BRA interface Layer 2 monitoring is enabled.

Step 5 Run:
work-mode { p2p | p2mp }
The working mode is set.

Step 6 Run:
remote-power enable
Remote power supply is enabled.

Step 7 Run:
auto-deactive enable
Automatic deactivation is enabled.

Step 8 Run:
auto-deactive delay delay
The automatic deactivation delay is set.

Step 9 Run:
active-mode { unstable | stable }
The Layer 1 activation mode is set.

Step 10 Run:
alarm enable
The alarm function is enabled.

----End

Run the display voice port bra [ state ] [ slotid/subcardid/portid ] command to check the BRA
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2.4.4 Setting Parameters for a PRA Interface

This section describes how to set parameters for a PRA interface.
A PRA interface connects to a PBX or PSTN network. On the AR2200, you can enable the
CRC4 check, E1 interface Layer 2 monitoring, and E1 interface pulse code modulation (PCM)
alarm functions, and set the CRC alarm threshold and E1 interface signaling mode on a PRA
interface.
Before setting parameters for a PRA interface, complete the following task:
l
Ensuring that the 1E1T1-M/2E1T1-M board is working properly
Data Preparation
To set parameters for a PRA interface, you need the following data.
No.
Data
Access digital section mode
CRC alarm threshold and signaling mode of the E1 interface
Procedure
Step 1 Run:
system-view

Step 2 Run:
set workmode slot slot-id
e1t1 e1-voice
The E1 interface board is configured as an E1 voice card.

Step 3 Run:
voice

Step 4 Run:
port ve1 slotid/subcardid/portid
The PRA interface view is displayed.

Step 5 Run:
access-mode { digital | direct }
The access digital section mode is set.

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Step 6 Run:
crc4 enable
CRC4 check is enabled.

Step 7 Run:
crc-alarm-threshold { es es-threshold | cses cses-threshold | dm dm- threshold }
The CRC alarm threshold is set.

Step 8 Run:
l2-monitor enable
E1 interface Layer 2 monitoring is enabled.

Step 9 Run:
pcm-alarm
The PCM alarm function is enabled.

Step 10 Run:
signal { CCS | CAS }
The signaling mode is set.

----End

Run the display voice port ve1 [ state ] [ slotid/subcardid/portid ] command to check the PRA
2.5 Setting PBX Parameters

This section describes how to set PBX parameters.
2.5.1 Configuring the AR2200 to Work in PBX Mode

You can configure the AR2200 to work in SIP AG or PBX mode. Before configuring PBX
service features, configure the AR2200 to work in PBX mode. You can run the display voice
service-mode command to view the working mode of the AR2200. If the AR2200 works in SIP
AG mode, delete the SIP AG configurations and configure the AR2200 to work in PBX mode.
If the AR2200 works in PBX mode, skip this configuration.
Before configuring the AR2200 to work in PBX mode, complete the following task:
l
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
service-mode pbx
The AR2200 is configured to work in PBX mode.

Step 4 Run:
quit

Step 5 Run:
quit

Step 6 Run:
save
The configuration is saved.

Step 7 Run:
reboot
The AR2200 is restarted.

NOTE
After the AR2200 is configured to work in PBX mode, restart the AR2200 to make the configuration take
effect.
----End

After the configuration is complete, the AR2200 works in PBX mode.
Run the display voice service-mode command to check the working mode of the AR2200.
<Huawei> display voice service-mode
The voice service mode is PBX
2.5.2 Configuring Media and Signaling IP Address Pools

Media and signaling IP address pools provide media and signaling IP addresses for SIP AG
interfaces.
Context
A SIP AG interface must obtain media and signaling IP addresses from media and signaling IP
address pools respectively. The signaling IP address pool stores IP addresses of PBX interfaces
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and the media IP address pool stores IP addresses of media streams. Media streams and signaling
streams can use the same IP address. Media and signaling IP addresses must be available and
routes are reachable.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
voip-address media interface interface-type interface-number { ip-address |
dynamic }
A media IP address pool is configured.

Step 4 Run:
voip-address signalling interface interface-type interface-number { ip-address |
dynamic }
A signaling IP address pool is configured.

----End
2.5.3 (Optional) Setting System Parameters

This section describes how to set system parameters.

Different countries and regions use different voice parameter standards; therefore, set voice
parameters on the SIP AG in accordance with local standards.
Before setting PBX parameters, complete the following task:
l
Data Preparation
To set PBX parameters, you need the following data.
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No.
Data
Country/Region identifier
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No.
Data
Country code
Area code of national toll calls
Lower threshold and upper threshold for hookflash pressing
MWI mode
Codec mode for voice services
AC amplitude of the ringing current, frequency of the ringing current, and cadence
ratio
CLIP parameters in offhook state:

l Interval between the time when the ACK message is received and the time when
the FSK is transmitted
l Maximum duration between the time when the DT-AS is transmitted and the time
when the ACK message is received
l Level of the FSK
CLIP parameters in onhook state:

l Number of bits seized by a channel
l Interval between the time when the DT-AS is transmitted and the time when the
FSK is transmitted
l Level of the FSK
10
Maximum uplink bandwidth and reserved bandwidth for emergency calls
Configuring a Country/Region Identifier

This section describes how to configure a country/region identifier.
A country/region identifier is configured on a SIP AG so that user terminals connect to the SIP
AG can comply with the local standard.
Procedure
Step 1 Run:
system-view
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Step 2 Run:
voice

Step 3 Run:
country { brazil | britain-msfuk | britain-etsi | bulgaria | china-hongkong
china-mainland | egypt | france | singapore | thailand }
A country/region identifier is configured.

By default, a country/region identifier is China mainland.
----End
Configuring a Country/Region Code

Context
Before configuring a country/region code, run the display voice country-code command to
view the pre-configured country/region codes on the AR2200.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
country-code country-code-value [ country-name country-name | international-prefix
international-prefix-value | national-prefix national-prefix-value ]*
A country/region code is configured.

NOTE
The AR does not support the user-defined country code and region code. If the user-defined country code
and region code are used, communication may fail.
To configure the prefix of an international toll call, specify international-prefix international-prefixvalue.
To configure the prefix of a national toll call, specify national-prefix national-prefix-value.
Step 4 Run:
pbx { default-country-code dcc-value | default-area-code dac-value }*
The default country/region is configured.

----End
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Configuring the Area Code of Toll Calls

Context
Before configuring the area code of toll calls, run the display voice country-code country-codevalue command to view the pre-configured area codes.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
area-code country-code-value area-code-value [ area-name area-name-value ]
The area code of toll calls is configured.

----End
Setting Upper and Lower Thresholds for Hookflash Pressing

The upper and lower thresholds for hookflash pressing determine the hookflash duration.
Context
Hookflash or flash is a button on a telephone that simulates quickly hanging up and then picking
up again (a quick off-hook/on-hook/off-hook cycle). The hookflash can be pressed by a calling
party or a called party:
l
Hookflash pressed by a called party: If the called party user A wants to transfer an incoming
call to user B, user A can press the hookflash and dial the number of user B.
Hookflash pressed by a calling party: User A calls user B. User B answers the call and talks
with user A. User A can press the hookflash and dial the number of user C after hearing a
special dial tone.
After user A presses the hookflash:

l
If user C is busy, user A can press the hookflash and talk with user B.
If user C does not respond for a long period of time, user A can press the hookflash and
talk with user B.
If the phone of user C rings, user A hangs up and user B hears the ringback tone. User C
picks up the phone and talks with user B.
Whether a called party can be transferred to a toll call is restricted by the outgoing right of
the called party.
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
flash-hook lower lower-value
The lower threshold for hookflash pressing is set.

By default, the lower threshold for hookflash pressing is 100 ms.
Step 4 Run:
flash-hook upper upper-value
The upper threshold for hookflash pressing is set.

By default, the upper threshold for hookflash pressing is 350 ms.
NOTE
The lower threshold for hookflash pressing must be 50 ms less than the upper threshold for hookflash
pressing.
----End
Setting the MWI Mode

You need to set the MWI mode according to user habits.
Context
If there are leave messages, the user device configured with the MWI function makes the
indicator on or plays a tone, indicating that there are leave messages. You can set the MWI mode
according to user habits.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
mwi-mode { fsk-with-ring | fsk-without-ring | high-voltage }
The MWI mode is set.

----End
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Configuring the Codec Mode for Voice Services

You need to configure the codec mode for voice services so that the user device can work in
compliance with the local standard.
Context
G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.
711 defines two main compression algorithms, the -law algorithm (used in North America &
Japan) and A-law algorithm (used in Europe and China). A-law encoding takes a 13-bit signed
linear audio sample as input. -law encoding takes a 15-bit signed linear audio sample as input.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pcm { a-law | u-law }
The codec mode is configured for voice services.

----End
Setting Ringing Parameters

You can set the AC amplitude of the ringing current, frequency of the ringing current, cadence
ratio, and initial ringing function on the AR2200 to meet local standards.
Context
Different countries and regions use different ringing standards. You can set the AC amplitude
of the ringing current to adjust the ringing tone volume, voice pitch, cadence ratio, and initial
ringing function on the AR2200 to meet local standards.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run the following command according to user requirements.
l
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ring frequency { 16hz | 25hz | 50hz }
The frequency of the ringing current is set.

l
Run:
user-defined-ring ring-index { first-ring first-ring-period | first-interval
first-interval | second-ring second-ring-period | second-interval secondinterval | third-ring third-ring-period | third-interval third-interval }*
The cadence ratio of a specified user-defined ring type is set.

l
Run:
stop-initial-ring { enable | disable }
The initial ringing function is enabled.

----End
Enabling the Function That Reduces the Feed on Locked Ports

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
park-feed enable
The function that reduces the feed on locked ports is enabled.

This function takes effect for only FXS ports.
----End
Setting CLIP Parameters in Offhook State

You can set CLIP parameters in offhook state on the AR2200 so that the AR2200 can work with
the phone terminal.
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be configured
on the AR2200 so that the AR2200 can work with the phone terminal. Generally, default
parameter settings are used. If CLIP parameters are not set properly, change relevant CLIP
parameters.
Procedure
Step 1 Run:
system-view

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Step 2 Run:
voice

l
Run:
clip offhook ack-fsk-interval ack-fsk-interval
The interval between the time when the ACK message is received and the time when the
frequency-shift keying (FSK) is transmitted in offhook state is set.
l
Run:
clip offhook dtas-ack-interval dtas-ack-interval
The maximum duration between the time when the dual tone-alerting signal (DT-AS) is
transmitted and the time when the ACK message is received in offhook state is set.
l
Run:
clip offhook dtas-duration dtas-dur-value
The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.
l
Run:
clip offhook dtas-level dtas-level
The level of the DT-AS in offhook state is configured.

l
Run:
clip offhook fsk-level fsk-level
The level of the FSK in offhook state is configured.

l
Run:
clip offhook mark-signal-bit mark-signal-bit
The number of bits of the FSK synchronization mask in offhook state is set.
----End
Setting CLIP Parameters in Onhook State

You can set CLIP parameters in onhook state on the AR2200 so that the AR2200 can work with
the phone terminal.
Context
When the CLIP service is registered, CLIP parameters in onhook state need to be configured on
the AR2200 so that the AR2200 can work with the phone terminal. Generally, default parameter
settings are used. If CLIP parameters are not set properly, change relevant CLIP parameters.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
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l
Run:
clip onhook channel-seize-bit channel-seize-bit
The number of bits seized by a channel in onhook state is set.

l
Run:
clip onhook dtas-duration dtas-dur-value
The duration of the DT-AS in onhook state is set.

l
Run:
clip onhook dtas-fsk-interval dtas-fsk-interval
The interval between the time when the DT-AS is transmitted and the time when the FSK
is transmitted in onhook state is set.
l
Run:
clip onhook dtas-level dtas-level
The level of the DT-AS in onhook state is configured.

l
Run:
clip onhook fsk-level fsk-level
The level of the FSK in onhook state is configured.

l
Run:
clip onhook mark-signal-bit marksignal-bit
The number of bits of the FSK synchronization mask in onhook state is set.
----End
Setting Uplink Bandwidth Parameters

This section describes how to set uplink bandwidth parameters.
Context
The AR2200 provides uplink bandwidth control. When the system detects that the uplink
bandwidth usage reaches the configured upper threshold, it restricts calls and generates an alarm.
If the uplink bandwidth is insufficient, the system processes calls based on user levels. Common
users may not obtain services.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

l
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media-bandwidth-control enable
The AR2200 is enabled to restrict calls when the uplink bandwidth is enabled.
l
Run:
media-bandwidth-control maximum max-bandwidth
The maximum uplink bandwidth is set.

l
Run:
media-bandwidth-control reserved-for-emergency reserved-bandwidth
The reserved bandwidth is set for emergency calls.

NOTE
The reserved bandwidth for emergency calls must be smaller than the maximum uplink bandwidth
configured by the media-bandwidth-control maximum command.
----End

After the configuration of system parameters is complete, you can view the system parameters
such as the SIP interface's IP address, upper and lower thresholds for hookflash pressing, and
CLIP parameters.
Procedure
l
Run the display voice configuration command to check the voice configuration.
Run the display voice user-defined-ring [ring-index ] command to check user-defined

ring information.
Run the display voice clip command to check CLIP parameters.
Run the display voice sip-reg-count-per-second command to check the number of SIP
Register messages initiated per second.
----End
2.5.4 (Optional) Setting SIP Stack Parameters

This section describes how to set SIP stack parameters.

Before setting SIP stack parameters, familiarize yourself with the applicable environment,
SIP is an IETF-defined signaling protocol widely used for controlling communication sessions
such as voice and video calls over Internet Protocol (IP). SIP, RTP, RTCP, RTSP, and other
protocols constitute a SIP protocol stack.
Before setting SIP protocol stack parameters, complete the following task:
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Data Preparation
To set SIP protocol stack parameters, you need the following data.
No.
Data
Minimum value of the session timer, value of the unreliable provisional

response retransmission timer, value of the reliable provisional response
retransmission timer, value of the T1 timer, value of the T2 timer, value of the
T4 timer, and value of the TD timer
Values of the Max-forwards field, Organization header field, Server header

field, and User-Agent header field
Setting the Values of Timers

This section describes how to set values of timers of the SIP stack.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sip

l
Run:
entity-based-sessions-timer enable
The session timer defined by RFC 4028 based on the UA is defined.

l
Run:
min-se min-seperiod
The minimum value of the session timer is set.

l
Run:
session-progress session-progressperiod
The value of the unreliable provisional response retransmission timer is set.

l
Run:
session-rel-progress session-rel-progressperiod
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The value of the reliable provisional response retransmission timer is set.

l
Run:
t1 t1period

l
Run:
t2 t2period

l
Run:
t4 t4period

l
Run:
td tdperiod
The value of the TD timer is set.

----End
Setting Values of Header Fields

This section describes how to set values of header fields.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sip

l
Run:
header-folding enable
Header folding in the authentication header field is enabled.

l
Run:
field-header max-forwards max-forwards-value
The value of the Max-forwards header field is set.

l
Run:
field-header organization organization-head
The value of the Organization header field is set.

l
Run:
field-header server server-head
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The value of the Server header field is set.

l
Run:
field-header user-agent user-agent-head
The value of the User-agent header field is set.

----End

After the configuration is complete, you can view the configured SIP stack parameters.
Procedure
l
Run the display voice sip command to check SIP parameters.
----End
2.5.5 (Optional) Setting DSP Parameters

This section describes how to set DSP parameters.

Before setting DSP parameters, familiarize yourself with the applicable environment, complete
the pre-configuration tasks, and obtain the data required for the configuration. This will help
you complete the configuration task quickly and accurately.
The digital signal processing (DSP) collects, converts, filters, measures, enhances, compresses,
or identifies signals and coverts the signal from an analog to a digital form.
The DSP module converts analog voice signals into digital signals and stores a certain number
of digital signals into packets for transmission. To improve the voice communication quality,
the DSP needs to further process voice signals.
Before setting DSP parameters, complete the following task:
l
Data Preparation
To set DSP parameters, you need the following data.
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No.
Data
VBD switching mode
Default DSP channel code type and default interval at which the DSP channel
packetizes RTP packets
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No.
Data
T.30 redundancy parameter value of the T.38 fax, T.4 redundancy parameter value
of the T.38 fax, fax training mode, and maximum fax training rate
Alarm threshold of the dynamic jitter buffer, initial value of the dynamic jitter buffer,
maximum value of the dynamic jitter buffer, maximum value of the static jitter buffer,
minimum value of the dynamic jitter buffer, minimum value of the static jitter buffer,
and initial value of the static jitter buffer of a DSP channel
RTP payload type value, G.726-16k payload type value, G.726-24k payload type
value, G.726-32k payload type value, G.726-40k payload type value, NTE payload
type value, redundancy payload type value, and VBD payload value of a DSP channel
Interval at which a DSP channel sends RTCP packets and threshold for the number
of severe degrade seconds
Resource threshold in hierarchical control mode
Data event transmission mode, special process, DTMF transmission mode, echo
cancellation function, input gain, output gain, jitter buffer mode, NLP mode, and DSP
working mode in a DSP template
DSP loopback type and DSP channel parameters
Enabling Echo Cancellation on a DSP Channel

This section describes how to enable echo cancellation on a DSP channel.
Context
A user may hear the user's echo in the phone receiver in a conversation. If a proper delay in the
transmitted or received signal is set, the echo can be removed. If the delay exceeds 25 ms, the
voice quality deteriorates and the conversation ends. You can enable echo cancellation on a DSP
channel to remove echoes.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
echo enable
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Echo cancellation is enabled on the DSP channel.

----End
Enabling the PLC Function on a DSP Channel

This section describes how to enable the packet loss concealment (PLC) function on a DSP
channel.
Context
PLC is a technique that masks the effects of packet loss in VoIP communications. PLC is
effective only when the packet loss ratio is low. During communication, the average packet loss
ratio may be low, but a high burst packet loss ratio results in severe voice quality deterioration.
PLC can insert a static frame in the place where a packet is lost, regenerate a packet received
prior to the lost one, or generate an analog voice packet. If packets are lost during communication
and PLC is not used, the voice communication is interrupted. You can use a proper PLC
algorithm to minimize effects of packet losses.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
plc enable
The PLC function is enabled on the DSP channel.

----End
Enabling Silence Compression on a DSP Channel

You can enable silence compression on a DSP channel to reduce the number of sent voice
packets.
Context
To save network bandwidth, enable silence compression on a DSP channel. When no voice is
detected, the encoder generates short silence codes, but does not generate voice compression
codes. In addition, the encoder notifies the receiver of silence start until the voice is restored.
The silence compression function reduces the number of sent voice packets.
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
silence enable
Silence compression is enabled on the DSP channel.

----End
Setting the VBD Switching Mode

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
autovbd { auto | host-controlled }
The VBD switching mode is configured.

NOTE
By default, the VBD switching mode is host-controlled.
----End
Setting Fax Parameters

This section describes how to set fax parameters.
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Set fax parameters.
l
Run:

l
Run:

l
Run:
fax training-mode { e2e | local }
The fax training mode is set.

l
Run:
fax training-rate { v17 | v29 | v27 }
The maximum fax training rate is set.

l
Run:
fax v8negotiate enable
V8 negotiation is enabled.
----End
Setting Jitter Buffer Parameters

This section describes how to set jitter buffer parameters to improve the voice quality.
Context
Delay variations in voice packet arrival time can occur because of network congestion or route
changes. To reduce sound distortion caused by the delay jitter and packet loss, a jitter buffer is
used. You can set proper jitter buffer parameters to minimize delay variations so that packets
can be processed in a timely manner and smooth voice communication can be provided as much
as possible.
Procedure
Step 1 Run:
system-view
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Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Set parameters for the Jitter Buffer.
l
Run:
jitter-buffer adapt-jb-threshold adapt-jb-threshold
The alarm threshold of the dynamic jitter buffer is set.

l
Run:
jitter-buffer init-adapt-jb init-adapt-jb-value
The initial value of the dynamic jitter buffer is set.

l
Run:
jitter-buffer init-fixed-jb normal-fixed-jb-value
The initial value of the static jitter buffer is set.

l
Run:
jitter-buffer max-adapt-jb max-adapt-jb-value
The maximum value of the dynamic jitter buffer is set.

l
Run:
jitter-buffer min-adapt-jb min-adapt-jb-value
The minimum value of the dynamic jitter buffer is set.

l
Run:
jitter-buffer max-fixed-jb max-fixed-jb-value
The maximum value of the static jitter buffer is set.

l
Run:
jitter-buffer min-fixed-jb min-fixed-jb-value
The minimum value of the static jitter buffer is set.

----End
Setting the Payload Type Value

This section describes how to set the payload type value.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
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Step 3 Run:
dsp-attribute

Step 4 Set Payload Type parameters.
NOTE
The following payload type values must be different.
Run:
payload-type clear-mode clearmode-value
The RTP payload type value of a DSP channel is set.

l
Run:

l
Run:

l
Run:

l
Run:

l
Run:
payload-type nte nte-value
The NTE payload type value of a DSP channel is set.

l
Run:
payload-type redundancy redundancy-value
The redundancy payload type value of a DSP channel is set.

l
Run:
payload-type vbd vbd-value
The VBD payload type value of a DSP channel is set.

----End
Setting RTCP Parameters

This section describes how to set RTP control protocol (RTCP) parameters.
Context
RTCP monitors the quality of service and conveys information about participants in an on-going
session. RTCP periodically sends packets to all the participants in the session to monitor the
quality of service and obtain identity information about the participants.
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

l
Run:
rtcp rtcp-interval rtcp-interval
The interval at which a DSP channel sends RTCP packets is set.

l
Run:
rtcp rtcpxr enable
The RTP Control Protocol Extended Reports (RTCP XR) function is enabled.
l
Run:
rtcp sev-degradethreshold sev-degradethresholdval
The threshold for the number of severe degrade seconds is set.

l
Run:
rtcp vqm enable
The Voice Quality Monitor (VQM) function is enabled.

----End
Setting DSP Resource Parameters

This section describes how to set DSP resource parameters.
Context
DSP resources are limited and users have different requirements for DSP resources. To control
and allocate DSP resources properly, set the DSP resource control mode and the resource
threshold in hierarchical control mode.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
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Step 3 Run:
dsp-attribute

l
Run:
resource-threshold mode { priority | normal }
The DSP resource control mode is configured.

l
Run:
resource-threshold { threshold1 threshold1 | threshold2 threshold2 |
threshold3 threshold3 }
The resource threshold in hierarchical control mode is set.

The resource threshold in hierarchical control mode can be set only when the DSP resource
control mode is priority.
----End
Configuring a DSP Template

This section describes how to configure a DSP template.
Context
To customize DSP parameters for data services, configure a DSP template. After a DSP template
is configured, specify the template for users according to the port and phone number. The DSP
template improves the call connection rate. After a template is specified successfully, parameters
in the DSP template take effect immediately.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

Step 4 Run:
template name
A DSP template is created.

Step 5 Set parameters in the DSP template.
l
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data-event data-event-value
The data event transmission mode is configured.

l
Run:
dsp-special-flow dsp-special-flow-value
A special process is configured.

l
Run:
dtmf dtmf-value
The Dual-Tone Multi-frequency (DTMF) transmission mode is configured.

l
Run:
echo-cancel { enable | disable }
The echo cancellation function is enabled or disabled.

l
Run:
input-gain input-gain-value
The input gain is set.

l
Run:
output-gain output-gain-value
The output gain is set.

l
Run:
jitter-buffer { dynamic | static }
The jitter buffer mode is configured.

l
Run:
nlp nlp-value
The normal link pulse (NLP) transmission mode is configured.

l
Run:
work-mode work-mode-value
The DSP working mode is configured.

l
Run:
vbd redundancy
Voice band data (VBD) redundancy transmission is enabled.

----End
Configuring a DSP Channel

This section describes how to configure a DSP channel.
Context
You can enable a digital signal processor (DSP) channel to work in loopback mode, and set the
loopback mode (PCM-side loopback test and IP-side loopback test). When the DSP channel
between the calling party and called party cannot transmit signals or can transmit signals only
in one direction, run the loop-back command to locate the fault. If the calling party hears the
echo in a PCM-side loopback test, the speech channel between the calling phone and the calling
DSP channel is functioning properly. If the called party hears the echo in an IP-side loopback
test, the speech channel between the called phone and the calling DSP channel is functioning
properly.
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To control resources of DSP channels, prohibit the DSP channels. The prohibited DSP channels
cannot participate in resource allocation.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp slot/dsp-index
The DSP resource management view is displayed.

l
Run:
loop-back loopback-type channel
The loopback function is configured on a DSP channel.

l
Run:
prohibit channel [ count ]
DSP channels are prohibited.

----End

After the configuration of DSP parameters is complete, you can view the DSP parameters
including fax parameters and jitter buffer parameters.
Procedure
l
Run the display voice dsp-attribute command to check the DSP configuration.
Run the display voice dsp state { slot/dsp-index | channel slot/dsp-index/channel }

command to check the status of a DSP or DSP channel.
Run the display voice dsp-template command to check the DSP template configuration.
----End
2.6 Configuring an Enterprise

After an enterprise is created, you can configure the Centrex group, prefix, and DN set.
When multiple enterprises access a PBX, the PBX can be divided into multiple virtual PBXs so
that the enterprises can use one PBX. Configuring enterprises on the PBX facilitate user
management.
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Before configuring an enterprise, complete the following task:
l
Data Preparation
To configure an enterprise, you need the following data.
No.
Data
Enterprise name
(Optional) Centrex group
DN set
(Optional) Ring delay
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
enterprise enterprise-name
An enterprise is created and the enterprise view is displayed.

By default, the enterprise default exists on the AR2200.
Step 4 Run:
centrex centrex-name [ description description ]
A Centrex group is created.

By default, no Centrex group is configured for an enterprise.
centrex-amount centrex-amount-value
By default, a maximum of 16 Centrex groups can be configured in the enterprise view for
different devices
l AR2220: 16
l AR2240: 32
Step 6 Run:
dn-set dn-set-name [ description description ]
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A DN set is configured.
By default, the default DN set defaultdialplan is generated after an enterprise is created.
description enterprise-name
The description of the enterprise is configured.

By default, no description is configured for an enterprise.
----End

Run the display voice enterprise [ enterprise-name ] command to check the enterprise
configuration.
2.7 Configuring a Call Prefix

A call prefix is located at the beginning of a called number.
Usage Scenario
A call prefix, an important attribute of the call service, defines a call number rule and describes
the call number distribution and routing plans in an exchange office.
Before configuring a call prefix, complete the following task:
l
Configuring an enterprise
Data Preparation
To configure a call prefix, you need the following data.
No.
Data
Call prefix profile name, enterprise and dial number (DN) set to be bound to the call
prefix, call type, call attribute, maximum length and minimum length of a number
that can be parsed, (optional) home area attribute, and (optional) ringing delay
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice
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Step 3 Run:
callprefix callprefix-name
A call prefix profile is created and its view is displayed.

By default, no call prefix profile is configured.
Step 4 Run:
prefix prefix
A call prefix is configured.

Step 5 Run:
enterprise enterprise-name [ dn-set dn-set-name ] [ centrex centrex-name ]
An enterprise, Centrex group, and DN set are bound to a call prefix.

By default, no enterprise, Centrex group, or DN set is bound to a call prefix.
Step 6 Run:
call-type category callcategory attribute attribute
The call type and call attribute are configured for the call prefix.
By default, no call type or call attribute is configured for a call prefix.
Step 7 Run:
digit-length maximum-length-value minimum-length-value
The length of a number that can be parsed is set.

By default, the length of a number that can be parsed is not set.
NOTE
The maximum length of a number that can be parsed must be greater than or equal to the minimum length
of a number that can be parsed.

destination-location { intra-office | inter-office }
The home area attribute of the call prefix is configured.

By default, the home area attribute of a call prefix is intra-office.
ring-delay ringdelay-value
The ringing delay is configured.

By default, the ringing delay is 0s.
----End

Run the display voice callprefix [ callprefix-name ] command to check the call prefix
configuration.
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2.8 Configuring a PBX User

A user identifier includes the user number and DN set.
Usage Scenario
When a user is needed for the voice service, you need to create a PBX user on the PBX.
Before configuring a PBX user, complete the following tasks:
l
Configuring the device to work in PBX mode
Setting global PBX parameters
Data Preparation
To configure an enterprise, you need the following data.
No.
Data
User name
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name [ msn-sub | { pots | sipue | bra } [ enterprise enterprise-name ] ]
A PBX user is created and the PBX user view is displayed.

By default, no PBX user is configured.
Step 4 Run the following commands to configure a port or device ID for the PBX user.
l
Run the port slotid/subcard/portid command to configure a port for the PBX user.
Run the sipue eid-value command to configure a device ID for the SIP UE user.
Step 5 Run:
telno [ country-code country-code-value ] [ area-code area-code-value ] telno-value
The country code, area code, and telephone number of the PBX user are configured.
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By default, no country code, area code, or telephone number is configured for a PBX user.
Step 6 Run:
dn-set dn-set-name
A DN set is bound to the PBX user identifier.

By default, no DN set is configured for a PBX user.
centrex centrex-name centrex-telno-value
The Centrex group that the PBX user belongs to is configured.

By default, no Centrex group is configured.
call-right { in in-right-value | out out-right-value }*
The incoming and outgoing call rights are configured.

By default:
l Incoming call: Only intra-office users can make calls to intra-office users.
l Outgoing call: Only intra-office users can receive calls from intra-office users.
user-type type
The user type is configured.

By default, users are common users.
user-status status
The user service status is configured.

By default, the user service status is correct.
eid-para description description
The description of the PBX user is configured.

By default, no description is configured for a PBX user.
----End

Run the display voice pbxuser command to check the configuration of the PBX user.
2.9 (Optional) Configuring a SIP Server

A SIP server is the main component of the IP PBX and establishes SIP sessions.
Usage Scenario
If SIP users need to communicate with each other through the AR2200, configure a SIP server.
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Before configuring a SIP server, complete the following tasks:
l
Setting global PBX parameters
Assigning IP addresses to interfaces
Data Preparation
To configure a SIP server, you need the following data.
No.
Data
Signaling IP address and signaling port number of the SIP server
Media IP address of the SIP server
Uniform Resource Identifier (URI) of the SIP server
Home domain of the SIP server
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sipserver
The SIP server view is displayed.

Step 4 Run:
signaling-address{ ip ip-address | addr-name addr-name-value } port port-value
A signaling IP address and signaling port number are configured for the SIP server.
By default, no signaling IP address or signaling port number is configured.
Step 5 Run:
signalling-domain signaling-domain-value
The signaling domain name is configured for the SIP server using a dynamic signaling IP address.
NOTE
When a SIP server uses a dynamic signaling IP address, configure a signaling domain name for the SIP
server.
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Step 6 Run:
ddns-client ddns-client-name
A dynamic domain name system (DDNS) name is configured for a SIP server using a dynamic
signaling IP address.
NOTE
When a SIP server uses a dynamic signaling IP address, configure a DDNS client name for the SIP server so
that the DDNS server can update the mapping between the signaling domain name and IP address.
Step 7 Run:
media-ip { ip-address | addr-name addr-name-value }
The media IP address of the SIP server is configured.

By default, no media IP address of the SIP server is configured.
Step 8 Run:
register-uri uri
The URI of the SIP server is configured.

By default, the URI of the SIP server is not configured.
Step 9 Run:
home-domain domain
A home domain that the SIP server belongs to is configured.

By default, no home domain is configured.
Step 10 Run:
reset
The SIP server is reset.
CAUTION
Exercise caution when you run this command because resetting a SIP server interrupts running
services.
----End

Run the display voice sipserver command to check the SIP server configuration.
2.10 (Optional)Configuring SBC Proxy Function

for SIPUEs.
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2.10.1 Configuring the SBC Proxy Function (Public IP Address for

the SIP Server and Private IP Addresses for SIPUEs)
for SIPUEs when the SIP server uses a public IP address and the SIPUEs use private IP addresses.
As shown in Figure 2-5, the SIP server (router A) at the headquarters uses a public IP address,
and traveling SIPUEs and some SIPUEs in the branch use private IP addresses. To provide voice
services through the PBX at the headquarters for SIPUEs, the SBC proxy function must be
configured on router A.
Figure 2-5 SBC proxy configuration networking diagram (public IP address for the SIP server
and private IP addresses for SIPUEs)
Branch
SIP UE
Enterprise A
Router C
SIP trunk
Router B
ISDN
IP network
SIP UE
(Travelling staff)
SIP trunk
Router D
Router A
Headquarters
User A
Enterprise A
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Before configuring the SBC proxy function, complete the following tasks:
l
Configuring the SIP server
Configuring the SIP trunk group
Configuring IP addresses to enable SIPUEs and SIP servers to communicate with each
other
Data Preparation
To configure the SBC proxy function, you need the following data.
No.
Data
Media trunk type, and signaling and media proxy policies for the SIP server
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sbc media-relay interface interface-type interface-number relay-type
The media relay interface and relay type are configured.

Step 4 Run:
sipserver

Step 5 Run:
sbc signalling-proxy value
The signaling proxy policy for the SIP server is configured.

NOTE
If all SIPUEs are located on the private network, set the value to Enable. If only some of the SIPUEs are located
on the private network, set the value to Auto.
Step 6 Run:
sbc media-proxy value
The media proxy policy for the SIP server is configured.

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NOTE
If all SIPUEs are located on the private network, set the value to Enable. If only some of the SIPUEs are located
----End

2.10.2 Configuring the SBC Proxy Function (Private IP Address for

the SIP Server and Public IP Addresses for SIP UEs)
for SIP UEs when the SIP server uses a private IP address and the SIP UEs use public IP
addresses.
Usage Scenario
As shown in Figure 2-6, the SIP server (router B) at the headquarters uses a private IP address,
and SIP UEs in the branch or on business trips use private IP addresses. The SBC proxy function
must be configured for the SIP server to provide voice services using the PBX at the headquarters
for SIP UEs.
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Figure 2-6 SBC proxy configuration networking diagram (private IP address for the SIP server
and public IP addresses for SIP UEs)
Headquarters
SIP UE
Enterpeise A
Router B
SIP trunk
ISDN
Router A
IP network
SIP UE
(Travelling staff)
l
Configuring SIP UEs
Configuring IP addresses to enable SIP UEs and SIP servers to communicate with each
other
Data Preparation
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No.
Data
Media trunk type, and signaling and media proxy policies for the SIP server
Mapping public signaling and media IP addresses of the SIP server
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:

Step 4 Run:
sipserver

Step 5 Run:
The signaling proxy policy for the SIP server is configured.

NOTE
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are located
Step 6 Run:
The media proxy policy for the SIP server is configured.

NOTE
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are located
Step 7 Run:
sbc mapped-signalling-address ip ip-address port port-value
A public IP address and port number are configured for signaling mapping of the SIP server.
Step 8 Run:
sbc mapped-media-ip ip-address
The public media IP address mapping the SIP server is configured.

----End
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2.10.3 Configuring the SBC Proxy Function (Private IP Address for

the SIP Server and Private IP Addresses for SIP UEs)
for SIP UEs when the SIP server uses a private IP address and the SIP UEs use private IP
addresses.
Usage Scenario
As shown in Figure 2-7, the SIP server (router A) at the headquarters uses a private IP address,
and SIP UEs in the branch or on business trips use private IP addresses. To provide voice services
through the PBX at the headquarters for SIP UEs, the SBC proxy function must be configured
on router A.
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Figure 2-7 SBC proxy configuration networking diagram (private IP address for the SIP server
and private IP addresses for SIP UEs)
Branch
SIP UE
Enterpeise A
Router C
SIP trunk
Router B
IP network
Headquarters
Router A
SIP trunk
Enterpeise A
l
Configuring SIP UEs
Configuring IP addresses to enable SIP UEs and SIP servers to communicate with each
other
Data Preparation
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No.
Data
Media proxy port number, media trunk type, and signaling and media proxy policies
for the SIP server
Mapping public signaling and media IP addresses of the SIP server
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:

Step 4 Configure the SBC proxy function for the SIP server.
l Run:
sipserver

l Run:
The signaling proxy policy is configured.

NOTE
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are
located on the private network, set the value to Auto.
l Run:
The media proxy policy is configured.

NOTE
l Run:
l Run:
sbc mapped-media-ip port-value
The public media IP address mapping the SIP server is configured.

l Run:
sbc mapped-media-port-start port-value
The media port number mapping the SIP server is configured.

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l Run:
sbc mapped-media-proxy-port-start port-value
The media proxy port number mapping the SIP server is configured.
l Run:
return
The user view is displayed.

Step 5 Configure the SBC proxy function for the SIP trunk group.
l Run:
system-view

l Run:
voice

l Run:
trunk-group name
The trunk group view is displayed.

l Run:
The media proxy policy is configured.

NOTE
l Run:
l Run:
sbc mapped-media-ip ip-address
The public media IP address mapping the SIP trunk group is configured.
l Run:
sbc mapped-media-port-start port-value
The media start port number mapping the SIP trunk group is configured.
l Run:
sbc mapped-media-proxy-port-start port-value
The media proxy start port number mapping the SIP trunk group is configured.
----End

2.11 (Optional)Configuring a CDR Server

The call detail record (CDR) of users can be queried in real time, and the CDR data can be
analyzed by using a third-party tool so that users can quickly know call fees.
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To record and display CDRs of users, specify a CDR server on the AR2200.
Before configuring a CDR server, complete the following tasks:
l
Assigning IP addresses to interfaces
Data Preparation
To configure a CDR server, you need the following data.
No.
Data
IP address and port number of the CDR server
User name and password
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
cdr-server
The CDR server view is displayed.

Step 4 Run:
server ip ip-address server-port port-number
The IP address and port number of the CDR server are specified.
By default, no IP address or port number is configured for a CDR server.
Step 5 Run:
username username password { cipher cipher-password | simple simple-password}
The user name and password of the CDR server is configured.

By default, no user name or password of the CDR server is configured.
----End

Run the display voice cdr-server command to check the CDR server configuration.
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2.12 Configuring a Trunk Group

A trunk group is associated with a call route to enable users from different exchange offices to
call each other.

Before configuring a trunk group, familiarize yourself with the applicable environment,
Usage Scenario
As shown in Figure 2-8, the AR2200 functions as a PBX and needs to communicate with other
PBXs or external networks. The AR2200 supports the following connection modes:
l
Connects to a PSTN network through an AT0 trunk.
Connects to a PSTN network through a PRA trunk.
Connects to an IMS network through a SIP trunk.
Connects to another PBX through a PRA trunk.
Connects to another PBX through a E1R2 trunk.
Connects to another PBX through a SIP trunk.
Connects to an IMS network through an H323 trunk.
Connects to another PBX through an H323 trunk.
Trunks for inter-office calls with the same attributes are added to a trunk group. A trunk group
is associated with a call route to establish call sessions between exchange offices.
Figure 2-8 Trunk application
PSTN
IMS
SIP/H323
AT0/PRA
AR
SIP/PRA/E1R2/H323
PBX
Before configuring a trunk group, complete the following tasks:
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Data Preparation
To configure a trunk group, you need the following data.
No.
Data
Common parameters of a trunk group: trunk group name, signaling type, DN set and
enterprise to be bound to the trunk group, and optional parameters including the callin right, call-out right, country code, area code, default number displayed, trunk
circuit selection mode, and trunk group description
Parameters of a SIP trunk group: SIP registration mode, home domain of the peer
SIP trunk, signaling IP address or dynamic signaling IP address name of the SIP trunk
group, media IP address or dynamic media IP address name of the SIP server,
signaling port of the local end, IP address and port of the remote end, registrar
Uniform Resource Identifier (URI), and optional parameters including the
authentication password, trunk group identifier, maximum number of concurrent
calls, transmission mode, numeral software parameter and its index, Dual-Tone
Multi-frequency (DTMF) parameters, fax/modem codec negotiation mode,
packetization interval, voice band data (VBD) attribute type, VBD codec mode, VBD
payload type, modem transmission mode, RFC 2198 negotiation start mode, RFC
2198 redundancy transmission start mode, RFC 2833 negotiation start mode, and
RFC 2833 fax and modem transmission
Parameter of a PRA trunk group: protocol used
Parameters of a SIP AT0 trunk: ID and password for registration, trunk name, called
number of incoming calls, and calling number of outgoing calls
Parameters of an AT0 trunk: trunk name, location of the Foreign Exchange Office
(FXO) interface connected to the trunk, signal transmission type for the calling line
identification presentation (CLIP) service, dial delay, call prefix, dial delay after the
call prefix is inserted, called number of incoming calls, dialing mode, and trunk status
Parameters of a PRA trunk: trunk name, and location of the physical interface
connected to the trunk
Parameters of a PRA trunk: trunk name and member interfaces in the PRA trunk
Parameters of an E1R2 trunk: R2 signaling type, sending parameter of R2 line
signaling and register signaling, receiving parameter of R2 line signaling and register
signaling, line signaling and register signaling of R2 profile, register's receiving
address of R2 profile, signaling profile name of E1R2, and multiple-language
adaptive signaling of R2 profile
2.12.2 Configuring a Trunk Group

Creating a Trunk Group
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Context
When the signaling type of a trunk group is dss1-net, dss1-user, qsig-net, or qsig-user, the trunk
group is a PRA trunk group.
When the signaling type of a trunk group is FXO, the trunk group is an AT0 trunk group.
When the signaling type of a trunk group is SIP, the trunk group is a SIP trunk group.
When the signaling type of a trunk group is e1-r2, the trunk group is an E1R2 trunk group.
When the signaling type of a trunk group is H323, the trunk group is an H323 trunk group.
In PBX mode:
l
When the AR2200 functions as a network-side device and uses DSS1 signaling to connect
to the remote network through a PRA trunk, set the signaling type to dss1-net.
When the AR2200 functions as a user-side device and uses DSS1 signaling to connect to
the remote network through a PRA trunk, set the signaling type to dss1-user.
When the AR2200 functions as a network-side device and uses QSIG signaling to connect
to the remote network through a PRA trunk, set the signaling type to qsig-net.
When the AR2200 functions as a user-side device and uses QSIG signaling to connect to
the remote network through a PRA trunk, set the signaling type to qsig-user.
When the AR2200 connects to the remote network through an AT0 trunk, set the signaling
type to FXO.
When the AR2200 connects to the remote network through a SIP trunk, set the signaling
type to SIP.
If the AR2200 uses R2 signaling to connect to a network through the PRA trunk and is used
as a user-side device, the signaling type is set to e1-r2.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name [ dss1-net | dss1-user | fxo | h323 h323-trunktype-value | qsignet | qsig-user | e1-r2 | sip register-mode ];
A trunk group is created and the trunk group view is displayed.

----End
Configuring Common Parameters of a Trunk Group

Procedure
Step 1 Run:
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system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name

Step 4 Set mandatory parameters for the trunk group.
l Run:
enterprise enterprise-name [ dn-set dn-set-name ]
An enterprise and a DN set are bound to the trunk group.

Step 5 Set optional parameters for the trunk group.
l Run:
select-mode selectmode-value
The trunk circuit selection mode is configured.

l Run:
call-right
{ in in-right-value | out out-right-value }
The incoming and outgoing call rights are configured for the trunk group.
l Run:
default-caller-telno country-code-value area-code-value value
The country code, area code, and default number displayed are configured for the trunk group.
l Run:
description desc-value
The description of the trunk group is configured.

----End
Setting Parameters for a PRA Trunk Group

Context
When the trunk group type is PRA, perform the following operations.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name
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Step 4 Run:
pra standard pra-standard-value
The protocol type is specified for the PRA trunk.

----End
Setting Parameters for a SIP Trunk Group

Context
When the trunk group type is SIP, perform the following operations.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name

Step 4 Set mandatory parameters for SIP trunks.
l Run:
mgc-type maxcr-value
The adaptation type of a SIP trunk group is configured.

l Run:
home-domain value
The home domain to which the SIP trunk of the peer device belongs is configured.
l Run:
signalling-address { ip ip-address | addr-name signal-addr-name-value } port
port-value
A signaling IP address and port number are configured for the SIP trunk group.
l Run:
media-ip {ip-address | addr-name addr-name-value}
The media IP address or dynamic media IP address name is configured for the SIP server.
l Run:
peer-address static primary-ip-value primary-port-value [ secondary
ip-value secondary-port-value ]
secondary-
The remote IP address and port are configured for the SIP trunk group.
l Run:
register-uri register-uri-value
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The registrar URI is configured.

l Run:
register-id register-uri-value password { simple password-value | cipher
password-value }
A trunk group ID and an authentication password are configured.

NOTE
The trunk authentication password and trunk group identifier need to be configured only when the SIP
trunk registration mode is trunk group registration.
Step 5 Set optional parameters for the SIP trunk group.

l Run:
maxcr maxcr-value
The maximum number of concurrent calls is set.

l Run:
transfer maxcr-value
The transmission mode is set.

l Run:
number-parameter numpara-no numpara-value
The numeral software parameter and numeral software parameter index are set.
l Run:
dtmf-transmission-mode { thoroughly | erase }
Dual-Tone Multi-frequency parameters are set.

l Run:
fax-modem common negotiation-mode { negotiate | self-switch }
The fax/modem codec negotiation mode is set.

l Run:
fax-modem common rtp-interval { 5ms | 10ms | 20ms | 30ms }
The packetization interval in transparent transmission mode is set.

l Run:
fax-modem common bd-attribute-type { chinatelecom | extchinatelecom | ietf |
v152 }
The VBD attribute type is set.

l Run:
fax-modem common vbd-codec { g711a | g711u }
The VBD codec mode is set.

l Run:
fax-modem common vbd-payload-type { dynamic | static }
The VBD payload type is set.

l Run:
fax-modem modem transmission-mode { thoroughly | relay }
The modem transmission mode is set.

l Run:
redundancy-negotiation-mode { negotiation | fixed-start | no-initiative-start }
The mode in which RFC 2198 redundancy transmission negotiation is started is set.
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l Run:
redundancy-start-mode { ordinary2198 | smart2198 }
The mode in which RFC 2198 redundancy transmission is started is set.

l Run:
redundancy-vbd enable
The PBX is enabled to transmit VBD using RFC 2198 redundancy transmission.
l Run:
nte-flashhook enable
RFC 2833 hookflash event transmission is enabled.

l Run:
nte-negotiation-mode { negostart | fixedstart | no-initiative-start }
The mode in which RFC 2833 transmission negotiation is started is set.

l Run:
nte-fax-modem enable
RFC 2833 fax and modem transmission is enabled.

----End
Setting Parameters for an H323 Trunk Group

Context
When an H323 trunk is used, the AR2200 can function as the gatekeeper and gateway and is
used in the following scenarios:
l
Peer mode
As shown in Figure 2-9, RouterA and RouterB are enterprise routers that can communicate.
Voice services between RouterA and RouterB are transmitted through the H323 trunk. The
H323 trunk uses the peer mode, so it does not need to be registered with the gatekeeper.
Figure 2-9 Networking in peer mode
N e tw
IP o rk
U s e rA
R o u te rA
GW
R o u te rB
GW
U se rB
G W s a re p e e rs
Registration mode
As shown in Figure 2-10, the gatekeeper is the carrier device and the gateway is the
enterprise device. The gateway can register with the gatekeeper to transmit voice services.
The gatekeeper needs to identify the registered gateway. After the gateway is registered on
the gatekeeper, the gatekeeper completes voice services initiated by the gateway. The
AR2200 does not function as the gateway, that is, the AR2200 does not send registration
packets.
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Figure 2-10 Networking in registration mode
IP o rk
N e tw
R o u te rB
GW
U se rA
G a te w a y re g iste rs w ith th e
g a te ke e p e r
GW
GK
R o u te rA
GK
Procedure
Step 1 Set H323 attributes.
l (Mandatory) Set H323 attribute.
Run:
system-view

Run:
voice

Run:
h323-attribute
The H323 configuration view is displayed.

Run:
localip ip-address
The default local IP address is configured.

l (Optional) Set H323 attributes.
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Action
Command
Remarks
Configure the
Q931
listening port.
q931-listenport q931listenport-value
By default, the Q931

listening port is 1720.
Configure the
AR2200
whether to
enable fast
start.
faststart-flag { enable| disable }
By default, fast start is

enabled.

NOTE
After the fast start function is
enabled, sessions can be set up
quickly.
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Action
Command
Remarks
Configure the
AR2200
whether to
support the
H245 control
signaling
channel.
support-h245tul { enable | disable }
By default, the AR2200 does

not support the H245 control
signaling channel.
Set the
operation
mode for the
H323 stack.
workmode workmode-value
The values are as follows:

l international
l national
l VT
By default, the operation
mode is international.
Configure the
UDP port
number of
RAS signals.
rasport rasport-value
By default, the UDP port

number of RAS signals is
1719.
Configure the
H323 stack
system name.
h323-systemname h323systemnamevalue
Configure the
H323 stack
product name.
productname productname-value
Configure the
H323 stack
version.
h323version h323version-value

l V2
l V3
l V4
l V5
The default H323 stack
version is V4.
Configure the
H245 version.
h245version h245version-value

l V2
l V3
l V4
l V5
The default H245 version is
V4.
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Action
Command
Remarks
Configure the
display
header field in
Q931
messages.
displayname displayname-value
Set the value

of the TCP
keepalive
timer.
tcp-keepalive-timerlen tcp-keepalivetimerLen-value
By default, the value of the

TCP keepalive timer is 3600
ms.
Configure the
AR2200
whether to
support
automatic
setup of the
H245
channel.
auto-connect-controlchannel { enable | By default, H245 channels

cannot be set up
disable }
automatically.
Configure the
H323 stack
whether to
receive the
H245 address
in Setup
messages.
earlyh245 { enable | disable }
By default, a Setup message

does not contain the H245
address.
Set the
timeout
interval for
TCP
connections.
tcp-connection-timeout tcp-connectiontimeout-value
By default, the timeout

interval for TCP connections
is 15000 ms.
Set the
timeout
interval for
Setup
messages.
setup-wait-response-timeout setupwait-response-timeout-value

interval for Setup messages
is 500000 ms.
Set the
timeout
interval for
Alerting and
Connect
messages.
callproc-wait-response-timeout
callproc-wait-response-timeout-value

interval for Alerting and
Connect messages is 500000
ms.
Set the
timeout
interval for
Connect
messages.
recvalert-wait-response-timeout
recvalert-wait-response-timeout-value

interval for Connect
messages is 500000 ms.

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Action
Command
Remarks
Set the
channel
establishment
timeout.
channels-timeout channels-timeoutvalue
By default, the channel

establishment timeout is
15000 ms.
Set the
timeout
interval of the
roundtrip
delay process.
roundtrip-delay-timeout roundtripdelay-timeout-value

interval is 15000 ms.
Set the
timeout
interval for
mode
switching.
requestmode-timeout requestmodetimeout-value

Set the
timeout
interval of the
media loop
process.
medialoop-timeout medialoop-timeoutvalue

Set the
timeout
interval for
the peer end to
send
EndSessionC
ommand
messages.
endsession-timeout endsession-timeoutvalue

Set the value

for the H323
timer.
h323-app-timer timerindex timerlen
l Run:
reset
The H323 system is reset.

NOTE
After setting H323 system parameters, run the reset command to reset the H323 system to make the
parameters take effect.
l Run:
quit
Return to the voice view.

Step 2 Set parameters for an H323 trunk group.
l Run:
trunk-group name
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l Run:
signalling-ip ip ip-address
The signaling IP address is configured for the H323 trunk group.

l Run:
media-ip { ip-address | addr-name addr-name-value }
The media IP address or dynamic media IP address name is configured for the H323 trunk
group.
l Run:
gwid gwid-value
The gateway ID of the peer device connected to the H323 trunk is configured.
NOTE
When the AR2200 functions as the gatekeeper, you must configure the gateway ID of the peer device.
l Run:
peer-address static primary-ip-value primary-port-value
The remote IP address and port number are configured for the remote H323 trunk of the H323
trunk group.
NOTE
When the AR2200 functions as the gateway, you must configure the remote IP address and port number.
l When there are trunk groups of various H323 types, perform the following operations.
Run:
techprefix techprefix-value
A technical prefix is configured.

Run:
send-techprefix h323-send-techprefix-value
The AR2200 is configured whether to send the technical prefix.

Run:
del-techprefix h323-send-techprefix-value
The AR2200 is configured whether to delete the technical prefix.

----End
Configuring E1R2 Trunk Group Parameters

Context
When the trunk group type is E1R2, perform the following operations:
Procedure
Step 1 Run:
system-view

Step 2 Run:
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voice

Step 3 Configure R2 signaling parameters:
l Run:r2 signalling-type signalling-type-name A R2 signaling is created and the R2 signaling
view is displayed. Or a R2 signaling view that has been created is displayed.
l Run:line-signal-send line-logic-cmd line-logic-signal-type { Line-phy-cmd-para Line-callstate Line-phy-send-time | no-process } The sending parameters of R2 line signaling are
configured.
l Run:line-signal-receive Line-phy-signal-type Line-phy-cmd Line-phy-cmd-para Line-callstate{ Line-logic-cmd | no-process } The receiving parameter of R2 line signaling is
configured.
l Run:register-signal-receive Register-phy-cmd Register-call-state { Register-logic-cmd
Register-logic-cmd-para | no-process } The receiving parameter of R2 register signaling is
configured.
l Run: register-signal-send Register-logic-cmd Register-logic-cmd-para { register-phycmd | no-process } The sending parameter of R2 register signaling is configured.
l Run:quit The voice view is displayed.
Step 4 Configure R2 profile parameters:
l Run: r2 profile profile-name A R2 profile is created and the R2 profile view is displayed.
l Run: ling-signalling { line-signalling-type line-signalling-type-value | wait-answer-time
wait-answer-time-value | wait-clear-forward-time wait-clear-forward-time-value | waitprotect-time wait-protect-time-value | wait-seize-ack-time wait-seize-ack-time-value } The
line signaling of R2 profile is configured.
l Run: register-address-receive { country-code value | distinctive-indication value | echocancellation value | circuit-character value | calling-number value | calling-user-type
value | calling-user-priority value | called-number value | calling-number-location
calling-number-location-value | user-type-location user-type-location-value } The register
receiving address of R2 profile is configured.
l Run: register-address-send { country-code value | distinctive-indication value | echocancellation value | circuit-character value | calling-number value | calling-user-type
value | calling-user-priority value | called-number value } The register sending address of
R2 profile is configured.
l Run: register-signalling { back-pulse-time back-pulse-time-value | calling-number-maxlength calling-number-max-length-value | compelling-sequence-keepalive-time
compelling-sequence-keepalive-time-value | dtmf-send-interval dtmf-send-interval-time |
dtmf-send-time dtmf-send-time-value | incoming-type incoming-type-value | numbermax-interval number-max-interval-time | number-send-mode number-send-mode-value |
outgoing-type outgoing-type-value | receive-number-max-interval receive-number-maxinterval-time | send-pulse-signal signal-value | signal-send-voltage signal-send-voltagevalue | signal-stop-permit-time signal-stop-permit-time-value | wait-Bsignal-time waitBsignal-time-value | wait-caller-category-time wait-caller-category-time-value } The
register signaling of R2 profile is configured.
Step 5 Run:
signalling-type signalling-type-name
The multi-language adaptive signaling of R2 profile is configured.

Step 6 Run:
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quit

Step 7 Run:
trunk-group name

Step 8 Run:
r2-profile profile-name
The E1R2 signaling profile used by the E1R2 trunk group is configured.
Step 9 Run:
r2-receive-earlymedia r2-receive-earlymedia
Whether to enable the early media for E1R2 trunk group is configured.
Step 10 Run:
r2-play-ringback r2-play-ringback
Whether to enable the ringback tone (RBT) for E1R2 trunk group is configured.
----End
2.12.3 Binding Trunks in a Trunk Group

Binding Trunks to a SIP AT0 Trunk Group
Context
When the SIP trunk registration mode is trunk circuit registration, perform the following
operations.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name
The SIP AT0 trunk group view is displayed.

Step 4 Run:
trunk-sipat0 name
The SIP AT0 trunk is added to a trunk group.

----End
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Binding Trunks to an AT0 Trunk Group

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name
The AT0 trunk group view is displayed.

Step 4 Run:
trunk-at0 slotid/subcardid/portid [ config-status config-status-value | cid-transtype cid-trans-type-value | dial-delay dial-delay-value | dial-mode dial-modevalue | insert-prefix insert-prefix-value | insert-prefix-delay insert-prefixdelay-value | reversepole-detect reversepole-detect-value | default-called-telno
default-called-telno-value] *
Trunks are bound to an AT0 trunk group.

----End
Binding Trunks to a PRA Trunk Group

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name
The PRA trunk group view is displayed.

Step 4 Run:
trunk-pra slotid/subcardid/portid
Trunks are bound to the PRA trunk group.

----End
Binding Trunks to an E1R2 Trunk Group

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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
trunk-group name
The E1R2 trunk group view is displayed.

Step 4 Run:
trunk-e1r2 slotid/subcardid/portid
Trunks are bound to the E1R2 trunk group.

NOTE
After the E1R2 trunk group is complete, to ensure that call services passing through the E1R2 trunk group
are transmitted correctly, local loopback cannot be configured on the VE1 interface. If local loopback has
been configured on the VE1 interface, cancel the local loopback configuration immediately. One minute
after the local loopback configuration is canceled, call services passing through the E1R2 trunk group are
restored.
----End
2.12.4 Resetting a SIP Trunk Group

To make SIP trunk group parameters take effect, reset the SIP trunk group.
To make modified SIP trunk group parameters take effect or restart a new SIP trunk group, reset
the SIP trunk group.
Before resetting a SIP trunk group, complete the following task:
l
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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Step 3 Run:
trunk-group name

Step 4 Run:
reset
The SIP trunk group is reset.
CAUTION
Exercise caution when you run this command because resetting a SIP trunk group affects running
services.
----End

Prerequisites
All configurations of a trunk group are complete.
Procedure
l
Run the display voice trunk-group command to check the configuration of a trunk group.
----End
2.13 Configuring a Call Route

A call route is associated with a trunk group to enable users from different offices to call each
other.
As shown in Figure 2-11, the headquarters and branch of enterprise A are connected through a
SIP trunk. The headquarters is connected to PSTN networks of carrier A and carrier B. To reduce
call fees, configure call routes on the AR2200, which meet the following requirements:
l
Calls between users in the headquarters and branch are connected through the SIP trunk.
When users in the enterprise call external users, the calls are connected to the called users
through the network of carrier A or carrier B, depending on the charge rates used by the
carriers in different time ranges.
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Figure 2-11 Call route application
Branch
User B
PSTN
Carrier A
AR
Enterprise A
SIP
PSTN
Carrier B
IP
SIP
PRA
AT0
AR
Headquarters
PBX
Enterprise A
User A
Before configuring a call route, complete the following tasks:
l
Configuring a Call Prefix
Configuring a trunk group
Data Preparation
To configure a call route, you need the following data.
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No.
Data
Name of the call route
Enterprise, Centrex group, and DN set to which the calling number is bound, and
calling condition
Prefix of the called numbers to which the call route applies
Validity period of the call route
Trunk group associated with the call route

143

No.
Data
Trunk circuit selection mode
2 PBX Configuration
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
A call prefix profile is created and its view is displayed.

By default, no call prefix profile is configured.
Step 4 Run:
callroute condition { caller-telno caller-telno-value | time-period { from fromdate [ from-time ] | to to-date [ to-time ] } * | time-repeat { { yearly |
monthly }begin-date [ begin-time ] [ end-date [ end-time ] ] | weekly beginweekday [ begin-time ] [ end-weekday [ end-time ] ] | daily begin-time [ endtime ] } }* trunkgroup1 trunkgroup1-name [ trunkgroup2 trunkgroup2-name |
trunkgroup3 trunkgroup3-name | trunkgroup4 trunkgroup4-name ]*
A call route is created.

----End

Run the display voice callroute command to check the configuration of a call route.
2.14 (Optional)Configuring Individualized PBX Services

The AR2200 supports multiple individualized PBX services.
2.14.1 Configuring the Abbreviated Dialing Service

The abbreviated dialing service allows a user to initiate a call by dialing the 2-digit abbreviated
code instead of the original called number.
After the abbreviated dialing service is configured on the AR2200, users can dial the 2-digit
abbreviated code instead of the original called number, which is convenient for users to operate
and manage the phone numbers.
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Before configuring the abbreviated dialing service, complete the following tasks:
l
Configuring a PBX user
Data Preparation
To configure the abbreviated dialing service, you need the following data.
No.
Data
Abbreviated code
PBX user name
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name
The PBX user identifier view is displayed.

Step 4 Run:
service-right abbdial enable
The abbreviated dialing service is enabled.

Step 5 Run:
service abbdial abb-telno called-telno
An abbreviated dialing service scheme is created.

By default, no abbreviated dialing service scheme is configured.
----End

Run the display voice [ pbxuser name ] service abbdial command to check the configuration
of the abbreviated dialing service.
2.14.2 Configuring the Call-out Restriction Service

You can restrict some outgoing calls of a phone, for example, toll calls.
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To restrict outgoing calls such as international toll calls, configure the call-out restriction service.
Before configuring the call-out restriction service, complete the following tasks:
l
Data Preparation
To configure the call-out restriction service, you need the following data.
No.
Data
User name
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name
The PBX user view is displayed.

Step 4 Run:
service-right cba enable
The outgoing call barring service is enabled.

Step 5 Run:
service cba { all | idd_cdd | idd }
A call-out restriction service scheme is created.

By default, no call-out restriction service scheme is configured.
----End

Run the display voice [ pbxuser name ] service cba command to check the configuration of
the call-out restriction service.
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2.14.3 Configuring the Call Forwarding Service

If the call forwarding service of a called party is activated and the call forwarding conditions
are met, calls are forwarded to a preset number.
Usage Scenario
To ensure that users can answer incoming calls, configure the following types of call forwarding
service:
l
Call Forwarding Busy (CFB): Incoming calls of a CFB user are forwarded to a preset
number when the user line is busy.
Call Forwarding No Reply (CFNR): Incoming calls of a CFNR user are forwarded to a
preset number when the user does not answer the call.
Call Forwarding Offline (CFO): Incoming calls of a CFO user are forwarded to a preset
number when the user is offline.
Call Forwarding Unconditional (CFU): All incoming calls of a CFU user are forwarded to
a preset number.
Before configuring the call forwarding service, complete the following tasks:
l
Data Preparation
To configure the call forwarding service, you need the following data.
No.
Data
PBX user name
Forwarded-to number
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

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l To configure the CFB service, run:
service-right cfb enable
The CFB service is enabled.
service cfb fwd-telno
The CFB service is configured.

l To configure the CFNR service, run:
service-right cfnr enable
The CFNR service is enabled.
service cfnr { telno fwd-telno | time telno fwd-telno }
The CFNR service is configured.

l To configure the CFO service, run:
service-right cfo enable
The CFO service is enabled.
service cfo fwd-telno
The CFO service is configured.

l To configure the CFU service, run:
service-right cfu enable
The CFU service is enabled.
service cfu name
The CFU service is configured.

----End

Run the following commands to check the call forwarding service:
l
Run the display voice [ pbxuser name ] service cfb command to check the CFB service.
Run the display voice [ pbxuser name ] service cfb command to check the CFNR service.
Run the display voice [ pbxuser name ] service cfo command to check the CFO service.
Run the display voice [ pbxuser name ] service cfu command to check the CFU service.
2.14.4 Configuring the Number Barring Service

To prevent a user from dialing a number, configure the number barring service.
Before configuring the number barring service, complete the following tasks:
l
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Data Preparation
To configure the number barring service, you need the following data.
No.
Data
PBX user name
Restricted number
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right dlc enable
The number barring service is enabled.

Step 5 Run:
service dlc dlc-telno
The number to be rejected is configured.

By default, no number is rejected.
----End

Run the display voice [ pbxuser name ] service dlc command to check the configuration of the
number barring service.
2.14.5 Configuring the DND Service

When a user calls a DND user, the user will hear the DND tone or busy tone.
The DND service allows a user to reject all incoming calls during a certain period.
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Before configuring the DND service, complete the following tasks:
l
Data Preparation
To configure the DND function, you need the following data.
No.
Data
User identifier name
DND tone
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right dnd enable
The do-not-disturb service is enabled.

Step 5 Run:
service dnd [ tone-id tone-id ]
A DND service is created .

By default, no DND service scheme is configured.
----End

Run the display voice [ pbxuser name ] service dnd command to check the configuration of
the DND service.
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2.14.6 Configuring the RAC Service

The Reject Anonymous Call (RAC) service allows a user to reject an anonymous call (not
automatic number identification) and allows the system to play an announcement to the calling
number after call rejection.
To prevent anonymous calls, configure the RAC service.
Before configuring the RAC service, complete the following tasks:
l
Data Preparation
To configure the RAC service, you need the following data.
No.
Data
PBX user name
RAC tone
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right rac enable
The reject anonymous call service is enabled.

Step 5 Run:
service rac [ tone-id tone-id ]
An RAC service is configured.

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By default, no RAC service scheme is configured.

----End

Run the display voice [ pbxuser name ] service rac command to check the configuration of the
RAC service.
2.14.7 Configuring the Remote Office Service

The remote office service allows a user to access from any terminal and share original services
such as short number dialing and call transfer.
The remote office service allows a user to access from any terminal and share original services
such as short number dialing and call transfer. Because the call initiator still dials the original
number in the remote office service, the user privacy is well protected.
Before configuring the remote office service, complete the following tasks:
l
Data Preparation
To configure the remote office service, you need the following data.
No.
Data
PBX user name
Destination number of the remote office service
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

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Step 4 Run:
service-right remote-office enable
The reject anonymous call service is enabled.

Step 5 Run:
service remote-office telno { centrex centrex-name short-num | [ country-code
countrycode ] [ area-code areacode ] telno }
The destination number of the remote office service is configured.

By default, the destination number of the remote office service is not configured.
----End

Run the display voice [ pbxuser name ] service remote-office command to check the
configuration of the remote office service.
2.14.8 Configuring the Secretary Service

The secretary service allows a user to designate another phone number (for example, the
secretary's phone number) to process all incoming calls. That is, all incoming calls of the user
are transferred to the secretary's phone first, and only the secretary can call the user without
transferring.
You can configure the secretary service to filter incoming calls and prevent interruption.
Before configuring the secretary service, complete the following tasks:
l
Data Preparation
To configure the secretary service, you need the following data.
No.
Data
PBX User name
Secretary phone number
Procedure
Step 1 Run:
system-view
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Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right scr enable
The secretary service is enabled.

Step 5 Run:
service scr telno { centrex centrex-name short-num | countrycode areacode telno }
The destination number of the secretary is configured.

By default, the destination number of the secretary is configured.
----End

Run the display voice service scr command to check the configuration of the secretary service.
2.14.9 Configuring the Wake-up Service

The wake-up service allows the system to send the wake-up tone to the user at the preset time.
When the wake-up time is due, the system sends the wake-up tone to the user.
Before configuring the wake-up service, complete the following tasks:
l
Data Preparation
To configure the wake-up service, you need the following data.
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Data
User name
Wake-up time

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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right wake-call enable
The wake-up service is enabled.

Step 5 Run:
service wake-call wake-time wake-time-value [ begin-date begin-date-value | wakedays wake-days-value | tone-id tone-id-value ] *
A wake-up service scheme is created.

By default, no wake-up service scheme is configured.
----End

Run the display voice [ pbxuser username ] service wake-call command to check the
configuration of the wake-up service.
2.14.10 Configuring the RBT Service

The Ring Back Tone (RBT) service allows the calling party to hear the music or sound effect
customized by a called party but not the traditional beeper.
Usage Scenario
After a user registers the RBT service, the user can set different RBTs for a calling party or a
group of calling parties in different periods.
Before configuring the RBT service, complete the following tasks:
l
Data Preparation
To configure the RBT service, you need the following data.
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No.
Data
PBX user name
RBT file name
Number or trunk group that receives RBTs
Period in which an RBT is played
2 PBX Configuration
Context
If the RBT played for the caller through the FXO port is too loud, the FXO port may fail to detect
the busy tone from the caller; therefore, if the caller hangs up before the call is answered, the
called user still hears the ring tone for a period. To solve this problem, run the gain send send
command to adjust the send gain on an FXO interface.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right crbt enable
The RBT service is enabled.

Step 5 Run:
service crbt [ condition { caller-trunkgroup trunkgroup-name | caller-telno callertelno-value | time-period { from from-date [ from-time ] | to to-date [ to-time ] }
* | time-repeat { { yearly | monthly }begin-date [ begin-time ] [ end-date [ endtime ] ] | weekly begin-weekday [ begin-time ] [ end-weekday [ end-time ] ] |
daily begin-time [ end-time ] }* }*] crbt-file file-name-value
The RBT service is configured.

By default, no RBT service is configured.
----End

Run the display voice service-crbt command to check the configuration of the RBT service.
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2.14.11 Configuring the SCR Service

The Selective Call Rejection (SCR) service filters every incoming call against a list of rejected
numbers created by a user and rejects those calls on the list.
Usage Scenario
If dialing some numbers is prevented, configure the SCR service.
Before configuring the SCR service, complete the following tasks:
l
Data Preparation
To configure the SCR service, you need the following data.
No.
Data
PBX User name
Rejected number
Period in which a call is rejected
Period in which a call is rejected repeatedly
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxuser name

Step 4 Run:
service-right rcs enable
The selective call rejection service is enabled.

Step 5 Run:
service rcs judgegrp judgegrp-value [ condition { caller-telno caller-telno-value
| time-period { from from-date [ from-time ] | to to-date [ to-time ] } * | time-
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repeat { { yearly | monthly }begin-date [ begin-time ] [ end-date [ end-time ] ] |

weekly begin-weekday [ begin-time ] [ end-weekday [ end-time ] ] | daily begintime [ end-time ] }* }*] [ active active-value ]
The SCR service is configured.

By default, no SCR service is configured.
----End

Run the display voice [ pbxuser name ] service rcs command to check the configuration of the
SCR service scheme.
2.14.12 Configuring the SCA Service

The Selective Call Acceptance (SCA) service filters every incoming call against a list of numbers
specified by a user and accepts calls from only those numbers on the list.
Usage Scenario
To receive only some incoming calls, configure the SCA service.
Before configuring the SCA service, complete the following tasks:
l
Data Preparation
To configure the SCA service, you need the following data.
No.
Data
PBX user name
Accepted phone number
Period in which a call is accepted
Period in which a call is accepted repeatedly
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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Step 3 Run:
pbxuser name

Step 4 Run:
service-right sca enable
The selective call acceptance service is enabled.

Step 5 Run:
service sca judgegrp judgegrp-value [ condition { caller-telno caller-telno-value
| time-period { from from-date [ from-time ] | to to-date [ to-time ] } * | timerepeat { { yearly | monthly }begin-date [ begin-time ] [ end-date [ end-time ] ] |
weekly begin-weekday [ begin-time ] [ end-weekday [ end-time ] ] | daily begintime [ end-time ] }* }*] [ active active-value ]
The SCA service is configured.

By default, no SCA service is configured.
----End

Run the display voice [ pbxuser name ]service sca command to check the configuration of the
SCA service.
2.15 (Optional) Configuring Group Services on the PBX

When functioning as a PBX, the AR2200 supports various individual services. You can
configure services for individual users as required.
2.15.1 Configuring the Call Interception Service

The call interception service provides a voice prompt when a call fails to be connected. The user
can learn about the failure cause and determine whether to redial the called number according
to the voice prompt.
Usage Scenario
A call may fail to be connected for some reasons, for example, the called user is busy, the called
user does not answer the call, the called user is offline, the called number does not exist, the
calling user does not have right to make this call, or the called user is unreachable. If the call
interception service is not configured, the calling user only hears the busy tone but does not
know the reason. If the user redials the called number multiple times but the call still fails to be
connected, the user experience is degraded and network resources are wasted. The call
interception service allows the system to play a user-friendly voice prompt when a call failed to
be connected. The called user knows the cause of the call failure and determines whether to
redial the called number according to the voice prompt. This improves the user experience,
reduces invalid dial attempts, and saves resources on the AR2200.
Before configuring the call interception service, complete the following tasks:
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Data Preparation
To configure the call interception service, you need the following data.
No.
Data
Call interception plan name, enterprise, DN set, and call prefix to be bound to the
call interception plan, and call status.
(Optional) ID of the prompt tone used for call interception
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
service call-intercept call-intercept-name
A call interception service plan is created and the call interception service view displayed.
By default, no call interception service plan is configured.
NOTE
Before creating a call interception plan, run the service-right call-intercept enable command in the PBX
user view to enable the call interception service right.
Step 4 Set mandatory parameters for the call interception service plan.
l Run:
enterprise enterprise-name [ dn-set dn-set-name | centrex centrex-name ]
An enterprise, a DN set, and a Centrex group are bound to the call interception service plan.
l Run:
A call prefix is bound to the call interception service plan.

l Run:
call-status { busy | no_answer | offline | dnerror | no_right | unreachable }
The call status is configured.

tone-id tone-id-value
A prompt tone is specified for call interception.

----End
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Run the display voice service call-intercept [ call-intercept-name ] command to check the call
interception service configuration.
2.15.2 Configuring the Distinctive Ringing Service

The distinctive ringing service plays different ring tones to users based on call prefixes (local
calls, national toll calls, international toll calls, and intra-group calls).
User A and user B belong to the same Centrex group, and user C is out of the Centrex group. If
the distinctive ringing service is not configured, the same ring tone is played to user B no matter
whether user A or user C calls user B. User B cannot know whether the call is from a Centrex
user or an external user according to the ring tone. After the distinctive ringing service is
configured, users in the Centrex group can identify calls from users in the same Centrex group
and outside the Centrex group.
Before configuring the distinctive ringing service, complete the following tasks:
l
Creating an enterprise
Creating a Centrex group
Data Preparation
To the distinctive ringing service, you need the following data.
No.
Data
Distinctive ringing service plan name, and enterprise and Centrex group bound to the
plan
(Optional) Ring IDs for intra-group calls, local calls, national toll calls, and
international toll calls
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
service alert-ring alert-ring-name
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A distinctive ringing service plan is created and the distinctive ringing service view is displayed.
By default, no distinctive ringing service plan is configured.
Step 4 Set mandatory parameters for the distinctive ringing service plan.
l Run:
enterprise enterprise-name [ centrex centrex-name ]
An enterprise is bound to the distinctive ringing service plan.

Step 5 Set optional parameters for the distinctive ringing service plan.
l Run:
centrex-ring-id centrex-ring-id-value
The ring ID for intra-group calls is specified.

l Run:
out-local-ring-id out-local-ring-id-value
The ring ID for local calls is specified.

l Run:
out-ddd-ring-id out-ddd-ring-id-value
The ring ID for national toll calls is specified.

l Run:
out-idd-ring-id out-idd-ring-id-value
The ring ID for international toll calls is specified.

----End

Run the display voice service alert-ring [ alert-ring-name ] command to check the
configuration of the distinctive ringing service plan.
2.15.3 Configuring the Enterprise RBT Service

The enterprise RBT service allows an enterprise to play a customized audio file to calling users.
Usage Scenario
If the RBT service is not configured for an enterprise, a user only hears the common ring back
tone when calling a user in the enterprise. With the enterprise RBT service, the enterprise can
play an advertisement or enterprise information to calling users. The enterprise can also
configure different RBTs for customers. The enterprise RBT service improves user experience
and helps the enterprise raise brand image.
Before configuring the enterprise RBT service, complete the following tasks:
l
Creating a trunk group
Data Preparation
To configure the enterprise RBT service, you need the following data.
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No.
Data
Name of the RBT service plan and enterprise bound to the plan
(Optional) Trunk group that can receive the enterprise RBT
(Optional) Calling numbers that can receive the enterprise RBT
(Optional) Duration in which the enterprise RBT can be played
2 PBX Configuration
Context
If the RBT played for the caller through the FXO port is too loud, the FXO port may fail to detect
the busy tone from the caller; therefore, if the caller hangs up before the call is answered, the
called user still hears the ring tone for a period. This problem can be solved by adjusting the
send gain on the FXO interface.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
The enterprise view is displayed.

Step 4 Run:
service crbt [ condition { caller-telno caller-telno-value | caller-trunkgroup
trunkgroup-name | time-period { from from-date [ from-time ] | to to-date [ totime ] } * | time-repeat { yearly begin-date [ begin-time ] [ end-date [ endtime] ] | monthly begin-date [ begin-time ] [ end-date [ end-time] ] | weekly beginweekday [ begin-time ] [ end-weekday [ end-time ] ] | daily begin-time [ endtime] } } ] file-name file-name-value
The enterprise RBT service is configured.

By default, no enterprise RBT service is configured.
NOTE
Before configuring the enterprise RBT service, run the service-right crbt enable command in the PBX user
view to enable the enterprise RBT service.
----End

Run the display voice service crbt [ name ] command to check the configuration of the RBT
service.
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2.15.4 Configure the IVR Service

The Interactive Voice Response (IVR) service allows enterprises to customize their IVR menu
and prompt tone, improving user experiences.

Before configuring an IVR service, familiarize yourself with the applicable environment,
The interactive voice response (IVR) service provides the automated attendant function and
allows enterprises to make their own IVR menus and voice prompts.
Before configuring an IVR service, complete the following task:
NOTE
If the codec mode is set to G.723, access terminals cannot use the IVR service.
Data Preparation
To configure the IVR service, you need the following data.
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No.
Data
Name, enterprise, description, and status of the IVR file
IVR menu name, type, enterprise, prompt tone, and optional parameters, including
the menu tone, prompt tone played when there is no input on the IVR menu within
the specified period, prompt tone played when the input information does not match
the preconfigured information of the IVR menu, prompt tone played when the number
of incorrect inputs on the IVR menu reaches the maximum value, tone interruption
flag, whether to replay the prompt tone when there is no input on the IVR menu within
the specified period, whether to replay the prompt tone when the input information
does not match the preconfigured information of the IVR menu, wait duration, and
maximum number of times the input information does not match the preconfigured
information of the IVR menu
IVR action menu name, current action status of the menu, operation code, IVR action
code, and optional parameters, including the prompt tone, sub-menu, forward-to
number, DN set of the forward-to number, Centrex group, and enterprise of the
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No.
Data
Enterprise, DN set, access number, validity period, validity cycle mode, calling
number, switchboard number, and optional parameters, including the prompt tone,
destination enterprise name used when the IVR group connects a call to an extension
phone number, maximum queue duration and maximum number of calls in a queue,
(optional) ringing duration and ringing mode, flag indicating whether the IVR group
can directly connect to the switchboard, and menu name of the IVR group
Enterprise, DN set, and optional parameters, including the maximum queue duration
and maximum number of calls in a queue and ringing duration and ringing mode of
the NAVI group
Configuring an IVR File

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
ivr-file file-name-value
The IVR file name is configured.

Step 4 Run:
An enterprise is bound to the IVR file.

Step 5 Set optional parameters for the IVR file.
l Run:
description desc-value
The IVR file description is configured.

l Run:
status status-value
The IVR file status is specified.

----End
Configuring an IVR Menu
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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
ivr-menu ivr-menu-name [ root-menu | sub-menu ]
An IVR menu is created and the IVR menu view is displayed.

Step 4 Run:
An enterprise is bound to the IVR menu.

Step 5 Run:
prompt-tone { toneid
| ivr-file name }
The prompt tone is configured for the IVR menu.

Step 6 Run:
father-menu father-menu-name
The parent menu of the IVR is configured.

NOTE
When the IVR menu is a root menu, this step can be skipped.
Step 7 Set optional parameters for the IVR menu.

l Run:
menu-tone { toneid | ivr-file name }
The IVR menu tone is configured.

l Run:
no-input-tone { toneid | ivr-file name }
The prompt tone played when there is no input on the IVR menu within the specified period
is configured.
l Run:
no-match-tone { toneid | ivr-file name }
The prompt tone played when the input information does not match the preconfigured
information of the IVR menu.
l Run:
max-error-tone { toneid | ivr-file name }
The prompt tone played when the number of incorrect inputs on the IVR menu reaches the
maximum value is configured.
l Run:
menu-bargein menubargein-value
The tone interruption flag is configured.

l Run:
no-input-reprompt { enable | disable }
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The device determines whether to replay the prompt tone when there is no input on the IVR
menu within the specified period.
l Run:
no-match-reprompt { enable | disable }
The device determines whether to replay the prompt tone when the input information does
not match the preconfigured information of the IVR menu.
l Run:
menu-wait-timer value
The wait duration is set for the IVR menu.

l Run:
max-error-num value
The maximum number of times the input information does not match the preconfigured
information of the IVR menu is set.
----End
Configuring an IVR Action

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
ivr-action name
An IVR action is created.

Step 4 Run:
ivr-menu name
The menu name of the IVR action is configured.

Step 5 Run:
current-state current-menu-state-value
The current action status of the menu is configured.

Step 6 Run:
menu-actcode { call-navi-group | call-transfer transfer-telephone-number | go-submenu sub-menu-name | play-prompt-tone { tone-id | ivr-file file-name } | releasecall | replay-menu-tone | return-father-menu | return-root-menu }
An action code is set.

Step 7 Run:
operation-code menu-opercode-value
An IVR operation code is set.

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Step 8 Set optional parameters for the IVR action.

l Run:
next-state out-menu-state-value
The IVR menu status displayed after the IVR action is taken is configured.
l Run:
destination enterprise destination-enterprise-name [ dn-set dn-set-name |
centrex centrex-name ]*
The destination enterprise name, destination DN set, and destination Centrex group are
configured for the forward-to number of the IVR action.
----End
Configuring an IVR Group

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name ivr [ enterprise enterprise-name ]
An IVR group is created and the IVR group view is displayed.

Step 4 Run:
dn-set dn-set-value
A DN set is bound to the IVR group.

Step 5 Run:
access-telno { centrex centrex-name centrex-telno-value | [ country-code countrycode-value ] [ area-code area-code-value ] telno-value }
An access number is set for the IVR group.

Step 6 Run:
condition { caller-telno caller-telno-value | time-period { from from-date [ fromtime ] | to to-date [ to-time ] } * | time-repeat { { yearly | monthly }begin-date
[ begin-time ] [ end-date [ end-time ] ] | weekly begin-weekday [ begin-time ]
[ end-weekday [ end-time ] ] | daily begin-time [ end-time ] }* }*
The calling party, validity period, and time repeat mode are configured for the IVR group.
Step 7 Run:
console-telno value
A switchboard number is set for the IVR group.

Step 8 Set optional parameters for the IVR group.
l Run:
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tone-id { tone-id-value | file name }
A tone is configured for the IVR group.

l Run:
destination { dn-set dn-set-name | centrex centrex-name | enterprise enterprisename}*
The destination enterprise name used when the IVR group connects a call to an extension
phone number is configured.
l Run:
queue enable [ maximum-queue
value ]
maximum-queue-value | queue-time queue-time-
The maximum queue duration and maximum number of calls in a queue are set.
l Run:
ring { mode ring-mode-value | time ring-time-value | select select-mode-value }
The ringing duration and ringing mode are set.

l Run:
direct-trans-console { disable
| enable }
The flag indicating whether the IVR group can connect to the switchboard is set.
l Run:
navigate-menu menu-name
A menu name is configured for the IVR group.

----End
Configuring an IVR Navigation Group

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name navi [ enterprise enterprise-name ]
The navigation group is configured.

Step 4 Run:
dn-set dn-set-value
A DN set is bound to the IVR navigation group.

Step 5 Run:
menu menu-name
A menu name is configured for the IVR navigation group.

Step 6 Run:
menu-opercode value
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An operation code input when the IVR navigation group takes the menu action is configured.
Step 7 Set optional parameters for the IVR navigation group.
l Run:
queue enable [ maximum-queue
value ]
maximum-queue-value | queue-time queue-time-
The maximum queue duration and maximum number of calls in a queue are set.
l Run:
ring { mode ring-mode-value | time ring-time-value
value }
| select
select-mode-
The ringing duration and ringing mode are set.

----End

After the configurations of IVR navigation service are complete, you can verify the
configurations.
Procedure
l
Run the display voice ivr-action [ ivr-action-name ] command to view information about
one IVR action or all IVR actions.
Run the display voice ivr-file [ ivr-file-name ] command to view information about one
IVR file or all IVR files.
Run the display voice ivr-menu [ ivr-menu-name ] command to view information about
one IVR menu or all IVR menus.
Run the display voice pbxusergroup [ pbxusergroup-name ] command to check the

configuration of the PBX user group.
----End
2.15.5 Configuring the Number Change Service

You can configure rules to change calling numbers or DN sets.
Usage Scenario
To hide calling numbers or display the same calling number for all outgoing calls, configure the
number change service.
Before configuring the number change service, complete the following tasks:
l
Configuring PBX users
Data Preparation
To configure the number change service, you need the following data.
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No.
Data
Name of the number change plan
Enterprise and DN set bound to the number change plan, calling number that needs
to be changed, and number change rule
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
caller-change [ dn-set dn-set-name ] [ enterprise enterprise-name ] caller
[ centrex centrex-name centrex-telno-value ] [ country-code country-code-value ]
[ area-code area-code-value ] telno-value operation-type { new-enterprise newenterprise-name | new-dn-set new-dn-set-name | new-centrex new-centrex-name | [ newcountry-code new-country-code-value ] [ new-area-code new-area-code-value ] } *
A number change service plan is configured.

By default, no number change service plan is configured.
----End

Run the display voice caller-change [ caller [ centrex centrex-name centrex-telno-value ]
[ country-code country-code-value] [ area-code area-code-value ] telno-value ] command to
check the configuration of the number change service.
2.15.6 Configuring the Pre-routing Number Change Service

Usage Scenario
You can configure pre-routing number change plans to define various dialing modes and change
the calling number displayed on the called party's phone. For example, a POTS user (using the
number 28761000) connected to an IP PBX makes a local call by dialing 0755 28961000. The
configured call route connects local outgoing calls with call prefix 2896 through an AT0 trunk.
Therefore, a pre-routing number change plan needs to be configured to remove 0755 from the
called number.
Before configuring the pre-routing number change service, complete the following tasks:
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Data Preparation
To configure the pre-routing number change service, you need the following data.
No.
Data
Name of the pre-routing number change plan
Enterprise, DN set, Centrex group, and call prefix bound to the pre-routing number
change plan
Calling numbers that need to be changed before routing, calling number change rule,
and called number change rule
(Optional) New enterprise name and Centrex group name used after number change
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
beforeroute-change name
A pre-routing number change plan is created and its view is displayed.

By default, no pre-routing number change service plan is configured.
Step 4 Run:
A call prefix is bound to the pre-routing number change plan.

Step 5 Run:
condition caller-telno { [ country-code country-code-value ] [ area-code area-codevalue ] telno-value | centrex centrex-telno-value }
A calling number that needs to be changed before routing is specified.

Step 6 Configure a pre-routing number change rule.
NOTE
A calling number change rule and a called number change rule must be configured simultaneously. If the
calling number or called number keeps unchanged, set the number rule mode to no-change.
l Run:
caller { del-then-insert del-offset del-len insert-telnum | del del-offsetval
del-lenval | insert insert-offset insert-telnum-val | no-change }
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A calling number change rule is configured.

l Run:
called { del-then-insert del-offset del-len insert-telnum | del del-offsetval
A called number change rule is configured.

Step 7 (Optional) Perform the following steps as required.
l Run:
call new-attribute { enterprise enterprise-name | dn-set dn-set-name | centrex
centrex name } *
The pre-routing number change service is configured.

l Run:
call re-analyse reanalyze-value
The AR2200 is configured to or not to reanalyze the changed numbers.

l Run:
call dual-dial-tone-flag dual-dial-tone-flag-value
The AR2200 is configured to or not to play second dial tone.

----End

Run the display voice beforeroute-change [ name ] command to check the configuration of the
pre-routing number change service.
2.15.7 Configuring the Post-routing Number Change Service

Usage Scenario
You can configure post-routing number change plans to define various dialing modes and change
the calling number displayed on the called party's phone. A post-routing number change plan
changes a called number to a long number. A post-routing number change plan can change a
called number to a long number ensure that it complies with the required number format. For
example, a POTS user (using the number 2876100) connected to an IP PBX makes a national
toll call by dialing 075528560982. A post-routing number change plan adds 12523 to the called
number 07552856098. 12523 is the call prefix defined by the carrier for the enterprise. When
the carrier's device detects the call prefix 12523, it connects the outgoing call through the
matching trunk. This reduces the call fees of the enterprise.
Before configuring the post-routing number change service, complete the following tasks:
l
Data Preparation
To configure the post-routing number change service, you need the following data.
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No.
Data
Name of the post-routing number change plan
Enterprise and DN set bound to the post-routing number change plan
Calling numbers that need to be changed after routing
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
afterroute-change name
A post-routing number change plan is created and its view is displayed.

By default, no post-routing number change service plan is configured.
Step 4 Run:
A call prefix is bound to the post-routing number change plan.

Step 5 Run:
condition caller-telno { [ country-code country-code-value ] [ area-code area-codevalue ] telno-value | centrex centrex-telno-value }
A calling number that needs to be changed after routing is specified.

Step 6 Run:
trunk-group trunk-group-name
A trunk group is bound to the post-routing number change plan.

Step 7 Configure a post-routing number change rule.
NOTE
A calling number change rule and a called number change rule must be configured simultaneously. If the
calling number or called number keeps unchanged, set the number rule mode to no-change.
l Run:
caller { del-then-insert del-offset del-len insert-telnum | del del-offsetval
A calling number change rule is configured.

l Run:
called { del-then-insert del-offset del-len insert-telnum | del del-offsetval
A called number change rule is configured.

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call new-attribute { enterprise enterprise-name | centrex centrex-name }
The pre-routing number change service is configured.

----End

Run the display voice afterroute-change [ name ] command to check the configuration of the
post-routing number change service.
2.15.8 Configuring the PBX Line Selection Service

After the PBX line selection service is configured on the AR2200, a calling party can dial the
access number of a PBX group, and the AR2200 uses the configured line selection mode to
connect the call to a user in the group.
Usage Scenario
To configure the PBX line selection service, create a PBX group and allocate an access number
to all users in the PBX group. When a user on the public network dials the access number, the
call processing program selects an idle line to connect the call by using the configured line
selection mode.
Before configuring the PBX line selection service, complete the following tasks:
l
Creating a trunk group
Data Preparation
To configure the PBX line selection service, you need to the following data.
No.
Data
Name of a PBX group
Enterprise bound to the PBX group
access number of the PBX group
DN set bound to the PBX group
(Optional) Ringing duration, ringing mode, and line selection mode
Procedure
Step 1 Run:
system-view
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Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name hunt [ enterprise enterprise-name ]
The PBX line selection service is configured.

By default, no PBX line selection service is configured.
Step 4 Run:
dn-set dn-set-value
A DN set is bound to the PBX group.

Step 5 Run:
The access number of the PBX group is configured.

NOTE
The access number must be a local user number.

The ringing duration, ringing mode, and line selection mode are configured.
----End

2.15.9 Configuring the Co-Group Pickup Service

The co-group pickup service allows users in the same group to answer calls for each other on
their own phones. For example, when the phone of user A rings, user B in the same group can
dial the service access code plus user A's phone number to answer the call.
Usage Scenario
Before the co-group pickup service is configured, some calls will be missing if the called parties
do not answer the calls in a timely manner. After the co-group pickup service is configured,
users in the same group can answer calls for each other on their own phones. For example, if
user A does not answer a call, user B in the same group can dial the service access code plus
user A's phone number to answer the call. This service reduces missing calls.
Before configuring the co-group pickup service, complete the following tasks:
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Data Preparation
To configure the co-group pickup service, you need the following data.
No.
Data
Name of the pickup group
Enterprise bound to the pickup group
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name pickup [ enterprise enterprise-name ]
The pickup service is configured.

By default, no pickup service is configured.
NOTE
Before configuring the pickup service, run the service-right pickup-in-group enable command in the PBX
user view to enable the pickup service.
----End

2.15.10 Configuring Simultaneous Ringing

When a calling party dials the access number of a simultaneous ringing group, all member phones
in the group ring simultaneously, and the called party can answer the call using any ringing
phone.
Usage Scenario
Before simultaneous ringing is configured, some calls will be missing if the called parties do
not answer the calls in a timely manner. After simultaneous ringing is configured, when user B
calls user A using the access number of a simultaneous ringing group, all the idle member phones
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in the group ring simultaneously, and user A can answer the call using any ringing phone.
Simultaneous ringing reduces missing calls and improves the call connection ratio without
requiring additional devices.
Before configuring simultaneous ringing, complete the following tasks:
l
Data Preparation
To configure simultaneous ringing, you need the following data.
No.
Data
Simultaneous ringing group name
Enterprise and DN set bound to the simultaneous ringing group
Access number of the simultaneous ringing group
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name ring-parallel [ enterprise enterprise-name ]
The simultaneous ringing group is configured.

By default, no simultaneous ringing group is configured.
Step 4 Run:
dn-set dn-set-value
A DN set is bound to the simultaneous ringing group.

Step 5 Run:
An access number is set for the simultaneous ringing group.

----End
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2.15.11 Configuring Sequential Ringing

When a calling party dials the access number of a sequential ringing group, member phones in
the group ring in the configured sequence.
Usage Scenario
Before sequential ringing is configured, some calls will be missing if the called parties do not
answer the calls in a timely manner. After sequential ringing is configured, when user B calls
user A using the access number of a sequential ringing group, member phones in the group ring
in the configured sequence. Sequential ringing reduces missing calls and improves the call
connection ratio without requiring additional devices.
Before configuring sequential ringing, complete the following tasks:
l
Data Preparation
To configure sequential ringing, you need the following data.
No.
Data
Sequential ringing group name
Enterprise and DN set bound to the sequential ringing group
Access number of the sequential ringing group
(Optional) Ringing duration and line selection mode of the sequential ringing group
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name hunt [ enterprise enterprise-name ]
The sequential ringing group is configured.

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By default, no sequential ringing group is configured.

Step 4 Run:
dn-set dn-set-value
A DN set is bound to the sequential ringing group.

Step 5 Run:
An access number is set for the sequential ringing group.

ring { time ring-time-value | select select-mode-value }
The ringing duration and line selection mode are set for the sequential ringing group.
----End

2.15.12 Configuring the One Number Link You Service

When the calling party calls the one number link you (ONLY) number of the called party,
multiple terminals of the called party ring according to the configured rules, and the called party
can select one terminal to answer the incoming call.
Usage Scenario
Before the ONLY service is configured, some calls will be missing if the called parties do not
answer the calls in a timely manner. After the ONLY service is configured, a user is bound to
multiple terminals. When user B calls user A using the access number of the ONLY service,
multiple terminals of user A ring according to the configured rules, and user A can select one
terminal to answer the call. Therefore, the ONLY service reduces missing calls and improves
the call connection ratio.
Before configuring the ONLY service, complete the following tasks:
l
Data Preparation
To configure the ONLY service, you need the following data.
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Data
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No.
Data
Enterprise and DN set bound to the ONLY service and access number of the ONLY
service
(Optional) Ringing duration, ringing mode, and line selection mode of the ONLY
service
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
pbxusergroup user-group-name unicall [ enterprise enterprise-name ]
The ONLY service is configured.

By default, no ONLY service is configured.
Step 4 Run:
dn-set dn-set-value
A DN set is bound to the ONLY service.

Step 5 Run:
An access number is set for the ONLY service.

The ringing duration, ringing mode, and line selection mode are set for the ONLY service.
----End

2.15.13 Configuring the Blacklist or Whitelist Function

You can implement the blacklist or whitelist function by configuring a rule set blacklist or
whitelist for a trunk group to control call rights of users.
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After a trunk group is configured, all calls meeting the call conditions configured in the trunk
group are allowed. To restrict call rights of users, you can configure a rule set for the trunk group
to allow or reject calls of specified users.
Before configuring the blacklist or whitelist function, complete the following task:
l
Configuring the trunk group
Data Preparation
To configure the blacklist or whitelist function, you need the following data.
No.
Data
Name of the rule set
Enterprise to which the rule set is applied
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
rule-set rule-set-name
A rule set is configured to implement the blacklist or whitelist function.

Step 4 Run:
rule { deny | permit } { incoming | outgoing } caller-telno caller-telno-value
The rule set type (blacklist or whitelist) and restricted call type are specified.
Step 5 Run:
An enterprise is bound to the rule set.

Step 6 Run:
quit

Step 7 Run:
trunk-group name
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The trunk group view is displayed.

Step 8 Run:
use-rule-set rule-set-name
The rule set is applied to the trunk group.

----End

Run the display voice rule-set [ rule-set-name ] command to check the blacklist/whitelist
configuration.
2.15.14 Configuring the DISA Service

The direct inward system access (DISA) service controls the incoming calls and outgoing calls
on a PBX. Only authenticated users can use voice services on the PBX.
The DISA service can be used in the following scenarios:
l
Controlling incoming calls made by external users

To prevent unauthorized users to connect to an enterprise's PBX, the DISA service can be
configured on the PBX to enable the PBX to authenticate PBX users. To use the voice
service provided by the PBX, an external user needs to dial a specified phone number of
the enterprise and then enter authentication information as prompted. The user can use
voice services provided by the PBX only after being authenticated by the PBX. As shown
in Figure 2-12, the PBX has the DISA service configured, with the access number 800.
When user A dials 25600000, a voice message is played, prompting user A to enter
authentication information. After user A is authenticated, the PBX provides voice services
to user A.
NOTE
The DISA service access number is an internal number of the enterprise.
Figure 2-12 Controlling incoming calls from external users
PSTN
User A
Port 1/0/4
AT0 trunk
External number:
25600000
Internal number: 800
PBX
User B
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Controlling outgoing calls made by internal users

When an enterprise wants to provide differentiated voice services to internal users, the
DISA service can be configured on the enterprise's PBX. After internal users are
authenticated, they can obtain voice services in addition to the basic voice service. As shown
in Figure 2-13, the DISA service is configured on the PBX to control the call-out right of
internal users. User B and user C are internal users, and user A is an external user. When
user B and user C need to call user A, they dial the DISA service access number and enter
authentication information as prompted. They can call user A only after being
authenticated.
Figure 2-13 Controlling outgoing calls from internal users
PSTN
User A
Port 1/0/4
AT0 trunk
PBX
User B
User C
Before configuring the DISA service, complete the following task:
l
Data Preparation
To configure the DISA service, you need the following data.
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No.
Data
Name of the DISA service account set
Enterprise name

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Procedure
Step 1 Create an account set and configure the authentication mode for the account set.
1.
Run:
system-view

2.
Run:
voice

3.
Run:
account-set account-set-name [account | password ]
An account set is created and the authentication mode is configured.

4.
Run:
An enterprise is bound to the account set.

Step 2 Configure an account and password for the account set.
l
When the account set uses account authentication, run the account account-value
password { simple | cipher } password-value command to configure an account and
password for the account set.
When the account set uses password authentication, run the password { simple | cipher}
password-value command to configure a password for the account set.
Step 3 Run:
quit

Step 4 Configure the DISA service.
1.
Run:
disa disa-name [ enterprise enterprise-name ]
A DISA service profile is created and the DISA service view is displayed.
2.
Run:
dn-set dn-set-name
A DN set is bound to the DISA service.

3.
Use either of the following methods to configure the DISA service access number. (Use
the first method when the PBX needs to control outgoing calls from external users.)
l Run the access-telno [ country-code country-code-value ] [ area-code area-codevalue ] telno-value command to set to the DISA service access number to a user phone
number.
l Run the access-telno centrex centrex-name centrex-number command to set the DISA
service access number to a Centrex group number.
4.
Run:
use-account-set account-set-name
The account set is bound to the DISA service.

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(Optional) Run:
dst-enterprise dst-enterprise-name [ dst-dnset dst-dnset-name ] [ dst-centrex
dst-centrex-name ]
The destination enterprise, destination DN set, and destination Centrex are specified for
the DISA service.
----End

Run the following commands to verify the configuration:
l
Run the display voice account-set [ account-value ] command to check the configuration
of the account set.
Run the display voice disa [ disa-name ] command to check the DISA service
configuration.
2.15.15 Configuring the Immediate Conference Service

The intermediate conference service allows more than three users to join a voice conversation.
During a conversation between user A and user B, user A or user B can press the hookflash and
then dial 3 and user C's phone number to invite user C to the conversation. If they want to invite
other users to the conversation, the intermediate service is required. If user A has the intermediate
conference right, user A can press hookflash and dial 6 and user D's phone number to invite user
D to the conference.
Figure 2-14 shows the intermediate conference usage scenario.
Figure 2-14 Networking of the intermediate conference service
IMS/IP
network
PBX
User A
(POTS)
User C
(POTS)
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User D
(POTS)
User B
(SIPUE)
User E
(SIPUE)

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NOTE
To use the intermediate conference service, the user that initiates an intermediate conference must be in a
three-party conversation.
Before configuring the intermediate conference service, complete the following task:
l
Perform the following steps to configure the intermediate conference service for the POTS
and SIPUE users connected to the PBX:
Procedure
1.
Run:
system-view

2.
Run:
voice

3.
Run:
pbxuser name

4.
Run:
service-right instant-conference enable
The intermediate conference service right is granted to the PBX user.

5.
Run:
quit

6.
Run:
The call prefix profile view is displayed.

7.
Run:
call-type category supplementary-service attribute 191
The call type and call attribute are configured for the intermediate conference service.
8.
Run:
quit
Return to the PBX view.

9.
(Optional) Run:
pbx service-priority
The priority of the intermediate conference service is configured.
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NOTE
l By default, the three-party service has a higher priority than the intermediate conference
service. The three-party and intermediate conference services have the same service process
when there are three parties in a conversation. When a user has the rights of both the threeparty service and intermediate conference service, the PBX determines whether to start the
three-party service or intermediate conference service based on priorities of the two
services.
l To configure the PBX to start the intermediate conference service, you can also run the
service-right three-party disable command to disable the three-party service. In this case,
you do not need to perform step 9.
Perform the following steps to enable the POTS and SIPUE users connected to the SIP AG
to use the intermediate conference service:
1.
Run:
system-view

2.
Run:
voice

3.
Run:
sipag
The SIP AG view is displayed.

4.
Run:
conference-factory-uri
The conference factory URI is specified for the SIP AG.

This section provides a PBX configuration example.
2.16.1 Example for Configuring Voice Services for a Small- or

Medium-sized Enterprise
As shown in Figure 2-15, an enterprise has POTS users: User A, User B, and User C. The
requirements are as follows:
l
Internal calls of the enterprises are connected through the PBX, and outgoing calls from
the enterprises are connected to external users through the AT0 trunk.
The carrier allocates the number 56623000 to the enterprise. External users can dial the
number 56623000 to query internal extension number. External users can also dial the
number 56623000, and then the call is transferred to an internal user.
NOTE
This example uses the voice tone "Please dial the extension number, or dial zero for the operator."
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Figure 2-15 Networking for configuring voice services for a small- or medium-sized enterprise
PSTN
AT0 trunk
Port 1/0/4
RouterA
Port 1/0/0
Port 1/0/2
Port 1/0/1
User A
User C
User B
1.
Set the service mode to PBX.
2.
Configure the default country code and area code.
3.
Configure the enterprise, DN set, and RTB file.
4.
Configure prefixes.
5.
Configure PBX users.
6.
Configure a trunk and a trunk group for inter-office calls.
7.
Configure a call route and a post-routing number change plan.
8.
Configure an IVR group.
Data Preparation
l
Country code and region code: 86 and 25

NOTE
The country code and region code in China are used as an example.
Extension numbers of User A, User B, and User C: 800, 801, and 802 (the access number
of the IVR group is 800)
Enterprise that User A, User B, and User C belongs to: hw
DN set: local
Intra-office and inter-office call prefixes: 8 and 9
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Call-in and call-out permissions: 3
IVR group name and user-defined RBT file name: ivr1 and sss.wav
NOTE
If the user-defined RBT is used, ensure that the RBT file has been made and uploaded/downloaded
to the storage media
Procedure
Step 1 Set the service mode to PBX.
NOTE
The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
purchased device. To use the PBX functions, apply for and purchase the license from the Huawei local
office.
<RouterA> system-view
[RouterA] voice
[RouterA-voice] service-mode pbx
Changing of the protocol configuration takes effect after you save the data and
then reboot the system. Are you sure to change the protocol configuration? (y/n
)[n] : y
[RouterA-voice] quit
[RouterA] quit
<RouterA> save
Are you sure to continue? [y/n]y
<RouterA> reboot
Info: The system is now comparing the configuration, please wait.
Step 2 Configure the default country code and area code.

[RouterA] voice
[RouterA-voice] pbx default-country-code 86 default-area-code 25
Step 3 Configure the enterprise, DN set, and RTB file.

[RouterA-voice] enterprise hw
[RouterA-voice-enterprise-hw] dn-set local
[RouterA-voice-enterprise-hw] crbt-file flash:/sss.wav status pass
[RouterA-voice-enterprise-hw] quit
Step 4 Configure prefixes.

[RouterA-voice] callprefix 8
[RouterA-voice-callprefix-8]
enterprise hw dn-set local

prefix 8
digit-length 3 4
quit
prefix 9
digit-length 1 15
destination-location inter-office
quit
Step 5 Configure PBX users.

# Configure User A.
[RouterA-voice] pbxuser 800
[RouterA-voice-pbxuser-800]
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pots enterprise hw
port 1/0/0
telno country-code 86 area-code 25 800
dn-set local

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2 PBX Configuration
[RouterA-voice-pbxuser-800] call-right in international-toll out internationaltoll

[RouterA-voice-pbxuser-800] service-right call-transfer enable
[RouterA-voice-pbxuser-800] quit
# Configure User B.
toll
pots enterprise hw
port 1/0/1
dn-set local
call-right in international-toll out internationalquit
# Configure User C.
toll
pots enterprise hw
port 1/0/2
dn-set local
Step 6 Configure a trunk and a trunk group.

[RouterA-voice] trunk-group at0 fxo
[RouterA-voice-trunkgroup-at0] enterprise hw dn-set local
[RouterA-voice-trunkgroup-at0] call-right in international-toll out internationaltoll
[RouterA-voice-trunkgroup-at0] trunk-at0 1/0/4 default-called-telno 800
reversepole-detect disable
[RouterA-voice-trunkgroup-at0] quit
Step 7 Configure a call route and a post-routing number change plan.

Configure a call route.
[RouterA-voice-callprefix-9] callroute trunkgroup1 at0
[RouterA-voice-callprefix-9] quit
# Configure a post-routing number change plan.

[RouterA-voice] afterroute-change 9
[RouterA-voice-afterroute-change-9]
callprefix 9
trunk-group at0
caller no-change
called del 7 1
quit
Step 8 Configure an IVR group.

[RouterA-voice] pbxusergroup ivr1 ivr enterprise hw
[RouterA-voice-ivr-group-ivr1] dn-set local
[RouterA-voice-ivr-group-ivr1] access-telno 800
[RouterA-voice-ivr-group-ivr1] console-telno 0
[RouterA-voice-ivr-group-ivr1] tone-id file flash:/sss.wav
[RouterA-voice-ivr-group-ivr1] group-member pbxuser 800
[RouterA-voice-ivr-group-ivr1] quit

1.
When external users dial the number 56623000, they can dial extension numbers to
communicate with internal users.
2.
User A, User B, and User C can call each other.
3.
User A, User B, and User C can make inter-office calls.
----End
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Configuration Files
Configuration file of RouterA
voice
pbx default-area-code 25
#
enterprise hw
crbt-file flash:/sss.wav status pass
dn-set local
#
r2 signalling-type argentina
#
r2 signalling-type brazil
#
r2 signalling-type mexico
#
r2 signalling-type standard
#
trunk-group at0 fxo
call-right in international-toll out international-toll
trunk-at0 1/0/4 default-called-telno 800 reversepole-detect disable
#
callprefix 8
prefix 8
digit-length 3 4
#
callprefix 9
prefix 9
digit-length 1 15
callroute trunkgroup1 at0
#
pbxuser 800 pots enterprise hw
port 1/0/0
telno 800
dn-set local
#
port 1/0/1
telno 801
dn-set local
#
port 1/0/2
telno 802
dn-set local
#
pbxusergroup ivr1 ivr enterprise hw
dn-set local
access-telno 800
console-telno 2
tone-id file flash:/sss.wav
destination dn-set DefaultDialPlan
group-member pbxuser 800 member-index 1
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
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called del 7 1
#
return
2.16.2 Example for Configuring the AR to Implement

Communication for Different Enterprises
As shown in Figure 2-16, enterprise A and enterprise B are located in the same industry park.
User A and User B belong to enterprise A, and User C and User D belong to enterprise B. The
AR logically isolates voice services of the two enterprises. Internal calls of the enterprises are
connected through their respective virtual PBXs on the AR, and outgoing calls from the
enterprises are connected to external users through one port. This networking helps enterprises
lower investment on equipment and reduces the number of access nodes on the carrier network.
The requirements are as follows:
l
The carrier allocates the number 56623000 to enterprise A. If external users dial the number
56623000, the phone of User A rings and the call transfer service is enabled. When external
users call other internal users, the phone of User A transfers the calls.
The carrier allocates the number 56623001 to enterprise B. If external users dial the number
56623001, the phone of User C rings and the call transfer service is enabled. When external
users call other internal users, the phone of User C transfers the calls.
Figure 2-16 Networking for configuring the AR to implement communication for different
enterprises
IMS/IP
network
Trunk
Router Eth2/0/0
Port 1/0/0
User A
Port
1/0/1
Port 1/0/3
Port
1/0/2
User B
User D
User C
Enterprise A
Enterprise B
Campus network
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1.
2.
Configure signaling and media IP addresses.
3.
4.
Configure the enterprise and DN set.
5.
Configure the SIP server.
6.
Configure prefixes.
7.
8.
Configure trunks, trunk groups, and routes for inter-office calls.
9.
Configure call routes and post-routing number change plans.
Data Preparation
l

NOTE
Numbers of User A, User B, User C, and User D: 2000, 2001, 3000, and 3001
Signaling and media IP addresses: 192.168.1.3
Enterprise huawei to which user A belongs, DN set local, intra-office call prefix 2, and
inter-office call prefix 8
Enterprise huawei to which user B belongs, DN set local1, intra-office call prefix 3, and
inter-office call prefix 9
IP address and port number of the IMS: 192.168.1.1 and 5060
Procedure
Step 1 Set the service mode to PBX.
NOTE
office.
[Huawei] voice
[Huawei-voice] service-mode pbx
)[n] : y
[Huawei-voice] quit
[Huawei] quit
<Huawei> save
<Huawei> reboot
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Step 2 Configure signaling and media IP addresses.

[Huawei] interface ethernet2/0/0
[Huawei-Ethernet2/0/0] ip address 192.168.1.3 24
[Huawei-Ethernet2/0/0] quit
# Configure signaling and media IP addresses.

[Huawei] voice
[Huawei-voice] voip-address signalling interface ethernet 2/0/0 192.168.1.3
[Huawei-voice] voip-address media interface ethernet 2/0/0 192.168.1.3

[RouterA] voice
Step 4 Configure the enterprise and DN set.

[Huawei-voice] enterprise hw
[Huawei-voice-enterprise-hw] dn-set local
[Huawei-voice-enterprise-hw] quit
[Huawei-voice] enterprise hw1
[Huawei-voice-enterprise-hw1] dn-set local1
[Huawei-voice-enterprise-hw1] quit

[Huawei-voice] sipserver
[Huawei-voice-sipserver] signalling-address ip 192.168.1.3 port 5060
[Huawei-voice-sipserver] media-ip 192.168.1.3
[Huawei-voice-sipserver] register-uri huawei.com
[Huawei-voice-sipserver] home-domain huawei.com
[Huawei-voice-sipserver] reset
SIP server reset succeeds.
[Huawei-voice-sipserver] quit


prefix 2
digit-length 4 8
quit
prefix 8
digit-length 1 15
quit
enterprise hw1 dn-set local1
prefix 3
digit-length 4 8
quit
prefix 9
digit-length 1 15
quit

# Configure User A.
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[Huawei-voice] pbxuser 2000
[Huawei-voice-pbxuser-2000]
toll
2 PBX Configuration
pots enterprise hw
port 1/0/0
dn-set local
call-right in international-toll out internationalservice-right call-transfer enable
quit
# Configure User B.
toll
pots enterprise hw
port 1/0/1
dn-set local
# Configure User C.
toll
pots enterprise hw1

port 1/0/2
dn-set local1
call-right in international-toll out internationalservice-right call-transfer enable
quit
# Configure User D.
toll
pots enterprise hw1

port 1/0/3
dn-set local1
Step 8 Configure trunks, trunk groups, and routes.

# Configure a SIP AT0 trunk group.
[Huawei-voice] trunk-group sipat0 sip trunk-circuit
[Huawei-voice-trunkgroup-sipat0] enterprise hw dn-set local
[Huawei-voice-trunkgroup-sipat0] call-right in international-toll out
international-toll
[Huawei-voice-trunkgroup-sipat0] default-caller-telno 2000
[Huawei-voice-trunkgroup-sipat0] signalling-address ip 192.168.1.3 port 5070
[Huawei-voice-trunkgroup-sipat0] media-ip 192.168.1.3
[Huawei-voice-trunkgroup-sipat0] peer-address static 192.168.1.1 5060
[Huawei-voice-trunkgroup-sipat0] register-uri huawei.com
[Huawei-voice-trunkgroup-sipat0] home-domain huawei.com
[Huawei-voice-trunkgroup-sipat0] register-id
56623000
[Huawei-voice-trunkgroup-sipat0] trunk-sipat0 56623000 default-called-telno 2000
[Huawei-voice-trunkgroup-sipat0] reset
Note: Trunkgroup reset succeeds.
[Huawei-voice-trunkgroup-sipat0] quit
[Huawei-voice] trunk-group sipat01 sip trunk-circuit
[Huawei-voice-trunkgroup-sipat01] enterprise hw1 dn-set local1
[Huawei-voice-trunkgroup-sipat01] call-right in international-toll out
international-toll
[Huawei-voice-trunkgroup-sipat01] default-caller-telno 3000
[Huawei-voice-trunkgroup-sipat01] signalling-address ip 192.168.1.3 port 5080
[Huawei-voice-trunkgroup-sipat01] callout-right 3
[Huawei-voice-trunkgroup-sipat01] media-ip 192.168.1.3
[Huawei-voice-trunkgroup-sipat01] peer-address static 192.168.1.1 5060
[Huawei-voice-trunkgroup-sipat01] register-uri huawei.com
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[Huawei-voice-trunkgroup-sipat01]
huawei.com
3000
home-domain
register-id 56623001
trunk-sipat0 56623001 default-called-telno
reset
quit
Step 9 Configure call routes and post-routing number change plans.

# Configure call routes.
callroute trunkgroup1 sipat0

quit
quit
# Configure post-routing number change plans.

callprefix 8
trunk-group sipat0
caller no-change
called del 7 1
quit
callprefix 9
trunk-group sipat01
caller no-change
called del 7 1
quit

1.
User A and User B can talk with each other.
2.
User C and User D can talk with each other.
3.
User A, User B, User C, and User D can call external users.
4.
When dialing the number 56623000, external users can talk with User A and User B.
5.
When dialing the number 56623001, external users can talk with User C and User D.
----End
Configuration Files
Configuration file of the router
#
voice
#
dspattribute
#
enterprise hw
dn-set local
#
enterprise hw1
dn-set local1
#
sipserver
media-ip 192.168.1.3
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#
#
#
#
#
trunk-group sipat0 sip trunk-circuit
default-caller-telno 2000
media-ip 192.168.1.3
peer-address static 192.168.1.1 5060
trunk-sipat0 56623000 default-called-telno 2000
#
trunk-group sipat01 sip trunk-circuit
default-caller-telno 3000
media-ip 192.168.1.3
trunk-sipat0 56623001 default-called-telno 3000
#
callprefix 2
prefix 2
digit-length 4 8
#
callprefix 3
prefix 3
digit-length 4 8
#
callprefix 8
prefix 8
digit-length 1 15
#
callprefix 9
prefix 9
digit-length 1 15
#
port 1/0/0
telno 2000
dn-set local
#
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afterroute-change 8
callprefix 8
trunk-group sipat0
caller no-change
called del 7 1
#
afterroute-change 9
callprefix 9
trunk-group sipat01
caller no-change
called del 7 1
#
return
2.16.3 Example for Using an H.323 Trunk Group in Peer Mode to

Configure Calls Between the Headquarters and Branch
As shown in Figure 2-17, the headquarters and branch of enterprise A are located in different
areas. RouterA and RouterB function as gateways and are connected through the H323 trunk.
After voice services are deployed on RouterA and RouterB, enterprise users can use the voice
services across areas. Internal users use the AT0 trunk to call external users.
l
The carrier allocates the number 56623000 to the enterprise headquarters. If external users
dial the number 56623000, the phone of User A rings and the call transfer service is enabled.
When external users call other internal users, the phone of User A transfers the calls.
The carrier allocates the number 28963000 to the enterprise branch. If external users dial
the number 28963000, the phone of User C rings and the call transfer service is enabled.
When external users call other internal users, the phone of User C transfers the calls.
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Figure 2-17 Networking diagram for configuring calls between the headquarters and branch
Enterprise A
Branch
User C
User D
Port1/0/0
Router B
Port1/0/1
Port1/0/4
Eth2/0/0
PSTN
H323 trunk
IP network
PSTN
H323 trunk
Eth2/0/0
Port1/0/4
Router A
Port1/0/0
Port1/0/1
User A
User B
Enterprise A
Headquarters
1.
2.
3.
Configure a country code and a region code.
4.
5.
Set H323 system parameters.
6.
Configure prefixes.
7.
8.
9.
Configure a call route and post-routing number change.
Data Preparation
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Country code 86, area code 25 of Router A, and area code 755 of Router B
NOTE
The country code and region code in China are used as an example. During deployment, configure
the country code and region code based on actual networking.
Number of user A: 22223000
Number of user B: 22223001
Number of user C: 33333000
Number of user D: 33333001
Media and signaling IP address of the headquarters: 192.168.1.1
Media and signaling IP address of the branch: 192.168.1.2
Enterprise hw to which user A and user B belong, DN set local, call prefix 2222, interoffice prefix 9 of the AT0 trunk, and inter-office prefix 20000 between the headquarters
and branch
Enterprise hw to which user C and user D belong, DN set local, call prefix 3333, interoffice prefix 9 of the AT0 trunk, and inter-office prefix 20000 between the headquarters
and branch
Procedure
Step 1 Set the service mode to PBX on RouterA and RouterB.
NOTE
office.
# Configure RouterA to work in PBX mode.

[RouterA] voice
)[n] : y
[RouterA] quit
<RouterA> save
<RouterA> reboot
# Configure RouterB to work in PBX mode.

<RouterB> system-view
[RouterB] voice
[RouterB-voice] service-mode pbx
)[n] : y
[RouterB-voice] quit
[RouterB] quit
<RouterB> save
Are you sure to continue? [Y/N]y
<RouterB> reboot
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# Configure signaling and media IP addresses on RouterA.
l Configure a signaling IP address.
[RouterA] interface ethernet 2/0/0
[RouterA-Ethernet2/0/0] ip address 192.168.1.1 24
[RouterA-Ethernet2/0/0] quit
l Configure a media IP address.

[RouterA] voice
[RouterA-voice] voip-address signalling interface ethernet 2/0/0 192.168.1.1
[RouterA-voice] voip-address media interface ethernet 2/0/0 192.168.1.1
# Configure signaling and media IP addresses on RouterB.

[RouterB] interface ethernet 2/0/0
[RouterB-Ethernet2/0/0] ip address 192.168.1.2 24
[RouterB-Ethernet2/0/0] quit

[RouterB] voice
[RouterB-voice] voip-address signalling interface ethernet 2/0/0 192.168.1.2
[RouterB-voice] voip-address media interface ethernet 2/0/0 192.168.1.2
Step 3 Configure a country code and a region code.

# Configure RouterA.
[RouterA] voice
# Configure RouterB.
[RouterB] voice
[RouterB-voice] pbx default-country-code 86 default-area-code 755
Step 4 Configure the enterprise, centrex group, and DN set.

[RouterB-voice] enterprise hw
[RouterB-voice-enterprise-hw] dn-set local
[RouterB-voice-enterprise-hw] quit
Step 5 Set H323 attribute.

# Set H323 attribute on RouterA.
[RouterA-voice] h323-attribute
[RouterA-voice-h323-attribute] localip 192.168.1.1
[RouterA-voice-h323-attribute] reset
H323 system parameters reset successfully!
[RouterA-voice-h323-attribute] quit
# Set H323 attribute on RouterB.

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[RouterB-voice] h323-attribute
[RouterB-voice-h323-attribute] localip 192.168.1.2
[RouterB-voice-h323-attribute] reset
H323 system parameters reset successfully!

# Configure prefixes on RouterA.
[RouterA-voice-callprefix-2222] enterprise hw dn-set local
[RouterA-voice-callprefix-2222] prefix 2222
[RouterA-voice-callprefix-2222] digit-length 8 9
[RouterA-voice-callprefix-2222] call-type category basic-service attribute 0
[RouterA-voice-callprefix-9] destination-location inter-office
# Configure prefixes on RouterB.

[RouterB-voice] callprefix 3333
[RouterB-voice-callprefix-3333] enterprise hw dn-set local
[RouterB-voice-callprefix-3333] prefix 3333
[RouterB-voice-callprefix-3333] call-type category basic-service attribute 0
[RouterB-voice-callprefix-3333] digit-length 8 9
[RouterB-voice-callprefix-3333] quit
[RouterB-voice-callprefix-9] destination-location inter-office

# Configure user A.
international-toll
pots enterprise hw
port 1/0/0
dn-set local
call-right in international-toll out
service-right call-transfer enable
quit
# Configure user B.
[RouterA-voice] pbxuser 22223001 pots enterprise hw
[RouterA-voice-pbxuser-22223001] port 1/0/1
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international-toll
2 PBX Configuration
dn-set local
quit
# Configure user C.
[RouterB-voice] pbxuser 33333000
[RouterB-voice-pbxuser-33333000]
international-toll
pots enterprise hw
port 1/0/0
dn-set local
quit
# Configure user D.
international-toll
pots enterprise hw
port 1/0/1
dn-set local
quit
Step 8 Configure trunk groups and routes.

# Configure trunk groups and routes on RouterA.
l Configure an H323 trunk group.
[RouterA-voice] trunk-group h323 h323 symmetrical
[RouterA-voice-trunkgroup-h323] enterprise hw dn-set local
[RouterA-voice-trunkgroup-h323] call-right in international-toll out
international-toll
[RouterA-voice-trunkgroup-h323] signalling-ip ip 192.168.1.1
[RouterA-voice-trunkgroup-h323] media-ip 192.168.1.1
[RouterA-voice-trunkgroup-h323] peer-address 192.168.1.2 1720
[RouterA-voice-trunkgroup-h323] quit
l Configure an AT0 trunk group.

[RouterA-voice-trunkgroup-at0] call-right in international-toll out
international-toll
l Configure a call route.

[RouterA-voice-callprefix-20000] callroute trunkgroup1 h323
# Configure trunks, trunk groups, and routes on RouterB.

[RouterB-voice] trunk-group h323 h323 symmetrical
international-toll
[RouterA-voice-trunkgroup-h323] peer-address static 192.168.1.1 1720

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[RouterB-voice] trunk-group at0 fxo

[RouterB-voice-trunkgroup-at0] enterprise hw dn-set local
[RouterB-voice-trunkgroup-at0] call-right in international-toll out
international-toll
[RouterB-voice-trunkgroup-at0] trunk-at0 1/0/4 default-called-telno 33333000

[RouterB-voice-callprefix-9] callroute trunkgroup1 at0
[RouterB-voice-callprefix-20000] callroute trunkgroup1 h323
Step 9 Configure post-routing number change.

# Configure post-routing number change on RouterA.
[RouterA-voice-afterroute-change-9] callprefix 9
[RouterA-voice-afterroute-change-9] trunk-group at0
[RouterA-voice-afterroute-change-9] caller no-change
[RouterA-voice-afterroute-change-9] called del 7 1
[RouterA-voice-afterroute-change-9] quit
[RouterA-voice-afterroute-change-20000] trunk-group h323
# Configure post-routing number change on RouterB.

[RouterB-voice-afterroute-change-9] callprefix 9
[RouterB-voice-afterroute-change-9] trunk-group at0
[RouterB-voice-afterroute-change-9] caller no-change
[RouterB-voice-afterroute-change-9] called del 8 1
[RouterB-voice-afterroute-change-9] quit
[RouterB-voice] afterroute-change 20000
[RouterB-voice-afterroute-change-20000] trunk-group h323

1.
User A and user B can talk with each other.
2.
User C and user D can talk with each other.
3.
User A, user B, user C, and user D can call external users.
4.
When dialing the number 56623000, external users can talk with user A and user B.
5.
When dialing the number 28963000, external users can talk with user C and user D.
----End
Configuration Files
# Configuration file of RouterA
#
voice
sip-reg-count-per-second 4294967295
#
h323-
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attribute
localip 192.168.1.1
#
enterprise hw
dn-set local
#
#
#
#
#
trunk-group at0 fxo
#
trunk-group h323 h323 symmetrical
signalling-ip ip 192.168.1.1
media-ip 192.168.1.1
#
callprefix 9
prefix 9
digit-length 1 15
#
callprefix 2222
prefix 2222
digit-length 8 9
#
callprefix 20000
prefix 20000
digit-length 5 20
callroute trunkgroup1 h323
#
port 1/0/0
telno 22223000
dn-set local
#
port 1/0/1
telno 22223001
dn-set local
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 7 1
#
afterroute-change 20000
callprefix 20000
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trunk-group h323
caller no-change
called del 7 5
#
return
# Configuration file of RouterB

#
voice
voip-address signalling interface GigabitEthernet 0/0/1 192.168.1.2
voip-address media interface GigabitEthernet 0/0/1 192.168.1.2
#
h323attribute
localip 192.168.1.2
#
enterprise hw
dn-set local
#
#
#
#
#
trunk-group at0 fxo
#
trunk-group h323 h323 symmetrical
media-ip 192.168.1.2
#
callprefix 9
prefix 9
digit-length 1 15
#
callprefix 3333
prefix 3333
digit-length 8 9
#
callprefix 20000
prefix 20000
digit-length 5 20
#
port 1/0/0
telno 33333000
dn-set local
#
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port 1/0/1
telno 33333001
dn-set local
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 8 1
#
callprefix 20000
trunk-group h323
caller no-change
called del 8 5
#
return
2.16.4 Example for Using an H.323 Trunk Group in Registration

Mode to Configure Calls Between the Headquarters and Branch
areas. RouterA functions as the gateway, and RouterB is a non-Huawei device and functions as
the gatekeeper. RouterA and RouterB are connected through the H323 trunk. After voice services
are deployed on RouterA and RouterB, enterprise users can use the voice services across areas.
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Figure 2-18 Networking diagram for configuring calls between the headquarters and branch
Enterprise A
Branch
User C
User D
Port1/0/0
Router B
H323 trunk
Port1/0/1
Eth2/0/0
IP network
Eth2/0/0
H323 trunk
Router A
Port1/0/0
Port1/0/1
User A
User B
Enterprise A
Headquarters
1.
2.
3.
Configure a country code and a region code.
4.
5.
Set H323 attribute.
6.
Configure prefixes.
7.
8.
Configure trunk groups and routes for inter-office calls.
9.
Configure a call route and post-routing number change.
Data Preparation
To complete the configuration, you need the following data (data of Router A):
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NOTE
The country code and region code in China are used as an example. During deployment, configure
the country code and region code based on actual networking.
Number of user A: 22223000 and number of user B: 22223001 Number of user C: 33333000
and number of user D: 33333001
Signaling and media IP addresses and signaling port number of the headquarters:
192.168.1.1 Signaling and media IP addresses and signaling port number of the branch:
192.168.1.2
Enterprise huawei to which user A and user B belong, DN set local, call prefix 2222, and
inter-office prefix 20000 between the headquarters and branch
Procedure
Step 1 Configure RouterA to work in PBX mode.
NOTE
l The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
office.
l The preceding configuration is the configuration of RouterA.

[RouterA] voice
)[n] : y
[RouterA] quit
<RouterA> save
<RouterA> reboot
Step 2 Configure signaling and media IP addresses on RouterA.


[RouterA] voice
Step 3 Configure a country code and a region code.

[RouterA] voice
Step 4 Configure the enterprise, centrex group, and DN set.

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Step 5 Set H323 attribute.

[RouterA-voice] h323-attribute
[RouterA-voice-h323-attribute] localip 192.168.1.1
[RouterA-voice-h323-attribute] reset
H323 attribute reset successfully!


# Configure user A.
international-toll
pots enterprise hw
port 1/0/0
dn-set local
quit
# Configure user B.
international-toll
pots enterprise hw
port 1/0/1
dn-set local
quit

NOTE
The same gateway ID must be configured on devices of the headquarters and branch.
[RouterA-voice] trunk-group h323 h323 register-gateway
international-toll
[RouterA-voice-trunkgroup-h323] gwid h323
[RouterA-voice-trunkgroup-h323] quit

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[RouterA-voice-callprefix-20000] callroute trunkgroup1 h323

callprefix 20000
trunk-group h323
caller no-change
called del 7 5

1.
User A and user B can talk with each other.
2.
User A and user B can talk with user C and user D.
----End
Configuration Files
#
voice
voip-address signalling interface GigabitEthernet 0/0/1 192.168.1.1
voip-address media interface GigabitEthernet 0/0/1 192.168.1.1
#
h323attribute
localip 192.168.1.1
#
enterprise hw
dn-set local
#
#
#
#
#
trunk-group h323 h323 register-gateway
media-ip 192.168.1.1
gwid h323
#
callprefix 2222
prefix 2222
digit-length 8 9
#
callprefix 20000
prefix 20000
digit-length 5 20
#
port 1/0/0
telno 22223000
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dn-set local
#
port 1/0/1
telno 22223001
dn-set local
#
callprefix 20000
trunk-group h323
caller no-change
called del 7 5
#
return
2.16.5 Example for Using the SIP Trunk Group to Configure Calls
Between the Headquarters and Branch
areas. RouterA and RouterB use SIP trunks to connect the headquarters and branch. After voice
services are deployed on RouterA and RouterB, enterprise users can use the voice services across
areas.Internal users call external users through the AT0 trunk. The requirements are as follows:
l
The carrier allocates the number 56623000 to the enterprise headquarters. If external users
dial the number 56623000, the phone of User A rings and the call transfer service is enabled.
When external users call other internal users, the phone of User A transfers the calls.
The carrier allocates the number 28963000 to the enterprise branch. If external users dial
the number 28963000, the phone of User C rings and the call transfer service is enabled.
When external users call other internal users, the phone of User C transfers the calls.
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Figure 2-19 Networking for configuring communication between the headquarters and branch
Enterprise A
Branch
User C
User D
Port1/0/0
Port1/0/1
Router B
Port1/0/4
Eth2/0/0
SIP trunk
PSTN
IP network
PSTN
SIP trunk
Port1/0/4
Eth2/0/0
Router A
Port1/0/0
Port1/0/1
User A
User B
Enterprise A
Headquarters
1.
2.
3.
4.
5.
Configure the SIP server.
6.
Configure prefixes.
7.
8.
9.
Configure call routes and post-routing number change plans.
Data Preparation
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Country code 86, area code 2 of Router A, and area code 755 of Router B
NOTE
Internal numbers of User A, User B, User C, and User D: 22223000, 22223001, 33333000,
and 33333001
Signaling and media IP addresses and signaling port number of the headquarters:
192.168.1.1 and 5070
Signaling and media IP addresses and signaling port number of the branch: 192.168.1.2
and 5070
Enterprise huawei to which user A and user B belong, DN set local, call prefix 2222, interoffice call prefix 9 of the AT0 trunk, and inter-office call prefix 20000 between the
headquarters and branch
Enterprise huawei to which user C and user D belong, DN set local, call prefix 3333, interoffice call prefix 9 of the AT0 trunk, and inter-office call prefix 20000 between the
headquarters and branch
Procedure
Step 1 Set the service mode to PBX on RouterA and RouterB.
NOTE
office.

[RouterA] voice
)[n] : y
[RouterA] quit
<RouterA> save
<RouterA> reboot
# Configure RouterB to work in PBX mode.

[RouterB] voice
[RouterB-voice] service-mode pbx
)[n] : y
[RouterB-voice] quit
[RouterB] quit
<RouterB> save
Are you sure to continue? [Y/N]y
<RouterB> reboot
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# Configure signaling and media IP addresses on RouterA.

[RouterA] voice
# Configure signaling and media IP addresses on RouterB.

[RouterB] interface ethernet 2/0/0
[RouterB-Ethernet2/0/0] ip address 192.168.1.2 24
[RouterB-Ethernet2/0/0] quit

[RouterB] voice
[RouterB-voice] voip-address signalling interface ethernet 2/0/0 192.168.1.2
[RouterB-voice] voip-address media interface ethernet 2/0/0 192.168.1.2

# Configure the default country code and area code on RouterA.
[RouterA] voice
# Configure the default country code and area code on RouterB.

[RouterB] voice
[RouterB-voice] pbx default-country-code 86 default-area-code 755
Step 4 Configure the enterprise and DN set.

[RouterA-voice] pbx
[RouterB-voice] pbx
[RouterB-voice] enterprise hw
[RouterB-voice-enterprise-hw] dn-set local
[RouterB-voice-enterprise-hw] quit

# Configure RouterA as the SIP server.
[RouterA-voice] sipserver
[RouterA-voice-sipserver] signalling-address ip 192.168.1.1 port 5060
[RouterA-voice-sipserver] media-ip 192.168.1.1
[RouterA-voice-sipserver] register-uri huawei.com
[RouterA-voice-sipserver] home-domain huawei.com
[RouterA-voice-sipserver] reset
[RouterA-voice-sipserver] quit
# Configure RouterB as the SIP server.

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[RouterB-voice] sipserver
[RouterB-voice-sipserver] signalling-address ip 192.168.1.2 port 5060
[RouterB-voice-sipserver] media-ip 192.168.1.2
[RouterB-voice-sipserver] register-uri huawei.com
[RouterB-voice-sipserver] home-domain huawei.com
[RouterB-voice-sipserver] reset
[RouterB-voice-sipserver] quit

# Configure prefixes on RouterA.
# Configure prefixes on RouterB.


# Configure User A.
international-toll
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port 1/0/0
dn-set local
quit

217

2 PBX Configuration
# Configure User B.
international-toll
pots enterprise hw
port 1/0/1
dn-set local
quit
# Configure User C.
international-toll
pots enterprise hw
port 1/0/0
dn-set local
quit
# Configure User D.
international-toll
pots enterprise hw
port 1/0/1
dn-set local
quit

# Configure trunks, trunk groups, and routes on RouterA.
l Configure a SIP trunk group.
[RouterA-voice] trunk-group sipip sip no-register
[RouterA-voice-trunkgroup-sipip] enterprise hw dn-set local
[RouterA-voice-trunkgroup-sipip] call-right in international-toll out
international-toll
[RouterA-voice-trunkgroup-sipip] signalling-address ip 192.168.1.1 port 5070
[RouterA-voice-trunkgroup-sipip] media-ip 192.168.1.1
[RouterA-voice-trunkgroup-sipip] peer-address static 192.168.1.2 5070
[RouterA-voice-trunkgroup-sipip] register-uri huawei.com
[RouterA-voice-trunkgroup-sipip] home-domain huawei.com
[RouterA-voice-trunkgroup-sipip] reset
[RouterA-voice-trunkgroup-sipip] quit

[RouterA-voice-trunkgroup-at0] call-right in international-toll out
international-toll

[RouterA-voice-callprefix-20000] callroute trunkgroup1 sipip
# Configure trunks, trunk groups, and routes on RouterB.

l Configure a SIP trunk group.
[RouterB-voice] trunk-group sipip sip no-register
[RouterB-voice-trunkgroup-sipip] enterprise hw dn-set local
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[RouterB-voice-trunkgroup-sipip] call-right in international-toll out

international-toll
[RouterB-voice-trunkgroup-sipip] signalling-address ip 192.168.1.2 port 5070
[RouterB-voice-trunkgroup-sipip] media-ip 192.168.1.2
[RouterB-voice-trunkgroup-sipip] peer-address static 192.168.1.1 5070
[RouterB-voice-trunkgroup-sipip] register-uri huawei.com
[RouterB-voice-trunkgroup-sipip] home-domain huawei.com
[RouterB-voice-trunkgroup-sipip] reset
[RouterB-voice-trunkgroup-sipip] quit

[RouterB-voice] trunk-group at0 fxo
[RouterB-voice-trunkgroup-at0] enterprise hw dn-set local
[RouterB-voice-trunkgroup-at0] call-right in international-toll out
international-toll
[RouterB-voice-trunkgroup-at0] trunk-at0 1/0/4 default-called-telno 33333000

[RouterB-voice-callprefix-9] callroute trunkgroup1 at0
[RouterB-voice-callprefix-20000] callroute trunkgroup1 sipip

# Configure post-routing number change on RouterA.
[RouterA-voice-afterroute-change-9] trunk-group at0
[RouterA-voice-afterroute-change-9] quit
[RouterA-voice-afterroute-change-20000] trunk-group sipip
# Configure post-routing number change on RouterB.

[RouterB-voice-afterroute-change-9] trunk-group at0
[RouterB-voice] afterroute-change 20000
[RouterB-voice-afterroute-change-20000] trunk-group sipip

1.
User A and User B can talk with each other.
2.
User C and User D can talk with each other.
3.
User A, User B, User C, and User D can call external users.
4.
When dialing the number 56623000, external users can talk with User A and User B.
5.
When dialing the number 28963000, external users can talk with User C and User D.
----End
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Configuration Files
voice
#
dspattribute
#
enterprise hw
dn-set local
#
sipserver
media-ip 192.168.1.1
#
#
#
#
#
trunk-group at0 fxo
#
trunk-group sipip sip no-register
media-ip 192.168.1.1
#
callprefix 9
prefix 9
digit-length 1 15
#
callprefix 2222
prefix 2222
digit-length 8 9
#
callprefix 20000
prefix 20000
digit-length 5 20
callroute trunkgroup1 sipip
#
port 1/0/0
telno 22223000
dn-set local
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#
port 1/0/1
telno 22223001
dn-set local
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 7 1
#
callprefix 20000
trunk-group sipip
caller no-change
called del 7 5
#
# Configuration file of RouterB

voice
#
port fxo 1/0/4
#
enterprise hw
dn-set local
#
sipserver
media-ip 192.168.1.2
#
#
#
#
#
trunk-group at0 fxo
#
trunk-group sipip sip no-register
media-ip 192.168.1.2
#
callprefix 9
prefix 9
digit-length 1 15
#
callprefix 3333
prefix 3333
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#
callprefix 20000
prefix 20000
callroute trunkgroup1 sipip
#
port 1/0/0
telno 33333000
dn-set local
#
port 1/0/1
telno 33333001
dn-set local
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 8 1
#
callprefix
20000
trunk-group at0
caller no-change
called del 8 5
#
return
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222

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Huawei AR2200 Series Enterprise Routers

Configuration Guide - Voice

HUAWEI TECHNOLOGIES CO., LTD.

Copyright Huawei Technologies Co., Ltd. 2012. All rights reserved.

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Huawei Technologies Co., Ltd.

Huawei Industrial Base

Huawei Proprietary and Confidential

Huawei AR2200 Series Enterprise Routers

About This Document

About This Document

Data configuration engineers

Network monitoring engineers

System maintenance engineers

Indicates a hazard with a high level of risk, which if not

Indicates a tip that may help you solve a problem or save

Provides additional information to emphasize or supplement

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About This Document

The keywords of a command line are in boldface.

Command arguments are in italics.

Items (keywords or arguments) in brackets [ ] are optional.

Optional items are grouped in braces and separated by vertical

Optional items are grouped in brackets and separated by vertical

Optional items are grouped in braces and separated by vertical

Optional items are grouped in brackets and separated by vertical

The parameter before the & sign can be repeated 1 to n times.

A line starting with the # sign is comments.

Interface Numbering Conventions

Changes in Issue 01 (2012-04-20)

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2.1 PBX Overview..................................................................................................................................................68

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2.14.10 Configuring the RBT Service............................................................................................................155

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About This Chapter

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1.1 SIP AG Overview

1.2 SIP AG Features Supported by the AR2200

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Using a SIP AG as a Voice Gateway

Figure 1-1 SIP AG as a voice gateway

SIP AG call flow

The DSP module of the SIP AG generates a dial tone.

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Voice Services Supported by the SIP AG

Basic voice service

The basic voice service is the basic call connection

The third-party service allows a calling party or called

When user A is talking with user B over the phone and

The message waiting indicator (MWI) service allows a

The call transfer service allows the called party to

The call conference service allows more than three

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