NetEngine AR V300R019 CLI-based Configuration Guide - Voice

NetEngine AR
V300R019
CLI-based Configuration Guide -

Voice
Issue 07
Date 2021-12-01
HUAWEI TECHNOLOGIES CO., LTD.

Copyright © Huawei Technologies Co., Ltd. 2021. 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 a 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: https://e.huawei.com
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CLI-based Configuration Guide - Voice Contents
Contents
1 About This Document.............................................................................................................1

2 PBX Configuration................................................................................................................... 5
2.1 Overview of PBX...................................................................................................................................................................... 6
2.1.1 Voice Communication Function...................................................................................................................................... 7
2.1.2 Concepts Related to PBX................................................................................................................................................... 8
2.1.3 PBX Call Processes............................................................................................................................................................... 9
2.2 Configuration Process for PBX.......................................................................................................................................... 14
2.3 Preparing for the PBX Configuration..............................................................................................................................15
2.4 Licensing Requirements and Limitations for PBX...................................................................................................... 16
2.5 Configuring Basic PBX Information................................................................................................................................ 18
2.5.1 Configuring a Device to Work in PBX Mode............................................................................................................ 18
2.5.2 Configuring Media and Signaling IP Address Pools.............................................................................................. 19
2.5.3 Configuring a Country Code and an Area Code..................................................................................................... 22
2.5.4 Configuring an Enterprise and a DN Set................................................................................................................... 27
2.6 Configuring a SIP Server..................................................................................................................................................... 29
2.7 Configuring Toll Fraud Prevention.................................................................................................................................. 33
2.7.1 Context for Configuring Toll Fraud Prevention....................................................................................................... 33
2.7.2 Preventing Theft of the Administrator Right........................................................................................................... 37
2.7.3 Preventing Calls from Being Transferred Through a Trunk and the IVR........................................................ 38
2.7.4 Preventing Registration or Login of Unauthorized Users.................................................................................... 41
2.7.5 Preventing Unauthorized Call Forwarding................................................................................................................42
2.7.6 Deploying the CDR Server.............................................................................................................................................. 43
2.7.7 Verifying the Toll Fraud Prevention Configuration................................................................................................ 43
2.7.8 Troubleshooting Toll Fraud Prevention...................................................................................................................... 43
2.8 Configuring a PBX User...................................................................................................................................................... 44
2.8.1 Configuring a POTS User................................................................................................................................................ 44
2.8.2 Configuring a SIP User.....................................................................................................................................................47
2.8.3 Configuring an ISDN User.............................................................................................................................................. 50
2.9 Configuring a Call Prefix.................................................................................................................................................... 55
2.10 Configuring Number Change......................................................................................................................................... 59
2.10.1 Calling Number Discrimination..................................................................................................................................59
2.10.2 Configuring Long and Short Codes...........................................................................................................................62
2.10.3 Configuring Number Mapping................................................................................................................................... 64
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2.10.4 Configuring a Pre-routing Number Change Plan................................................................................................ 67

2.10.5 Configuring a Post-routing Number Change Plan.............................................................................................. 69
2.10.6 Configuring GNR Number Negotiation................................................................................................................... 72
2.11 Configuring Trunk Groups............................................................................................................................................... 76
2.11.1 Configuring an AT0 Trunk Group.............................................................................................................................. 76
2.11.2 Configuring SIP Trunk Groups.................................................................................................................................... 80
2.11.2.1 Configuring a SIP AT0 Trunk Group...................................................................................................................... 80
2.11.2.2 Configuring a SIP PRA Trunk Group...................................................................................................................... 89
2.11.2.3 Configuring a SIP IP Trunk Group.......................................................................................................................... 96
2.11.3 Configuring an H.323 Trunk Group........................................................................................................................ 101
2.11.4 Configuring a BRA Trunk Group.............................................................................................................................. 113
2.11.5 Configuring a PRA Trunk Group.............................................................................................................................. 117
2.11.6 Configuring an R2 Trunk Group.............................................................................................................................. 124
2.12 Configuring a Call Route............................................................................................................................................... 133
2.13 Configuring the Default Call Route........................................................................................................................... 135
2.14 Configuring Services........................................................................................................................................................137
2.14.1 Configuring the Fax Service...................................................................................................................................... 137
2.14.2 Configuring Voice Mailbox Services....................................................................................................................... 146
2.14.2.1 Context for Configuring Voice Mailbox Services............................................................................................ 146
2.14.2.2 Preparing for the Voice Mailbox Configuration.............................................................................................. 146
2.14.2.3 Configuring the CFU Voice Mailbox Service.................................................................................................... 146
2.14.2.4 Configuring the CFNR Voice Mailbox Service..................................................................................................151
2.14.2.5 Configuring the CFB Voice Mailbox Service..................................................................................................... 156
2.14.2.6 Configuring the CFO Voice Mailbox Service.................................................................................................... 161
2.14.2.7 Customizing Greeting Phrases.............................................................................................................................. 166
2.14.3 Configuring Supplementary Telephony Services................................................................................................168
2.14.3.1 Configuring the Number Query Service............................................................................................................ 169
2.14.3.2 Configuring Calling Line Identification Services............................................................................................. 170
2.14.3.2.1 Configuring CLIP.................................................................................................................................................... 170
2.14.3.2.2 Configuring CLIR.................................................................................................................................................... 171
2.14.3.2.3 Enabling CLIR Override........................................................................................................................................ 172
2.14.3.2.4 Temporarily Activating CLIR............................................................................................................................... 173
2.14.3.2.5 Temporarily Deactivating CLIR.......................................................................................................................... 175
2.14.3.3 Configuring Name Identification Presentation Services.............................................................................. 177
2.14.3.3.1 Configuring CONP................................................................................................................................................. 177
2.14.3.3.2 Configuring CNIP................................................................................................................................................... 178
2.14.3.4 Configuring Call Forwarding Services................................................................................................................ 179
2.14.3.4.1 Configuring CFU..................................................................................................................................................... 180
2.14.3.4.2 Configuring CFNR.................................................................................................................................................. 182
2.14.3.4.3 Configuring CFB..................................................................................................................................................... 184
2.14.3.4.4 Configuring CFO..................................................................................................................................................... 187
2.14.3.4.5 Configuring CFC..................................................................................................................................................... 189
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2.14.3.5 Configuring Call Transfer........................................................................................................................................ 190

2.14.3.6 Configuring Call Hold.............................................................................................................................................. 192
2.14.3.7 Configuring Call Waiting........................................................................................................................................ 193
2.14.3.8 Configuring Call Barring Services........................................................................................................................ 197
2.14.3.8.1 Configuring Number Barring............................................................................................................................. 197
2.14.3.8.2 Configuring Call Barring...................................................................................................................................... 200
2.14.3.8.3 Configuring Password Call.................................................................................................................................. 203
2.14.3.8.4 Configuring the Blacklist and Whitelist......................................................................................................... 207
2.14.3.8.5 Configuring RAC..................................................................................................................................................... 210
2.14.3.8.6 Configuring DND................................................................................................................................................... 214
2.14.3.8.7 Configuring SCA..................................................................................................................................................... 217
2.14.3.8.8 Configuring SCR..................................................................................................................................................... 222
2.14.3.9 Configuring Call Back Services............................................................................................................................. 227
2.14.3.9.1 Configuring CBB..................................................................................................................................................... 227
2.14.3.9.2 Configuring ACB..................................................................................................................................................... 229
2.14.3.10 Configuring Pickup Services................................................................................................................................ 232
2.14.3.10.1 Configuring Co-group Pickup.......................................................................................................................... 232
2.14.3.10.2 Configuring Designated Pickup...................................................................................................................... 234
2.14.3.11 Configuring Secretary Services........................................................................................................................... 235
2.14.3.11.1 Configuring the Secretary Service................................................................................................................. 235
2.14.3.11.2 Configuring the Advanced Secretary Service............................................................................................. 239
2.14.3.12 Configuring Hotline Services...............................................................................................................................243
2.14.3.12.1 Configuring Delay Hotline................................................................................................................................243
2.14.3.12.2 Configuring Instant Hotline............................................................................................................................. 246
2.14.3.13 Configuring Ringing Services.............................................................................................................................. 248
2.14.3.13.1 Configuring Simultaneous Ringing............................................................................................................... 248
2.14.3.13.2 Configuring Sequential Ringing..................................................................................................................... 250
2.14.3.14 Configuring Conference Call Services.............................................................................................................. 252
2.14.3.14.1 Configuring Individuals Dialing-In.................................................................................................................252
2.14.3.14.2 Configuring System Convening...................................................................................................................... 256
2.14.3.14.3 Configuring Host Convening........................................................................................................................... 259
2.14.3.14.4 Configuring Instant Conference..................................................................................................................... 262
2.14.3.15 Configuring Enhanced Services.......................................................................................................................... 263
2.14.3.15.1 Configuring Abbreviated Dialing................................................................................................................... 263
2.14.3.15.2 Configuring ONLY................................................................................................................................................266
2.14.3.15.3 Configuring Three-Party Calls......................................................................................................................... 269
2.14.3.15.4 Configuring Enterprise CRBT........................................................................................................................... 273
2.14.3.15.5 Configuring the Wakeup Service....................................................................................................................274
2.14.3.15.6 Configuring the PBX Line Selection Service............................................................................................... 277
2.14.3.15.7 Deregistering All Services................................................................................................................................. 282
2.14.3.15.8 Configuring the MWI Service.......................................................................................................................... 283
2.14.3.15.9 Configuring the Busy Lamp Field Service................................................................................................... 285
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2.14.4 Configuring the Service Password.......................................................................................................................... 287

2.15 Advanced Configurations.............................................................................................................................................. 289
2.15.1 Creating and Switching a Phone System Announcement.............................................................................. 289
2.15.2 Configuring High-level Users to Preempt Trunks to Implement Preferred Calls....................................291
2.15.3 Configuring Preemption for High-Priority Users............................................................................................... 294
2.15.4 Configuring a Call Limiting Policy.......................................................................................................................... 296
2.15.4.1 Configuring Call Limiting Over a Trunk............................................................................................................ 296
2.15.4.2 Configuring 32-Level Right Call Barring........................................................................................................... 298
2.15.4.3 Configuring Call Limiting Upon CPU Overload.............................................................................................. 304
2.15.4.4 Configuring Static Bandwidth Control for Call Barring................................................................................305
2.15.4.5 Configuring Call Limiting When the CDR Pool Is Full.................................................................................. 311
2.15.4.6 Configuring Reanalysis Prefix Call Restriction.................................................................................................313
2.15.4.7 Configuring Call Limiting Based on the Remote IP Address...................................................................... 315
2.15.5 Configuring Intelligent Routing............................................................................................................................... 316
2.15.5.1 Configuring Re-routing Upon a Failure............................................................................................................. 317
2.15.5.2 Configuring a Call Route Plan Based on Charge Rates................................................................................319
2.15.5.3 Configuring a Call Route Plan Based on Load Balancing........................................................................... 322
2.15.5.4 Configuring a Call Route Plan Based on the Call Percentage................................................................... 324
2.15.5.5 Configuring a Call Route Plan Based on User Rights................................................................................... 327
2.15.5.6 Configuring a Call Route Plan Based on Link Balancing............................................................................. 331
2.15.5.7 Routing Base on Time Segment and Calling Number................................................................................. 333
2.15.6 Configuring Connection Data for the CDR Server.............................................................................................336
2.15.7 Configuring PRA Trunk Timeslot Binding............................................................................................................. 340
2.15.8 Configuring Local Survival......................................................................................................................................... 342
2.15.9 Setting SIP Stack Parameters................................................................................................................................... 347
2.15.10 Setting DSP Parameters........................................................................................................................................... 349
2.15.11 Setting Parameters for Physical Voice Interfaces............................................................................................ 357
2.15.12 Configuring a mapping of a built-in tone......................................................................................................... 363
2.16 Configuring the Signaling Proxy and Media Proxy...............................................................................................364
2.16.1 Context for Configuring the Signaling Proxy and Media Proxy.................................................................... 364
2.16.2 Configuring the Signaling Proxy and Media Proxy on a Public Network..................................................369
2.16.3 Configuring the Signaling Proxy and Media Proxy on the Private/Public Network Boundary.......... 373
2.17 Configuring SIP PRA Agent Registration and SIP AT0 Agent Registration...................................................382
2.17.1 Configuring SIP PRA Agent Registration.............................................................................................................. 382
2.17.2 Configuring SIP AT0 Agent Registration...............................................................................................................395
2.18 Follow-up Procedure for PBX....................................................................................................................................... 406
2.19 Maintaining PBX............................................................................................................................................................... 407
2.19.1 Checking Call Information......................................................................................................................................... 407
2.19.2 Checking Call Logs....................................................................................................................................................... 408
2.20 Configuration Examples for PBX................................................................................................................................. 408
2.20.1 Example for Configuring Voice Services for Small- and Medium-sized Enterprises..............................408
2.20.2 Example for Configuring Distributed Networking............................................................................................ 414
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2.20.3 Example for Expanding the Capacity of the Live-Network PBX................................................................... 423
2.20.4 Example for Configuring PBX Sharing for Different Enterprises.................................................................. 433
2.20.5 Example for Configuring an AR as a Branch Gateway to Access UC......................................................... 440
3 SIP AG Configuration......................................................................................................... 450

3.1 Overview of the SIP AG.................................................................................................................................................... 450
3.2 Understanding the SIP AG...............................................................................................................................................452
3.2.1 SIP AG Registration........................................................................................................................................................ 452
3.2.2 SIP AG Call Flow.............................................................................................................................................................. 454
3.3 SIP AG Features Supported by the Device................................................................................................................. 455
3.4 Application Scenarios for the SIP AG...........................................................................................................................458
3.5 Licensing Requirements and Limitations for the SIP AG...................................................................................... 460
3.6 Setting Voice Parameters on a SIP AG........................................................................................................................ 462
3.6.1 Configuring the Device to Work in SIP AG Mode................................................................................................ 462
3.6.2 Configuring Media and Signaling IP Address Pools............................................................................................ 463
3.7 Configuring a SIP AG Interface...................................................................................................................................... 464
3.8 Configuring a SIP AG User.............................................................................................................................................. 474
3.8.1 Setting Parameters for a SIP AG User......................................................................................................................475
3.8.2 (Optional) Configuring a SIP AG User Group....................................................................................................... 477
3.8.3 (Optional) Configuring Functions or Services in a SIP Service Data Profile...............................................478
3.8.4 Verifying the SIP AG User Configuration................................................................................................................ 480
3.9 Maintaining a SIP AG........................................................................................................................................................ 481
3.9.1 Clearing SIP AG Statistics............................................................................................................................................. 481
3.10 Advanced Configurations.............................................................................................................................................. 481
3.10.1 (Optional) Setting Parameters for an FXS Interface........................................................................................ 481
3.10.2 Setting Parameters for a BRA Interface................................................................................................................ 483
3.10.3 Setting Parameters for a VE1 Interface................................................................................................................ 484
3.10.4 (Optional) Setting System Parameters................................................................................................................. 485
3.10.4.1 Setting Upper and Lower Thresholds for Hookflash Pressing Parameters........................................... 486
3.10.4.2 Setting the MWI Mode............................................................................................................................................ 487
3.10.4.3 Configuring the Codec Mode for Voice Services............................................................................................ 487
3.10.4.4 Setting Ringing Parameters................................................................................................................................... 488
3.10.4.5 Enabling the Function That Reduces the Feed on Locked Ports...............................................................489
3.10.4.6 Setting CLIP Parameters in Offhook State....................................................................................................... 489
3.10.4.7 Setting CLIP Parameters in Onhook State........................................................................................................ 490
3.10.4.8 Setting the Number of SIP Register Messages Sent Every Second.......................................................... 491
3.10.4.9 Loading the Voice Prompt File and Voice Service Logic File...................................................................... 491
3.10.4.10 Configuring a Media Port.................................................................................................................................... 492
3.10.4.11 Verifying the System Configuration................................................................................................................. 493
3.10.5 (Optional) Setting SIP Stack Parameters............................................................................................................. 493
3.10.6 (Optional) Setting DSP Parameters....................................................................................................................... 495
3.10.6.1 Enabling Echo Cancellation on a DSP Channel.............................................................................................. 495
3.10.6.2 Enabling PLC on a DSP Channel.......................................................................................................................... 496
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3.10.6.3 Enabling Silence Compression on a DSP Channel......................................................................................... 497

3.10.6.4 Setting Fax Parameters........................................................................................................................................... 497
3.10.6.5 Setting Jitter Buffer Parameters........................................................................................................................... 498
3.10.6.6 Setting the Payload Type Value........................................................................................................................... 499
3.10.6.7 Setting RTCP Parameters........................................................................................................................................ 500
3.10.6.8 Setting DSP Resource Parameters....................................................................................................................... 501
3.10.6.9 Configuring a DSP Channel................................................................................................................................... 502
3.10.6.10 Verifying the DSP Configuration....................................................................................................................... 503
4 H.248 AG Configuration.................................................................................................... 504

4.1 Overview of the H.248 AG.............................................................................................................................................. 504
4.2 Understanding the H.248 AG......................................................................................................................................... 506
4.2.1 H.248 Protocol Mechanism..........................................................................................................................................506
4.2.2 H.248 AG Call Flow........................................................................................................................................................ 510
4.3 H.248 Features Supported by the Device................................................................................................................... 512
4.4 Application Scenarios for the H.248 AG..................................................................................................................... 515
4.5 Licensing Requirements and Limitations for the H.248 AG.................................................................................516
4.6 Setting Voice Parameters on an H.248 AG................................................................................................................ 517
4.6.1 Configuring the Device to Work in H.248 AG Mode.......................................................................................... 517
4.6.2 Configuring Media and Signaling IP Address Pools............................................................................................ 518
4.7 Configuring an H.248 AG Interface.............................................................................................................................. 519
4.8 Configuring an H.248 AG User...................................................................................................................................... 524
4.8.1 Configuring a POTS User.............................................................................................................................................. 524
4.8.2 Configuring an ISDN User............................................................................................................................................ 527
4.9 Advanced Configurations................................................................................................................................................. 528
4.9.1 (Optional) Setting Parameters for an FXS Interface...........................................................................................528
4.9.2 Setting Parameters for a BRA Interface.................................................................................................................. 529
4.9.3 Setting Parameters for a VE1 Interface................................................................................................................... 531
4.9.4 (Optional) Setting System Parameters....................................................................................................................532
4.9.4.1 Setting Upper and Lower Thresholds for Hookflash Pressing Parameters.............................................. 533
4.9.4.2 Setting the MWI Mode.............................................................................................................................................. 534
4.9.4.3 Configuring the Codec Mode for Voice Services............................................................................................... 534
4.9.4.4 Setting Ringing Parameters..................................................................................................................................... 535
4.9.4.5 Enabling the Function That Reduces the Feed on Locked Ports................................................................. 536
4.9.4.6 Setting CLIP Parameters in Offhook State.......................................................................................................... 536
4.9.4.7 Setting CLIP Parameters in Onhook State.......................................................................................................... 537
4.9.4.8 Loading the Voice Prompt File and Voice Service Logic File........................................................................ 538
4.9.4.9 Configuring a Media Port......................................................................................................................................... 539
4.9.4.10 Verifying the System Configuration.................................................................................................................... 539
4.9.5 (Optional) Setting DSP Parameters.......................................................................................................................... 540
4.9.5.1 Enabling Echo Cancellation on a DSP Channel.................................................................................................540
4.9.5.2 Enabling PLC on a DSP Channel............................................................................................................................. 541
4.9.5.3 Enabling Silence Compression on a DSP Channel............................................................................................541
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4.9.5.4 Setting Fax Parameters.............................................................................................................................................. 542

4.9.5.5 Setting Jitter Buffer Parameters............................................................................................................................. 543
4.9.5.6 Setting the Payload Type Value.............................................................................................................................. 544
4.9.5.7 Setting RTCP Parameters.......................................................................................................................................... 545
4.9.5.8 Setting DSP Resource Parameters..........................................................................................................................546
4.9.5.9 Configuring a DSP Channel......................................................................................................................................547
4.9.5.10 Verifying the DSP Configuration.......................................................................................................................... 547
4.9.6 (Optional) H.248 Common Configuration............................................................................................................. 548
4.9.6.1 Setting RFC 2198 Parameters.................................................................................................................................. 548
4.9.6.2 Setting FAX/Modem Parameters............................................................................................................................ 548
4.9.6.3 Configuring a TID Profile.......................................................................................................................................... 549
4.9.6.4 Configuring a Local Digitmap................................................................................................................................. 550
4.9.7 Configuration Examples for the H.248 AG............................................................................................................. 550
4.9.7.1 Example for Configuring the Voice Service........................................................................................................ 550
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CLI-based Configuration Guide - Voice 1 About This Document
1 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:
● 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
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 level of risk which, if

not avoided, could result in death or serious injury.
Indicates a hazard with a 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 deterioration, or unanticipated results.
NOTICE is used to address practices not related to
personal injury.
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Symbol Description
Supplements the important information in the main text.

NOTE is used to address information not related to
personal injury, equipment damage, and environment
deterioration.
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.
Security Conventions
● Password setting
– When configuring a password, the cipher text is recommended. To ensure
device security, change the password periodically.

NetEngine AR
– When you configure a password in plain text that starts and ends with
%@%@, @%@%, %#%#, or %^%# (the password can be decrypted by
the device), the password is displayed in the same manner as the
configured one in the configuration file. Do not use this setting.
– When you configure a password in cipher text, different features cannot
use the same cipher-text password. For example, the cipher-text password
set for the AAA feature cannot be used for other features.
● Encryption algorithm
Currently, the device uses the following encryption algorithms: 3DES, AES,
RSA, SHA1, SHA2, and MD5. 3DES, RSA and AES are reversible, while SHA1,
SHA2, and MD5 are irreversible. The encryption algorithms DES, 3DES, RSA
(RSA-1024 or lower), MD5 (in digital signature scenarios and password
encryption), and SHA1 (in digital signature scenarios) have a low security,
which may bring security risks. If protocols allowed, using more secure
encryption algorithms, such as AES, RSA (RSA-2048 or higher), SHA2, and
HMAC-SHA2, is recommended. The encryption algorithm depends on actual
networking. The irreversible encryption algorithm must be used for the
administrator password, SHA2 is recommended.
● Personal data
Some personal data (such as MAC or IP addresses of terminals) may be
obtained or used during operation or fault location of your purchased
products, services, features, so you have an obligation to make privacy policies
and take measures according to the applicable law of the country to protect
personal data.
● The terms mirrored port, port mirroring, traffic mirroring, and mirroring in this
manual are mentioned only to describe the product's function of
communication error or failure detection, and do not involve collection or
processing of any personal information or communication data of users.
Reference Standards and Protocols

To obtain reference standards and protocols, log in to Huawei official website,
search for "protocol compliance list", and download the Huawei AR Series
Standard and Protocol Comply Table.
Declaration
● This manual is only a reference for you to configure your devices. The
contents in the manual, such as web pages, command line syntax, and
command outputs, are based on the device conditions in the lab. The manual
provides instructions for general scenarios, but do not cover all usage
scenarios of all product models. The contents in the manual may be different
from your actual device situations due to the differences in software versions,
models, and configuration files. The manual will not list every possible
difference. You should configure your devices according to actual situations.
● The specifications provided in this manual are tested in lab environment (for
example, the tested device has been installed with a certain type of boards or
only one protocol is run on the device). Results may differ from the listed
specifications when you attempt to obtain the maximum values with multiple
functions enabled on the device.

NetEngine AR
● In this document, public IP addresses may be used in feature introduction and

configuration examples and are for reference only unless otherwise specified.
● In this document, NetEngine access routers include
AR600&AR1600&AR6000&AR6000-S Series.
Mappings Between Product Software Versions and NMS Versions

The mappings between product software versions and NMS versions are as
follows.
AR Product eSight iManager U2000 NCE-IP

Software Version
V300R019C00 V300R010C00 V200R018C60 -
V300R019C10 V300R010C00 - V100R019C10
V300R019C11 V300R010C00 - V100R020C00
V300R019C13 V300R010C00 - V100R020C10

NetEngine AR
CLI-based Configuration Guide - Voice 2 PBX Configuration
2 PBX Configuration
2.1 Overview of PBX

2.2 Configuration Process for PBX
This section describes the process for configuring voice service data on the PBX. In
the configuration process, optional tasks and subtasks are performed according to
your site requirements.
2.3 Preparing for the PBX Configuration
Before configuring the PBX, verify the hardware deployment and loaded license. In
addition, plan required data for subsequent configuration tasks.
2.4 Licensing Requirements and Limitations for PBX
2.5 Configuring Basic PBX Information
This section describes how to configure basic information so that the device can
work in PBX mode. Such data is used for subsequent service configuration.
2.6 Configuring a SIP Server
This section describes how to configure parameters of a SIP server, including the IP
address, URI, and home domain. SIP server is an important entity in the SIP
protocol architecture. The PBX can function as the SIP server to accept registration
information of SIP users, save the information in the address information
database, and manage and maintain users' registration information.
2.7 Configuring Toll Fraud Prevention
You can configure limiting conditions on the PBX to prevent extra expenses due to
call fraud.
2.8 Configuring a PBX User
You can configure different types of PBX users according to actual networking and
user planning. Terminal users on the PBX contain POTS, SIP, and ISDN users.
2.9 Configuring a Call Prefix
After intra-office and inter-office prefixes are configured on the PBX, intra-office
users can call each other, and intra-office users can call inter-office users.
2.10 Configuring Number Change
This section describes how to configure calling/called number change, long and
short codes, and number mapping.

NetEngine AR
2.11 Configuring Trunk Groups

A trunk group is used to transmit signaling and media data between central office,
and is used to constitute a route. In TDM scenarios, a trunk group is a set of trunk
circuits with the same attributes. Generally, a group of trunk circuits with the same
attributes in the same direction are called a trunk group.
2.12 Configuring a Call Route
A call route defines routing rules for outgoing calls, and is bound to a call prefix
and a trunk group. After a call route is configured, the PBX can correctly route
outgoing calls.
2.13 Configuring the Default Call Route
The default call route defines routing rules for inter-office calls, and is bound to a
trunk group. A call for which no prefix can be matched is routed out through the
default call route.
2.14 Configuring Services
2.15 Advanced Configurations
This section describes special topics including number change, call limiting, and
intelligent routing.
2.16 Configuring the Signaling Proxy and Media Proxy
The PBX media proxy and signaling proxy implement address translation between
network segments and enable signaling and media streams to traverse NAT.
2.17 Configuring SIP PRA Agent Registration and SIP AT0 Agent Registration
2.18 Follow-up Procedure for PBX
After completing all the configurations, provide user information, server IP
addresses, and documents to the customer.
2.19 Maintaining PBX
2.20 Configuration Examples for PBX
2.1 Overview of PBX

The voice feature of the AR router provides the IP-PBX service based on the RFC
standard. The signaling is transmitted using SIP, and the voice media is transmitted
using RTP. In this version, SIP TLS and SRTP are not supported; therefore, VoIP
signaling and voice media in the IP-PBX service are transmitted in plaintext over
the network.
The voice feature of the AR router provides PBX services, including call barring,
password-based calling, secretary service, and deactivation of all services. These
services involve transmission of service passwords in plaintext using SIP between
VoIP terminals and the AR router. Therefore, ensure that these services are used in
trusted networks. Note that the service passwords here are bound to specific VoIP
terminals.
The voice feature of the AR router provides voice mailbox and outgoing call
barring services. These two services involve transmission of service passwords in
plaintext using RTP between VoIP terminals and the AR router. Therefore, ensure
that these services are used in trusted networks. Note that the service passwords
here are bound to specific VoIP terminals.

NetEngine AR
The voice feature of the AR router provides the voice conference services. This
service involves transmission of conference access passwords in plaintext using
RTP between VoIP terminals and the AR router. Therefore, ensure that these
services are used in trusted networks. Note that the conference access passwords
here are valid within the specific conference time.
If the above-mentioned services are used in non-trusted networks, encrypted IPSec
channels can be set up between the AR router on the terminal side and the AR
router that provides such services to ensure password transmission security. For
details, see CLI-based Configuration > Configuration Guide - VPN.
Figure 2-1 Encrypting transmission channels using VPN
AR PBX
IPSec
LAN
IP
Network
AR AG
Encrypted channels are set up using IPSec between the AR router deployed in the
terminal area and the AR router that provides voice PBX services in the service
area. Then, passwords involved in the above-mentioned services are transmitted
over the encrypted IPSec channels.
2.1.1 Voice Communication Function
Basic Concept of PBX

A traditional PBX is a program control voice exchange used for voice
communication between enterprise users and between enterprise users and PSTN
users. However, a traditional PBX cannot well support computer telephony
integration (CTI) and voice over Internet Protocol (VoIP), and is not open enough
because of the dedicated technology employed.
With the development of Internet and VoIP technologies, the IP PBX is introduced.
IP PBXs integrate voice communication into enterprise data networks to provide
services for both traditional POTS users and VoIP users. IP PBXs are able to provide
a unified voice and data access network for enterprise offices and employees
distributed all over the world, greatly enhancing their working efficiency and
saving their communication costs.

NetEngine AR
2.1.2 Concepts Related to PBX
Intra-office
A logical PBX is an exchange office. When the calling and called parties are in the
same exchange office, the call is an intra-office call.
Inter-office
When an exchange office communicates with another exchange office through a
trunk line, the call is an inter-office call.
Trunk
A trunk is a logical link between two exchange offices. Inter-office calls must be
transmitted by trunks. To set up a connection channel with a remote office, a
trunk is bound to the signaling and media addresses of the local office and
signaling address of the remote office.
Trunk Group
A trunk group is a set of trunks for inter-office calls with the same attributes.
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 identifies the service attribute (basic service or supplementary service;
intra-office call, national long-distance call, or international long-distance 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 the 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, a
correct call prefix configuration is the key to the call service. A call must match at
least one call prefix.
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.
Enterprise
A PBX system is divided into multiple virtual PBX systems based on enterprise
names. Users in an enterprise can directly call each other, while users in different
enterprises cannot directly call each other.
DN set
A DN set is also called a dial plan. DN sets divide a physical network or a device
into multiple logical networks.

NetEngine AR
Short Number and Long Number

A short number is an internal number of an enterprise, and a long number is
allocated by the carrier to an enterprise. Internal users of an enterprise can dial
the short numbers of each other. External users need to dial the long number of
the enterprise.
2.1.3 PBX Call Processes
Call Flowchart
The calling party can be a trunk user, intra-office user, or moderator of a
conference system or IVR process. The following information is required for
initiating a call: enterprise, DN set, calling information (includes number, call
source, and rights), called number, and so on. The PBX sends the call information
to the number analysis module.
Figure 2-2 shows the call routing process and Figure 2-3 shows the pre-routing
number change process.

NetEngine AR
Figure 2-2 Call processes
Intra-office user
Conference/IVR call Trunk intra-office call
initiates a call
No Rejected by Yes
Obtain calling right whitelist/blacklist?
according to service user Country code/
and conference/IVR rights Release
area code
call
regulation
Information carried in a calling
number: such as enterprise, DN set,
calling information (includes
number, call source, and rights), and Calling number
called number discrimination?
No
Yes
Calling number change
Prefix analysis
Does
No
calling party right meet the Release
prefix right requirement? call
Yes
Long-/short-number change
Number mapping
Pre-routing number change
Yes Number reanalysis and

number of reanalysis times
<5?
No
Inter- Intra- New/supplemental Conference/
office call office call service IVR call
Intra-office user
Select route Yes Trunk user? No Conference/
location
IVR
Post-routing
Selective call Yes Release Meet service
number change Yes No
rejection? call right and status
requirement?
Rejected by Yes Release No
blacklist/whitelist? call Service is Service
No successful fails
Inter-office call Intra-office call

NetEngine AR
Figure 2-3 Pre-routing number change
Start
Prefix analysis is
successful
Calling number changes according
Reanalyze called number. The calling

Reanalyze the new called number
to prefix configuration
number does not change

Calling short-/long- N Calling number N Pre-routing calling N
number change? mapping? number change?
Y Y Y
Calling number changes Calling number Calling number

from short- number to changes from changes according to
long- number InnNum to ExtNum rule
Called number
changes from Y Called number Called number
ExtNum to mapping? changes according
InnNum to rule
N
Pre-routing called N
number change?
Y
Called number changes
according to rule
Prefix reanalysis?
N
End
Call Process Description

1. If a trunk user calls an intra-office user, the PBX checks whether the user has
the right to call an intra-office user against the blacklist and whitelist. If the
user does not have the right, the PBX releases the call; otherwise, the PBX
regulates the country code and area code for the call.
2. When receiving the call information, the PBX matches the calling number
against the calling number discrimination list. If the calling number does not
match the list, the PBX does not discriminate the calling number; otherwise,
the PBX checks whether the calling number has the calling number
discrimination right. If so, the PBX changes the calling and called information;
if not, the PBX releases the call.
3. The PBX then analyzes the called number prefix, and determines whether the
calling party has the right to dial the called number. If the calling party has
the right, the PBX performs pre-routing number change; otherwise, the PBX
releases the call. If the changed number needs to be reanalyzed, the PBX
sends the call to the prefix analysis module again.
4. After analyzing the prefix, the PBX perform the following operations:

NetEngine AR
a. For an inter-office outgoing prefix, the PBX finds the corresponding trunk
according to the route bound to the prefix, performs post-routing number
change, and then checks whether the calling party has the inter-office
outgoing call against the blacklist and whitelist. If the calling party has
the right, the call is connected; otherwise, the call is released.
b. For an intra-office prefix, the PBX determines whether to accept or reject
the call according to the call acceptation/rejection conditions configured
by the called user. If the called user does not configure the call
acceptation/rejection conditions, the PBX directly calls the intra-office
user.
c. For a new/supplementary service prefix, if the calling party is a trunk
user, the call is released; if the calling party is not a trunk user, the PBX
checks whether the calling party meets the service right and status
requirements. If so, the call is connected; otherwise, the call fails.
d. For a conference access prefix or IVR, the PBX connects the calling party
to the conference or completes the IVR call.
Table 2-1 Call process description

Process Description How to Configure
ing
Metho
d
Blacklis When receiving an inter-office For the concept and

t and incoming call, the PBX checks the configuration method, see
whitelis calling number against the 2.14.3.8.4 Configuring the
t blacklist and whitelist. The calls Blacklist and Whitelist.
rejected by blacklist or whitelist
will be released.
Numbe Supplement the calling number For the concept and

r according to user customization. configuration method, see 2.5.3
regulati Regulate the called number to Configuring a Country Code
on quickly and accurately locate the and an Area Code.
called user.
Calling Before analyzing the called number For the configuration method,
numbe prefix, the PBX changes the see 2.10.1 Calling Number
r number according to the Discrimination.
discrimi enterprise, DN set, and number of
nation the calling party. When the calling
party initiates an inter-office
outgoing call, the new number is
used.

NetEngine AR

ing
Metho
d
Prefix Prefixes include basic call prefixes, For the configuration method,
analysi new service management prefixes, see 2.9 Configuring a Call
s new service triggering prefixes, and Prefix.
IVR prefixes. The rights of a basic
call prefix include Inter, Local,
National-toll, and International-
toll. The PBX handles the prefix
dialed by the calling party only
when the calling party has the
rights of the prefix.
Long-/ The long number is allocated by a For the concept and

short- carrier to short-number users. configuration method, see
numbe Short-number users can call each 2.10.2 Configuring Long and
r other by dialing the short numbers. Short Codes.
change The inter-office users need to dial
the long number to call intra-office
users, and the long number is
displayed when the intra-office
users call inter-office users.
Numbe Number mapping can be For the concept and

r configured for the users with no configuration method, see
mappin long numbers configured. When 2.10.3 Configuring Number
g receiving an inter-office incoming Mapping.
call, the PBX maps the external line
number to the private line number
and analyzes the number. The PBX
then locates the called party
according to the obtained prefix.
Pre- This function allows the PBX to For the concept and
routing delete, modify, and insert calling configuration method, see
numbe and called numbers. You can 2.10.4 Configuring a Pre-
r configure whether to perform routing Number Change Plan.
change reanalysis and play reanalysis dial
tone.
Route The PBX associates the prefixes to For the concept and
selectio routes, and find routes for calls. If a configuration method, see
n route selection policy is configured, 2.15.5 Configuring Intelligent
the PBX selects routes according to Routing.
the policy.
Post- The PBX changes calling and called For the concept and
routing numbers according to prefixes and configuration method, see
numbe trunk groups. 2.10.5 Configuring a Post-
r routing Number Change Plan.
change

NetEngine AR

ing
Metho
d
Selectiv After a call acceptation/rejection For the concept and

e call list is configured, the PBX configuration method, see
accepta determines whether to accept or 2.14.3.8.7 Configuring SCA and
tion/ release the call according to the 2.14.3.8.8 Configuring SCR.
rejectio list.
n
2.2 Configuration Process for PBX

This section describes the process for configuring voice service data on the PBX. In
the configuration process, optional tasks and subtasks are performed according to
your site requirements.
Figure 2-4 illustrates the PBX configuration process.

NetEngine AR
Figure 2-4 PBX configuration process
Start
Prepare for
configuration
Configure basic Working Interface IP Country/Area Enterprise/

information mode address pool code DN set
Configure the SIP

server
Configuring Toll
Fraud Prevention
Configure a PBX
POTS user SIP user ISDN user
user
Configure a call
AT0 SIP AT0 SIP PRA
prefix
Configure a trunk
BRA H.323 SIP IP
group
Configure a call
PRA R2
route
Configure
services
Advanced
configuration Mandatory Mandatory
task subtask
Follow-up
procedure Optional Optional
task subtask
End
NOTICE
Plan and configure the service data by strictly following the information described
in 2.7 Configuring Toll Fraud Prevention. Otherwise, toll fraud (placing
international toll calls or premium-rate telephone numbers bypassing the security
policies) may occur, causing economic losses.
2.3 Preparing for the PBX Configuration

Before configuring the PBX, verify the hardware deployment and loaded license. In
addition, plan required data for subsequent configuration tasks.
Table 2-2 describes the tasks to perform before the configuration.

NetEngine AR
Table 2-2 PBX configuration preparations
Task Description
Verify hardware Ensure that the following requirements are met:

deployment. ● Cards and cables for voice services are available.
For example, equipped with FXS ports and FXS cables for
the applicable device model are required to support POTS
user access.
● Cables are properly routed and connected.
● Cards are installed and powered on.
For more information about hardware deployment, see
Hardware Description, Quick Start Guide, and Quick
Installation Guide.
Verify the Run the display license command to query the license
loaded license. information. Verify that the voice license file has been
activated and that the configuration items are the same as
those specified duration license application.
For details about the voice license description and the
methods of applying for, loading, and activating a license, see
the License Usage Guide.
Plan data. Based on enterprise requirements, data provided by the

network carrier, and product conditions, plan the following
PBX data:
● User numbers and prefixes
● Trunk types and interworking parameters
● Voice service types and service data
2.4 Licensing Requirements and Limitations for PBX
Involved Network Elements

None
Licensing Requirements
For devices that support voice functions (such as PBX, SIP AG, and H.248 AG),
their licensing requirements for the voice functions are as follows:
By default, voice functions cannot be used. To use voice functions, apply for and
purchase the following license from the Huawei local office.
NOTE
This function is not under license control on AR617VW, AR617VW-LTE4, and AR617VW-
LTE4EA.

NetEngine AR
● AR6100 series: AR6100 Value-Added Voice Package

If phones of any type are deployed on an AR router, a CM&BEST license must be

purchased based on the number of deployed phones.
● CM&BEST license-5 phones
Hardware Requirements
Table 2-3 Hardware requirements
Series Feature Support
AR300, AR600, AR700, ● Only the AR6120-VW, AR617VW, AR617VW-

AR1600, and AR6000 series LTE4, and AR617VW-LTE4EA support voice
functions (such as the PBX and SIP AG).
● Only the AR6120-VW supports voice
functions (such as the H.248 AG).
AR6000-S series Not supported
Interface card/SRU Only the SRU-200H support voice functions

(such as PBX, SIP AG, and H.248 AG).
● The 4FXS1FXO, 16FXS, 32FXS, and 4FXS voice cards support POTS users. For
the mapping between the device and voice card, see 4FXS1FXO (4-Port FXS
+ 1-Port FXO Voice Interface Card), 16FXS (16-Port FXS Voice Interface Card),
32FXS (32-Port FXS Voice Interface Card), 4FXS (4-Port FXS Voice Interface
Card) in the NetEngine AR Get to Know the Product-Hardware Description-
Cards-Voice Card.
● The 2BST voice card supports providing voice services to ISDN phone users.
For the mapping between the device and voice card, see 2BST (2-Port ISDN
S/T Voice Interface Card) in the NetEngine AR Get to Know the Product-
Hardware Description-Cards-Voice Card.
● The E1T1-M interface card supports connecting a router to a TDM PBX
through E1 interface. For the mapping between the device and voice card, see
1E1/T1-M (1-Port Channelized E1/T1/PRI/VE1 Multiflex Trunk Interface Card)
in the NetEngine AR Get to Know the Product-Hardware Description-Cards-
E1/T1 Card.
● The 4FXS voice card only supports PBX.
● The 4FXS voice card does not support three-party services.
● The 4FXS voice card of the AR6300, AR6300-S, and AR6300K does not support
Master/Slave Switchover.

NetEngine AR
Feature Limitations
None.
2.5 Configuring Basic PBX Information

This section describes how to configure basic information so that the device can
work in PBX mode. Such data is used for subsequent service configuration.
2.5.1 Configuring a Device to Work in PBX Mode

You can use PBX voice services after configuring the device to work in PBX mode.
Context
When providing voice communication, you can configure the device to work in
PBX, SIP AG, or H.248 AG mode. Before using PBX voice services, configure the
device to work in PBX mode. Before the configuration, run the display voice
service-mode command to query the working mode of the device. If the working
mode is not PBX, you need to run the display current-configuration command to
query the configurations and then delete all configurations in the voice view.
Then, set the working mode of the device to PBX. If the device works in PBX mode,
skip this configuration.
In PBX mode, the device exchanges data between enterprise users and the Public
Switched Telephone Network (PSTN). As the IP PBX, the device can connect to SIP
users and integrate voice communication into enterprise data networks so that an
integrated voice and data network is established to connect offices and employees
around the world.
Prerequisites
The license for PBX voice services has been obtained. For details about the voice
license description and the methods of applying for, loading, and activating a
license, see the License Usage Guide.
Procedure
Step Action Command Description
1 View the working display voice ● If the device is configured

mode of the device. service-mode to work in PBX mode, you
do not need to perform
the following operations.
● If the device is configured
to work in H.248 AG or
SIP AG mode, manually
delete the AG
configuration.
2 Enter the system system-view -

view.

NetEngine AR
3 Enter the voice voice -

view.
4 Configure the service-mode pbx -

device to work in
PBX mode.
5 Return to the user return -

view.
6 Save the save NOTICE

configuration. After the device is configured to
work in PBX mode, restart the
7 Restart the device. reboot device to make the
configuration take effect.
Verification
Action Command Expected Result
View the display voice The command output shows that the
working service-mode device works in PBX mode.
mode of the <Huawei>display voice service-mode
The voice service mode is PBX.
device.
2.5.2 Configuring Media and Signaling IP Address Pools

This section describes how to configure Ethernet IP addresses, media IP address
pools, and signaling IP address pools for interfaces on the PBX, so that the PBX
can connect to the IP network.
Context
An interface is used by a PBX to exchange data with other network devices.
Interfaces on the PBX are numbered in format of slot ID/card ID/interface ID.
The Ethernet IP address for an interface is to implement LAN networking, while
media and signaling IP address pools provide alternate IP addresses for the SIP
server and SIP trunk.
Configuring an Ethernet IP Address


view.
2 Enter the interface interface-type -

interface view. interface-number

NetEngine AR
3 (Optional) Bind ip binding vpn-instance Before running the ip

a VPN instance vpn-instance-name binding vpn-instance
to an interface. command, create a VPN
instance.
When you run the ip
binding vpn-instance
command on an
interface, all Layer 3
features on the interface,
such as the IP address
and routing protocol, are
deleted. To use these
features, re-configure
them.
By default, no VPN
instance is bound to an
interface.
4 Configure an ip address ip-address If you configure a new IP

Ethernet IP { mask | mask-length } address for an interface
address. that already has an IP
address, the new IP
address overwrites the
old one.
5 Return to the quit -

system view.
NOTE
To improve network reliability, configure the loopback interface IP address for voice
communication. A loopback interface is always Up at the physical layer and link layer
unless it is manually shut down. For details on how to configure a loopback interface, see
Configuring a Loopback Interface.
Configuring Media and Signaling IP Address Pools


view.

NetEngine AR
2 Add an IP voip-address signalling To specify the ip-address

address to the interface interface-type parameter, ensure that
signaling IP interface-number { ip- an IP address has been
address pool. address | dynamic } configured for the
specified interface. To
specify the dynamic
parameter, ensure that
an IP address has been
dynamically allocated to
the specified interface.
3 Add an IP voip-address media It is recommended that

address to the interface interface-type the media and signaling
media IP address interface-number { ip- IP address pools be the
pool. address | dynamic } same.
4 (Optional) voip-vpn-instance vpn- Only SIP trunk groups

Configure a instance-name can be bound to VPN
bound VPN instances.
instance. Before running the voip-
vpn-instance command,
ensure that a VPN
instance has been
created and interfaces to
which the signaling IP
addresses and media IP
addresses belong have
been bound to the
instance.
By default, no VPN
instance is specified to
be used by voice services.
5 Save the save -

configuration.
Example
# Set the Ethernet IP address of interface GE0/0/0 to 192.168.1.3, and add IP
address 192.168.1.3 to media and signaling IP address pools.
<Huawei> system-view
[Huawei] interface gigabitethernet 0/0/0
[Huawei-GigabitEthernet0/0/0] ip address 192.168.1.3 24
[Huawei-GigabitEthernet0/0/0] quit
[Huawei] voice
[Huawei-voice] voip-address media interface gigabitethernet 0/0/0 192.168.1.3
[Huawei-voice] voip-address signalling interface gigabitethernet 0/0/0 192.168.1.3
[Huawei-voice] save

NetEngine AR
Verification
Check the display ip The value of IP Address/Mask is consistent

Ethernet IP interface brief with the setting.
address of [ interface-type <Huawei> display ip interface brief gigabitethernet
0/0/0
the interface. [ interface- Interface IP Address/Mask Physical Protocol
number ] ] GigabitEthernet0/0/0 192.168.1.3/24 up up
Check media display voice The value of IP address is consistent with

and signaling voip-address the setting.
IP address <Huawei> display voice voip-address
IP type : Media IP
pools. Method : Static
Interface : GigabitEthernet0/0/0
IP address : 192.168.1.3
IP type : Signal IP
Method : Static
Interface : GigabitEthernet0/0/0
IP address : 192.168.1.3
2.5.3 Configuring a Country Code and an Area Code

After a country code and an area code are configured, the PBX regulates calling
and called numbers from different countries or areas.
Context
The PBX adds digits to calling numbers to display incoming numbers, meeting user
needs. It regulates called numbers to accurately locate called parties.
NOTE
● Digits are collected at one time. The PBX cannot collect digits one by one.
● The PBX does not support number regulation over an R2 trunk.
● Figure 2-5 shows the calling number regulation process for incoming calls
over a trunk.

NetEngine AR
Figure 2-5 Calling number regulation process

Incoming calls over a trunk
Match preconfigured country

code prefix and country code
Yes
Are one or more country
code prefixes and country
codes matched?
No
Yes Delete the area code prefix and

Is the area code prefix on
supplement the country code
the trunk matched?
prefix and country code
No
Supplement the country code,

area code, and country code
prefix
Use the default

Yes No country code prefix
Is an intra-office user Is the country code of the
called? calling number the same as to replace the
the default one? country code of
the calling number
No Yes
Restore calling number Delete the country code
Pre-routing Number Change Use the default

No area code prefix
Is the area code of the to replace the
calling number the same as area code of the
Inter-office call the default one? calling number
Yes
Delete the area code
Intra-office call
● Figure 2-6 shows the called number regulation process for incoming calls
over a trunk.

NetEngine AR
Figure 2-6 Called number regulation process

Incoming calls over a trunk
No Check whether country

code prefix of calling party is the
same as the first digits of the called
number
Yes
Delete the prefix of the
country code
No
No Is area code prefix the same Is country code the same as
as default value? default value?
Yes Yes
Delete the prefix of the area code Delete the country code
Is area code the same as default No Supplement

value? the area code
Yes
Delete the area code
Yes
Intra-office call Is intra-office user called?
No
Pre-routing Number Change
Restore the deleted country code

and area code
Inter-office call
● Figure 2-7 shows the intra-office call number regulation process.

NetEngine AR
Figure 2-7 Intra-office call number regulation process

Outgoing call
No Does called
number contain
default country prefix
code?
Yes
Does called Does calling

No number contain Yes number contain
default area code default country prefix
prefix? code?
Yes No
Does calling Does calling

Yes number contain number contain No
default area code default area code
prefix? prefix?
No Yes
Delete area code prefix and Supplement default area
Supplement default area supplement default country code , country code and
code and area code prefix code and country code country code prefix
prefix
Inter-office call
Procedure
Ste Action Command Description
p
1 Enter the system-view -

system
view.
2 Enter the voice -

voice
view.

NetEngine AR

p
3 (Optional country-code country-code-value Before configuring a

) [ country-name country-name | country code, run
Configur international-prefix international- the display voice
ea prefix-value | national-prefix country-code
country national-prefix-value ]* command to view
code. the default country
code on the PBX.
NOTE
The PBX does not
support voice services
using non-standard
country or area
codes. If non-
standard country or
area codes are used,
calls may be
disconnected.
4 Configur pbx { default-country-code dcc-value By default, the

e the | default-area-code dac-value }* country code is 86
default and area code is
country 755.
code and
area
code.
5 Enable pbx enable-country-area-transform -

country enable
code and
area
code
change.
6 Save the save -

configura
tion.
Example
Set the default country code to 86 and area code to 571, and enable country code
and area code change.
<Huawei> system-view [Huawei] voice [Huawei-voice] pbx default-country-code 86 default-area-code
571 [Huawei-voice] pbx enable-country-area-transform enable [Huawei-voice] save

NetEngine AR
Verification
Verify the display voice pbx [Huawei-voice]dis voice pbx country-area-transform

Default country code:
default country-area- 86 Default area
and transform code:
configured 571 Enable
Country Area Transform Flag: enable
country
codes and
area codes.
2.5.4 Configuring an Enterprise and a DN Set

After multiple enterprises are configured on the device in PBX mode, they share
the PBX. A DN set consists of an enterprise's number rules. When one enterprise
shares a PBX, you do not need to configure enterprises. When one DN set meets
enterprise requirements, you do not need to configure DN sets.
Context
When multiple enterprises need to share one PBX, you can configure enterprises
on the PBX and virtualize the PBX into multiple PBXs. Configuring enterprises on
the PBX facilitates user management. Each terminal user is included in an
enterprise, and enterprises are independent and make inter-office calls.
NOTE
● By default, the enterprise default exists on the PBX.

● For detailed about the PBX shared by enterprises, see 2.20.4 Example for Configuring
PBX Sharing for Different Enterprises.
A DN set consists of a group of prefixes and is a set of an enterprise's number

rules. For example, if a user must dial 9 for a local call and 8288 for a long-
distance call, 9 and 8288 can be contained in a DN set. When configuring an
enterprise, you can create a DN set. When configuring a PBX user, you need to
specify the bound DN set. You can define multiple DN sets for an enterprise based
on number rules.
NOTE
By default, the PBX provides the DN set defaultdialplan for the enterprise default and new
enterprises.
Figure 2-8 shows the mapping between the PBX, enterprise, and DN set.

NetEngine AR
Figure 2-8 Mapping between the PBX, enterprise, and DN set
PBX
Enterprise name: default
Default PBX DN set: defaultdialplan
Enterprise A
Enterprise A
Virtual PBX DN set: defaultdialplan
(Optional) DN set
Enterprise B Enterprise B
Virtual PBX
… ... DN set: defaultdialplan
Procedure

view.

view.
3 Create an enterprise enterprise- If an enterprise already

enterprise and name exists, this command
enter the directly displays the
enterprise view. enterprise view.
By default, the enterprise
default exists on the
PBX.
4 Configure a DN set dn-set dn-set-name By default, the DN set

for the enterprise. [ description defaultdialplan exists on
description ] the PBX.
5 (Optional) description description -

Configure the
description.

NetEngine AR
6 Save the save -

configuration.
Example
Create an enterprise named hw and a DN set hwdnset.
[Huawei] voice
[Huawei-voice] enterprise hw
[Huawei-voice-enterprise-hw] dn-set hwdnset
[Huawei-voice-enterprise-hw] save
Verification
Verify the display voice The parameter values in the command

enterprise enterprise output are consistent with the settings.
configuration [ enterprise-name ] <Huawei>display voice enterprise hw
Enterprise : hw
. Max dn-set number :5
Dn-set : hwdnset
Dn-set description :
Dn-set : DefaultDialPlan
Dn-set description : DefaultDialPlan
2.6 Configuring a SIP Server

This section describes how to configure parameters of a SIP server, including the IP
address, URI, and home domain. SIP server is an important entity in the SIP
protocol architecture. The PBX can function as the SIP server to accept registration
information of SIP users, save the information in the address information
database, and manage and maintain users' registration information.
Prerequisites
You have finished 2.5.2 Configuring Media and Signaling IP Address Pools.
Procedure
1 Enter the system view. system-view -

NetEngine AR
2 (Optional) Create an ssl policy policy-name ● When the AR

SSL policy [ type { client | functions as the SIP
server } ] server and the
security of voice
signal transmission
needs to be
enhanced, you can
run this command
to configure an SSL
policy of server
type on the AR and
configure an SSL
policy of client type
on the remote
device. Data
encryption, identity
authentication, and
integrity
authentication
mechanisms of SSL
ensure data
transmission
security.
● When the AR
functions as the SIP
server and the
security of voice
signal transmission
needs to be
enhanced, you can
run this command
to configure an SSL
on the AR and
configure an SSL
policy of server
type on the remote
device. Data
authentication, and
integrity
authentication
mechanisms of SSL
ensure data
transmission
security.
3 Enter the voice view. voice -
4 Enter the SIP server sipserver -

view.

NetEngine AR
5 Configure a signaling signalling-address The signaling IP

IP address and { ip ip-address | addr- address must be
signaling port number name addr-name- selected from the
for the SIP server. value } port port- signaling IP address
value pool configured on an
interface.
6 Configure a signaling signalling-domain If the SIP server uses a

domain name for the signaling-domain- dynamic signaling IP
SIP server that uses a value address, you must
dynamic signaling IP configure a signaling
address. domain name for the
SIP server.
7 Configure a media IP media-ip { ip-address The media IP address

address for the SIP | addr-name addr- must be selected from
server. name-value } the media IP address
pool configured on an
interface.
8 Configure a uniform register-uri uri SIP users use this URI

resource identifier to register with the
(URI) for the SIP SIP server.
server.
9 Configure the home home-domain The SIP server uses

domain for the SIP domain the home domain
server. name as the user
domain name in the
From field in SIP
message headers.
10 (Optional) Enable number-parameter To use the registration

registration status 56 1 status subscription
subscription. function, set
Registration status
Subscription on SIP
terminals to Enable.
By default,
registration status
subscription is not
enabled.
11 (Optional) Configure transfer transfer- To configures an SSL

the transfer mode for value policy of server type
a SIP server. or client type when
the SIP server uses
TLS, set the transfer
mode for a SIP server
to TLS.

NetEngine AR
12 (Optional) Configure cs-mode cs-mode- When the transfer

the reliable connection value mode for a SIP server
mode for the SIP is TCP or TLS, this
server. command needs to be
configured.
● When the AR
functions as the
server, the reliable
connection mode
for the SIP server
need to be set to
server or
client_server.
● When the AR
functions as the
client, the reliable
connection mode
for the SIP server
need to be set to
client or
client_server.
13 (Optional) Configure ● the SSL policy of ● When the AR

an SSL policy of server server type: ssl- functions as the
type or client type server-policy server and is bound
when the SIP server server-policy-name to an SSL policy of
uses TLS. ● the SSL policy of server type, the
client type: ssl- remote device
client-policy needs to be
server-policy-name configured with an
SSL policy of client
type.
● When the AR
functions as the
client and is bound
to an SSL policy of
client type, the
remote device
needs to be
configured with an
SSL policy of server
type.
14 Reset the SIP server. reset -
15 Save the save -

configuration.

NetEngine AR
Example
# Configure a SIP server. Set its signaling IP address to 192.168.1.3, signaling port
number to 5060, media IP address to 192.168.1.3, registration URI to abcd.com,
and home domain to abcd.com.
[Huawei] voice
[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 abcd.com
[Huawei-voice-sipserver] home-domain abcd.com
[Huawei-voice-sipserver] reset
[Huawei-voice-sipserver] save
Verification
Check the display voice The parameter values in the command

configuration sipserver output are consistent with the settings.
of a SIP If the configuration information (Config) is
server. inconsistent with the running information
(Running), the current configuration does
not take effect. Run the reset command to
reset the SIP server for the configuration to
take effect.
2.7 Configuring Toll Fraud Prevention

You can configure limiting conditions on the PBX to prevent extra expenses due to
call fraud.
NOTICE
Plan and configure the service data according to this document. Otherwise, toll
fraud (making international toll calls or premium-rate telephone numbers
bypassing security policies) may occur, causing economic losses.
2.7.1 Context for Configuring Toll Fraud Prevention
Toll Fraud Scenarios

● Theft of the administrator account and password
A user illegally obtains the administrator account and password, logs in to the
device as the administrator, and modifies configuration data, such as the
number, trunk, and call detail record (CDR) configurations. By configuring the
required number and trunk and disabling the CDR function, the unauthorized
user can make fraudulent calls for which no CDR will be generated.

NetEngine AR
● Transfer by the automatic switchboard

As shown by line (1) in Figure 2-9, outer-office user B dials the automatic
switchboard number of the company with PBX1 deployed, and dials user C's
number when hearing an announcement or a two-stage dialing tone. If the
call is connected by the automatic switchboard, user B is charged only for a
local call, while the company must pay the toll call fee.
● Transfer through a trunk
As shown by line (2) in Figure 2-9, outer-office user C dials the incoming
prefix configured on PBX1, the outgoing prefix for making calls from PBX1 to
PBX2, and a toll number in sequence to make a toll call. For example, if 0 is
the incoming prefix configured on PBX1, and 00 is the outgoing prefix for
making calls from PBX1 to PBX2, user C can dial 0+00+123456789 to make a
call to toll number 123456789.
● Unauthorized registration
As shown by line (3) in Figure 2-9, an authorized user who is registered with
PBX2 by impersonating a registered user makes a call to outer-office user D.
The call is completely free of charge to the unauthorized user.
● Misuse of call forwarding services
As shown by line (4) in Figure 2-9, user B illegally obtains the administrator
account and password for logging to PBX1 or user A's self-service account and
password, and enables the call forwarding unconditional (CFU) service for
user A with user C's number configured as the forwarded-to number. When
user B makes a call to user A, the call is forwarded to user C regardless of
user A's status. User B pays only the local call fee to get connected to user C,
while the company must pay the toll call fee.

NetEngine AR
Figure 2-9 Toll fraud scenarios
User C User D
2
1
4
User B User E
PBX1 PBX2
PRA
User A User F
IP
Network
Unauthorized Unauthorized
user registartion
Recommended Prevention Measures

1. Prevent theft of the administrator right.
As long as the administrator right is used by an unauthorized user,
configuration data may be modified and other prevention measures take no
effect.
– Isolate private and public networks.
– Configure the administrator whitelist.
– Configure the user login right and login authentication.
2. Prevent calls from being transferred through a trunk and the IVR.
The PBX determines whether the call rights of the calling party meet the
prefix call attributes.

NetEngine AR
Figure 2-10 Determining calls over a trunk or the IVR
Incoming call
(Optional) Calling number

discrimination
Do the call rights No

of the calling party meet the The call is released
prefix call attributes?
Yes
No Intra-office/Outgoing
Is the call an IVR call?
call
Yes
Do the call rights No

of the calling party meet the The call is released
prefix call attributes?
Yes
Intra-office/Outgoing call
Configure data on the PBX to prevent call fraud.

– Configure the call-out right for calling number discrimination.
– Configure call attributes for the call prefix.
– Configure the call-out right.
– Configure the maximum number length for the IVR.
– Configure the IVR call-out right.
3. Prevent registration or login of unauthorized users.
When measures of 1 and 2 are taken, prevent SIP UE users from registering
on the PBX or logging in to the self-help service without authorization.
Configure an IP whitelist to prevent users using IP addresses not included in
the whitelist to initiate registration to the PBX. If no IP whitelist is configured,
unauthorized users can initiate registration to the PBX through the IP
network. If password authentication is configured during digit allocation,
registration becomes difficult. Unauthorized users can log in to the self-help
service through the IP network. If the self-help service password is changed,
login becomes difficult.
– Configure the IP whitelist.
– Configure password authentication for digit allocation.
– Change the self-help service password.
– Configure defense against brute force registration.

NetEngine AR
4. Configure call forwarding to prevent call fraud.

When the preceding measures are taken, properly configure the call
forwarding service to prevent unnecessary call forwarding and save enterprise
costs.
5. Deploy the CDR server.
2.7.2 Preventing Theft of the Administrator Right

This topic describes how to isolate the public network from the private network,
configure the whitelist, and configure the user login right and login authentication
for preventing theft of the administrator right.
Isolating Private and Public Networks

To ensure security, the device is required to be deployed on the private network. If
the device is deployed on the public network, risks (such as network attacks,
unauthorized registration, and account theft) may occur. If the device must be
deployed on the public network, take network isolation approaches, for example,
disabling SIP port 5060, SSH service port 22, and Telnet management port 23 on
the SBC and firewall to prevent communication through these ports. For detailed
firewall configuration, see Firewall Configuration.
NOTICE
If the device is deployed on the public network or the preceding ports are not
disabled on the SBC or firewall, unauthorized users may log in to the device and
modify the configuration data.
Configuring the Whitelist

If users on the public network need to access the device, configure the whitelist.
Users with the whitelisted IP addresses can perform administrator operations on
the device through the public network. For details about how to configure the
whitelist, see Configuring the Whitelist.
Configuring the User Login Right and Login Authentication

When a user logs in to the device through Telnet or SSH for local or remote
maintenance, perform the operation of Configuring the VTY User Interface
according to user and security requirements.
● User levels correspond to command levels. A user can run only the commands
of the same level or lower levels. You can configure different user levels for
different Telnet or SSH login users to ensure device security.
● The device provides AAA and password authentication. Configuring a user
authentication mode enhances device security. Ensure complexity of the
authentication password to prevent the password from being easily cracked.
To ensure password security, change the password periodically.

NetEngine AR
NOTICE
If the user level and login authentication are not configured, unauthorized users
can easily log in to the device and modify the configuration data.
2.7.3 Preventing Calls from Being Transferred Through a

Trunk and the IVR
This section describes how to configure call attributes for a call prefix, call-out
right for call transfer through a trunk, maximum number length for the IVR, and
call-out right for call transfer through the IVR to prevent inter-office users from
making outgoing calls through the PBX or IVR.
Context
Call rights are classified into five types:
● 0: intra-office call
● 1: local call
● 2: national toll call
● 3: international toll call
● 4: local survival call
It is recommended that you configure prefixes for intra-office, local, national toll,
and international toll calls. By default, even if the national and international toll
call prefixes are not configured, the system automatically determines whether a
call is a local, national toll, or international toll call according to the number a
user has dialed. If the user does not have the corresponding call rights, the system
denies the call. If a user has the requirements to make national or international
toll calls, configure the national (ddd) or international (idd) toll call rights.
NOTE
The 214 control point enables or disables automatic determination and call restriction
functions, and it does not affect the call restriction functions when national and
international toll call prefixes are configured.
In the tandem scenario, if the outgoing prefix of PBX2 is the same as the country
code prefix or area code prefix of PBX1, users under PBX1 need to dial the
outgoing prefix (for example, 9) of PBX1, outgoing prefix (for example, 0) of
PBX2, and then a user number, which is 90+user number, and one digit of the
called number needs to be deleted on PBX1. Figure 2-11 shows the trunk tandem
network. Assuming that the country code prefix is 00 and area code prefix is 0, if a
user under PBX1 has only the right to make local calls, the user cannot make
outgoing calls by dialing 90+user number. This is because PBX1 considers 0+user
number as a toll call number. When the national and international toll call
prefixes are not configured, configure the 214 control point so that the system will
restrict calls not according to the calling parties' call-out rights. For details, see
pbx number-parameter.

NetEngine AR
Figure 2-11 Trunk tandem network
User C
PBX1 PBX2
User C
0+User C
90+User C
User A User B
NOTICE
If the 214 control point is configured for the system to restrict calls not according
to the calling parties' call-out rights, toll fraud may occur. In this case, the national
and international toll call prefixes must be configured to prevent toll fraud.
Mapping between the call prefix attribute and call-out rights for calls transferred
over a trunk and the IVR:
Assume that the right for the inter-office call prefix between PBX1 and PBX2 is 1.
If the scenario where PSTN users make incoming calls through PBX1 and calling
users on PBX2 is not limited, the following setting is recommended:
● If a PSTN user calls a user on PBX2 through the trunk of PBX1, the call-out
right of the trunk must contain the call-out right (right 1) of the inter-office
call prefix.
● If a PSTN user dials the IVR through PBX1 and calls a user on PBX2, the IVR
and trunk call-out rights of PBX1 must contain the call-out right (right 1) of
the inter-office call prefix.
Configuring the Call-Out Right for Calling Number Discrimination

For the calling party whose incoming calls are made over a trunk, the default call-
out right is the same as that for calls transferred over a trunk. If the call-out right
for calling number discrimination is configured on the PBX, the call-out of the
calling party is that configured for calling number discrimination. To prevent users
from calling inter-office users after calls are connected over a trunk, configure the
call-out right for calling number discrimination.
It is recommended that the call-out right for calling number discrimination be
configured in the following two scenarios:

NetEngine AR
● Specific users need to have special rights.

● Calling numbers other than the specific ones (7000–7999) are not allowed to
make outgoing calls.
This example assumes that the call-out right is inter for call prefix 7*. For details,
see 2.10.1 Calling Number Discrimination.
<Huawei> system-view [Huawei] voice [Huawei-voice] caller-change 0 [Huawei-voice-caller-change-0]
callerprefix 7* [Huawei-voice-caller-change-0] outgoing-right
inter enable [Huawei-voice-caller-change-0] caller no-change [Huawei-voice-caller-change-0] called no-
change [Huawei-voice-caller-change-0] save
Configuring Call Attributes for the Call Prefix

Assume that the prefixes 9, 90, and 900 indicate the local calls, national toll calls,
and international toll calls. Call prefix 9 is used as an example. For call prefixes 90
and 900, set attribute values to 2 and 3. For details, see 2.9 Configuring a Call
Prefix.
<Huawei> system-view [Huawei] voice [Huawei-voice] callprefix 9 [Huawei-voice-callprefix-9] prefix 9
[Huawei-voice-callprefix-9] call-type category basic-service attribute 1 [Huawei-voice-callprefix-9] digit-
length 1 32 [Huawei-voice-callprefix-9] save
Configuring the Call-Out Right for Calls Transferred Through a Trunk

Call-out rights are classified into inter, local, ddd, idd, all. You can configure
multiple call-out rights simultaneously. The default call-out rights for calls
transferred through a trunk are inter and local. To enable the PBX not to forward
incoming calls over a trunk, configure inter. If a user makes a call over a trunk
and wants to call a local user, configure inter and local. If a user wants to make
national or international toll calls, configure ddd or idd. For details on how to
configure call-out rights, see 2.15.4.1 Configuring Call Limiting Over a Trunk.
It is recommended that outgoing-right of the trunk between PBX1 and PBX2 be
set to inter&local and outgoing-right of the trunk between the PBXs and the
PSTN be set to inter.
For example, the trunk between PBX1 and PBX2 is sippra and the trunk between
PBX1 and the PSTN is sipip1. The outgoing-right of the trunk sippra is local and
is bound to call route 1. For details, see 2.11 Configuring Trunk Groups.
<Huawei> system-view [Huawei] voice [Huawei-voice] trunk-group sippra [Huawei-voice-trunkgroup-
sippra] outgoing-right local enable [Huawei-voice-trunkgroup-sippra] callroute 1 [Huawei-voice-
trunkgroup-sippra] save
NOTICE
If a trunk has local right or more rights, a user can make a call over this trunk and
calls inter-office users, which may cause a call fraud.
Configuring the Maximum Number Length for the IVR

Assume that a 4-digit extension number is used. Set the second-dialing number
length to 4 so that an inter-office user only needs to dial the IVR and extension
number to call the intra-office user. If the second-dialing number length is larger
than or equal to the inter-office number length, when inter-office call conditions
are met, an inter-office user can dial the IVR and another inter-office user. As a

NetEngine AR
result, a call fraud occurs. To prevent users from dialing the IVR and inter-office
user's number, set the second-dialing number length to the intra-office number
length. For details, see IVR Configuration.
Configuring the IVR Call-Out Right

Call-out rights are classified into inter, local, ddd, idd, all. You can configure
multiple call-out rights simultaneously. The default call-out right is inter. By
default, after a user dials the IVR, the user cannot call an inter-office user. If a user
wants to dial the IVR and make outgoing calls, configure the call-out right.
In this example, the IVR call-out right is local, and the call-out right
vuoutgoingright is used.
<Huawei> system-view [Huawei] voice [Huawei-voice] callprefix 888 [Huawei-voice-callprefix-888]
vuoutgoingright
local enable [Huawei-voice-callprefix-888] save
NOTICE
When the IVR has local right or more rights, a user can dial the IVR, hear the
second-stage dialing tone, and dial the inter-office user's number, which may
cause a call fraud.
2.7.4 Preventing Registration or Login of Unauthorized Users

This section describes how to configure the IP whitelist, configure user password
authentication, and change the self-help service password to prevent unauthorized
users from registering with the PBX or using the default password to log in to the
self-help service to modify data or make a call fraud.
Configuring the IP Whitelist

After IP addresses of IP phones are added to the whitelist, such IP addresses can
be correctly registered. For details, see Configuring the Whitelist.
Configuring User Password Authentication

It is recommended that password authentication be configured when users are
added. Password authentication prevents unauthorized users from using user
names to register with the PBX. For details, see 2.8.2 Configuring a SIP User.
NOTICE
If no password is configured for a registered user or the password leaks,

unauthorized users can make a call fraud through the registered user.
Changing the Self-Help Service Password

The default username and password are available in AR Router Default
Usernames and Passwords (Enterprise Network or Carrier). If you have not
obtained the access permission of the document, see Help on the website to find

NetEngine AR
out how to obtain it. To ensure password security, change the password
periodically.
● For enterprise users, access the web self-help service system using the
corresponding user name and password. After a successful login, click Change
Password in the upper right corner to change the password.
● For the administrator, run the web-password cipher command in the user
view to change the password.
NOTICE
If the self-help service password is not changed or kept, unauthorized users may
use this password to log in to the PBX to modify conference management and
service registration data, for example, configure the PBX to forward all calls to a
destination toll call number.
Defense Against Brute Force Registration

After the user registration number fails authentication, the PBX records the failure
time. When the user registers with the PBX again, the PBX determines the
difference between the current system time and last failure time. If the difference
is larger than the protection time, the registration process is used. Otherwise, the
system displays a message indicating registration failure. The protection time after
registration failure can be set through control point 205. For details, see pbx
number-parameter.
NOTICE
The default value of control point 205 is 10. The value of control point 205 ranges
from 0 to 300, in seconds. It is recommended that you adjust the protection time
to a proper value. If a short protection time is used, unauthorized users may
continuously use different passwords to initiate registration to the PBX, and the
user passwords may finally be cracked.
2.7.5 Preventing Unauthorized Call Forwarding

This section describes how to properly configure the call forwarding service to
prevent unnecessary call forwarding and save enterprise costs.
The call forwarding service is configured according to user requirements. For
example, employees are allowed to user the call forwarding service at work time.
The following example configures the CFC service for intra-office user 6000, and
configures the PBX to forward calls to 6001 from 09:00 to 17:00 at Monday when
there is no response. For details, see 2.14.3.4 Configuring Call Forwarding
Services.
[Huawei] voice
[Huawei-voice] pbxuser 6000
[Huawei-voice-pbxuser-6000] service-right cfc enable
[Huawei-voice-pbxuser-6000] forward-condition id 1 forwardnum 6001 state noanswer weekly mon
09:00:00 17:00:00

NetEngine AR
[Huawei-voice-pbxuser-6000] service cfc condition 1

[Huawei-voice-pbxuser-6000] save
2.7.6 Deploying the CDR Server

You can query call fraud information on the CDR server when a fraud occurs.
For details on how to configure a CDR server, see 2.15.6 Configuring Connection
Data for the CDR Server.
2.7.7 Verifying the Toll Fraud Prevention Configuration

After the configuration is complete, check whether an inter-office user can log in
to the enterprise PBX through the Internet, whether a user makes an outgoing call
by calling the IVR or through a trunk, whether an unauthorized user can be
registered with the PBX, and whether call forwarding can be implemented when
call forwarding conditions are not met.
● An inter-office user cannot log in to the enterprise PBX through the Internet.
● An inter-office user calls the IVR, but outgoing calls cannot be made. For
example, when an inter-office user calls the IVR, performs second-stage
dialing, and dials a number, the call fails.
● When an inter-office user dials the intra-office call prefix + inter-office call
prefix + toll call number, the toll call fails. For example, user C cannot make a
call by dialing 0 + 00 + 001123456789.
● A user cannot be registered with the PBX through no-authentication or an IP
address not whitelisted, or cannot log in to the self-help service through the
default password.
● Call forwarding cannot be implemented when call forwarding conditions are
not met. For example, call forwarding is performed at work hours. Call
forwarding cannot be implemented at off-work hours.
2.7.8 Troubleshooting Toll Fraud Prevention

This section describes exception handling.
● If an external user can log in to the PBX, solve the problem according to 2.7.2
Preventing Theft of the Administrator Right.
● If a user calls the IVR, performs second-stage dialing as prompted, and
successfully makes a call, solve the problem according to 2.7.3 Preventing
Calls from Being Transferred Through a Trunk and the IVR.
● If an inter-office user dials the intra-office call prefix + inter-office call prefix +
toll call number, solve the problem according to 2.7.3 Preventing Calls from
Being Transferred Through a Trunk and the IVR.
● If a user cannot be registered with the PBX through no-authentication or an
IP address not whitelisted, or cannot log in to the self-help service through
the default password, solve the problem according to 2.7.4 Preventing
Registration or Login of Unauthorized Users.
● If the call forwarding service is implemented when call forwarding conditions
are not met, solve the problem according to 2.7.5 Preventing Unauthorized
Call Forwarding.

NetEngine AR
2.8 Configuring a PBX User

You can configure different types of PBX users according to actual networking and
user planning. Terminal users on the PBX contain POTS, SIP, and ISDN users.
2.8.1 Configuring a POTS User

After a number and call right are configured for a POTS user, the POTS user device
such as a POTS phone can make incoming and outgoing calls.
Context
POTS users refer to POTS phone users or fax machine users. POTS user devices
connect to voice cards (4FXS1FXO or 16/32FXS , and 4FXS card) of the PBX
through common phone lines.
As shown in Figure 2-12, the PBX connects to POTS users.
Figure 2-12 PBX connects to POTS users

PBX
POTS POTS
Fax machine
phone phone
The PBX can control call-in and call-out rights. Calls are classified into intra-office
calls, local calls, national toll calls, international toll calls and local survival call.
Procedure

view.

view.

NetEngine AR
3 (Optional) Create pbxuser-batch pots You can run this command

POTS users in a pbxuser name port to connect multiple POTS
batch. slotid/subcardid/ phones to FXS interfaces of
portid number num- the same voice card and
value allocate consecutive user
numbers to the POTS
users.
By default, no POTS user is
created.
4 Create a POTS pbxuser name pots By default, a SIP user

user, (optional) [ enterprise belongs to the enterprise
specify the enterprise-name ] default. You need to
enterprise that the configure enterprise
POTS user belongs enterprise-name parameter
to, and enter the only when multiple
POTS user view. enterprises share the PBX.
For details on how to
configure an enterprise, see
2.5.4 Configuring an
Enterprise and a DN Set.
After a POTS user is
created, you can run this
command to enter the
POTS user view.
5 Bind a physical port slotid/ The physical interface is in

interface to the subcardid/portid the format of slot ID/card
POTS user. ID/interface ID. If no card is
installed, the card ID is 0.
If you create POTS users in
a batch, this step is
optional.
6 Configure a telno telno-value telno is generally set to an

telephone number [ long-telno long- intra-office number for
and long code. telno-value ] intra-office calls. long-
telno is generally set to a
number allocated by the
carrier for DDI incoming
and outgoing calls.
If you create POTS users in
optional.
7 (Optional) call-right in { { inter By default, intra-office,

Configure the call- inter-value | local local, national, and
in right. local-value | ddd international incoming
ddd-value | idd idd- calls are all allowed.
value } * | all }

NetEngine AR
8 (Optional) call-right out By default, only intra-office

Configure the call- { { inter inter-value | and local call can be made.
out right. local local-value |
ddd ddd-value | idd
idd-value } * | all }
9 (Optional) Bind a dn-set dn-set-name By default, a POTS user
DN set to the SIP belongs to the DN set
user. DefaultDialPlan. You need
to perform this step only
when a DN set is created.
configure a DN set, see
10 (Optional) user-type type By default, users are

Configure the user common users.
type.
11 (Optional) user-name user- You can run this command

Configure the name-value to configure a name for a
name of a POTS POTS user. When a remote
user. user has enabled the name
identification presentation
service, the configured
name will be displayed on
the phone of the remote
user.
By default, the name of
POTS user is not
configured.
12 (Optional) eid-para description By default, the description

Configure the description of a user is not configured.
description.
13 Save the save -

configuration.
Example
Configure a POTS user. Set the user name to 7000, telephone number to 7000,
and long code to 28987000, enable the national toll call right, and bind physical
interface 4/0/1 to the POTS user.
[Huawei] voice
[Huawei-voice] pbxuser 7000 pots
[Huawei-voice-pbxuser-7000] port 4/0/1
[Huawei-voice-pbxuser-7000] telno 7000 long-telno 28987000

NetEngine AR
[Huawei-voice-pbxuser-7000] call-right out ddd enable

Verification
Verify the POTS display voice pbxuser The parameter values in the
user [ pbxuser-name ] command output are
configuration. consistent with the setting.
2.8.2 Configuring a SIP User

After a number and call right are configured for a Session Initiation Protocol (SIP)
user, the SIP user device such as an IP phone can make incoming and outgoing
calls.
Context
A SIP user connects calls on the SIP server through SIP. SIP user devices can be IP
phones, eSpace software terminals, and POTS phones connected to the eSpace
IAD. The PBX as the SIP server receives registration and session requests of SIP
users.
As shown in Figure 2-13, the PBX connects to SIP users.
Figure 2-13 PBX connects to SIP users
PBX
IAD
IP eSpace soft
phone terminal
POTS POTS
phone phone

NetEngine AR
Procedure

view.
3 (Optional) Create SIP pbxuser-batch You can run this

users in a batch. sipue pbxuser command to create
name sipue eid- multiple SIP users and
value number num- allocate consecutive user
value numbers to the SIP users.
It is recommended that
eid-value (device
identifier) be the same as
name (user name).
By default, no SIP user is
created.
4 Create a SIP user, pbxuser name By default, a SIP user

(optional) specify the sipue [ enterprise belongs to the enterprise
enterprise that the enterprise-name ] default. You need to
SIP user belongs to, configure enterprise
and enter the SIP enterprise-name
user view. parameter only when
multiple enterprises share
the PBX.
configure an enterprise,
see 2.5.4 Configuring an
After a SIP user is
command to enter the
SIP user view.
5 Configure a device sipue eid-value It is recommended that

identifier. the device identifier be
the same as the user
name.
If you create SIP users in
optional.

NetEngine AR
6 Configure a telno telno-value A long code is allocated

telephone number [ long-telno long- by a carrier. After a long
and long code. telno-value ] code is configured,
external users can dial
the long code to call
users. They do not need
to dial the switchboard
number and extension
number.
If you create SIP users in
optional.
7 Configure an eid-para password After creating a SIP user,

authentication cipher password- configure an
password. value authentication password
for the user. If no
authentication password
is configured, the user
account may be stolen by
unauthorized users.
8 (Optional) Configure call-right in By default, intra-office,

the call-in right. { { inter inter-value local, national, and
| local local-value | international incoming
ddd ddd-value | idd calls are all allowed.
9 (Optional) Configure call-right out By default, only intra-
the call-out right. { { inter inter-value office and local call can
| local local-value | be made.
ddd ddd-value | idd
10 (Optional) Bind a DN dn-set dn-set-name By default, a SIP user
set to the SIP user. belongs to the DN set
DefaultDialPlan. You
need to perform this step
only when a DN set is
created.
configure a DN set, see
11 (Optional) Configure user-type type By default, users are

the user type. common users.

NetEngine AR
12 (Optional) Configure user-name user- You can run this

the name of a SIP name-value command to configure a
user. name for a SIP user.
When a remote user has
enabled the name
identification
presentation service, the
configured name will be
displayed on the phone
of the remote user.
SIP user is not configured.
13 (Optional) Configure eid-para By default, the

the description. description description of a user is
description not configured.
14 Save the save -

configuration.
Example
Configure a SIP user with the user name 7100, device identifier 7100, telephone
number 7100, long number 28987100, and authentication password a12345.
Grant the national long-distance call-out permission to the SIP user.
[Huawei] voice
[Huawei-voice] pbxuser 7100 sipue
[Huawei-voice-pbxuser-7100] sipue 7100
[Huawei-voice-pbxuser-7100] eid-para password cipher
Please input user password(6-64 chars): ******
Verification
Verify the SIP display voice pbxuser The parameter values in the
2.8.3 Configuring an ISDN User

After a number and call rights are configured for an ISDN user, its ISDN phone can
make incoming and outgoing calls.

NetEngine AR
Context
When an ISDN user connects an ISDN telephone line to the BRA interface of a
PBX, the BRA interface must work in NT mode.
As shown in Figure 2-14, the PBX connects to ISDN users.
Figure 2-14 PBX connects to ISDN users

PBX
ISDN ISDN ISDN

phone phone phone
Configuring the Working Mode of the 2BST Card or BRA Interface
NOTICE
After the working mode of the 2BST card is configured, original service data on
the 2BST card and interfaces of the 2BST card will become invalid.

NetEngine AR
2 Configure the Configure set workmode slot After this

card or the 2BST slot-id bri bri-voice command is
interface to card to work nt-mode executed, the
work in NT in NT mode. system asks
mode. you whether to
(Configure restart the
the working 2BST card. If
mode of the you select Y,
card or the 2BST card
interface as restarts. If the
required.) 2BST card does
not restart in
this situation,
run the reset
slot command
to restart the
2BST card.
If the 2BST card
works in TE
mode and has
BRA trunk data
configured,
delete the BRA
trunk data after
you change the
working mode
to NT.
(Optional) Configuring Remote Power Supply on the BRA Interface

An ISDN phone is powered by a dedicated transformer or the BRA interface of the
2BST card.
If no dedicated transformer is available, enable the remote power supply function

on the BRA interface connected to the ISDN phone.

view.
2 Enter the BRA port bra slotid/ The physical interface is in

interface view. subcardid/portid the format of slot ID/card
ID/port number. If no card is
3 Enable the remote-power By default, the remote

remote power enable power supply function is
supply function. disabled.

NetEngine AR

voice view.
Configuring an ISDN User


view.
2 Create an ISDN pbxuser name bra By default, an ISDN user

user, (optional) [ enterprise belongs to the enterprise
specify the enterprise-name ] default. You need to
enterprise that configure an enterprise only
the ISDN user when multiple enterprises
belongs to, and share the PBX.
enter the ISDN For details on how to
user view. configure an enterprise, see
After an ISDN user is
command to enter the ISDN
user view.
3 Bind a physical port slotid/ The physical interface is in

interface to the subcardid/portid the format of slot ID/card
ISDN user. ID/port number. If no card is
4 Configure a telno telno-value A long code is allocated by a

telephone [ long-telno long- carrier. After a long code is
number and long telno-value ] configured, external users
code. can dial the long code to call
users. They do not need to
dial the switchboard number
and extension number.
5 (Optional) call-right in { { inter By default, intra-office, local,

Configure the inter-value | local national, and international
call-in right. local-value | ddd incoming calls are all
ddd-value | idd idd- allowed.
value } * | all }
6 (Optional) call-right out By default, only intra-office
Configure the { { inter inter-value | and local call can be made.
call-out right. local local-value |
ddd ddd-value | idd

NetEngine AR
7 (Optional) Bind dn-set dn-set-name By default, a SIP user

a DN set to the belongs to the DN set
ISDN user. DefaultDialPlan. You need
to perform this step only
when a DN set is created.
configure a DN set, see 2.5.4
Configuring an Enterprise
and a DN Set.
8 (Optional) user-type type By default, users are

Configure the common users.
user type.
9 (Optional) user-name user- You can run this command

Configure the name-value to configure a name for an
name of an ISDN ISDN user. When a remote
user. user has enabled the name
identification presentation
service, the configured name
will be displayed on the
phone of the remote user.
ISDN user is not configured.
10 (Optional) eid-para signalling When the PBX functions as a

Configure the sigtype user-side device and uses
user signaling DSS1 signaling to connect to
type. the remote network, set the
signaling type to dss1
(default).
When the PBX functions as a
user-side device and uses
QSIG signaling to connect to
the remote network, set the
signaling type to qsig.
11 (Optional) eid-para description By default, the description of

Configure the description a user is not configured.
description.
12 Save the save -

configuration.
Example
Configure an ISDN user. Set the user name to 7200, telephone number to 7200,
and long code to 28987200, enable the national toll call right, and bind physical
interface 2/0/1 of the 2BST card to the ISDN user.
[Huawei] set workmode slot 1 bri bri-voice nt-mode

NetEngine AR
[Huawei] voice
[Huawei-voice] pbxuser 7200 bra
Verification
Verify the ISDN display voice pbxuser The parameter values in the
2.9 Configuring a Call Prefix

After intra-office and inter-office prefixes are configured on the PBX, intra-office
users can call each other, and intra-office users can call inter-office users.
Context
A call prefix is a string of consecutive digits starting from the first digit of a called
number. It can be the first digit or several digits starting from the first digit of a
called number. That is, a call prefix is a subset of a called number. For example,
you can define either of the following intra-office call prefixes for the called
number 1234:
● First digit: 1
● First two digits: 12
● First three digits: 123
● Called number: 1234
A group of call prefixes configured on the PBX constitutes a DN set. If the

preceding call prefixes all exist in the DN set, the PBX analyzes the called number
according to the longest match principle. For example, a user calls 1234. If call
prefixes 1, 12, and 1234 are configured in a DN set, the PBX matches the called
number with the call prefix 1234 according to the longest match principle.
Call prefixes are classified into the following types:
● Intra-office call prefix: is applicable to scenarios where intra-office and inter-

office users call intra-office users. For example, the intra-office number range
is 7000 to 7099. You can configure the intra-office call prefix 7. You need to
dial only the intra-office user number such as 7001 when calling an intra-
office user.
● Inter-office call prefix: Inter-office call prefixes are used by intra-office users
to make outgoing calls, for example, intra-office users make local calls,
national toll calls, and international toll calls. Assuming that the inter-office
call prefix is 9, intra-office user 7000 can dial 912345678 when calling inter-
office user 12345678. The PBX parses and changes the number (for example,
delete 9) to make outgoing calls, as shown in Figure 2-15.

NetEngine AR
Figure 2-15 Inter-office call

Inter-office
call prefix:
Dial 12345678 9
PBX
Dial
912345678
12345678 7000
NOTE
A call prefix can be flexibly configured, and depends on the user number plan.
It is recommended that you configure prefixes for intra-office, local, national toll,
and international toll calls. By default, even if the national and international toll
call prefixes are not configured, the system automatically determines whether a
call is a local, national toll, or international toll call according to the number a
user has dialed. If the user does not have the corresponding call rights, the system
denies the call. If a user has the requirements to make national or international
toll calls, configure the national (ddd) or international (idd) toll call rights.
NOTE
The 214 control point enables or disables automatic determination and call restriction
functions, and it does not affect the call restriction functions when national and
international toll call prefixes are configured.
In the tandem scenario, if the outgoing prefix of PBX2 is the same as the country
code prefix or area code prefix of PBX1, users under PBX1 need to dial the
outgoing prefix (for example, 9) of PBX1, outgoing prefix (for example, 0) of
PBX2, and then a user number, which is 90+user number, and one digit of the
called number needs to be deleted on PBX1. Figure 2-16 shows the trunk tandem
network. Assuming that the country code prefix is 00 and area code prefix is 0, if a
user under PBX1 has only the right to make local calls, the user cannot make
outgoing calls by dialing 90+user number. This is because PBX1 considers 0+user
number as a toll call number. When the national and international toll call
prefixes are not configured, configure the 214 control point so that the system will
restrict calls not according to the calling parties' call-out rights. For details, see
pbx number-parameter.

NetEngine AR
Figure 2-16 Trunk tandem network
User C
PBX1 PBX2
User C
0+User C
90+User C
User A User B
NOTICE
If the 214 control point is configured for the system to restrict calls not according
to the calling parties' call-out rights, toll fraud may occur. In this case, the national
and international toll call prefixes must be configured to prevent toll fraud.
Procedure

view.
3 Create a call prefix callprefix callprefix- -

profile and enter the name
call prefix profile
view.
4 Configure a call prefix prefix -

prefix.
5 Configure the call call-type category -

type and call callcategory
attribute for the call attribute attribute
prefix.

NetEngine AR
6 Set the length of a digit-length The length of a number

number that can be minimum-length- that can be parsed
parsed. value maximum- contains the call prefix.
length-value For example, when the
length of a number that
can be parsed is 5 and
the call prefix is 88, the
PBX parses 88***.
The maximum length of
a number that can be
parsed must be larger
than or equal to the
minimum length of a
number that can be
parsed.
7 (Optional) Configure enterprise By default, a call prefix

the enterprise and enterprise-name belongs to the enterprise
DN set that the call [ dn-set dn-set- default and DN set
prefix belongs to. name ] DefaultDialPlan.
You need to perform this
step only when multiple
enterprises and DN sets
are configured. For
details on how to
configure an enterprise
and a DN set, see 2.5.4
Configuring an
8 (Optional) Configure vuoutgoingright If in step 5, you run the

the interactive voice { { inter value | call-type command to
response (IVR) local value | idd set vu-service to the IVR
service to transfer value | ddd value } * service, run the
inter-office calls. | all value } vuoutgoingright
command to enable the
IVR to transfer inter-
office calls.
9 (Optional) Configure re-route re-route- By default, the function

the function of value of rerouting upon a call
rerouting upon a call failure is disabled.
failure.
10 (Optional) Configure collectcall By default, the function

the function of the collectcall-value of the called party
called party payment is disabled.
payment.
11 Save the save -

configuration.

NetEngine AR
Example
Set the intra-office call prefix to 7 and inter-office call prefix to 9.
[Huawei] voice
[Huawei-voice] callprefix 7
[Huawei-voice-callprefix-7] prefix 7
[Huawei-voice-callprefix-7] call-type category basic-service attribute 0
[Huawei-voice-callprefix-7] digit-length 3 4
[Huawei-voice-callprefix-7] quit
[Huawei-voice-callprefix-9] save
Verification
Verify the call display voice callprefix The parameter values in the
prefix [ callprefix-name ] command output are
2.10 Configuring Number Change

This section describes how to configure calling/called number change, long and
short codes, and number mapping.
2.10.1 Calling Number Discrimination

Calling number discrimination is a type of number change. This service allows the
PBX to change outgoing called and calling numbers and to limit the call-out right
of a calling party using a prefix.
Context
Calling number discrimination can be used in the following scenarios:
● To show a specific number to the called party, configure the specific number
to replace the calling number.
● To change the number as a specific number, configure a new number to
replace the called number.
● To control the call-out right for calling numbers over a trunk, configure the
call-out right.
Figure 2-17 shows how an example of calling number discrimination. In this

example, the calling number 7000 is changed to 28987000 on the phone of the
called party through the calling number discrimination function.

NetEngine AR
Figure 2-17 Calling number discrimination
Call 12345678
PBX
Called number: Calling number:
12345678 28987000
Calling number: Called number:
7000 12345678
User A User B
7000 12345678
Prerequisites
● A PBX user has been configured. For details about the configuration, see 2.8
Configuring a PBX User.
● A call prefix has been configured. For details about the configuration, see 2.9
Configuring a Call Prefix.
● The PSTN assigns number 28987000 to the PBX user.
Procedure
p
3 (Optional) Create an enterprise enterprise- If the enterprise

enterprise and enter name exists, this
the enterprise view. command displays
the enterprise
view directly.
The device
provides a default
enterprise name
default.
4 (Optional) Configure a dn-set dn-set-name A new enterprise

DN set for the [ description description ] does not have any
enterprise. DN set. You must
configure a DN set
for the enterprise.
The enterprise
default has a
default DN set
named
defaultdialplan.

NetEngine AR

p
5 (Optional) Return to quit -

the voice view.
6 Create a calling caller-change caller- By default, no

number change plan change-name calling number
and enter its name. change plan exists.
7 Configure a call prefix. callerprefix callprefix- -

name
8 Configure a calling caller { del-then-insert -
number change rule. del-offset del-len insert-
telnum | del del-offset del-
len | insert insert-offset
insert-telnum | no-
change }
9 Configure a called called { del-then-insert -

number change rule. del-offset del-len insert-
telnum | del del-offset del-
len | insert insert-offset
insert-telnum | no-
change }
10 (Optional) Configure outgoing-right By default, the

the calling party call- call-out right for
out right. calls transferred is
used.
11 Save the configuration. save -
Example
Crete calling number change plan 0. Bind it to calling prefix 7* for changing the
calling numbers to 28987000. Set the called number change rule to deleting the
first digit.
[Huawei] voice
[Huawei-voice] caller-change 0
[Huawei-voice-caller-change-0] callerprefix 7*
[Huawei-voice-caller-change-0] caller del-then-insert 1 32 28987000
[Huawei-voice-caller-change-0] called del 1 1
[Huawei-voice-caller-change-0] save

NetEngine AR
Verification
Verify the display voice caller-change The parameter values in

configuratio the command output
n of calling are consistent with the
number settings.
change.
2.10.2 Configuring Long and Short Codes

A short code is used by intra-office users, and a long code is used to display
numbers of outgoing calls or used by inter-office users.
Context
Short codes can be only used by intra-office users, and cannot be identified by
carrier-side devices. To implement communication between enterprise users and
carrier-side users, configure long and short codes and number mapping so that
the numbers displayed can be identified by the carrier-side devices. For example,
the short and long codes of an intra-office user are 7000 and 28987000. When
short code 7000 is used to call user 28980808, calling number 28987000 is
displayed. Figure 2-18 shows networking of long and short codes.
Figure 2-18 Networking of long and short codes
PBX
Calling party: 7000 Calling party: 28987000

Called party: Called party: 28980808
928980808
User A User B User C

Short code: 7000 Short code: 7001 Number:
Long code: Long code: 28980808
28987000 28987001
Prerequisites
● A PBX user has been configured. For details on how to configure a PBX user,
see 2.8 Configuring a PBX User.
● A call prefix has been configured. For details on how to configure a call prefix,
see 2.9 Configuring a Call Prefix.
● The long codes allocated by the PSTN to PBX users are 28987000 and
28987001.

NetEngine AR
Procedure

system view.
2 Enter the voice -

voice view.
3 Enter the pbxuser name -

user view.
4 Configure a telno telno-value [ long-telno long- -

long code. telno-value ]
5 Enter the quit -
voice view.
6 Enter the callprefix callprefix-name -

call prefix
profile view.
7 Configure special-deal long-caller enable By default, the

the PBX to PBX does not
display the display the long
long code of code of a calling
a calling number.
number.
8 Save the save -

configuratio
n.
Example
Configure short code 7000 and long code 28987000 for user 7000, and configure
the PBX to display 28987000 for inter-office call prefix 9.
<Huawei> system-view [Huawei] voice [Huawei-voice] pbxuser 7000 [Huawei-voice-pbxuser-user7000]
telno 7000 long-telno 28987000 [Huawei-voice-pbxuser-user7000] quit [Huawei-voice] callprefix 9
[Huawei-voice-callprefix-9] special-deal long-caller enable [Huawei-voice-callprefix-9] save
Verification
Verify display voice pbxuser The configured value of Long telno is

the PBX pbxuser-name displayed.
user [Huawei-voice]display voice pbxuser 7000
configu PBX
ration. user : 7000 Enterprise : default
Port : 4/0/1 Sipue Type
: Dn-set : DefaultDialPlan
Telephone
number : 7000 Long telno : 28987000

NetEngine AR
Verify display voice callprefix The configured value of Long cli switch is
the call callprefix-name displayed.
prefix [Huawei-voice]display voice callprefix 9
configu Callprefix
ration. :9 Enterprise : default Dn-set
: DefaultDialPlan Prefix :9 Call
category
: Basic service Call attribute : 1(Local)
Minimum length
:1 Maximum length : 15 Ring delay(s) :0
Long
cli switch : enable
2.10.3 Configuring Number Mapping

Number mapping allows the PBX to map external numbers to internal numbers
and re-parse numbers and locate called parties according to the matched prefix.
Context
When number discrimination and long and short codes number change rules are
not matched, number mapping is recommended. PBX1 and PBX2 in a company
connect to the PSTN through PRA trunks, and they are connected through a trunk.
On PBX1, the intra-office call prefix is 70 and the call attribute is inter; the inter-
office call prefix is 71 and the call attribute is local. For example, 7000, 7001, and
7100 are mapped to 28987000, 28987001, and 28987100 respectively. When
carrier-side user 28980808 calls 28987100 through PBX1, PBX1 determines that
the number is not an intra-office number. Then PBX1 re-parses the number and
forwards the call to PBX2 according to call prefix 71. Figure 2-19 shows
networking of number mapping.
Figure 2-19 Networking of number mapping
User D
Number: 28980808
PBX1 PBX2
Calling party: 28980808

Called party: 28987100

Short code: 7000 Short code: 7001 Short code: 7100
Mapped number: Mapped number: Mapped number:
28987000 28987001 28987100

NetEngine AR
Prerequisites
● A user number has been configured. For details on how to configure a user
number, see 2.8.1 Configuring a POTS User.
● A trunk group has been created and its connection parameters have been
configured. For details about the configuration, see 2.11 Configuring Trunk
Groups.
● The long codes allocated by the PSTN to PBX users are 28987000, 28987001,
and 28987100.
Procedure

system view.

view.
3 Enter the number-map number-map- -

number name
mapping view.
4 Configure internal-num internal-num -

internal
numbers.
5 Configure external-num external-num -

external
numbers.
6 Configure an enterprise enterprise-name If the intra-office

enterprise. user belongs to the
default enterprise,
the enterprise
parameter does not
need to be set;
otherwise, set the
enterprise
parameter to the
enterprise to which
the user belongs.
7 Enter the voice quit -

view.
8 Enter the call callprefix callprefix-name -

prefix profile
view.

NetEngine AR
9 Enable number special-deal { caller-map long- -

mapping. cli-flag | called-map cld-map-
flag | long-caller long-cli-flag } *
10 Save the save -

configuration.
Example
Map 7000 and 7001 to 28987000 and 28987001 of enterprise default respectively,
and enable called number mapping for intra-office call prefix 2 and calling
number mapping for inter-office call prefix 9.
[Huawei] voice
[Huawei-voice] number-map 0
[Huawei-voice-number-map-0] internal-num 7000
[Huawei-voice-number-map-0] external-num 28987000
[Huawei-voice-number-map-0] quit
[Huawei-voice] number-map 1
[Huawei-voice-number-map-1] internal-num 7001
[Huawei-voice-number-map-1] external-num 28987001
[Huawei-voice-number-map-1] quit
[Huawei-voice-callprefix-2] special-deal called-map enable
[Huawei-voice-callprefix-9] special-deal caller-map enable
Verification
Verify display voice number-map [Huawei]display voice number-

the map
number Spec: 250, Records: 2
------------------------
mapping Internal-num value: 7000
configur External-num value: 28987000
ation. Enterprise name : default
Internal-num value: 7001

External-num value: 28987001
Enterprise name : default
Verify display voice callprefix The values of Called map switch

the call callprefix-name for call prefix 7 and Caller map
prefix switch for call prefix 9 are
configur consistent with settings.
ation.

NetEngine AR
2.10.4 Configuring a Pre-routing Number Change Plan

Before route selection, complete calling or called number change.
Context
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 a PBX makes a
local call by dialing 0755 28961000. The configured call route connects local
outgoing calls with call prefix 2896. Therefore, a pre-routing number change plan
needs to be configured to remove 9 from the called number. Then the number is
re-analyzed. You can configure the PBX to delete a call prefix in the pre-routing
number change plan so that the PBX can accurately locate the called party. In this
plan, the calling or called number is changed after number analysis and before
route selection. After a number changes, the PBX re-analyzes the number and
plays the two-stage dial tone.
Prerequisites
● An inter-office call prefix has been configured. For details on how to configure
an inter-office call prefix, see 2.9 Configuring a Call Prefix.
Procedure
p

system view.
2 Enter the voice -

voice view.
3 Create a pre- beforeroute-change name By default, no

routing pre-routing
number number
change plan change plan is
and enter the configured.
pre-routing
number
change plan
view.
4 Bind a call callprefix callprefix-name -

prefix to the
pre-routing
number
change plan.

NetEngine AR

p
5 Configure a caller { del-then-insert del-offset del- -

calling len insert-telnum | del del-offset del-len
number | insert insert-offset insert-telnum | no-
change rule. change }
6 Configure a called { del-then-insert del-offset del- -

called len insert-telnum | del del-offset del-len
7 (Optional) call new-attribute { enterprise -

Configure an enterprise-name | dn-set dn-set-name }
enterprise *
and a DN set
in the pre-
routing
number
change plan.
8 (Optional) call re-analyse reanalyze-value The PBX

Configure the supports re-
PBX whether analysis for a
to re-analyze maximum of
changed five times.
numbers.
9 (Optional) call dual-dial-tone-flag dual-dial-tone- To enable the

Configure the flag-value PBX to play
PBX whether the two-stage
to play the dial tone,
two-stage delete the
dial tone. dialed number.
10 (Optional) condition caller-telno telno-value By default, no

Configure a caller number
calling is configured
number for a for a pre-
pre-routing routing
number change plan.
change plan.

NetEngine AR

p
11 (Optional) condition called-telno telno-value The prefix of

Configure a the called
called number must
number that be the same
needs to be as the prefix
changed configured in
before route step 4.
selection. By default, no
called number
is configured
for a pre-
routing
change plan.
12 Save the save -

configuration.
Example
Bind call prefix 9 to pre-routing number change plan 0, and configure the PBX to
map calling numbers to 28980808 and to delete the first digit from called
numbers.
[Huawei] voice
[Huawei-voice] beforeroute-change 0
[Huawei-voice-beforeroute-change-0] callprefix 9
[Huawei-voice-beforeroute-change-0] caller del-then-insert 1 32 28980808
[Huawei-voice-beforeroute-change-0] called del 1 1
[Huawei-voice-beforeroute-change-0] save
Verification
Verify the display voice beforeroute-change The parameter values

pre- [ name ] in the command output
routing are consistent with the
number settings.
change
plan
configura
tion.
2.10.5 Configuring a Post-routing Number Change Plan

After route selection, complete calling or called number change.

NetEngine AR
Context
A post-routing number change plan 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 code to ensure that it complies
with the required number format. For example, a POTS user (using the number
7000) connected to a PBX makes a national toll call by dialing 057128980000. A
post-routing number change plan adds 12523 to the called number
057128980000. 12523 is the call prefix defined by the carrier for the enterprise.
When the carrier's device detects call prefix 12523, it connects the outgoing call
through the matching trunk. This reduces the call fees of the enterprise. You can
configure the PBX to delete a call prefix in the post-routing number change plan
so that the PBX can accurately locate the called party.
Prerequisites
● An inter-office call prefix has been configured. For details on how to configure
an inter-office call prefix, see 2.9 Configuring a Call Prefix.
● An inter-office trunk group has been configured. For details on how to
configure an inter-office trunk group, see 2.11 Configuring Trunk Groups.
Procedure
p

system view.

view.
3 Create a post- afterroute-change name By default, no

routing number post-routing
change plan number
and enter the change plan
post-routing is configured.
number change
plan view.

prefix to the
post-routing
number change
plan.
5 Bind a trunk trunk-group trunk-group-name -

group to the
post-routing
number change
plan.

NetEngine AR

p
6 Configure a caller { del-then-insert del-offset -

calling number del-len insert-telnum | del del-offset
change rule. del-len | insert insert-offset insert-
telnum | no-change }
7 Configure a called { del-then-insert del-offset -
called number del-len insert-telnum | del del-offset
change rule. del-len | insert insert-offset insert-
telnum | no-change }
8 (Optional) condition caller-telno telno-value -
Configure a
calling number
for a post-
routing number
change plan.
9 Save the save -

configuration.
Example
Bind post-routing number change plan 0 to inter-office call prefix 8 and SIP IP
trunk group, and configure the PBX to map calling numbers to 28980808 and to
add 12523 to called numbers.
[Huawei] voice
[Huawei-voice] afterroute-change 0
[Huawei-voice-afterroute-change-0] callprefix 8
[Huawei-voice-afterroute-change-0] trunk-group sipip
[Huawei-voice-afterroute-change-0] caller del-then-insert 1 32 28980808
[Huawei-voice-afterroute-change-0] called insert 1 12523
[Huawei-voice-afterroute-change-0] save
Verification
Verify the display voice afterroute-change The parameter values in the

post- [ name ] command output are
routing consistent with the settings.
number
change
plan
configurati
on.

NetEngine AR
2.10.6 Configuring GNR Number Negotiation

The device functioning as the trunk gateway, following the GNR negotiation rules,
can meet the hunt group service requirements in the scenario of tandem calls to
the iMSS through trunks.
Context
In a voice project, the existing TDM PBX needs to connect to the iMSS through
trunks for bidirectional trunk-based tandem calls. Figure 2-20 shows the network
diagram.
Figure 2-20 Network diagram

TDM PBX PBX
PRA/BRA SIP
iMSS
Analog phone Peer phone
To meet the hunt service requirements on the iMSS, with regard to trunk-based
tandem calls from the TDM PBX (narrowband side) to the iMSS (broadband side),
also ensure that calling numbers in SIP messages can be recognized on the
broadband side. To achieve so, the trunk gateway uses the GNR list to check
whether the calling numbers can be recognized on the broadband side. Table 2-4
describes the negotiation rules.
Table 2-4 GNR number negotiation rules
Result of Calling SIP Message (From Field) SIP Message (PAI Field)
Number
Matching
Calling number in From: <SIP:Calling P-Asserted-Identity:

the GNR list number@Calling <SIP:Calling
address:Port> number@Calling
address:Port>
Calling number From: P-Asserted-Identity:
not in the GNR <SIP:GNRNumber@Calling <SIP:GNRNumber@Calling
list address:Port> address:Port>
Anonymous call From: P-Asserted-Identity:
<anonymous@anonymous.i <SIP:GNRNumber@Calling
nvalid> address:Port>

NetEngine AR
NOTE
The trunk gateway uses the GNR list to check calling numbers, and this list contains several full
numbers that can be recognized on the broadband side.
The GNR number uniquely identifies a GNR list. When matching fails, the GNR number is sent
to the broadband side as the default calling number, and the GNR number also must be
recognized on the broadband side.
Prerequisites
The following data has been configured on the device for implementing basic
bidirectional trunk-based tandem calling:
● Digital trunks (PRA/BRA) to the TDM PBX and SIP trunks to the iMSS
● Data (such as DN set) of trunks to the TDM PBX and to the iMSS
Configuration Procedure
3 Create a GNR list. gnr-number gnr-number- The gnr-number-

value value parameter
uniquely identifies
a GNR list, and its
value is also the
default calling
number entered in
the SIP message
when matching
fails.
The system
supports a
maximum of 30
GNR lists.
4 Add full numbers in full-number full-number- A GNR list can

the GNR list. value [ number number- contain a
value steplen steplen- maximum of 50
value ] full numbers.
5 Return to the voice quit -

view.
6 Enter the trunk group trunk-group trunk-group- Bind the same

view. name GNR list to the SIP
trunk group
7 Bind the GNR list to gnr-number gnr-number- destined for the
the trunk groups. value [ num count [ step- iMSS and
lenth step-lenth-value ] ] PRA/BRA trunk

NetEngine AR
8 Configure GNR number-parameter group destined for

numbers for trunk numpara-no numpara- the TDM PBX.
groups. value NOTE
● Each trunk
9 Return to the voice quit group can be
view. bound to 30
GNR lists.
● If multiple GNR
lists are
configured on
the narrowband
side and calling
numbers of
incoming calls
all fail to be
matched, the
device sends the
GNR number of
the GNR list
that is first
created to the
broadband side.
As for trunks on
broadband side,
configure GNR
numbers based on
the trunk type by
referring to Table
2-5.
10 Save the configuration. save -
Table 2-5 Related GNR

Broadband GNR Value
Trunk Type
125 126 127
SIP IP 2 default default
SIP AT0 2 0 0
SIP PRA 2 default 0

NetEngine AR
Broadband GNR Value

Trunk Type
125 126 127
Description When an If the trunk on the If the trunk on the

anonymous call broadband side is broadband side is
comes into an SIP AT0, ensure SIP AT0 or SIP PRA
office from the that calling trunk group, ensure
narrowband side numbers received that the Invite
and goes out of the on the iMSS side message carries the
office from the are not forcibly PAI domain rather
broadband side, changed to than PPI domain.
ensure that the allocated
From domain registration
content meets numbers.
requirements in
Table 2-4:
From:
<anonymous@anonymou
s.invalid>
Configuration Example
Create GNR list 123456, add numbers 1234567890 to 1234567894 as full numbers
to the GNR list, and bind the GNR list to trunk groups pra1 and sip1.
[Huawei] voice
[Huawei-voice] gnr-number 123456
[Huawei-voice-gnrnumber-123456] full-number 1234567890 number 5 steplen 1
[Huawei-voice-gnrnumber-123456] quit
[Huawei-voice] trunk-group pra1
[Huawei-voice-trunkgroup-pra1] gnr-number 123456
[Huawei-voice-trunkgroup-pra1] quit
[Huawei-voice] trunk-group sip1
[Huawei-voice-trunkgroup-sip1] gnr-number 123456
[Huawei-voice-trunkgroup-sip1] number-parameter 125 2
[Huawei-voice-trunkgroup-sip1] quit
[Huawei-voice] save
Usage Example
Query GNR display voice gnr-number [ GNR- Parameter values in the

configuratio number-value ] command output are
n the same as the preset
information. ones during data
configuration.
Query the display voice trunk-group [ trunk- The GNR list bound to
GNR list group-name [ para-value ] ] the trunk groups in the
bound to command output is the
the trunk desired one.
groups.

NetEngine AR
2.11 Configuring Trunk Groups

A trunk group is used to transmit signaling and media data between central office,
and is used to constitute a route. In TDM scenarios, a trunk group is a set of trunk
circuits with the same attributes. Generally, a group of trunk circuits with the same
attributes in the same direction are called a trunk group.
2.11.1 Configuring an AT0 Trunk Group

An AT0 trunk group implements voice communication between PBX users and
inter-office users. It is an analog circuit trunk and uses the FXO interface of the
PBX to connect to the PSTN through a common telephone line.
Context
Enterprises request PSTN telephone numbers of a certain number, and use
common telephone lines as AT0 trunk lines so that enterprise users can share the
trunk lines. Outgoing calls occupy one trunk line, and the PSTN telephone number
is displayed. After calls are ended, the trunk line is released. Incoming calls reach
the PBX through the AT0 trunk, and then the PBX forwards the incoming calls to a
PBX user or access number of the enterprise switchboard. The trunk line use
efficiency is high, and enterprises do not need to request independent PSTN
telephone numbers for all enterprise employees.
An AT0 trunk uses FXO interfaces, so the PBX must be configured with the voice
card such as the 4FXS1FXO or 4FXO card providing FXO interfaces.
The PBX can add several trunks of the same type to a trunk group, which are
invoked by call routes. Even if there is only one trunk, a trunk group needs to be
configured to facilitate trunk management.
The PBX connects to the PSTN through an AT0 trunk group, as shown in Figure
2-21.
Figure 2-21 Connecting the PBX to the PSTN through an AT0 trunk group
AT0 trunk
group
PBX
POTS Fax ISDN IP

phone machine phone phone

NetEngine AR
Procedure

view.

view.
3 Create an AT0 trunk-group name If an AT0 trunk group

trunk group and [ fxo ] already exists, this
enter the AT0 command directly displays
trunk group view. the AT0 trunk group view.

NetEngine AR
4 Bind a trunk to trunk-at0 slotid/ The default-called-telno

the AT0 trunk subcardid/portid parameter is mandatory.
group. [ default-called-telno You can specify default-
default-called-telno- called-telno to transmit
value | caller-telno incoming calls through a
[ enterprise trunk to an intra-office
enterprise-name ] user (dedicated access) or
caller-telno-value | cid- access number of the
transmit-type cid- switchboard (switchboard
transmit-type-value | access).
config-status config- By configuring caller-
status-value | dial- telno, you can specify a
delay dial-delay-value | number for outgoing calls
dial-mode dial-mode- through the AT0 trunk.
value | insert-prefix Only the outgoing calls
insert-prefix-value | that are made using the
insert-prefix-delay number can be routed
insert-prefix-delay- through the AT0 trunk.
value ] * When the AT0 is
unavailable, the outgoing
calls that are made using
the number specified by
caller-telno can be routed
through other trunks.
NOTICE
● If dial-mode is set to
pulse, run the pbx
number-parameter
command in the voice
view to change the value
of control point 40 to 5
after completing the
configuration of the AT0
trunk group. If this
control point value is not
changed, pulse dialing
fails on the AT0 trunk.
The default value of dial-
mode is dtmf, and the
default value of control
point 40 is 0.
V300R019C10 and later
versions do not support
pulse dialing.
● When the 4FXS1FXO
board functions as an
AT0 trunk board, the port
number must be set to 4;
otherwise, the
configuration fails. For
other boards, the actual
port in use prevails.

NetEngine AR
5 (Optional) enterprise enterprise- By default, an AT0 trunk

Configure the name [ dn-set dn-set- group belongs to the
enterprise and name ] enterprise default and DN
DN set to which set DefaultDialPlan.
the AT0 trunk You need to perform this
group belongs. step only when multiple
enterprises and DN sets
are configured. For details
on how to configure an
enterprise and a DN set,
6 (Optional) select-mode The following circuit

Configure a selectmode-value selection modes are
trunk group available:
circuit selection ● loop: polling mode
mode.
● increase: ascending
mode
● decrease: descending
mode
● port-loop: ascending
mode. (In the case of
call failure, re-select a
time slot on the same
port in ascending
order.)
● port-increase:
ascending mode. (In
the case of call failure,
re-select a port in
ascending order.)
The default value is loop.
7 (Optional) default-caller-telno -
Configure the telno-value
default number
displayed.
8 (Optional) description desc-value -

Configure the
description for
the AT0 trunk
group.
9 Save the save -

configuration.

NetEngine AR
Example
# Create an AT0 trunk group named at0a, bind the AT0 trunk connected to FXO
interface 4/0/4 to the AT0 trunk group, and set the default called number to the
access number 7200.
[Huawei] voice
[Huawei-voice] trunk-group at0a fxo
[Huawei-voice-trunkgroup-at0a] trunk-at0 4/0/4 default-called-telno 7200
[Huawei-voice-trunkgroup-at0a] save
Verification
Verify the AT0 display voice trunk-group The parameter values in the
trunk group [ name [ para-value ] ] command output are
configuration. consistent with the settings.
2.11.2 Configuring SIP Trunk Groups

The PBX supports three types of SIP trunk groups: SIP AT0, SIP PRA, and SIP IP.
2.11.2.1 Configuring a SIP AT0 Trunk Group

A SIP AT0 trunk group implements voice communication between PBX users and
inter-office users. A SIP AT0 trunk connects the PBX to the carrier network by
forging a SIP user. SIP users can register on the carrier network separately.
Background
After an enterprise applies for SIP users from the carrier, configure a SIP AT0 trunk
group to implement voice communication between PBX users and inter-office
users. A SIP AT0 trunk group, similar to an AT0 trunk group based on POTS users,
works over the IP network and uses SIP.
The SIP users are separately registered on the carrier network such as the IMS
network. Then trunks of the SIP AT0 trunk group are formed. The carrier network
connects to common SIP users, and does not learn about the private network. An
enterprise user occupies one trunk of the SIP AT0 trunk group to make outgoing
calls. Incoming calls are transmitted to the enterprise switchboard or an enterprise
user.
By default, a SIP AT0 trunk allocated by the carrier network to the PBX cannot
support multiple calls at the same time. For example, when a SIP AT0 trunk is
used for the automatic switchboard, multiple calls cannot be set up using the
automatic switchboard. If a SIP AT0 trunk group needs to support multiple
concurrent calls, configure the SIP AT0 multi-call service. For details, see trunk-
sipat0.

NetEngine AR
Table 2-6 describes the transport protocols that can be used by a SIP AT0 trunk.
Table 2-6 Transport protocols that can be used by a SIP AT0 trunk
Transport Description
Protocol
UDP Connectionless transport-layer protocol that provides event-based,

simplified, and unreliable information transmission. When UDP is
used to transmit data, each datagram is independent and contains
the source and destination. Because UDP is unreliable, data may
not reach the destination, and the time data reaches the
destination and data integrity cannot be ensured.
TCP Connection-oriented protocol. Before data transmission, both ends

must establish a virtual channel.
The PBX connects to the IMS network through a SIP AT0 trunk group, as shown in
Figure 2-22.
Figure 2-22 Connecting the PBX to the IMS network through a SIP AT0 trunk
group
IMS
network
SIP AT0
trunk group
PBX
POTS Fax ISDN IP

Procedure

view.

NetEngine AR
2 (Optional) Create ssl policy policy-name ● When the AR

an SSL policy [ type { client | functions as the SIP
server } ] server and the
security of voice
signal transmission
needs to be
enhanced, you can
run this command
to configure an SSL
policy of server type
on the AR and
configure an SSL
on the remote
device. Data
authentication, and
integrity
authentication
mechanisms of SSL
ensure data
transmission
security.
● When the AR
functions as the SIP
server and the
security of voice
signal transmission
needs to be
enhanced, you can
run this command
to configure an SSL
on the AR and
configure an SSL
policy of server type
on the remote
device. Data
authentication, and
integrity
authentication
mechanisms of SSL
ensure data
transmission
security.

view.

NetEngine AR
4 (Optional) cause-mapping { isdn- -

Configure the cause isdn-cause-value |
mappings between sip-response sip-
ISDN cause values response-value } *
and SIP error codes.
5 Create a SIP AT0 trunk-group trunk- If a SIP AT0 trunk

trunk group and group-name [ sip group already exists,
enter the SIP AT0 trunk-circuit ] this command directly
trunk group view. displays the SIP AT0
trunk group view.
6 Configure the signalling-address { ip -

signaling IP address ip-address | addr-name
and port number at signal-addr-name-
the local end. value } port port-value
7 Configure the media-ip { ip-address | -
media IP address or addr-name addr-name-
dynamic media IP value }
address name at
the local end.
8 Configure the peer-address static Use the static mode.

remote IP address primary-ip-address The data of the trunk
and port number in primary-port-value group must be the
static or DNS [ secondary secondary- same as the data at the
mode. ip-address secondary- remote end.
port-value ]
peer-address { dns-a Use the DNS mode.
primary-domain-value The data of the trunk
primary-port-value group must be the
[ secondary secondary- same as the data at the
domain-value remote end.
secondary-port-value ] |
dns-srv domain-value | If the DNS mode is
dns-naptr domain- used, the SIP AT0 trunk
value } group can be redirected
for registration. The
registration redirection
process of a SIP AT0
trunk group is the
same as that of a SIP
AG. For details, see SIP
AG Registration
Redirection Process.

NetEngine AR
9 Configure the home-domain value Set the parameter

home domain to based on the
which the SIP IP requirements of the
trunk group of the remote end.
peer device
belongs.
10 Configure a register-uri register-uri- Set the parameter

registrar URI. value based on the
requirements of the
remote end.
11 Bind a trunk to a trunk-sipat0 register-id You can specify

SIP AT0 trunk { password cipher default-called-telno to
group. cipher-password | transmit incoming calls
caller-telno over a trunk to an
[ enterprise enterprise- intra-office user
name ] telno-value | (dedicated access) or
default-called-telno access number of the
default-called-telno- switchboard
value | max-calls max- (switchboard access).
calls-value } * If register-id is set to
esn, the system
automatically replaces
the esn with the actual
ESN.
12 Configure the user number-parameter 19 If the user information

name format for numpara-value provided by the IMS for
authentication. registration is a
number, such as
+865717000, set this
parameter to 0. If the
user information is in
User number@Domain
name format, such as
+865717000@abcd.co
m, set this parameter
to 1. Set this parameter
to 2 for other formats.
Enumerated type. The
options are as follows:
● 0: Userinfo
● 1: Userinfo@domain
● 2: User-defined
The default value is
0.

NetEngine AR
13 Reset the SIP AT0 reset You must reset the SIP
trunk group. AT0 trunk group to
make new or modified
parameters of the SIP
AT0 trunk group take
effect.
14 (Optional) enterprise enterprise- By default, a SIP IP

Configure the name [ dn-set dn-set- trunk group belongs to
enterprise and DN name ] the enterprise default
set that the SIP AT0 and DN set
trunk group DefaultDialPlan.
belongs to. You need to perform
this step only when
multiple enterprises
and DN sets are
configured. For details
enterprise and a DN
set, see 2.5.4
Configuring an
Enterprise and a DN
Set.
15 (Optional) transfer transfer-value By default, a transport

Configure a protocol for the SIP
transport protocol trunk group is UDP.
for the SIP trunk
group.

NetEngine AR
16 (Optional) cs-mode mode-value When the transfer

Configure the mode for a SIP server is
reliable connection TCP or TLS, this
mode. command needs to be
configured.
● When the AR
functions as the
server, the reliable
connection mode for
the SIP trunk group
need to be set to
server or
client_server.
● When the AR
functions as the
client, the reliable
connection mode for
the SIP trunk group
need to be set to
client or
client_server.
17 (Optional) ● the SSL policy of ● When the AR

Configure an SSL server type: ssl- functions as the
policy of server server-policy server- server and is bound
type or client type policy-name to an SSL policy of
when a trunk group ● the SSL policy of server type, the
uses TLS. client type: ssl- remote device needs
client-policy server- to be configured
policy-name with an SSL policy of
client type.
● When the AR
functions as the
client and is bound
to an SSL policy of
client type, the
remote device needs
to be configured
with an SSL policy of
server type.

NetEngine AR
18 (Optional) select-mode select- The following circuit

Configure a trunk mode-value selection modes are
group circuit available:
selection mode. ● loop: polling mode
mode
● decrease:
descending mode
mode. (In the case
of call failure, re-
select a time slot on
the same port in
ascending order.)
● port-increase:
ascending mode. (In
the case of call
failure, re-select a
port in ascending
order.)
loop.
19 (Optional) country-code country- This command

Configure the code-value area-code configures the country
default country area-code-value code and area code for
code and area incoming calls over a
code. SIP AT0 trunk group.
When incoming calls
do not carry the
country code and area
code, the PBX adds the
configured country
code and area code so
that intra-office users
can view calls and call
back through call
display.

Configure the
description for the
SIP AT0 trunk
group.

NetEngine AR
22 (Optional) circuit-state-relate This command

Configure trunk trunk-group-name associates other trunk
groups that are groups with the SIP
associated with the AT0 trunk group. When
SIP AT0 trunk the status of the SIP
group. AT0 trunk group
changes, status of the
associated trunk groups
also changes.
The SIP AT0 trunk
group can be
associated only with
PRA trunk groups or
BRA trunk groups.
By default, the SIP AT0
trunk group is not
associated with any
trunk group.
23 Save the save -

configuration.
Example
Configure a SIP AT0 trunk group named sipat0 that connects the enterprise and
IMS network of the carrier, and set the signaling and media IP addresses and
signaling port number at the local end to 192.168.1.3 and 5061 respectively. The
SIP user number allocated by the carrier is 56623000, and the authentication
password is 123456. When an external user makes a call to 56623000, the phone
of user A (phone number: 7222) rings. If an external user wants to make a call to
an intra-office user other than user A, the external user can make a call to user A
and ask user A to transfer the call to the target user. Set the IP address and port
number of the IMS network to 192.168.10.10 and 5060 respectively. Set the user
name format for authentication to 0 (Userinfo).
[Huawei] voice
[Huawei-voice] trunk-group sipat0 sip trunk-circuit
[Huawei-voice-trunkgroup-sipat0] signalling-address ip 192.168.1.3 port 5061
[Huawei-voice-trunkgroup-sipat0] media-ip 192.168.1.3
[Huawei-voice-trunkgroup-sipat0] peer-address static 192.168.10.10 5060
[Huawei-voice-trunkgroup-sipat0] register-uri abcd.com
[Huawei-voice-trunkgroup-sipat0] home-domain abcd.com
[Huawei-voice-trunkgroup-sipat0] trunk-sipat0 56623000 default-called-telno 7222 password cipher
Please input user password(1-32 chars):******
[Huawei-voice-trunkgroup-sipat0] number-parameter 19 0
[Huawei-voice-trunkgroup-sipat0] reset
[Huawei-voice-trunkgroup-sipat0] save

NetEngine AR
Verification
Check the SIP display voice trunk-group The parameter values in the
AT0 trunk [ name [ para-value ] ] command output are
group consistent with the settings.
configuration.
2.11.2.2 Configuring a SIP PRA Trunk Group

A SIP PRA trunk group implements voice communication between PBX users and
inter-office users. A SIP PRA trunk connects the PBX to the carrier network by
forging a SIP user. SIP users can register on the carrier network separately.
Context
After an enterprise applies for SIP users from the carrier, configure a SIP PRA trunk
group to implement voice communication between PBX users and inter-office
users.
Unlike a SIP AT0 trunk group, a SIP PRA trunk group uses trunk group registration.
That is, the SIP PRA trunk group sends a registration message to complete number
registration of a group of SIP users. Then trunks of the SIP PRA trunk group are
formed. The carrier network connects to common SIP users, and does not learn
about the private network. An enterprise user occupies one trunk of the SIP PRA
trunk group to make outgoing calls. Incoming calls are transmitted to the
enterprise switchboard or an enterprise user.
Table 2-7 describes the transport protocols that can be used by a SIP PRA trunk.
Table 2-7 Transport protocols that can be used by a SIP PRA trunk
Protocol


The PBX connects to the IMS network through a SIP PRA trunk group, as shown in
Figure 2-23.

NetEngine AR
Figure 2-23 Connecting the PBX to the IMS network through a SIP PRA trunk
group
IMS
network
SIP PRA
trunk group
PBX
POTS Fax ISDN IP

Procedure

view.

view.
3 (Optional) cause-mapping { isdn- -

Configure the cause isdn-cause-value
mappings | sip-response sip-
between ISDN response-value } *
cause values and
SIP error codes.
4 Create a SIP PRA trunk-group trunk- If a SIP PRA trunk group

trunk group and group-name [ sip already exists, this
enter the SIP trunk-group ] command directly displays
PRA trunk group the SIP PRA trunk group
view. view.
5 Configure the signalling-address { ip -

signaling IP ip-address | addr-name
address and port signal-addr-name-
number at the value } port port-value
local end.

NetEngine AR
6 Configure the media-ip { ip-address | -

media IP address addr-name addr-
or dynamic name-value }
media IP address
name at the
local end.
7 Configure the peer-address static Use the static mode.

remote IP primary-ip-address The data of the trunk
address and port primary-port-value group must be the same
number in static [ secondary secondary- as the data at the remote
or DNS mode. ip-address secondary- end.
port-value ]
primary-port-value group must be the same
[ secondary secondary- as the data at the remote
domain-value end.
secondary-port-value ]
| dns-srv domain-value If the DNS mode is used,
| dns-naptr domain- the SIP PRA trunk group
value } can be redirected for
registration. The
registration redirection
process of a SIP PRA trunk
group is the same as that
of a SIP AG. For details,
see SIP AG Registration
Redirection Process.
8 Configure the home-domain value Set the parameter based

home domain to on the requirements of the
which the SIP IP remote end.
trunk group of
the peer device
belongs.
9 Configure a register-uri register- Set the parameter based

registrar URI. uri-value on the requirements of the
remote end.

NetEngine AR
10 Configure a register-id register-uri- A SIP PRA trunk uses the

group value [ password group registration mode,
registration ID cipher password- and you must configure a
and value ] group registration ID and
authentication authentication password
password. for it.
The group registration ID
and authentication
password need to be
obtained from the service
provider, for example,
carrier of the IMS network.
11 Specifies the number-parameter 19 Enumerated type. The

user name num-para options are as follows:
format for user ● 0: Userinfo
authentication
for the trunk ● 1: Userinfo@domain
group. ● 2: User-defined
The default value is 0.
12 Specifies the number-parameter 33 Enumerated type. The

mode of num-para options are as follows:
checking the ● 0: OPTIONS
proxy server for
the trunk group. ● 1: Heartbeat detection
in probe mode
● 2: REGISTER
The probe mode takes
effect only when the IMS
Core dual-homing function
is not supported. If the
server is configured to
support the IMS Core dual-
homing function, the
OPTIONS and REGISTER
modes are used. The probe
mode is the default mode
in a common process. The
REGISTER mode is the
default mode in the DT
process.
13 Reset the SIP reset You must reset the SIP PRA
PRA trunk group. trunk group to make new
or modified parameters of
the SIP PRA trunk group
take effect.

NetEngine AR
14 (Optional) digest-username If no authentication digest

Configure the digest-username-value user name is configured,
authentication the trunk group sends the
digest user encrypted registration ID
name. to the peer device for
authentication.
15 (Optional) enterprise enterprise- By default, a SIP IP trunk

Configure the name [ dn-set dn-set- group belongs to the
enterprise and name ] enterprise default and DN
DN set that the set DefaultDialPlan.
SIP PRA trunk You need to perform this
group belongs step only when multiple
to. enterprises and DN sets
are configured. For details
enterprise and a DN set,
16 (Optional) transfer transfer-value The PBX supports the

Configure a following transport
transport protocols for a SIP trunk
protocol for the group: UDP (default), and
SIP trunk group. TCP.
17 (Optional) cs-mode mode-value Configure this parameter

Configure the when the transport
reliable protocol of the trunk
connection group is TCP. The reliable
mode. connection modes include
client, server, and
client_server.

NetEngine AR
18 (Optional) select-mode select- The following circuit

Configure a mode-value selection modes are
trunk group available:
circuit selection ● loop: polling mode
mode.
mode
● decrease: descending
mode
time slot on the same
port in ascending
order.)
● port-increase: ascending
port in ascending
order.)
The default value is loop.
19 (Optional) country-code country- This command configures

Configure the code-value area-code the country code and area
default country area-code-value code for incoming calls
code and area over a SIP PRA trunk
code. group. When incoming
calls do not carry the
country code and area
code, the PBX adds the
configured country code
and area code so that
intra-office users can view
calls and call back through
call display.

Configure the
description for
the SIP PRA
trunk group.

NetEngine AR
21 (Optional) circuit-state-relate This command associates

Configure trunk trunk-group-name other trunk groups with
groups that are the SIP PRA trunk group.
associated with When the status of the SIP
the SIP PRA PRA trunk group changes,
trunk group. status of the associated
trunk groups also changes.
The SIP PRA trunk group
can be associated only
with PRA trunk groups or
BRA trunk groups.
By default, the SIP PRA
trunk group is not
associated with any trunk
group.
22 Save the save -

configuration.
NOTE
On the PBX, the same sipue user as the one on the IMS needs to be added, and this user
needs to register with the PBX.
Example
An enterprise connects to the IMS network of a carrier through a SIP PRA trunk
group named sippra, and the group registration ID is 81234567. The signaling and
media IP addresses are both 192.168.1.3, and the signaling port number is 5062.
The IP address of the IMS network is 192.168.10.10, and the port number is 5060.
The home domain and registrar URI of the peer trunk group are both abcd.com.
[Huawei] voice
[Huawei-voice] trunk-group sippra sip trunk-group
[Huawei-voice-trunkgroup-sippra] signalling-address ip 192.168.1.3 port 5062
[Huawei-voice-trunkgroup-sippra] media-ip 192.168.1.3
[Huawei-voice-trunkgroup-sippra] peer-address static 192.168.10.10 5060
[Huawei-voice-trunkgroup-sippra] home-domain abcd.com
[Huawei-voice-trunkgroup-sippra] register-uri abcd.com
[Huawei-voice-trunkgroup-sippra] register-id 81234567 password cipher
[Huawei-voice-trunkgroup-sippra] reset
[Huawei-voice-trunkgroup-sippra] save

NetEngine AR
Verification
Verify the SIP display voice trunk-group The parameter values in the
PRA trunk [ name [ para-value ] ] command output are
configuration.
2.11.2.3 Configuring a SIP IP Trunk Group

A SIP IP trunk group implements voice communication between PBX users and
inter-office users. A SIP IP trunk is a packet trunk that uses SIP based on IP.
Context
The PBX and the peer device connected by a SIP IP trunk have equivalent
positions. That is, the PBX at one end of a SIP IP trunk does not need to register
with the device at the other end. Unlike a circuit trunk that defines a physical
channel, a SIP IP trunk defines a logical channel and solves authentication and
addressing problems between local and remote offices.
When you configure a SIP IP trunk for the PBX, the remote end must be the device
supporting SIP IP trunks.
Table 2-8 describes the transport protocols that can be used by a SIP IP trunk.
Table 2-8 Transport protocols that can be used by a SIP IP trunk
Protocol


The PBX connects to the remote device through SIP IP trunk groups, as shown in
Figure 2-24.

NetEngine AR
Figure 2-24 Networking of SIP IP trunk groups
PBX SIP IP IP PBX

trunk group
POTS Fax ISDN IP

Procedure
3 (Optional) Configure cause-mapping -

the mappings between { isdn-cause isdn-
ISDN cause values and cause-value | sip-
SIP error codes. response sip-
response-value } *
4 Create a SIP IP trunk trunk-group trunk- If a SIP IP trunk group
group and enter the group-name [ sip no- already exists, this
SIP IP trunk group register ] command directly
view. displays the SIP IP
trunk group view.
5 Configure the signalling-address -

signaling IP address { ip ip-address | addr-
and port number at name signal-addr-
the local end. name-value } port
port-value
6 Configure the media media-ip { ip-address -
IP address or dynamic | addr-name addr-
media IP address name-value }
name at the local end.
7 Configure the remote peer-address static Use the static mode.

IP address and port primary-ip-address The data of the trunk
number in static or primary-port-value group must be the
DNS mode. [ secondary same as the data at
secondary-ip-address the remote end.
secondary-port-value ]

NetEngine AR

primary-port-value group must be the
[ secondary same as the data at
secondary-domain- the remote end.
value secondary-port-
value ] | dns-srv
domain-value | dns-
naptr domain-value }
8 Configure the home home-domain value Set the parameter

domain to which the based on the
SIP IP trunk group of requirements of the
the peer device remote end.
belongs.
9 Configure a registrar register-uri register- Set the parameter

URI. uri-value based on the
requirements of the
remote end.
10 Reset the SIP IP trunk reset You must reset the

group. SIP IP trunk group to
make new or
modified parameters
of the SIP IP trunk
group take effect.
11 (Optional) Configure enterprise enterprise- By default, a SIP IP

the enterprise and DN name [ dn-set dn-set- trunk group belongs
set that the SIP IP name ] to the enterprise
trunk group belongs default and DN set
to. DefaultDialPlan.
You need to perform
this step only when
and DN sets are
on how to configure
an enterprise and a
DN set, see 2.5.4
Configuring an
Enterprise and a DN
Set.
12 (Optional) Configure a transfer transfer-value The PBX supports the

transport protocol for following transport
the SIP trunk group. protocols for a SIP
trunk group: UDP
(default) and TCP.

NetEngine AR
13 (Optional) Configure cs-mode mode-value Configure this

the reliable connection parameter when the
mode. transport protocol of
the trunk group is
TCP. The reliable
connection modes
include client, server,
and client_server.
14 (Optional) Configure a select-mode select- The following circuit

trunk group circuit mode-value selection modes are
selection mode. available:
● loop: polling mode
mode
● decrease:
descending mode
● port-loop:
ascending mode.
(In the case of call
time slot on the
same port in
ascending order.)
● port-increase:
ascending mode.
port in ascending
order.)
loop.
15 (Optional) Configure country-code This command

the default country country-code-value configures the
code and area code. area-code area-code- country code and area
value code for incoming
calls over a SIP IP
trunk group. When
incoming calls do not
carry the country
code and area code,
the PBX adds the
configured country
back through call
display.

NetEngine AR
16 (Optional) Configure description desc-value -

the description for the
SIP IP trunk group.
17 (Optional) Configure circuit-state-relate This command

trunk groups that are trunk-group-name associates the SIP IP
associated with the SIP trunk group with
IP trunk group. other trunk groups.
When the status of
the SIP IP trunk group
changes, status of the
associated trunk
groups changes
synchronously.
The SIP IP trunk
group can be
associated only with
PRA trunk groups or
BRA trunk groups.
By default, the SIP IP
trunk group is not
associated with any
trunk group.
18 Save the save -

configuration.
Example
An enterprise network connects an IP PBX through a SIP IP trunk group named
sipip01. The signaling and media IP addresses are both 192.168.1.3, and the
signaling port number is 5063. The IP address of the peer IP PBX is 192.168.10.10,
and its port number is 5060. The home domain and registrar URI of the peer trunk
group are both abcd.com.
[Huawei] voice
[Huawei-voice] trunk-group sipip01 sip no-register
[Huawei-voice-trunkgroup-sipip01] signalling-address ip 192.168.1.3 port 5063
[Huawei-voice-trunkgroup-sipip01] media-ip 192.168.1.3
[Huawei-voice-trunkgroup-sipip01] peer-address static 192.168.10.10 5060
[Huawei-voice-trunkgroup-sipip01] home-domain abcd.com
[Huawei-voice-trunkgroup-sipip01] register-uri abcd.com
[Huawei-voice-trunkgroup-sipip01] reset
[Huawei-voice-trunkgroup-sipip01] save

NetEngine AR
Verification
Verify the SIP IP display voice trunk-group The parameter values in the
2.11.3 Configuring an H.323 Trunk Group

An H.323 trunk group implements voice communication between PBX users and
inter-office users. The H.323 trunk is a packet trunk based on the IP network, and
it uses H.323.
Context
Unlike a circuit trunk that defines a physical channel, an H.323 trunk defines a
logical channel. The H.323 trunk is used to authenticate the gateway, gatekeeper,
and external gatekeeper on the H.323 network where the PBX is used.
The PBX can function as a gatekeeper or a gateway in the following scenarios:

● Peer gateway mode
As shown in Figure 2-25, PBX1 and PBX2 are enterprise routers that can
communicate with each other on a LAN (or WAN). Voice data between PBX1
and PBX2 is transmitted through the H.323 trunk. The H.323 trunks on PBX1
and PBX2 use the peer gateway mode, and do not need to register with the
gatekeeper.
Figure 2-25 Peer gateway mode networking

PBX1 Peer gateway mode PBX2
(GW) (GW)
User A H.323 User B
trunk
IP
network
H.323
trunk
PBX4 Peer gatekeeper mode PBX3
(GK) (GK)
User D User C
● Peer gatekeeper mode

As shown in Figure 2-25, PBX3 and PBX4 are enterprise routers that can
communicate with each other on a LAN (or WAN). Voice data between PBX3
and PBX4 is transmitted through the H.323 trunk. The H.323 trunks on PBX3
and PBX4 use the peer gatekeeper mode.

NetEngine AR
● Registration mode
– One-to-one registration mode
As shown in Figure 2-26, PBX5 is a gateway and PBX6 is a gatekeeper.
PBX6 needs to authenticate the registered PBX5 through gwid. After PBX5
is successfully authenticated, PBX5 and PBX6 can communicate with each
other.
Figure 2-26 One-to-one registration mode
IP
network
H.323 trunk
One-to-one
PBX5 Registration mode PBX6
(GW) (GK)
User F
User E
– One-to-many registration mode

As shown in Figure 2-27, PBX7 and PBX8 are gateways, and PBX9 is a
gatekeeper. PBX9 needs to authenticate the registered PBX7 and PBX8
through gkid. After PBX7 and PBX8 are successfully authenticated, PBX7,
PBX8, and PBX9 can communicate with one another.
Figure 2-27 One-to-many registration mode

PBX7
(GW)
User G H.323
trunk
IP
network
H.323 PBX9 User I
trunk (GK)
One-to-many
PBX8
(GW) Registration mode
User H
Prerequisites
Media and signaling IP address pools have been configured. For details, see 2.5.2
Configuring Media and Signaling IP Address Pools.

NetEngine AR
Configuring Protocol Attributes

3 Enter the H.323 h323-attribute -

attribute view.
4 Configure the default localip ip-address -

local IP address.
5 (Optional) Configure q931-listenport By default, the Q.931

Q.931 signaling port. q931listenport-value signaling port is 1720.
6 (Optional) Configure faststart-flag By default, fast start

whether to enable the { enable | disable } is enabled.
fast start mode. NOTE
After the fast start
function is enabled,
sessions can be set up
quickly.
7 (Optional) Configure support-h245tul By default, the PBX

whether to support { enable | disable } does not support the
the H.245 signaling H.245 signaling
tunnel. tunnel.
8 (Optional) Configure workmode The values are as

the work mode of the workmode-value follows:
H.323 protocol stack. ● international:
International mode
● national: National
mode
● VT: VT mode of the
H.323 protocol
stack
Default: international
9 (Optional) Configure rasport rasport-value By default, the UDP

the UDP port of RAS port number of RAS
signaling. signaling in peer
gateway mode is
1720, and that in peer
gatekeeper mode and
registration mode is
1719.
(Optional) Configure h323-systemname H. By default, the H.323

the name of the H.323 323systemname-value stack system name is
protocol stack. HUAWEI.

NetEngine AR
10 (Optional) Configure productname By default, the name

the name of the H.323 productname-value of the product using
protocol stack. the H.323 stack is
HUAWEI-AR.
11 (Optional) Set the H. h323version H. The values are as

323 protocol version. 323version-value follows:
● V2
● V3
● V4
● V5
The default H.323
protocol version is V4.
12 (Optional) Set the H. h245version The values are as

245 protocol version. h245version-value follows:
● V2
● V3
● V4
● V5
The default H.245
protocol version is V4.
13 (Optional) Configure displayname By default, the display

the display header in displayname-value header in Q.931
Q.931 messages. messages is HUAWEI.
14 (Optional) Set the TCP tcp-keepalive- By default, the TCP

keepalive timer. timerlen tcp- keepalive timer is
keepalive-timerLen- 3600 ms.
value
15 (Optional) Configure auto-connect- By default, H.245
whether to support controlchannel channels cannot be
automatic setup of the { enable | disable } set up automatically.
H.245 channel.
16 (Optional) Configure earlyh245 { enable | By default, the H.245

whether the H.245 disable } address is not
address is received in received in setup
the setup messages. messages.
17 (Optional) Set the TCP tcp-connection- By default, the

connection timeout timeout tcp- timeout interval of
interval. connection-timeout- TCP connections is
value 15000 ms.

NetEngine AR
18 (Optional) Configure setup-wait-response- By default, the setup

the setup message timeout setup-wait- message timeout is
timeout. response-timeout- 500000 ms.
value
19 (Optional) Set the callproc-wait- By default, the
timeout interval for response-timeout interval between
the Alerting or callproc-wait- Alerting and Connect
Connect message from response-timeout- messages is 500000
the called party. value ms.
20 (Optional) Configure recvalert-wait- By default, the

Connect message response-timeout Connect message
timeout. recvalert-wait- timeout is 500000 ms.
response-timeout-
value
21 (Optional) Set the channels-timeout By default, the
channel establishment channels-timeout- channel
timeout. value establishment
timeout is 15000 ms.
22 (Optional) Set the roundtrip-delay- By default, the

timeout interval of the timeout roundtrip- timeout interval of
roundtrip delay delay-timeout-value the roundtrip delay
process. process is 15000 ms.
23 (Optional) Set the requestmode- By default, the

timeout interval for timeout timeout interval for
mode switching. requestmode-timeout- mode switching is
value 15000 ms.
24 (Optional) Set the medialoop-timeout By default, the

timeout interval of the medialoop-timeout- timeout interval of
media loop process. value the media loop
process is 15000 ms.
25 (Optional) Set the endsession-timeout By default, the

timeout interval for endsession-timeout- timeout interval for
the peer end to send value the peer end to send
EndSessionCommand EndSessionCommand
messages. messages is 3000 ms.
26 (Optional) Configure h323-app-timer For the default values,

the H.323 timer. timerindex timerlen see the command
reference.
27 Restart the H.323 reset After configuring the

system. H.323 attributes,
restart the H.323
system to make the
parameters effective.

NetEngine AR
28 (Optional) Return to quit -

the voice view.
Configuring the Trunk Group

1 Create an H.323 trunk trunk-group name ● In the peer

group and enter the H. [ h323 h323-type ] gateway mode, the
323 trunk group view. trunk group type is
set to
symmetrical.
● In the peer
gatekeeper mode,
the trunk group
type is set to gk-
gk.
● Registration mode:
– In one-to-one
registration
mode, the trunk
type on the
gateway is set
to gw-gk and
that on the
gatekeeper is
set to register-
gateway.
– In one-to-many
registration
mode, the trunk
type on the
gateway is set
to gw-gk and
that on the
gatekeeper is
set to
gatekeeper.
If a trunk group has
been created, you can
directly enter the
trunk group view.
2 Configure the local media-ip { ip-address -

media IP address or | addr-name addr-
dynamic media IP name-value }
address name.

NetEngine AR
3 Configure the gateway gwid gwid-value This command is

ID of the trunk group. applicable to the H.
323 trunks of the
register-gateway and
gw-gk types.
The H.323 trunk
group of the gw-gk
type must have a
gateway ID
configured, and the
gateway ID must be
the same as that on
the peer H.323 trunk
group of the register-
gateway type;
otherwise,
authentication will
fail.
4 Configure the remote peer-address static This command is

IP address and port. primary-ip-address applicable to the H.
primary-port-value 323 trunks of the
[ secondary symmetrical, gw-gk,
secondary-ip-address and gk-gk types.
secondary-port-value ] The connection data
on the trunk group
must be the same as
that on the peer.
5 (Optional) Configure gkid id secondary This command is

IDs of the active and secondary-id applicable to the H.
standby gatekeepers 323 trunk groups of
with which a gateway the gw-gk type.
registers on the
gateway.
6 Reset the H.323 trunk reset After creating or

group. modifying an H.323
trunk group, run this
command to make
the configuration
effective.
7 (Optional) Configure a techprefix techprefix- When multiple H.323

technical prefix. value trunk groups are
configured, this step
is required.

NetEngine AR
8 (Optional) Configure send-techprefix H. When multiple H.323

whether to send the 323-send-techprefix- trunk groups are
technical prefix. value configured, this step
is required.
9 (Optional) Configure del-techprefix H.323- When multiple H.323

whether to delete the send-techprefix-value trunk groups are
technical prefix. configured, this step
is required.
10 (Optional) Configure ttlrrq id This command is

the gateway applicable to the H.
registration timeout 323 trunk groups of
for the trunk group. the gw-gk type.
11 (Optional) Configure send-heartbeat This command is

the gatekeeper to or { disable | enable } applicable to the H.
not to send heartbeat 323 trunk groups of
messages. the gk-gk type.
12 (Optional) Configure heartbeat { timeout This command is

the heartbeat time | retrycount applicable to the H.
detection timeout and count } * 323 trunk groups of
retransmission times the gk-gk type.
for the gatekeeper. By default, the
heartbeat detection
timeout is 10s and
retransmission times
is 3.
13 (Optional) Configure enterprise enterprise- By default, an H.323

the enterprise that the name [ dn-set dn-set- trunk group belongs
H.323 trunk group name ] to the enterprise
belongs to and the default and DN set
bound DN set. DefaultDialPlan.
You need to perform
this step only when
and DN sets are
on how to configure
an enterprise and a
DN set, see 2.5.4
Configuring an
Enterprise and a DN
Set.

NetEngine AR
14 (Optional) Configure a select-mode The following circuit

trunk group circuit selectmode-value selection modes are
mode
● decrease:
descending mode
● port-loop:
ascending mode.
time slot on the
same port in
ascending order.)
● port-increase:
ascending mode.
port in ascending
order.)
loop.

the default country country-code-value configures the
code and area code. area-code area-code- country code and area
value code for incoming
calls over an H.323
trunk group. When
carry the country
code and area code,
the PBX adds the
configured country
back through call
display.
16 (Optional) Configure default-caller-telno -

the default number telno-value
displayed.
17 (Optional) Configure description desc-value -

the description for the
H.323 trunk group.

NetEngine AR
18 (Optional) Configure a register-E164- This step needs to be

registration E.164 number E164-telno- performed when the
number on an H.323 value remote GK uses the
trunk group. registration E.164
number.
19 (Optional) Configure a gatekeeper- This operation is

gateway prefix. techprefix applicable to gw-gk
gatekeeper-techprefix- type H.323 trunk
value groups.
By default, no
gateway technical
prefix is configured in
the system.
20 Restart the H.323 reset After configuring the

trunk group. H.323 trunk group,
restart the H.323
trunk group to make
the parameters
effective.
21 Save the save -

configuration.
Example 1 (H.323 Peer Gateway Mode)

PBX1 of the enterprise headquarters and PBX2 of the branch are located in
different areas. They communicate with each other through the H.323 trunk group
(h323a) in peer gateway mode. The media IP address of PBX1 is 192.168.1.3 and
of PBX2 is 192.168.1.4. The port numbers of the PBX1 and PBX2 are both 1720.
# Configure the H.323 attributes for PBX1.
[Huawei] voice
[Huawei-voice] h323-attribute
[Huawei-voice-h323-attribute] localip 192.168.1.3
[Huawei-voice-h323-attribute] reset
[Huawei-voice-h323-attribute] quit
# Configure the trunk group for PBX1.
[Huawei-voice] trunk-group h323a h323 symmetrical
[Huawei-voice-trunkgroup-h323a] media-ip 192.168.1.3
[Huawei-voice-trunkgroup-h323a] peer-address static 192.168.1.4 1720
[Huawei-voice-trunkgroup-h323a] reset
[Huawei-voice-trunkgroup-h323a] save
# The methods on PBX1 and PBX2 are the same except that the local and remote IP addresses on the two
ends are different.
Example 2 (H.323 Peer Gatekeeper Mode)

(h323a) in peer gatekeeper mode. The media IP address of PBX3 is 192.168.3.3

NetEngine AR
and of PBX4 is 192.168.3.4. The port numbers of the PBX3 and PBX4 are both
1719.
[Huawei] voice
[Huawei-voice-h323-attribute] save
[Huawei-voice] trunk-group h323a h323 gk-gk
[Huawei-voice-trunkgroup-h323a] send-heartbeat enable
[Huawei-voice-trunkgroup-h323a] quit
# The methods on PBX3 and PBX4 are the same except that the local and remote IP addresses on the two
ends are different.
Example 3 (One-to-One Registration Mode)

(h323a) in registration mode. PBX5 functions as the gateway, and PBX6 functions
as the gatekeeper. The media IP address of PBX5 is 192.168.5.5. The media IP
address of PBX6 is 192.168.5.6 and port number is 1719.
[Huawei] voice
[Huawei-voice] trunk-group h323a h323 gw-gk
[Huawei-voice-trunkgroup-h323a] gwid gwpbx6
[Huawei] voice
[Huawei-voice] trunk-group h323a h323 register-gateway
Example 4 (One-to-Many Registration Mode)

PBX9 of the enterprise headquarters, and PBX7 and PBX8 of the branch are
located in different areas. PBX7 and PBX8 communicate with PBX9 through the H.
323 trunk group (h323a) in registration mode. PBX7 and PBX8 function as the
gateways, and PBX9 functions as the gatekeeper. The media IP addresses of PBX7,
PBX8, and PBX9, are 192.168.7.7, 192.168.7.8, and 192.168.7.9, respectively. The
port number of PBX9 is 1719.

NetEngine AR
# Configure H.323 attributes for PBX7.

[Huawei] voice
[Huawei-voice-trunkgroup-h323a] gkid gkpbx9
[Huawei] voice
[Huawei] voice
[Huawei-voice] trunk-group h323a h323 gatekeeper
Verification
Verify the H. display voice trunk-group The parameter values in the

323 trunk [ name [ para-value ] ] command output are
configuration.
Verify display voice gatekeeper The prefix information of the

information callprefix gateway registered with the
about the gatekeeper is displayed in the
prefix used by command output.
the H.323 trunk
group
registered with
a gatekeeper.

NetEngine AR
Verify the display voice gatekeeper The dynamic information of

dynamic endpoints the gateway registered with
information of the gatekeeper is displayed in
a gateway the command output.
registered with
a gatekeeper.
2.11.4 Configuring a BRA Trunk Group

A BRA trunk group implements voice communication between PBX users and
inter-office users. A BRA trunk is a digital circuit trunk. If a BRA trunk is bound to
a BRA interface, the BRA trunk supports a maximum of two simultaneous voice
calls. If the BRA trunk is bound to multiple BRA interfaces, the number of
supported voice calls is twice the number of bound BRA interfaces.
Context
If the voice interface card 2BST that supports the BRA interface or a router with a
BRA interface is deployed in a PBX, the PBX can connect to the ISDN through a
BRA trunk. In this case, the BRA interface must work in TE mode.
configured to facilitate trunk management. BRA trunk groups use Digital
Subscriber Signaling No.1 (DSS1) or QSIG as the control signaling.
The PBX connects to the ISDN through a BRA trunk group, as shown in Figure
2-28.
Figure 2-28 Connecting the PBX to the ISDN through a BRA trunk group
ISDN
BRA Trunck
group
BRA2/0/0
PBX
POTS Fax ISDN IP


NetEngine AR
Configuring the Card and Interface
NOTICE
After the working mode of the 2BST card is configured, original service data on
the 2BST card and interfaces of the 2BST card will become invalid.
2 Configure the Configure set workmode After this command is

card or the 2BST slot slot-id bri executed, the system
interface to card to bri-voice nt- asks you whether to
work in NT work in mode restart the 2BST card.
mode. NT mode. If you select Y, the
(Configure 2BST card restarts. If
the working the 2BST card does
mode of the not restart in this
card or situation, run the
interface as reset slot command
required.) to restart the 2BST
card.
If the 2BST card
works in TE mode and
has BRA trunk data
configured, delete the
BRA trunk data after
you change the
working mode to NT.
3 Configure a clock source clock source You are advised to

for the BRA interface. source-id [ slotid/ configure a clock
subcardid/portid ] source for the trunk
priority priority- interface.
value If no clock source is
configured, the main
control unit transmits
clock signals
generated by the local
crystal oscillator to
other cards as the
transmit reference
clock.

NetEngine AR
Configuring a BRA Trunk Group


ISDN cause values and cause-value sip-
response-value } *
3 Create a BRA trunk trunk-group name If a BRA trunk group
group and enter the [ bra-dss1 | bra-qsig ] already exists, this
BRA trunk group view. command directly
displays the BRA trunk
group view.
4 Bind a trunk to the trunk-bra slotid/ You can specify

BRA trunk group. subcardid/portid default-called-telno
[ default-called- to transmit incoming
telno default-called- calls over a trunk to
telno-value | caller- an intra-office user
telno [ enterprise (dedicated access) or
enterprise-name ] access number of the
telno-value | default- switchboard
cli default-cli-value | (switchboard access).
[ b-channels b-
channels-value [ num
num-value ] ] config-
status config-status-
value ] *
5 (Optional) Configure enterprise enterprise- By default, a BRA
set that the BRA trunk name ] to the enterprise
group belongs to. default and DN set
DefaultDialPlan.
You need to perform
this step only when
and DN sets are
on how to configure
an enterprise and a
DN set, see 2.5.4
Configuring an
Enterprise and a DN
Set.

NetEngine AR
6 (Optional) Configure a select-mode The following circuit

trunk group circuit selectmode-value selection modes are
mode
● decrease:
descending mode
● port-loop:
ascending mode.
time slot on the
same port in
ascending order.)
● port-increase:
ascending mode.
port in ascending
order.)
loop.

the default country country-code-value configures the country
code and area code. area-code area-code- code and area code
value for incoming calls
over a BRA trunk
group. When
carry the country code
and area code, the
PBX adds the
configured country
back through call
display.

displayed.
9 (Optional) Configure description desc- -

the description for the value
BRA trunk group.

NetEngine AR
10 (Optional) Configure typeofnumber caller -

the calling number callernumber-value
type.
11 (Optional) Configure typeofnumber called -

the called number callednumber-value
type.
12 (Optional) Configure num-plan-identify This step is required

the caller number plan caller only when the remote
identify. numplanidentify-value device has special
requirements on the
numbering plan.

the called number called only when the remote
plan identify. numplanidentify-value device has special
requirements on the
numbering plan.
14 Save the save -

configuration.
Example
Create a BRA trunk group named braa, bind the BRA trunk connected to BRA
interface 2/0/0 to the BRA trunk group, and set the default called number to the
access number 7200. Configure a clock source for the BRA interface.
[Huawei] set workmode slot 2 bri bri-voice te-mode
[Huawei] clock source 0 2/0/0 priority 5
[Huawei] voice
[Huawei-voice] trunk-group braa bra-dss1
[Huawei-voice-trunkgroup-braa] trunk-bra 2/0/0 default-called-telno 7200
[Huawei-voice-trunkgroup-braa] save
Verification
Verify the BRA display voice trunk-group The parameter values in the
2.11.5 Configuring a PRA Trunk Group

A PRA trunk group implements voice communication between PBX users and
inter-office users. A PRA trunk is a digital circuit trunk and uses an E1 interface on
the PBX to connect to the remote device through an E1 trunk line.

NetEngine AR
Context
A PRA trunk can use the E1 interface not the T1 interface to connect the PBX to
the remote device. The E1 interface card must work in voice mode, and E1
interfaces on this card are called VE1 interfaces. A VE1 interface supports
concurrent calls over a maximum of 30 channels, and the signaling type must be
set to CCS.
A PRA trunk group can use Digital Subscriber Signaling System No.1 (DSS1) or Q
signaling (QSIG), and the PBX can function as the user-side or network-side
device. At both ends of a PRA trunk, one end must function as the user-side
device, and the other end must function as the network-side device.
● If the PBX uses DSS1 signaling to connect to a network through a PRA trunk
group and is used as a network-side device, the signaling type is set to dss1-
net.
● If the PBX uses DSS1 signaling to connect to a network through a PRA trunk
group and is used as a user-side device, the signaling type is set to dss1-user.
● If the PBX uses QSIG signaling to connect to a network through a PRA trunk
group and is used as a network-side device, the signaling type is set to qsig-
net.
● If the PBX uses QSIG signaling to connect to a network through a PRA trunk
group and is used as a user-side device, the signaling type is set to qsig-user.
The PBX can use a PRA trunk group to connect to the PSTN and traditional PBXs
to protect investments and implement smooth expansion. Figure 2-29 shows
networking of a PRA trunk group.
Figure 2-29 Networking diagram for configuring a PRA trunk group
PRA trunk
group
PRA trunk
group Traditional
PBX
PBX
POTS Fax ISDN IP


NetEngine AR
Configuring a VE1 Interface

2 Configure the E1 set workmode slot After this command is

interface card as an E1 slot-id e1t1 e1-voice executed, the
voice card. configured service
becomes invalid.
After you use this
command, the system
displays a message
asking you whether to
reset the E1 voice
card. If you enter Y,
the system resets the
E1 voice card to make
the configuration take
effect. Otherwise, run
the reset slot
command to restart
the E1 voice card.
4 Enter the VE1 port ve1 slotid/ -

interface view. subcardid/portid
5 Configure CSS on the signal ccs -
VE1 interface.

view.
7 Return to the system quit -

view.
8 Configure a clock clock source sourceid When the local end is

source. [ slotid/cardid/portid ] the user end, it is
priority priority-value recommended that
you configure a clock
source for the trunk
interface.
If no clock source is
clock signals
generated by the local
crystal oscillator to
other cards as the
transmit reference
clock.

NetEngine AR
Configuring a Trunk Group


ISDN cause values and cause-value sip-
response-value | sip-
response sip-
response-value isdn-
cause isdn-cause-
value }
3 Create a PRA trunk trunk-group trunk- When configuring a
group and enter the group-name [ dss1- trunk group, specify
PRA trunk group view. net | dss1-user | qsig- DSS1 or QSIG
net | qsig-user ] signaling and the PBX
as the user-side or
network-side device.
For details, see
Context.
The signaling and
device types are
determined through
negotiation.
If a PRA trunk group
already exists, this
command directly
displays the PRA trunk
group view.
4 Bind a trunk to the trunk-pra slotid/ -

PRA trunk group. subcardid/portid [ [ b-
channels b-channels-
value [ num num-
value ] ] config-
status config-status-
value | default-cli
default-cli-value ]

NetEngine AR
5 (Optional) Configure enterprise enterprise- By default, a SIP IP

set that the SIP PRA name ] to the enterprise
trunk group belongs default and DN set
to. DefaultDialPlan.
You need to perform
this step only when
and DN sets are
on how to configure
an enterprise and a
DN set, see 2.5.4
Configuring an
Enterprise and a DN
Set.
6 (Optional) Configure a select-mode select- The following circuit

trunk group circuit mode-value selection modes are
mode
● decrease:
descending mode
● port-loop:
ascending mode.
time slot on the
same port in
ascending order.)
● port-increase:
ascending mode.
port in ascending
order.)
loop.

NetEngine AR

the default country country-code-value configures the country
code and area code. area-code area-code- code and area code
value for incoming calls
over a PRA trunk
group. When
carry the country code
and area code, the
PBX adds the
configured country
back through call
display.

displayed.
9 (Optional) Configure description desc- -

the description for the value
PRA trunk group.
10 (Optional) Configure typeofnumber caller -

the calling number callernumber-value
type.
11 (Optional) Configure typeofnumber called -

the called number callednumber-value
type.

the caller number plan caller only when the remote
identify. numplanidentify-value device has special
requirements on the
numbering plan.

the called number called only when the remote
plan identify. numplanidentify-value device has special
requirements on the
numbering plan.

NetEngine AR
14 (Optional) Configure circuit-state-relate This command

trunk groups that are trunk-group-name associates other trunk
associated with the groups with the PRA
PRA trunk group. trunk group. When
the status of the PRA
trunk group changes,
status of the
associated trunk
groups also changes.
The PRA trunk group
can be associated only
with SIP trunk groups.
By default, the PRA
trunk group is not
associated with any
trunk group.
15 (Optional) Configure stable delay-time The value is in

status synchronization time-value seconds. By default,
delay of trunk groups the status
associated with the synchronization delay
PRA trunk group. of trunk groups
associated with the
PRA trunk group is
30s.
16 Save the save -

configuration.
Example
An enterprise connects to a PSTN through a PRA trunk group, which occupies E1
interface 0 in slot 1 of the PBX. Configure a clock source for the E1 interface.
Create a PRA trunk group named pra01, specify DSS1 as the signaling protocol,
and configure the PRA trunk group as the user end.
# Configure a VE1 interface.
[Huawei] 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.
[Huawei] voice
[Huawei-voice] port ve1 1/0/0
[Huawei-voice-ve1-1/0/0] signal ccs
[Huawei-voice-ve1-1/0/0] quit
[Huawei-voice] quit
# Configure a trunk group.
[Huawei] voice
[Huawei-voice] trunk-group pra01 dss1-user
[Huawei-voice-trunkgroup-pra01] trunk-pra 1/0/0
[Huawei-voice-trunkgroup-pra01] save

NetEngine AR
Verification
Verify the PRA display voice trunk-group The parameter values in the
2.11.6 Configuring an R2 Trunk Group

An R2 trunk group implements voice communication between PBX users and
inter-office users. An R2 trunk is a digital circuit trunk and uses an E1 interface on
the PBX to connect to the remote device through an E1 trunk line. It uses R2
signaling.
Context
R2 signaling is a type of Channel Associated Signaling (CAS). R2 signaling is
defined by ITU-T Q.400-Q.490. Different countries define different R2 signaling
standards, and there are great differences between CAS in some versions and
standard R2 signaling. The PBX supports R2 signaling types of Brazil(MFC-5C),
Argentina, and Mexico. Figure 2-30 describes the mapping between the R2
signaling type, R2 profile, and trunk group on the PBX.
Figure 2-30 Mapping between the R2 signaling type, signaling profile, and trunk
group on the PBX
R2 signaling type R2 signaling profile R2 trunk group
ITU-T (Standard) R2 signaling type R2 signaling profile
brazil Line signaling attribute Call right
Input register address

argentina Bound trunk
format
Output register address …

mexico
format
Register signaling
User-defined attributes
Line signaling output
parameters
Register signaling
input parameters
Register signaling
output parameters
An R2 trunk can use the E1 interface not the T1 interface to connect the PBX to
the remote device. The E1 interface card must work in voice mode, and E1

NetEngine AR
interfaces on this card are called VE1 interfaces. A VE1 interface supports
concurrent calls over a maximum of 30 channels.
NOTE
AR6120-VW provide E1 interfaces but do not support the R2 trunk.
The PBX connects to the PSTN through an R2 trunk group, as shown in Figure
2-31.
Figure 2-31 Connecting the PBX to the PSTN through an R2 trunk group
R2 trunk
group
PBX
POTS Fax ISDN IP

Configuring a VE1 Interface


NetEngine AR
2 Configure the E1 set workmode slot After this command is

interface card as an E1 slot-id e1t1 e1- executed, the configured
voice card. voice service becomes invalid.
After you use this
command, the system
displays a message
asking you whether to
reset the E1 voice card.
If you enter Y, the
system resets the E1
voice card to make the
configuration take
effect. Otherwise, run
the reset slot command
to restart the E1 voice
card.

5 Configure CAS on the signal { cas } -
VE1 interface.

view.
7 Return to the system quit -

view.
8 Configure a clock clock source It is recommended that

source. source-id [ slotid/ you configure a clock
subcardid/portid ] source for the trunk
priority priority- interface.
value If no clock source is
clock signals generated
by the local crystal
oscillator to other cards
as the transmit
reference clock.
(Optional) Setting the R2 Signaling Type

The PBX provides R2 signaling types that adapt to ITU-T, Brazil, Argentina, and
Mexico standards by default. To define a new R2 signaling type, perform the
following operations.

NetEngine AR
Step Action Command
1 Enter the voice view. voice
2 Create an R2 signaling r2 signalling-type signalling-type-name

type and enter the R2
signaling type view, or
directly enter the
existing R2 signaling
type view.
3 Set line signaling output line-signal-send line-logic-cmd { line-

parameters. logic-signal-type line-phy-cmd-para line-
call-state line-phy-send-time | no-
process }
4 Set register signaling register-signal-receive register-phy-cmd

input parameters. register-call-state { register-logic-cmd
register-logic-cmd-para | no-process }
5 Set register signaling register-signal-send register-logic-cmd
output parameters. register-logic-cmd-para { register-phy-cmd
| no-process }
6 Return to the voice view. quit
Configuring an R2 Profile
p
1 Create an R2 r2 profile profile-name -

profile and enter
the R2 profile
view.
2 (Optional) line-signalling { line- -

Configure line signalling-type line-
signaling signalling-type-value | wait-
attributes. answer-time wait-answer-
time-value | wait-clear-
forward-time wait-clear-
forward-time-value | wait-
protect-time wait-protect-
time-value | wait-seize-ack-
time wait-seize-ack-time-
value }

NetEngine AR

p
3 (Optional) Set the register-address-receive -

input register { country-code value |
address format. distinctive-indication value
| echo-cancellation 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 }
4 (Optional) Set the register-address-send -
output register { country-code value |
address format. distinctive-indication value
| echo-cancellation value |
circuit-character value |
calling-number value |
calling-user-type value |
calling-user-priority value |
called-number value }

NetEngine AR

p
5 (Optional) register-signalling { back- -

Configure register pulse-time back-pulse-
signaling time-value | calling-
attributes. number-max-length
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 | number-max-
interval number-max-
interval-time | number-
send-mode number-send-
mode-value | outgoing-type
outgoing-type-value |
receive-number-max-
interval receive-number-
max-interval-time | send-
pulse-signal signal-value |
signal-send-voltage signal-
send-voltage-value | signal-
stop-permit-time signal-
stop-permit-time-value |
wait-Bsignal-time wait-
Bsignal-time-value | wait-
caller-category-time wait-
caller-category-time-value }
6 Set the adaptation signalling-type signalling- Select standard (ITU-
signaling type. type-name T), brazil, argentina,
mexico, or another
user-defined R2
signaling type.

voice view.

NetEngine AR
Configuring a Trunk Group

p
1 Create an R2 trunk trunk-group name [ e1- -

group and enter r2 ]
the R2 trunk group
view.
2 Bind an R2 profile r2-profile profile-name -

to the R2 trunk
group.
3 Bind a trunk to the trunk-r2 slotid/subcardid/ After the R2 trunk

R2 trunk group. portid group is complete, to
ensure that call
services passing
through the R2 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
R2 trunk group are
restored.
4 (Optional) r2-receive-earlymedia r2- -

Configure an R2 receive-earlymedia
trunk group
whether to use
early media.
5 (Optional) r2-play-ringback r2-play- -

Configure an R2 ringback
trunk group
whether to play
ringback tones to
the remote end.

NetEngine AR

p
6 (Optional) enterprise enterprise- By default, an R2

Configure the name [ dn-set dn-set- trunk group belongs
enterprise and DN name ] to the enterprise
set that the R2 default and DN set
trunk group DefaultDialPlan.
belongs to. You need to perform
this step only when
and DN sets are
on how to configure
an enterprise and a
DN set, see 2.5.4
Configuring an
Enterprise and a DN
Set.
7 (Optional) select-mode selectmode- The following circuit

Configure a trunk value selection modes are
group circuit available:
selection mode. ● loop: polling mode
mode
● decrease:
descending mode
● port-loop:
ascending mode.
time slot on the
same port in
ascending order.)
● port-increase:
ascending mode.
port in ascending
order.)
loop.
8 (Optional) default-caller-telno telno- -

Configure the value
default number
displayed.

NetEngine AR

p

Configure the
description for the
R2 trunk group.
10 (Optional) reanswer reanswer-value By default, the calling

Configure the party control function
function of the is disabled for an
calling party E1R2 trunk group.
control.
11 Save the save -

configuration.
Example
An enterprise connects to a PSTN through a R2 trunk group, which occupies E1
interface 0 in slot 1 of the PBX. Configure a clock source for the E1 interface.
Create a R2 trunk group named r201, and set the R2 signaling type to standard.
# Configure a VE1 interface.
[Huawei] voice
[Huawei-voice-ve1-1/0/0] signal cas
[Huawei-voice] quit
# Configure a signaling profile.
[Huawei] voice
[Huawei-voice] r2 profile r201
[Huawei-voice-r2-profile-r201] signalling-type standard
[Huawei-voice-r2-profile-r201] quit
# Configure a trunk group.
[Huawei-voice] trunk-group r201 e1-r2
[Huawei-voice-trunkgroup-r201] r2-profile r201
[Huawei-voice-trunkgroup-r201] trunk-r2 1/0/0
[Huawei-voice-trunkgroup-r201] save
Verification
Verify the R2 display voice trunk-group The parameter values in the


NetEngine AR
2.12 Configuring a Call Route

A call route defines routing rules for outgoing calls, and is bound to a call prefix
and a trunk group. After a call route is configured, the PBX can correctly route
outgoing calls.
Context
NOTE
A trunk group supports a maximum of 48 routes, and the system supports a maximum of
1024 routes.
When a user calls an inter-office user, the PBX extracts the inter-office call prefix
from the number, finds a trunk group bound to the call prefix, and then finds the
called party.
Figure 2-32 Call route

Call route:
Bind inter-office call prefix 9 to
trunk group 1
Bind inter-office call prefix 8 to
trunk group 2
Trunk Trunk
group 1 group 2
PBX PBX B
Dial 8001
User User User

28980001 7001 8001
According to the example in Figure 2-32, the process of an outgoing call is as

follows:
1. Intra-office user 7001 calls inter-office user 8001.

2. After the PBX receives the number, it finds the prefix in the DN set bound to
the calling number.
3. The PBX finds a trunk group bound to the call route according to inter-office
call prefix 8.
4. The PBX selects an inter-office line to PBX B according to trunk group 2.
Prerequisites
● The operations of 2.9 Configuring a Call Prefix have been performed.
● The operations of 2.11 Configuring Trunk Groups have been performed.

NetEngine AR
Procedure
Step Action Command
1 Enter the system view. system-view
3 Create a call route. callroute callroute-name
4 Return to the system view. quit
5 Enter the call prefix profile view. callprefix callprefix-name
6 Bind a call route to the call callroute callroute-name

prefix.
8 Enter the SIP AT0 trunk group trunk-group trunk-group-name

view.
9 Bind a call route to the trunk callroute callroute-name

group.
11 Save the configuration. save
Follow-up Procedure
● Perform the operations of 2.10.5 Configuring a Post-routing Number
Change Plan.
● Configure the custom routing selection policy. For details, see 2.15.5
Configuring Intelligent Routing.
Example
Bind inter-office call prefix 8 and trunk group pra01 to the call route route01.
When an intra-office user dials the inter-office number starting from 8, the
outgoing call is made over the trunk group pra01.
# Create a call route plan.
[Huawei] voice
[Huawei-voice] callroute route01
[Huawei-voice-callroute-route01] quit
# Bind a call prefix to the call route plan.
[Huawei-voice-callprefix-8] callroute route01
# Bind a trunk group to the call route plan.
[Huawei-voice] trunk-group pra01
[Huawei-voice-trunkgroup-pra01] callroute route01
[Huawei-voice-trunkgroup-pra01] save

NetEngine AR
Verification
Verify the call display voice callprefix The value of Call route name
prefix [ callprefix-name ] is consistent with the setting.
configuration.
Verify the trunk display voice trunk-group The value of Relate call
group [ name [ para-value ] ] route name is consistent
configuration. with the setting.
2.13 Configuring the Default Call Route

The default call route defines routing rules for inter-office calls, and is bound to a
trunk group. A call for which no prefix can be matched is routed out through the
default call route.
Context
When a user makes an inter-office call, the PBX analyzes the dialed number to
obtain the inter-office call prefix. If the inter-office call prefix fails to be analyzed,
the PBX sends the call request to the office where the called party is located
through the trunk group corresponding to a call route bound to the default call
route.
Figure 2-33 PBX default call route
In Figure 2-33, the process of the inter-office call is as follows:

1. Intra-office user 7001 calls inter-office user 8001.
2. After collecting the digits, the PBX searches for the prefix in the DN set
corresponding to the calling number.
3. If no prefix is found, the PBX searches for the corresponding trunk group in
the bound call routes based on the default call route.
4. Based on the found trunk group (trunk group 2 in the preceding figure), the
PBX routes the inter-office call to PBX B.

NetEngine AR
Prerequisites
Trunk groups have been configured. For details, see 2.11 Configuring Trunk
Groups.
Procedure
Step Operation Command
3 Create a call route. callroute callroute-name
5 Create a default call route and default-callroute

enter its view.
6 (Optional) Configure the enterprise enterprise-name dn-

enterprise and DN set to which set dn-set-name
the default call route belongs.
7 Bind the call route to the default callroute callroute-name

call route.
9 Enter the trunk group view. trunk-group trunk-group-name
10 Bind the call route in the trunk callroute callroute-name

group view.
Follow-up Procedure
Configure user-defined route selection policies. For details, see 2.15.5 Configuring
Intelligent Routing.
Bind the default call route and trunk group pra01 to the call route route01 to
route calls for which no prefix can be matched through the call route route01.
# Create a call route.
[Huawei] voice
[huawei-voice] callroute route01
[huawei-voice-callroute-route01] quit
# Bind the call route in the default call route view.
[huawei-voice] default-callroute
[Huawei-voice-default-callroute] enterprise default dn-set DefaultDialPlan
[huawei-voice-default-callroute] callroute route01
[huawei-voice-default-callroute] quit

NetEngine AR
# Bind the call route in the trunk group view.

[huawei-voice] trunk-group pra01
[huawei-voice-trunkgroup-pra01] callroute route01
[huawei-voice-trunkgroup-pra01] save
Verification
Operation Command Expected Result
Display display voice default- <Huawei> display voice default-

callroute
information callroute Enterprise :
about the default
default call
Dn-set :
route. DefaultDialPlan
Call route name :

default
Call route name : route01
2.14 Configuring Services
2.14.1 Configuring the Fax Service

This section describes the concept and configuration of the fax service.
Context
Depending on the transport protocol used, the fax service can be provided in
transparent or T.38 mode.
In transparent mode, modulated fax data from a PSTN is forwarded over an end-
to-end voice tunnel on an IP network. The gateway between the PSTN and the IP
network directly forwards fax data to the peer gateway and fax machine as voice
data, and does not perform fax modulation and demodulation. Fax transparent
transmission is implemented using either of the following methods:
● Preconfigured voice code is used to transmit fax data.
● During fax processing, the gateway converts the fax code into G.711 lossless
code through media negotiation or digitally re-encodes the fax signals using
T.38. This helps reduce the impact on fax data caused by IP network jitter.
Figure 2-34 shows the process of fax transparent transmission.

NetEngine AR
Figure 2-34 Process of fax transparent transmission

Analog data passes the
Analog 64kbit/s VoIP tunnel Analog
data data
Gateway Gateway
IP
network
Fax Fax
G.711 coding G.711 coding

64kbit/s 64kbit/s
RTP data stream
In T.38 mode, the sender gateway demodulates the T.30 fax data from the PSTN.
The demodulated fax data is encapsulated in datagrams and sent to the receiver
across the IP network. The receiver gateway modulates the datagrams into T.30
fax data and sends the fax data to the receiver. Figure 2-35 shows the process of
T.38 fax transmission.
Figure 2-35 Process of T.38 fax transmission
Data packet
transmission
Analog Analog
data Gateway Gateway data
IP
network
Fax Fax
DSP DSP
demodulation modulation
RTP data RTP data
stream T.38 signaling stream
A fax machine can be directly connected to the FXS interface of the PBX or
connected to the PBX through an integrated access device (IAD). Figure 2-36
shows a typical networking of the fax service.

NetEngine AR
Figure 2-36 Fax service networking
IP
PBX network IP PBX
IAD
Fax A Fax B Fax C
On this network:
● The PBX connects to the IP PBX through a SIP trunk.
● The IAD registers fax machine A with the PBX.
● Fax machine B is a POTS user of the PBX.
Prerequisites
Before configuring the fax service, complete the following tasks:
● Configure PBX users. For details about the configuration, see 2.8 Configuring
a PBX User.
● Configure a call prefix for inter-office calls. For details about the
configuration, see 2.9 Configuring a Call Prefix.
● Configure a SIP or H.323 trunk group. For details about the configuration, see
2.11 Configuring Trunk Groups.
Procedure
The fax service can be transmitted in transparent or T.38 mode, and the codec
negotiation mode can be negotiation or auto-switching. By default, transparent
transmission and negotiation mode are used. The PBX must negotiate with the
remote IP PBX about the transmission protocol or mode to improve fax
transmission efficiency. To configure the transmission mode, perform the following
steps.

p

system
view.
2 Enter the voice -

voice view.

NetEngine AR

p
3 Enter the sipserver This configuration applies to

SIP server communication between fax
view. machines connected to the
same PBX, for example, fax
4 Configure fax-modem fax machine A and fax machine
the fax transmission-mode { T38 | B.
transmissio throughly | T38_throughly |
Both ends must use the same
n mode for throughly_T38 }
codec negotiation mode.
the SIP
server. NOTE
In the fax interaction process, if
5 Configure fax-modem common the fax recipient gives priority to
the remote end, the fax
the fax negotiation-mode recipient determines the fax
codec { negotiate | self-switch } transmission mode as follows:
negotiation (Whether the remote end takes
mode. priority is controlled by control
point 8. For details, see pbx
number-parameter.)
● If the transmission mode of
the fax sender is throughly
or T38, check whether the
local end supports this
transmission mode. If the
local end does not support
the fax transmission mode,
fax transmission fails.
the fax sender is
T38_throughly, check
whether the local end
supports the T.38 mode and
transparent transmission
mode in sequence, and use
the transmission mode
matched first to transmit
data. If the transmission
mode of the fax sender is
throughly_T38, check
supports the transparent
transmission mode and T.38
data.
If the fax recipient gives priority
to the local end, the fax
recipient performs similar
negotiation based on the
transmission mode supported by
the local end and the fax
transmission mode specified in
the fax message.

NetEngine AR

p
6 Return to quit -
the voice
view.
7 Enter the trunk-group name This configuration applies to

trunk communication between fax
group view. machines connected to
different PBXs, for example,
8 Configure fax-modem fax fax machine A/B and fax
the fax transmission-mode { T38 | machine C.
transmissio throughly | T38_throughly |
Both ends must use the same
n mode for throughly_T38 }
codec negotiation mode.
the trunk
group.

NetEngine AR

p
9 Configure fax-modem common NOTE

the fax negotiation-mode In the fax interaction process, if
the fax recipient gives priority to
codec { negotiate | self-switch }
the remote end, the fax
negotiation recipient determines the fax
mode. transmission mode as follows:
(Whether the remote end takes
priority is controlled by control
point 8. For details, see pbx
number-parameter.)
the fax sender is throughly
or T38, check whether the
local end supports this
transmission mode. If the
local end does not support
the fax transmission mode,
fax transmission fails.
the fax sender is
T38_throughly, check
supports the T.38 mode and
transparent transmission
data. If the transmission
mode of the fax sender is
throughly_T38, check
supports the transparent
transmission mode and T.38
data.
If the fax recipient gives priority
to the local end, the fax
recipient performs similar
negotiation based on the
transmission mode supported by
the local end and the fax
transmission mode specified in
the fax message.
10 (Optional) nte-fax-modem enable -

Enable RFC
2833 fax
and
modem
transmissio
n.

NetEngine AR

p
11 (Optional) fax-modem common rtp- Both ends must have the

Configure interval { 5ms | 10ms | same packetization interval
the 20ms | 30ms } configured.
packetizati
on interval
in
transparent
transmissio
n mode.
12 (Optional) fax-modem common vbd- Both ends must have the

Configure attribute-type same VBD attribute type
the VBD { chinatelecom | configured.
attribute extchinatelecom | ietf |
type. v152 }
13 (Optional) fax-modem common vbd- -

Configure codec { g711a | g711u }
the VBD
codec
mode.
14 (Optional) redundancy-negotiation- -
Configure mode { negotiation | fixed-
the mode start | no-initiative-start }
to start the
RFC 2198
redundancy
transmissio
n
negotiation
.
15 (Optional) redundancy-start-mode -
Configure { ordinary2198 |
the mode smart2198 }
to start the
RFC 2198
redundancy
transmissio
n.
16 (Optional) redundancy-vbd enable -

Enable RFC
2198
redundancy
transmissio
n of VBD.

NetEngine AR

p
17 (Optional) nte-negotiation-mode -
Configure { negotiation | fixed-start |
the mode no-initiative-start }
to start the
RFC 2833
transmissio
n
negotiation
.
18 (Optional) quit -
Return to
the voice
view.
19 (Optional) dsp-attribute -
Enter the
DSP
attribute
view.
20 (Optional) fax redundancy-t4 -

Configure redundancy-t4-value
the T.4
redundancy
parameters
for T.38 fax
transmissio
n.
21 (Optional) fax redundancy-t30 -

Configure redundancy-t30–value
the T.30
redundancy
parameters
for T.38 fax
transmissio
n.
22 (Optional) fax training-mode { e2e | -

Configure local }
the fax
training
mode.
23 (Optional) fax training-rate { v17 | Set this parameter according

Configure v29 | v27 } to the maximum rate
the supported by the fax
maximum machines.
fax training
rate.

NetEngine AR

p
24 (Optional) fax v8negotiate enable -

Configure
V8
negotiation
parameters
.
25 (Optional) fax ecm enable -

Enable the
error
correction
mode
(ECM) for
T.38 fax
transmissio
n.
Example
● To implement communication between fax machines connected to the same
PBX, set the fax transmission mode to T.38 and set the codec negotiation
mode to auto-switching.
● To implement communication between fax machines connected to different
PBXs, set the fax transmission mode to T.38 and set the codec negotiation
mode to auto-switching.
[Huawei] voice
[huawei-voice-sipserver] fax-modem fax transmission-mode t38
[huawei-voice-sipserver] fax-modem common negotiation-mode self-switch
[huawei-voice-sipserver] quit
[huawei-voice-trunkgroup-1] fax-modem fax transmission-mode t38
[huawei-voice-trunkgroup-1] fax-modem common negotiation-mode self-switch
[huawei-voice-trunkgroup-1] nte-fax-modem enable
[huawei-voice-trunkgroup-1] quit
Verification
Verify the fax display voice sipserver The parameter values

transmission parameter fax-modem displayed in the
settings on the SIP command output are
server. consistent with the
settings.
Verify the fax display voice trunk-
transmission parameter group[ name [ fax-
settings in the trunk modem ] ]
group.

NetEngine AR
2.14.2 Configuring Voice Mailbox Services

This section describes the configurations of the voice mailbox service supported by
PBX.
NOTE
The AR617VW, AR617VW-LTE4, AR617VW-LTE4EA, and AR6120-VW do not support this

function.
2.14.2.1 Context for Configuring Voice Mailbox Services

The PBX has a built-in voice mailbox services (VMS) module, which provides voice
mailbox services.
The voice mailbox service uniformly stores and manages all voice messages. Users
can listen to voice messages anytime and anywhere through the intra-office or
inter-office phones.
The PBX supports the following voice mailbox services:

● Call forwarding unconditional (CFU)
● Call forwarding no reply (CFNR)
● Call forwarding busy (CFB)
● Call forwarding offline (CFO)
● Greeting phrases customization
NOTE
● Voice messages for the voice mailbox service are stored in the USB flash drive, or SD
card.
● By default, a message can last 120 seconds.
● By default, a user can leave 20 messages.
2.14.2.2 Preparing for the Voice Mailbox Configuration

Before configuring the voice mailbox service, ensure that the space on USB flash
drive, or SD card is sufficient.
The storage space is calculated as follows:
Required storage space = [PCM voice data size per second (8 KB) x Maximum
duration of each message (120s) + File header and index (0.1 KB)] x Maximum
number of messages a user can leave (20) x Number of users. Therefore, an 8 GB
space can accommodate the messages of about 430 users.
2.14.2.3 Configuring the CFU Voice Mailbox Service

After a user has registered the CFU voice mailbox service, the PBX unconditionally
transfers the calls to the user to the voice mailbox. The callers leave messages to
this user according to the prompts of the voice mailbox system.

NetEngine AR
Prerequisites
The intra-office user configurations are complete. For the configuration procedure,
Configuring Prefixes of Voice Messages

p

system view.
2 Enter the voice -

voice view.
3 Enter the vms -

VMS view.
4 Configure vmsdata-storage-path Before configuring the

the path for vmsdata-storage-filepath- VMS, this path must be
storing voice value specified.
messages.
5 Return to quit -
the voice
view.
6 Create a callprefix callprefix-name -

prefix
template
and enter
the prefix
template
view.
7 Configure prefix prefix -

the call
prefix.
8 Configure call-type category vu- Set the vu-service

the call type service vu-service-name parameter to
and vuservicename vuvmsleavemsg.
attributes
for the
prefix.
9 Set the digit-length minimum- -

length of a length-value maximum-
number that length-value
can be
parsed.
10 Return to quit -
the voice
view.

NetEngine AR
Configuring the Message Extraction Prefix

p

prefix
template
and enter
the prefix
template
view.

the call
prefix.

the call type service vu-service vu- parameter to
and service-name vuvmsgetmsg.
attributes vuservicename
for the
prefix.

number length-value
that can be
parsed.
5 Return to quit -
the voice
view.
Configuring Service Rights and Activating Service Functions

p
1 Enter the vms -

voice view.
2 Configure a vmsno vmsno callprefix -

voice callprefix-value
mailbox.
3 Return to quit -
the voice
view.

NetEngine AR

p

user view.
5 Configure service-right voicemailcfu Set the service-right

the message enable parameter to voicemailcfu.
leaving
rights for
the user.
6 Activate the service voicemailcfu -

message vmsno vmsno-value active
leaving enable
function.
7 Return to quit -
the voice
view.
(Optional) Configuring the Activation and Deactivation Prefixes

When the PBX needs to provide the self-activation or self-deactivation service for
end users, the service prefixes must be configured.
Service prefixes include service activation and deactivation prefixes. Configure the
prefixes one by one according to the following steps.

p

prefix
template
and enter
the prefix
template
view.

the call
prefix.
3 Configure call-type category The call type category

the call type callcategory attribute must be set to new-
and attribute supplementary-
attributes management.
for the For the service activation
prefix. prefix, set attribute to 113.
For the service deactivation
prefix, set attribute to 114.

NetEngine AR

p
4 Set the digit-length minimum- The minimum length and

length of a length-value maximum- maximum length can be set
number that length-value to the actual prefix length.
can be
parsed.
5 Save the save -

configuratio
n.
Example
Configure the CFU voice mailbox service rights and activation prefix for user 7000.
Step 1 Set the voice message prefix to 10000.

Store voice data to the SD card.
[Huawei] voice
[Huawei-voice] vms
[Huawei-voice-vms] vmsdata-storage-path sd:
[Huawei-voice-vms] quit
[Huawei-voice] callprefix leave
[Huawei-voice-callprefix-leave] prefix 10000
[Huawei-voice-callprefix-leave] call-type category vu-service vu-service-name vuvmsleavemsg
[Huawei-voice-callprefix-leave] digit-length 5 32
[Huawei-voice-callprefix-leave] quit
Step 2 Set the message extraction prefix to 10001.

[Huawei-voice] callprefix get
[Huawei-voice-callprefix-get] prefix 10001
[Huawei-voice-callprefix-get] call-type category vu-service vu-service-name vuvmsgetmsg
[Huawei-voice-callprefix-get] digit-length 5 32
[Huawei-voice-callprefix-get] quit
Step 3 Configure the service rights and activate the service function.
[Huawei-voice] vms
[Huawei-voice-vms] vmsno 0 callprefix leave
[Huawei-voice-pbxuser-7000] service-right voicemailcfu enable
[Huawei-voice-pbxuser-7000] service voicemailcfu vmsno 0 active enable
[Huawei-voice-pbxuser-7000] quit
Step 4 Configure the service prefix.

Configure the service activation prefix. The configuration of service deactivation
prefix is similar.
NOTE
Before configuring the service activation prefix, run the service voicemailcfu vmsno 0
command to bind the VMS number in the pbxuser view.
[Huawei-voice] callprefix *113*
[Huawei-voice-callprefix-*113*] prefix *113*
[Huawei-voice-callprefix-*113*] call-type category new-supplementary-management attribute 113

NetEngine AR
[Huawei-voice-callprefix-*113*] digit-length 5 32
[Huawei-voice-callprefix-*113*] save
----End
Verification
Verify the call prefix display voice callprefix The parameter values in
configuration. callprefix-name the command output are
consistent with the
settings.
Verify the service rights. display voice pbxuser [Huawei-voice]display

service-right voice pbxuser 7000
service-right
forwarding call to voice
mail unconditional : Enable
Service Verification
After the configuration is complete, verify the CFU voice mailbox service.
1. User B calls user A. When hearing the prompts, user B leaves messages
according to the system prompts.
2. If the terminal of user A is an IP phone, the system notifies user A of the new
messages using message indicator or displays the messages on the screen of
the IP phone. If the terminal of user A is a POTS phone, the system notifies
user A of the new messages through voice.
To configure the message indicator on the SIP phone, such as an IP phone,
see the Administrator Guide of IP phone. If you need to configure the MWI
mode for POTS users, see mwi-mode for the detailed operations.
3. User A dials the message extraction prefix to listen to the voice messages.
NOTE
● User A can extract voice messages on an intra-office or inter-office phone. When

extracting leave messages, you need to enter the service password.
● The default username and password are available in AR Router Default Usernames
and Passwords (Enterprise Network or Carrier). If you have not obtained the
access permission of the document, see Help on the website to find out how to
obtain it. If you use the password for the first time, the password must be
modified. For details about how to change the password, see 2.14.4 Configuring
the Service Password.
2.14.2.4 Configuring the CFNR Voice Mailbox Service

After a user has registered the CFNR voice mailbox service, the PBX transfers the
calls to the user to the voice mailbox when times out. The callers leave messages
to this user according to the prompts of the voice mailbox system.

NetEngine AR
Prerequisites

p

system view.
2 Enter the voice -

voice view.
3 Enter the vms -

VMS view.

messages.
5 Return to quit -
the voice
view.

prefix
template
and enter
the prefix
template
view.

the call
prefix.

attributes
for the
prefix.

can be
parsed.
10 Return to quit -
the voice
view.

NetEngine AR

p

prefix
template
and enter
the prefix
template
view.

the call
prefix.

for the
prefix.

number length-value
that can be
parsed.
5 Return to quit -
the voice
view.
Configuring the Service Rights and Activating the Service Function

p
1 Enter the vms -

voice view.
2 Configure a vmsno vmsno callprefix callprefix-value -

voice
mailbox.
3 Return to quit -
the voice
view.

NetEngine AR

p

user view.
5 Configure service-right voicemailcfnr enable Set the

the message service-right
leaving parameter to
rights for voicemailcfn
the user. r.
6 Activate the service voicemailcfnr time vmsno -

message vmsno-value active enable
leaving
function.
7 Save the quit -

configuratio
n.


p

prefix
template
and enter
the prefix
template
view.

the call
prefix.


NetEngine AR

p

can be
parsed.
5 Save the save -

configuratio
n.
Example
Configure the CFNR voice mailbox service rights and activation prefix for user
7001.

[Huawei] voice
[Huawei-voice] vms

[Huawei-voice] vms
[Huawei-voice-pbxuser-7001] service-right voicemailcfnr enable
[Huawei-voice-pbxuser-7001] service voicemailcfnr vmsno 0 active enable

prefix is similar.
NOTE
Before configuring the service activation prefix, run the service voicemailcfnr vmsno 0

NetEngine AR

----End
Verification
consistent with the
settings.

service-right
mail on no reply : Enable
After the configuration is complete, verify the CFNR voice mailbox service. Assume
that the no-reply duration is set to 20s when activating the message leaving
function.
1. User B calls user A. User A does not answer the call within 20s. When hearing
the prompts, user B leaves messages according to the system prompts.
the SIP phone. If the terminal of user A is a POTS phone, the system notifies
NOTE

2.14.2.5 Configuring the CFB Voice Mailbox Service

After a user has registered the CFB voice mailbox service, the PBX transfers the
calls to the user to the voice mailbox if the user is busy. The callers leave
messages to this user according to the prompts of the voice mailbox system.

NetEngine AR
Prerequisites

p

system view.
2 Enter the voice -

voice view.
3 Enter the vms -

VMS view.

messages.
5 Return to quit -
the voice
view.

prefix
template
and enter
the prefix
template
view.

the call
prefix.

attributes
for the
prefix.

can be
parsed.
10 Return to quit -
the voice
view.

NetEngine AR

p

prefix
template
and enter
the prefix
template
view.

the call
prefix.

for the
prefix.

number length-value
that can be
parsed.
5 Return to quit -
the voice
view.

1 Enter the vms -

voice view.

voice
mailbox.
3 Return to quit -
the voice
view.

user view.

NetEngine AR
5 Configure service-right voicemailcfb enable Set the

rights for voicemailcfb.
the user.
6 Activate the service voicemailcfb vmsno vmsno-value -

message active enable
leaving
function.
7 Save the save -

configuratio
n.


p

prefix
template
and enter
the prefix
template
view.

the call
prefix.
3 Configure call-type category The call type category must

the call type callcategory attribute be set to new-

NetEngine AR

p

can be
parsed.
5 Save the save -

configuratio
n.
Example
Configure the CFB voice mailbox service rights and activation prefix for user 7002.

[Huawei] voice
[Huawei-voice] vms

[Huawei-voice] vms
[Huawei-voice-pbxuser-7002] service-right voicemailcfb enable
[Huawei-voice-pbxuser-7002] service voicemailcfb vmsno 0 active enable

prefix is similar.
NOTE
Before configuring the service activation prefix, run the service voicemailcfb vmsno 0

NetEngine AR
----End
Verification
consistent with the
settings.

service-right
mail on busy : Enable
After the configuration is complete, verify the CFB voice mailbox service.
1. User B calls user A. User A is busy. When hearing the prompts, user B leaves
messages according to the system prompts.
NOTE

2.14.2.6 Configuring the CFO Voice Mailbox Service

After a user has registered the CFO voice mailbox service, the PBX transfers the
calls to the user to the voice mailbox if the user is offline. The callers leave
messages to this user according to the prompts of the voice mailbox system.

NetEngine AR
Prerequisites

p

system view.
2 Enter the voice -

voice view.
3 Enter the vms -

VMS view.

messages.
5 Return to quit -
the voice
view.

prefix
template
and enter
the prefix
template
view.

the call
prefix.

attributes
for the
prefix.

can be
parsed.
10 Return to quit -
the voice
view.

NetEngine AR

p

prefix
template
and enter
the prefix
template
view.

the call
prefix.

for the
prefix.

number length-value
that can be
parsed.
5 Return to quit -
the voice
view.

p
1 Enter the vms -

voice view.

voice
mailbox.
3 Return to quit -
the voice
view.

NetEngine AR

p

user view.
5 Configure service-right voicemailcfo enable Set the

rights for voicemailcfo.
the user.
6 Activate the service voicemailcfo vmsno vmsno-value -

message active enable
leaving
function.
7 Save the save -

configuratio
n.


p

prefix
template
and enter
the prefix
template
view.

the call
prefix.


NetEngine AR

p

can be
parsed.
5 Save the save -

configuratio
n.
Example
Configure the CFO voice mailbox service rights and activation prefix for user 7003.

[Huawei] voice
[Huawei-voice] vms

[Huawei-voice] vms
[Huawei-voice-pbxuser-7003] service-right voicemailcfo enable
[Huawei-voice-pbxuser-7003] service voicemailcfo vmsno 0 active enable

prefix is similar.
NOTE
Before configuring the service activation prefix, run the service voicemailcfo vmsno 0

NetEngine AR
----End
Verification
consistent with the
settings.

service-right
mail on offline : Enable
After the configuration is complete, verify the CFO voice mailbox service.
1. User B calls user A. User A is offline. When hearing the prompts, user B leaves
messages according to the system prompts.
To configure the message indicator on the IP phone, such as an IP phone, see
the Administrator Guide of IP phone. If you need to configure the MWI mode
for POTS users, see mwi-mode for the detailed operations.
NOTE

2.14.2.7 Customizing Greeting Phrases

When a user enters the voice mailbox system, the user can hear the greeting
phrases.

NetEngine AR
Prerequisites
Configuring Prefixes of Greeting Phrases

p

system view.
2 Enter the voice -

voice view.
3 Enter the vms -

VMS view.
4 Configure vmsdata-storage-path vmsdata-storage- Before

the path for filepath-value configuring
storing voice the VMS, this
messages. path must be
specified.
5 Return to quit -
the voice
view.

prefix
template
and enter
the prefix
template
view.

the call
prefix.
8 Configure call-type category vu-service vu-service- Set the vu-

the call type name vuservicename service
and parameter to
attributes vuvmswelco
for the metone.
prefix.
9 Set the digit-length minimum-length-value -

length of a maximum-length-value
number that
can be
parsed.

NetEngine AR

p
10 Return to quit -
the voice
view.
Example
Set the prefix for greeting phrases to 10002.
[Huawei] voice
[Huawei-voice] vms
[Huawei-voice-callprefix-10002] call-type category vu-service vu-service-name vuvmswelcometone
Verification
consistent with the
settings.
User A dials the greeting phrases prefix, and records and saves the greeting
phrases according to the system prompts. User B calls user A. When the call is
transferred to the voice mailbox, user B hears the greeting phrases recorded by
user A.
NOTE
● By default, the greeting phrases does not include the keypad tone at the end.
● A user can dial the greeting phrases prefix on another phone, and enter its own number
and password to record the greeting phrases for its own voice mailbox. When record the
greeting phrases, you need to enter the service password.
● The default username and password are available in AR Router Default Usernames and
Passwords (Enterprise Network or Carrier). If you have not obtained the access
permission of the document, see Help on the website to find out how to obtain it. If you
use the password for the first time, the password must be modified. For details about
how to change the password, see 2.14.4 Configuring the Service Password.
2.14.3 Configuring Supplementary Telephony Services

NetEngine AR
2.14.3.1 Configuring the Number Query Service

After a user dials the number query prefix, the PBX announces the phone number
or ONLY number of the user.
Configuring Service Prefixes

Service prefixes include local number query prefix and ONLY number query prefix.
Configure the prefixes one by one according to the following steps.
Step Operation Command Description

view.

view.
3 Create a prefix callprefix callprefix- -

profile view and name
access the view.
4 Configure a prefix. prefix prefix -
5 Configure the call call-type category The call type category

type and call callcategory attribute must be set to
attribute for the attribute supplementary-service.
prefix. When configuring the
local number query
prefix, set attribute to
192.
When configuring the
ONLY number query
prefix, set attribute to
173.
6 Configure the digit-length -

minimum length minimum-length-value
and maximum maximum-length-
length of a number value
for number analysis.
7 Save the save -

configuration.

view.
Configuring the Service Right

By default, PBX users have been assigned the number query service right. If you
want to configure the service right, perform the steps described in the following
table.

NetEngine AR
1 Enter the PBX user pbxuser name

view.
2 Enable the number service-right query-dialno enable

query service right.
Set the local number query prefix and ONLY number query prefix to *192* and
*173* respectively, and activate service rights for user 6000.
[Huawei] voice # Configure the service prefix (In this example, the service prefix is *192*. The method of
configuring the ONLY number prefix
is the same.)
[Huawei-voice-callprefix-*192*] call-type category supplementary-service attribute 192
[Huawei-voice-callprefix-*192*] quit #
Configure the service right.
[Huawei-voice-pbxuser-6000] service-right query-dialno enable
Usage Example
This example assumes that user A has been assigned the number query service
right. The local number query prefix is *192* and ONLY number query prefix is
*173*.
User A hangs up and dials *192*. The PBX announces the local number to user A.
User A hangs up and dials *173*. The PBX announces the ONLY number to user A.
2.14.3.2 Configuring Calling Line Identification Services

Calling line identification (CLI) services determine whether calling numbers are
displayed on the phone of a called party. The services include calling line
identification presentation (CLIP), calling line identification restriction (CLIR), and
CLIR override. Calling parties and called parties can enable the services as
required.
2.14.3.2.1 Configuring CLIP

If a user number has been assigned the calling line identification presentation
(CLIP) service right, calling numbers will be displayed on the phone of this user
number.

NetEngine AR
By default, PBX users have been assigned the CLIP service right. If you want to
configure the service right, perform the steps described in the following table.

system view.
2 Enter the voice -

voice view.
3 Enter the PBX pbxuser name -

user view.
4 Enable the service-right clip enable To disable the CLIP service

CLIP service right, run the following
right. command:
service-right clip disable
5 Save the save -

configuration.
Enable the CLIP service right for user 6000.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right clip enable
Usage Example
This example assumes that user A has been assigned the CLIP service right.
● Scenario 1: User A is idle. If user B calls user A at this time, user B's number is
displayed on user A's phone.
● Scenario 2: User A has been assigned the call waiting service right and is
engaged in a call. If user B calls user A at this time, user B's number is
displayed on user A's phone.
NOTE
● If user B has been assigned the calling line identification restriction (CLIR) service right,
user B's number will not be displayed on user A's phone.
● In scenario 2, user A's phone needs to support CLIP in off-hook state. IP phones and
some POTS phones (for example, Panasonic KX-TSC11MX) support this function.
2.14.3.2.2 Configuring CLIR

If a user that has been assigned the calling line identification restriction (CLIR)
service right places a call, the number of this user (that is, the calling number) will
not be displayed on the called party's phone.

NetEngine AR

system view.
2 Enter the voice -

voice view.

user view.
4 Enable the service-right clir enable To disable the CLIR service

CLIR service right, run the following
right. command:
service-right clir disable
5 Save the save -

configuration.
Enable the CLIR service right for user 6000.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right clir enable
Usage Example
This example assumes that user A has been assigned the CLIR service right.
If user A places a call to user B, user A's number will not be displayed on user B's
phone regardless of whether user B has been assigned the CLIP service right.
NOTE
If user B has been assigned the CLIR override service right, user A's number will still be
displayed on user B's phone.
2.14.3.2.3 Enabling CLIR Override

After this service is enabled by a called party, the called party can see the phone
number of the calling party no matter whether the calling party has enabled the
calling line identification restriction (CLIR) service.

system view.

NetEngine AR
2 Enter the voice -

voice view.

user view.
4 Enable the service-right overstepclir To disable the CLIR

CLIR override enable override service right, run
service right. the following command:
service-right overstepclir
disable
5 Save the save -

configuration.
Enable the CLIR override service right for user 6000.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right overstepclir enable
Usage Example
This example assumes that user A has been assigned the CLIR override service
right.
If user B places a call to user A, user B's number is displayed on user A's phone.
NOTE
● Even if user B has been assigned the CLIR service right, user B's number is still displayed
on user A's phone.
● For incoming calls routed through the R2 trunk groups, the CLIR override service does
not take effect.
● When the calling party user B is an inter-office user, user A cannot see the phone
number of user B.
2.14.3.2.4 Temporarily Activating CLIR

This topic describes how to configure the service of temporarily activating the
calling line identification restriction (CLIR) service.
Generally, if the called party has the calling line identification presentation (CLIP)
service enabled, the calling party's number can be displayed to the called party.
However, if the calling party does not want to display the calling number to the
called party, the calling party can add a prefix to the called number during dialing
to temporarily activate the CLIR service.

NetEngine AR
Configuring a Prefix

view.

access the view.
5 Configure the call call-type category Set category (call

type and call callcategory attribute type) to
attribute for the attribute supplementary-
prefix. service and attribute
to 175.

minimum length and minimum-length-value
maximum length of maximum-length-
a number for value
number analysis.
7 Save the save -

configuration.
Configure prefix *175* for temporarily activating the CLIR service.
[Huawei] voice
Using the Service of Temporarily Activating the CLIR Service

This example assumes that user B has the CLIP service enabled.
1. User A picks up the phone and hears the dial tone.

2. If user A directly dials user B's number, user A hears the ringback tone, user
B's phone rings, and user A's number is displayed on user B's phone. If user A
dials *175*User B's number#, user A hears the ringback tone, user B's phone
rings, and user A's number is not displayed on user B's phone.
3. User B picks up the phone and talks with user A.
4. User A or B hangs up. The call ends.

NetEngine AR
2.14.3.2.5 Temporarily Deactivating CLIR
If the calling party has the CLIR service enabled, the calling number is not
displayed to the called party even when the called party has the CLIP service
enabled. However, if the calling party wants to display the calling number to the
called party, the calling party can add a prefix to the called number during dialing
to temporarily deactivate the CLIR service.


system view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right clir enable To disable the CLIR

CLIR service service right, run the
right. following command:
service-right clir disable
5 Return to quit -
the voice
view.
Configuring a Prefix

system view.
2 Enter the voice -

voice view.
3 Create a callprefix callprefix- -

prefix profile name
view and
access the
view.
4 Configure a prefix prefix -

prefix.

NetEngine AR
5 Configure call-type category Set category (call type)

the call type callcategory attribute to supplementary-
and call attribute service and attribute to
attribute for 176.
the prefix.
6 Configure digit-length minimum- -

the length-value maximum-
minimum length-value
length and
maximum
length of a
number for
number
analysis.
7 Save the save -

configuratio
n.
Configure the CLIR service for number 6000. Prefix *176* is the prefix for
temporarily deactivating the CLIR service.
[Huawei] voice
# Configure the service right.
[Huawei-voice-pbxuser-6000] service-right clir enable
# Configure a service prefix.
Using the Service of Temporarily Deactivating the CLIR Service

This example assumes that user A has been assigned the CLIR service right and
user B has the CLIP service enabled.
1. User A picks up the phone and hears the dial tone.

2. If user A directly dials user B's number, user A hears the ringback tone, user
B's phone rings, and user A's number is not displayed on user B's phone. If
user A dials *176*User B's number#, user A hears the ringback tone, user B's
phone rings, and user A's number is displayed on user B's phone.
3. User B picks up the phone and talks with user A.
4. User A or B hangs up. The call ends.

NetEngine AR
2.14.3.3 Configuring Name Identification Presentation Services

Name identification presentation services determine whether the name of a
calling user is displayed on the called user's phone.
2.14.3.3.1 Configuring CONP

A called user's name is displayed on the calling user's terminal.
Context
The Connected Name Identification Presentation (CONP) service displays a called
user's name on the calling user's terminal. Assume that A has the CONP service
rights. When A calls B, B's name is displayed on A's phone. If B has subscribed to
the explicit communication transfer (ECT) service, the call is transmitted to C.
When C answers the call, C's name is displayed on A's phone. If B belongs to a
user group, the name of the user group will be displayed on A's phone.
Requirements for implementing the CONP service are as follows:
● The phone of the calling user must support the CONP service and the
connected line identification presentation service (number obtained from the
PAI header field first).
● A name has been configured for the called user. Otherwise, there will be no
name displayed on the calling user's phone.
● During intra-office calls, the PBX user type must be SIP.
● During inter-office calls, the CONP service can be implemented only by SIP
users, SIP trunks, and H.323 trunks.
Prerequisites
A PBX user has been configured. For details about the configuration, see 2.8
Procedure

system view.
2 Enter the voice -

voice view.
3 Enter the PBX pbxuser name The CONP service right

user view. can be configured only by
SIP users.

NetEngine AR
4 Enable the service-right conp enable You can run the service-
CONP service right conp disable
right. command to delete the
CONP service right.
By default, the CONP
service right is disabled for
PBX users.
5 Save the save -

current
configuration.
Example
Enable the CONP service right for the user 6000.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right conp enable
Enable the CONP service right for an intra-office user A. Set the name of an out-
office user B to UserB. When A calls B over a SIP trunk, B's name UserB is
displayed on A's phone.
2.14.3.3.2 Configuring CNIP

A calling user's name is displayed on the called user's terminal.
Context
The Calling Name Identification Presentation (CNIP) service, also the called Calling
Line Identification Presentation (CLIP) service, displays a calling user's name on
the called user's terminal. If A expects a calling user's name to be displayed on the
terminal, A can enable the CNIP service. When A's terminal rings or A is answering
a call, the calling user's name is displayed on A's phone.
Requirements for implementing the CNIP service are as follows:
● The phone of the called user must support the CNIP service.
● A name has been configured for the calling user. Otherwise, there will be no
name displayed on the called user's phone.
● The CNIP service can be implemented only by SIP users, SIP trunks, and H.323
trunks.

NetEngine AR
Prerequisites
A PBX user has been configured. For details about the configuration, see 2.8
Procedure
By default, PBX users have the CNIP service right. To configure the CNIP service,
perform the following steps.

system view.
2 Enter the voice -

voice view.
3 Enter the PBX pbxuser name The CNIP service right can
user view. be configured only by SIP
users.
4 Enable the service-right cnip enable You can run the service-
CNIP service right cnip disable
right. command to delete the
CNIP service right.
5 Save the save -

current
configuration.
Example
Enable the CNIP service right for the user 6000.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right cnip enable
Set the name of an out-office user A to UserA. Enable the CNIP service right for
an intra-office user B. When A calls B over a SIP trunk, A's name UserA is
displayed on B's phone.
2.14.3.4 Configuring Call Forwarding Services

Call forwarding services forward incoming calls to a preset number when certain
conditions are met. Users can enable a call forwarding service based on service
requirements.

NetEngine AR
2.14.3.4.1 Configuring CFU

The call forwarding unconditional (CFU) service allows all incoming calls destined
for a user to be forwarded to a preset number regardless of the user status.
Configuring the Service Right and Forwarding Scheme


system view.

view.

user view.
4 Enable the CFU service-right cfu By default, PBX users have

service right. enable been assigned the CFU
service right.
To disable the CFU service
right, run the following
command:
service-right cfu disable
5 Configure the service cfu fwd-telno If the service prefixes are

forwarded-to configured, this step is not
number. required. Users can enable or
disable the service using
their phones.
6 Save the save -

configuration.

voice view.
(Optional) Configuring Service Prefixes

A service prefix is required for the PBX to allow terminal users themselves to
enable or disable a service.
Two service prefixes are involved: one for enabling the service and the other for
disabling the service. Perform the steps described in the following table to
configure the service prefixes one by one.

access the view.

NetEngine AR
3 Configure the call call-type category Set category (call type)

type and call callcategory attribute to new-supplementary-
attribute for the attribute management.
prefix. For the prefix used to
enable the service, set
attribute (call attribute)
to 21. For the prefix used
to disable the service, set
attribute to 22.

minimum length minimum-length-value
and maximum maximum-length-
length of a value
number for
number analysis.
5 Save the save -

configuration.

voice view.
Enable the CFU service right for user 6000 and set the forwarded-to number to
6001. On the PBX, set the prefix for enabling the CFU service to *21* and the
prefix for disabling the CFU service to *22*.
[Huawei] voice
# Configure the service right and forwarding scheme.
[Huawei-voice-pbxuser-6000] service-right cfu enable
[Huawei-voice-pbxuser-6000] service cfu 6001
# Configure a service prefix, for example, *21* for enabling the CFU service.
Enabling or Disabling the CFU Service

This example assumes that on the PBX, the prefix for enabling the CFU service is
set to *21* and that for disabling the CFU service is set to *22*. User A has been
assigned the CFU service right.
● Enabling the service
User A picks up the phone and dials *21*forwarded-to number. An
announcement is played, indicating that the service is enabled.

NetEngine AR
● Disabling the service

User A picks up the phone and dials *22*. An announcement is played,
indicating that the service is disabled.
Using the CFU Service

This example assumes that user A has been assigned the CFU service right and
enabled the CFU service that forwards calls to user B's number.
User C places a call to user A. The call is directly forwarded to user B.
2.14.3.4.2 Configuring CFNR

The call forwarding on no reply (CFNR) service allows calls destined for a user to
be automatically forwarded to a preset number if this user does not answer the
calls within a specified period.


system view.

view.

user view.
4 Enable the service-right cfnr By default, PBX users have

CFNR service enable been assigned the CFNR
right. service right.
To disable the CFNR service
command:
service-right cfnr disable
5 Configure the service cfnr { telno The default waiting time is

waiting time fwd-telno | time telno 30s, which means that a call
and fwd-telno } is forwarded if the called
forwarded-to party does not answer the
number. call within 30s.
If the service prefixes are
configured, this step is not
required. Users can enable or
disable the service using their
phones.
6 Save the save -

configuration.

voice view.

NetEngine AR


access the view.
3 Configure the call call-type category Set category (call type) to

type and call callcategory new-supplementary-
attribute for the attribute attribute management.
prefix. For the prefix used to
attribute (call attribute) to
25. For the prefix used to
disable the service, set
attribute to 26.

minimum length minimum-length-
and maximum value maximum-
length of a length-value
number for
number analysis.
5 Save the save -

configuration.

voice view.
Enable the CFNR service right for user 6000 and set the forwarded-to number to
6001. On the PBX, set the prefix for enabling the CFNR service to *25* and the
prefix for disabling the CFNR service to *26*.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right cfnr enable
[Huawei-voice-pbxuser-6000] service cfnr telno 6001

NetEngine AR
# Configure a service prefix, for example, *25* for enabling the CFNR service.
Enabling or Disabling the CFNR Service

This example assumes that on the PBX, the prefix for enabling the CFNR service is
set to *25* and that for disabling the CFNR service is set to *26*. User A has been
assigned the CFNR service right.
NOTE
The waiting time is 30s by default if a user dials a prefix to enable the CFNR service.
Using the CFNR Service

This example assumes that user A has been assigned the CFNR service right and
enabled the CFNR service that forwards calls to user B's number. The waiting time
is 30s.
User C places a call to user A. The call is forwarded to user B if user A does not
answer the call within 30s.
2.14.3.4.3 Configuring CFB

The call forwarding on busy (CFB) service allows calls destined for a user to be
automatically forwarded to a preset number if this user is busy.


system view.

view.

user view.

NetEngine AR
4 Enable the CFB service-right cfb enable By default, PBX users have
service right. been assigned the CFB
service right.
To disable the CFB service
command:
service-right cfb disable
5 Configure the service cfb fwd-telno If the service prefixes are

number. required. Users can enable
or disable the service using
their phones.
6 Save the save -

configuration.

voice view.


access the view.

type and call callcategory attribute to new-
prefix. management.
For the prefix used to
to 23. For the prefix
used to disable the
service, set attribute to
24.

NetEngine AR
4 Configure the digit-length minimum- -

minimum length length-value maximum-
and maximum length-value
length of a
number for
number analysis.
5 Save the save -

configuration.

voice view.
Enable the CFB service right for user 6000 and set the forwarded-to number to
6001. On the PBX, set the prefix for enabling the CFB service to *23* and the prefix
for disabling the CFB service to *24*.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right cfb enable
[Huawei-voice-pbxuser-6000] service cfb 6001
# Configure a service prefix, for example, *23* for enabling the CFB service.
Enabling or Disabling the CFB Service

This example assumes that on the PBX, the prefix for enabling the CFB service is
set to *23* and that for disabling the CFB service is set to *24*. User A has been
assigned the CFB service right.
Using the CFB Service

This example assumes that user A has been assigned the CFB service right and
enabled the CFB service that forwards calls to user B's number.
User C places a call to user A. The call is forwarded to user B if user A is already
engaged in a call.

NetEngine AR
2.14.3.4.4 Configuring CFO

The call forwarding offline (CFO) service allows calls destined for a SIP user to be
automatically forwarded to a preset number if this user is offline. If a SIP user
does not initiate a registration update request upon its registration failure, the PBX
sets the status of this SIP user to offline.


system view.

view.

user view.
4 Enable the service-right cfo enable By default, PBX users have

CFO service been assigned the CFO
right. service right.
To disable the CFO service
command:
service-right cfo disable
5 Configure the service cfo fwd-telno If the service prefixes are

their phones.
6 Save the save -

configuration.

voice view.


NetEngine AR

access the view.

type and call callcategory attribute to new-
prefix. management.
to 27. For the prefix
used to disable the
service, set attribute to
28.

minimum length length-value maximum-
and maximum length-value
length of a
number for
number analysis.
5 Save the save -

configuration.

voice view.
Enable the CFO service right for user 6000 and set the forwarded-to number to
6001. On the PBX, set the prefix for enabling the CFO service to *27* and the
prefix for disabling the CFO service to *28*.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right cfo enable
[Huawei-voice-pbxuser-6000] service cfo 6001
# Configure a service prefix, for example, *27* for enabling the CFO service.

NetEngine AR
Enabling or Disabling the CFO Service

This example assumes that on the PBX, the prefix for enabling the CFO service is
set to *27* and that for disabling the CFO service is set to *28*. User A has been
assigned the CFO service right.
Using the CFO Service

This example assumes that user A has been assigned the CFO service right and
enabled the CFO service that forwards calls to user B's number.
User C places a call to user A. The call is forwarded to user B if user A is offline.
2.14.3.4.5 Configuring CFC

The call forwarding conditional (CFC) service forwards calls meeting specified
conditions to preset numbers. The conditions include the specified called party
status, calling numbers, call attributes, and time segments.
The PBX does not allow users to dial the service prefix to register or cancel the
CFC service. An administrator can run the following commands to configure the
service rights and call forwarding plan.

system view.

view.

user view.
4 Enable the CFC service-right cfc To disable the CFC service

service right. enable right, run the following
command:
service-right cfc disable

NetEngine AR
5 Configure a forward-condition id The value of id (condition ID)

forwarding id-value forwardnum ranges from 0 to 9.
condition. forwardnum-value The options of state (called
[ umsno umsno- party status) are any, busy,
value ] state state- noanswer, offline, and
value [ caller-telno donotdisturb.
caller-telno-value ]
[ call-attribute call- The options of call-attribute
attribute-value ] (call attribute) are internal
[ weekly weekly- and external.
value ] [ begin-time ] The options of weekly (day)
[ end-time ] are sun, mon, tue, wed, thu,
fri, and sat.
The time format is hh:mm:ss.
6 Bind the service cfc condition -

forwarding condition-index-value
condition.
7 Save the save -

configuration.

voice view.
Enable the CFC service right for user 6000 and set the forwarded-to number to
6001. Specify the forwarding condition to 09:00-12:00 every Monday when the
user does not answer the calls.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right cfc enable
[Huawei-voice-pbxuser-6000] forward-condition id 1 forwardnum 6001 state noanswer weekly mon
09:00:00 12:00:00
[Huawei-voice-pbxuser-6000] service cfc condition 1
Using the CFC Service

This example assumes that user A has been assigned the CFC service right and the
forwarding condition is 09:00-12:00 every Monday when the user does not answer
the calls.
If user C places a call to user A between 09:00-12:00 on Monday, the call is
directly forwarded to user B.
2.14.3.5 Configuring Call Transfer

The call transfer service allows a user to transfer the ongoing call to a third-party
user.

NetEngine AR
Using the Call Transfer Service (POTS Phone)

There are two methods for the call transfer service: consult transfer and semi-
consult transfer. For details, see Figure 2-37.
Figure 2-37 Call transfer service through the POTS phone

User A and user B
talk with each
other User A presses the hookflash
or when user C does not
User A presses answer the call, user A
the hookflash. presses the hookflash
1 User A calls 2 User A 3
Call on hold Call being Call being
user C hangs up
(waiting for user transferred (user transferred (user
A to dial) C's phone rings) C's phone rings)
User C picks Consult Semi-consult User C picks

up the phone transfer transfer up the phone
4
User B and user C
Call being
talk with each
transferred (User User A
other
A and user C talk hangs up
with each other)
In Figure 2-37:
● : User A hears the special dial tone, and user B hears the call hold tone.
● : User A hears the ringback tone, and user B hears the call hold tone.
● : User B hears the ringback tone.
● : User B hears the call hold tone.

NOTE
● On POTS phones of some types such as TCL37 phones, you can press button R
(hookflash action).
● When hookflash cannot be implemented by pressing button R, run the flash-hook
lower command in the voice view to adjust the lower threshold for hookflash pressing.
Using the Call Transfer Service (IP Phone)

There are two methods for the call transfer service: busy transfer, consult transfer,
and semi-consult transfer. For details, see Figure 2-38. eSpace 7950 is used as an
example.

NetEngine AR
Figure 2-38 Call transfer service through the IP phone

User A and user B
talk with each
other
User A presses the
call transfer button
1 User A
User A dials the
presses the 3
Call being number of user C 2 Call being Call being
transferred done button
and press Call. transferred (user transferred (user
(waiting for user
C's phone rings) C's phone rings)
A to dial)
User C picks
User A dials the number User C picks Consult Semi-consult up the
of user C and presses the up the phone transfer transfer phone
3 done button 4
Call being Call being User B and user C
transferred (user transferred (user talk with each
C's phone rings) Blind transfer A and user C talk User A other
with each other) presses the
User C picks
up the phone done button
In Figure 2-38:
● : User A hears the dialing tone, and user B hears the call hold tone.
● : User A hears the ringback tone, and user B hears the call hold tone.
● : User B hears the ringback tone.
● : User B hears the call hold tone.
2.14.3.6 Configuring Call Hold

The call hold service allows a user to hold an ongoing call and resume it later.
When the call is placed on hold, the user can call a third party.
Usage Example
This example assumes that user A has been assigned the call hold service right.
If user A uses an analog phone, the service usage method and call state transition
are shown in Figure 2-39.

NetEngine AR
Figure 2-39 Call hold

User A and
user B talk
with each User A presses the
other hookflash or does
not perform any
User A calls User A operation
User C user C, and presses
hangs up user C picks the hookflash
up the phone User A does not
perform any 2 The call
3 The call 1 operation or
User A and The call between user the dial times between user A
between user and user B is
user C talk A and user B is placed out.
A and user B placed on hold
with each on hold (waiting for
is placed on (prompt before
other user A to dial)
hold call end)
User A presses the User A or

hookflash user B
hangs up User A or
4 The call user B
User A and hangs up
between user
The call user B talk The call
A and user C
ends with each between user
is placed on
other A and user B
hold
ends
User B
hangs up
User A and The call

user C talk between user
with each A and user B
other ends
The following describes the prompt tones played for each call state in Figure 2-39:
● State : User A hears a special dial tone, and user B hears a call hold tone.
● State : User A hears a busy tone, and user B hears a call hold tone.
● State : User B hears a call hold tone.
● State : User C hears a call hold tone.

NOTE
● In double communication state, if user A hangs up, user B and user C can hear the busy
tone, and the call is ended.
(hookflash action).
Multiple calls can be placed on hold at the same time on an IP phone. If using an
IP phone, user A can press the call hold button to place a current call on hold and
press the corresponding line button to resume the call.
2.14.3.7 Configuring Call Waiting

The call waiting service allows a user engaged in a call to hear the call waiting
tone (telling the user that a new call comes in) when a third party calls the user.

NetEngine AR


system view.

view.

user view.
4 Enable the call service-right cw By default, PBX users have

waiting service enable been assigned the call
right. waiting service right.
To disable the call waiting
service right, run the
following command:
service-right cw disable
5 Enable the call service-active cw If the service prefixes are

waiting service. configured, this step is not
required. Users can enable
their phones.
6 Save the save -

configuration.

voice view.


access the view.

NetEngine AR

type and call callcategory attribute type) to new-
prefix. management.
attribute (call
attribute) to 31. For
the prefix used to
attribute to 32.

a number for value
number analysis.
5 Save the save -

configuration.

view.
Enable the call waiting service for user 6000. On the PBX, set the prefix for
enabling the call waiting service to *31* and the prefix for disabling the call
waiting service to *32*.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right cw enable
[Huawei-voice-pbxuser-6000] service-active cw
# Configure a service prefix, for example, *31 for enabling the call waiting service.
Enabling or Disabling the Call Waiting Service

This example assumes that on the PBX, the prefix for enabling the call waiting
service is set to *31* and that for disabling the call waiting service is set to *32*.
User A has been assigned the call waiting service right.
indicating that the service is enabled.

NetEngine AR

Using the Call Waiting Service (POTS Phone)

This example assumes that user A has the call waiting service enabled.
For details about the usage and call status change, see Figure 2-40.
Figure 2-40 Call waiting using a POTS phone
User A and user

User C is idle B talk with each After user C hangs up,
other user A presses the
hookflash; or user A
User A waits for the operation timeout; or presses the hookflash
User C calls user A presses the hookflash and presses 0 and presses 1
user A
User A presses
the hookflash
and presses 2 The call between User A and user
Call waiting (user A hears the call
user A and user B C talk with
waiting tone)
is placed on hold each other
After user B hangs
User A presses the up, user A presses User A presses the
hookflash and presses 1 the hookflash; or hookflash and presses 2
user A presses the
hookflash and User A and user The call between
The call User A and user presses 1
between user A C talk with each B talk with each user A and user C
and user B ends other other is placed on hold
NOTE
(hookflash action).
Using the Call Waiting Service (IP Phone)

This example assumes that user A has the call waiting service enabled.
For details about the usage and call status change, see Figure 2-41. (for example,
eSpace 7950)
Figure 2-41 Call waiting using an IP phone

User A and user
User C is idle B talk with each
After user C
other
hangs up, user A
resumes the call
User C calls User A rejects the
user A call or waits for the
operation timeout
Call waiting User A answers The call between User A and user C
(User C's number is displayed on user A's the call user A and user B talk with each
phone) is placed on hold other
User A switches
the call.
After user B
The call between User A and user hangs up, user A User A and user B The call between
resumes the call
user A and user C talk with each talk with each user A and user C
B ends other other is placed on hold

NetEngine AR
2.14.3.8 Configuring Call Barring Services

This topic describes how the call barring services are defined, configured, and
used.
2.14.3.8.1 Configuring Number Barring

Certain rules can be specified to prevent a user from placing some types of calls.


system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right dlc enable To disable the number

number barring service right, run
barring the following command:
service service-right dlc disable
right.
5 Enable the service-active dlc If the service prefixes are

number configured, this step is not
barring required. Users can enable
service. or disable the service using
their phones.
6 Specify the service dlc dlc-telno The dlc-telno parameter

number to supports fuzzy match. An
have asterisk (*) indicates
number multiple digits. For
barring example, 7* means a calling
enabled. number starting with 7. An
x indicates one digit. For
example, 7x00 means a
four-digit number whose
hundreds place can be any
digit.
7 Return to quit -
the voice
view.

NetEngine AR


prefix
profile view
and access
the view.

prefix.
3 Configure call-type category Set category (call type) to

the call callcategory attribute new-supplementary-
type and attribute management.
call ● For the prefix used to
attribute change service password,
for the set attribute to 121.
prefix.
● For the prefix used to
query services, set
attribute to 87.
attribute to 88.
attribute to 89.
activate call barring, set
attribute to 90.
deactivate call barring,
set attribute to 91.
cancel call restriction of
all numbers, set
attribute to 92.

NetEngine AR

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configurati
on.
Configure number barring for number 7000, preventing the number from calling
intra-office user 7001. Prefix *90* is the prefix for activating number barring.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right dlc enable
[Huawei-voice-pbxuser-7000] service-active dlc
[Huawei-voice-pbxuser-7000] service dlc 7001
# Configure a service prefix, for example, *90* for activating call barring.
Activating and Deactivating Number Barring

This example assumes that the prefix for activating number barring is *90* and
that for deactivating number barring is *91*.
● Activating number barring: Dial *90*SSSSSS#.
● Deactivating number barring: Dial *91*SSSSSS#.
NOTE
The variable SSSSSS is the service password, The default username and password are
available in AR Router Default Usernames and Passwords (Enterprise Network or Carrier).
If you have not obtained the access permission of the document, see Help on the website
to find out how to obtain it. You must change the service password when you use the
service for the first time. For details about how to change the service password, see 2.14.4
Configuring the Service Password.

NetEngine AR
Enabling and Disabling the Number Barring Service
NOTICE
To enable or disable the number barring service for a user number, you can dial
Prefix for enabling or disabling the service + Service password + Called number
that has call barring enabled#. You can also dial Prefix for disabling the service +
Service password# to disable the service. Note that the AR supports only an at-
most-32-digit number. If the number you dial exceeds 32 digits, the AR does not
process it.

User A that has been assigned the number barring service right picks up the
phone, and dials *88*SSSSSSTN#. An announcement is played, indicating that
the service is enabled.
NOTE
● The variable TN indicates the number that user A cannot dial.

● If you cannot hear the system prompts when using the prefix to register service,
check whether the service information is correct, for example, whether the format,
prefix length, or user number length meet requirements and whether a conflict
occurs in the registered service.
User A picks up the phone, and dials *89*SSSSSSTN#. An announcement is
played, indicating that the service is disabled.
Using the Number Barring Service

This example assumes that user A has been assigned the number barring service
right.
User A initiates a restricted call. An announcement is played, indicating that the
call fails. User A initiates a call that is not restricted. The call is connected.
2.14.3.8.2 Configuring Call Barring

Users' call-out rights can be restricted as required.
A user's call-out rights depend on the call-out rights configured during number
allocation and the call barring policy enabled by the user. For example, although
international and national toll call rights are configured for a user during number
allocation, the user still cannot make international toll calls when the user enables
international toll call barring.

NetEngine AR

Step Operatio Command Description
n

system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right cba enable By default, the service right

call has been enabled for users.
barring To disable the call barring
service service right, run the
right. following command:
service-right cba disable
5 Configure service cba { all | IDD_DDD If the service prefixes are

restriction | IDD } configured, this step is not
rules. required. Users can enable
their phones.
6 Return to quit -
the voice
view.


prefix
profile
view and
access the
view.

a prefix.

NetEngine AR

call For the prefix used to enable
attribute the service, set attribute to
for the 98. For the prefix used to
prefix. disable the service, set
attribute to 101.

length
and
maximum
length of
a number
for
number
analysis.
5 Save the save -

configurat
ion.
Configure call barring for number 7000, preventing the number from making
international toll calls. Prefix *98* is the prefix for enabling call barring, and prefix
*101* is the prefix for disabling call barring.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right cba enable
[Huawei-voice-pbxuser-7000] service cba IDD
# Configure a service prefix, for example, *98* for enabling call barring. The procedure for configuring other
service prefixes is similar.
Enabling and Disabling the Call Barring Service

User A picks up the phone and dials *98*. When hearing the dial tone, user A
enters SSSSSSK#. An announcement is played, indicating that the service is
successfully enabled.

NetEngine AR
– When K is 1, the user cannot make any calls.

– When K is 2, the user cannot make local calls, national toll calls, or
international toll calls.
– When K is 3, the user cannot make international toll calls.
NOTE
available in AR Router Default Usernames and Passwords (Enterprise Network or
Carrier). If you have not obtained the access permission of the document, see Help on
the website to find out how to obtain it. You must change the service password when
you use the service for the first time. For details about how to change the service
password, see 2.14.4 Configuring the Service Password.
User A picks up the phone and dials *101*. When hearing the dial tone, user A
enters SSSSSSK#. An announcement is played, indicating that the service is
successfully disabled.
Using the Call Barring Service

Assume that user A has enabled the call barring service.
User A initiates a restricted call. An announcement is played, indicating that the

call fails. User A initiates a call that is not restricted. The call is connected.
2.14.3.8.3 Configuring Password Call

This topic describes how to configure the password call service.
The password call service restricts the calls that a user can make using a
password. The password used in the password call service is the service password.
For details about how to change the service password, see 2.14.4 Configuring the
Service Password.
When the user and the prefix of a call both have the password call service
enabled, the user can set up the call only by first dialing the call access code and
user password. It is recommended that the password call service be disabled for
intra-office prefixes and service prefixes. If not, users with the password call
service enabled need to first dial the call access code and user password even
before calling intra-office users or before dialing service prefixes. For details about
how to disable the password call service for a prefix, see pwdcalllimit disable.

n

system
view.
2 Enter the voice -

voice view.

NetEngine AR

n

PBX user
view.
4 Enable the service-right pwcb enable To disable the password call

password service right, run the
call following command:
service undo service-right pwcb
right.
5 Enable the service-active pwcb -

password
call
service.
6 Return to quit -
the voice
view.


prefix
profile view
and access
the view.

prefix.

call For the prefix used to
attribute enable the service, set
for the attribute to 95. For the
prefix. prefix used to disable the
service, set attribute to 96.

NetEngine AR

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configurati
on.
Configuring the Password Call Access Code


prefix
profile view
and access
the view.

prefix.
3 Configure call-type category vu- Set vu-service-name to

the call service vu-service-name vupwdcalllimit.
type and vuservicename
call
attribute
for the
prefix.
4 Disable the pwdcalllimit disable -

password
call service
right.

length and
maximum
length of a
number for
number
analysis.

NetEngine AR
6 Save the save -

configurati
on.
Configure password call for number 7000. Prefix *95* is the prefix for enabling
password call, and prefix *96* is the prefix for disabling password call. The
password call access code is 10003.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right pwcb enable
[Huawei-voice-pbxuser-7000] service-active pwcb
# Configure a service prefix, for example, *95* for enabling password call. The procedure for configuring
other service prefixes is similar.
[Huawei-voice-callprefix-*95*] pwdcalllimit disable
# Configure the password call access code.
[Huawei-voice-callprefix-10003] call-type category vu-service vu-service-name vupwdcalllimit
[Huawei-voice-callprefix-10003] pwdcalllimit disable
Enabling and Disabling the Password Call Service

This example assumes that user A has been assigned the password call service
right. The administrator has configured the prefix for enabling the service (*95*)
and the prefix for disabling the service (*96*) on the PBX.

User A that has been assigned the password call service right picks up the
phone, and dials *95*SSSSSS#. An announcement is played, indicating that the
service is enabled.
User A picks up the phone, and dials *96*SSSSSS#. An announcement is
Using the Password Call Service

This example assumes that user A has been assigned the password call service
right.
User A makes a call to user B. An announcement is played, indicating the call fails.

NetEngine AR
User A dials the password call access code, and enters the password and user B's
number in sequence as prompted. Then the call is connected.
NOTE
available in AR Router Default Usernames and Passwords (Enterprise Network or Carrier).
If you have not obtained the access permission of the document, see Help on the website
to find out how to obtain it. You must change the service password when you use the
service for the first time. For details about how to change the service password, see 2.14.4
Configuring the Service Password.
2.14.3.8.4 Configuring the Blacklist and Whitelist

After a trunk group is configured, users who meet the call conditions specified for
the trunk group can enjoy call services using this trunk group. To restrict the use
of this trunk group, you can configure a whitelist to allow specified users to use
the trunk group or configure a blacklist to prevent specified users from using the
trunk group.
Context
A rule consists of the rule attribute (two options: permit and deny) and the calling
number.
A rule set is a collection of blacklist/whitelist rules.
Check the blacklist and whitelist according to the following rules:
1. Check whether a whitelist is available. If a whitelist is available, check
whether the calling party belongs to the whitelist. If the calling party belongs
to the whitelist, the system selects a route for the call; otherwise, the system
rejects the call.
2. Check whether a blacklist is available. If a blacklist is available, check whether
the calling party belongs to the blacklist. If the calling party belongs to the
blacklist, the system rejects the call; otherwise, the system selects a route for
the call.
The call right of a number must be permit if the calls from this number want to
be routed over a trunk that is bound to a rule set.
This topic assumes that the blacklist/whitelist rule set is configured as follows:
● The value of outgoing is set to permit for intra-office user 7000.
● The value of outgoing is set to deny for intra-office user 7001.
● The value of incoming is set to permit for external user 1000.
Figure 2-42 shows the networking.

NetEngine AR
Figure 2-42 Blacklist/Whitelist networking
PBX IP PBX

7000 7001 1000
If the trunk group connecting the PBX to the IP PBX has the preceding blacklist/
whitelist rule set configured, user 1000 can call user 7000 and user 7001, user
7000 can call user 1000, but user 7001 cannot call user 1000.
Prerequisites
The desired trunk group has been configured. For details, see 2.11 Configuring
Trunk Groups.
3 Configure a rule set. rule-set Rule set -

name

NetEngine AR
4 Configure a rule. rule { deny | permit } Rule deny indicates a

{ incoming | blacklist, while rule
outgoing } { caller- permit indicates a
telno caller-telno- whitelist.
value | called-telno The calling number
called-telno-value } caller-telno supports
fuzzy matching. *
indicates multiple
digits, for example, 7*
indicates a number
starting with 7. x
indicates one digit, for
example, 7x00
indicates a 4-digit
number in which x is
any numeral.
The called number
called-telno supports
fuzzy matching. *
indicates multiple
digits, for example, 7*
indicates a number
starting with 7. x
indicates one digit, for
example, 7x00
indicates a 4-digit
number in which x is
any numeral.
5 (Optional) Configure rule disconnect- By default, the error

the cause code cause incoming code call reject is
returned when the returned when an
incoming call is incoming call is
blocked by the blocked by the
blacklist. blacklist.
If the enterprise
wants to return a
specific cause value to
the peer end rather
than the default call-
reject, you can run
the rule disconnect-
cause incoming
command to
configure the cause
value.

view.

NetEngine AR
7 Enter the group trunk trunk-group name -

view.
8 Configure the rule set use-rule-set rule-set- -

for the trunk group. name
9 Save the save -
configuration.
Configure blacklist/whitelist rule set 1, where user 7000 is whitelisted for outgoing
calls, user 7001 is blacklisted for outgoing calls, and user 1000 is blacklisted for
incoming calls. Bind trunk group sipip to blacklist/whitelist rule set 1.
[Huawei] voice
[Huawei-voice] rule-set 1
[Huawei-voice-rule-set-1] rule permit outgoing caller-telno 7000
[Huawei-voice-rule-set-1] rule deny outgoing caller-telno 7001
[Huawei-voice-rule-set-1] rule deny incoming caller-telno 1000
[Huawei-voice-rule-set-1] quit
[Huawei-voice] trunk-group sipip
[Huawei-voice-trunkgroup-sipip] use-rule-set 1
[Huawei-voice-trunkgroup-sipip] save
Usage Example
To prevent a user from placing outgoing calls over a trunk, configure a blacklist/
whitelist rule set, set outgoing of the user to deny in the rule set, and bind the
rule set to that trunk.
2.14.3.8.5 Configuring RAC

This topic describes how to configure the reject anonymous call (RAC) service.
In an anonymous call, the calling number is not displayed to the called party. The
RAC service rejects an anonymous call, and plays an announcement to the calling
party. The RAC service can be configured to prevent harassment of anonymous
calls.


system
view.
2 Enter the voice -

voice view.

NetEngine AR

PBX user
view.
4 Enable the service-right rac enable To disable the RAC service

RAC service right, run the following
right. command:
service-right rac disable

NetEngine AR
5 Configure service rac [ tone-id tone- If the service prefixes are

the RAC id ] configured, this step is not
service. required. Users can enable
their phones.
The default value of tone-id
is 539. If you want to set
the parameter to a non-
default value, you can
achieve this only by using
the preceding command.
On a phone, you can set
the parameter only to the
default value.
Corresponding control
points need to be enabled
for different trunks;
otherwise, calling parties
cannot hear the prompt
tones.
● For incoming
anonymous calls routed
through SIP and H.323
trunks, enable control
point 120 (for playing
announcements to the
peer party) in the trunk
view on the PBX. For
details, see number-
parameter.
● For incoming
anonymous calls routed
through the other
trunks, enable control
point 202 (playing
announcements on the
trunk side) on the PBX.
For details, see pbx
number-parameter.
NOTICE
For incoming anonymous calls
routed through R2 trunks,
besides enabling control point
202 on the local end, enable
local early media on the peer
end. For details, see r2-
receive-earlymedia.

NetEngine AR
6 Return to quit -
the voice
view.


prefix
profile view
and access
the view.

prefix.

the call type callcategory attribute new-supplementary-
and call attribute management.
attribute for For the prefix used to
the prefix. enable the service, set
attribute to 39. For the
prefix used to disable the

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configuratio
n.

NetEngine AR
Configure RAC for number 7000. Prefix *39* is the prefix for enabling RAC, and
prefix *40* is the prefix for disabling RAC.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right rac enable
[Huawei-voice-pbxuser-7000] service rac
# Configure a service prefix, for example, *39* for enabling RAC. The procedure for configuring other service
prefixes is similar.
Enabling and Disabling the RAC Service

User A that has been assigned the RAC service right picks up the phone, and
dials *39*#. An announcement is played, indicating that the service is enabled.
User A picks up the phone, and dials *40*#. An announcement is played,
Using the RAC Service

This example assumes that user A has configured the RAC service.
1. User B makes an anonymous call to user A.
2. User B hears an announcement, indicating that the call is rejected.
2.14.3.8.6 Configuring DND

The do not disturb (DND) service blocks all incoming calls for a user. If the DND
service is enabled for a user, others who call this user hear the DND
announcement, and no call is put through to the user. With this service enabled,
the user cannot receive any calls but can still place calls.


system
view.
2 Enter the voice -

voice view.

PBX user
view.

NetEngine AR
4 Enable the service-right dnd enable To disable the DND service

DND right, run the following
service command:
right. service-right dnd disable
5 Enable the service dnd If the service prefixes are

DND configured, this step is not
their phones.
6 Return to quit -
the voice
view.


prefix
profile view
and access
the view.

prefix.

for the attribute (call attribute) to
prefix. 35. For the prefix used to
attribute to 36.

NetEngine AR

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configuratio
n.
User A has enabled the DND service. The administrator configures service
registration prefix (for example, *35*) and service deregistration prefix (for
example, *36*) on the PBX.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right dnd enable
[Huawei-voice-pbxuser-7000] service dnd
# Configure a service prefix, for example, *35* for enabling the DND service.
Enabling or Disabling the DND Service

This example assumes that on the PBX, the prefix for enabling the DND service is
set to *35* and that for disabling the DND service is set to *36*.

User A that has been assigned the DND service right picks up the phone and
dials *35*#. An announcement is played, indicating that the service is enabled.
User A picks up the phone and dials *36*#. An announcement is played,
Using the DND Service

User A has been assigned the DND service right.
Others call user A and hear the DND announcement or the busy tone. User A can
still place calls.

NetEngine AR
2.14.3.8.7 Configuring SCA

Conditions can be specified for selective call acceptance (SCA). The system accepts
calls that meet the specified conditions. The SCA service helps mitigate the impact
of frequent calls.
Configuring the Service Right and Judgment Group


system view.
2 Enter the voice -

voice view.

user view.
4 Enable the service-right sca To disable the SCA service

SCA service enable right, run the following
right. command:
service-right sca disable

NetEngine AR
5 Configure a service sca judgegrp Judgment group conditions

judgment judgegrp-value include the calling number,
group and [ condition { caller- time range, and interval. One
enable it. telno caller-telno-value or more conditions can be
| time-period { from specified for a judgment
from-date [ from- group.
time ] | to to-date [ to- The parameters are described
time ] } * | time-repeat as follows:
{ { yearly |
● judgegrp: indicates the
monthly }begin-date
judgment group ID, which
[ begin-time ] [ end-
ranges from 0 to 99.
date [ end-time ] ] |
weekly begin-weekday ● caller-telno: indicates the
[ begin-time ] [ end- calling number, which
weekday [ end-time ] ] supports fuzzy match. The
| daily begin-time asterisk (*) represents
[ end-time ] }* }*] multiple digits. For
[ active active-value ] example, 6* indicates a
number starting with 6.
The variable x represents
one digit. For example,
6x00 is a four-digit number,
with x being any digit.
● Date format: yyyy-mm-dd.
● Time format: hh:mm:ss.
● weekly: indicates a day,
with options being sun,
mon, tue, wed, thu, fri,
and sat.
required. Users can dial the
corresponding service prefix
for the related function using
their phones.
6 Save the save -

configuration.

voice view.

enable judgment group conditions and to enable and disable a judgment group.
The service prefixes include those used to enable/disable the SCA number, time,
and day and to query/delete/enable/disable a judgment group.

NetEngine AR
Perform the steps described in the following table to configure the service prefixes
one by one.

access the view.

prefix.
3 Configure the call-type category Set category (call type) to

call type and call callcategory new-supplementary-
prefix. Set attribute (call attribute)
to the following values for
the specific service prefixes:
enable an SCA number: 59
disable an SCA number:
60
enable an SCA time
period: 61
disable an SCA time
period: 62
enable an SCA day: 63
disable an SCA day: 64
enable a judgment group:
65
disable a judgment group:
66
verify a judgment group:
67
delete a judgment group:
68

NetEngine AR

number for
number analysis.
5 Save the save -

configuration.

voice view.
Enable the SCA service right for user 6000. Enable the system to accept calls for
user 6000 when either of the following conditions are met:
● The calling number is 6001 every Tuesday to Friday.
● 10:00-17:00 every day from 2013-01-01 to 2014-12-31.
On the PBX, set a service prefix to *+Call attribute of the service prefix+*. For
example, set the prefix used to enable an SCA number to *59*.
# Configure the service right and judgment group.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right sca enable
[Huawei-voice-pbxuser-6000] service sca judgegrp 1 condition caller-telno 6001 time-repeat weekly
tue fri active enable
[Huawei-voice-pbxuser-6000] service sca judgegrp 2 condition time-period from 2013-01-01 to
2014-12-31 time-repeat daily 10:00:00 17:00:00 active enable
# Configure a service prefix, for example, *59* for enabling an SCA number.
Enabling or Disabling the SCA Service

This example assumes that on the PBX, a service prefix is set to *+Call attribute of
the service prefix+*, for example, the prefix used to enable an SCA number being
set to *59*.
User A has been assigned the SCA service right.
NOTE
If you cannot hear the system prompts when using the prefix to operate service, check
whether the service information is correct, for example, whether the format, prefix length,
or user number length meet requirements and whether a conflict occurs in the service.
User A can enable the judgment group conditions in the following ways:

NetEngine AR
● Adding an SCA number to a judgment group: User A picks up the phone and
dials *59*+2-digit group ID+Accepted user number#. For example, if user A
dials *59*0128971799#, accepted number 28971799 is added to judgment
group 1.
● Adding an SCA time period to a judgment group: User A picks up the phone
and dials *61*+2-digit group ID+Accepted time period#. For example, if user A
dials *61*0208301030#, accepted time period 08:30-10:30 everyday is added
to judgment group 2.
● Adding SCA days to a judgment group: User A picks up the phone and dials
*63*+2-digit group ID+Accepted days in a week#. The value for accepted days
in a week is a 7-digit binary number, with the first digit being Monday and
last digit being Sunday. For example, if user A dials *63*030111110#, accepted
days Tuesday to Saturday are added to judgment group 3.
NOTE
● A judgment group can contain one condition, or a combination of number and time or
number and day.
● Time and day are mutually exclusive. If the time and day conditions are both enabled,
the condition that is enabled later takes effect.
User A can disable the judgment group conditions in the following ways:
● Disabling the number condition from a judgment group: User A picks up the
phone and dials *60*+2-digit group ID#, for example, *60*01#.
● Disabling the time condition from a judgment group: User A picks up the
● Disabling the day condition from a judgment group: User A picks up the
User A can enable one or more judgment groups in the following ways:
● Enabling a judgment group: User A picks up the phone and dials *65*+2-digit
group ID#, for example, *65*01#.
● Enabling all judgment groups: User A picks up the phone and dials *65*#.
User A can disable one or more judgment groups in the following ways:
● Disabling a judgment group: User A picks up the phone and dials *66*+2-digit
● Disabling all judgment groups: User A picks up the phone and dials *66*#.
User A can verify a judgment group or delete one or more judgment groups in the
following ways:
● Verifying a judgment group: User A picks up the phone and dials *67*+2-digit
group ID#, for example, *67*03#. If the judgment group is enabled, an
announcement is played, indicating that the judgment group is enabled.
Otherwise, an announcement is displayed, indicating a failure to enable the
judgment group.
● Deleting a judgment group: User A picks up the phone and dials *68*+2-digit
● Deleting all judgment groups: User A picks up the phone and dials *68*#.

NetEngine AR
Using the SCA Service

This example assumes that user A has been assigned the SCA service right and
enabled a judgment group with SCA conditions specified.
User B places a call to user A. If the SCA conditions are met, the PBX connects the
call to user A. Otherwise, the PBX releases the call.
2.14.3.8.8 Configuring SCR

Conditions can be specified for selective call rejection (SCR). The system rejects
calls that meet the specified conditions. The SCR service helps mitigate the impact
of frequent calls.
Configuring the Service Right and Judgment Group


system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right rcs enable To disable the SCR service right,
SCR service run the following command:
right. service-right rcs disable

NetEngine AR
5 Configure a service rcs judgegrp Judgment group conditions

judgment judgegrp-value include the calling number, time
group and [ condition { caller- range, and interval. One or
enable it. telno caller-telno-value more conditions can be
| time-period { from specified for a judgment group.
from-date [ from-time ] The parameters are described
| to to-date [ to-time ] } as follows:
* | time-repeat
● judgegrp: indicates the
{ { yearly |
judgment group ID, which
monthly }begin-date
ranges from 0 to 99.
[ begin-time ] [ end-
date [ end-time ] ] | ● caller-telno: indicates the
weekly begin-weekday calling number, which
[ begin-time ] [ end- supports fuzzy match. The
weekday [ end-time ] ] | asterisk (*) represents
daily begin-time [ end- multiple digits. For example,
time ] }* }*] [ active 6* indicates a number
active-value ] starting with 6. The variable
x represents one digit. For
example, 6x00 is a four-digit
number, with x being any
digit.
● Date format: yyyy-mm-dd.
● Time format: hh:mm:ss.
● weekly: indicates a day, with
options being sun, mon, tue,
wed, thu, fri, and sat.
required. Users can dial the
corresponding service prefix for
the related function using their
phones.
6 Save the save -

configurati
on.
7 Return to quit -
the voice
view.

enable judgment group conditions and to enable and disable a judgment group.
The service prefixes include those used to enable/disable the SCR number, time,
and day and to query/delete/enable/disable a judgment group.

NetEngine AR
Perform the steps described in the following table to configure the service prefixes
one by one.

access the view.

prefix.

call type and call callcategory new-supplementary-
prefix. Set attribute (call attribute)
to the following values for the
specific service prefixes:
enable an SCR number: 49
disable an SCR number: 50
enable an SCR time period:
51
disable an SCR time period:
52
enable an SCR day: 53
disable an SCR day: 54
enable a judgment group:
55
disable a judgment group:
56
verify a judgment group:
57
delete a judgment group:
58

NetEngine AR

number for
number analysis.
5 Save the save -

configuration.

voice view.
Enable the SCR service right for user 6000. Enable the system to reject calls for
user 6000 when either of the following conditions are met:
● The calling number is 6001 every Monday and Tuesday.
● 09:00-10:00 every day from 2013-01-01 to 2014-12-31.
On the PBX, set a service prefix to *+Call attribute of the service prefix+*. For
example, set the prefix used to enable an SCR number to *49*.
# Configure the service right and judgment group.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right rcs enable
[Huawei-voice-pbxuser-6000] service rcs judgegrp 1 condition caller-telno 6001 time-repeat weekly
mon tue active enable
[Huawei-voice-pbxuser-6000] service rcs judgegrp 2 condition time-period from 2013-01-01 to
2014-12-31 time-repeat daily 09:00:00 10:00:00 active enable
# Configure a service prefix, for example, *49* for enabling an SCR number.
Enabling or Disabling the SCR Service

This example assumes that on the PBX, a service prefix is set to *+Call attribute of
the service prefix+*, for example, the prefix used to enable an SCR number being
set to *49*.
User A has been assigned the SCR service right.
NOTE
If the system prompts an error when you use the service prefixes, check whether the
registration information is correct, for example, the format and prefix length. In addition,
check whether a conflict occurs in the service.
User A can enable the judgment group conditions in the following ways:

NetEngine AR
● Adding an SCR number to a judgment group: User A picks up the phone and
dials *49*+2-digit group ID+Rejected user number#. For example, if user A
dials *49*0128971799#, rejected number 28971799 is added to judgment
group 1.
● Adding an SCR time period to a judgment group: User A picks up the phone
and dials *51*+2-digit group ID+Rejected time period#. For example, if user A
dials *51*0208301030#, rejected time period 08:30-10:30 everyday is added to
judgment group 2.
● Adding SCR days to a judgment group: User A picks up the phone and dials
*53*+2-digit group ID+Rejected days in a week#. The value for rejected days
in a week is a 7-digit binary number, with the first digit being Monday and
last digit being Sunday. For example, if user A dials *53*030111110#, rejected
days Tuesday to Saturday are added to judgment group 3.
NOTE
● A judgment group can contain one condition, or a combination of number and time or
number and day.
● Time and day are mutually exclusive. If the time and day conditions are both enabled,
the condition that is enabled later takes effect.
User A can disable the judgment group conditions in the following ways:
● Disabling the number condition from a judgment group: User A picks up the
● Disabling the time condition from a judgment group: User A picks up the
● Disabling the day condition from a judgment group: User A picks up the
User A can enable one or more judgment groups in the following ways:
● Enabling a judgment group: User A picks up the phone and dials *55*+2-digit
● Enabling all judgment groups: User A picks up the phone and dials *55*#.
User A can disable one or more judgment groups in the following ways:
● Disabling a judgment group: User A picks up the phone and dials *56*+2-digit
● Disabling all judgment groups: User A picks up the phone and dials *56*#.
User A can verify a judgment group or delete one or more judgment groups in the
following ways:
● Verifying a judgment group: User A picks up the phone and dials *57*+2-digit
group ID#, for example, *57*03#. If the judgment group is enabled, an
announcement is played, indicating that the judgment group is enabled.
Otherwise, an announcement is displayed, indicating a failure to enable the
judgment group.
● Deleting a judgment group: User A picks up the phone and dials *58*+2-digit
● Deleting all judgment groups: User A picks up the phone and dials *58*#.

NetEngine AR
Using the SCR Service

This example assumes that user A has been assigned the SCR service right and
enabled a judgment group with SCR conditions specified.
User B places a call to user A. If the SCR conditions are met, the PBX releases the
call. Otherwise, the PBX sends the call request to user A.
2.14.3.9 Configuring Call Back Services

This topic describes how to configure and use call back services.
2.14.3.9.1 Configuring CBB

When a user makes a call to an enterprise user and the enterprise user is busy, the
system prompts the user to enable the call back on busy (CBB) service.


system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right ccbs enable To disable the CBB service

CBB service right, run the following
right. command:
service-right ccbs disable
5 Return to quit -
the voice
view.


NetEngine AR

prefix
profile view
and access
the view.

prefix.

type and attribute management and
call attribute to 104.
attribute
for the
prefix.

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configuratio
n.
Enable CBB service right for number 7000. Prefix *104* is the prefix for disabling
the CBB service.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right ccbs enable
Enabling and Disabling the CBB Service

This example assumes that user A has been assigned the CBB service right. The
administrator has configured the prefix for disabling the service (*104*) on the
PBX.

NetEngine AR

a. User A makes a call to an enterprise user, and the enterprise user is busy.
b. User A hears an announcement, prompting the user to enable the CBB
service.
c. User A dials 1 to enable the service.
User A picks up the phone, and dials *104*. An announcement is played,
NOTE
If call back is successfully performed after the CBB service is enabled, the service is
automatically disabled.
Using the CBB Service

● The called party is an intra-office user.
After a user enables the CBB service, the system monitors the status of the
called party. When detecting that the called party is idle, the system makes a
call to the called party, and then makes a call to the calling party when the
called party picks up the phone. If the call fails to be connected, the call
attempt process is similar to that where the called party is an outer-office
user.
● The called party is an outer-office user.
After a user enables the CBB service, the system makes a call to the called
party at the end of the first call attempt interval, and then makes a call to the
calling party when the called party picks up the phone. If the call is
successfully connected, the system stops initiating call attempts. If the call
fails to be connected, the system initiates a call attempt at the end of the
next call attempt interval. When the number of call attempts reaches the
maximum or the total time period reaches the maximum, the system stops
initiating call attempts.
NOTE
The call back control points are as follows:

● 68: call attempt time period
● 69: maximum number of call attempts
● 70: call attempt interval
You can run the pbx number-parameter command to modify call back time control points.
2.14.3.9.2 Configuring ACB

The automatic callback (ACB) service enables a user who makes a call to register
the call when the called user does not answer the call.

NetEngine AR


system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right ccnr enable To disable the ACB service

ACB service right, run the following
right. command:
service-right ccnr disable
5 Return to quit -
the voice
view.


prefix
profile view
and access
the view.

prefix.

type and attribute management and
call attribute to 107.
attribute
for the
prefix.

NetEngine AR

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configuratio
n.
Enable ACB service right for number 7000. Prefix *107* is the prefix for disabling
the ACB service.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right ccnr enable
Enabling and Disabling the ACB Service

This example assumes that user A has been assigned the ACB service right. The
administrator has configured the prefix for disabling the service (*104*) on the
PBX.
a. User A makes a call to an enterprise user, and the enterprise user does
not answer the call.
b. User A hears an announcement, prompting the user to enable the ACB
service.
c. User A dials 1 to enable the service.
User A picks up the phone, and dials *107*. An announcement is played,
NOTE
If call back is successfully performed after the ACB service is enabled, the service is
automatically disabled.

NetEngine AR
Using the ACB Service

After a user has the ACB service enabled, the system calls the called party when
the call attempt interval expires, and the system calls the calling party after the
called party goes offhook. If the call attempt fails, the system initiates another call
attempt when the next call attempt interval expires. If the call attempt is
successful, the system does not initiate a call attempt again; but if the call
attempt still fails, the system initiates another call attempt when the next call
attempt interval expires. When the maximum number of call attempts or the
maximum call attempt duration is reached, the system no longer initiates another
call attempt even if the previous call attempt fails.
NOTE
The call back control points are as follows:

● 68: call attempt time period
● 69: maximum number of call attempts
● 70: call attempt interval
You can run the pbx number-parameter command to modify call back time control points.
2.14.3.10 Configuring Pickup Services

This topic describes how the pickup services are defined, configured, and used.
2.14.3.10.1 Configuring Co-group Pickup

The co-group pickup service allows a user to answer the calls for another user in
the same pickup group by dialing the pickup access code.
3 Enter the PBX user pbxuser name -

view.
4 Enable the co-group service-right pickup- To disable the co-

pickup service right. in-group enable group pickup service
right, run the
following command:
service-right pickup-
in-group disable
5 Add a user to the co- pickupgroup-name -

group pickup group. pickupgroup-name-
value
view.

NetEngine AR
7 Create a prefix profile callprefix callprefix- -

view and access the name
view.
8 Configure a co-group prefix prefix -

pickup prefix.
9 Configure the call type call-type category Set category (call

and call attribute for callcategory attribute type) to
the prefix. attribute supplementary-
service.
attribute (call
attribute) to 171.

minimum length and minimum-length-
maximum length of a value maximum-
number for number length-value
analysis.
11 Save the save -

configuration.
Configure the co-group pickup service rights for user 7000 and add user 7000 to
group aaa. Add user 7001 to group aaa. Set the co-group pickup prefix to *171*.
# Configure the co-group pickup service.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right pickup-in-group enable
[Huawei-voice-pbxuser-7000] pickupgroup-name aaa
Usage Example
This example assumes that user A has been assigned the co-group pickup service
right, user A and user B are intra-office users, user C is an external user, and the
co-group pickup prefix is *171*.
1. User C dials user B's number, and user B's phone rings.

NetEngine AR
2. User A picks up his or her own phone and dials *171*. User B's phone stops
ringing.
3. User A talks with user C.
NOTE
If multiple phones in the same pickup group are ringing, the system automatically
selects the phone that rings first for pickup.
2.14.3.10.2 Configuring Designated Pickup

The designated pickup service allows a user to answer the calls for another user
by dialing the pickup access code and called number.

view.
4 Enable the designated service-right pickup- To cancel the service

pickup service right. special enable right, run the
following command:
service-right pickup-
special disable
5 Add a user to the pickupgroup-name -

designated pickup pickupgroup-name-
group. value
view.
7 Create a prefix profile callprefix callprefix- -

view and access the name
view.

designated pickup
prefix.
9 Configure the call type call-type category Set category (call

and call attribute for callcategory attribute type) to
the prefix. attribute supplementary-
service.
attribute (call
attribute) to 172.

NetEngine AR

minimum length and minimum-length-
maximum length of a value maximum-
number for number length-value
analysis.
11 Save the save -

configuration.
Configure the designated pickup service rights for user 7000. Add users 7000 and
7001 to the designated pickup group aaa. Set the designated pickup prefix to
*172*.
# Configure the designated pickup service.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right pickup-special enable
Usage Example
This example assumes that user A has been assigned the designated pickup
service right and the designated pickup prefix is set to *172*.
1. User C dials user B's number, and user B's phone rings.
2. User A picks up his or her own phone and dials *172*TN#. User B's phone
stops ringing. The variable TN indicates user B's number.
3. User A talks with user C.
2.14.3.11 Configuring Secretary Services

This topic describes how to configure and use secretary services.
2.14.3.11.1 Configuring the Secretary Service

All the calls made to the manager are forwarded to the secretary, and only the
secretary can put calls through to the manager.

NetEngine AR


system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right scr enable To disable the secretary

secretary service right, run the
service following command:
right. service-right scr disable
5 Enable the service scr telno telno- If the service prefixes are
secretary value configured, this step is not
their phones.
6 Return to quit -
the voice
view.


prefix
profile view
and access
the view.

prefix.

NetEngine AR

for the attribute to 72. For the
prefix. prefix used to disable the

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configuratio
n.
Configure the secretary service for number 7000. Prefix *72* is the prefix for
enabling the secretary service, and prefix *73*# is the prefix for disabling the
secretary service.
# Configure user 7000 as the manager and user 7001 as the secretary.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right scr enable
[Huawei-voice-pbxuser-7000] service scr telno 7001
# Configure a service prefix, for example, *72* for enabling the secretary service. The procedure for
configuring other service prefixes is similar.
Enabling and Disabling the Secretary Service

This example assumes that users A and B have been assigned the secretary service
right. The administrator has configured the prefix for enabling the service (*72*)
and the prefix for disabling the service (*73*) on the PBX.
Assumes that user A is a manager who has been assigned the secretary
service right, and user A's number is DN. User B has been assigned the
secretary rights, and user B's number is TN.

NetEngine AR
If user A wants to register user B as the secretary, user A can pick the phone
and dial *72*TN#. An announcement is played, indicating that the service is
enabled.
If user B wants to register user A as the manager, user B can pick the phone
and dial *72*DN*SSSSSS#. An announcement is played, indicating that the
service is enabled.
NOTE
Carrier). If you have not obtained the access permission of the document, see Help on
the website to find out how to obtain it. You must change the service password when
you use the service for the first time. For details about how to change the service
password, see 2.14.4 Configuring the Service Password.
– Manager A picks up the phone, and dials *73*#. An announcement is
– Secretary B picks up the phone, and dials *73*TN#. An announcement is
played, indicating that the service is disabled. In the dialed number, TN is
manager A's number.
Using the Secretary Service

● If secretary B uses an analog phone, the process of using the secretary service
is as follows:
a. User C makes a call to manager A. Secretary B's phone rings.
b. Secretary B picks up the phone, talks with user C, and presses the
hookflash key.
User C hears the waiting tone. If secretary B does not perform any
operations within 10 seconds, secretary B hears the busy tone.
▪ If secretary B presses the hookflash key within 5 seconds after

hearing the busy tone, the conversation with user C resumes.
▪ If secretary B does not perform any operations within 5 seconds after

hearing the busy tone, the system resumes the conversation between
secretary B and user C automatically.
c. Secretary B dials manager A's number.
If secretary B presses the hookflash key before manager A picks up the
phone, the connection with manager A is released and the conversation
with user C resumes.
d. Manager A picks up the phone and talks with secretary B.
▪ If manager A hangs up the phone first and secretary B does not

perform any operations within 5 seconds after hearing the busy tone,
the system resumes the conversation between secretary B and user C
automatically.
▪ If secretary B hangs up the phone first, manager A talks with user C.

● If secretary B uses an IP phone, the process of using the secretary service is as
follows:

NetEngine AR
The following assumes that secretary B uses eSpace 7950.

a. User C makes a call to manager A. Secretary B's phone rings. Secretary B
picks up the phone and talks with user C. This scenario assumes that the
conversation occupies line 1.
b. Secretary B presses Transfer and dials manager A's number.
The call with user C is held, and user C hears the waiting tone. If
secretary B presses the key corresponding to the line on which secretary B
and user C are having a conversation (line 1 in this example) twice, the
connection with manager A is released and the conversation with user C
resumes.
c. Manager A picks up the phone and talks with secretary B.
▪ If manager A hangs up, the message "call hold" is displayed on

secretary B's phone. If secretary B presses the key corresponding to
the line on which secretary B and user C are having a conversation
(line 1 in this example), the conversation with user C resumes.
▪ If secretary B presses Done, secretary B's phone automatically hangs

up and manager A talks with user C.
2.14.3.11.2 Configuring the Advanced Secretary Service

The advanced secretary service allows a manager to specify any intra-office user
as the secretary. When a user calls the manager, both the phones of the manager
and the secretary ring and the shared line indicator blinks. The secretary answers
the call, and forwards the call to the manager through the private line. The
secretary and manager can know the phone status of each other through the
phone status indicators.
Context
For the service of one manager with multiple secretaries, one manager with a
maximum of six secretaries can be configured. For the service of multiple
managers with multiple secretaries, a maximum of four managers with two
secretaries can be configured. These two types of services have the same
configuration principles. This topic uses one manager with two secretaries as an
example. Figure 2-43 shows the networking.
Figure 2-43 Network for the advanced secretary service

Manager A PBX
Private line number: 7102

Shared line number: 7100
Secretary Secretary User C

B1 B2
Private line number: 7201 Private line number: 7301

Shared line number: 7200 Shared line number: 7300

NetEngine AR
Service Scenario
After shared lines are configured for the manager and secretaries, all the phones
with shared numbers, including the manager's and secretaries' phones, ring when
a call destined for the manager's shared number comes in. The manager and the
secretaries can learn about each other's status over a shared line. For example, the
manager can know whether a secretary is busy or idle over the shared line.
Data listed in Table 2-9 is used as an example.
Table 2-9 Line sharing relationships between the manager and two secretaries
Manger's Shared Line EID for the Manager's EID for the Secretary's
Number Phone Number Phone Number
7100 7100 Secretary B1: 7200
7101 Secretary B2: 7300
Private line number:

● Manager A: 7102
● Secretary B1: 7201
● Secretary B2: 7301
NOTE
Only one number is required for multiple shared lines of the manager. Multiple equipment
IDs (EIDs) share this number.
Configuration Procedure (on the PBX)

3 Configure a manager. ssca-group pbxuser -

name
4 Bind a secretary. bind main-eid -
Maineid value shared-
eid Shared-eid-value
5 Configure a private config-dn dn-value -

line. private-dn private-
dn-value
6 (Optional) Turn mute config-dn dn-value -
on for calls destined alerting-silence
for the manager. enable

NetEngine AR
7 (Optional) Enable config-dn dn-value -

automatic answering auto-answer enable
for the manager.
8 Configure the line line-share-mode If line-share-mode is

sharing mode. set to all, all the
manager's and
secretary's lines
bound to the same
manager's number
are used as the same
line for sharing.
9 Save the save -

configuration.
[Huawei] voice
[Huawei-voice] ssca-group pbxuser 7100
[Huawei-voice-ssca-group-7100] bind main-eid 7100 shared-eid 7200
[Huawei-voice-ssca-group-7100] bind main-eid7101 shared-eid 7300
[Huawei-voice-ssca-group-7100] config-dn 7100 private-dn 7102
[Huawei-voice-ssca-group-7100] config-dn 7100 auto-answer enable
[Huawei-voice-ssca-group-7100] config-dn 7100 alerting-silence enable
[Huawei-voice-ssca-group-7100] line-share-mode all
[Huawei-voice-ssca-group-7100] save
Configuration Procedure (on the IP Phone)

The manager's account is configured in the same way as the secretary's account
on an IP phone. The following uses the manager's account as an example to
describe the configuration on the eSpace 7950 IP phone. The configuration varies
depending on the IP phone model. For details, see the Administrator Guide of the
corresponding model.
NOTE
● The following settings are the special settings for the advanced secretary service. For
common settings, see the Administrator Guide of the corresponding IP phone.
● The accounts to configure include the manager's and secretary's shared line numbers
and private line numbers.
● For details about the shared line settings, see Table 2-9.
Step 1 Open Internet Explorer and enter the IP address of the IP phone in the address box
to access the web configuration page of the IP phone.
Step 2 On the login page, enter the user name and password to log in to the web
configuration page of the IP phone, The default username and password are

NetEngine AR
Carrier). If you have not obtained the access permission of the document, see
Help on the website to find out how to obtain it.
Step 3 Configure the manager's and secretary's accounts.
1. Choose Advanced > Account. On the Account Setting page, click Add
Account.
2. In the Add Account dialog box that is displayed, set Account.
NOTE
Ensure that the settings of Account, Password, and User Name are the same as those
configured during SIP number allocation on the PBX.
3. In the Line Match area, specify accounts for different lines.
4. Click Save.
Step 4 Configure shared lines.
The following shared lines need to be configured:
● Shared lines 7100 and 7101 for manager A's phone
● Shared line 7200 for secretary B1's phone
● Shared line 7300 for secretary B2's phone
1. Choose Advanced > Service Rights Management. On the page that is
displayed, click the tab of an account, for example, 7100.
2. Select Yes from the Authorized drop-down list box for Manager Service.
NOTE
If the configuration is performed on the secretary's phone, select Yes from the
Authorized drop-down list box for Secretary Service.
3. Click Save.
4. Choose Call Features > Secretary.
5. In the Manager Line area, select the checkbox for account 7100.
6. Click Save.
----End
Usage Example
The following uses the eSpace 7950 IP phone as an example to describe how to
use the advanced secretary service, as shown in Figure 2-44.

NetEngine AR
Figure 2-44 Advanced secretary service
A user calls Manager's Secretary Secretary

the manager
phone 1's phone 2's phone
muted ring ring
1 Secretary 1
answers the call
Secretary 1 or
user on-hook Secretary 1
Call end talks to the
user
Manager or 2 Secretary 1 presses the line
user on-hook Manager button of the private line to
3 Manager presses on-hook call the manager's private
the line button of line
the shared line
Manager Secretary 1
User on
talks to the talks to the
hold
user manager
● – Scenario : The manager does not need to answer the call.
– Scenario : The secretary needs to ask the manager for permission, but
the manager does not need to answer the call.
– Scenario : The manager needs to answer the call.
2.14.3.12 Configuring Hotline Services

This topic describes how the hotline services are defined, configured, and used.
Currently, only intra-office POTS users support hotline services. Intra-office SIP
users do not support the services.
2.14.3.12.1 Configuring Delay Hotline

If a user that has been assigned the delay hotline service right does not dial a
number within 5s after picking up a phone, a call is automatically placed to a
preset number (hotline number).


system
view.
2 Enter the voice -

voice view.

NetEngine AR

PBX user
view.
4 Enable the service-right hotline To disable the hotline

hotline enable service right, run the
right. service-right hotline
disable
5 Configure a service hotline -

hotline { immediate | delay }
number. hotline-telno hotline-
telno-value
6 Return to quit -
the voice
view.


prefix
profile view
and access
the view.

prefix.

the call type callcategory attribute new-supplementary-
and call attribute management.
attribute for For the prefix used to
the prefix. enable the service, set
attribute (call attribute) to
111. For the prefix used to
attribute to 112.

NetEngine AR

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configuratio
n.
Enable the hotline service right for user 7000 and set the hotline number to 6000.
Set the prefix for enabling the delay hotline service to *111* and that for disabling
the delay hotline service to *112*.
# Configure the delay hotline service.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right hotline enable
[Huawei-voice-pbxuser-7000] service hotline delay hotline-telno 6000
# Configure a service prefix, for example, *111* for enabling the delay hotline service.
Enabling or Disabling the Delay Hotline Service

User A picks up the phone and dials *111*TN#. An announcement is played,
NOTE
● User A has been assigned the hotline service right.

● The variable TN is the hotline number (either an intra-office number or external
number), which is the number of user B.
Using the Delay Hotline Service

This example assumes that user A has been assigned the delay hotline service
right.

NetEngine AR
User A picks up the phone and waits for 5s. A call is initiated to user B
automatically. User A hears the ringback tone, and can talk with user B when user
B picks up the phone.
2.14.3.12.2 Configuring Instant Hotline

If a user that has been assigned the instant hotline service right picks up the
phone, a call is automatically placed to a preset number (hotline number).

n

system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right hotline enable To disable the hotline

hotline service right, run the
right. service-right hotline
disable
5 Configure service hotline { immediate If the service prefixes are

a hotline | delay } hotline-telno configured, this step is not
number. hotline-telno-value required. Users can enable
their phones.
6 Return to quit -
the voice
view.


NetEngine AR

prefix
profile
view and
access the
view.

a prefix.

call For the prefix used to enable
attribute the service, set attribute
for the (call attribute) to 109. For
prefix. the prefix used to disable
the service, set attribute to
110.

length
and
maximum
length of
a number
for
number
analysis.
5 Save the save -

configurat
ion.
Enable the hotline service right for user 7000 and set the hotline number to 7001.
Set the prefix for enabling the instant hotline service to *109* and that for
disabling the instant hotline service to *110*.
# Configure the instant hotline service.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right hotline enable
[Huawei-voice-pbxuser-7000] service hotline immediate hotline-telno 7001
# Configure a service prefix, for example, *109* for enabling the instant hotline service.

NetEngine AR
Enabling or Disabling the Instant Hotline Service

User A picks up the phone and dials *109*TN#. An announcement is played,
NOTE
● User A has been assigned the hotline service right.

● The variable TN is the hotline number (either an intra-office number or external
number), which is the number of user B.
Only the user with the instant hotline service right but without the instance
hotline service configured can cancel the instance hotline service of other
users. For example, when user 7001 dials *110*7000#, user 7001 is prompted
that the instant hotline service of user 7000 is canceled.
Using the Instant Hotline Service

This example assumes that user A has been assigned the instant hotline service
right.
User A picks up the phone. A call is automatically placed to user B. User A hears
the ringback tone, and can talk with user B if user B picks up the phone.
2.14.3.13 Configuring Ringing Services

This topic describes how the ringing services are defined, configured, and used.
2.14.3.13.1 Configuring Simultaneous Ringing

When a user with simultaneous ringing configured is called, the user's phone and
other specified phones ring simultaneously. The simultaneous ringing service can
only be configured for intra-office users.

view.
4 Enable the ringing service-right ring- To disable the ringing

service right. service enable service right, run the
following command:
service-right ring-
service disable

NetEngine AR

view.
6 Create a simultaneous ring-parallel-group -

ringing user group and pbxuser pbxuser
access the group. name
7 Configure the ringing triggertype trigger Three trigger types
trigger type. type value are available:
● internal:
Simultaneous
ringing is triggered
if the calling party
is an intra-office
user.
● external:
Simultaneous
if the calling party
is an external user.
● unspecified:
Simultaneous
regardless of
whether the calling
party is an intra-
office or external
user.
8 Configure numbers for group-member telno -

members in the telno value
simultaneous ringing
group.
9 Save the save -

configuration.
Configure simultaneous ringing for user 7000, with the trigger type being internal
and the number for simultaneous ringing being 7001.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right ring-service enable
[Huawei-voice] ring-parallel-group pbxuser 7000
[Huawei-voice-ring-parallel-group-7000] triggertype internal
[Huawei-voice-ring-parallel-group-7000] group-member telno 7001
[Huawei-voice-ring-parallel-group-7000] save

NetEngine AR
Usage Example
This example assumes that user A has been assigned the ringing service right,
numbers of users B and C are configured as the numbers for simultaneous ringing
with user A, and user D is an intra-office user.
1. User D places a call to user A. The phones of users A, B, and C ring at the
same time.
2. User A, B, or C picks up the phone and talks with user D. The phones of the
other two users stop ringing.
2.14.3.13.2 Configuring Sequential Ringing

When a user with sequential ringing configured is called and does not answer the
call within a specified period, the user's phone stops ringing and other specified
phones ring in sequence. The sequential ringing service can only be configured for
intra-office users.

view.

view.

user view.
4 Enable the ringing service-right ring- To disable the ringing

service right. service enable service right, run the
following command:
service-right ring-
service disable

voice view.
6 Create a ring-serial-group -
sequential ringing pbxuser pbxuser name
user group and
access the group.

NetEngine AR
7 Configure the triggertype trigger type Three trigger types are

ringing trigger value available:
type. ● internal: Sequential
ringing is triggered if
the calling party is
an intra-office user.
● external: Sequential
ringing is triggered if
the calling party is
an external user.
● unspecified:
Sequential ringing is
triggered regardless
of whether the
calling party is an
intra-office or
external user.
8 Configure group-member telno -

numbers for telno value
members in the
simultaneous
ringing group.
9 Configure the selectinterval select Value range: 5-30

interval for interval value Unit: second
sequential
ringing. Default: 20
10 Save the save -

configuration.
Configure sequential ringing for user 7000, with the trigger type being internal
and the number for sequential ringing being 7001.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right ring-service enable
[Huawei-voice] ring-serial-group pbxuser 7000
[Huawei-voice-ring-serial-group-7000] triggertype internal
[Huawei-voice-ring-serial-group-7000] group-member telno 7001
[Huawei-voice-ring-serial-group-7000] save
Usage Example
This example assumes that user A has been assigned the ringing service right,
numbers of users B and C are configured as the numbers for sequential ringing

NetEngine AR
with user A, and user D is another user. The interval for sequential ringing is the
default value, which is 20s.
1. User D places a call to user A.
2. The phone of user A stops ringing if user A does not pick up the phone with
20s. Then, the phone of user B rings.
3. The phone of user B stops ringing if user B does not pick up the phone with
20s. Then, the phone of user C rings.
4. User C picks up the phone and talks with user D.
NOTE
If user A picks up the phone within 20s, the phone of user B does not ring. If user B picks
up the phone within 20s, the phone of user C does not ring.
2.14.3.14 Configuring Conference Call Services
NOTE
The AR617VW, AR617VW-LTE4, AR617VW-LTE4EA, and AR6120-VW do not support this

function.
2.14.3.14.1 Configuring Individuals Dialing-In

In individuals dialing-in, participants dial a preset conference access code and
enter related conference information as prompted to join a conference.
Configuring Conference Information


view.

view.

NetEngine AR
3 Create a conference conference Configure the enterprise

scheduled name enterprise information if the
conference. enterprise name list conference to schedule
is enterprise-specific. By
default, the enterprise
information is not
provided.
NOTE
● To query the
information about a
scheduled conference,
run the display voice
conference conference
name command.
● To cancel a conference,
run the undo
conference conference
name command.
4 Configure the description description -

conference information
description.
5 Specify the regnum register -

maximum number value
number of users
to join the
conference.
6 Configure the attendee-password ● The password must

passwords for the cipher be a string of 6-16
common digits.
participants to ● The password for the
join the participants cannot
conference. be the same as or the
reverse of the
password for the
moderator.
● The conference
password cannot be
the same as or the
reverse of the
account of the user
who schedules the
conference.
● When the conference
subject or name is a
string of digits only,
the conference
password cannot be
the same as or the

NetEngine AR
7 Configure the chairman-password reverse of the

passwords for the cipher conference subject
moderator to join and conference
the conference. name.
● If you do not config
the password, the
system will use the
random attendee
password and
chairman password.
Both chairman and
attendees can see
their password
through self-services
http web page.
8 Configure the schedule from date -

conference time to date time
duration.

voice view.
Configuring a Conference Access Code

1 Create a prefix callprefix callprefix-name

profile view and
access the view.
2 Configure a prefix. prefix prefix
3 Configure the call call-type category callcategory attribute

type and call attribute
attribute for the
prefix.
4 Configure the digit-length minimum-length-value

minimum length maximum-length-value
and maximum
length of a number
5 Save the save

configuration.

NetEngine AR
Create conference 1. Set the maximum number of participants to 3, participant
password to 132456, moderator password to 142356, conference description to
test, and conference duration to 2013-09-10 10:00:00 to 2013-09-10 12:00:00.
Configure a conference access code. Set callprefix to conference, prefix to 888,
and category to vu-service.
[Huawei] voice
[Huawei-voice] conference 1
[Huawei-voice-conference-1] description test
[Huawei-voice-conference-1] regnum 3
[Huawei-voice-conference-1] attendee-password cipher Please input user password(6-16 chars):******
[Huawei-voice-conference-1] chairman-password cipher Please input user password(6-16 chars):******
[Huawei-voice-conference-1] schedule from 2013-09-10 10:00:00 to 2013-09-10 12:00:00
[Huawei-voice-conference-1] quit
[Huawei-voice] callprefix conference
[Huawei-voice-callprefix-conference] prefix 888
[Huawei-voice-callprefix-conference] call-type category vu-service vu-service-name vuconference
[Huawei-voice-callprefix-conference] digit-length 3 32
[Huawei-voice-callprefix-conference] save
Usage Example (for Participants)

All participants (including the moderator) can join a conference as follows:
1. Dial the conference access code.
2. Press 1 after hearing the announcement "Welcome to the conference system.
To join a conference, please press 1. To create an instant conference, please
press 2."
3. Enter the conference ID when hearing the announcement "Please input the
conference ID followed by the pound key."
NOTE
You can run the display voice conference [conference-name]command to view the
conference ID.
4. Enter the password when hearing the announcement "Please input the
password followed by the pound key."
5. When you join the voice conference, press 8# to mute the terminal and press
7# to cancel the muting.
Usage Example (for the Moderator)

After joining the conference, the moderator perform the hookflash operation.
● If the moderator uses a POTS phone, the moderator presses the hookflash
button.
NOTE
(hookflash action).
lower command in the voice view to adjust the lower threshold for hookflash
pressing.
● If the moderator uses an IP phone, the moderator presses the star (*) key.

NetEngine AR
Only eSpace7800 series support this operation. This section uses eSpace7850
to describe the configuration method.
NOTICE
The G.711 codec number collection must be preferred on the moderator

phone or DTMF type must be set to AUTO + SIP INFO; otherwise, the
moderator cannot add participants after pressing hook. For the configuration
procedure, see the Administrator Guide of the IP phone.
● After the hookflash operation is performed, the moderator can perform the
following operations:
Press 1 to invite participants. Dial the number of a participant as prompted.
After the call is answered, perform the hookflash operation again.
– Press 1 to allow the user to join the conference.
– Press 2 to reject the user. The moderator returns to the conference, and
the user hears the busy tone.
2.14.3.14.2 Configuring System Convening

In system convening, the moderator sets a participant list in advance, and the
system calls the listed participants at the scheduled time. Participants can join the
conference by simply picking up the phone.


view.

view.
3 (Optional) conference-attribute -
Configure
conference
attributes and
enter the global
conference
attribute view.
4 (Optional) conference- If a called user is busy,

Configure the reconvoke-attempts does not respond, or set
maximum number reconvoke-times DND, the system recalls
of recall attempts. the user after 60 seconds.
The value ranges from 0 to
16.
5 (Optional) Return quit -

to the voice view.

NetEngine AR
6 Create a conference Configure the enterprise

scheduled conference name information if the
conference. enterprise enterprise conference to schedule is
name list enterprise-specific. By
information is not
provided.
NOTE
● To query the information
about a scheduled
conference, run the
display voice conference
conference name
command.
run the undo conference
conference name
command.
7 Configure the description -

conference description
description. information
maximum number number value
of users to join
the conference.
9 Configure the attendee-password ● The password must be a

passwords for the cipher string of 6-16 digits.
common ● The password for the
participants to participants cannot be
join the the same as or the
conference. reverse of the password
for the moderator.
● The conference
password cannot be the
same as or the reverse
of the account of the
user who schedules the
conference.
string of digits only, the
conference password
cannot be the same as
or the reverse of the
conference subject and
conference name.
● If you do not config the
password, the system

NetEngine AR
10 Configure the chairman-password will use the random

passwords for the cipher attendee password and
moderator to join chairman password.
the conference. Both chairman and
attendees can see their
password through self-
services http web page.

duration.
12 Conference the attendee-telno Available voice access

rights for users to attendee telephone modes for participants:
join the number mode mode ● listenspeak: listen and
conference. type talk
● listenonly: listen only
● chairman: moderator
mode (The right to
perform the hookflash
operation is also
available except for the
rights to talk and
listen.)
● isolate: isolate
Default: listenspeak

voice view.


profile view and
access the view.

attribute for the
prefix.

NetEngine AR

and maximum
length of a number
5 Save the save

configuration.
test, conference duration to 2013-09-10 10:00:00 to 2013-09-10 12:00:00, user
7100 to chairman, participant 7101 to listenspeak, and participant 7102 to
listenonly. Configure a conference access code. Set callprefix to conference,
prefix to 888, and category to vu-service.
[Huawei] voice
[Huawei-voice-conference-1] attendee-telno 7100 mode chairman
[Huawei-voice-conference-1] attendee-telno7101 mode listenspeak
[Huawei-voice-conference-1] attendee-telno 7102 mode listenonly
Usage Example
When the start time of a scheduled conference arrives, the phones of all
participants ring at the same time. They can join the conference by simply picking
up the phone.
● During the conference, the moderator can invite participants and perform
other operations. For details, see Usage Example (for the Moderator).
● If a called user is busy, does not respond, or set DND, the system recalls the
user after 60 seconds.
2.14.3.14.3 Configuring Host Convening

The moderator joins a conference and invites or removes participants by phone.

NetEngine AR


view.

view.
3 Create a conference Configure the enterprise

scheduled conference name information if the
conference. enterprise enterprise conference to schedule is
name list enterprise-specific. By
information is not
provided.
NOTE
● To query the information
about a scheduled
conference, run the
display voice conference
conference name
command.
run the undo conference
conference name
command.
4 Configure the description -

conference description
description. information
maximum number number value
of users to join
the conference.
6 Configure the attendee-password ● The password must be a

passwords for the cipher string of 6-16 digits.
common ● The password for the
participants to participants cannot be
join the the same as or the
conference. reverse of the password
for the moderator.
● The conference
password cannot be the
same as or the reverse
of the account of the
user who schedules the
conference.
string of digits only, the

NetEngine AR
7 Configure the chairman-password conference password

passwords for the cipher cannot be the same as
moderator to join or the reverse of the
the conference. conference subject and
conference name.
● If you do not config the
password, the system
will use the random
attendee password and
chairman password.
Both chairman and
attendees can see their
passwords on the self-
services web page.

duration.
9 Conference the attendee-telno Available voice access

rights for users to attendee telephone modes for participants:
join the number mode mode ● listenspeak: listen and
conference. type talk
● listenonly: listen only
● chairman: moderator
mode (The right to
perform the hookflash
operation is also
available except for the
rights to talk and
listen.)
● isolate: isolate
Default: listenspeak

voice view.
test, conference duration to 2013-09-10 10:00:00 to 2013-09-10 12:00:00, user
7100 to chairman, participant 7101 to listenspeak, and participant 7102 to
listenonly.
[Huawei] voice

NetEngine AR

[Huawei-voice-conference-1] attendee-telno 7100 mode chairman
[Huawei-voice-conference-1] attendee-telno 7101 mode listenspeak
[Huawei-voice-conference-1] attendee-telno 7102 mode listenonly
Usage Example
After a moderator joins or is invited to a conference, the moderator can invite
other participants. For details, see Usage Example (for the Moderator).
2.14.3.14.4 Configuring Instant Conference

If a user has been assigned the instant conference service right, the user can start
instant conferences by phone.


system view.
2 Enter the voice -

voice view.

PBX user
view.
4 Enable the service-right instant- To disable the instant

instant conference enable conference service right,
conference run the following
service command:
right. service-right instant-
conference disable
5 Return to quit -
the voice
view.


profile view and
access the view.

NetEngine AR

attribute for the
prefix.

and maximum
length of a number
5 Save the save

configuration.
Configure the instant conference service right for user 7100. Configure a
conference access code. Set callprefix to conference, prefix to 888, category to
vu-service.
[Huawei] voice
[Huawei-voice-pbxuser-7100] service-right instant-conference enable
Usage Example
1. The user that has been assigned the instant conference service right dials the
conference access code.
2. The user presses 2 as prompted.
3. The user invites participants. For detailed operations, see Usage Example (for
the Moderator).
2.14.3.15 Configuring Enhanced Services

This topic describes how the enhanced services are defined, configured, and used.
2.14.3.15.1 Configuring Abbreviated Dialing

The abbreviated dialing service allows a user to dial the service prefix and a two-
digit abbreviated number instead of a complete called number to connect to the
called party.

NetEngine AR

system view.
2 Enter the voice -

voice view.

user view.
4 Enable the service-right abbdial To disable the abbreviated

abbreviated enable dialing service right, run
dialing service the following command:
right. service-right abbdial
disable
5 Configure an service abbdial abb-telno If the service prefixes are

abbreviated called-telno configured, this step is not
or disable the service
using their phones.

voice view.

1 Create a prefix callprefix callprefix-name -

profile view
and access the
view.

prefix.

NetEngine AR

call type and callcategory attribute new-supplementary-
call attribute attribute management.
for the prefix. For the prefix used to
to 37. For the prefix used
to disable the service, set
attribute to 38.

minimum length-value maximum-
length and length-value
maximum
length of a
number for
number
analysis.
5 Save the save -

configuration.
Enable the abbreviated dialing service right for user 7000 and set the abbreviated
number for dialing user 7001 to 11. Set the prefix for enabling the abbreviated
dialing service to *37*, prefix for disabling the abbreviated dialing service to *38*,
and abbreviated number prefix to *174*.
NOTICE
For abbreviated dialing forwarding to *174*, set category to supplementary-

service and attribute to 174. If the prefix is not configured, users cannot dial the
abbreviated number.

[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right abbdial enable
[Huawei-voice-pbxuser-7000] service abbdial 11 7001
# Configure a service prefix, for example, *37* for enabling the abbreviated dialing service.
[Huawei-voice-callprefix-*37*] quit
# Configure an abbreviated dialing prefix.

NetEngine AR
Enabling or Disabling the Abbreviated Dialing Service

User A that has been assigned the abbreviated dialing service right picks up
the phone and dials *37*MN*TN#. An announcement is played, indicating that
the service is enabled.
NOTE
● *37*: indicates the abbreviated dialing prefix.

● MN: indicates a 2-digit abbreviated number that ranges from 00 to 99.
● TN: indicates user B's number.
● If you hear the announcement indicating that the service registration fails when
using the prefix to register service, check whether the service information is
correct, for example, whether the format, prefix length, or user number length
meet requirements and whether a conflict occurs in the registered service.
– Deleting a single abbreviated number
User A picks up the phone and dials *38*MN#. An announcement is
played, indicating that the abbreviated number is deleted.
– Deleting all abbreviated numbers
indicating that all abbreviated numbers are deleted.
Usage Example
This example assumes that user A has been assigned the abbreviated dialing
service right.
User A picks up the phone and dials *174*MN to place a call to user B.
2.14.3.15.2 Configuring ONLY

The one number link you (ONLY) service allows multiple terminals of a user to
share the same number and the same supplementary services. The ONLY number
is a number of an intra-office terminal, which is the main terminal. When others
call a user's ONLY number, the user's terminals ring according to the preset rules,
and the user can select a terminal to answer the call.

view.

view.

NetEngine AR
3 Configure the time time-name { { from -

time index. from-date [ from-time ]
to to-date [ to-time ] } |
time-repeat { { yearly |
monthly } begin-date
[ begin-time ] [ end-date
[ end-time ] ] |weekly
begin-weekday [begin-
time1 ] [end-
weekday[end-time1 ] ] |
daily begin-time2 [ end-
time2 ] } }
4 Create an ONLY unicall-group pbxuser -
view and access pbxuser-name
the view.
5 Configure an bind bind-type bind- Available types of

ONLY terminal value priority Priority- terminals bound to the
value [ time-templet ONLY service:
time-templet-name ] ● internal-telno:
indicates an intra-
office terminal.
● external-telno:
indicates an external
terminal.
● sipue: indicates a SIP
terminal.
6 (Optional) busyselectmode The following modes are

Configure the busymodeselect value available:
mode of ● null: No terminal is
processing a new called.
call when the
ONLY user is ● all: All terminals are
engaged in a called.
call. ● active: The busy
terminal is called.
● inactive: Idle
terminals are called.
Default: all

NetEngine AR
7 (Optional) prioritymode priority Available terminal

Configure the mode value priority modes:
terminal priority ● dynamic: The
mode. terminal priorities are
dynamically adjusted.
● static: The terminal
priorities remain
unchanged.
Default: static
8 (Optional) ringmode ring mode Available terminal

Configure the value ringing modes:
terminal ringing ● alter: When the
mode. terminal with a lower
priority start to ring,
terminals with higher
priorities stop ringing.
● append: When the
terminal with a lower
priority start to ring,
terminals with high
priorities continue to
ring.
Default: append
9 (Optional) selectinterval select Value range: 5-30

Configure the interval value Unit: second
ringing interval
between Default: 12
terminals of
adjacent
priorities.
10 Save the save -

configuration.
Configure the ONLY service for intra-office user 7000 and bind intra-office
terminals 7001 and 7002, with priorities being 1 and 2. Configure time index 0,
and specify the time period of time index 0 as 08:00:00 to 17:30:00 from 2013-1-1
to 2014-1-1. Enable the ONLY service to take effect only within this time period.
[Huawei] voice
[Huawei-voice] time 0 time-period from 2013-1-1 08:00:00 to 2014-1-1 18:00:00
[Huawei-voice] unicall-group pbxuser 7000
[Huawei-voice-unicall-group-7000] bind internal-telno 7001 priority 1 time-temple 0
[Huawei-voice-unicall-group-7000] bind internal-telno 7002 priority 2 time-temple 0
[Huawei-voice-unicall-group-7000] save

NetEngine AR
Usage Example
This example assumes that user A's office phone is used as the main terminal of
the ONLY service. User A binds lab phone 1 (priority being 1) and lab phone 2
(priority being 2). When a new call comes in while user A is busy, the idle
terminals ring. The ringing mode is append, and the ringing interval is 15s.
The service process is described as follows when user A is a called party:
1. User B places a call to user A. User A's office phone rings, and user B hears
the ringback tone.
– If user A picks up the office phone within 15s, the call is connected. If
user C places a call to user A at this moment, the idle terminals ring.
– If user A does not answer the call within 15s, lab phone 1 rings and the
office phone continues to ring. Go to 2.
2. User B hears the ringback tone. Within 30s (15s for the office phone and 15s
for lab phone 1):
– If user A picks up the phone, the call is connected. All the other terminals
stop ringing. If user C places a call to user A at this moment, the idle
terminals ring.
– If user A does not answer the call, lab phone 2 rings, and the office
phone and lab phone 1 continue to ring. Go to 3.
3. If none of the phones is picked up within 60s, user B hears the busy tone. If
user C places a call to user A at this moment, user C hears the busy tone.
2.14.3.15.3 Configuring Three-Party Calls

The three-party call service allows the calling or called party in a two-party
conversation to call a third party to start a three-party conversation or talk to the
other two parties separately without ending the current conversation.
Context
The three-party call service allows a user to start a three-party conversation or
talk to two parties separately. In some scenarios, only three-party conversations
are required. The pbx number-parameter command can be used to deploy a
simplified three-party call service.
● The three-party call service allows a user to start a three-party conversation
or talk to two parties separately. After calling a third party, the user who
initiates a three-party call needs to press the hook flash button and dial a
number to select a conversation mode.
● A simplified three-party call service allows a user to start a three-party
conversation, but does not allow the user to talk to two parties separately.
After calling a third party, the user who initiates a three-party call only needs
to press the hook flash button to select the conversation mode.

system view.

NetEngine AR
2 Enter the voice -

voice view.

user view.
4 Enable the service-right three-party To disable the three-party

three-party enable call service right, run the
call service following command:
right. service-right three-party
disable
5 Save the save -

configuration.
Enable the three-party call service right for user 6000.
[Huawei] voice
[Huawei-voice-pbxuser-6000] service-right three-party enable
Configuration Procedure for Simplified Three-Party Calls

Two types of simplified three-party call services are provided to meet requirements
in different scenarios.

system view.
2 Enter the voice -

voice view.
3 Enable a pbx number-parameter By default, an AR

simplified 236 1 implements the three-
three-party party call service flow.
call service. This command configures
(Select a type the IMS to control the
of simplified three-party call service
three-party flow. The AR only forwards
call service as packets and does not
required.) control the three-party call
service flow.
Figure 2-46 shows the
service usage and call
status change.

NetEngine AR
pbx number-parameter By default, an AR

242 1 implements the three-
party call service flow.
If a user only requires the
three-party conversation
service, and does not need
to talk to other two parties
separately during a call,
the user can run this
command to configure a
simplified three-party call
service. After the
configuration takes effect,
the user only needs to
press the hook flash
button without dialing a
number when initiating a
three-party conversation.
Figure 2-47 shows the
service usage and call
status change.
4 Save the save -

configuration.
Usage Example
NOTE
Currently, the PBX supports only a POTS phone as the party initiating a three-party call.
There is no restriction on the type of the phones invited to a three-party call.
This example assumes that user A has been assigned the three-party call service
right and is engaged in a call with user B using a POTS phone.
For details about the usage and call status change, see Figure 2-45.
Figure 2-45 Three-party call

User A and user B are
engaged in a call User C does not
(user C is idle) answer the call
User A presses and user A
the hookflash presses the
1 2 hookflash
User A calls
Call on hold (waiting user C Dual-communication
for user A to dial) (user C's phone rings)
User C picks
User A presses up the phone User A presses the
4 hookflash and presses 1
the hookflash 3
Dual-communication and presses 2 Dual-communication
(user A and user B (user A and user C Three-party call
talk with each other) talk with each other)
User A presses the
hookflash and presses 3

NetEngine AR
● State : User A hears a ringback tone, and user B hears a call hold tone.
● State : User C hears a call hold tone.

This example assumes that user A runs the pbx number-parameter command to
configure a simplified three-party call service.
Figure 2-46 shows the usage of the IMS-controlled simplified three-party call
service and call status change.
Figure 2-46 IMS-controlled simplified three-party call

1 hookflash
User A calls 2
User C picks
up the phone
Three-party call
Figure 2-47 shows the usage of the AR-controlled simplified three-party call
service and call status change.
Figure 2-47 AR-controlled simplified three-party call

1 hookflash
User A calls 2
User C picks
up the phone
3
Dual-communication
(user A and user C Three-party call
talk with each other) User A presses
the hookflash

NetEngine AR
In a three-party call, if either user B or C hangs up, user A can continue to talk
with the party who does not hang up; if user A hangs up, users B and C hear the
busy tone.
2.14.3.15.4 Configuring Enterprise CRBT

Enterprise CRBT is a piece of music or sound customized by an enterprise. After an
enterprise registers the CRBT service, the user can set different CRBTs for a calling
party or a group of calling parties in different periods.
Prerequisites
The CRBT file has been recorded and loaded. The methods of creating and loading
a CRBT file are the same as those for system announcements. For details, see
2.15.1 Creating and Switching a Phone System Announcement.
Procedure
St Operation Command Description
ep

system
view.
2 Enter the voice -

voice view.
3 Create an enterprise enterprise-name -

enterprise
and enter
the
enterprise
view.
4 Activate crbt enable To cancel the service right,

the service run the following command:
right. crbt disable
5 Configure crbt-file crbt-file-name The name of a CRBT file

the status-value [ description (including the path) must be
enterprise description ] a string of 1 to 64 characters.
CRBT file. The value is in the format
path:/file-name, for example,
sd:/crbt.wav.

NetEngine AR
St Operation Command Description

ep
6 Configure service crbt [ condition -

the service { caller-telno caller-telno-
attributes value | caller-trunkgroup
for the trunkgroup-name | time-
enterprise period { from from-date
CRBT. [ from-time ] | to to-date
[ to-time ] } * | time-repeat
{ yearly begin-date [ begin-
time ] [ end-date [ end-
time] ] | monthly begin-date
[ end-time] ] | weekly begin-
weekday [ begin-time ]
[ end-weekday [ end-time ] ]
| daily begin-time [ end-
time] } }* ] file-name file-
name-value
7 Save the save -
configurati
on.
Example
Assume that the path for storing enterprise CRBT files is sd:/crbt.wav. Configure
the CRBT crbt.wav played between 10:00:00 and 18:00:00 from 2013-10-10 to
2013-11-10 for inter-office user 1000.
[Huawei] voice
[Huawei-voice] enterprise default
[Huawei-voice-enterprise-default] crbt enable
[Huawei-voice-enterprise-default] crbt-file sd:/crbt.wav status pass
[Huawei-voice-enterprise-default] service crbt condition caller-telno 1000 time-repeat daily 10:00:00
18:00:00 time-period
from 2013-10-10 to 2013-11-10 file-name sd:/crbt.wav
[Huawei-voice-enterprise-default] save
An enterprise has registered the CRBT service. When inter-office users call the
enterprise, the users can hear the CRBT customized by the enterprise.
2.14.3.15.5 Configuring the Wakeup Service

After a phone has the wakeup service enabled, the phone rings at the specified
time to notify the user.

NetEngine AR
Procedure
By default, PBX users have the wakeup service right. To configure the wakeup
service, perform the following steps.

system
view.
2 Enter the voice -

voice view.

PBX user
view.
4 Activate service-right wake-call To cancel the service right,

the service enable run the following command:
right. service-right wake-call
disable
5 Register service wake-call wake- After the service prefix is

the service. time wake-time-value configured, the user can
[ begin-date begin-date- register and cancel the
value | wake-days wake- service through the phone.
days-value | tone-id tone- This step is not required.
id-value ] *
6 Return to quit -
the voice
view.


prefix
profile
view and
access the
view.

NetEngine AR

a prefix.

for the attribute (call attribute) to
prefix. 41. For the prefix used to
attribute to 42.

length and
maximum
length of a
number for
number
analysis.
5 Save the save -

configurati
on.
Example
Assign the wakeup service right for user 7000, and enable the wakeup service with
the wakeup time being 08:00 every morning. Set the service registration prefix to
*41* and service deregistration prefix to *42*.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right wake-call enable
[Huawei-voice-pbxuser-7000] service wake-call wake-time 08:00:00
# Configure the service prefix. (In this example, the service registration prefix is *41*. The methods of
configuring other service prefixes are the same.)
Registering or Deregistering Service

Assume that the wakeup service registration prefix is *41* and deregistration prefix
is *42*.

NetEngine AR
● Register the service.

User A with the wakeup service right picks up and dials *41*H1H2M1M2#. User
A is prompted that service is successfully registered.
NOTE
● H1H2 is the two-digit value of the hour. The value ranges from 00 to 23.
● M1M2 is the two-digit value of the minute. The value ranges from 00 to 59.
● If you cannot hear the system prompts when using the prefix to register service,
check whether the service information is correct, for example, whether the format
or prefix length meet requirements and whether a conflict occurs in the registered
service.
● Deregistering the service.
When user A dials *42*#, user A is prompted that the service is successfully
deregistered.
User A has enabled the wakeup service.
● The phone of user A rings at the specified wakeup time. User A picks up the
phone and hears the voice prompt.
● If user A does not respond within one minute, the system rings again after
five minutes. If user A still does not respond, the system does not ring on the
same day.
2.14.3.15.6 Configuring the PBX Line Selection Service

This topic describes how to configure the PBX line selection service.
Context
A PBX group refers to a group of numbers allocated under a PBX. When a user
outside of a PBX group dials the access code of the PBX group, the PBX line
selection service enables the call processing program to select and ring one or
more idle lines based on the preset line selection method and connect the call.
The PBX line selection service supports simultaneous ringing, sequential ringing,
ringing of a single phone, and cyclic ringing.
Procedure
3 Enter the PBX user pbxusergroup user-group- -

group view. name [ hunt [ enterprise
enterprise-name ] ]
4 Configure an access access-telno access-telno- -
code. value

NetEngine AR
5 (Optional) Configure dn-set dn-set-name -

a DN set.
6 (Optional) Configure master-first enable When an intra-

the primary number office number
preference rule. is the same as
the access
code, the intra-
office number
is called the
primary
number.
By default, the
primary
number
preference rule
is enabled. To
disable the
primary
number
preference rule,
run the
master-first
disable
command.
7 Configure group group-member { pbxuser Configure

members. pbxuser-name | external- external-telno
telno external-telno-value } external-telno-
[ condition { time-period value to specify
{ from from-date [ from- members of a
time ] | to to-date [ to- group as outer-
time ] } * | time-repeat office numbers.
{ { yearly | monthly }begin- The specified
date [ begin-time ] [ end- numbers must
date [ end-time ] ] | weekly be different
begin-weekday [ begin- from intra-
time ] [ end-weekday [ end- office numbers
time ] ] | daily begin-time and access
[ end-time ] } } * ] [ admin- codes.
state admin-state-value ]
[ member-index member-
index-value ]

NetEngine AR
8 (Optional) Configure ring { mode ring-mode- By default, the

the line selection value | time ring-time-value ringing
mode and ringing | select select-mode-value } duration is 12s,
mode. ringing mode is
ring-serial, and
line selection
mode is loop.
If the ringing
mode is
sequential
ringing, you
can configure
the 156 control
point to specify
whether to
release a call
when no
sequential
ringing group
member
answers the
call. For details,
see pbx
number-
parameter.
9 (Optional) Configure group-name group-name- By default, the

the name of a user value name of a user
group. group is not
configured.
10 (Optional Enable crbt enable By default, the

) the user user group RBT
Configure group service is
the user RBT disabled.
group service.
RBT
service. Specify crbt-file crbt-file-name RBT files must
an RBT status state-value be in .wav
If the user
file for a [ description description ] format.
group
user By default, no
RBT
group. RBT file is
service
and specified for a
enterprise user group.
RBT
service
are
configure
d
simultane

NetEngine AR

ously, the Configure service crbt file-name file- RBT files must
user an name-value be in .wav
group attribute format.
RBT of the By default, an
service user attribute of the
takes group user group RBT
precedenc RBT service is not
e over the service. configured.
enterprise
RBT
service.
11 (Optional) Configure call-right { in | out } By default,

call-in and call-out { { inter inter-value | local users of a user
rights for users in a local-value | ddd ddd-value | group have
user group. idd idd-value } * | all } call-in rights of
intra-office,
local, national
toll, and
international
toll calls, and
call-out rights
of intra-office
and local calls.
12 (Optional) Configure service cfnr [ time ] telno The

a CFNR forwarded-to fwd-telno configuration
number for all users takes effect
in a user group. only when
simultaneous
ringing is
configured for
a user group.
You can run the
ring command
to configure
the ringing
mode, and set
the ring-mode-
value
parameter to
ring-parallel.
By default, no
CFNR
forwarded-to
number is
specified.

NetEngine AR
13 (Optional) Configure service cfo fwd-telno By default, no

a CFO forwarded-to CFO
number for users in a forwarded-to
user group. number is
specified.
14 Save the save -

configuration.
Configure number 7000 as the access code of a PBX group, group member 7001
whose index is 1, and group member 7002 whose index is 2. The PBX group takes
effect between 08:00:00 and 17:00:00 from 2013-11-01 to 2013-11-30. The ringing
mode is sequential ringing. The ringing interval is 12s. The line selection mode is
ascending. The call-out rights are of intra-office, local, and national toll calls.
[Huawei] voice
[Huawei-voice] pbxusergroup 0 hunt
[Huawei-voice-pbxusergroup-0] access-telno 7000
[Huawei-voice-pbxusergroup-0] group-member pbxuser 7000 condition time-period from 2013-11-01 to
2013-11-30 time-repeat daily 08:00:00 17:00:00 member-index 0
[Huawei-voice-pbxusergroup-0] ring mode ring-serial
[Huawei-voice-pbxusergroup-0] ring time 12
[Huawei-voice-pbxusergroup-0] ring select increase
[Huawei-voice-pbxusergroup-0] call-right out ddd enable
[Huawei-voice-pbxusergroup-0] save
Using the PBX Line Selection Service

This example assumes that a PBX group consists of users A, B, and C. User B's
number is the primary number, and users A and C's numbers are secondary
numbers. The indexes of users A, B, and C are in the following order: A < B < C.
The line selection mode is ascending. The ringing mode is sequential ringing.
● The primary number preference rule is disabled.
Based on the line selection mode, the ringing priority is as follows: A > B > C.
If user A is idle, user A's phone rings. If user A is busy or does not answer the
call within the ringing interval, user B's phone rings. The same rule applies to
the rest.
● The primary number preference rule is enabled.
Based on the primary number preference rule and the line selection mode,
the ringing priority is as follows: B > A > C.
a. If user B is idle, user B's phone rings. If user B is busy or does not answer
the call within the ringing interval, user A's phone rings.
b. If user A is idle, user A's phone rings. If user A is busy or does not answer
the call within the ringing interval, user C's phone rings.

NetEngine AR
2.14.3.15.7 Deregistering All Services

Deregistering all services means deleting all registered services of a user.
Configuring Service Rights


system view.
2 Enter the voice -

voice view.

user view.
4 Activate the service-right scc-cancel To cancel the service right,

service right. enable run the following
command:
service-right scc-cancel
disable

voice view.
Configuring the Service Prefix


prefix profile
view and
access the
view.

prefix.
3 Configure the call-type category The call type category

call type and callcategory attribute must be set to new-
call attribute attribute supplementary-
for the prefix. management.
For the service registration

NetEngine AR

minimum length-value maximum-
length and length-value
maximum
length of a
number for
number
analysis.
5 Save the save -

configuration.
Example
Activate the service right for user 7000 and set the service registration prefix to
*75*.
[Huawei] voice
[Huawei-voice-pbxuser-7000] service-right scc-cancel enable
#Configure a service prefix
When user A dials *75*SSSSSS#, all services are deregistered.
NOTE
● *75* is the prefix for deregistering all services.

● The default username and password are available in AR Router Default Usernames and
Passwords (Enterprise Network or Carrier). If you have not obtained the access
permission of the document, see Help on the website to find out how to obtain it. If you
use the password for the first time, the password must be modified. For details about
how to change the password, see 2.14.4 Configuring the Service Password.
2.14.3.15.8 Configuring the MWI Service

The message waiting indicator (MWI) function applies to messages of the
incoming calls routed through SIP trunks.
Context
The PBX connects or registers with PBX1 using the SIP trunk, and the voice
mailbox service is configured on PBX1. As long as user C can directly call user B,
user C can leave a message to user B. When user C is leaving a message, PBX1
plays a message leaving announcement. PBX1 directly sends the message to user

NetEngine AR
B. The system can notify user B of the message only when the MWI function is
configured on the PBX. Figure 2-48 shows the networking.
Figure 2-48 MWI networking
PBX1
SIP Trunk
PBX
Prerequisites
● Intra-office users have been configured. For details, see 2.8 Configuring a
PBX User.
● SIP trunks have been configured. For details, see 2.11.2 Configuring SIP
Trunk Groups.
Configuring the Service Rights

1 Access the system-view -

system
view.
2 Access the voice -

voice view.
3 (Optional) mwi-mode { fsk-with-ring Configure the MWI mode

Configure | fsk-without-ring } when the MWI user is a
the MWI POTS user.
mode. To restore the default
settings, run the following
command:
undo mwi-mode
The default value is fsk-
without-ring.
4 Save the save -

configuratio
n.

NetEngine AR
Configure the MWI function for user 7000 and set the MWI mode to fsk-with-
ring.
[Huawei] voice
[Huawei-voice] mwi-mode fsk-with-ring
[Huawei-voice] save
Usage Example
This example assumes that user A has been assigned the MWI service right.
1. User C sends a message to user A through the SIP trunk, and PBX1 sends an
MWI notification to user A through the SIP trunk.
2. If user A uses an IP phone, the system notifies user A of the new message by
means of, for example, MWI or phone screen. If user A is a POTS phone user,
the system notifies user A of the new message by means of, for example,
MWI, voice message, or ringing.
see the Administrator Guide of IP phone.
NOTE
● The prefix for user A to retrieve messages is configured on PBX1.

● User A can retrieve voice messages on an intra-office or inter-office phone. When
retrieving messages, you need to enter the service password.
2.14.3.15.9 Configuring the Busy Lamp Field Service

The Busy Lamp Field (BLF) service allows a user to check the conversation status
(including idle, in conversation, ringing, and offline) of another IP phone in real
time.
Context
The BLF service allows a user to check the conversation status (including idle, in
conversation, ringing, and offline) of another IP phone in real time. A user can use
the multi-purpose keys on an IP phone to subscribe to the conversation status of
another intra-office user. After subscription, the system will inform the subscriber
of the conversation status of the subscribed user in real time. This service allows
the subscriber to call the subscribed user when it is idle, which increases call
success rate. Additionally, the BLF service can be used with the Pickup Services.
For example, when user A subscribes to the status of user B, the conversation
status and incoming call information about user B are displayed on the IP phone
of user A, so user A can pick up the phone for user B.

NetEngine AR
As shown in Figure 2-49, the router functions as the PBX of the internal voice
network of an enterprise, and both user A and user B are intra-office SIP users.
User B makes phone calls frequently, and user A fails to call user B many times.
User A uses the multi-purpose keys on an IP phone to subscribe to the
conversation status of user B and calls user B when it is idle.
Figure 2-49 User conversation status subscription process
PBX
Subscriber Subscriber
Using the Service

The device supports the BLF service by default. The BLF service requires that both
the subscriber and subscribed user should be intra-office SIP users, and the IP
phone of the subscriber should support multi-purpose keys for subscribing to the
conversation status of another user.
Figure 2-50 shows the user conversation status subscription process between user
A and user B based on the BLF service.
Figure 2-50 User conversation status subscription process

Subscriber Subscribed user
PBX
User A User B
1. Registration request 1. Registration request

2. Registration succeeds 2. Registration succeeds
3. Subscription request
4. Subscription result
1. User A and user B (SIP users) send registration requests to the SIP server of
the PBX.
2. User A and user B register with the SIP server successfully.
3. User A sends a request to the PBX to subscribe to the conversation status of
user B.
4. The PBX checks the status of user B. If the status meets the subscription
requirement, the PBX sends a message to user A, indicating that the

NetEngine AR
subscription is successful. Otherwise, the subscription fails, and a message is

sent to inform user A.
After user A subscribes to the conversation status of user B, the PBX displays user
B status on the IP phone of user A. For example, when user C calls user B and the
status of user B changes, the PBX informs user A of the conversation status of user
B, as shown in Figure 2-51.
Figure 2-51 BLF service process

Subscribed
Subscribed user B PBX User C
user A
1. User C calls user B 1. User C calls user B

2. The IP phone of user B
is ringing
Display the 3. Inform the ringing status
4. Play ringback tones
ringing status 1
of user B 5. User B picks up
the phone
User B is in conversation with user C
Display the in 6. Inform the in conversation status
conversation 2
7. User B or C hangs up,
status of user B
and the conversation ends
Display the idle 8. Inform the idle status
3
status of user B
After the BLF service is enabled successfully, user A can obtain the conversation
status of user B based on the multi-purpose keys on the IP phone. Description of
multi-purpose keys on the IP phone:
● Steady green: The subscribed user is idle, including pickup and dialing.
● Steady red: The subscribed user is in conversation.
● Blinking red: The subscribed user is ringing.
● Off: The subscribed user is offline.
2.14.4 Configuring the Service Password

This section describes how to change the service password. You can configure the
service password call prefix through the administrator and dial the call prefix to
change the service password, or directly change the service password through the
administrator.
Configuring the Service Password

3 Enter the PBX user view. pbxuser name
4 Configure the service password. telno service-password cipher

NetEngine AR
(Optional) Configuring a Call Prefix

When the PBX needs to provide the service password change service for end users,
the call prefix must be configured.

view.

access the view.

type and call callcategory attribute type) to new-
prefix. management.
Set attribute to 121
for the service
password call prefix.

a number for value
number analysis.
7 Save the save -

configuration.
Change the service password to 012345 for user 7000 and configure the call prefix
*121* as the service password call prefix.
[Huawei] voice
[huawei-voice] pbxuser 7000
[Huawei-voice-pbxuser-7000] pbxuser 7000
[Huawei-voice-pbxuser-7000] telno service-password cipher
Please input user password(6-6 chars):*******

NetEngine AR
Using the Service

User A picks up the phone and dials *121*SSSSSS*NNNNNN#. If the old password
is correct and the new password is valid, the announcement "The new service you
applied for has been registered" is played, indicating that the password is changed
successfully.
NOTE
● SSSSSS indicates the six-digit old password. NNNNNN indicates the six-digit new
password. The default username and password are available in AR Router Default
Usernames and Passwords (Enterprise Network or Carrier). If you have not obtained
the access permission of the document, see Help on the website to find out how to
obtain it.
● Both the old password and new password must contain six digits.

This section describes special topics including number change, call limiting, and
intelligent routing.
2.15.1 Creating and Switching a Phone System Announcement

This section describes how to record and switch a phone system announcement of
the PBX to the required language.
NOTE
The methods of creating, saving, and loading RBT are the same as those for system
announcements. However, the RBT file can be loaded only after it is saved in .wav format.
Context
A phone system announcement is automatically played by the system.
The PBX provides the bilingual announcement.
You can load the announcement file package to switch the announcement to the
required language. If the announcement of the required language is unavailable,
record the phone system announcement and load it.
Creating the Phone System Announcement and RBT

Pay attention to the following points:
● The recorded announcement file must be in .wav format.
● The creating duration cannot exceed 60s, and the file size between 400 KB
and 480 KB. The recorded file must be saved in CCITT A-Law format and with
the attributes of 8.000 kHz, 8 bit, Mono.
● Notice the announcement speed by referring to typical phone system
announcements at the local area.
● The recorded announcement file is named with the channel name. When you
load an announcement file, the new announcement will overwrite the original

NetEngine AR
announcement with the same name. Name the announcement file according
to Voice List.
It is recommended that you use third-party software (not described in this
document), for example, Audacity, to record custom announcements.
NOTE
When creating the digit collection announcement for the automatic switchboard service, you are
advised to add the description "followed by the pound key." For example, you are advised to
record "Please dial the extension number" as "Please dial the extension number followed by the
pound key."
Switching the Phone System Announcement

1. (For the RBT, skip this step.) Obtain the phone system announcement file.
– If there is no required announcement file, record the phone system
announcement according to Creating the Phone System
Announcement and RBT, and contact the service provider to convert the
recorded announcement file into .res or .cc.
– If the required announcement file already exists, download it.
2. Load the announcement file.
NOTE
Using FTP to transfer files may bring some security risks because FTP is an insecure
protocol. It is recommended that you use the FTPS protocol.
St Action Command Description

ep
1 Enter the FTP ftp [ [ -a source-ip-address | -i Enter the FTP

client view and interface-type interface-number ] user name and
establish a host-ipv4 [ port-number ] ] password.
control
connection with
the FTP server.
2 Download the get remote-filename [ local- Before

announcement filename ] downloading
file from the the
remote FTP announcement
server and save file, enable the
it locally. FTP server. The
FTP user name
and password
are the same
as those for
entering the
FTP client view.

system view.

NetEngine AR

ep
4 Load the load voice-package filename -

announcement
file.
5 Specify the startup voice-package filename After the

announcement announcement
file for the next file for the next
startup. startup is
specified, the
PBX uses the
specified
announcement
file for the next
startup.
6 Save the save -

configuration.
Example
Load the announcement file voice_uc_english_v11.res and specify the
announcement file for the next startup.
<Huawei> ftp 172.16.15.201
[huawei-ftp] get voice_uc_english_v11.res
[huawei-ftp] quit
<Huawei> load voice-package voice_uc_english_v11.res
<Huawei> startup voice-package voice_uc_english_v11.res
<Huawei> save
Verification
Check display startup The Startup voice-files command output

whether the displays the loaded announcement file:
announceme [Huawei-voice]display startup
nt file is Next startup patch package: null
successfully Startup voice-files: flash:/
voice_uc_english_v11.res
loaded. Next startup voice-files: flash:/
voice_uc_english_v11.res
2.15.2 Configuring High-level Users to Preempt Trunks to

Implement Preferred Calls
This section describes how to configure reserved trunk resources. When an
emergency occurs and outgoing calls need to be made but trunks are all occupied,
high-level users can use reserved trunk resources to call inter-office users.

NetEngine AR
Context
When trunk resources are all occupied, high-level users can use reserved trunk
resources to call inter-office users. When there is no reserved trunk resource, users
with super rights can preempt trunk resources occupied by other users to call
inter-office users.
The user rights are as follows:
● Default right: default
● Common right: normal
● Advanced right: advanced
● Super right: super
User rights of default, normal, advanced, and super are in ascending order of
priority.
Prerequisites
A PBX user has been configured. For details on how to configure a PBX user, see
2.8 Configuring a PBX User.
Procedure

system
view.
2 Enter the voice -

voice view.

user view.
4 Configure priority { default |normal | advanced | -

the user super }
right.
5 Return to quit -
the voice
view.
6 Enter the trunk-group trunk-group-name -

trunk
group
view.

NetEngine AR
7 Configure reserved-circuit number number value After reserved

reserved priority { default |normal | advanced | trunk
trunk super } resources are
resources. configured for
a user level,
usage of
reserved trunk
resources is
limited by the
maximum
number of
concurrent
calls. For
example, when
there are 10
channels of
reserved trunk
resources and
the maximum
number of
concurrent
calls is 5, only
5 channels of
reserved trunk
resources can
be used.
NOTE
Run the maxcr
maxcr-value
command in
the trunk view
to set the
maximum
number of
concurrent
calls.
8 Save the save -

configurati
on.
Example
Configure the advanced right for user 7000 and 10 reserved trunk resources for
users with the advanced right.
[Huawei] voice
[Huawei-voice-pbxuser-7000] priority advanced
[Huawei-voice] trunk-group 1
[Huawei-voice-trunkgroup-1] reserved-circuit number 10 priority advanced
[Huawei-voice-trunkgroup-1] save

NetEngine AR
Verification
Verify the trunk group display voice trunk- The parameter values in
configuration. group the command output are
consistent with the
settings.
2.15.3 Configuring Preemption for High-Priority Users

This section describes how to configure preemption for high-priority users.
Context
Important calls to user A cannot be connected when user A is engaged in a call. In
this case, preemption can be configured for user A. For example, user C has a
higher priority than user B. If user C calls user A during a conversation between
user A and user B, user A ends the current conversation immediately and
establishes a call with user C.
Prerequisite
A PBX user has been configured. For details on how to configure a PBX user, see
2.8 Configuring a PBX User.
Procedure

system view.

view.
3 Create a call priority-group priority-group-name If a call priority

priority group. group has been
created, this
command
directly
displays the
priority-group
view.
By default, no
call priority
group is
created.

NetEngine AR
4 Configure the telno telno-value priority priority- A smaller value

priority of each level indicates a
user. higher priority.
By default, the
priority is not
configured for
a calling user.

voice view.

user view.
7 Associate the priority-group priority-group-name Associate a

call priority PBX user with
group with a the call priority
PBX user. group created
in step 3.
By default, no
call priority
group is
associated with
a PBX user.
8 Save the save -

configuration.
Example
# Set the name of the call priority group associated with the SIP user 7001 to
group1, and the priority of the user 28987000 to 0.
[Huawei] voice
[Huawei-voice] priority-group group1
[Huawei-voice-priority-group-group1] telno 28987000 priority 0
[Huawei-voice-priority-group-group1] quit
[Huawei-voice-pbxuser-7001] telno 7001
[Huawei-voice-pbxuser-7001] priority-group group1

NetEngine AR
Verification
Verify the configuration display voice priority- The parameter values in

of users in the call group the command output are
priority group. consistent with the
settings.
<Huawei> display voice
priority-group
Priority group name :
group1
Priority telno :
28987000
Priority level :0
2.15.4 Configuring a Call Limiting Policy

The PBX limits calls meeting conditions.
2.15.4.1 Configuring Call Limiting Over a Trunk

This section describes how to configure call limiting over a trunk. Call limiting over
a trunk prevents toll fraud.
Context
Call rights are classified as follows:
● inter: intra-office call
● local: local call
● ddd: national toll call
● idd: international toll call
● all: all calls
Call limiting over a trunk prevents incoming calls from being forwarded over a
trunk. As shown in Figure 2-52, user A on the PBX is not allowed to make a
national or international toll call to user E.
Figure 2-52 Networking of call limiting over a trunk
User A User E
AT0
trunk
SIP IP
PBX trunk
IP PBX
User B User C User D

NetEngine AR
NOTE
Call limiting over a trunk applies to the scenario where incoming calls are forwarded over a
trunk.
Prerequisites
An AT0 trunk group and SIP IP trunk have been configured. For detailed
configuration, see 2.11 Configuring Trunk Groups.
Procedure
p

system
view.
2 Enter the voice -

voice view.
3 The trunk trunk-group trunk group name -

group view
is
displayed.
4 Configure outgoing-right To prevent the

the call-out inter-office users
right. who initiate
incoming calls
from initiating
outgoing calls, set
the call right to
inter.
5 Save the save -

configurati
on.
Example
When you configure the trunk outgoing call right, call fraud should be prevented.
You are advised to set the outgoing call right of AT0 trunk group 2 to inter and
set the outgoing call right of trunk group 1 of SIP IP to local. The default value of
Outgoing right is inter&local.
[Huawei] voice
[Huawei-voice-trunkgroup-2] outgoing-right inter enable
[Huawei-voice-trunkgroup-2] quit
[Huawei-voice-trunkgroup-1] outgoing-right local enable
[Huawei-voice-trunkgroup-1] save

NetEngine AR
Verification
Verify the display voice trunk-group The parameter

trunk values in the
group command output
configurati are consistent
on. with the settings.
2.15.4.2 Configuring 32-Level Right Call Barring

32-level rights are user-defined extended rights. These rights can be used together
with call prefixes and trunk groups to bar calls of specified users.
Context
Basic user call rights include Inter (intra-office call), Local (local call), DDD
(national toll call), and IDD (international toll call). As enterprises require more
and more call barring modes, the four basic call rights cannot meet enterprises'
service requirements. To extend users' call rights, the PBX provides new 32-level
rights (identified as c1 to c32).
For calls initiated by intra-office users, the PBX matches user rights with call prefix
rights to perform call barring control. Calls can be connected only when users
have the basic call rights and 32-level extended rights of call prefixes. Figure 2-53
shows the detailed call barring process.

NetEngine AR
Figure 2-53 32-level right call barring process for calls initiated by intra-office
users
An intra-office user initiates a call
The PBX analyzes the called number

(parses the call prefix)
No
Does the user have the basic
call right of the call prefix?
Yes
No
Does the user have the 32-level
extended right of the call prefix?
Yes
Process the call Reject the call
For trunk-based incoming calls, the PBX first analyzes called numbers. If calling
number discrimination configurations are detected, the PBX matches calling
number discrimination rights with call prefix rights to perform call barring control;
otherwise, the PBX matches trunk group rights with call prefix rights to perform
call barring control. Calls can be connected only when calling number
discrimination configurations or trunk groups have the basic call rights and 32-
level extended rights of call prefixes. Figure 2-54 shows the detailed call barring
process.

NetEngine AR
Figure 2-54 32-level right call barring process for trunk-based incoming calls
A trunk-based incoming call

arrives
The PBX analyzes the called

number
Yes No
Is calling number
discrimination configured?
Does calling number

Does the trunk group have the
discrimination have the basic basic call right of the call prefix?
call right of the call prefix? No No
Yes Yes
Does calling number

discrimination have the 32-level Does the trunk group have the
extended right of the call prefix? No No 32-level extended right of the
call prefix?
Yes Yes
Process the call Process the call
Reject the call
Prerequisites
● Prefixes have been configured. For details, see 2.9 Configuring a Call Prefix.
● For barring of calls initiated by intra-office users, users have been configured.
For details, see 2.8 Configuring a PBX User.
● For barring of trunk-based incoming calls, calling number discrimination has
been configured or trunk groups have been configured. For details, see 2.10.1
Calling Number Discrimination or 2.11 Configuring Trunk Groups.

NetEngine AR
Configuring 32-Level Rights for a Call Prefix


system view.

view.
3 Enter the prefix callprefix callprefix- -

profile view. name
4 Configure 32- custom-right right- Only one 32-level right can
level rights. level be configured.
To clear configured 32-
level rights, run the undo
custom-right command.
5 (Optional) vu-custom-right right- If the call type of the

Configure 32- level current call prefix is
level rights for interactive voice response
two-stage (IVR) service, and the
dialing. service involves two-stage
dialing, you can configure
32-level rights for two-
stage dialing as required.
Multiple 32-level rights can
be configured.
vu-custom-right
command.

voice view.
7 Save the save -

configuration.
Configuring 32-Level Rights for a User


user view.
2 Configure 32- custom-right right- Multiple 32-level rights can


NetEngine AR

voice view.
4 Save the save -

configuration.
Configuring 32-Level Rights for a Trunk Group

1 Enter the trunk trunk-group name -

group view.


voice view.
4 Save the save -

configuration.
Configuring 32-Level Rights for Calling Number Discrimination

1 Enter the calling caller-change caller- -

number change-name
discrimination
view.


voice view.
4 Save the save -

configuration.

NetEngine AR
NOTE
Rights configured using the outgoing-right command in the calling number discrimination
view apply only to barring of trunk-based outgoing calls, and do not apply to barring of
trunk-based incoming calls.
PBX1 has users 7xxx and 8xxx, which correspond to intra-office prefixes 7 and 8
respectively. PBX1 connects to PBX2 through trunk group sipip01, and the
corresponding outgoing prefix is 5. You can configure 32-level extended rights on
PBX1 to implement the following:
● On PBX1, only users 7xxx can make calls to users on PBX2.
● Trunk-based incoming calls from PBX2 can be connected only to users 7xxx on
PBX1.
● On PBX1, users 7xxx and 8xxx can make calls to each other.
Figure 2-55 shows the configuration scenario.
Figure 2-55 Configuration scenario example

Outgoing prefix 5: making calls to PBX2 users
Intra-office prefix 7: making calls to users 7xxx
Intra-office prefix 8: making calls to users 8xxx
Trunk group
PBX1 (sipip01)
PBX2
... ... ...

7000 7001 8000 8001 5000 5001
[Huawei] voice
# Configure 32-level rights for a call prefix.
[Huawei-voice-callprefix-5] custom-right c1
# Configure 32-level rights for users. (Users 7000 and 8000 are used as examples.)
[Huawei-voice-pbxuser-7000] custom-right c1 c2 c3
[Huawei-voice-pbxuser-8000] custom-right c2 c3
# Configure 32-level rights for a trunk group.
[Huawei-voice] trunk-group sipip01
[Huawei-voice-trunkgroup-sipip01] custom-right c2
[Huawei-voice-trunkgroup-sipip01] quit
[Huawei-voice] save

NetEngine AR
Verification
Verify calls based on the preceding configuration example. Table 2-10 describes
the expected result.
Table 2-10 32-level right call barring verification
- User 7001 on User 8001 on User 5001 on

PBX1 PBX1 PBX2
User 7000 on √ √ √
PBX1
User 8000 on √ √ x
PBX1
User 5000 on √ x -
PBX2
√ indicates that the call is connected; x indicates that the call is rejected.
2.15.4.3 Configuring Call Limiting Upon CPU Overload

This section describes how to configure call limiting upon CPU overload.
Context
CPU overload flow control allows the PBX to monitor the CPU usage and limit
calls according to upper and lower thresholds for CPU overload flow control,
preventing device breakdown caused by high CPU overload. Table 2-11 describes
CPU overload levels.
Table 2-11 CPU overload levels
CPU Overload Level Call Limiting Default CPU Overload

Percentage Threshold
1 30% 95%
2 100% 99%
NOTE
By default, the CPU overload flow control is as follows:

● When the CPU usage is smaller than 95%, CPU overload flow control is not performed.
● When the CPU usage is larger than 95% but smaller than 99%, the PBX limits 30% calls.
● When the CPU usage is larger than 99%, the PBX limits all calls.

NetEngine AR
Procedure
Step Action Command
3 Set upper and lower thresholds for CPU cpu-flowcontrol { low-level

overload flow control. low-level-value | high-level
high-level-value }*
Example
Set the thresholds to 70% and 90% for CPU overload levels 1 and 2 respectively.
[Huawei] voice
[Huawei-voice] cpu-flowcontrol low-level 70 high-level 90
[Huawei-voice] save
Verification
Verify the display voice cpu-flowcontrol [Huawei]display voice cpu-

CPU flowcontrol
overload CPU-FlowControl Lowlevel-value: [70%]
CPU-FlowControl Highlevel-value: [90%]
flow
control
configurati
on.
2.15.4.4 Configuring Static Bandwidth Control for Call Barring

You can configure static bandwidth control to limit the bandwidth consumed by
calls. This function ensures quality of voice services.
Context
If voice flows are transmitted over an IP network, bandwidth for voice services is
limited. For example, when The device is used as a media proxy agent for SIP and
POTS users, calls of these users are transmitted over an IP network. The more calls
are transmitted, the more bandwidth is consumed. The static bandwidth control
function limits the bandwidth for voice calls and rejects new calls when the
bandwidth limit is exceeded. Figure 2-56 shows the networking.

NetEngine AR
Figure 2-56 Networking for static bandwidth control
PBX IP PBX
POTS phone IP phone IP phone
Call admission control (CAC) limits new voice calls to ensure quality of existing
calls when the uplink bandwidth of the device is insufficient.
The traditional phone system is a circuit switching system where voice calls
exclusively occupy lines. The number of voice calls depends on the number of
trunk lines on a TDM PBX, as shown in Figure 2-57. User A and user B exclusively
occupy trunk lines of the TDM PBX, so the TDM PBX rejects calls initiated by user
C. User C hears the busy tone.
Figure 2-57 Traditional Voice Call Control
Physical Trunks
Third call
rejected
TDM
PBX STOP
VoIP runs over the IP network and the VoIP network is a packet switched network
(PSN) where calls do not exclusively occupy lines. The limited link bandwidth of
the IP network is used to transmit data of other applications in addition to voice
data. Calls on the VoIP network are limited by the IP network bandwidth, and too
many calls may exhaust bandwidth. If new calls are connected, voice quality is
affected. CAC allows the device to reject new voice calls when the number of
existing voice calls reaches the maximum value. This ensures voice quality. As
shown in Figure 2-58, the IP link allows a maximum of two-channel voice calls. If
one-channel voice calls are connected, voice call quality on this channel is

NetEngine AR
affected. After CAC is enabled, the device rejects voice calls of user C because user
A and user B have occupied two-channel voice calls. In this manner, voice call
quality of user A and user B is ensured.
Figure 2-58 VoIP Voice Call Control
IP IP
network network
After CAC is
Before CAC is enabled, voice calls
IP WAN link enabled, voice over the third
Bandwidth meets calls on the third channel are
requirements for channel can be rejected to ensure
high-quality voice set up. The voice quality of voice
communication quality is affected calls over the other
over two channels. due to insufficient two channels.
bandwidth.
Router Router
STOP
User A User B User C User A User B User C
Different voice codec modes and packetization intervals affect bandwidth

occupied by each voice channel.
● Bandwidth of each voice channel = INT [(Prefix_length/Page_Time) +
PayLoad]
Then the maximum number of concurrent calls is calculated based on service
characteristics and the maximum number of calls allowed by CAC is set.
● Prefix_length is the packet prefix length and has a fixed value of 624 bits.
● Page_Time specifies the packetization interval.
● PayLoad specifies the size of each packet.
● INT[] indicates that the value is rounded down. For example, INT[95.6]
indicates that value 95 is used.
In different codec modes, the size of each voice data packet is as follows:
● G.711U: 64 K
● G.711A: 64 K
● G.729: 8 K
● G.723_1: 7 K
● G.723_Low: 6 K

NetEngine AR
● G.726_16K: 16 K
● G.726_24K: 24 K
● G.726_32K: 32 K
● G.726_40K: 40 K
In different codec modes, the packetization interval is as follows:
● G.711U: 5 ms, 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
● G.711A: 5 ms, 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
● G.729: 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
● G.723_1: 30 ms, 60 ms
● G.723_Low: 30 ms, 60 ms
● G.726_16K: 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
● G.726_24K: 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
● G.726_32K: 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
● G.726_40K: 10 ms, 20 ms, 30 ms, 40 ms, 50 ms, 60 ms
When G.711A is used, the bandwidth occupied by voice calls in each channel is as
follows:
● INT [(624/20) + 64] = 95 kbit/s
If the number of concurrent calls is 100, the maximum CAC bandwidth should be
9500 kbit/s (95 kbit/s x 100).
NOTE
CAC bandwidth occupied by voice calls is relevant to DSP resources. If a call occupies one
DSP resource, the bandwidth occupied by the call is the required bandwidth. If a call
occupies two DSP resources, the required bandwidth is 2 multiplied by the bandwidth
calculated through the algorithm. If signaling media proxy is used, communication between
the local user and signaling media proxy occupies one DSP resource, and communication
between the signaling media proxy and the remote user occupies one DSP resource, the call
between the local user and remote user occupies two DSP resources.
The procedure for implementing CAC is as follows:

1. When a call is to be set up, the device uses the occupied bandwidth
subtracted from the maximum CAC bandwidth and checks whether the
remaining bandwidth is 0. If the remaining bandwidth is less than or equal to
0, the call is rejected. If the remaining bandwidth is larger than 0, the reserved
bandwidth is used.
2. The device reserves bandwidth for the call.
Reserved bandwidth = INT[(624/10 ms) + PayLoad]
The packetization interval is 10 ms, and PayLoad is the size of each data
packet in the specified codec mode.

NetEngine AR
NOTE
● If the remaining bandwidth is larger than 0 but smaller than the bandwidth
required by the call, the device still reserves bandwidth for the call. The reserved
bandwidth is the remaining bandwidth. For example, a voice call occupies
bandwidth of 95 kbit/s and the remaining bandwidth is 90 kbit/s. The device does
not reject the call, and still reserves the bandwidth for the call.
● During call negotiation, set the reserved bandwidth to be larger than the
bandwidth required by a call to ensure voice quality of the call to be set up.
3. After call negotiation is complete, the call is set up and the reserved
bandwidth is released. Then the bandwidth occupied by the call is used.
As shown in Figure 2-59, the router uses G.711 and packetization interval of 20
ms. During call negotiation, the router reserved bandwidth of 126 kbit/s for user
B. After a call is set up for user B, the call occupies actual bandwidth of 95 kbit/s.
Figure 2-59 Bandwidth usage during call negotiation and call setup
Maximum CAC bandwidth: 1024 kbit/s

Remaining bandwidth = 1024 - 95 - 126 = 803 kbit/s
Router
User A
G.711A IP Wan Link
IP
Call occupies of User A: 95 kbit/s.
network
User B
The device reserves 126 kbit/s
bandwidth for user B call negotiation.
Remaining bandwidth = 1024 – 95 - 95 = 834

kbit/s
User A Router
G.711A IP Wan Link
Call occupies of User A: 95 kbit/s. IP

network
User B User B call is set up.
Voice communication
occupies 95 kbit/s.
Prerequisites
● A PBX user has been configured. For details about the configuration, see 2.8
Groups.

NetEngine AR
Procedure
ep

system
view.
2 Enter the voice -

voice view.
3 Enter the trunk-group trunk-group-name NOTE

trunk If The device
functions as a media
group
proxy agent to route
view. outgoing calls of SIP
users, configure a SIP
server. For details
about the
configuration, see
2.5.2 Configuring
Media and Signaling
IP Address Pools.
4 Enable media-bandwidth-control enable -

static
bandwidth
control.
5 Set the media-bandwidth-control max- The value is an

maximum bandwidth max-bandwidth-value integer that ranges
uplink from 1 to 5000. The
media unit is 100 kbit/s.
bandwidth The default value is
for voice 5000, representing
flows. 500000 kbit/s.
NOTE
If the bandwidth
consumed by
common voice calls
on the IP network
reaches the
maximum uplink
bandwidth, new calls
are rejected.
6 Save the save -

configurati
on.
Example
Enable static bandwidth control for the trunk group sipip and set the maximum
uplink media bandwidth to 25600 kbit/s.

NetEngine AR
[Huawei] voice
[Huawei-voice] trunk-group sipip
[Huawei-voice-trunkgroup-sipip] media-bandwidth-control enable
[Huawei-voice-trunkgroup-sipip] media-bandwidth-control max-bandwidth 256
[Huawei-voice-trunkgroup-sipip] save
Verification
Verify the configuration display voice media- The parameter values in

of the post-routing bandwidth-control the command output are
number change plan. consistent with the
settings.
2.15.4.5 Configuring Call Limiting When the CDR Pool Is Full

This section describes how to configure call limiting when the CDR pool is full.
Procedure
The PBX enforces the following call limiting policies when the CDR pool usage
reaches 90%:
● The PBX does not limit calls, and overwrites old CDRs with new CDRs.
● The PBX does not limit calls, and does not generate CDR for new calls. The
old CDRs are kept.
● The PBX does not allow new calls.
By default, the PBX does not limit calls, and overwrites old CDRs with new CDRs.
To cancel call limiting and keep old CDRs, perform the configuration on the CDR
server. For low to configure call limiting, refer to the call limiting policy
configuration performed when the CDR pool is full.

system
view.
2 Enter the voice -

voice
view.
3 Enter the cdr-server -

CDR
server
view.

NetEngine AR
4 Configur billpoolfull { billdiscard ● billdiscard: indicates

e call enable | calleelimit enable | whether new CDRs
limiting callerlimit enable } overwrite old CDRs. The
when value disable indicates
the CDR that new CDRs overwrite
pool is old CDRs, while enable
full. indicates that new CDRs
do not overwrite old
CDRs.
● calleelimit: limit local
calls of the called. The
value disable indicates
that allowed to limit local
calls of the called, while
enable indicates that not
allowed to limit local calls
of the called.
● callerlimit: limit local
calls of the caller. The
value disable indicates
that allowed to limit local
calls of the caller, while
enable indicates that not
allowed to limit local calls
of the caller.
5 Save the save -

configur
ation.
Example
Configure the PBX to limit local calls of the called party when the CDR pool is full.
[Huawei] voice
[Huawei-voice] cdr-server
[Huawei-voice-cdr-server] billpoolfull calleelimit enable local enable
[Huawei-voice-cdr-server] save
Verification
Verify the call limiting display voice cdr-server The parameter values in
configuration when the the command output are
CDR pool is full. consistent with the
settings.

NetEngine AR
2.15.4.6 Configuring Reanalysis Prefix Call Restriction

This topic describes how to configure the reanalysis prefix call restriction.
Context
Based on the user dialing habit (9+number of an external user), prefix 9 is
configured for prefix reanalysis. If prefix 9 has the local call right, a user with the
local call right can make a toll call by dialing 9 and a toll call number if the
reanalysis prefix is not configured. After the reanalysis prefix is configured, a user
can make toll calls only when the user has the toll call right. If prefix 9 is
configured for playing the two-stage dial tone, a user can hear the two-stage dial
tone when dialing the prefix. The principle is as follows: If POTS user 77771000
wants to call local number 88881000, the POTS user dials 9 and when hearing the
two-stage dial tone, dials 88881000. Prefix 9 is deleted for the called number
change, and then number reanalysis is performed.
NOTE
The system can play the two-stage dial tone only for POTS users.
Prerequisites
The local call right has been configured for prefix 9 and the local outgoing prefix,
the ddd call right has been configured for prefix 0, and the idd call right has been
configured for prefix 00. For details, see 2.9 Configuring a Call Prefix.
Procedure
p

system view.
2 Enter the voice -

voice view.
3 Create a pre- beforeroute-change name By default, no

routing pre-routing
number number
change plan change plan is
and enter the configured.
pre-routing
number
change plan
view.

prefix to the
pre-routing
number
change plan.

NetEngine AR

p
5 Configure a caller { del-then-insert del-offset del- -

calling len insert-telnum | del del-offset del-len
6 Configure a called { del-then-insert del-offset del- -

called len insert-telnum | del del-offset del-len
7 (Optional) call new-attribute enterprise -

Configure an enterprise-name [ dn-set dn-set-
enterprise name ]
and a DN set
in the pre-
routing
number
change plan.
8 Configure the call re-analyse reanalyze-value The PBX

PBX whether supports
to reanalyze reanalysis for
changed a maximum of
numbers. five times.
9 Configure the call dual-dial-tone-flag dual-dial-tone- To enable the

PBX whether flag-value PBX to play
to play the the two-stage
two-stage dial tone,
dial tone. delete the
dialed number.
10 Save the save -

configuration.
Example
Bind prefix 9 to pre-routing number change plan 0, configure prefix 9 as the
outgoing prefix, and enable the system to delete prefix 9 for prefix reanalysis.
[Huawei] voice
[Huawei-voice] beforeroute-change 0
[Huawei-voice-beforeroute-change-0] callprefix 9
[Huawei-voice-beforeroute-change-0] caller no-change
[Huawei-voice-beforeroute-change-0] called del 1 1
[Huawei-voice-beforeroute-change-0] call re-analyse enable
[Huawei-voice-beforeroute-change-0] call dual-dial-tone-flag enable
[Huawei-voice-beforeroute-change-0] save

NetEngine AR
Verification
Verify the pre- display voice beforeroute-change The parameter values

routing [ name ] in the command output
number are consistent with the
change plan settings.
configuration.
2.15.4.7 Configuring Call Limiting Based on the Remote IP Address

This section describes how to configure call limiting based on the remote IP
address.
Context
When the same remote IP address is configured for multiple SIP trunks including
SIP IP trunks, SIP AT0 trunks, and SIP PRA trunks, call limiting can be configured
based on the remote IP address. Figure 2-60 shows the networking.
Figure 2-60 Configuring call limiting based on the remote IP address
IP:192.168.1.1
SIP PRA Port:5061
SIP AT0 IP:192.168.1.1

PBX Port:5062
SIP IP
IP:192.168.1.1
Port:5063
Analog IP
phone phone
The PBX connects to remote devices through SIP PRA trunks, SIP AT0 trunks, and
SIP IP trunks, the remote IP address of the trunks is 192.168.1.1, and the call
limiting policy is configured on the PBX. The policy defines a maximum number of
200 online users based on the destination IP address of 192.168.1.1.
When more than 200 online calls are initiated by the PBX to remote devices or
remote devices to the PBX through trunks, the PBX cannot initiate calls through
the trunks. Other devices with the IP address of 192.168.1.1 cannot initiate calls to
the PBX.
The call liming policy is valid for only SIP trunks. If route re-selection upon a
failure is enabled, calls can be still initiated through H.323 and BRA trunks.

NetEngine AR
Procedure
Ste Operation Command Description
p

system view.
2 Enter the voice -

voice view.
3 Enable call cac destination-ip ip-address max- -

limiting by call-number max-call-number-value
setting the
maximum
uplink
bandwidth.
4 Save the save -

configuration.
Example
Set the maximum number of calls over SIP trunks based on the remote IP address
of 192.168.1.1 to 200.
[Huawei] voice
[Huawei-voice] cac destination-ip 192.168.1.1 max-call-number 200
[Huawei-voice] save
Verification
Check the display voice cac destination-ipip- The parameter values

maximum address in the command output
number of are consistent with the
online calls settings.
and the
number of
current online
calls based on
a specified IP
address.
2.15.5 Configuring Intelligent Routing

Intelligent routing allows the PBX to select routes according to user-defined route
selection policies. That is, the PBX intelligently selects trunk links to transmit voice
services to implement load balancing with the minimum expense.

NetEngine AR
2.15.5.1 Configuring Re-routing Upon a Failure

When the PBX fails to select a route according to the configured route selection
policy, it re-selects a route according to the backup route policy specified by the
failure index.
Context
Re-routing is configured in the following scenarios:
● Routing failure
The cause is that the trunk link is faulty.
● Call failure
The possible cause is that the SIP trunk link is faulty or the remote device is
faulty. The remote device is faulty when the remote device cannot respond to
a call request initiated by the PBX. The PBX determines whether the SIP trunk
link is normal according to the heartbeat. After the SIP trunk configuration is
complete, run the reset command to activate the trunk link. After the trunk
link is activated, the heartbeat is enabled.
As shown in Figure 2-61, the PBX connects to carrier A through the R2 trunk, and
the callroute is 1; the PBX connects to carrier B through the SIP trunk, the
callroute is 2, and the inter-office call prefix is 1. Re-routing is configured upon a
route selection failure. Normally, outgoing calls on the PBX are made through the
R2 trunk. When calls fail to be routed to carrier A over the R2 trunk, calls are
routed over the SIP trunk.
Figure 2-61 Networking of re-routing upon a route selection failure
Carrier A PBX Carrier B

R2 SIP IMS
network
Callroute 1 Callroute 2
Prerequisites
● A trunk group has been configured. For details on how to configure a trunk
group, see 2.11 Configuring Trunk Groups.
● Call routes 1 and 2 have been configured. For details on how to configure a
call route, see 2.12 Configuring a Call Route.
Configuring Re-routing
Step Action Command
3 Create a call route plan. callroute callroute-name

NetEngine AR
Step Action Command
5 Create re-routing. reroute-deal name
6 Configure re-routing re-callroute name

information.
7 Configures a failure type of re- failed-type type value

routing.
Configure Prefix Re-routing

Step Action Command
1 Create a call prefix profile and callprefix callprefix-name

enter its view.
2 Configure a call prefix. prefix prefix
3 Configure routing information callroute callroute-name

for the call prefix.
Example
Configure callroute 1 to carrier A and set the call prefix to 2, configure callroute 2
to carrier B, and configure the link of callroute 2 as the backup of call route 1
when route selection fails. Specify re-routing upon a routing failure as the re-
routing type.
[Huawei] voice
[Huawei-voice] callroute 1
[Huawei-voice-trunkgroup-1] callroute 1
[Huawei-voice] reroute-deal 1
[Huawei-voice-reroute-deal-1] re-callroute 2
[Huawei-voice-reroute-deal-1] failed-type route-failed
[Huawei-voice-reroute-deal-1] callroute 1
[Huawei-voice-reroute-deal-1] quit
[Huawei-voice-callprefix-1] callroute 1

NetEngine AR
Verification
Verify display voice callroute [Huawei] display voice

the callroute
configura Call route name :1
Call route select type : none
tion of Call route name :2
the call Call route select type : none
route
plan.
Verify display voice callprefix [Huawei] display voice

the call [ callprefix-name ] callprefix 1
prefix Callprefix :1
Enterprise : default
configura Dn-set : DefaultDialPlan
tion. Prefix :1
Call category :
Call attribute :
Minimum length :4
Maximum length :8
Ring delay(s) :0
Long cli switch : disable
Caller map switch : disable
Called map switch : disable
Call route name : 1
Check re- display voice reroute-deal [Huawei] display voice reroute-

routing [ reroute-deal-name ] deal 1
informati Spec: 256, Records : 1
--------------------------
on. Reroute-deal name : 1
Recall route name : 2
Call route name : 1
Failed type value : route-failed
2.15.5.2 Configuring a Call Route Plan Based on Charge Rates

The PBX searches for routes in ascending order of charge routes until it finds idle
circuits. If routes have the same charge rate, the PBX searches for routes in
ascending order of trunk group numbers. If no idle circuit is found, the PBX
searches for routes for which no charge rate is configured in ascending order of
trunk group numbers until it finds idle circuits.
Context
To select routes for outgoing calls based on charge rates, configure intelligent
routing based on charge rates. For example, the charge rates for outgoing calls
over the R2 trunk and SIP trunk are 1 and 2 respectively. As shown in Figure 2-62,
the PBX connects to the carrier network through the R2 trunk, and the trunk
group number is 1; the PBX connects to the carrier network through the SIP trunk,
the trunk group number is 2, and the inter-office call prefix is 1.

NetEngine AR
Figure 2-62 Networking of routing

Carrier PBX
trunkgroup 1 R2
trunkgroup 2 SIP
Prerequisites
Groups.
Creating Routing Information


system view.
2 Enter the voice -

voice view.
3 Create a call callroute callroute-name -

route plan.
4 Configure selecttype call route select type Set the route

intelligent value selection type to
routing. chargebase.
5 Return to quit -
the voice
view.

trunk group
view.
7 Bind a call callroute callroute-name charge must be

route to the specified.
trunk group.
8 Save the save -

configuratio
n.

NetEngine AR
Configure Prefix Re-routing

Step Action Command

enter its view.

Example
Configure a call route plan based on charge routes. The charge rates of call routes
1 and 2 are 1 and 2 respectively. Call route 1 corresponds to trunk group 1 and
call route 2 corresponds to trunk group 2.
[Huawei] voice
[Huawei-voice-callroute-1] selecttype chargebase
[Huawei-voice-callroute-1] quit
[Huawei-voice-trunkgroup-1] callroute 1 charge 1
[Huawei-voice-trunkgroup-2] callroute 1 charge 2
Verification
Verify the display voice callroute [Huawei]display voice callroute

configura 1
tion of
Call route name :1
the call Call route select type : chargebase
route
plan.

NetEngine AR
Verify the display voice callprefix [Huawei]display voice callprefix

call prefix [ callprefix-name ]
Callprefix :1
configura Enterprise : default
tion. Dn-set : DefaultDialPlan
Prefix :1
Call category :
Call attribute :
Minimum length :4
Maximum length :8
Ring delay(s) :0
Call route name : 1
Callprefix :2
Prefix :1
Call category :
Call attribute :
Minimum length :4
Maximum length :8
Ring delay(s) :0
Call route name : 2
2.15.5.3 Configuring a Call Route Plan Based on Load Balancing

The PBX searches for routes in ascending order of trunk group numbers until it
finds a route with idle circuits.
Context
To select routes for outgoing calls based on load balancing, configure intelligent
routing based on load balancing. As shown in Figure 2-63, the PBX connects to
the carrier network through the R2 trunk, and the trunk group number is 1; the
PBX connects to the carrier network through the SIP trunk, the trunk group
number is 2, and the inter-office call prefix is 1.

Carrier PBX
trunkgroup 1 R2
trunkgroup 2 SIP
Prerequisites
Groups.

NetEngine AR

p

system
view.
2 Enter the voice -

voice view.
3 Create a callroute callroute-name -

call route
plan.
4 Configure selecttype call route select type value Set the route
intelligent selection type
routing. to loadshare.
5 Return to quit -
the voice
view.

trunk group
view.
7 Bind a call callroute callroute-name -

route to the
trunk group.
8 Return to quit -
the voice
view.
Configuring a Call Route for a Call Prefix

Step Action Command

enter its view.


NetEngine AR
Example
Configure a call route plan based on load balancing and set the inter-office call
prefix to 0. In the call route plan, there are routes with trunk groups 1 and 2.
[Huawei] voice
[Huawei-voice-callroute-1] selecttype loadshare
Verification
Verify display voice callroute The parameter values in

the the command output are
configur consistent with the
ation of settings.
the call
route
plan.
Verify display voice callprefix [ callprefix-

the call name ]
prefix
configur
ation.
2.15.5.4 Configuring a Call Route Plan Based on the Call Percentage

The PBX allocates calls to trunks bound to the same route according to the
configured percentage to flexibly control the load.
Context
To enable the PBX to forward outgoing calls based on the call percentage,
configure intelligent routing based on the call percentage. For example, 30%
outgoing calls are forwarded over an R2 trunk, and 70% outgoing calls are
forwarded over a SIP trunk. As shown in Figure 2-64, the PBX connects to the
carrier network through the R2 trunk and call route 1; the PBX connects to the
carrier network through the SIP trunk and call route 2, and the inter-office call
prefix is 1.

NetEngine AR

Carrier PBX
trunkgroup 1 R2
trunkgroup 2 SIP
Prerequisites
Groups.

p

system view.
2 Enter the voice -

voice view.

route plan.
4 Configure selecttype call route select type Set the route

intelligent value selection type to
routing. perloadshare.
5 Return to quit -
the voice
view.

trunk group
view.
7 Bind a call callroute callroute-name percent must be

route to the specified.
trunk group.
8 Return to quit -
the voice
view.

NetEngine AR

Step Action Command

enter its view.

Example
Configure the PBX to forward 30% calls through trunk group 1 and 70% calls
through trunk group 2, and set the inter-office call prefix to 1.
[Huawei] voice
[huawei-voice] callroute 1
[Huawei-voice-callroute-1] selecttype perloadshare
[Huawei-voice-trunkgroup-1] callroute 1 percent 30
[Huawei-voice-trunkgroup-2] callroute 1 percent 70
Verification
Verify display voice callroute [Huawei]display voice callroute

the 1
configur
Call route name :1
ation of Call route select type : perloadshare
the call
route
plan.

NetEngine AR
Verify display voice callprefix [Huawei]display voice callprefix

the call [ callprefix-name ]
Callprefix :1
prefix Enterprise : default
configur Dn-set : DefaultDialPlan
ation. Prefix :1
Call category :
Call attribute :
Minimum length : 4
Maximum length : 8
Ring delay(s) :0
Call route name : 1
Callprefix :2
Prefix :1
Call category :
Call attribute :
Minimum length : 4
Maximum length : 8
Ring delay(s) :0
Call route name : 2
2.15.5.5 Configuring a Call Route Plan Based on User Rights

The PBX assigns different routes for users with different rights. It preferentially
selects a route corresponding to a user with advanced right for outgoing calls
from the user. If route selection fails, the PBX selects a route corresponding to a
user with lower user right.
Context
When a call route plan based on user rights is configured, the PBX preferentially
selects a trunk group corresponding to a user right to forward outgoing calls. If
the trunk group has no idle circuit, the PBX selects a trunk group corresponding to
lower user right. If trunk groups correspond to the same user right, the PBX
searches for routes in ascending order of trunk group numbers. After trunk groups
with good performance are configured for advanced user right, users with
advanced user right can enjoy good call quality, while users with lower user rights
cannot use the trunk groups.
The user rights are as follows:
● Default right: default
● Common right: normal
● Advanced right: advanced
● Super right: super
User rights of default, normal, advanced, and super are in ascending order of
priority.

NetEngine AR
To forward outgoing calls based on user rights, configure intelligent routing based
on user rights. For example, outgoing calls from users with default right and
common right are forwarded over an R2 trunk and a SIP trunk respectively. As
shown in Figure 2-65, the PBX connects to the carrier network through the R2
trunk, and the trunk group number is 1; the PBX connects to the carrier network
through the SIP trunk, the trunk group number is 2, and the inter-office call prefix
is 1.

Carrier PBX
trunkgroup 1 R2
trunkgroup 2 SIP
Prerequisites
configured. For details about the configuration, see Configuring a Trunk
Group.

p

system
view.
2 Enter the voice -

voice view.

user view.
4 Configure priority { default |normal | -

the user advanced | super}
right.
5 Return to quit -
the voice
view.

call route
plan.

NetEngine AR

p
7 Configure selecttype selecttype-value Set the route

intelligent selection type to
routing. userpriorlevel.
8 Return to quit -
the voice
view.

trunk group
view.
10 Bind a call callroute callroute-name [ [ [ time Set the type of

route to the time-name | callertelno binding a call route
trunk group. callertelno-value ] *| charge to userpriorlevel.
charge-value | percent percent-
value | userpriorlevel
userpriorilevel-value ] | [ select-
level select-level-value ] ] *
11 Return to quit -
the voice
view.

Step Action Command

enter its view.

Example
Configure default right for user 7000 and common right for user 7001, configure
routes for trunk groups 1 and 2 based on user rights, and configure the PBX to
forward outgoing calls from users with default right through trunk group 1 and
outgoing calls from users with common right and higher rights through trunk
group 2.
[Huawei] voice
[Huawei-voice-pbxuser-7000] priority default

NetEngine AR
[Huawei-voice-pbxuser-7001] priority normal
[Huawei-voice-callroute-1] selecttype userpriorlevel
[Huawei-voice-trunkgroup-1] callroute 1 userpriorlevel default
[Huawei-voice-trunkgroup-2] callroute 1 userpriorlevel normal
Verification

the 1
configur
Call route name :1
ation of Call route select type : userpriorlevel
the call
route
plan.

the call [ callprefix-name ]
Callprefix :1
prefix Enterprise : default
configur Dn-set : DefaultDialPlan
ation. Prefix :1
Call category :
Call attribute :
Minimum length : 4
Maximum length : 8
Ring delay(s) :0
Call route name : 1
Callprefix :2
Prefix :1
Call category :
Call attribute :
Minimum length : 4
Maximum length : 8
Ring delay(s) :0
Call route name : 2

NetEngine AR
2.15.5.6 Configuring a Call Route Plan Based on Link Balancing

The PBX searches for routes in ascending order of trunk group numbers until it
finds a route with the maximum of idle circuits.
Context
To select routes for outgoing calls based on link balancing, configure intelligent
routing based on link balancing. As shown in Figure 2-66, the PBX connects to the
carrier network through the R2 trunk, and the trunk group number is 1; the PBX
connects to the carrier network through the SIP trunk, the trunk group number is
2, and the inter-office call prefix is 1.

Carrier PBX
trunkgroup 1 R2
trunkgroup 2 SIP
Prerequisites
Groups.


system
view.
2 Enter the voice -

voice view.

call route
plan.
4 Configure selecttype call route select type value Set the route
intelligent selection type
routing. to loadbalance.
5 Return to quit -
the voice
view.

trunk group
view.

NetEngine AR
7 Bind a call callroute callroute-name -

route to the
trunk group.
8 Return to quit -
the voice
view.

Step Action Command

enter its view.

Example
Configure a call route plan based on link balancing. Set trunkgroup to 1 and
outgoing prefix to 1.
[Huawei] voice
[Huawei-voice-callroute-1] selecttype loadbalance
Verification
Verify the configuration display voice callroute The parameter values in

of the call route plan. the command output are
consistent with the
Verify the call prefix display voice callprefix settings.
configuration. [ callprefix-name ]

NetEngine AR
2.15.5.7 Routing Base on Time Segment and Calling Number

This topic describes how to configure routing based on the time segment and
calling number.
Context
To meet the requirements of selecting different routes in different time segments
and for different calling numbers, you can configure intelligent routing based on
the time segment and calling number. For example, you can enable the system to
route the outgoing calls made between 08:00:00 to 18:00:00 by intra-office user
6000 through SIP1, route the outgoing calls made in other time segments by
intra-office user 6000 through SIP2, and route the outgoing calls made by intra-
office user 8000 through SIP2 preferentially. Figure 2-67 shows the network.
Figure 2-67 Routing network
Carrier A Carrier B
PBX
SIP1 SIP2 IMS
network
6000 8000
Prerequisites
Groups.
● A call prefix has been configured. For details about the configuration, see 2.9
Procedure
p

view.

view.

NetEngine AR

p
3 Configure a time time time-name {{ from -

index. from-date [ from-time ]
to to-date [ to-time ] } |
time-repeat { { yearly |
monthly } begin-date
[ end-time ] ] | weekly
begin-weekday [ begin-
time1 ] [end-weekday
[ end-time1 ] ] | daily
begin-time2 [ end-
time2 ] } }
route plan.
5 Configure selecttype call route Set the route selection

intelligent select type value type to callertimebase.
routing.

voice view.
7 Enter the trunk trunk-group trunk- -

group view. group-name
8 Bind a call route callroute callroute-name When binding a call route
to the trunk to the trunk group, you
group. must set either the time
or callertelno parameter,
or set neither parameters.

voice view.

Step Action Command

enter its view.
3 Configure routing information for callroute callroute-name

the call prefix.

NetEngine AR
Configure the system to route the outgoing calls made between 08:00:00 to
18:00:00 by intra-office user 6000 to outer-office user 1000 through SIP1, route
the outgoing calls made in other time segments by intra-office user 6000 to
outer-office users 1000 through SIP2, and route the outgoing calls made by intra-
office user 8000 to outer-office user 1000 through SIP2 preferentially. Enable
routing based on the time segment and calling number for call route 1.
[Huawei] voice
[Huawei-voice-callroute-1] selecttype callertimebase
[Huawei-voice-callroute-2] selecttype callertimebase
[Huawei-voice] trunk-group SIP1
[Huawei-voice-trunkgroup-SIP1] callroute 1 time 1 callertelno 6* select-level 0
[Huawei-voice-trunkgroup-SIP1] callroute 2 callertelno 8* select-level 1
[Huawei-voice-trunkgroup-SIP1] quit
[Huawei-voice] trunk-group SIP2
[Huawei-voice-trunkgroup-SIP2] callroute 1 time 2 callertelno 6*
[Huawei-voice-trunkgroup-SIP2] callroute 2 callertelno 8* select-level 0
[Huawei-voice-trunkgroup-SIP2] quit
Verification

the call Call route name :1
Call route select type : callertimebase
route Call route name :2
plan Call route select type : callertimebase
configur
ation.

NetEngine AR

the call [ callprefix-name ] Callprefix :1
Enterprise :
prefix default
configur Dn-set :
ation. DefaultDialPlan
Prefix :1
Call category : Basic
service
Call attribute :
1(Local)
Minimum length :
4
Maximum length :
32
Long cli switch :
disable
Caller map switch :
disable
Called map switch :
disable
Call route name :
1
Call route name : 2
2.15.6 Configuring Connection Data for the CDR Server

This section describes the connection data for the CDR server on the PBX.
Context
The CDR data generated for voice service is stored to the internal CDR pool on the
PBX. Because the CDR pool capacity is limited, connect the PBX to the CDR server.
Otherwise, CDRs are lost when the CDR pool is full or the PBX is powered off. The
data in CDR pool can be saved to the CDR server through the CDR interface, or
saved to the FTP/SFTP server in binary format or txt format. Figure 2-68 shows
the networking.
Figure 2-68 Connecting to the CDR server
IP
Network
PBX
CDR Server / Third-party

FTP Server / application
SFTP Server
CDR flows
POTS POTS
phone phone

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NOTICE
The PBX and CDR server must be deployed on a trusted network. Otherwise, there
will be security risks.
Visit https://support.huawei.com/enterprise and download the software package

eSpace ECS.
The PBX can connect to the following CDR server through either of the following
protocols:
● TCP: The PBX transfers the CDRs to the CDR server through TCP. The
corresponding CDR format is UCBILL.
● FTP/SFTP: The PBX transfers the CDRs to the FTP/SFTP server through FTP/
SFTP. The third-party systems or billing centers can obtain CDR information
from the FTP/SFTP server. The corresponding CDR format is CDR for CC08,
SOFTX for SOFTX3000, or MINI if only CDRs are required but billing process is
ignored.
NOTE
UCBILL, CDR, and SOFTX are the binary CDR format, and MINI is the txt CDR format.
For the call restriction policy used when the CDR pool is approximately full (90%),
see 2.15.4.5 Configuring Call Limiting When the CDR Pool Is Full.
Prerequisites
● The work mode of the device has been set to PBX. For how to configure the
work mode of device, see 2.5.1 Configuring a Device to Work in PBX Mode.
● The read, write, and overwrite permissions have been granted to the login
account of the PBX when FTP/SFTP is used to transfer CDRs.
Providing CDRs through TCP


view.

view.
3 Enter the CDR cdr-server -

server view.
4 Configure the IP server ip ip-address The default server port

address and port port port-number number is 21.
number of the CDR
server.

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5 (Optional) username username The user name and

Configure the user password cipher password must be
name and cipher-password consistent with those
password. on the CDR server.
NOTE
If no user name or
password is configured,
the system uses
anonymous user name
and password for login.
The anonymous user
name and password are
anonymous and
user@anonymous.com,
respectively.
6 Configure the CDR bill format format- For the TCP protocol,
format. value the CDR format is
UCBILL.
NOTE
● To modify the CDR
format, run the bill
format command in
the CDRServer view.
Do not use global
control point 53 to
modify the CDR
format.
● When the CDR
format is UCBILL, the
default server port
number is 2020.
7 (Optional) bill textfield index -

Configure the CDR
output information.

to the voice view.
9 (Optional) pbx number- Enumerated type. The

Configure the CDR parameter 85 value value is 0 or 1.
local storage ● 0: not supported
adaptation.
● 1: supported
10 Save the save -

configuration.

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Providing CDRs through FTP/SFTP


view.

view.
3 Enter the CDR cdr-server -

server view.
4 Configure transfer bill transfertype { FTP | The default transfer

mode. SFTP } mode is SFTP.
5 Configure the IP server ip ip-address The default server port

address and port port port-number number is 21.
number of the CDR
server.
6 Configure the user username username The user name and

name and password cipher password must be
password. cipher-password consistent with those
on the FTP/SFTP server.
NOTE
Enable the FTP/SFTP CDR
server to ensure that all
CDRs are saved.
7 Configure the CDR bill format format- When FTP/SFTP is used,

format. value the CDR formats
include:
● CDR
● SOFTX
● MINI
8 (Optional) bill textfield index -

Configure the CDR
output information.

to the voice view.
10 (Optional) pbx number- Enumerated type. The

Configure the CDR parameter 85 value value is 0 or 1.
local storage ● 0: not supported
adaptation.
● 1: supported
11 Save the save -

configuration.

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Example
When the PBX uses TCP to transfer CDRs to the CDR server, the connection data
for the CDR server is configured as follows: (Assume that the CDR server IP
address is 192.168.1.14, port number is 2020, and CDR format is UCBILL.)
[Huawei] voice
[Huawei-voice-cdr-server] server ip 192.168.1.14 port 2020
[Huawei-voice-cdr-server] bill format UCBILL
When the PBX uses FTP/SFTP to transfer CDRs to the CDR server, the connection
data for the CDR server is configured as follows: (Assume that the CDR server IP
address is 192.168.1.15, port number is 21, FTP user name and password are
admin and huawei123, and CDR format is CDR.)
[Huawei] voice
[Huawei-voice-cdr-server] bill transfertype FTP
[Huawei-voice-cdr-server] server ip 192.168.1.15 port 21
[Huawei-voice-cdr-server] bill format CDR
[Huawei-voice-cdr-server] username admin password cipher
Please input user password(1-15 chars):*********
Verification
Verify the display voice cdr-server The parameter values in the

connection command output are
configuration consistent with the settings.
between PBX
and CDR server.
2.15.7 Configuring PRA Trunk Timeslot Binding

This section describes how to bind trunk timeslots to a PRA trunk group to
implement call barring.
Context
In peak hours, all timeslots of a PRA trunk may be occupied by outgoing calls, and
incoming calls cannot be connected. To solve this problem, you can bind specified
trunk timeslots to a PRA trunk group and reserve the other timeslots for incoming
calls. Figure 2-69 shows the networking.

NetEngine AR
Figure 2-69 Networking for PRA trunk timeslot binding
PRA trunk
PBX IP PBX
Prerequisites
● A PRA trunk group has been configured. For details about the configuration,
see Configuring a Trunk Group.
● A call prefix has been configured. For details about the configuration, see
Procedure
Step Action Command
3 Enter the PRA trunk group view. trunk-group trunk-group-name
4 Bind trunk timeslots to the PRA trunk-pra slotid/subcardid/portid

trunk group. [ [ b-channels b-channels-value
[ num num-value ] ] [ config-
status config-status-value ] ]
Example
Assume that trunk group pra from the PBX to the IP PBX is configured on E1
interface 0 in slot 1. Bind E1/T1 timeslots 1 through 10 to the PRA trunk group.
[Huawei] voice
[Huawei-voice] trunk-group pra
[Huawei-voice-trunkgroup-pra] trunk-pra 1/0/0 b-channels 1 num 10
[huawei-voice-trunkgroup-pra] save

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Verification
Verify the PRA trunk display voice trunk-pra The parameter values in
group configuration. the command output are
consistent with the
settings.
2.15.8 Configuring Local Survival

In the PBX mode, The device can function as the local gateway of an enterprise
branch network to implement the local survival function.
Context
As the VoIP technology is rapidly developed and widely used, enterprise users
deploy IP phones in the remote branch and deploy the SIP server at the central
site to manage all calls of the remote branch. When communication between the
branch and the headquarters fails, the local voice gateway in the branch must
replace the headquarters SIP server at the central site to manage local voice
services. When communication between the branch and headquarters is restored,
the headquarters SIP server controls all calls.
In normal situations, the local gateway only provides network data switching and
trunk interface functions. SIP terminals located in the branch register with the
central node. POTS phones register with the central node through the local
gateway agent which enables call outgoing through trunks. Figure 2-70 shows the
call process.

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Figure 2-70 Local survival networking (normal situations)
Enterprise
Central headquarters
gateway
IP network
U1900
Enterprise
branch
Local
gateway
Registration
SIP terminal Call process

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If communication between the local gateway and center gateway fails (for
example, the network fails or center gateway fails), the local gateway
automatically takes charge of the voice service in the branch. After the SIP
terminal and analog phone of the branch registers with the local gateway, the
outgoing calls of the SIP terminal can be implemented by the local gateway.
Figure 2-71 shows the call process.
Figure 2-71 Local survival networking (communication between the local gateway
and center gateway fails)
Enterprise
Headquarters
U1900
Central Gateway
PSTN Fault
IP Phone
VPN
Enterprise
Branch
Local
Gateway
Registration
IP Phone Call process
NOTE
● On the local survival network, the SIP server must be configured on the SIP terminals
(IP phones or IADs). The SIP terminals (IP phones or IADs) register with the center
gateway and local gateway in the descending order of priority.
● On the local survival network, configure SIP AT0 trunks for POTS phones.

NetEngine AR
Prerequisites
● The local survival node has been configured.
● The license controlling the local survival function is loaded.
NOTICE
When the local node synchronizes data with the central node, the central node
checks a user according to the user number (telno) reported by the local node. If
the user is not registered on the central node, the central node deletes the user
according to the user name (pbxuser name).
The central node cannot delete a user whose user name and user number
configured on the local node are different. For example, if a user with user name
6000 and user number 5000 is configured on the local node, the central node
cannot delete this user. Therefore, it is recommended that you manually delete
such users.
Procedure
p

view.

view.
3 Enter the local local-survival -

survival view.
4 Configure the dataserver ip ip-address port Default port value:

server IP address port-number 8099
and port.
5 Set the server type. dataservertype type-value Enumerated type.

The values are as
follows:
● bmp
● u1900
Default value:
bmp
6 Configure the IP local-address ip ip-value port Default port value:

address and port port-value 8000
bound to the BIN
channel.

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p
7 Configure the password cipher password- The connection

authentication value data between
password for the center gateway
BIN channel. and local gateway
must be
configured.
8 Configure and sync-interval synchronization- The BIN channel

enable the interval-value sends
synchronization synchronization
interval of the BIN request to the
channel. center gateway at
intervals.
9 Configure the transfer transfer-value Enumerated type.

transfer mode for The values are as
the BIN channel. follows:
● TLS
● TCP
Default value: TLS
10 Configure the ssl-server-policy server-policy- This command is

server SSL policy to name valid only when
be bound to the the transfer mode
BIN channel. of the BIN channel
is set to TLS.
11 Configure the trunk primary-trunk-group trunk- Trunks configured

group between the group-name secondary-trunk- for central and
local gateway and group trunk-group-name local gateways in
center gateway. proxyreg-trunk-group trunk- this topic are SIP
group-name trunks. To delete
the trunk group,
run the undo
primary-trunk-
group command.
12 Reset the local reset -

survival
configuration.
13 Save the save -

configuration.
Example
Configure the SIP IP trunk group uc, and SIP AT0 trunk group sipat0 between the
local gateway and center gateway, set the IP address and port number of the
center gateway U1900 series gateways to 172.16.15.160 and 8099, set the IP

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address and port of the local gateway to 172.16.15.201 and 8000, set the
authentication password for the BIN channel to abcdefgh12345678 and transfer
mode to TLS, and bind the client SSL policy client to the BIN channel.
[Huawei] voice
[Huawei-voice] local-survival
[Huawei-voice-local-survival] dataserver ip 172.16.15.160 port 8099
[Huawei-voice-local-survival] dataservertype U1900
[Huawei-voice-local-survival] primary-trunk-group uc proxyreg-trunk-group sipat0
[Huawei-voice-local-survival] local-address ip 172.16.15.201 port 8000
[Huawei-voice-local-survival] password cipher
Please input user password(16-32 chars):****************
[Huawei-voice-local-survival] sync-interval 2
[Huawei-voice-local-survival] transfer tls
[Huawei-voice-local-survival] ssl-server-policy client
[Huawei-voice-local-survival] reset
[Huawei-voice-local-survival] save
Verification
Verify the configurations display voice local- The parameter values in

of local survival. survival status the command output are
consistent with the
settings.
Check the remaining display voice local- NOTE

available time of the survival lefttime The maximum remaining
time of local survival is 720
license for the best item.
hours, and it is updated
every 6 hours.
2.15.9 Setting SIP Stack Parameters

This section describes how to set SIP stack parameters.
Context
The Session Initiation Protocol (SIP) is a text-based signaling protocol. SIP
messages are classified into Request and Response messages. As an application
layer protocol, SIP establishes, modifies, or terminates multimedia sessions and
creates and controls multimedia sessions among two or more parties. SIP can
work with specified protocols to complete session setup and media negotiation,
such as Real-Time Transport Protocol (RTP), Real-Time Transport Control Protocol
(RTCP), Session Description Protocol (SDP), Real-time Stream Protocol (RTSP),
Domain Name System (DNS), and Transmission Control Protocol (TCP).
Prerequisites
The working mode of the device has been set to PBX. For details on how to
configure the working mode of the device, see 2.5.1 Configuring a Device to
Work in PBX Mode.

NetEngine AR
Procedure
Step Action Command
3 Enter the SIP user view. sip
4 Enable the session timer defined entity-based-sessions-timer enable

by RFC4028 based on the UA.
5 Set the minimum value of the min-se min-se–period

session timer.
6 Set the value of the unreliable session-progress session-progress–

provisional response period
retransmission timer.
7 Set the value of the reliable session-rel-progress session-rel-

provisional response progress–period
retransmission timer.
8 Set the value of the T1 timer. t1 t1–period
11 Set the value of the TD timer. td td–period
12 Enable header folding in the header-folding enable

authentication header field.
13 Set the value of the Max- field-header max-forwards max-

forwards header field. forwards-value
14 Set the value of the Organization field-header organization
header field. organization-head
15 Set the value of the Server field-header server server-head
header field.
16 Set the value of the User-agent field-header user-agent user-agent-

header field. head
Verification
Verify the of SIP stack display voice sip The parameter values in the
parameter settings. command output are
consistent with the settings.

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2.15.10 Setting DSP Parameters
Context
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.
Prerequisites
configure the working mode of device, see 2.5.1 Configuring a Device to Work in
PBX Mode.
Procedure
p

view.

view.
3 Enter the DSP view. dsp-attribute -
4 Enable echo echo enable By default, echo

cancellation on a cancellation is
DSP channel. disabled on the DSP
channel.
5 Enable the packet plc enable By default, the PLC

loss concealment function is disabled.
(PLC) function on a
DSP channel.
6 Enable silence silence enable By default, silence

compression on a compression is
DSP channel. disabled on the DSP
channel.

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p
7 Set fax parameters. ● Run: -

fax redundancy-t4
redundancy-t4-value
The T.4 redundancy
parameter value of the
T.38 fax is set.
By default, the T.4
redundancy parameter
value of the T.38 fax is
3.
● Run:
fax redundancy-t30
redundancy-t30–value
The T.30 redundancy
parameter value of the
T.38 fax is set.
By default, the T.30
redundancy parameter
value of the T.38 fax is
3.
● Run:
fax training-mode { e2e |
local }
The fax training mode
is set.
By default, end-to-end
training is used.
● Run:
fax training-rate { v17 | v29
| v27 }
The maximum fax
training rate is set.
By default, the
maximum fax training
rate is V17.
● Run:
fax v8negotiate enable
V8 negotiation is
enabled.
By default, V8
negotiation is disabled.
● Run:
fax ecm enable
T.38 ECM is enabled.
By default, T.38 ECM is
enabled.

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p
8 Set jitter buffer ● Run: -

parameters. jitter-buffer adapt-jb-
threshold adapt-jb-threshold
The alarm threshold of
the dynamic jitter
buffer is set.
By default, the alarm
threshold of the
dynamic jitter buffer of
the DSP channel is 150
ms.
● Run:
jitter-buffer init-adapt-jb
init-adapt-jb-value
The initial value of the
dynamic jitter buffer is
set.
By default, the initial
value of the dynamic
jitter buffer of the DSP
channel is 2 ms.
● Run:
jitter-buffer init-fixed-jb
normal-fixed-jb-value
The initial value of the
static jitter buffer is
set.
By default, the initial
value of the static jitter
buffer of the DSP
channel is 60 ms.
● Run:
jitter-buffer max-adapt-jb
max-adapt-jb-value
The maximum value of
the dynamic jitter
buffer is set.
By default, the
maximum value of the
ms.
● Run:
jitter-buffer min-adapt-jb
min-adapt-jb-value
The minimum value of
the dynamic jitter
buffer is set.

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p
By default, the
minimum value of the
ms.
● Run:
jitter-buffer max-fixed-jb
max-fixed-jb-value
The maximum value of
the static jitter buffer
is set.
By default, the
maximum value of the
static jitter buffer of
ms.
● Run:
jitter-buffer min-fixed-jb
min-fixed-jb-value
The minimum value of
the static jitter buffer
is set.
By default, the
minimum value of the
static jitter buffer of
ms.

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p
9 Set payload type ● Run: -

parameters. payload-type clear-mode
clearmode-value
The RTP payload type
value of a DSP channel
is set.
By default, the RTP
payload type value of
the DSP channel is
100.
● Run:
payload-type g726-16k
g726-16k-value
The G.726-16k payload
type value of a DSP
channel is set.
By default, the G.
726-16k payload type
value of the DSP
channel is 103.
● Run:
g726-24k-value
type value of a DSP
channel is set.
By default, the G.
value of the DSP
channel is 104.
● Run:
payload-type
g726-32k g726-32k-value
type value of a DSP
channel is set.
By default, the G.
value of the DSP
channel is 101.
● Run:
g726-40k-value
type value of a DSP
channel is set.
By default, the G.

NetEngine AR

p
value of the DSP

channel is 105.
● Run:
payload-type nte nte-value
The NTE payload type
is set.
By default, the NTE
the DSP channel is 97.
● Run:
payload-type redundancy
redundancy-value
The redundancy
a DSP channel is set.
By default, the
redundancy payload
type value of the DSP
channel is 96.
● Run:
payload-type
vbd vbd-value
The VBD payload type
is set.
By default, the VBD
the DSP channel is
102.

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p
10 Set RTCP ● Run: -

parameters. rtcp rtcp-interval rtcp-
interval
The interval at which a
DSP channel sends
RTCP packets is set.
By default, the interval
at which the DSP
channel sends RTCP
packets is 5000 ms.
● Run:
rtcp rtcpxr enable
The RTP Control
Protocol Extended
Reports (RTCP XR)
function is enabled.
By default, the RTCP
XR function is disabled.
● Run:
rtcp sev-degradethreshold
sev-degradethresholdval
The threshold for the
number of severe
degrade seconds is set.
By default, the
threshold for the
number of severe
degrade seconds is
400.
● Run:
rtcp vqm enable
The Voice Quality
Monitor (VQM)
function is enabled.
By default, the VQM
function is disabled.

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p
11 Set DSP resource ● Run: The resource threshold

parameters. resource-threshold mode in hierarchical control
{ priority | normal }
mode can be set only
The DSP resource
when the DSP
control mode is
resource control mode
configured.
is priority.
By default, the DSP
resource control mode
is normal.
● Run:
resource-threshold
{ threshold1 threshold1 |
threshold2 threshold2 |
threshold3 threshold3 }
The resource threshold
in hierarchical control
mode is set.
The default resource
threshold in
hierarchical control
mode is 100.
12 Set the mode in Run: nte2833-volume By default, the

which the DSP mode mode-value. The adjustment mode is
adjusts the volume mode in which the DSP disabled.
of DTMF tones. adjusts the volume of
DTMF tones is set.
13 Set a value by Run: nte2833-volume The default value is

which the DSP deviation deviation-value. 20, indicating that the
adjusts the volume A value that indicates DSP adjusts the
of DTMF tones in how much the DSP volume of DTMF tones
fixed mode. adjusts the volume of by 0 dBm.
DTMF tones in fixed
mode is set.

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p
14 Configure a DSP ● Run: -

channel. quit
The DSP attribute view
is closed.
● Run:
dsp slot/dsp-index
The DSP resource
management view is
displayed.
● Run:
loop-back loopback-type
channel
The loopback function
is configured on a DSP
channel.
By default, the
loopback function is
not configured on a
DSP channel.
● Run:
prohibit channel [ count ]
DSP channels are
prohibited.
By default, no DSP
channel is prohibited.
Verification
Verify the DSP display voice dsp- The parameter values in the
configuration. attribute command output are
consistent with the settings.
Check DSP statistics. display voice dsp
statistic
Check the DSP status. display voice dsp

state { slot/dsp-index
| channel slot/dsp-
index/channel }
2.15.11 Setting Parameters for Physical Voice Interfaces

NetEngine AR
Prerequisites
configure the working mode of the device, see 2.5.1 Configuring a Device to
Work in PBX Mode.
Setting Parameters for an FXS Interface


view.
3 Enter the FXS port fxs slotid/ By default, the region

interface view. subcardid/portid identifier is China
mainland.
4 Set physical attributes ● Run: -

for the FXS interface. reverse-pole-pulse
level level-value
The polarity
reversal pulse
level width is set.
● Run:
line-lock enable
Port locking is
enabled.
● Run:
nlp-mode { disable |
normal | ec-gain-
adjust | enhance |
reduce }
The non-linear
processing (NLP)
mode is set.

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5 Set electrical ● Run: -

attributes for the FXS electric current current-
value
interface.
The current is set.
● Run:
electric impedance
impedance-value
The impedance is
set.
● Run:
electric send-gain
sendgain-value
The send gain is
set.
● Run:
electric receive-gain
recvgain-value
The receive gain is
set.
Setting Parameters for an FXO Interface


view.
3 Enter the FXO port fxo slotid/ -

interface view subcardid/portid
4 Set the gain. gain { send If the RBT played for
sendgain-value | the caller through the
receive recvgain- FXO interface is loud,
value } the FXO interface
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.
5 Set the impedance. impedance { dc -

value | ac value }

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6 Set the ring current. ring { minimum- -

duration min-
duration-value |
maximum-interval
max-interval-value |
frequence
frequence-value | ac-
amplitude ac-
amplitude-value |
offset offset-value }
7 Set the feed. feed { maximum- -

voltage max-vol-
value | minimum-
voltage min-vol-
value | maximum-
current max-current-
value | minimum-
current min-current-
value }
Setting Parameters for a BRA Interface


view.

NetEngine AR
2 Configure the set workmode slot When the BRA

working mode of the slot-id bri bri-voice interface connects to
2BST interface card. { nt-mode | te- an ISDN phone, set
mode } the working mode of
the 2BST interface
card to nt-mode.
When the BRA
interface connects to
an ISDN network, set
the working mode of
the 2BST interface
card to te-mode.
NOTE
After this command is
executed, the
configured service is
invalid.
After you run this
command, the system
asks you whether to
reset the 2BST
interface card. If you
enter Y, the system
resets the 2BST
interface card to make
effect. If the system
does not reset the 2BST
interface card, run the
reset slot command to
reset the 2BST
interface card.
4 Enter the BRA port bra slotid/ -

6 Enable BRA interface l2-monitor enable -
Layer 2 monitoring.
7 Set the working work-mode { p2p | -

mode. p2mp }
8 Enable remote power remote-power -

supply. enable
9 Enabled the alarm alarm enable -

function.

NetEngine AR
Setting Parameters for a VE1 Interface


view.
2 Configure the E1 card set workmode slot After this command is

as the E1 voice card. slot-id e1t1 e1-voice executed, the
configured service is
invalid. The system
asks you whether to
reset the E1 voice
card. If you enter Y,
the system resets the
E1 voice card to make
effect. If the system
does not reset the E1
voice card, run the
reset slot command
to reset the E1 voice
card.

5 Set the signaling signal { ccs | cas } When the VE1
mode. interface is bound to
a PRA trunk,
configure common
channel signaling
(CCS) on the VE1
interface. When the
VE1 interface is
bound to an E1R2
trunk, configure
channel associated
signaling (CAS) on
the VE1 interface.
By default, a VE1
interface uses the CCS
mode.
6 Set the access digital access-mode -

section mode. { digital | direct }
7 Enable the CRC4 crc4 enable -

function.

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8 Set the CRC alarm crc-alarm-threshold -

threshold. { es es-threshold |
cses cses-threshold |
dm dm-threshold }
9 Enable VE1 interface l2-monitor enable -

Layer 2 monitoring.
10 Enable the PCM pcm-alarm -

alarm function.
11 Set the impedance of impedance -

the VE1 interface. impedance-value
Verification
Verify the FXS interface display voice port The parameter values in the
configuration. fxs [ slotid/subcardid/ command output are
portid ] consistent with the settings.
Verify the FXO interface display voice port

configuration. fxo [ slotid/
subcardid/portid ]
Verify the BRA interface display voice port
configuration. bra [ slotid/
subcardid/portid ]
Verify the VE1 interface display voice port
configuration. ve1 [ slotid/
subcardid/portid ]
2.15.12 Configuring a mapping of a built-in tone

This section describes how to set a built-in tone mapping.
Prerequisites
configure the working mode of device, see 2.5.1 Configuring a Device to Work in
PBX Mode.
NOTE
V300R019C13 and later versions support the command.

NetEngine AR
Procedure
p

view.

view.
3 Mapping a built-in tone-set id toneid-value1 -

tone of a system to { toneid toneid-value2 |
a built-in tone of file-name file-name }
another system or
an external tone
file.
Verification
Querying the Mapping display voice tone- The parameter values in the
Between Reason Codes set command output are
and Announcements. consistent with the settings.
2.16 Configuring the Signaling Proxy and Media Proxy

The PBX media proxy and signaling proxy implement address translation between
network segments and enable signaling and media streams to traverse NAT.
2.16.1 Context for Configuring the Signaling Proxy and Media

Proxy
By using an internal signaling media proxy module, the PBX can implement NAT
traversal and communicate with the devices on other network segments.
Overview of the Signaling Proxy and Media Proxy

Network address translation (NAT) technology was developed to alleviate IPv4
address exhaustion. Unlike traditional gateways that connect different network
segments, NAT devices can be regarded as special gateways that connect private
and public IP networks. When an IP packet passes through a NAT device, its
source IP address is translated into a routable public address. In addition, the NAT
device creates a binding covering the private source address, public source address,
and public destination address. In this way, the response packet from the public
network can be routed to the source on the private network.
Although NAT technology can mitigate IP address exhaustion, the technology
brings about the following problems: Most of the existing protocols are

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incompatible with NAT technology. IP addresses can be translated by NAT devices

at the network and transport layers but cannot be translated at the application
layer. As a result, IP addresses contained in the application-layer protocol are still
private addresses and response packets sent based on these IP addresses cannot
be routed to the source network elements. A NAT device binds the private source
address, public source address, and public destination address together only for IP
packets sent from a private network to a public network. Public network entities
cannot initiate connection requests to private network entities before the bindings
are created. NAT traversal includes four modes: static NAT, Simple Traversal of
User Datagram Protocol (UDP) Through Network Address Translators (NATs)
(STUN), ALG, and proxy. The signaling media proxy is the proxy mode.
The signaling media proxy is a proxy-based IP gateway. The signaling media proxy
supports signaling proxy and media proxy for VoIP services. It directionally
transmits all signaling and media streams, and re-specifies a destination address
and port for a signaling or media stream from a private or public address to help
achieve address translation between various network domains, including address
translation between private and public networks.
The signaling media proxy is a logical function entity and provides two functions:
SIP signaling proxy and media proxy.
● SIP signaling proxy: All registration and call messages of intra-office users are
sent to the signaling media proxy. The signaling media proxy processes
signaling messages and forwards them to a control node such as a trunk or a
SIP server. A call request from an inter-office user is sent to the signaling
media proxy first, and then the signaling media proxy processes the request
and forwards it to the called party. The signaling media proxy processes and
analyzes the signaling to obtain address change and bandwidth requirement
information about calls, and determines whether the media streams pass
through the signaling media proxy based on the network resource usage. This
helps to protect networks, prevent bandwidth theft, and achieve NAT
traversal.
● Media proxy: All RTP media streams pass through the signaling media proxy.
The signaling media proxy processes and forwards media streams to allow
communication between intra-office and inter-office users. The signaling
media proxy checks whether packets are valid and determines a forwarding
policy for the media streams based on the signaling processing results. The
forwarding policy covers packet filtering, QoS, and address translation. The
signaling media proxy specifies IP addresses and ports for intra-office and
inter-office users to receive RTP media streams to correctly forward the media
streams and ensure QoS and security.
SIP Signaling Proxy

To implement signaling NAT traversal, the INVITE request must reach the
destination user connected to the NAT device. In Figure 2-72, a private SIP UE
registered with the SIP server is used as an example.

NetEngine AR
Figure 2-72 Registration process

Contact Register Contact Register Contact
Aa Aa Dd
a c n c d e
A C N C Signaling D E
SIP user NAT media proxy SIP Server
IP:A IP:N IP:C IP:D IP:E

Port:a Port:n Port:c Port:d Port:e
Contact Contact Contact
Cc Cc Ee
c a 200 OK c n 200 OK e d
C A C N E D
1. A SIP UE sends a REGISTER request to the NAT device. The source IP address
contained in the REGISTER packet header and the contact address contained
in the payload are both the private address/port (Aa) of the SIP UE.
2. The NAT device allocates a public address/port (Nn) to the UE, generates a
mapping between Aa and Nn, translates Aa in the packet header into Nn, and
forwards the REGISTER request to the signaling media proxy.
3. The signaling media proxy receives the REGISTER request, allocates a public
signaling address/port (Dd), translates the address contained in the REGISTER
packet header and payload, records the mapping between Nn/Cc and Dd/Ee,
and sends the REGISTER request to the SIP server to which the SIP UE
belongs.
4. The SIP server authenticates the SIP UE and sends a response packet to the
signaling media proxy.
5. The signaling media proxy receives the response packet, modifies the address
contained in the packet header and payload according to the address
mapping, and forwards the response packet to the NAT device.
6. The NAT device translates the IP address contained in the response packet
into Aa and forwards the packet to the SIP UE.
Media Proxy
Media streams are transmitted over the VoIP network using RTP. RTP is carried
over UDP. The IP addresses and ports used for the RTP media streams are
negotiated using the signaling messages sent for establishing calls.
SIP uses the SDP information of the calling user and called user to negotiate the
media addresses and ports for the calling user and called user. When the signaling
carrying SDP information passes through the NAT device, the NAT device
translates only the IP, TCP, or UDP packet header, but not the IP address and port.
The media address obtained by a called user is the private address and port a
calling user. As a result, the called user cannot use the private address to access
the calling user on the private network. Deploying a media proxy on the network
is an effective way to implement media NAT traversal. The media proxy translates
private media addresses and ports into public addresses and ports during end-to-
end media negotiation.

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The signaling media proxy provides the media proxy function to support media
NAT traversal without the need to upgrade the existing NAT device on the
network. Media NAT traversal on the signaling media proxy is divided into two
stages: signaling negotiation and media transmission.
● Signaling negotiation stage, at which media address mappings are set up

by SDP negotiation
Figure 2-73 shows media address mapping setup through SDP negotiation.
Figure 2-73 Media address mapping setup through SDP negotiation
10.211.5.9
Signaling Called user
NAT media
Calling user Trunk IMS
proxy
network
192.168.1.2 192.168.1.1 | 10.211.2.3 10.211.3.8 | 10.10.3.5 10.10.5.7

INVITE: INVITE: INVITE:
Source IP:192.168.1.2 Source IP:10.211.2.3 Source IP:10.10.3.5
Dest IP:10.211.3.8 Dest IP:10.211.3.8 Dest IP:10.10.5.7
SDP:192.168.1.2/3008 SDP:192.168.1.2/3008 SDP:10.10.3.5/5007
10.211.5.9
Signaling Called user

Calling user NAT media Trunk IMS
proxy
network
192.168.1.2 192.168.1.1 | 10.211.2.3 10.211.3.8 | 10.10.3.510.10.5.7

INVITE: INVITE: INVITE:
SDP:10.211.3.8/7003 SDP:10.211.3.8/7003 SDP:10.211.5.9/9000
Before a calling user and a called user make a call, they must send signaling
packets to negotiate a channel for transmitting media streams. The signaling
media proxy obtains the calling user's and called user's IP addresses and ports
for receiving media streams according to SDP information contained in the
signaling packets, allocates the access-side and core-side media addresses and
ports to the calling user and called user, and creates an address mapping
entry (192.168.1.2:3008, 10.211.3.8:7003)<->(10.10.3.5:5007, 10.211.5.9:9000)
for media sessions. All media streams will pass through the signaling media
proxy, but only the media streams matching media session entries on the
signaling media proxy will be forwarded.
● Media transmission stage, at which the IP addresses for media packets
are learned and translated
The media transmission stage is divided into three sub stages: pre-media-
latching, media latching, and post-media-latching.

NetEngine AR
a. Pre-media-latching sub stage: As shown in Figure 2-74, because a

terminal at 192.168.1.2 does not send media packets to the signaling
media proxy, the media address mapping between the terminal and
signaling media proxy is not generated on the NAT device. As a result,
the NAT device discards all media packets destined for the terminal.
Figure 2-74 Pre-media-latching sub stage

Signaling
A NAT media proxy B
192.168.1.2 192.168.1.1 | 10.211.2.3 10.211.3.8 | 10.10.3.5 10.211.5.9
RTP: RTP:
Cannot reach
Source IP:10.211.3.8 Source IP:10.211.5.9
private IP address
S Port:7003 S Port:9000
Dest IP:192.168.1.2 Dest IP:10.10.3.5
D Port:3008 D Port:5007
b. Media latching sub stage: As shown in Figure 2-75, a terminal sends the
first media packet to the signaling media proxy. After the first media
packet passes through the NAT device, the NAT device creates an address
mapping between 192.168.1.2:3008 and 10.211.2.3:8028. The signaling
media proxy receives the media packet processed by the NAT device,
learns the transport-layer address and port (10.211.2.3:8028) contained in
the media packet, and updates the address mapping entry
(10.211.2.3:8028, 10.211.3.8:7003)<->(10.10.3.5:5007, 10.211.5.9:9000) for
media sessions.
Figure 2-75 Media latching sub stage

Signaling
A NAT media proxy B
192.168.1.2 192.168.1.1 | 10.211.2.3 10.211.3.8 | 10.10.3.5 10.211.5.9
RTP: RTP:
Source IP:192.168.1.2 Source IP:10.211.2.3
S Port:3008 S Port:8028
Dest IP:10.211.3.8 Dest IP:10.211.3.8
D Port:7003 D Port:7003
c. Post-media-latching sub stage: As shown in Figure 2-76, the signaling

media proxy queries the updated address mapping entry
(10.211.2.3:8028, 10.211.3.8:7003)<->(10.10.3.5:5007, 10.211.5.9:9000)
after it receives media packets destined for a terminal at 192.168.1.2 and
forwards the media packets to 10.211.2.3:8028. The NAT device queries
the address mapping entry (192.168.1.2:3008)<->(10.211.2.3:8028) and
forwards the media packets to the terminal.

NetEngine AR
Figure 2-76 Post-media-latching sub stage

Signaling
A NAT media proxy B
192.168.1.2 192.168.1.1 | 10.211.2.3 10.211.3.8 | 10.10.3.5 10.211.5.9
RTP: RTP: RTP:

S Port:7003 S Port:7003 S Port:9000
D Port:3008 D Port:8028 D Port:5007
2.16.2 Configuring the Signaling Proxy and Media Proxy on a

Public Network
When the PBX is deployed on a public network, configure signaling media proxy to
process and forward packets and media streams.
Context
The PBX is deployed on the public network and the SIP server of the PBX is
configured to provide registration services for private network users that register
with the PBX through NAT. Signaling media proxy implements voice
communication between SIP user C, POTS user B, and SIP user A. Figure 2-77
shows the networking.
Figure 2-77 Signaling media proxy on a public network
PBX Router
IP
IP:172.16.2.2
network
IP:10.10.10.2
A B C
Prerequisites

NetEngine AR
Configuring Interface Addresses and Address Pools


system view.

3 (Optional) Bind ip binding vpn-instance vpn- Before running the ip
a VPN instance instance-name binding vpn-instance
to an interface. command, create a
VPN instance.
When you run the ip
command on an
features on the
interface, such as the
IP address and
routing protocol, are
them.
By default, no VPN
instance is bound to
an interface.
4 Configure an ip address ip-address { mask If you configure a

Ethernet IP | mask-length } new IP address for an
address. interface that already
has an IP address, the
new IP address
overwrites the old
one.

system view.

view.

NetEngine AR
7 Add an IP voip-address signalling To specify the ip-

address to the interface interface-type address parameter,
signaling IP interface-number { ip-address ensure that an IP
address pool. | dynamic } address has been
configured for the
specify the dynamic
parameter, ensure
that an IP address has
been dynamically
allocated to the
specified interface.
8 Add an IP voip-address media It is recommended

address to the interface interface-type that the media and
media IP interface-number { ip-address signaling IP address
address pool. | dynamic } pools be the same.

instance. Before running the
voip-vpn-instance
command, ensure
that a VPN instance
has been created and
interfaces to which
the signaling IP
addresses and media
IP addresses belong
have been bound to
the instance.
By default, no VPN
instance is specified
to be used by voice
services.

system view.
Configuring the SIP Server and signaling media proxy

1 Enter the SIP sipserver -

server view.

NetEngine AR
2 Configure the signaling-address { ip ip- The signaling IP

signaling IP address | addr-name addr- address of the SIP
address and name-value } port port-value server must be
signaling port allocated from the
number of the signaling IP address
SIP server. pool on the
corresponding
interface.
3 Configure the media-ip { ip-address | addr- The media IP address

media IP name addr-name-value } of the SIP server must
address of the be allocated from the
SIP server. media IP address pool
on the corresponding
interface.
4 Configure the register-uri uri The Uniform Resource

URI for the SIP Identifier (URI) is
server. used for SIP user
registration.
5 Configure the home-domain domain The SIP server fills

home domain this parameter into
for the SIP the From field in SIP
server. messages.
6 Enable sbc signalling-proxy -

signaling proxy.
7 Enable media sbc media-proxy -

proxy.
8 Reset the SIP reset NOTE

server. Resetting the SIP server
requires a dozen of
seconds. Wait until the
system displays SIP
server reset succeeds.
before performing
other operations;
otherwise,
configurations will fail.
9 Save the save -

configuration.
Example
Step 1 Configure interface addresses and address pools.
Set the public network address of GE0 to 172.16.2.2 and add this address to the
media IP address pool and signaling IP address pool.

NetEngine AR

[Huawei] voice
[Huawei-voice] quit
Step 2 Configure the SIP server and signaling media proxy.

Configure the SIP server, and enable the signaling proxy and media proxy.
[Huawei-voice-sipserver] sbc signalling-proxy enable
[Huawei-voice-sipserver] sbc media-proxy enable
[Huawei-voice-sipserver]
SIP server reset succeeds.
[Huawei-voice-sipserver] save
----End
Verification
Verify the configuration display voice sipserver [Huawei]display voice sipserver

Config
of a SIP server. Signalling IP :
172.16.2.2
Signalling IP address
name :
Signalling domain :
DDNS client name :
Signalling port :
5060
Media IP :
172.16.2.2
Media IP address
name :
Register URI :
abcd.com
Home domain :
abcd.com
2.16.3 Configuring the Signaling Proxy and Media Proxy on

the Private/Public Network Boundary
This section describes how to configure signaling media proxy on the boundary
between the private and public networks to implement processing and directional
forwarding for call packets and media streams.
Context
The PBX is deployed at the edge of the public and private networks. Signaling
media proxy is configured on the private network and NAT is configured to
provide registration services for public network users. The PBX connects to PBX1
through the public network and PBX2 through the private network. The signaling

NetEngine AR
media proxy is configured to allow the voice service between SIP user G who is on
the public network, PBX, and users connected to the PBX1 and PBX2. Figure 2-78
shows the networking.
Figure 2-78 Networking of signaling media proxy deployed on the boundary

between private network and public network
G
PBX2 PBX PBX1
IP
network
IP:10.10.10.3 IP:172.16.2.3
NAT
A B C D E F
Prerequisites
● The user configurations are complete. For the configuration procedure, see 2.8
● The prefix configurations are complete. For the configuration procedure, see
2.9 Configuring a Call Prefix.
● The trunk group configurations are complete. For the configuration procedure,
see 2.11 Configuring Trunk Groups.
Configuring Interface Addresses and Address Pools


system view.


NetEngine AR
3 (Optional) Bind ip binding vpn-instance vpn- Before running the ip

a VPN instance instance-name binding vpn-instance
to an interface. command, create a
VPN instance.
When you run the ip
command on an
features on the
interface, such as the
IP address and
routing protocol, are
them.
By default, no VPN
instance is bound to
an interface.
4 Configure an ip address ip-address { mask If you configure a

Ethernet IP | mask-length } new IP address for an
address. interface that already
has an IP address, the
new IP address
overwrites the old
one.

system view.

view.
7 Add an IP voip-address signalling To specify the ip-

address to the interface interface-type address parameter,
signaling IP interface-number { ip-address ensure that an IP
address pool. | dynamic } address has been
configured for the
specify the dynamic
parameter, ensure
that an IP address has
been dynamically
allocated to the
specified interface.
8 Add an IP voip-address media It is recommended

address to the interface interface-type that the media and
media IP interface-number { ip-address signaling IP address
address pool. | dynamic } pools be the same.

NetEngine AR

instance. Before running the
voip-vpn-instance
command, ensure
that a VPN instance
has been created and
interfaces to which
the signaling IP
addresses and media
IP addresses belong
have been bound to
the instance.
By default, no VPN
instance is specified
to be used by voice
services.
10 Configure the sbc media-relay -

interface and
type for the
media relay.
Configuring NAT
Step Action Command
1 Enter the ACL view. acl number
2 Add an ACL rule. rule [ rule-id ] { deny | permit } [ source

{ source-address source-wildcard | any } ]
3 Return to the system quit

view.
4 Enter the interface interface interface-type interface-number

view.
5 Configure NAT to nat static protocol { imcp | tcp | udp }

translate private global { global-address | current-interface
network addresses to | interface } global-port inside host-address
public network [ host-port ] [ vpn-instance vpn-instance-
addresses. name ] [ netmask mask ] [ acl acl-
number ] [ description description ]
6 Return to the system quit
view.

NetEngine AR
Configuring the SIP Server and signaling media proxy

1 Enter the SIP sipserver -

server view.
2 Configure the signaling-address { ip ip- The signaling IP

signaling IP address | addr-name addr- address of the SIP
address and name-value } port port-value server must be
signaling port allocated from the
number of the signaling IP address
SIP server. pool on the
corresponding
interface.
3 Configure the media-ip { ip-address | addr- The media IP address

media IP name addr-name-value } of the SIP server must
address of the be allocated from the
SIP server. media IP address pool
on the corresponding
interface.
4 Configure the register-uri uri The Uniform Resource

URI for the SIP Identifier (URI) is
server. used for SIP user
registration.
5 Configure the home-domain domain The SIP server fills

home domain this parameter into
for the SIP the From field in SIP
server. messages.
6 Enable sbc signalling-proxy -

signaling proxy.

proxy.
8 Configures a sbc mapped-signalling- -

public IP address ip ip-address port
address and port-value
port number for
the signaling
mapping of the
SIP server.
9 Configure a sbc mapped-media-ip ip- -

media IP address
address for a
SIP server.

NetEngine AR
10 Reset the SIP reset NOTE

server. Resetting the SIP server
requires a dozen of
system displays SIP
before performing
other operations;
otherwise,

voice view.
Configuring the SIP IP Trunk (Connecting to PBX1)

1 Create a SIP IP trunk-group name sip no- -

trunk group register
and enter the
SIP IP trunk
group view.
2 Configure the signalling-address { ip ip- -

local signaling address | addr-name signal-
IP address and addr-name-value } port port-
port. value
3 Configure the media-ip { ip-address | addr- -
local media IP name addr-name-value }
address.
4 Configure the peer-address static primary- The remote IP address

remote IP ip-address primary-port-value and port must be
address and [ secondary secondary-ip- manually specified.
port. address secondary-port-
value ]
5 Configure the home-domain value -
home domain
of the remote
trunk group.
6 Configure the register-uri register-uri-value The URI must be the

URI for the same as home-
registration domain.
server.

proxy.

NetEngine AR
8 Reset the trunk. reset NOTE

Resetting the SIP server
requires a dozen of
system displays SIP
before performing
other operations;
otherwise,

voice view.
Configuring the SIP IP Trunk (Connecting to PBX2)
NOTICE
When the PBX functions as the aggregation node for the calls from PBX1 to PBX2,
enable the media proxy in the voice view by running the following command:
pbx number-parameter 104 0
1 Create a SIP IP trunk-group name sip no- -

trunk group register
and enter the
SIP IP trunk
group view.
2 Configure a call callroute callroute-name -

route.
3 Configure the signalling-address { ip ip- -

local signaling address | addr-name signal-
IP address and addr-name-value } port port-
port. value
4 Configure the media-ip { ip-address | addr- -
local media IP name addr-name-value }
address.
5 Configure the peer-address static primary- -

remote IP ip-address primary-port-value
address and [ secondary secondary-ip-
port. address secondary-port-
value ]

NetEngine AR
6 Configure the home-domain value -

home domain
of the remote
trunk group.
7 Configure the register-uri register-uri-value The URI must be the

URI for the same as home-
registration domain.
server.
8 Configure the sbc mapped-signalling- -

signaling IP address ip ip-address port
address and port-value
port number
for a SIP trunk.
9 Configure the sbc mapped-media-ip ip- -

media IP address
address for the
SIP trunk.
10 Reset the trunk. reset NOTE

Resetting the SIP server
requires a dozen of
system displays SIP
before performing
other operations;
otherwise,
11 Save the save -

configuration.
Example
Step 1 Configure interface addresses and address pools.
Set the public network address of GE0 to 172.16.2.2, private network address of
GE1 to 10.10.10.10, and add the two addresses to the media IP address pool and
signaling IP address pool.
[Huawei] voice

NetEngine AR
[Huawei-voice] sbc media-relay interface GigabitEthernet 0/0/0 external

[Huawei-voice] quit
Step 2 Configure NAT.

Allow the packets from GE0 with private network address 10.10.10.10 to traverse
NAT to reach GE1 on the public network.
[Huawei] acl 2000
[Huawei-acl-basic-2000] rule permit source 10.10.10.0 0.0.0.255
[Huawei-acl-basic-2000] quit
[Huawei] interface GigabitEthernet 0/0/1
[Huawei-GigabitEthernet0/0/1] nat static protocol udp global current-interface 5070 inside 10.10.10.10
5060 netmask 255.255.255.255
[Huawei-GigabitEthernet0/0/1] nat static protocol udp global current-interface 5071 inside 10.10.10.10
5061 netmask 255.255.255.255
Step 3 Configure the SIP server and signaling media proxy.

[Huawei-voice-sipserver] sbc signalling-proxy enable
[Huawei-voice-sipserver] sbc mapped-signalling-address ip 172.16.2.2 port 5070
[Huawei-voice-sipserver] sbc mapped-media ip 172.16.2.2
[Huawei-voice-sipserver]
SIP server reset succeeds.
[Huawei-voice-sipserver] quit
Step 4 Configure the SIP IP trunk (connecting to PBX1).

Configure the route with callroute 1 destined for PBX1. Assume that the IP address
and port number of PBX1 are 172.16.2.3 and 5061 respectively.
[Huawei-voice] trunk-group PBX1 sip no-register
[Huawei-voice-trunkgroup-PBX1] callroute 1
[Huawei-voice-trunkgroup-PBX1] signalling-address ip 172.16.2.2 port 5060
[Huawei-voice-trunkgroup-PBX1] media-ip 172.16.2.2
[Huawei-voice-trunkgroup-PBX1] peer-address static 172.16.2.3 5061
[Huawei-voice-trunkgroup-PBX1] home-domain abcd.com
[Huawei-voice-trunkgroup-PBX1] register-uri abcd.com
[Huawei-voice-trunkgroup-PBX1] sbc media-proxy enable
[Huawei-voice-trunkgroup-PBX1] reset
[Huawei-voice-trunkgroup-PBX1]
Note: Trunkgroup reset succeeds.
[Huawei-voice-trunkgroup-PBX1] quit
Step 5 Configure the SIP IP trunk (connecting to PBX2).

Configure the route with callroute 2 destined for PBX2. Assume that the IP address
and port number of PBX2 are 10.10.10.13 and 5061 respectively.
[Huawei-voice] trunk-group PBX2 sip no-register
[Huawei-voice-trunkgroup-PBX2] callroute 2
[Huawei-voice-trunkgroup-PBX2] signalling-address ip 10.10.10.10 port 5061
[Huawei-voice-trunkgroup-PBX2] media-ip 10.10.10.10
[Huawei-voice-trunkgroup-PBX2] peer-address static 10.10.10.13 5061
[Huawei-voice-trunkgroup-PBX2] home-domain abcd.com
[Huawei-voice-trunkgroup-PBX2] register-uri abcd.com
[Huawei-voice-trunkgroup-PBX2] sbc mapped-signalling-address ip 172.16.2.2 port 5071
[Huawei-voice-trunkgroup-PBX2] sbc mapped-media ip 172.16.2.2
[Huawei-voice-trunkgroup-PBX2] reset
[Huawei-voice-trunkgroup-PBX2]
[Huawei-voice-trunkgroup-PBX2] save
The current configuration will be written to the device.
Are you sure to continue? (y/n)[n]:y
----End

NetEngine AR
Verification
Verify the trunk group display voice trunk- The parameter values in
configuration. group [ name [ para- the command output are
value ] ] consistent with the
settings.
Verify the configuration display voice sipserver [Huawei]display voice sipserver

Config
of a SIP server. Signalling IP :
10.10.10.10
Signalling IP address
name :
Signalling domain :
DDNS client name :
Signalling port :
5060
Media IP :
10.10.10.10
Media IP address
name :
Register URI :
abcd.com
Home domain :
abcd.com
2.17 Configuring SIP PRA Agent Registration and SIP

AT0 Agent Registration
2.17.1 Configuring SIP PRA Agent Registration

When receiving a SIP PRA registration message from the IP PBX, the AR works as
the agent for the IP PBX and initiates SIP PRA registration to the IMS network.
Context
To implement NAT traversal at the edge between enterprise and carrier networks,
deployed the AR at the enterprise side to perform agent registration. The IP PBX
initiates a registration request to the AR which then works as the agent for the IP
PBX and initiates SIP PRA registration to the IMS network. When the registration
succeeds, PBX users within the enterprise can perform voice communication with
terminal users in the IMS network.
In group registration mode, the SIP PRA uses one registration message to register
a group of SIP user numbers. A registration group contains the primary number
and numbers to be registered in agent mode. Numbers to be registered in agent
mode can be consecutive or scattered. Consecutive numbers correspond to
wildcard characters (!.*!) in a registration group on the HSS. Scattered numbers
correspond to other numbers in a registration group on the HSS. The AR initiates a
registration message only for the primary number in a registration group. The
primary number can substitute all numbers in the registration group to perform
the registration.

NetEngine AR
Figure 2-79 shows the network diagram for SIP PRA agent registration.
Figure 2-79 SIP PRA agent registration
Enterprise
IP PBX
Ethernet cable
Telephone line
Analog Fax SIP soft IP Registration message
phone machine terminal phone
Procedure
Ste Operation Command Remarks
p
1 Configur Enter the system-view -

e system view.
interface
addresse Enter the interface interface-type -
s and interface view. interface-number
address
pools

NetEngine AR

p
(Optional) ip binding vpn-instance Before running

Bind a VPN vpn-instance-name the ip binding
instance to an vpn-instance
interface. command,
create a VPN
instance.
When you run
the ip binding
vpn-instance
command on
an interface,
all Layer 3
features on the
interface, such
as the IP
address and
routing
protocol, are
deleted. To use
these features,
re-configure
them.
By default, no
VPN instance is
bound to an
interface.
Configure an ip address ip-address If you

Ethernet IP { mask | mask-length } configure a
address. new IP address
for an interface
that already
has an IP
address, the
new IP address
overwrites the
old one.
Return to the quit -

system view.
Enter the voice -

voice view.

NetEngine AR

p
Add an IP voip-address signalling To specify the

address to the interface interface-type ip-address
signaling IP interface-number { ip- parameter,
address pool. address | dynamic } ensure that an
IP address has
been
configured for
the specified
interface. To
specify the
dynamic
parameter,
ensure that an
IP address has
been
dynamically
allocated to
the specified
interface.
Add an IP voip-address media It is

address to the interface interface-type recommended
media IP interface-number { ip- that the media
address pool. address | dynamic } and signaling
IP address
pools be the
same.

NetEngine AR

p
(Optional) voip-vpn-instance vpn- Only SIP trunk

Configure a instance-name groups can be
bound VPN bound to VPN
instance. instances.
Before running
the voip-vpn-
instance
command,
ensure that a
VPN instance
has been
created and
interfaces to
which the
signaling IP
addresses and
media IP
addresses
belong have
been bound to
the instance.
By default, no
VPN instance is
specified to be
used by voice
services.
Configure the sbc media-relay -

interface and
type for the
media relay.
2 Configur Create an enterprise enterprise-name -

e an enterprise and
enterpris enter the
e and a enterprise
DN set view
Configure the dn-set dn-set-name -

DN set [ description description ]
3 Configuri Enter the SIP sipserver -

ng the server view.
SIP
server

NetEngine AR

p
Configure the signalling-address { ip ip- The signaling

signaling IP address | addr-name addr- IP address of
address and name-value } port port- the SIP server
signaling port value must be
number of the allocated from
SIP server. the signaling IP
address pool
on the
corresponding
interface.
Configure a signalling-domain If the SIP

signaling signaling-domain-value server uses a
domain name dynamic
for the SIP signaling IP
server that address, you
uses a must configure
dynamic a signaling
signaling IP domain name
address. for the SIP
server.
Configure the media-ip { ip-address | The media IP

media IP addr-name addr-name- address of the
address of the value } SIP server must
SIP server. be allocated
from the
media IP
address pool
on the
corresponding
interface.
Configure the register-uri uri The Uniform

URI for the Resource
SIP server. Identifier (URI)
is used for SIP
user
registration.
Configure the home-domain domain The SIP server

home domain fills this
for the SIP parameter into
server. the From field
in SIP
messages.
Enable media sbc media-proxy enable -

proxy.

NetEngine AR

p
Reset the SIP reset -

server.
4 Configur Create a call callroute callroute-name -

e a call route
route
Configure a selecttype selecttype-value -
call route
policy
5 Configur Create and trunk-group name sip -

e a trunk enter the SIP trunk-group
group PRA trunk
group view
Bind a route callroute callroute-name -

mode
Configure a signalling-address { ip ip- -

local signaling address | addr-name
IP address and signal-addr-name-value }
port number port port-value
Configure a media-ip { ip-address | -

local media IP addr-name addr-name-
address or value }
dynamic
media IP
address name
Configure a peer-address static Configure in

peer IP primary-ip-address static mode.
address and primary-port-value Interconnectio
port number [ secondary secondary-ip- n data for a
(select a static address secondary-port- trunk group
or DNS mode value ] must be
based on consistent with
actual needs) the actual peer
data.
peer-address { dns-a Configure in

primary-domain-value DNS mode.
primary-port-value Interconnectio
[ secondary secondary- n data for a
domain-value secondary- trunk group
port-value ] | dns-srv must be
domain-value | dns-naptr consistent with
domain-value } the actual peer
data.

NetEngine AR

p
Configure the home-domain value Set this

home domain parameter
name of a based on peer
peer trunk requirements.
group
Configure the register-uri register-uri- Set this

URI of the value parameter
registration based on peer
server requirements.
Configure an register-id register-uri- To implement

identifier and value [ password cipher SIP PRA group
authentication password-value ] registration, an
password for identifier and
a registration authentication
group password must
be configured
for a
registration
group.
You can obtain
an identifier
and
authentication
password for a
registration
group from the
service
provider (such
as IMS
network
provider).
(Optional) enterprise enterprise-name Perform this

Configure the [ dn-set dn-set-name ] operation only
enterprise and when multiple
DN set for a enterprises and
trunk group DN sets are
configured.
Enable the sbc media-proxy enable -

media agent

NetEngine AR

p
Reset a trunk reset After a SIP

group trunk group is
created or its
parameters are
modified, reset
the SIP trunk
group for the
settings to take
effect.
6 Configur Create a SIP pbxuser name sipue -

e users user and enter [ enterprise enterprise-
the user view name ]
Configure the sipue eid-value It is
user device recommended
identifier that the device
identifier and
name of a user
be the same.
Configure the telno telno-value [ long- A long number

user telno long-telno-value ] is allocated by
telephone the carrier
number and network. When
long number a user's long
number is
configured,
extranet users
can directly
dial the long
number to
reach the user
without dialing
the
switchboard
number and
then the
extension
number.
Configure the eid-para password cipher Configure the

user password-value authentication
authentication password.
password Perform this
operation to
prevent
unauthorized
registration
and toll frauds.

NetEngine AR

p
7 Configur Create and groupreg groupreg-name -

e group enter the
registrati group
on registration
view
Configure an impi impi-value impu It is

identifier for impu-name [password recommended
group cipher] that IMPI and
registration IMPU values
be consistent
with those on
the IMS.
Add a SIP user pbxuser user-name [ step If multiple SIP

to the step-value ][number users are
registration number-value ] available, add
set all these SIP
users to the
registration
set.
Configure bind trunk-group trunk- -

agent group-name
registration
Configure the home-domain home- -

domain name domain-value
in group
registration
8 Configur Create and callprefix callprefix-name -

e enter the
incoming prefix
and template view
outgoing
prefixes Configure the prefix prefix -
prefix
Configure the call-type category -

prefix call callcategory attribute
attribute

NetEngine AR

p
Configure the digit-length minimum- The number

number length-value maximum- analysis length
analysis length-value also contains
length the prefix. For
example, if the
number
analysis length
is set to 5 and
the prefix is 88,
the analyzed
number is
88***.
The maximum
number
analysis length
must be
greater than or
equal to the
minimum
number
analysis length.
(Optional) callroute callroute-name Perform this

Bind a route operation only
mode when you need
to configure
the outgoing
prefix.
(Optional) enterprise enterprise-name Perform this

Configure the [ dn-set dn-set-name ] operation only
enterprise and when you need
DN set for the to configure
prefix the incoming
prefix.
9 Save the current settings save -
NOTE
The sipue users on the AR, IP PBX, and IMS must be the same.
The AR works as the agent for the IP PBX and initiates SIP PRA registration to the
IMS network. The enterprise DN set is proxy, trunk group name sippra-proxy,
group registration identifier 28988000, wildcard identifier 2898800!.*!, and bound
trunk group route pra-out. SIP user numbers allocated by the carrier range from

NetEngine AR
28988000 to 28988009. Intra-office users (28988000 to 28988009) call an outer-

office user through the sippra-proxy trunk.
NOTE
Here, !.*! are wildcard characters and can represent any digit from 0 to 9. 2898800!.*!
indicates the range from 28988000 to 28988009.
The SIP server IP address is 10.10.10.10. The local signaling and media IP address
is 172.16.2.2, and the signaling port number is 5062. The IP address for the IMS
network is 172.16.2.3, and the port number is 5060. The home domain name of
the peer trunk group and the URI of the registration server are abcd.com. The
incoming and outgoing prefixes are 10.
#Configure interface addresses and address pools.
[Huawei] voice
[Huawei-voice] quit
#Configure the DN set.

[Huawei-voice]enterprise default
[Huawei-voice-enterprise-default]dn-set proxy
[Huawei-voice-enterprise-default]quit
#Configure the SIP server.

#Configure a call route.

[Huawei-voice]callroute pra-out
[Huawei-voice-callroute-pra-out]selecttype callertimebase
[Huawei-voice-callroute-pra-out]quit
#Configure a trunk group.

[Huawei-voice]trunk-group sippra-proxy sip trunk-group
[Huawei-voice-trunkgroup-sippra-proxy]enterprise default dn-set proxy
[Huawei-voice-trunkgroup-sippra-proxy]callroute pra-out callertelno 10000x
[Huawei-voice-trunkgroup-sippra-proxy]signalling-address ip 172.16.2.2 port 5062
[Huawei-voice-trunkgroup-sippra-proxy]media-ip 172.16.2.2
[Huawei-voice-trunkgroup-sippra-proxy]peer-address static 172.16.2.3 5060
[Huawei-voice-trunkgroup-sippra-proxy]home-domain abcd.com
[Huawei-voice-trunkgroup-sippra-proxy]register-uri abcd.com
[Huawei-voice-trunkgroup-sippra-proxy]register-id 28988000 password cipher
[Huawei-voice-trunkgroup-sippra-proxy]sbc media-proxy enable
[Huawei-voice-trunkgroup-sippra-proxy]reset
Are you sure to continue?[Y/N]:y
[Huawei-voice-trunkgroup-sippra-proxy]quit
#Configure users. This example describes how to add two SIP users. You can add more SIP users by

NetEngine AR
referring to these operations.

[Huawei-voice]pbxuser sippra-28988000 sipue
[Huawei-voice-pbxuser-sippra-28988000]sipue 28988000
[Huawei-voice-pbxuser-sippra-28988000]telno 28988000
[Huawei-voice-pbxuser-sippra-28988000]eid-para password cipher
[Huawei-voice-pbxuser-sippra-28988000]pbxuser sippra-100001 sipue
[Huawei-voice-pbxuser-sippra-28988001]sipue 28988001
[Huawei-voice-pbxuser-sippra-28988001]telno 28988001
[Huawei-voice-pbxuser-sippra-28988001]eid-para password cipher
[Huawei-voice-pbxuser-sippra-100001]quit
#Configure group registration. Add 28988000 ~ 28988009 to the registration set.

[Huawei-voice]groupreg praproxy
[Huawei-voice-groupreg-praproxy]impi 28988000 impu 2898800!.*!
[Huawei-voice-groupreg-praproxy]pbxuser sippra-28988000 number 10
[Huawei-voice-groupreg-praproxy]bind trunk-group sippra-proxy
[Huawei-voice-groupreg-praproxy]quit
#Configure outgoing prefixes.

[Huawei-voice]callprefix pra-out
[Huawei-voice-callprefix-pra-out]prefix 10
[Huawei-voice-callprefix-pra-out]call-type category basic-service attribute 1
[Huawei-voice-callprefix-pra-out]digit-length 4 32
[Huawei-voice-callprefix-pra-out]callroute pra-out
[Huawei-voice-callprefix-pra-out]quit
#Configure incoming prefixes.

[Huawei-voice]callprefix pra-in
[Huawei-voice-callprefix-pra-in]enterprise default dn-set proxy
[Huawei-voice-callprefix-pra-in]prefix 10
[Huawei-voice-callprefix-pra-in]call-type category basic-service attribute 0
[Huawei-voice-callprefix-pra-in]digit-length 4 32
[Huawei-voice-callprefix-pra-in]quit
#Save the settings.

[Huawei-voice]save
Verification
Verify trunk group display voice trunk- Values in the command output
configuration group [ name [ para- are the specified values.
value ] ]
Verify user display voice pbxuser
configuration [ pbxuser-name ]
Verify group display voice

registration groupreg [ groupreg-
configuration name ]

NetEngine AR
Verify user display voice <Huawei> display voice groupreg

praproxy pbxuser brief
registration groupreg groupreg- Spec: 256, Records: 10
information name pbxuser brief
----------------------------------------------------
---------------------------
Index Telno Service-status SIPUE
Expires(s) Interval(s) Reg-status
0 28988000 Idle 10.10.10.3:5062
428 600 In service
1 28988001 Idle 10.10.10.3:5062
428 600 In service
Verify prefix display voice Values in the command output

configuration callprefix [ callprefix- are the specified values.
name ]
2.17.2 Configuring SIP AT0 Agent Registration

When receiving a SIP AT0 registration message from the IP PBX, the AR works as
the agent for the IP PBX and initiates SIP AT0 registration to the IMS network.
Context
To implement NAT traversal at the edge between enterprise and carrier networks,
deployed the AR at the enterprise side to perform agent registration. The IP PBX
initiates a registration request to the AR which then works as the agent for the IP
PBX and initiates SIP AT0 registration to the IMS network. When the registration
succeeds, PBX users within the enterprise can perform voice communication with
terminal users in the IMS network.
In user-by-user registration (SIP AT0 registration) mode, each SIP user under the IP
PBX must register with the AR which then registers each SIP user with the IMS.
Only one number can be registered at one time.
Figure 2-80 shows the network diagram for SIP AT0 agent registration.

NetEngine AR
Figure 2-80 SIP AT0 agent registration
Enterprise
IP PBX
Ethernet cable
Telephone line
Analog Fax SIP soft IP Registration message
phone machine terminal phone
Procedure
p
1 Configure Enter the system-view -

interface system view.
addresses
and Enter the interface interface-type -
address interface view. interface-number
pools

NetEngine AR

p
(Optional) ip binding vpn-instance Before running

Configure a vpn-instance-name the ip binding
bound VPN vpn-instance
instance. command,
create a VPN
instance.
When you run
the ip binding
vpn-instance
command on
an interface, all
Layer 3
features on the
interface, such
as the IP
address and
routing
protocol, are
deleted. To use
these features,
re-configure
them.
By default, no
VPN instance is
bound to an
interface.
Configure an ip address ip-address If you configure

Ethernet IP { mask | mask-length } a new IP
address. address for an
interface that
already has an
IP address, the
new IP address
overwrites the
old one.
Return to the quit -

system view.
Enter the voice -

voice view.

NetEngine AR

p
Add an IP voip-address signalling To specify the

address to the interface interface-type ip-address
signaling IP interface-number { ip- parameter,
address pool. address | dynamic } ensure that an
IP address has
been
configured for
the specified
interface. To
specify the
dynamic
parameter,
ensure that an
IP address has
been
dynamically
allocated to the
specified
interface.
Add an IP voip-address media It is

address to the interface interface-type recommended
media IP interface-number { ip- that the media
address pool. address | dynamic } and signaling
IP address
pools be the
same.

NetEngine AR

p
(Optional) voip-vpn-instance vpn- Only SIP trunk

Configure a instance-name groups can be
bound VPN bound to VPN
instance. instances.
Before running
the voip-vpn-
instance
command,
ensure that a
VPN instance
has been
created and
interfaces to
which the
signaling IP
addresses and
media IP
addresses
belong have
been bound to
the instance.
By default, no
VPN instance is
specified to be
used by voice
services.
Configure the sbc media-relay -

interface and
type for the
media relay.
2 Configure Create an enterprise enterprise- -

an enterprise and name
enterprise enter the
/DN set enterprise
view
Configure the dn-set dn-set-name -

DN set [ description description ]
3 Configuri Enter the SIP sipserver -

ng the SIP server view.
server

NetEngine AR

p
Configure the signalling-address { ip ip- The signaling IP

signaling IP address | addr-name address of the
address and addr-name-value } port SIP server must
signaling port port-value be allocated
number of from the
the SIP server. signaling IP
address pool on
the
corresponding
interface.
Configure a signalling-domain If the SIP server

signaling signaling-domain-value uses a dynamic
domain name signaling IP
for the SIP address, you
server that must configure
uses a a signaling
dynamic domain name
signaling IP for the SIP
address. server.
Configure the media-ip { ip-address | The media IP

media IP addr-name addr-name- address of the
address of the value } SIP server must
SIP server. be allocated
from the media
IP address pool
on the
corresponding
interface.
Configure the register-uri uri The Uniform

URI for the Resource
SIP server. Identifier (URI)
is used for SIP
user
registration.
Configure the home-domain domain The SIP server

home domain fills this
for the SIP parameter into
server. the From field
in SIP
messages.
Enable media sbc media-proxy -

proxy.
Reset the SIP reset -

server.

NetEngine AR

p
4 Configure Create a call callroute callroute-name -

a call route
route
Configure a selecttype selecttype- -
call route value
policy
5 Configure Create and trunk-group name sip -

a trunk enter the SIP trunk-group
group AT0 trunk
group view
(Optional) enterprise enterprise- Perform this

Configure the name [ dn-set dn-set- operation only
enterprise and name ] when multiple
DN set for a enterprises and
trunk group DN sets are
configured.
Bind a route callroute callroute-name -

mode
Configure a signalling-address { ip ip- -

local signaling address | addr-name
IP address signal-addr-name-value }
and port port port-value
number
Configure a media-ip { ip-address | -

local media IP addr-name addr-name-
address or value }
dynamic
media IP
address name
Configure a peer-address static Configure in

peer IP primary-ip-value primary- static mode.
address and port-value [ secondary Interconnection
port number secondary-ip-value data for a trunk
(select a secondary-port-value ] group must be
static or DNS consistent with
mode based the actual peer
on actual data.
needs)

NetEngine AR

p
peer-address { dns-a Configure in

primary-domain-value DNS mode.
primary-port-value Interconnection
[ secondary secondary- data for a trunk
domain-value secondary- group must be
port-value ] | dns-srv consistent with
domain-value | dns-naptr the actual peer
domain-value } data.
Configure the home-domain value Set this

home domain parameter
name of a based on peer
peer trunk requirements.
group
Configure the register-uri register-uri- Set this

URI of the value parameter
registration based on peer
server requirements.
Enable the sbc media-proxy enable -

media agent
Configure the trunk-sipat0 register-id By setting the

bound trunk { password cipher cipher- default-called-
of a trunk password | caller-telno telno
group [ enterprise enterprise- parameter, you
name ] telno-value | can specify a
default-called-telno specific user or
default-called-telno-value the automatic
| max-calls max-calls- switchboard to
value digest-username answer the
digest-username-value }* incoming call
from the trunk.
Reset a trunk reset After a SIP

group trunk group is
created or its
parameters are
modified, reset
the SIP trunk
group for the
settings to take
effect.

NetEngine AR

p
6 Configure Create a SIP pbxuser name sipue After a user is

users user and [ enterprise enterprise- created, you
enter the user name ] can run this
view command
(without
entering the
user type) to
enter the user
view.
Configure the sipue eid-value It is

user device recommended
identifier that the device
identifier and
name of a user
be the same.
Configure the telno telno-value [ long- A long number

user telno long-telno-value ] is allocated by
telephone the carrier
number and network. When
long number a user's long
number is
configured,
extranet users
can directly dial
the long
number to
reach the user
without dialing
the
switchboard
number and
then the
extension
number.
Configure the eid-para password cipher Configure the

user password-value authentication
authenticatio password.
n password Perform this
operation to
prevent
unauthorized
registration and
toll frauds.
Configure bind trunk-group trunk- -

agent group-name trunk-sipat0
registration register-id

NetEngine AR

p
7 Configure Create and callprefix callprefix-name -

incoming enter the
and prefix
outgoing template view
prefixes
Configure the prefix prefix -
prefix
Configure the call-type category -

prefix call callcategory attribute
attribute
Configure the digit-length minimum- The number

number length-value maximum- analysis length
analysis length-value also contains
length the prefix. For
example, if the
number
analysis length
is set to 5 and
the prefix is 88,
the analyzed
number is
88***.
The maximum
number
analysis length
must be
greater than or
equal to the
minimum
number
analysis length.
(Optional) callroute callroute-name Perform this

Bind a route operation only
mode when you need
to configure
the outgoing
prefix.
(Optional) enterprise enterprise- Perform this

Configure the name [ dn-set dn-set- operation only
enterprise and name ] when you need
DN set for the to configure
prefix the incoming
prefix.
8 Save the current settings save -

NetEngine AR
NOTE
The sipue users at the AR, IP PBX, and IMS must be the same.
The AR works as the agent for the IP PBX and initiates SIP AT0 registration to the
IMS network. The enterprise DN set is proxy, trunk group name sipat0-proxy, and
bound trunk group route at0-out. The SIP user number allocated by the carrier is
200000.
The SIP server IP address is 10.10.10.10. The local signaling and media IP address
is 192.168.10.2, and the signaling port number is 5080. The IP address for the IMS
network is 192.168.10.10, and the port number is 5060. The home domain name
of the peer trunk group and the URI of the registration server are abcd.com. The
incoming and outgoing prefixes are 10.
#Configure interface addresses and address pools.
<uawei> system-view
[Huawei] voice
[Huawei-voice] quit
#Configure the DN set.

[Huawei-voice]enterprise default
[Huawei-voice-enterprise-default]dn-set proxy
[Huawei-voice-enterprise-default]quit
#Configure the SIP server.

[Huawei-voice]sipserver
[Huawei-voice-sipserver]signalling-address ip 10.10.10.10 port 5060
[Huawei-voice-sipserver]media-ip 10.10.10.10
[Huawei-voice-sipserver]register-uri abcd.com
[Huawei-voice-sipserver]home-domain abcd.com
[Huawei-voice-sipserver]sbc media-proxy enable
[Huawei-voice-sipserver]reset
[Huawei-voice-sipserver]quit
#Configure a call route.

[Huawei-voice]callroute at0-out
[Huawei-voice-callroute-at0-out]selecttype callertimebase
[Huawei-voice-callroute-at0-out]quit
#Configure a trunk group.

[Huawei-voice]trunk-group sipat0-proxy sip trunk-circuit
[Huawei-voice-trunkgroup-sipat0-proxy]enterprise default dn-set proxy
[Huawei-voice-trunkgroup-sipat0-proxy]callroute at0-out callertelno 20000x
[Huawei-voice-trunkgroup-sipat0-proxy]signalling-address ip 192.168.10.2 port 5080
[Huawei-voice-trunkgroup-sipat0-proxy]media-ip 192.168.10.2
[Huawei-voice-trunkgroup-sipat0-proxy]peer-address static 192.168.10.10 5060
[Huawei-voice-trunkgroup-sipat0-proxy]register-uri abcd.com
[Huawei-voice-trunkgroup-sipat0-proxy]home-domain abcd.com
[Huawei-voice-trunkgroup-sipat0-proxy]sbc media-proxy enable
[Huawei-voice-trunkgroup-sipat0-proxy]trunk-sipat0 200000 default-called-telno 200000 caller-telno
200000
Huawei-voice-trunkgroup-sipat0-proxy]reset

NetEngine AR
Are you sure to continue?[Y/N]:y

[Huawei-voice-trunkgroup-sipat0-proxy]quit
#Configure users.
[Huawei-voice]pbxuser sipat0-200000 sipue
[Huawei-voice-pbxuser-sipat0-200000]sipue 200000
[Huawei-voice-pbxuser-sipat0-200000]telno 200000
[huawei-voice-pbxuser-sipat0-200000]eid-para password cipher
[Huawei-voice-pbxuser-sipat0-200000]bind trunk-group sipat0-proxy trunk-sipat0 200000
[Huawei-voice-pbxuser-sipat0-200000]quit
#Configure outgoing prefixes.

[Huawei-voice]callprefix at0-out
[Huawei-voice-callprefix-at0-out]prefix 20
[Huawei-voice-callprefix-at0-out]call-type category basic-service attribute 1
[Huawei-voice-callprefix-at0-out]digit-length 4 32
[Huawei-voice-callprefix-at0-out]callroute at0-out
[Huawei-voice-callprefix-at0-out]quit
#Configure incoming prefixes.

[Huawei-voice]callprefix at0-in
[Huawei-voice-callprefix-at0-in]enterprise default dn-set proxy
[Huawei-voice-callprefix-at0-in]prefix 20
[Huawei-voice-callprefix-at0-in]call-type category basic-service attribute 0
[Huawei-voice-callprefix-at0-in]digit-length 4 32
[Huawei-voice-callprefix-at0-in]quit
#Save the settings.

[Huawei-voice]save
Verification
Verify trunk group display voice trunk- Values in the command

configuration group [ name [ para- output are the specified
value ] ] values.
Verify user configuration display voice pbxuser Values in the command

[ pbxuser-name ] output are the specified
values.
Verify prefix display voice callprefix Values in the command

configuration [ callprefix-name ] output are the specified
values.
2.18 Follow-up Procedure for PBX

After completing all the configurations, provide user information, server IP
addresses, and documents to the customer.
Table 2-12 describes the follow-up procedure after you complete the
configurations.

NetEngine AR
Table 2-12 Follow-up procedure for PBX
Action Description
Record the Record the following items:

configured user ● Operating system users
names and
passwords and ● SIP users
submit them to
the customer.
Record IP Record IP addresses of PBX.

addresses of all
servers and
provide the IP
addresses to the
customer.
Provide backup Provide license files of PBX.

files to the
customer.
Provide required The product documentation includes:

product ● Product description
documentation
to the customer. ● Configuration guide
Tell the Visit https://support.huawei.com/enterprise/, choose Support

customer how to > Enterprise Networking > Router > AR, and select a
obtain product specific product to obtain the product documentation.
documentation.
2.19 Maintaining PBX
2.19.1 Checking Call Information
Context
During routine maintenance, you can run the related commands to check call
information.
Procedure
● Run the display voice online-info [ callee callee-number | caller caller-
number ] command to check information about all calls on a device.
● Run the display voice isdn active command to check information about ISDN
calls on a device.
----End

NetEngine AR
2.19.2 Checking Call Logs
Context
In routine maintenance, you can check or export call logs, and locate call failures
by analyzing the logs.
Procedure
Step 1 Run the display voice trace command to view the latest log information when a
call fails. The log information includes error and call logs related to the call failure.
Step 2 Run the display voice error command to view the latest call (including the
current call) failure logs.
Step 3 Run the save tracelog command to save the logs in the error log buffer and call
log buffer to the tracelog.zip file in the flash memory of a device.
----End
2.20 Configuration Examples for PBX
2.20.1 Example for Configuring Voice Services for Small- and

Medium-sized Enterprises
Networking Requirements
The PBX configures the external number allocated to an enterprise by the carrier
as the automatic switchboard number. An outer-office user dials the external
number and then dials an extension number as prompted to connect to an intra-
office user. Intra-office users make calls to each other by dialing short numbers.
Figure1 shows the voice service network.
This topic assumes that you want to implement the following requirements:
● The country code is 86, and the area code is 571.

● The internal numbers of users A, B, and C are 7000, 7001, and 7100
respectively.
● Users A, B, and C belong to enterprise hw. The DN set is local. The intra-
office call prefix is 7. The outgoing call prefix is 9. The incoming and outgoing
call rights of all users are all.
● Both the signaling IP address and media IP address are 192.168.1.2.
● The automatic switchboard name is ivr. A number allocated by the PSTN is
used as the automatic switchboard number, such as 28980808.
● Post-routing number change scheme 9 is configured to retain calling numbers
and delete the first digit of called numbers when PBX users make outgoing
calls through the FXO port.

NetEngine AR
Figure 2-81 Voice service network
AT0
Port 3/0/4
Port 3/0/0 PBX
Port 3/0/1

7000 7001 7100
Prerequisites
The IVR configuration has been completed. For details, see IVR. vu-service-name
configured for the automatic switchboard is service for the IVR.
Data Plan
The data plan provided in this example is for reference only. Plan data by
negotiating with users and the carrier.
Table 2-13 User number plan

POTS User SIP User Remarks
7000–7004 7100– PBX users

7104
Table 2-14 AT0 trunk plan

Slot ID/Subcard ID/Port Default Access Code Route ID
ID
3/0/4 28980808 0

NetEngine AR
Table 2-15 Prefix plan

Prefix Type Route
7 Intra-office N/A
9 Outgoing 0
Table 2-16 Post-routing number change plan

Prefix Outgoing Trunk Called Number Change
9 AT0 Deleting the first digit
Procedure
Step 1 Set the service mode to PBX.
[Huawei] voice
[Huawei-voice] service-mode pbx
[Huawei-voice] return
[Huawei] save
Are you sure to continue? (y/n)[n]:yIt 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]:yInfo: system is rebooting, please wait...
Step 2 Set the Ethernet IP address of interface GE0/0/0 to 192.168.1.2, and add
192.168.1.2 to the media IP address pool and signaling IP address pool of the
interface.
[Huawei] voice
Step 3 Configure the SIP server.

Step 4 Set the default country code to 86 and default area code to 571, and enable
country code change and area code change.
[Huawei-voice] pbx default-country-code 86 default-area-code 571
Step 5 Configure the enterprise and DN set for numbers.

[Huawei-voice-enterprise-hw] dn-set local
[Huawei-voice-enterprise-hw] quit

NetEngine AR
Step 6 Set the enterprise and DN set of prefixes to hw and local, and configure intra-
office call prefix 7 whose call attribute is 0 and local call prefix 9 whose call
attribute is 1. Configure national toll call prefix 90 whose call attribute is 2 and
international toll call prefix 900 whose call attribute is 3.
[Huawei-voice-callprefix-7] enterprise hw dn-set local
Step 7 Configure the automatic switchboard.
Set the automatic switchboard name to ivr and automatic switchboard number to
28980808.
[Huawei-voice] callprefix ivr
[Huawei-voice-callprefix-ivr] prefix 28980808
[Huawei-voice-callprefix-ivr] enterprise hw dn-set local
[Huawei-voice-callprefix-ivr] call-type category vu-service vu-service-name vudefault
[Huawei-voice-callprefix-ivr] digit-length 8 32
[Huawei-voice-callprefix-ivr] save
[Huawei-voice-callprefix-ivr] quit
Step 8 Configure a SIP user whose user number is 7100, authentication password is
a123456, and incoming and outgoing call rights are all.
[Huawei-voice] pbxuser 7100 sipue enterprise hw
[Huawei-voice-pbxuser-7100] dn-set local
[Huawei-voice-pbxuser-7100] call-right in all
[Huawei-voice-pbxuser-7100] call-right out all
Please input user password(6-64 chars): *******
Step 9 Configure POTS users whose user numbers are 7000 and 7001 and incoming and
outgoing call rights are all.
[Huawei-voice] pbxuser 7000 pots enterprise hw

NetEngine AR

Step 10 Configure an AT0 trunk group.

[Huawei-voice] trunk-group at0 fxo
[Huawei-voice-trunkgroup-at0] enterprise hw dn-set local
[Huawei-voice-trunkgroup-at0] trunk-at0 3/0/4 default-called-telno 28980808
[Huawei-voice-trunkgroup-at0] quit
Step 11 Configure a call route and post-routing number change.

[Huawei-voice-calldroute-9] quit
[Huawei-voice] trunk-group at0
[Huawei-voice-trunkgroup-at0] callroute 9
[Huawei-voice-trunkgroup-at0] quit
[Huawei-voice-afterroute-change-9] trunk-group at0
[Huawei-voice-afterroute-change-9] caller no-change
[Huawei-voice-afterroute-change-9] called del 1 1
Step 12 Verify the configuration.
Item Expected Result Possible Fault Cause
Intra-office call Calls can be made The intra-office call

properly, and the calling prefix is incorrectly
number is correctly configured.
displayed. For example,
user 7000 can dial 7100
to make a call to user
7100, and the calling
number displayed to
user 7100 is 7000.
Outgoing call Calls can be made ● The outgoing call

displayed. For example, ● The outgoing trunk is
user 7000 can make an incorrectly configured.
outgoing call through
the AT0 trunk, and the
calling number displayed
to the called party is
28980808.
Incoming call Calls can be made N/A

properly. For example, an
outer-office user can dial
28980808 and then dial
7000 as prompted to
connect to user 7000.

NetEngine AR
----End
Configuration Files
● Router configuration
#
interface GigabitEthernet0/0/0
ip address 192.168.1.2 255.255.255.0
#
voice
voip-address media interface GigabitEthernet 0/0/0 192.168.1.2
voip-address signalling interface GigabitEthernet 0/0/0 192.168.1.2
pbx default-area-code 571
#
callroute 9
#
enterprise hw
dn-set local
#
sipserver
signalling-address ip 192.168.1.2 port 5060
media-ip 192.168.1.2
register-uri abcd.com
home-domain abcd.com
#
trunk-group at0 fxo
enterprise hw dn-set local
trunk-at0 3/0/4 default-called-telno 28980808
callroute 9
#
callprefix 7
prefix 7
call-type category basic-service attribute 0
digit-length 3 32
#
callprefix 9
prefix 9
digit-length 1 32
callroute 9
#
callprefix 90
prefix 90
digit-length 2 32
#
callprefix 900
prefix 900
digit-length 3 32
#
callprefix ivr
prefix 28980808
call-type category vu-service vu-service-name vudefault
digit-length 8 32
#
pbxuser 7000 pots enterprise hw
telno 7000
dn-set local
port 3/0/0
call-right out all
#

NetEngine AR

telno 7001
dn-set local
port 3/0/0
call-right out all
#
pbxuser 7100 sipue enterprise hw
sipue 7100
telno 7100
dn-set local
call-right out all
eid-para password cipher %@%@nGE1Y)%q*~n14{5/1l2@,._1TrX7Eeq(Y>/,=AT'V"\~._4,%@%@
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 1 1
#
2.20.2 Example for Configuring Distributed Networking
To reduce toll call costs, an enterprise connects two branches in different cities
through a SIP trunk. Each branch connects to the IMS through a SIP AT0 trunk or
connects to the PSTN through a PRA trunk.
When an intra-office user in city A dials a PSTN number in city B, the call is routed
by the PBX to the IP PBX through the SIP IP trunk, routed by the IP PBX to the
PSTN through the PRA trunk, and finally connected to the outer-office user in city
B. When an intra-office user in city B dials a local number in city A, the call
process is similar, in which the call is first routed to the PBX.
This scenario reduces toll call costs. Figure 2-82 shows the distributed networking.
● The country code is 86, the area code of city A is 571, and the area code of
city B is 577.
● The IP address of the IMS is 192.168.1.4, and the port number is 5060.
● The automatic switchboard number of the PBX is 83787005, and the
automatic switchboard number of the IP PBX is 83786005.
● PBX users and IP PBX users make calls to each other by dialing short numbers.
● When a PBX user or an IP PBX user dials a local number in city B, the call is
routed through the IP PBX. If the calling user has a long number, the long
number is displayed as the calling number. If the calling user does not have a
long number, 83786005 is displayed as the calling number.
● When an IP PBX user or a PBX user dials a local number in city A, the call is
routed through the PBX. If the calling user has a long number, the long
number is displayed as the calling number. If the calling user does not have a
long number, 83787005 is displayed as the calling number.

NetEngine AR
Figure 2-82 Distributed networking
IMS
City A network
City B
PBX SIP AT0 PRA IP PBX

(192.168.1.2) (192.168.1.3)
SIP IP SIP IP
IP network
Port Port Port Port
User A 2/0/0 2/0/1 2/0/0 2/0/1 User D
Long: Long:
83787000 83786000
Short: Short:
7000 User B Fax1 Fax2 6000
User C
Short: Long: Long: Short:
7100 83787001 83786001 6100
Short: Short:
7001 6001
Prerequisites
The IVR configuration has been completed. For details, see IVR. The value of vu-
service-name configured for the automatic switchboard is the value of service for
the IVR.
Data Plan
Short Number Segment Long Number Segment Local Device
7000–7004 83787000–83787004 PBX
7100–7104 N/A
N/A 83786005 Automatic switchboard

of the PBX
6000–6004 83786000–83786004 IP PBX
6100–6104 N/A

of the IP PBX

NetEngine AR
Table 2-18 IP address plan
NE IP Address Subnet Mask
PBX 192.168.1.2 255.255.255.0
IP PBX 192.168.1.3 255.255.255.0
Table 2-19 PRA trunk plan
Slot ID/Subcard ID/Port ID Route Peer

Office
3/0/0 0 PSTN
Table 2-20 SIP IP trunk plan
Signaling IP Port Media IP Peer IP Peer Port Ro

Address Number Address Address Number ut
e
192.168.1.2 5062 192.168.1. 192.168.1.3 5062 1

2
Table 2-21 SIP AT0 trunk plan
Signaling Port Media IP Peer IP Peer Ro Regist Regist

IP Address Numbe Address Addres Port ut ration ration
r s Number e ID Passw
ord
192.168.1.2 5061 192.168.1. 192.168 5060 3 +8620 12345

2 .1.4 83787 678
005@
abcd.c
om
Prefix Type Route Calling Number Called Number Local

Change Change Device
7 Intra- N/A N/A N/A PBX

office
83787 Intra- N/A N/A N/A

office

NetEngine AR
Prefix Type Route Calling Number Called Number Local

Change Change Device
9 Outgoi 3 Changing the Deleting the first

ng number to digit
83786005
6 Outgoi 1 N/A N/A

ng
90577 Outgoi 1 N/A N/A

ng
6 Intra- N/A N/A N/A IP PBX

office
83786 Intra- N/A N/A N/A

office
9 Outgoi 0 Changing the Deleting the first

ng number to digit
83787005
7 Outgoi 2 N/A N/A

ng
90571 Outgoi 2 N/A N/A

ng

Prefix Outgoing Trunk Calling Number Called Number
Change Change
9 PRA 83786005 Deleting the first

digit
SIP AT0 83787005 Deleting the first

digit
Procedure
[Huawei] voice
[Huawei-voice] service-mode pbx
[Huawei-voice] return
[Huawei] save
<Huawei>reboot

NetEngine AR
Step 2 Set the Ethernet IP address of interface GE0/0/0 to 192.168.1.2, and add
192.168.1.2 to the media IP address pool and signaling IP address pool of the
interface.
[Huawei] voice

[Huawei-voice] pbx enable-country-area-transform enable

Step 6 Configure prefixes.

1. Configure intra-office call prefix 7 whose call attribute is 0.
The procedure for configuring outgoing call prefix 83787 is similar. You only
need to change the minimum number length to 8.
2. Configure prefix 9 whose call attribute is 1 and call route is 3.
Configure national toll call prefix 90 whose call attribute is 2 and call route is
0, and configure international toll call prefix 900 whose call attribute is 3 and
call route is 0. For details, see the configuration of prefix 9.
Configure outgoing call prefix 90577 whose call attribute is 1 and call route is
1. For details, see the configuration of prefix 6.

NetEngine AR
4. Configure the automatic switchboard.
Set the automatic switchboard name to ivr and automatic switchboard

number to 83786005.
[Huawei-voice-callprefix-ivr] enterprise hw dn-set hwdnset
Step 7 Configure user numbers.

1. Configure a SIP user whose user number is 7000, authentication password is
2. Configure a POTS user whose user number is 7100 and incoming and
Step 8 Configure trunk groups.

1. Configure a PRA trunk group.
[Huawei] voice
Set signal type successfully
[Huawei-voice] quit
[Huawei] clock source 0 3/0/0
[Huawei] voice
[Huawei-voice] trunk-group pra dss1-user
[Huawei-voice-trunkgroup-pra] trunk-pra 3/0/0
[Huawei-voice-trunkgroup-pra] callroute 0
[Huawei-voice-trunkgroup-pra] quit

NetEngine AR
2. Configure a SIP AT0 trunk group.

[Huawei-voice-trunkgroup-sipat0] trunk-sipat0 +862083787005@abcd.com default-called-telno
83787005 password cipher
Please input user password(1-32 chars):********
[Huawei-voice-trunkgroup-sipat0] callroute 3
[Huawei-voice-trunkgroup-sipat0] quit
3. Configure a SIP IP trunk group.

[Huawei-voice] trunk-group sipip01 sip no-register
[Huawei-voice-trunkgroup-sipip01] signalling-address ip 192.168.1.2 port 5062
[Huawei-voice-trunkgroup-sipip01] media-ip 192.168.1.2
[Huawei-voice-trunkgroup-sipip01] peer-address static 192.168.1.3 5062
[Huawei-voice-trunkgroup-sipip01] home-domain abcd.com
[Huawei-voice-trunkgroup-sipip01] register-uri abcd.com
[Huawei-voice-trunkgroup-sipip01] callroute 1
[Huawei-voice-trunkgroup-sipip01] reset
Step 9 Configure post-routing number change for the PBX.

The procedure for configuring post-routing number change for the IP PBX is
similar. You only need to change the trunk to PRA and calling number to
83786005.
[Huawei-voice] afterroute-change 9_6xxx_sipat0
[Huawei-voice-afterroute-change-9_6xxx_sipat0] callprefix 9
[Huawei-voice-afterroute-change-9_6xxx_sipat0] trunk-group sipat0
[Huawei-voice-afterroute-change-9_6xxx_sipat0] condition caller-telno 6xxx
[Huawei-voice-afterroute-change-9_6xxx_sipat0] caller del-then-insert 1 32 83786005
[Huawei-voice-afterroute-change-9_6xxx_sipat0] called del 1 1
[Huawei-voice-afterroute-change-9_6xxx_sipat0] quit
[Huawei-voice] afterroute-change 9_7xxx_sipat0
[Huawei-voice-afterroute-change-9_7xxx_sipat0] callprefix 9
[Huawei-voice-afterroute-change-9_7xxx_sipat0] trunk-group sipat0
[Huawei-voice-afterroute-change-9_7xxx_sipat0] condition caller-telno 7xxx
[Huawei-voice-afterroute-change-9_7xxx_sipat0] caller del-then-insert 1 32 83787005
[Huawei-voice-afterroute-change-9_7xxx_sipat0] called del 1 1
[Huawei-voice-afterroute-change-9_7xxx_sipat0] quit


number displayed to
user 7100 is 7000.

NetEngine AR
Outgoing call made by Calls can be made ● The outgoing call

an intra-office user with properly, and the calling prefix is incorrectly
a long number number is correctly configured.
displayed. For example: ● The outgoing trunk is
● User 7000 can dial incorrectly configured,
6000 to make a call or the reset
to user 6000, and the command is not
calling number executed after the
displayed to user configuration.
6000 is 7000.
● User 7000 can make
an outgoing call
through the SIP AT0
trunk, and the calling
number displayed to
the called party is
83787000.
an outgoing call
through the PRA
number displayed to
the called party is
83787000.
Outgoing call made by Calls can be made -

an intra-office user properly, and the calling
without a long number number is correctly
displayed. For example:
● User 7100 can dial
6000 to make a call
to user 6000, and the
calling number
displayed to user
6000 is 7100.
an outgoing call
through the SIP AT0
number displayed to
the called party is
83787005.
an outgoing call
through the PRA
number displayed to
the called party is
83786005.

NetEngine AR

7000 as prompted to
----End
Configuration Files
#
clock source 0 3/0/0 priority 9
#
set workmode slot 3 e1t1 e1-voice
#
ip address 192.168.1.2 255.255.255.0
#
voice
pbx enable-country-area-transform enable
#
port ve1 3/0/0
signal CCS
#
callroute 0
#
callroute 1
#
callroute 3
#
enterprise hw
dn-set hwdnset
#
sipserver
media-ip 192.168.1.2
#
trunk-group pra dss1-user
callroute 0
trunk-pra 3/0/0
#
trunk-group sipat0 sip trunk-circuit
callroute 3
media-ip 192.168.1.2
peer-address static 192.168.1.4 5060
number-parameter 19 1
trunk-sipat0 +862083787005@abcd.com password cipher %^%#sh1hK7Y[vIDIo]@
%y)"(^`xyQQLvuFT&:]Fob_b5%^%#
#
trunk-group sipip01 sip no-register
callroute 1

NetEngine AR
media-ip 192.168.1.2
#
callprefix 6
prefix 6
digit-length 4 32
callroute 1
#
callprefix 7
prefix 7
digit-length 4 32
#
callprefix 9
prefix 9
digit-length 1 32
callroute 3
#
callprefix ivr
prefix 83786005
digit-length 8 32
#
pbxuser 7000 sipue
sipue 7000
telno 7000 long-telno 83787000
call-right out all
eid-para password cipher %^%#%')'%i~C[2>B0.~$l6E@D)H|+:L0I!`Dg@,2>qjJ%^%#
#
pbxuser 7100 pots
port 2/0/0
call-right out all
#
afterroute-change 9_6xxx_sipat0
callprefix 9
trunk-group sipat0
condition caller-telno 6xxx
caller del-then-Insert 1 32 83786005
called del 1 1
#
afterroute-change 9_7xxx_sipat0
callprefix 9
trunk-group sipat0
called del 1 1
#
2.20.3 Example for Expanding the Capacity of the Live-

Network PBX
Connect the live-network PBX to a new PBX to meet the requirements of
expanding the capacity based on the live-network PBX and retaining the original
long and short user numbers, automatic switchboard number, and users' dialing
habit. Figure 2-83 shows the typical network for connecting the live-network PBX
to the PBX.

NetEngine AR
● The enterprise is hw, and the DN set is hwdnset.
● PBX users and live-network PBX users make calls to each other by dialing
short numbers.
● The PBX connects to the live-network PBX through an H.323 trunk, and the
call route is 2.
● The live-network PBX connects to the PBX through an H.323 trunk, and the
call route is 3.
● The PBX connects to carrier A through an AT0 trunk, and connects to carrier B
through a PRA trunk. The outgoing call routing mode based on load
balancing is used, and both trunks are bound to call route 0.
● Outgoing calls to the PSTN of carrier B are routed through a PRA trunk. If the
calling user has a long number, the long number is displayed as the calling
number. If the calling user does not have a long number, 28980808 is
displayed as the calling number.
● Outgoing calls to the PSTN of carrier A are routed through an AT0 trunk. If
the calling user has a long number, the long number is displayed as the
calling number. If the calling user does not have a long number, 83780808 is
displayed as the calling number.
● Users of carrier A or B dial the automatic switchboard of the PBX to make
incoming calls.
Figure 2-83 Typical network for connecting the live-network PBX to the PBX
PBX
Carrier A Carrier B
(192.168.1.2)
AT0 PRA
H.323
Live-network PBX
Fax1 User A User B User C User D Fax2

Long: Short: Long: Long: Short: Long:
83786001 6100 83786000 28987000 7100 28987001
Short: Short: Short: Short:
6001 6000 7000 7001

NetEngine AR
Prerequisites
The IVR configuration has been completed. For details, see IVR. The value of vu-
service-name configured for the automatic switchboard is the value of service for
the IVR.
Data Plan

Short Number Segment Long Number Segment Remarks
7000–7004 28987000–28987004 New PBX users
7100–7104 N/A
- 28980808 Automatic switchboard

of the new PBX
6000–6004 83786000–83786004 Live-network PBX users
6100–6104 N/A

of the live-network PBX
Table 2-25 IP address plan

NE IP Address Subnet Mask
New PBX 192.168.1.2 255.255.255.0
Live-network PBX 192.168.1.3 255.255.255.0
Table 2-26 PRA trunk plan

Slot ID/Subcard ID/Port ID Route Peer
Office
1/0/0 0 PSTN
Table 2-27 AT0 trunk plan

Slot ID/Subcard ID/Port Default Access Code Route ID
ID
3/0/4 83780808 0

NetEngine AR
Table 2-28 H.323 trunk plan

Media Port Number Peer Peer Route
IP IP Port ID
Addre Addre Numb
ss ss er
192.16 1720 192.16 1720 2

8.1.2 8.1.3

Prefix Type Route Local Device
6 Outgoing 2 New PBX
7 Intra-office N/A
2898 Intra-office N/A
8378 Outgoing 2
6 Intra-office N/A Live-network PBX
7 Outgoing 3
9 Outgoing 3
8378 Intra-office N/A

Prefix Outgoing Trunk Calling Number Called Number
Change Change
9 PRA 28980808 Deleting the first

digit
AT0 N/A Deleting the first

digit
Procedure
[Huawei] voice
[huawei-voice] service-mode pbx
[huawei-voice] return
[Huawei] save

NetEngine AR
<Huawei>reboot
Step 2 Set the Ethernet IP address of interface 0/0/0 to 192.168.1.2, and add 192.168.1.2
to the media IP address pool and signaling IP address pool of the interface.
[Huawei] voice



1. Configure intra-office call prefix 7 whose call attribute is 0, and set the
enterprise and DN set to hw and hwdnset.
NOTE
The procedure for configuring intra-office call prefix 2898 is similar. You only need to
change the minimum number length to 8.
[Huawei-voice-callprefix-7] enterprise hw dn-set hwdnset
NOTE
The procedure for configuring outgoing call prefix 8378 is similar. You only need to
change the minimum number length to 8.

NetEngine AR

NOTE
Configure national toll call prefix 90 whose call attribute is 2 and call route is 0, and
configure international toll call prefix 900 whose call attribute is 3 and call route is 0.
For details, see the configuration of prefix 9.
4. Configure the automatic switchboard.
Set the automatic switchboard name to ivr and automatic switchboard
number to 28980808.
[Huawei-voice-callprefix-ivr] enterprise hw dn-set hwdnset

1. Configure a SIP user whose user number is 7000, authentication password is
[Huawei-voice-pbxuser-7000] dn-set hwdnset
2. Configure a POTS user whose user number is 7100 and incoming and
[Huawei-voice-pbxuser-7100] dn-set hwdnset
[Huawei-voice-pbxuser-7100] return

1. Configure a PRA trunk group.
[Huawei] voice

NetEngine AR
[Huawei-voice] quit
[Huawei] voice
[Huawei-voice-trunkgroup-pra] enterprise hw dn-set hwdnset
2. Configure an AT0 trunk group.

[Huawei-voice] trunk-group AT0 fxo
[Huawei-voice-trunkgroup-AT0] trunk-AT0 3/0/4 default-called-telno 83780808
[Huawei-voice-trunkgroup-AT0] enterprise hw dn-set hwdnset
[Huawei-voice-trunkgroup-AT0] quit
3. Configure an H.323 trunk group.

Note: Mandatory parameter of h323 system completed, please reset h323 system.
H323 system parameters reset successfully!
[Huawei-voice] trunk-group h323 h323 symmetrical
[Huawei-voice-trunkgroup-h323] media-ip 192.168.1.2
[Huawei-voice-trunkgroup-h323] peer-address static 192.168.1.3 1720
Note: Mandatory parameter of the trunkgroup completed, please reset the trunkgroup.
[Huawei-voice-trunkgroup-h323] enterprise hw dn-set hwdnset
[Huawei-voice-trunkgroup-h323] callroute 2
[Huawei-voice-trunkgroup-h323] reset
[Huawei-voice-trunkgroup-h323] save
[Huawei-voice-trunkgroup-h323] quit
Step 9 Configure intelligent routing based on load balancing.

[Huawei-voice-callroute-0] selecttype loadshare
[Huawei-voice] trunk-group pra
[Huawei-voice] trunk-group AT0
[Huawei-voice-trunkgroup-AT0] callroute 0
[Huawei-voice-trunkgroup-AT0] quit
Step 10 Configure post-routing number change.

[Huawei-voice] afterroute-change 9_6xxx_pra
[Huawei-voice-afterroute-change-9_6xxx_pra] callprefix 9
[Huawei-voice-afterroute-change-9_6xxx_pra] trunk-group pra
[Huawei-voice-afterroute-change-9_6xxx_pra] condition caller-telno 6xxx
[Huawei-voice-afterroute-change-9_6xxx_pra] caller del-then-insert 1 32 28980808
[Huawei-voice-afterroute-change-9_6xxx_pra] called del 1 1
[Huawei-voice-afterroute-change-9_6xxx_pra] quit
[Huawei-voice] afterroute-change 9_71xx_pra
[Huawei-voice-afterroute-change-9_7xxx_pra] callprefix 9
[Huawei-voice-afterroute-change-9_7xxx_pra] trunk-group pra
[Huawei-voice-afterroute-change-9_7xxx_pra] condition caller-telno 71xx
[Huawei-voice-afterroute-change-9_7xxx_pra] caller del-then-insert 1 32 28980808
[Huawei-voice-afterroute-change-9_7xxx_pra] called del 1 1
[Huawei-voice-afterroute-change-9_7xxx_pra] quit
[Huawei-voice] afterroute-change 9_at0
[Huawei-voice-afterroute-change-9_at0] callprefix 9
[Huawei-voice-afterroute-change-9_at0] trunk-group AT0
[Huawei-voice-afterroute-change-9_at0] caller no-change
[Huawei-voice-afterroute-change-9_at0] called del 1 1
[Huawei-voice-afterroute-change-9_at0] quit
[Huawei-voice] afterroute-change 9_pra
[Huawei-voice-afterroute-change-9_at0] callprefix 9
[Huawei-voice-afterroute-change-9_at0] trunk-group pra

NetEngine AR
[Huawei-voice-afterroute-change-9_at0] caller no-change

[Huawei-voice-afterroute-change-9_at0] called del 1 1
[Huawei-voice-afterroute-change-9_at0] quit


number displayed to
user 7100 is 7000.
Outgoing call made by Calls can be made ● The outgoing call

an intra-office user with properly, and the calling prefix is incorrectly
a long number number is correctly configured.
displayed. For example: ● The outgoing trunk is
● User 7000 can dial incorrectly configured,
6000 to make a call or the reset
to user 6000, and the command is not
calling number executed after the
displayed to user configuration.
6000 is 7000.
an outgoing call
through the PRA
number displayed to
the called party is
28987000.
● For example, user
7000 can make an
the AT0 trunk, and
the calling number
displayed to the
called party is
28987000.

NetEngine AR
Outgoing call made by Calls can be made N/A

an intra-office user properly, and the calling
without a long number number is correctly
displayed. For example:
● User 7100 can dial
6000 to make a call
to user 6000, and the
calling number
displayed to user
6000 is 7100.
an outgoing call
through the PRA
number displayed to
the called party is
28980808.
● For example, user
7100 can make an
the AT0 trunk, and
the calling number
displayed to the
called party is
83780808.

7000 as prompted to
----End
Configuration Files
#
#
#
ip address 192.168.1.2 255.255.255.0
#
voice
#
port ve1 1/0/0

NetEngine AR
signal CCS
#
h323-attribute
localip 192.168.1.2
#
callroute 0
selecttype loadshare
#
callroute 2
#
enterprise hw
dn-set hwdnset
#
sipserver
media-ip 192.168.1.2
#
trunk-group at0 fxo
trunk-AT0 3/0/4 default-called-telno 83780808
enterprise hw dn-set hwdnset
callroute 0
#
trunk-group h323 h323 symmetrical
callroute 2
media-ip 192.168.1.2
#
trunk-pra 1/0/0
callroute 0
#
callprefix 6
prefix 6
digit-length 4 32
callroute 2
#
callprefix 7
prefix 7
digit-length 4 32
#
callprefix 9
prefix 9
digit-length 1 32
callroute 0
#
callprefix ivr
prefix 28980808
digit-length 8 32
#
sipue 7000
dn-set hwdnset
call-right out all
eid-para password cipher %^%#"(sq-~Wu6YD^RCIcKx:'6]z--N|iKU6DyrM4m&*X%^%#
#

NetEngine AR
telno 7100
port 3/0/0
dn-set hwdnset
call-right out all
#
afterroute-change 9_6xxx_pra
callprefix 9
trunk-group pra
called del 1 1
#
afterroute-change 9_71xx_pra
callprefix 9
trunk-group pra
condition caller-telno 71xx
called del 1 1
#
afterroute-change 9_at0
callprefix 9
trunk-group AT0
caller no-change
called del 1 1
#
afterroute-change 9_pra
callprefix 9
trunk-group pra
caller no-change
called del 1 1
#
2.20.4 Example for Configuring PBX Sharing for Different

Enterprises
Users A and B belong to enterprise 1. Users C and D belong to enterprise 2.
Enterprises 1 and 2 are in the same industrial campus. By configuring different
enterprises on the device, you can logically isolate multiple enterprises' voice
services, implementing PBX sharing. Enterprises 1 and 2 can use virtual PBXs to
implement voice services for intra-office users and use a unified egress to
implement voice services between intra-office and outer-office users. This reduces
enterprise costs as well as the carrier's access points. Figure 2-84 shows the PBX
sharing network.
● The external number allocated by the carrier to enterprise 1 is 56623000.
When an outer-office user dials 56623000, user B's phone rings. If the outer-
office user wants to make a call to an intra-office user other than user B, the
call can be transferred by user B to the target user.
● The external number allocated by the carrier to enterprise 2 is 56623001.
When an outer-office user dials 56623001, user C's phone rings. If the outer-
office user wants to make a call to an intra-office user other than user C, the
call can be transferred by user C to the target user.
● The internal numbers of users A, B, C, and D are 7100, 7000, 6000, and 6100
respectively.

NetEngine AR
● Both the signaling IP address and media IP address are 192.168.1.2.

● Users A and B belong to enterprise hw. The DN set is local. The intra-office
call prefix is 7. The outgoing call prefix is 8. Users C and D belong to
enterprise hw1. The DN set is local1. The intra-office call prefix is 6. The
outgoing call prefix is 9.
● A SIP AT0 trunk is used to route outgoing calls. The IP address of the IMS is
192.168.10.10, and the port number is 5060.
● Post-routing number change scheme 8 is configured to retain calling numbers
and delete the first digit of called numbers when users of enterprise 1 make
outgoing calls through the SIP AT0 trunk. Post-routing number change
scheme 9 is configured to retain calling numbers and delete the first digit of
called numbers when users of enterprise 2 make outgoing calls through the
SIP AT0 trunk.
● Figure 2-84 PBX sharing network
IMS network
SIP AT0
trunk
Eth1/0/0
PBX
Port Port
User A User D
2/0/0 2/0/1
User B User C
Enterprise 1 Enterprise 2
Campus
Configuration Roadmap
The configuration procedure is as follows:
1. Set the service mode of the router to PBX, and set public parameters.
Configure enterprises 1 and 2, and connect the enterprises to the router.
2. Configure the users, call prefixes, trunk group, call route, and post-routing
number change for enterprise 1 so that intra-office users of enterprise 1 can
make intra-office and outgoing calls.
3. Configure the users, call prefixes, trunk group, call route, and post-routing
number change for enterprise 2 so that intra-office users of enterprise 2 can
make intra-office and outgoing calls.

NetEngine AR
Data Plan

Short Number Segment Long Number Segment Remarks
7000–7004 N/A Enterprise 1
7100–7104 N/A
N/A 56623000
6000–6004 N/A Enterprise 2
6100–6104 N/A
N/A 56623001
Table 2-32 SIP AT0 trunk plan

Signal Port Media Peer Peer Route Remar Regist Regist
ing IP Numb IP IP Port ks ration ration
Addre er Addre Addre Numb ID Passw
ss ss ss er ord
192.16 5061 192.16 192.16 5060 0 Enterp 56623 12345

8.1.2 8.1.2 8.1.10 rise 1 000 678
192.16 5062 192.16 192.16 5060 0 Enterp 56623 12345

8.1.2 8.1.2 8.1.10 rise 2 001 678

Prefix Type Route Remarks
7 Intra-office N/A Enterprise 1
8 Outgoing 0
6 Intra-office N/A Enterprise 2
9 Outgoing 0

NetEngine AR
Prefix Outgoing Trunk Called Number Remarks

Change
8 SIP AT0 Deleting the first Enterprise 1

digit
9 SIP AT0 Deleting the first Enterprise 2

digit
Procedure
[Huawei] voice
[huawei-voice] quit
[Huawei] save
<Huawei>reboot
Step 2 Set the Ethernet IP address of interface 0/0/0 to 192.168.1.2, and add 192.168.1.2
to the media IP address pool and signaling IP address pool of the interface.
[Huawei] voice
[Huawei-voice] quit
[Huawei] voice


[Huawei-voice-enterprise-hw] dn-set local
[Huawei-voice] enterprise hw1
[Huawei-voice-enterprise-hw1] dn-set local1
[Huawei-voice-enterprise-hw1] quit

NetEngine AR
Step 6 Set the enterprise of users A and B to hw, the DN set to local, the intra-office call
prefix to 7, and the outgoing call prefix to 8.
NOTE
Set the enterprise of users C and D to hw1, the DN set to local1, the intra-office call prefix
to 6, and the outgoing call prefix to 9.
Step 7 Configure SIP user A whose user number is 7100 and authentication password is
a123456.
NOTE
Configure users D whose enterprise is hw1 and user number is 6100. For details, see the
configuration of user 7100.
[huawei-voice-pbxuser-7100] eid-para password cipher
Step 8 Configure user B 7000.

NOTE
Configure users C whose enterprise is hw1 and user number is 6000. For details, see the
configuration of user 7000.
[Huawei-voice-pbxuser-7000] call-right in idd enable
[Huawei-voice-pbxuser-7000] call-right out idd enable
Step 9 Configure a SIP AT0 trunk group.

[Huawei-voice-trunkgroup-sipat0] enterprise hw dn-set local

NetEngine AR

[Huawei-voice-trunkgroup-sipat01] enterprise hw1 dn-set local1
Step 10 Configure a call route and post-routing number change.

Configure post-routing number change scheme 8 to retain calling numbers and
delete the first digit of called numbers when users of enterprise 1 make outgoing
calls through the SIP AT0 trunk.
NOTE
Configure post-routing number change scheme 9 to retain calling numbers and delete the
first digit of called numbers when users of enterprise 2 make outgoing calls through the SIP
AT0 trunk.
[Huawei-voice] trunk-group sipat0
[Huawei-voice-trunkgroup-sipat0] callroute 8
[Huawei-voice-afterroute-change-8] trunk-group sipat0
[Huawei-voice-afterroute-change-8] caller no-change


number displayed to
user 7100 is 7000.

NetEngine AR
Outgoing call Calls can be made ● The outgoing call

displayed. For example, ● The outgoing trunk is
user 7000 can make an incorrectly configured.
the SIP AT0 trunk, and
the calling number
displayed to the called
party is 56623000.

56623000 and be
transferred to the target
user through user 7000.
----End
Configuration Files
#
ip address 192.168.1.2 255.255.255.0
#
voice
#
callroute 8
#
enterprise hw
dn-set local
#
enterprise hw1
dn-set local1
#
sipserver
media-ip 192.168.1.2
#
callroute 8
media-ip 192.168.1.2
trunk-sipat0 56623000 default-called-telno 7000
trunk-sipat0 56623000 password cipher %^%#wlyf~LstT9[[t|HjP*N3>t{Q3!f8>2e#PDM@Ga-:%^%#
#

NetEngine AR
enterprise hw1 dn-set local1

media-ip 192.168.1.2
trunk-sipat0 56623001 password cipher %^%##;d~MTyB}'|b2:@A=Rg25*=-5s^[<12[K8-M9pm5%^%#
#
callprefix 7
prefix 7
digit-length 4 32
#
callprefix 8
prefix 8
digit-length 1 32
callroute 8
#
telno 7000
port 2/0/0
dn-set local
call-right out idd enable
#
sipue 7100
telno 7100
dn-set local
eid-para password cipher %^%#Y;QQB*rk9,$7=K5av']G_[YWC~knl!4X[%Dvz%Q<%^%#
#
afterroute-change 8
callprefix 8
trunk-group sipat0
caller no-change
called del 1 1
#
2.20.5 Example for Configuring an AR as a Branch Gateway to

Access UC
● When the central node is correctly connected to the AR local node:
– All users at the headquarters and branches register with the central node.
– The central node processes all internal calls.
● When the central node is faulty or disconnects from the local node, local
users register with the local node, and the local node processes service
requests (including intra-office calls and incoming and outgoing calls) from
local users. This is known as local regeneration.
Figure 2-85 shows the typical network.
The unified gateway is connected to the AR to achieve the following objectives:
● Intra-office calls can be made by dialing short numbers between IP phones,
POTS phones, and fax machines at the headquarters, branch 1, and branch 2.
● The central node can be connected to local nodes through SIP trunks and to
the PSTN to implement incoming and outgoing calls.

NetEngine AR
Figure 2-85 Typical networking
PRA
POTS phone
U1900
IAD Central Node
10.10.10.3/24 10.10.10.2/24
Fax machine
IP Phone
10.10.10.4/24
Headquarters 10.10.10.1/24 Router
Network cable
192.168.1.2/24
E1/T1 trunk cable
Phone line
Router Router
192.168.1.1/24 172.16.1.1/24
POTS phone
IAD
192.168.1.3/24
Fax machine
IP Phone POTS IP Phone

192.168.1.4/24 phone 172.16.1.4/24
POTS
U1900 AR phone
Local Node 1 Fax Local Node 2
machine 172.16.1.2/24
PRA PRA
Branch 1 Branch 2
Data Plan
Table 2-35 User numbers
Endpoint Short Number Long number
IP phone 86000 28886000
POTS phone 88001 -

NetEngine AR
Endpoint Short Number Long number
Automatic switchboard - 28888999

access code
Table 2-36 SIP trunk group

Locatio NE Peer Trunk Trunk Media Local Peer
n Office Type Group and Port Port
Name Signali Numbe Numbe
ng IP r r
Branch Local Central SIP IP sipip 172.16. 5063 5060

2 Node 2 Node 1.2
SIP AT0 sipat0 172.16. 5061

1.2
Table 2-37 PRA trunk group

Location NE Peer Trunk Type Slot/ Rout
Office Subcard/
Port
Number
Branch 2 Local PSTN PRA DSS1 1/0/0 0

Node 2
Table 2-38 Prefix and number change

Loc NE Prefix Prefix Office Called Callin Remarks
atio Type Route Numbe g
n Select r Numb
ion Change er
Code Chang
e
Bra Loc Intra- 8 - - - Intra-office prefix

nch al office
2 Nod call
e2 prefix
Intra- 2888 - - - Intra-office prefix

office
call
prefix

NetEngine AR
Loc NE Prefix Prefix Office Called Callin Remarks

atio Type Route Numbe g
n Select r Numb
ion Change er
Code Chang
e
Nationa 90 Rout: Delete Chang Outgoing calls to

l toll 1 the first es the the local PSTN.
call digit. numbe
prefix r to
the
Internat 900 autom
ional atic
toll call switch
prefix board
Outgoin 9 access
g call code
prefix 28888
999 of
Outgoin 9021 branch
g call 2.
prefix
Table 2-39 Board

Location NE Board Type Slot Number
Branch 2 Local Node 2E1/T1–F 1

2
4FXS1FXO 2
Procedure
Step 1 Configure the service mode to IP PBX.
[Huawei] voice
[Huawei] save
<Huawei> reboot
Step 2 Set the IP address of interface 0/0/0 to 172.16.1.2, and add 172.16.1.2 to the
media and signaling IP address pools of the interface.

NetEngine AR
[Huawei] voice

Step 4 Set the default country code to 86 and default area code to 021. Enable country/
area code transformation.

1. Configure intra-office prefix 8 whose call attribute is 0.
The method for configuring intra-office prefix 2888 is similar. Set the minimal
number length to 8.
2. Configure inter-office prefix 9 whose call attribute is 1 and route is 1.
Use the same method to configure inter-office prefix 9021 whose call
attribute is 1, national inter-office prefix 90 whose call attribute is 2, and
international inter-office prefix 900 whose call attribute is 3.
3. Configure the automatic switchboard. The automatic switchboard name is ivr,

and number is 28888999.
NOTE
Before configuring the automatic switchboard, perform IVR configuration by referring to

the AR Product Documentation. Use the same value for vu-service-name of the
automatic switchboard and service of the IVR.

1. Configure the SIP user whose number is 86000.

NetEngine AR
SIP users at branches have been configured on the central node, and SIP user
numbers have been synchronized to the local node, so user numbers do not
need to be allocated to the SIP users on the local node.
2. Configure the POTS user whose number is 88001 and call rights to all.
[Huawei-voice-pbxuser-88001] proxyreg-id 88001
[Huawei-voice-pbxuser-88001] proxyreg-password cipher
//The password is the same as the authentication password for adding the POTS user of local node
AR on the U1900 central node.
//When a POTS user registers with the central node through the AR local agent, configure the header
field on the AR, enabling the unified gateway at the central node to identify the POTS user.
[Huawei-voice] sip
[Huawei-voice-sip] field-header user-agent HUAWEI-eSpace-UCExpress

1. Configure the PRA trunk group.
[Huawei] voice
[Huawei-voice] quit
[Huawei] voice
2. Configure the SIP AT0 trunk group.

[Huawei-voice-trunkgroup-sipat0] trunk-sipat0 28888001 default-called-telno 88001
3. Configure the SIP IP trunk group.

[Huawei-voice] trunk-group sipip sip no-register
[Huawei-voice-trunkgroup-sipip] callroute 2
[Huawei-voice-trunkgroup-sipip] signalling-address ip 172.16.1.2 port 5063
[Huawei-voice-trunkgroup-sipip] media-ip 172.16.1.2
[Huawei-voice-trunkgroup-sipip] peer-address static 10.10.10.2 5060
[Huawei-voice-trunkgroup-sipip] home-domain abcd.com
[Huawei-voice-trunkgroup-sipip] register-uri abcd.com
[Huawei-voice-trunkgroup-sipip] reset
[Huawei-voice-trunkgroup-sipip] quit

NetEngine AR
Step 8 Configure the post-routing number change.

[Huawei-voice-afterroute-change-9] trunk-group pra
[Huawei-voice-afterroute-change-9] caller del-then-insert 1 32 28888999
[Huawei-voice-afterroute-change-9] quit
Step 9 Configure local regeneration.

[Huawei-voice-local-survival] dataserver ip 10.10.10.2 port 8099
[Huawei-voice-local-survival] dataservertype u1900
[Huawei-voice-local-survival] primary-trunk-group sipip proxyreg-trunk-group sipat0
[Huawei-voice-local-survival] local-address ip 172.16.1.2 port 8000
[Huawei-voice-local-survival] password cipher
//Assume that the local regeneration BIN channel authentication password is a12345678987654321 which
is the same as the authentication password for adding the local node AR on the U1900 central node.
[Huawei-voice-local-survival] sync-interval 2
Step 10 Import certificate and private key files.

NOTE
Skip this step if the transmission mode in step 9 is configured to TCP. However, non-encrypted
TCP transmission has security risks. It is recommended that you use TLS transmission.
1. Obtain the servercert.pem certificate file and serverkey.pem private key file
from the U1900 series unified gateway host software package (if you do not
have the software package, download it from https://support.huawei.com/
enterprise).
Certificate and private key files are credentials for TLS transmission
authentication. Matched certificate and private key files are preconfigured
when the U1900 series unified gateways are delivered.
It is recommended that you replace the preconfigured files with certificate
and private key files generated by the customer or issued by an official
authority. After certificate and private key files are replaced on the AR, import
matched certificate and private key files to the U1900 series unified gateway.
2. Upload certificate and private key files to the AR.
3. Configure the policy.
[Huawei] pki realm u1900
[Huawei-pki-realm-u1900] quit
[Huawei] ssl policy u1900 type server
[Huawei-ssl-policy-u1900] pki-realm u1900
[Huawei-ssl-policy-u1900] quit
4. Import the files.

[Huawei] pki import-certificate local u1900 pem
Please enter the name of certificate file <length 1-127>: servercert.pem
You are importing a local certificate, the current private keyis required.
Please enter the name of private key file <length 1-127>: serverkey.pem
Please enter the type of private key file(pem , p12): pem
The current password is required, please enter your password <length 1-31 >:********
Successfully imported the certificate.

NetEngine AR
NOTE
You can obtain the decryption password for the private key file attached with the U1900
series unified gateway from Configuration > Configuration Guide > Advanced
Configuration > Configuring Signaling Encryption in the eSpace U1900 series unified
gateway product documentation.
5. Access the local-survival view and bind the policy.
[Huawei-voice-local-survival] ssl-server-policy u1900
Step 11 Configuration Files

No. Action Expected Result
1 Verifying intra-office calls 1. IP phone 81000 rings. The call is

made by short numbers set up after the IP phone is picked
between SIP users and POTS up.
users 2. POTS phone 88001 rings. The call
1. Pick up the IP phone 86000 is set up after the POTS phone is
at branch 2 and call IP picked up.
phone 81000 at the
headquarters.
2. Pick up IP phone 86000 at
branch 2 and call POTS
phone 88001 at branch 2.
2 Verifying outgoing calls 1. PSTN user B's phone rings. The

1. Pick up IP phone 86000 at call is set up after the phone is
branch 2 and dial picked up.
9XXXXXXXX to call PSTN 2. PSTN user A's phone rings. The
user B at branch 2. call is set up after the phone is
2. Pick up IP phone 86000 at picked up.
branch 2 and dial
9010XXXXXXXX to call PSTN
user A at the headquarters.
3 Verifying incoming calls 1. IP phone 28886000 rings. The call

1. Pick up the phone of PSTN is set up after the IP phone is
user B at branch 2 and dial picked up.
the long number 28886000 2. POTS phone 88001 rings. The call
to call IP phone 86000 at is set up after the POTS phone is
branch 1. picked up.
2. Pick up the phone of PSTN
user B at branch 2 and dial
the enterprise switchboard
number 28888999 and then
the extension number
88001.

NetEngine AR
No. Action Expected Result
4 Verifying local regeneration 1. POTS phone 88001 rings. The call

calls by disconnecting from the is set up after the POTS phone is
central node picked up.
1. Pick up IP phone 86000 at 2. PSTN user B's phone rings. The
branch 2 and call POTS call is set up after the phone is
phone 88001 at branch 1. picked up.
2. Pick up IP phone 86000 at 3. IP phone 28886000 rings. The call
branch 2 and dial is set up after the IP phone is
9XXXXXXXX to call PSTN picked up.
user B at branch 2. 4. POTS phone 88001 rings. The call
3. Pick up the phone of PSTN is set up after the POTS phone is
user B at branch 2 and dial picked up.
the long number 28886000
to call IP phone 86000 at
branch 2.
4. Pick up the phone of PSTN
user B at branch 2 and dial
the enterprise switchboard
number 28888999 and then
the extension number
88001.
----End
Verify the configuration.

#
#
#
pki realm u1900
#
ssl policy u1900 type server
pki-realm u1900
#
ip address 172.16.1.2 255.255.255.0
#
voice
#
port ve1 1/0/0
signal CCS
#
sip
field-header user-agent HUAWEI-eSpace-UCExpress
#
callroute 1
#
callroute 2
#

NetEngine AR
sipserver
media-ip 172.16.1.2
#
trunk-pra 1/0/0
callroute 1
#
media-ip 172.16.1.2
#
trunk-group sipip sip no-register
callroute 2
media-ip 172.16.1.2
#
callprefix 8
prefix 8
digit-length 5 32
#
callprefix 9
prefix 9
digit-length 1 32
callroute 1
#
callprefix ivr
prefix 28888999
digit-length 8 32
#
local-survival
dataserver ip 10.10.10.2
dataservertype u1900
local-address ip 172.16.1.2
sync-interval 2
password cipher %^%#nw@y%OP0$#],HR"wQH/3`|.@A7+ZttF2*1D!)C~.or3f~>0ZB#EX,3dEoR%^%#
ssl-server-policy u1900
primary-trunk-group sipip proxyreg-trunk-group sipat0
#
pbxuser 88001 pots
telno 88001
port 2/0/1
call-right out all
#
afterroute-change 9
callprefix 9
trunk-group pra
called del 1 1
#

NetEngine AR
CLI-based Configuration Guide - Voice 3 SIP AG Configuration
3 SIP AG Configuration
This document describes the implementation and configurations of the Session

Initiation Protocol access gateway (SIP AG).
3.1 Overview of the SIP AG

This section describes the Session Initiation Protocol access gateway (SIP AG) in
Voice over IP (VoIP) applications.
3.2 Understanding the SIP AG
3.3 SIP AG Features Supported by the Device
This section describes SIP AG features supported by the device.
3.4 Application Scenarios for the SIP AG
3.5 Licensing Requirements and Limitations for the SIP AG
This section describes the notes about configuring the SIP AG.
3.6 Setting Voice Parameters on a SIP AG
This section describes how to set voice parameters on a SIP AG.
3.7 Configuring a SIP AG Interface
You can configure a SIG AG interface to connect SIP users to the IMS for voice,
data, and multimedia services.
3.8 Configuring a SIP AG User
This section describes how to configure a SIP AG user.
3.9 Maintaining a SIP AG
This section describes how to maintain a SIP AG.
This topic describes how to set physical voice interface parameters and SIP AG
global voice parameters.
3.1 Overview of the SIP AG

This section describes the Session Initiation Protocol access gateway (SIP AG) in
Voice over IP (VoIP) applications.

NetEngine AR
SIP AG
A 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.
The packet switched network (PSN) development brings revolutionary changes to

the voice phone system and many new technologies emerge. VoIP transmits voice
services such as telephony services on an IP network, and the IP multimedia
subsystem (IMS) promotes development of VoIP applications. An IMS network is a
standard next generation carrier network that provides mobile or fixed-line
multimedia services. It supports traditional packet switched and circuit switched
telephony systems. Compared with the public switched telephone network (PSTN),
VoIP features higher resource utilization and VoIP calls do not occupy telephone
lines exclusively. VoIP has been applied for commercial use. The line switched
PSTN has developed for many years and currently there are a large number of
devices. Replacing the PSTN with VoIP takes high costs. A device can function as
the SIP AG to connect the PSTN to IP networks with low costs.
As shown in Figure 3-1, the device functions as a SIP AG to integrate voice

networks with IP networks.
Figure 3-1 SIP AG as a voice gateway
Terms
● 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.

NetEngine AR
● SIP
SIP is a text-based signaling protocol. SIP messages are classified into Request
and Response messages. As an application layer protocol, SIP establishes,
modifies, or terminates multimedia sessions and creates and controls
multimedia sessions among two or more parties. SIP can work with specified
protocols to complete session setup and media negotiation, such as 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).
3.2 Understanding the SIP AG

3.2.1 SIP AG Registration
SIP AG Registration Process
Before a SIP user initiates a call, the user must register user information on the
home network. After the user is added, the user registration process starts.
A SIP AG user needs to register with the IMS core when the user initiates a call for
the first time. Figure 3-2 shows the registration process.
1. The SIP AG sends a REGISTER message to the IMS core.
2. If the IMS core requires authentication but cannot find user information in the
database, it sends a 401 message, which contains information required for
authentication.
3. The SIP AG asks the user to enter the user name and password. It encrypts
the user name and password, encapsulates them into a REGISTER message,
and sends the REGISTER message to the IMS core.
4. The IMS core decrypts the user information from the REGISTER message and
sends the user information to the authentication/accounting center. After the
user is authenticated, the PBX records the user information in the database
and sends a 200 OK message to the SIP AG.
Figure 3-2 SIP AG registration process
SIP AG IMS Core
REGISTER (1)
401(2)
REGISTER (3)
200 OK (4)

NetEngine AR
SIP AG Registration Redirection Process

If a SIP AG obtains the IP address and port number of the remote end through
DNS, the SIP AG can be redirected for registration. By using the DNS, the SIP AG
can obtain one or more IP addresses. The address can be either an IMS core
address or a redirection server address. When a user initiates a call for the first
time, the SIP AG sends a REGISTER message to the IP address obtained through
the DNS. If this IP address is a redirection server address, the SIP AG initiates
registration redirection.
Figure 3-3 shows the SIP AG registration redirection process.
1. When a user initiates a call for the first time, the SIP AG sends a REGISTER
message to the redirection server.
2. The redirection server sends a 301 Moved Permanently message to the SIP
AG, which contains a redirection address. When receiving the 301 Moved
Permanently message, the SIP AG verifies whether the redirection address is
in the IP address list obtained through the DNS. If the address is not in the
list, the SIP AG repeats Step 1 and Step 2 to obtain a new redirection address,
until it obtains a redirection address that is in the IP address list.
3. If the address is in the list, the SIP AG sends a REGISTER message to the
corresponding IMS core.
4. After the IMS core receives the REGISTER message, it verifies the validity of
user information, saves the information in the database, and sends a 200 OK
message to the SIP AG. After registration redirection, the SIP AG connects to
the redirected IMS core to perform all interactions.
Figure 3-3 SIP AG registration redirection process
Redirected
SIP AG IMS Core
Server
REGISTER(1)
301(2)
REGISTER(3)
200 OK(4)
SIP AG Deregistration Process

Figure 3-4 shows the SIP AG deregistration process.
1. The SIP AG sends a REGISTER message (deregistration request) to the IMS
core.

NetEngine AR
2. When receiving the REGISTER message, the IMS core sends a 200 OK message
to the SIP AG and deregisters the user.
Figure 3-4 SIP AG deregistration process
SIP AG IMS Core
REGISTER (1)
200 OK (2)
3.2.2 SIP AG Call Flow

Figure 3-5 shows the SIP AG call flow.
1. The calling party POTS1 picks up a phone. When the SIP AG detects that the
calling party picks up a phone, it allocates DSP resources to the calling party
and plays a dial tone.
2. When detecting that the calling party dials the first digit of a dialing number,
SIP AG1 stops the dial tone and matches the digit with a digitmap.
3. When detecting that the calling party dials the last digit of the dialing
number, the SIP AG matches the dialing number with digitmaps. When the
dialing number matches a digitmap, the SIP AG constructs an Invite message
and sends it to the IMS proxy server.
4. The IMS proxy server receives the Invite message from the calling party,
analyzes the called number, finds SIP AG2 to which the called party belongs,
sends an Invite message to the called party, and returns a 100 Trying message
to SIP AG1.
5. The called party receives the Invite message and responds with a 100 Trying
message, indicating that it is receiving the called number.
6. SIP AG2 receives a complete called number, requests the called party POTS2
to ring, and sends a 180 Ringing message to SIP AG1. The calling party then
hears the ringback tone.
7. The called party POTS2 picks up a phone. When SIP AG2 detects that the
called party picks up a phone, it sends a 200 OK message to the calling party
POTS1.
8. The calling party stops playing the ringback tone, and SIP AG1 sends an ACK
message to the called party.
9. A call is established.
10. The calling party hangs up.
11. When detecting that the calling party hangs up the phone, the IMS proxy
server sends a BYE message to the called party.

NetEngine AR
12. SIP AG2 receives the BYE message, plays the busy tone to the called party,
and sends a 200 OK message to the calling party.
13. The called party hangs up the phone.
Figure 3-5 SIP AG called process
IMS
Network
Proxy
POTS1 SIPAG1 SIPAG2 POTS2
server
Off hook
Plays a dial tone
Dials the first digit
Stops the dial tone
Dials the last digit
Matches a digitmap
INVITE
INVITE
100 Trying
100 Trying
Plays ring tone
180 Ringing
180 Ringing
Plays ringback tone
Picks up phone
200 OK
200 OK
Stops ringback tone
ACK
ACK
Call set up
Hangs up
BYE
BYE
Plays busy tone
200 OK Hangs up
200 OK
3.3 SIP AG Features Supported by the Device

This section describes SIP AG features supported by the device.

NetEngine AR
Using a SIP AG as a Voice Gateway

As shown in Figure 3-6, a SIP AG functions as the IP voice gateway to implement
communication between an IP network and a public switched telephone network
(PSTN) or integrated services digital network (ISDN). The following describes the
access process of a POTS user as an example:
● 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.
Figure 3-6 SIP AGs functioning as voice gateways
IMS/IP
SIPAG
IP link
FXS link
E1 link
ISDN link
POTS Modem FAX ISDN TDM PBX
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.

NetEngine AR
Service Type Introduction Configure

d on the
SIP AG or
Not
Basic voice The basic voice service is the basic call Yes
service connection function, including intra-office calls,
local calls, national toll calls, international toll
calls, and transit calls.
Three-party The third-party service allows a calling party or Yes

service 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.
Call waiting When UserA is talking with UserB over the Yes
service phone and at this moment UserC is calling
UserA, UserA hears a call waiting tone,
indicating that there is a call waiting for UserA.
MWI service The message waiting indicator (MWI) service Yes

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.
Malicious call The user that registers the MCID service with Yes
identification the carrier can query the phone number of the
(MCID) service attacker that initiates malicious calls after
performing relevant operations.
Call transfer The call transfer service allows the called party Yes
service 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.
Call conference The call conference service allows more than Yes
service three parties to communicate together.
Calling line The CLIP service displays the calling number in No

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

identification a calling party on the terminal of a called party.
restriction (CLIR)
service
Distinctive The distinctive ringing service plays different No

ringing service ring tones for incoming calls.

NetEngine AR
Service Type Introduction Configure

d on the
SIP AG or
Not
Differentiated The differentiated ringback tone service enables No

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

(AoC) service the charge rate, fee notification during a call,
and the total fee of the call.
Urgent call If the SIP AG finds an urgent call, the SIP AG No

process inserts the urgent call flag into the SIP
message.
Completion of When the called number is busy, the CCBS No

Calls to Busy service enables the SIP AG to monitor the
Subscriber called party status. When the called party is
(CCBS) service 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.
Multiple MSN Multiple MSN numbers can be configured on a No

numbers on a POTS interface.
POTS interface
Hotline service ● Instant hotline service: After a user picks up Yes

a phone, the SIP AG dials the hotline
number for the user.
● 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.
Anonymous call The anonymous call service enables the called No

service party not to view information about incoming
calls.
3.4 Application Scenarios for the SIP AG

Using SIP AGs as Voice Gateways
As shown in Figure 3-7, 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:
● The SIP AG connects small-scale and distributed enterprise users to analog
phones or fax machines to implement voice communication.

NetEngine AR
● The SIP AG is registered with the IMS. Then the IMS provides and controls
user services on the SIP AG.
Figure 3-7 Networking where the SIP AG functions as the voice gateway
IMS
Network
SIPAG
POTS Modem FAX POTS
FXS (RJ11 telephone line)

IP link
Connecting to the PSTN Through a Trunk

As shown in Figure 3-8, a SIP AG connects to a PBX through a PRA trunk and to
an ISDN phone through a BRA interface.

NetEngine AR
Figure 3-8 SIP AGs connect to the PSTN through a trunk
IMS
Network
SIPAG
SIPAG
PRA trunk
BRA interface
PBX
POTS ISDN POTS
POTS
IP link
ISDN-ST cable
E1 cable
3.5 Licensing Requirements and Limitations for the SIP

AG
This section describes the notes about configuring the SIP AG.

None
NOTE
LTE4EA.

NetEngine AR




Cards-Voice Card.
E1/T1 Card.
Feature Limitations
None.

NetEngine AR
3.6 Setting Voice Parameters on a SIP AG

This section describes how to set voice parameters on a SIP AG.
3.6.1 Configuring the Device to Work in SIP AG Mode
Context
You can configure the device to work in SIP AG mode, H.248 AG mode, or PBX
mode. Before configuring SIP AG service features, configure the device to work in
SIP AG mode. Before the configuration, run the display voice service-mode
command to query the working mode of the device. If the working mode is not
SIP AG, you need to run the display current-configuration command to query
the configurations and then delete all configurations in the voice view. Then, set
the working mode of the device to SIP AG. If the device works in SIP AG mode,
skip this configuration.
Pre-configuration Tasks
Before configuring the device to work in SIP AG mode, complete the following
task:
● 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 SIP AG
The device is configured to work in SIP AG mode.

Step 4 Run:
quit
Return to the system view.

Step 5 Run:
quit
Return to the user view.

Step 6 Run:
save

NetEngine AR
The configuration is saved.
Step 7 Run:
reboot
The device is restarted.
NOTE
After the device is configured to work in SIP AG mode, restart the device to make the
configuration take effect.
----End
Verifying the Configuration

Run the display voice service-mode command to view the working mode of the
device.
<Huawei> display voice service-mode
The voice service mode is SIP AG.
After the configuration is complete, the device works in SIP AG mode.
Context
The SIP AG interface can be used only after the interface IP address is added to
the media and signaling IP address pools. A SIP AG interface must obtain media
and signaling IP addresses from media and signaling IP address pools respectively.
Media IP address pool is generally the same as the signaling IP address pool.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
voip-address media interface interface-type interface-number { ip-address | dynamic }
The IP address of the specified SIP AG is added to the media IP address pool.
Step 4 Run:
voip-address signalling interface interface-type interface-number { ip-address | dynamic }
The IP address of the specified SIP AG is added to the signaling IP address pool.
When configuring media and signaling IP address pools, note the following points:

NetEngine AR
● To specify the ip-address parameter, ensure that an IP address has been

configured for the specified interface.
● To specify the dynamic parameter, ensure that an IP address has been
dynamically assigned to the specified interface.
----End

Run the display voice voip-address command to verify the configured signaling
and media IP address pools.
3.7 Configuring a SIP AG Interface

You can configure a SIG AG interface to connect SIP users to the IMS for voice,
Context
If users want to connect to the IMS for voice, data, and multimedia services, a SIP
AG interface must be configured for registration on the IMS. SIP AG users connect
to the IMS through the SIP AG interface to use voice, data, and multimedia
services.
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.
Before configuring a SIP AG interface, complete the following task:
● 3.6.1 Configuring the Device to Work in SIP AG Mode
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
SIP AG SIP AG-interface-id
A SIP AG interface is created and the SIP AG interface view is displayed.
Step 4 Set mandatory parameters on the SIP AG interface.

NetEngine AR
Parameter Command Remarks
Media IP address media-addr { media-ip | addr- The media IP

name media-addr-name } address of the SIP
AG interface must
be obtained from
the media IP
address pool.
By default, no
media IP address or
media IP address
name is configured
for a SIP AG.
Signaling IP address signalling-addr { signal-ip | The signaling IP

and signaling port addr-name signal-addr-name } address of the SIP
number signal-port AG interface must
be obtained from
the signaling IP
address pool.
By default, no
signaling port
number is
configured for a SIP
AG interface.
Parame Static IP primary-proxy-addr static By default, no static

ters of address static-ip-address static-ip- IP address is
the address-port [ secondary static- configured for the
primar ip-address2 ] primary proxy
y proxy server.
server
DNS-A primary-proxy-addr dns-a dns- When this
domain a-domain-name dns-a-port- command is used
name and number to configure the
port primary proxy
number server, the SIP AG
can be redirected
for registration.
By default, no DNS-
A domain name or
port number is
configured for the
primary proxy
server.

NetEngine AR

NOTE DNS-NAPTR primary-proxy-addr dns-naptr When this
The domain dns-naptr-domain-name command is used
four
param
name to configure the
eters primary proxy
for server, the SIP AG
the can be redirected
primar for registration.
y
proxy By default, no DNS-
server NAPTR domain
canno name is configured
t be for the primary
config
proxy server.
ured
simult DNS-SRV primary-proxy-addr dns-srv When this
aneou
sly.
domain dns-srv-domain-name command is used
Only name to configure the
the primary proxy
latest server, the SIP AG
config can be redirected
uratio for registration.
n
takes By default, no DNS-
effect. SRV domain name
is configured for
the primary proxy
server.
Step 5 Run:
reset
The SIP AG is reset.
NOTICE
Exercise caution when you run this command because resetting a SIP AG affects
running services.
Do not frequently modify SIP AG interface parameters. Reset the SIP AG to make
the configuration take effect.
Step 6 Run:
quit
Return to the voice view.
Step 7 (Optional) Run:

local-digitmap name-value { append | type { normal | emergency | scc | direct-centrex | second-
centrex } }body-value
The digitmap is configured.

NetEngine AR
Step 8 (Optional) Set optional parameters on the SIP AG interface.

The SIP AG interface SIP AG SIP AG-interface-id Enter the SIP AG

view is displayed interface ID that is
created in step 3
and run this
command to enter
the SIP AG interface
view. If a new SIP
AG interface ID is
entered, this
command creates
the SIP AG interface
and displays the SIP
AG interface view.
Transmission protocol transfer transfer-protocol Enumerated type.

The values are as
follows:
● TCP: specifies the
Transmission
Control Protocol
(TCP) as the
transmission
protocol.
● UDP: specifies
UDP as the
transmission
protocol.
● SCTP: specifies
SCTP as the
transmission
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- By default, no home

value domain name is
AG interface.
Description description description By default, no

description is
AG interface.

NetEngine AR
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
AG interface.
Conference factory conference-factory-uri uri By default, no

URI conference factory
URI is configured
for a SIP AG
interface.
Authen Authenticati auth mode mode-value Enumerated type.

ticatio on mode The values are as
n follows:
param ● interface:
eters configures all the
users connected
to a SIP AG to
share the same
authentication
user name.
● user: specifies an
IP Multimedia
Subsystem
Private Identity
(IMPI) for a user.
By default, the
authentication
mode is user.
Authenticati auth username auth-username By default, no

on user password { cipher cipher- authentication user
name and password | ha1 cipher cipher- name or password
password password } is configured for a
SIP AG user.
Timer value digitmap-timer { long long- By default, the

timer-value | short short-timer- value of a digitmap
value | start start-timer-value } long timer is 20s,
the value of a
digitmap short
timer is 4s, and the
value of a digitmap
start timer is 20s.

NetEngine AR
DTMF parameters dtmf-transmission-mode By default, the

{ throughly | erase } DTMF transmission
mode is transparent
transmission.
Fax/ Fax/Modem fax-modem common By default, the fax/

Mode codec negotiation-mode { negotiate | modem codec
m negotiation self-switch } negotiation mode is
param mode negotiation.
eters
Packetizatio fax-modem common rtp- By default, the
n interval in interval { 5ms | 10ms | 20ms | packetization
transparent 30ms } interval in
transmissio transparent
n mode transmission mode
is 10 ms.
VBD fax-modem common vbd- By default, the VBD

attribute attribute-type { chinatelecom | attribute type is V.
type extchinatelecom | ietf | v152 } 152.
VBD codec fax-modem common vbd-codec By default, the VBD

mode { g711a | g711u } codec mode is A-
law algorithm in G.
711.
VBD fax-modem common vbd-pt- By default, the VBD

payload type { dynamic | static } payload type is
type static.
Fax fax-modem fax transmission- By default, the fax

transmissio mode { throughly | t38 } transmission mode
n mode of a SIP AG
interface is
transparent
transmission.
Modem fax-modem modem By default, the

transmissio transmission-mode { throughly modem
n mode | relay } transmission mode
of a SIP AG
interface is
transparent
transmission.
Param Static IP secondary-proxy-addr static By default, no static

eters address and static-ip-address static-ip- IP address or port
of the port address-port [ secondary static- number is
second number ip-address2 ] configured for the
ary secondary proxy
proxy server.
server

NetEngine AR
DNS-A secondary-proxy-addr dns-a By default, no DNS-

domain dns-a-domain-name dns-a-port- A domain name or
name and number port number is
port configured for the
number secondary proxy
server.
DNS-NAPTR secondary-proxy-addr dns- By default, no DNS-

domain naptr dns-naptr-domain-name NAPTR domain
name name is configured
for the secondary
proxy server.
DNS-SRV secondary-proxy-addr dns-srv By default, no DNS-

domain dns-srv-domain-name SRV domain name
name is configured for the
secondary proxy
server.
Mode used for proxy-dhcp-option option-value By default, the

sending DHCP mode used for
Request messages sending DHCP
Request messages is
none.
Control points ag number-parameter name Different control

value points may use
different value
ranges. For details,
see ag number-
parameter.
String parameters ag string-parameter strpara- For details about

name strpara-value string parameters,
see ag string-
parameter.
RFC RFC 2833 nte-dtmf { enable | disable } By default, RFC

2833 DTMF 2833 DTMF
transm transmissio transmission is
ission n enabled.
RFC 2833 nte-fax-modem { enable | By default, RFC

fax and disable } 2833 fax and
modem modem
transmissio transmission is
n enabled.
RFC 2833 nte-flash-hook { enable | By default, RFC

hookflash disable } 2833 hookflash
transmissio transmission is
n disabled.

NetEngine AR
Mode in nte-negotiation-mode mode- By default, RFC

which RFC value 2833 redundancy
2833 transmission is
transmissio started based on
n is started negotiation in
based on passive-nego mode.
negotiation
Proxy Mode used proxy-detect-mode { option | By default, a SIP AG

functio by a SIP AG probe | register } uses the probe
ns to detect a mode to detect a
proxy server proxy server.
Mode in proxy-dual-home { disable | By default, a SIP AG

which a SIP manual-switchover | auto- supports dual
AG is dual switchover } homing but does
homed to not support
proxy automatic
servers switchover.
Mode used proxy-refresh-mode { no-switch By default, a SIP AG

by a SIP AG | defer | immediate } updates the proxy
to update server IP address
the proxy after a delay (defer
server mode).
address
Registration of a SIP proxy-switchover { primary | To maintain or

AG with the specified secondary } troubleshoot the
proxy server current proxy server,
run this command.
RFC Mode in redundancy-negotiation-mode By default, RFC

2198 which RFC negotiation-mode 2198 redundancy
redund 2198 transmission is
ancy redundancy started in no-
transm transmissio initiative-start
ission n is started mode.
Transmissio redundancy-nte { enable | By default, a SIP AG

n of RFC disable } is disabled from
2833 audio using RFC 2198
data using redundancy
RFC 2198 transmission to
redundancy transmit the RFC
transmissio 2833 audio data.
n

NetEngine AR
Mode in redundancy-start-mode mode- By default, RFC

which RFC value 2198 redundancy
2198 transmission is
redundancy started in ordinary
transmissio mode.
n is started
VBD redundancy-vbd { enable | By default, a SIP AG

transmissio disable } is enabled to
n using RFC transmit VBD using
2198 RFC 2198
redundancy redundancy
transmissio transmission.
n
Voice redundancy-voice { enable | By default, a SIP AG

transmissio disable } does not use RFC
n using RFC 2198 redundancy
2198 transmission to
redundancy transmit voice.
transmissio
n
Registrar URI register-server-uri uri By default, no

registrar URI is
AG.
Ringing parameters ring-mode index ringmode- By default, ringing

name [ cadence-ring parameters of a SIP
cadencering-value | initial-ring AG is empty.
initialring-value ] *

NetEngine AR
SCTP stack sctp { assoc-maximum- After SCTP stack

parameters retransmission assoc-max- parameters are set,
retrans | checksum-arithmetic reset the SIP AG to
checksum-arith | heartbeat- make SCTP stack
interval hb-interval | high- parameters take
congestion-level high- effect.
congestion-level-value | in-
stream-number in-stream-num |
low-congestion-level low-
congestion-level-value |
maximum-init-retransmission
max-init-retrans | no-
congestion-level no-congestion-
level-value | out-stream-number
out-stream-num | path-
maximum-retransmission path-
max-retrans | rto-alpha rto-
alpha-value | rto-beta rto-beta-
value | rto-init rto-init-value |
rto-maximum rto-max | valid-
cookie-life valid-cookie-life-
value }
SDP negotiation sdp-negotiation-mode { local | By default, the
mode remote } remote priority
mode is used for
SDP negotiation.
Index of the service service-logic service-logic-index By default, the

logic used by a SIP service logic index
AG of a SIP AG is 0.
Mapping between a status-code-mapping scene After configuring

call scenario and a status-code the status code of a
status code call scenario, reset
the SIP AG to make
the configuration
take effect.
MWI MWI subscribe mwi { enable | By default, the MWI

subscri subscription disable } subscription
ption function function is disabled.
functio
n Register subscribe reg { enable | By default, the
status disable } register status
subscription subscription
function function is disabled.
UA profile subscribe ua-profile { enable | By default, the UA

subscription disable } profile subscription
function function is enabled
on a SIP AG.

NetEngine AR
Codec priority, codec user-defined-profile codec- Run the profile

value, and priority pri-value codec-value pt- command to set
packetization interval value profile to 0. If you
change the priority
of a codec value,
priorities of other
codec values are
also changed.
Control point with a user-defined-profile number- Sets the control

specified sequence parameter strpara-no strpara- point with a
number value specified sequence
number for a SIP
AG.
Priority of a service user-defined-profile service- If you change the

type priority pri-value srvid-value priority of a service
type, priorities of
other service types
are also changed.
Step 9 Run:
save
The configuration is saved.
----End

Run the display voice SIP AG [ SIP AG-interface-id { running | config } ]
command to verify the configuration of SIP AG interface parameters.
3.8 Configuring a SIP AG User

This section describes how to configure a SIP AG user.
Context
On the IMS, a SIP AG is directly connected to a user terminal. You need to set
attributes 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:
● 3.7 Configuring a SIP AG Interface

NetEngine AR
Procedure
3.8.1 Setting Parameters for a SIP AG User is mandatory and other
configurations are optional.
3.8.1 Setting Parameters for a SIP AG User
Context
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:
SIP AGuser SIP AGuser-name [ port [ pots | bra ] interface-number ]
NOTE
When creating a SIP AG user, specify the interface number using the port interface-number
parameter. When entering the view of an existing SIP AG user, you do not need to input the
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 SIP AG-interface-ID
A SIP AG ID is specified for the SIP AG user.
Step 5 Run:
base-telno telno-value [ SIP AGusergroup usergroup-id ]
The basic phone number is configured for the SIP AG user.
Step 6 Set optional parameters for the SIP AG user.
Parameter Command Default Setting
Timeslot active-mask time-slot time-slot- This command is valid

status value config-status config-status- only on VE1 interfaces.
value

NetEngine AR
Maximum auto-resume-limit auto-resume- 20

number of limit This command is valid
automatic only on VE1 and BRA
recovery interfaces.
attempts
from
deterioration
SIP AG's bell-ans-flag { enable | disable } disable

capability to This command is valid
support the only on FXS interfaces.
BELL ANS
flag
Calling clip-format { sdmffsk | mdmffsk | FSK simple data format

number dtmf | r15 | etsi }
format
Sequence in clip-transmission-sequence { after- after-ring

which the ring | before-ring }
calling
number of a
SIP AG user
is displayed
Power-off dc-time dc-time 10 ms

interval
Single-tone detect-ansbar-by-single-tone disable

ANSbar { enable | disable }
signal
detection
DSP chip dsp-input-gain dsp-input-gain 0 dB

input gain
DSP chip dsp-output-gain dsp-output-gain 0 dB

output gain
DSP dsp-template dsp-template No DSP template name

template configured
name
Service endservice { defer [ maxcobtime By default, services of a

termination maxcobtime-value ] | immediate } SIP AG user are not
of a terminated.
specified SIP
AG user
Extended extend-telno telno-value [ SIP No extended phone

phone AGusergroup usergroup-id ] number configured
number

NetEngine AR
FSK call fsk-mode { BELL_202 | CCITT_V23 | BELL_202

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

PayPhone }
SIP AG user priority { cat1 | cat2 | cat3 } cat3

priority
TAS mode of tas-pattern { NO-TAS | DT-AS } NO-TAS

CLIP
UNI fault uni-report { enable | disable } disable

reporting This command is valid
function only on VE1 and BRA
interfaces.
Voice quality vqe-agc { enable | disable } disable

enhancemen
t (VQE)
function
Target VQE vqe-agc-level vqe-agc-level 13

level
VQE noise vqe-sns { enable | disable } disable

suppression
Noise vqe-sns-level vqe-sns-level 12 dB

suppression
level
----End
3.8.2 (Optional) Configuring a SIP AG User Group

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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

NetEngine AR
Step 4 Set parameters for the SIP AG user group.

● Run:
group-name usergroup-name
The description is configured.

● Run:
auth username auth-username password { cipher cipher-password | ha1 cipher cipher-password }
The authentication user name and password are configured.

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

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

● Run:
uri uri
A URI is configured.
● Run:
endservice
Services of a specified user group are terminated.
----End
3.8.3 (Optional) Configuring Functions or Services in a SIP

Service Data Profile
Context
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.
By default, the user authority (UA) profile subscription function is enabled on a
SIP AG interface. The user authority set by the UA profile subscription function
takes preference over that set by local command lines. Run the subscribe ua-
profile disable command to disable the UA profile subscription function.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

NetEngine AR
Step 3 Run:
sipservicedata SIP AGuser-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.
Function/Service Command Default Setting
Authentication user auth username auth- No authentication user

name and password username password name and password
{ cipher cipher-password | configured.
ha1 cipher cipher-
password }
Dial tone type dial-tone { normal | Normal dial tone type
special | mwi }
Dial-out mode for centrex-dial { directly | Direct dialing

Centrex users secondary }
Centrex number centrex-number centrex- Empty

number
Centrex prefix centrex-prefix centrex- No Centrex prefix
prefix configured
CFB forward-to service cfb cfbnum-value No CFB forward-to

number number configured
CFNR forward-to service cfnr cfnrnumtime- No CFNR forward-to

number and timeout value cfnrnum-value number configured
interval
CFO forward-to service cfo cfbnum-value No CFO forward-to

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

Hotline number service hotline hotline- No hotline number

time hotline-number configured
Anonymous call service-right anonymous- disable

service call enable
Call forwarding service-right call-diversion disable
service enable
Call hold service service-right call-hold enable

enable
Call transfer service service-right call-transfer enable

enable
Call conference service-right conf enable enable

service

NetEngine AR
Function/Service Command Default Setting
Call waiting service service-right cw enable disable
Hotline service service-right hotline disable

enable
Malicious call service-right malicious- disable

identification (MCID) call-identification enable
service
MWI service service-right mwi enable disable
MWI service mode mwi-mode { deferred | Deferred MWI service

immediate | combine } mode
Terminal portability service-right terminal- disable

service portability enable
Three-party service service-right three-party enable
enable NOTE
By default, the call
conference service has a
higher priority than the
three-party service. Before
using the three-party service,
run the service-right conf
disable command in the SIP
AG service data profile to
disable the call conference
service.
UUS1 service service-right uus1 enable disable
----End
3.8.4 Verifying the SIP AG User Configuration
Procedure
● Run the display voice SIP AGuser [ SIP AGuser-name ] command to verify
the configuration of the SIP AG user.
● Run the display voice SIP AGusergroup [ SIP AG-interface-id [ usergroup-
id ] ] command to verify information about user group.
● Run the display voice sipservicedata SIP AG-user-name telephone-number
command to verify service data in the SIP service data profile.
----End

NetEngine AR
3.9 Maintaining a SIP AG

This section describes how to maintain a SIP AG.
3.9.1 Clearing SIP AG Statistics
Context
NOTICE
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 voice view to clear global
SCTP statistics.
----End

This topic describes how to set physical voice interface parameters and SIP AG
3.10.1 (Optional) Setting Parameters for an FXS Interface

This section describes how to set parameters for an FXS interface.
Context
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
device, including physical attributes and electrical attributes.
Before setting parameters for an FXS interface, complete the following task:

NetEngine AR
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
port fxs slotid/subcardid/portid
The FXS interface view is displayed.
Step 4 Set physical attributes for the FXS interface.

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

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

● 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.

● Run:
electric current current-value
The current is set.

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

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

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

Run the display voice port fxs [ slotid/subcardid/portid ] command to verify the
FXS interface configuration.

NetEngine AR
3.10.2 Setting Parameters for a BRA Interface

Context
A basic rate access (BRA) interface connects to an ISDN phone. On a device, you
can enable the BRA interface Layer 2 monitoring, remote power supply, alarm
functions, set the working mode and Layer 1 activation mode of a BRA interface.
Before setting parameters for a BRA interface, complete the following tasks:
Procedure
Step 1 Run:
system-view

Step 2 Run:
set workmode slot slot-id bri bri-voice { nt-mode | te-mode }
The working mode of the 2BST interface card is configured.
NOTE
When a BRA interface connects to an ISDN phone, set the working mode of the 2BST interface
card to nt-mode. When a device is configured to work in SIP AG mode, the BRA interface can
only connect to an ISDN phone.
After this command is executed, the configured service is invalid.
After you run this command, the system asks you whether to reset the 2BST interface card. If
you enter Y, the system resets the 2BST interface card to make the configuration take effect. If
the system does not reset the 2BST interface card, run the reset slot command to reset the
2BST interface card.
Step 3 Run:
voice

Step 4 Run:
port bra slotid/subcardid/portid
The BRA interface view is displayed.

l2-monitor enable
BRA interface Layer 2 monitoring is enabled.

work-mode { p2p | p2mp }
The working mode is set.


NetEngine AR
remote-power enable
Remote power supply is enabled.

alarm enable
The alarm function is enabled.
----End

Run the display voice port bra [ slotid/subcardid/portid ] command to verify the
BRA interface configuration.
3.10.3 Setting Parameters for a VE1 Interface

This section describes how to set parameters for a voice E1 (VE1) interface.
Context
When the device works in SIP AG mode, the VE1 interface is used often to connect
to PBXs. The PBX can connect to the SIP AG through the PRA trunk (bound to a
VE1 interface). On the device, you can enable the CRC4 check, VE1 interface Layer
2 monitoring, and VE1 interface pulse code modulation (PCM) alarm functions,
and set the CRC alarm threshold and VE1 interface signaling mode on a VE1
interface.
NOTE
VE1 interface refers to the E1 interface in voice mode.
Before setting parameters for a VE1 interface, complete the following task:
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.
NOTE
After this command is executed, the configured service takes no effect.

After you using this command, the system displays a message asking you whether to reset the
E1 voice card. If you enter Y, the system resets the E1 voice card to make the configuration take
effect. Otherwise, run the reset slot command to restart the E1 voice card.

NetEngine AR
Step 3 Run:
voice
Step 4 Run:
port ve1 slotid/subcardid/portid
The VE1 interface view is displayed.

signal { CCS | CAS }
The signaling mode is set.
NOTE
When the PBX connects to the SIP AG through a PRA trunk, configure CCS on the VE1 interface
of the SIP AG.

access-mode { digital | direct }
The access digital section mode is set.

crc4 enable
The CRC4 check is enabled.

crc-alarm-threshold { es es-threshold | cses cses-threshold | dm dm- threshold }
The CRC alarm threshold is set.

l2-monitor enable
VE1 interface Layer 2 monitoring is enabled.

pcm-alarm
The PCM alarm function is enabled.

impedance impedance-value
The impedance of the VE1 interface is set.
----End

Run the display voice port ve1 [ slotid/subcardid/portid ] command to verify the
VE1 interface configuration.
3.10.4 (Optional) Setting System Parameters

NetEngine AR
Context
The AR G3 router working in SIP AG mode can exchange information with a
softswitch device using SIP to implement call services. 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:
Procedure
The following configurations are optional.
3.10.4.1 Setting Upper and Lower Thresholds for Hookflash Pressing

Parameters
Context
The upper and lower thresholds for hookflash pressing determine the hookflash
duration. 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.
The softswitch provides services after the hookflash button is pressed. The
following describes the common scenarios after a hookflash press:
● Hookflash pressed by a called party: If the called party UserA wants to

transfer an incoming call to UserB, UserA can press the hookflash and dial the
number of UserB.
● Hookflash pressed by a calling party: UserA calls UserB. UserB answers the
call and talks with UserA. UserA can press the hookflash and dial the number
of UserC after hearing a special dial tone.
After UserA presses the hookflash:

● If UserC is busy, UserA can press the hookflash and talk with UserB.
● If UserC does not respond for a long period of time, UserA can press the
hookflash and talk with UserB.
● If the phone of UserC rings, UserA hangs up and UserB hears the ringback
tone. UserC picks up the phone and talks with UserB.
● 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

NetEngine AR
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
3.10.4.2 Setting the MWI Mode
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 }
The MWI mode is set.
----End
3.10.4.3 Configuring the Codec Mode for Voice Services
Context
You can configure the G.711 codec mode for voice services so that the user device
can work in compliance with the local standard G.711, also known as Pulse Code

NetEngine AR
Modulation (PCM), is a commonly used waveform codec. G.711 defines two main
compression algorithms, the µ-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. μ-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
3.10.4.4 Setting Ringing Parameters
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 device to meet local standards.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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

By default, the frequency of the ringing current is 25 Hz.
● Run:
user-defined-ring ring-index { first-ring first-ring-period | first-interval first-interval | second-ring
second-ring-period | second-interval second-interval | third-ring third-ring-period | third-interval
third-interval }*
The cadence ratio of a specified user-defined ring type is set.

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

By default, the initial ringing function is disabled.
----End
3.10.4.5 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
3.10.4.6 Setting CLIP Parameters in Offhook State
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be
configured on the device so that the device 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
Step 3 Run the following commands as required.

● Run:
clip offhook ack-fsk-interval ack-fsk-interval

NetEngine AR
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.
● 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.
● Run:
clip offhook dtas-duration dtas-dur-value
The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.
● Run:
clip offhook dtas-level dtas-level
The level of the DT-AS in offhook state is configured.

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

● Run:
clip offhook mark-signal-bit mark-signal-bit
The number of bits of the FSK synchronization mask in offhook state is set.
----End
3.10.4.7 Setting CLIP Parameters in Onhook State
Context
When the CLIP service is registered, CLIP parameters in onhook state need to be
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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

● Run:
clip onhook dtas-duration dtas-dur-value

NetEngine AR
The duration of the DT-AS in onhook state is set.

● 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.
● Run:
clip onhook dtas-level dtas-level
The level of the DT-AS in onhook state is configured.

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

● Run:
clip onhook mark-signal-bit marksignal-bit
The number of bits of the FSK synchronization mask in onhook state is set.
----End
3.10.4.8 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
3.10.4.9 Loading the Voice Prompt File and Voice Service Logic File
Context
A prompt tone is the voice prompt heard by the calling and called users when the
calling user initiates a call. For example, when the calling user calls the called user,
the calling user hears the prompt tone indicating that the called user is busy. A
voice prompt file stores all voice prompt tones. The voice service logic file defines
the service exchange process.
The system software has the default voice prompt file and voice service logic file.
Voice prompt tone rules and services (including three-way conversation, three-
party service, teleconference, call transfer, and call waiting) have different usage

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scenarios in different countries. You can customize the voice prompt voice and
voice service logic file according to service requirements.
Procedure
Step 1 Make the voice prompt file and voice service logic file and generate a file with file
name extension .res or .cc.
NOTE
For details about generating the voice prompt file and voice service logic file with file name
extension .res or .cc, please contact technical support personnel.
You make the voice prompt file and voice service logic file into the same file or two different
files with file name extension .res or .cc and then load the file to the device.
Step 2 Upload or download the voice prompt file and voice service logic file to the
device's memory.
Step 3 Load the voice prompt file and voice service logic file. You can load the voice
prompt file and voice service logic file using either of the following methods.
● Method 1: Perform the following operation in the user view:
a. Run the load voice-package filename command to load the voice
prompt file/voice service logic file.
NOTE
When you use this method to load the voice prompt file and voice service logic file,
the files take effect immediately after being loaded and the voice prompt file and
voice service logic file for next startup are modified.
a. Run the startup voice-package filename command to load the voice
prompt file.
NOTE
To make the current voice prompt file and voice service logic file continue to take
effect and the new voice prompt file and voice service logic file take effect during
next startup, use this method.
b. Run the reboot to restart the device.
----End
3.10.4.10 Configuring a Media Port
Context
Different media ports may be used in different scenarios, so media port number
range needs to be set.
Procedure
Step 1 Run:
system-view

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Step 2 Run:
voice
Step 3 Run:
media-port start start-port-value end end-port-value
The media port number range is set.
By default, the system allocates a media port number range.
----End
3.10.4.11 Verifying the System Configuration
Procedure
● Run the display voice configuration command to verify the voice
configuration.
● Run the display voice user-defined-ring [ring-index ] command to verify
user-defined ring information.
● Run the display voice clip command to verify CLIP parameter settings.
● Run the display voice sip-reg-count-per-second command to check the
number of SIP Register messages initiated per second.
● Run the display startup command to check the loaded voice prompt file and
voice service logic file.
● Run the display voice media-bandwidth-control command to verify the CAC
configuration on the SIP AG and uplink bandwidth occupied by voice data.
----End
3.10.5 (Optional) Setting SIP Stack Parameters
Context
The Session Initiation Protocol (SIP) is a text-based signaling protocol. SIP
messages are classified into Request and Response messages. As an application
layer protocol, SIP establishes, modifies, or terminates multimedia sessions and
creates and controls multimedia sessions among two or more parties. SIP can
work with specified protocols to complete session setup and media negotiation,
such as 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).
Before setting SIP protocol stack parameters, complete the following task:

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Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sip
The SIP view is displayed.

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

● Run:
min-se min-se–period
The minimum value of the session timer is set.

● Run:
session-progress session-progress–period
The value of the unreliable provisional response retransmission timer is set.

● Run:
session-rel-progress session-rel-progress–period
The value of the reliable provisional response retransmission timer is set.

● Run:
t1 t1–period
The value of the T1 timer is set.

● Run:
t2 t2–period

● Run:
t4 t4–period

● Run:
td td–period
The value of the TD timer is set.

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

● Run:
field-header max-forwards max-forwards-value

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The value of the Max-forwards header field is set.

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

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

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

Run the display voice sip command to verify the SIP parameter settings.
3.10.6 (Optional) Setting DSP Parameters
Context
form.
voice communication quality, the DSP needs to process voice signals.
Before setting DSP parameters, complete the following task:
Procedure
3.10.6.1 Enabling Echo Cancellation on a DSP Channel
Context
A user may hear the user's echo in the phone receiver in a conversation. If the
delay exceeds 25 ms, the voice quality deteriorates.
Users can enable echo cancellation on a DSP channel to eliminate the echo and
improve voice quality.

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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.

By default, echo cancellation is disabled on the DSP channel.
----End
3.10.6.2 Enabling PLC 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

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The PLC function is enabled on the DSP channel.

By default, the PLC function is disabled.
----End
3.10.6.3 Enabling Silence Compression on a DSP Channel
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.

By default, silence compression is disabled on the DSP channel.
----End
3.10.6.4 Setting Fax Parameters
Context
This section describes how to set fax parameters.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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Step 3 Run:
dsp-attribute

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

● Run:

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

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

● Run:
V8 negotiation is enabled.
----End
3.10.6.5 Setting Jitter Buffer Parameters
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

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Step 4 Set parameters for the Jitter Buffer.

● Run:
jitter-buffer adapt-jb-threshold adapt-jb-threshold
The alarm threshold of the dynamic jitter buffer is set.

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

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

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

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

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

● Run:
jitter-buffer min-fixed-jb min-fixed-jb-value
The minimum value of the static jitter buffer is set.
----End
3.10.6.6 Setting the Payload Type Value
Context
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.

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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.

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

● Run:

● Run:

● Run:

● Run:
The NTE payload type value of a DSP channel is set.

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

● Run:
payload-type vbd vbd-value
The VBD payload type value of a DSP channel is set.
----End
3.10.6.7 Setting 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. This section describes how to set RTP control protocol (RTCP)
parameters.
Procedure
Step 1 Run:
system-view

Step 2 Run:

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voice

Step 3 Run:
dsp-attribute

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

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

● Run:
rtcp vqm enable
The Voice Quality Monitor (VQM) function is enabled.
----End
3.10.6.8 Setting 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

Step 3 Run:
dsp-attribute

● Run:
resource-threshold mode { priority | normal }

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The DSP resource control mode is configured.

● 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
3.10.6.9 Configuring 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.
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.

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

● Run:
DSP channels are prohibited.
----End

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3.10.6.10 Verifying the DSP Configuration
Context
After the configuration of DSP is complete, you can view the DSP parameter
settings including fax parameter settings and jitter buffer parameter settings.
Procedure
● Run the display voice dsp-attribute command to verify the DSP
configuration.
● Run the display voice dsp statistic command to check the DSP statistics.
● 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.
----End

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CLI-based Configuration Guide - Voice 4 H.248 AG Configuration
4 H.248 AG Configuration
4.1 Overview of the H.248 AG

This section describes the development and working principle of the H.248 AG and
its application in the voice over IP (VoIP) field.
4.2 Understanding the H.248 AG
4.3 H.248 Features Supported by the Device
This topic describes the H.248 features supported by the device, the related
specifications, and supplementary services.
4.4 Application Scenarios for the H.248 AG
4.5 Licensing Requirements and Limitations for the H.248 AG
This section describes the notes about configuring the H.248 AG.
4.6 Setting Voice Parameters on an H.248 AG
This section describes how to set voice parameters on an H.248 AG.
4.7 Configuring an H.248 AG Interface
You can configure an H.248 AG interface to connect H.248 AG users to the
softswitch for voice, data, and multimedia services.
4.8 Configuring an H.248 AG User
This section describes how to configure an H.248 AG user.
This topic describes how to set physical voice interface parameters and H.248 AG
4.1 Overview of the H.248 AG

This section describes the development and working principle of the H.248 AG and
its application in the voice over IP (VoIP) field.
Definition
H.248 is a recommendation from the International Telecommunication Union -
Telecommunication Standardization Sector (ITU-T) and was first published
(version 1) in June 2000. It is a gateway control protocol that is used by a Media

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Gateway Controller (MGC) to control Media Gateways (MGs) to enable the

transmission of various media data.
Compared with the Media Gateway Control Protocol (MGCP), H.248 has the
following advantages:
● Supports access of varied data types.
● Enhances the description capability of the MGCP, supports a broader range of
networks, and provides far greater flexibility when defining media control.
● Supports various transport layer protocols such as the User Datagram
Protocol (UDP) and Stream Control Transmission Protocol (SCTP).
Figure 4-1 shows the process of setting up and releasing an H.248-based VoIP
call.
Figure 4-1 Application of H.248 in VoIP
Softswitch
(MGC)
IP
H.248 Network H.248
MG-0 MG-1
POTS-0 POTS-1
Context
RTP stream
Call
Terms
● MG
A core network device for converting the media format of a network to the
required format of another network. It can process audio, video, and data

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services, and convert the media format in full duplex mode. It can also play
certain audio and video signals and provide the interactive voice response
(IVR) function and media conference.
● MGC
In the ITU-T H.323 architecture, the MGC is responsible for call control on
MGs.
● MGCP
This protocol defines a call control architecture for signaling processing and
session management in multimedia conferences. In this architecture, call
control is separated from data transmission. MGCP is a master/slave protocol
that allows a call control device such as an MCG or Call Agent (CA) to take
control of a specific port on a Media Gateway.
● UDP
UDP is one of the core members of the Internet protocol suite. It allows
applications on a computer to send datagrams to other computers on an IP
network. UDP is a message-based connectionless protocol that provides
unreliable connections for applications. It provides no guarantees for message
delivery. UDP messages may be discarded, repeated, delayed, or transmitted
in an incorrect sequence. When data is transmitted using UDP, the destination
does not actively inform the source of the transmission result.
● SCTP
A transport layer protocol used between the SCTP user application and a
connectionless packet network. In the SIGTRAN protocol stack, the upper
layer of SCTP is the adaptation module of the SCN signaling, for example,
M2UA and M2UA, and the lower layer of SCTP is the IP network. The SCTP
protocol delivers the higher reliability, optimum real-time performance, and
multi-homing feature for signaling transmission.
● H.248
A media gateway control protocol used for communications between the
media gateway controller (MGC) and the media gateway (MGW) in the
detached gateway architecture so that the MGC can control the MGW. In
Universal Mobile Telecommunications System (UMTS) networks, the interface
between the MGC and the MGW is the MC interface and the 3GPP defines
specific usage of H.248 over the MC interface.
● VoIP
VoIP is a group of technologies used for the delivery of voice communications
and multimedia sessions over IP networks, such as the Internet. The VoIP
service is cheaper and is available on smartphones, personal computers, and
Internet access devices. VoIP calls are free for Internet users.
4.2 Understanding the H.248 AG
4.2.1 H.248 Protocol Mechanism

When an H.248 user is added, the user must register personnel information with
the home network so that the user can make calls.

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Termination ID
A termination ID identifies a termination that is going to register or deregister a
service. The termination ID of each termination is unique. During service
configuration, the termination ID corresponding to each termination must be
configured on the MG and MGC. The root termination ID represents an entire MG.
The ServiceChange command executed on the root termination ID is effective on
an entire MG. The wildcard can be used.
H.248 Interface Registration Mechanism

The MG sends the ServiceChangeRequest command to inform the MGC that a
user or a group of users are about to register or deregister a service. After this
command is executed successfully, the termination status is changed to InService
or OutOfService. In addition, the MGC can unsolicitedly send the
ServiceChangeRequest command to request the MG to register or deregister a
service for a user or a group of users.
Figure 4-2 shows the registration flow of the H.248 AG.
● The MG sends the ServiceChangeRequest command to the MGC. In the
command, TerminationId is Root, Method is Restart, and
ServiceChangeReason is 901 (cold boot), 902 (warm boot), or 900 (in other
cases).
● The MGC sends the Reply message to the MG indicating the successful
registration.
● The MGC sends the Modify command to the MG requesting the MG to detect
the offhook of all users (al/of).
● The MG responds to the MGC with the Reply message.
Figure 4-2 H.248 AG registration flow
MG MGC
ServiceChangerequest (1)
Reply (2)
Modify (3)
Reply (4)
H.248 Interface Heartbeat Mechanism

After the registration is successful, the MG and MGC maintain communication by
sending each other the heartbeat message Notify (it/ito). By default, the

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heartbeat message is sent every 60s. The sending interval can be set within the
range of 5s to 655s.
After the MG sends the first heartbeat message to the MGC, if the MG does not
receive the heartbeat response from the MGC before the preset interface
heartbeat timer (for example, the length of three sending intervals) times out, the
MG sets the interface status to wait for response. Then, the MG keeps initiating a
registration with the MGC. If dual-homing is configured, the MG initiates
registration with the two MGCs alternatively.
H.248 Interface Deregistration Mechanism

Figure 4-3 shows the unsolicited deregistration flow of the MG.
● The MG sends the ServiceChangeRequest command to the MGC. In the
command, TerminationId is Root, Method is Restart, and
ServiceChangeReason is 905 (indicates that the termination is taken out of
service because of maintenance operation, and now the MG uses 905 to
initiate a deregistration request through command lines).
● The MGC sends the Reply message to the MG indicating a successful
deregistration.
Figure 4-3 Unsolicited deregistration flow of the MG
MG MGC
Reply (2)
Figure 4-4 shows the flow of the MGC unsolicitedly deregistering the MG.
● The MGC sends the ServiceChangeRequest command to the MG. In the
command, TerminationId is Root, Method is Forced, and
ServiceChangeReason is 905.
● The MG responds to the MGC with the Reply message. The MG supports the
registration and deregistration of not only an entire MG but also a single
termination. The service status of a single user can be changed through the
registration and deregistration of a single termination.

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Figure 4-4 Unsolicited deregistration flow of the MGC
MG MGC
Reply (2)
H.248 Interface Authentication Mechanism

Authentication is a security mechanism through which the MGC authenticates the
legality of MG users. The purpose of authentication is to prevent unauthorized
entities from establishing illegal calls or interfering with legal calls through the H.
248 or MGCP protocol. Authentication can be implemented only when it is also
supported by the MGC interconnected with the MG.
● In H.248, the implementation of authentication complies with RFC2402.
● MD5 is adopted as the encryption algorithm.
Figure 4-5 shows the authentication flow.
● The MG sends the ServiceChange command to register with the MGC. The
command contains the digital signature of the MG.
● After receiving the ServiceChange command, the MGC verifies the MG and
sends a reply.
● The MGC sends the Modify message to the MG. The message contains the
required algorithm ID and random number.
● The MG verifies the message sent by the MGC and sends a reply.
● The MGC authenticates the MG periodically.
● The MG sends replies to the MGC.

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Figure 4-5 Authentication flow
MG MGC
Reply (2)
Modify (3)
Reply (4)
Modify (5)
Reply (6)
4.2.2 H.248 AG Call Flow

Figure 4-6 shows the H.248 AG call flow.
1. MG-0 detects the offhook of user A0 and notifies the MGC of the offhook
event through the Notify command.
2. When receiving the offhook event, the MGC sends a digitmap to MG-0,
requests MG-0 to play the dial tone to user A0, and at the same time checks
for the digit collection event.
3. User A0 dials a telephone number. MG-0 collects the digits according to the
digitmap issued by the MGC and reports the result of digit collection to the
MGC.
4. The MGC sends the Add command to MG-0 for creating a context and adding
the termination and RTP termination of user A0 into the context.
5. After creating the context, MG-0 responds to the MGC. The response contains
the session description that provides the necessary information such as the IP
address and UDP port number for the peer end to send packets to MG-0.
6. The MGC sends the Add command to MG-1 for creating a context and adding
the termination and RTP termination of user A1 into the context. Then the
MGC issues the IP address/UDP port number of user A0 to user A1.
7. After creating the context, MG-1 responds to the MGC. The response contains
the session description that provides the necessary information such as the IP
address and UDP port number for the peer end to send packets to MG-1.
8. The MGC sends the Modify command to enable user A1's phone to ring and
enable user A0 to play the ring back tone.
9. When detecting the offhook of user A1, MG-1 reports the offhook event to
the MGC. The MGC sends the Modify command to stop the ring back tone of
user A0 and the ringing of user A1.

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10. The MGC sends the session description of MG-1 to user A0 through the
Modify command. Then a call is set up between user A0 and user A1.
11. MG-0 detects the onhook of user A0 and notifies the MGC of the onhook
12. The MGC sends the Modify command to MG-0 and MG-1 respectively to
modify the RTP mode to receive-only.
13. The MGC sends the Modify command to MG-1 requesting MG-1 to play the
busy tone to user A1. At the same time, the MGC checks for the onhook
event.
14. The MGC sends the Subtract command to MG-0, requesting MG-0 to release
the resources that are occupied by the call of user A0.
15. MG-1 detects the onhook of user A1 and notifies the MGC of the onhook
16. The MGC sends the Subtract command to MG-1, requesting MG-1 to release
the resources that are occupied by the call of user A1.
17. The call between user A0 and user A1 is terminated, and all the resources
occupied by the call are released.

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Figure 4-6 H.248 AG call flow
Softswitch
User A0 MG-0 MGC MG-1 User A1
Off-hook Notify
Notify_Reply
Modify
Play dial-tone Modify_Reply
Dial the first digit of a number
Stop playing dial-tone
Dial the remaining digits
Match digitmap
Notify
Notify_Reply
Add
Add_Reply
Add
Add_Reply
Modify
Modify Ringing
Play ringback tone Modify_Reply
Modify_Reply Off-hook
Modify
Modify
Stop playing ringback tone Modify_Reply
Modify_Reply
Call set up
On-hook Notify
Notify_Reply
Modify
Modify_Reply
Subtract
Subtract_Reply
Modify
Modify
Modify_Reply
Modify_Reply Busy-tone
On-hook
Notify
Notify_Reply
Subtract
Subtract_Reply
Modify
Modify_Reply
4.3 H.248 Features Supported by the Device

This topic describes the H.248 features supported by the device, the related
specifications, and supplementary services.

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Using an H.248 AG as a Voice Gateway

As shown in Figure 4-7, an H.248 AG is used as an IP voice gateway to implement
communication between an IP network and a public switched telephone network
(PSTN) or integrated services digital network (ISDN). The details are as follows:
● A Plain Old Telephone Service (POTS) user connects to the H.248 AG through
the PSTN. The H.248 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 H.248 AG of the called party through the
IP network. The H.248 AG 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.
Figure 4-7 H.248 AG functioning as a voice gateway
Voice Services Supported by the H.248 AG

The H.248 AG supports the basic voice service and developed services such as the
three-party service and call waiting service. These services need to be configured
on the softswitch.
Service Type Introduction
Basic voice service The basic voice service provides call connections,
including intra-office calls, local calls, national long-
distance calls, international long-distance calls, and
transit calls.
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 make a three-party
conversation or talk to the other two parties separately.

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Call waiting If user C calls user A when user A is talking with user B,
user A hears a call waiting tone indicating that there is
an incoming call.
MWI service The message waiting indicator (MWI) service allows a

user to read unread or leave messages. When the called
user is busy, the MWI is on indicating that there are
messages.
Malicious call The MCID service allows users to perform certain

identification operations to find the phone number of an attacker that
(MCID) service initiates malicious calls.
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.
Call conference The call conference service allows more than three
service parties to talk to each other.
Calling line The CLIP service displays the calling number in onhook
identification state or offhook state (for call waiting). The displayed
presentation (CLIP) information includes the phone number, name, date, and
service time.
Calling line The CLIR service shields the calling number on the
identification terminal of a called party.
restriction (CLIR)
service
Distinctive ringing The distinctive ringing service plays different ring tones
service for incoming calls from different calling parties.
Differentiated The differentiated ringback tone service plays different

ringback tone ringback tones for different users.
service
Advice of charge The AoC service displays the charge rate, fee notification
(AoC) service during a call, and total fee of the call.
Emergency call If the H.248 AG detects an emergency call, it inserts an

process emergency call flag into the H.248 message.
Completion of Calls The CCBS service enables the H.248 AG to monitor the
to Busy Subscriber called party status when the called party is busy. When
(CCBS) service the called party is idle, the H.248 AG notifies the calling
party so that the calling party can determine whether to
make a call to the called party again.
Multiple MSN Multiple MSN numbers can be configured on a POTS

numbers on a POTS interface.
interface

NetEngine AR
Hotline service The hotline service is classified into:

● Instant hotline service: After a user picks up a phone,
the H.248 AG dials the hotline number for the user.
● Delayed hotline service: If a user does not dial any
number within the specified period of time after
picking up a phone, the H.248 AG dials the hotline
number for the user.
Anonymous call The anonymous call service prevents the called party
service from viewing information about incoming calls.
4.4 Application Scenarios for the H.248 AG

H.248 AG Functioning as the Voice Gateway
As shown in Figure 4-8, an H.248 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:
● The H.248 AG connects small-scale and distributed enterprise users to analog
phones or fax machines to implement voice communication.
● The H.248 AG is registered with the SoftSwitch. Then the SoftSwitch provides
and controls user services on the H.248 AG.
Figure 4-8 H.248 AG functioning as the voice gateway
Softswitch
H.248AG
POTS Modem FAX POTS

IP link

NetEngine AR
4.5 Licensing Requirements and Limitations for the H.

248 AG
This section describes the notes about configuring the H.248 AG.

None
NOTE
LTE4EA.




NetEngine AR
Cards-Voice Card.
E1/T1 Card.
Feature Limitations
None.
4.6 Setting Voice Parameters on an H.248 AG

This section describes how to set voice parameters on an H.248 AG.
4.6.1 Configuring the Device to Work in H.248 AG Mode
Context
You can configure the device to work in H.248 AG, SIP AG, or PBX mode. Before
configuring H.248 AG service features, configure the device to work in H.248 AG
mode. Before the configuration, run the display voice service-mode command to
query the working mode of the device. If the working mode is not H.248 AG, you
need to run the display current-configuration command to query the
configurations and then delete all configurations in the voice view. Then, set the
working mode of the device to H.248 AG. If the device is working in H.248 AG
mode, ignore this configuration.
Before configuring the device to work in H.248 AG mode, complete the following
task:
● Configuring IP addresses and routing protocols for interfaces to ensure
connectivity

NetEngine AR
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
service-mode h248ag
The device is configured to work in H.248 AG mode.

Step 4 Run:
quit

Step 5 Run:
quit
The user view is displayed.

Step 6 Run:
save
The current configuration is saved.

Step 7 Run:
reboot
The device is restarted.
NOTE
After the router is configured to work in H.248 AG mode, save the configuration and restart
the router to make the configuration take effect.
----End

Run the display voice service-mode command to verify the working mode of the
device.
<Huawei> display voice service-mode
The voice service mode is H248AG.
After the configuration is complete, the device works in H.248 AG mode.
Context
An H.248 AG interface can be used only after the interface IP address is added to
the media and signaling IP address pools. An H.248 AG interface must obtain

NetEngine AR
media and signaling IP addresses from media and signaling IP address pools
respectively.
The media and signaling IP address pools can be the same.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
voip-address media interface interface-type interface-number { ip-address | dynamic }
The IP address of the specified interface is added to the media IP address pool.
Step 4 Run:
voip-address signalling interface interface-type interface-number { ip-address | dynamic }
The IP address of the specified interface is added to the signaling IP address pool.
When configuring media and signaling IP address pools:
● To specify the ip-address parameter, ensure that an IP address has been
configured for the specified interface.
● To specify the dynamic parameter, ensure that an IP address has been
dynamically assigned to the specified interface.
----End

Run the display voice voip-address command to verify the configured signaling
and media IP address pools.
4.7 Configuring an H.248 AG Interface

You can configure an H.248 AG interface to connect H.248 AG users to the
softswitch for voice, data, and multimedia services.
Context
If users want to connect to the softswitch for voice, data, and multimedia services,
an H.248 AG interface must be configured for registration on the softswitch. H.248
AG users connect to the softswitch through the H.248 AG interface to use voice,
To allow the H.248 AG and softswitch to exchange media and signaling streams,
the media and signaling IP addresses, signaling port number, and transmission
protocol need to be configured for the H.248 AG interface.

NetEngine AR
Before configuring an H.248 AG interface, complete the following task:
● 4.6.1 Configuring the Device to Work in H.248 AG Mode
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
H248ag mgid
An H.248 AG interface is created, and the H.248 AG interface view is displayed.
Step 4 Set mandatory parameters on the H.248 AG interface.
Parameter Command Description
Media IP address mg-media-ip1 { mg-media-ip | The media IP

addr-name addr-name } address of the H.
248 AG interface
must be obtained
from the media IP
address pool.
Signaling IP address mg-signal-ip { ip_addr | signal- The signaling IP

and signaling port addr-name name } mgport port address of the H.
number 248 AG interface
must be obtained
from the signaling
IP address pool.
By default, no
signaling port
number is
configured for an H.
248 AG interface.
Parame Static IP primary-mgc-address1 static By default, no static

ters of address of ip_addr port [ primary-mgc- IP address is
the the primary address2 ip_addr2 ] configured for the
primar MGC server primary MGC
y and server.
second
ary
MGC
servers

NetEngine AR
DNS primary-mgc-address1 primary- By default, no DNS

domain mgc-domain-name domain name or
name and domainname port port number is
port configured for the
number of primary MGC
the primary server.
MGC server
Static IP secondary-mgc-address1 static By default, no static

address of ip_addr port [secondary-mgc- IP address is
the address2 ip_addr2] configured for the
secondary secondary MGC
MGC server server.
Domain secondary-mgc-address1 By default, no

name and secondary-mgc-domain-name domain name or
port domainname port port number is
number of configured for the
the secondary MGC
secondary server.
MGC server
Step 5 Run:
reset coldstart
The H.248 AG is reset.
NOTICE
● Exercise caution when you run this command because resetting an H.248 AG
affects running services.
● Do not frequently modify H.248 AG interface parameters. To make the
configuration take effect, you must reset the H.248 AG.
● The standby MGC server is optional. You must configure the standby MGC
server only when dual-homing is used.

local-digitmap name {bodyvalue | append bodyvalue}
Configure a digitmap for the H.248 AG.
Step 7 (Optional) Set optional parameters on the H.248 AG interface.

NOTE
The authentication mode on the H.248 MG interface is defined by the standard protocol, and
the encryption algorithm cannot be changed at random. A change of the encryption algorithm
will cause a failure to connect to the MGC.

NetEngine AR
Transmission protocol transfer protocol By default, an H.

248 AG uses the
User Datagram
Protocol (UDP) as
the transmission
protocol.
Profile negotiation profile-negotiation profile- By default, the

negotiation profile negotiation
is disabled for an H.
248 AG interface.
Authentication auth { initial-key cipher | auth- The default

parameters mgid auth_mgid } username and
password are
available in AR
Router Default
Usernames and
Passwords
(Enterprise
Network or
Carrier). If you
have not obtained
the access
permission of the
document, see Help
on the website to
find out how to
obtain it.
Timer value digitmap-timer { start By default, the

startvalue | short shortvalue | value of a digitmap
long longvalue | duration long timer is 20s,
durationvalue } the value of a
digitmap short
timer is 4s, and the
value of a digitmap
start timer is 20s.
Software parameters mg-software parameter name Different software

value parameters may use
different value
ranges. For details,
see mg-software
parameter.
Default ring type mg-ringmode default-ringpara By default, ringing

{ short-ring short-ring | aoce- parameters of an H.
ring aoce-ring } * 248 AG are empty.

NetEngine AR
Default ring mg-ringmode default-mgcpara The default value is

parameters for the { ring ring-default-mgcpara | 255.
MGC mwi mwi-default-mgcpara } *
Ring mode mapping mg-ringmode index [ cadence- The default value of

ring cadencering-value | initial- cadence-ring is 0.
ring initialring-value ] * The default value of
initial-ring is 4.
H.248 message MIDType midtypeval The H.248 message

identifier type identifier type
needs to be
specified.
The value can be
ip4Address,
domainName, or
deviceName.
ip4Address.
TID format of the H. tid-format { rtp | pstn | bra | The TID format of
248 AG interface pra } { prefix prefix-string | tid- the H.248 AG
template index } interface needs to
be specified.
NOTE
Make sure the tid is
configured as the
softswitch.
Otherwise, you will
fail to configure the
h.248 users.
IP address and port stand-alone-mgc-ip1 ip_addr IP address and port

number of the [ stand-alone-mgc-ip2 number of the
standalone MGC ip_addr2 ] stand-alone-mgc- stand-alone MGC
port port need to be
specified.
By default, the IP
address and port
number are empty.
----End

Run the display voice h248ag [ agid { running | config } ] command to verify the
H.248 AG interface configuration.

NetEngine AR
4.8 Configuring an H.248 AG User

This section describes how to configure an H.248 AG user.
Before configuring an H.248 AG user, complete the following task:
● 4.7 Configuring an H.248 AG Interface
4.8.1 Configuring a POTS User

Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
h248aguser slotid/subcard/portid [ agid agid [ terminalid terminalid ] * ]
A POTS user is created and the POTS user view is displayed.

Step 4 Set optional parameters for the POTS user.
User phone telno telno This number is used for

number auto-switching only.
User priority priority priority The priority of the POTS

user needs to be
specified.
The value can be cat1,
cat2, or cat3.
The default value is cat3.

NetEngine AR
User type line-type line-type Select an option

depending on the actual
user type.
The options are as
follows:
● DEL: Direct Exchange
Lines
● ECPBX: Earth Calling
PBX
● LCPBX: Loop Calling
PBX
● PayPhone: Pay Phone
DEL users' lines are
directly connected to
device ports. PBX users'
lines are connected to
device ports through the
PBX. Pay phone users'
lines are connected to
the accounting terminal
through the device ports.
The default value is DEL.
Sequence in clip-transmission-sequence clip- This parameter is

which the transmission-sequence optional. Set this
calling parameter when you
number is need to configure the
displayed sequence of displaying
the calling number of a
PSTN user.
The options are as
follows:
● after-ring: Configures
the calling number to
be displayed after
ringing.
● before-ring:
Configures the calling
number to be
displayed before
ringing.
after-ring.
Power-off dc-time dc-time-value The default value is 10

interval ms.

NetEngine AR
Single-tone detect-ansbar-by-single-tone The default value is

ANSbar { enable | disable } disable.
signal
detection
DSP chip dsp-input-gain dsp-input-gain-value The default value is 0 dB.

input gain
DSP chip dsp-output-gain dsp-output-gain- The default value is 0 dB.

output gain value
DSP dsp-template dsp-template-name By default, no DSP
template template name is
name configured.
Service endservice { defer [ maxcobtime By default, services of a

termination maxcobtime-value ] | immediate } user are not terminated.
FSK call fsk-mode { BELL_202 | CCITT_V23 | The default value is

display mode NTT } BELL_202.
FSK delay fsk-time fsk-time-value The default value is 10

ms.
TAS mode of tas-pattern { NO-TAS | DT-AS } The default value is NO-

the calling TAS.
line
identification
presentation
(CLIP)
Calling clip-format { sdmffsk | mdmffsk | The default number

number dtmf | r15 | etsi } format is FSK simple data
format format.
Voice quality vqe-agc { enable | disable } The default value is

enhancemen disable.
t (VQE)
function
Target VQE vqe-agc-level levelvalue The default value is 13.

level
VQE noise vqe-sns { enable | disable } The default value is

suppression disable.
Noise vqe-sns-level levelvalue The default value is 12

suppression dB.
level

NetEngine AR
H.248 AG's bell-ans-flag { enable | disable } The default value is

capability to disable.
support the This command is valid for
BELL ANS only FXS ports.
flag
----End
4.8.2 Configuring an ISDN User
Before configuring an ISDN user, complete the following task:
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
sigtran
The SIGTRAN view is displayed.

Step 4 Run:
iua-linkset linksetno [ mgid mgid [ jointly-work jointly-work ] | trafficmode trafficmode | pendingtime
pendingtime | iid-map iid-map | braprefix braprefix | brasuffix brasuffix | praprefix praprefix | prasuffix
prasuffix | cs-mode cs-mode ] *
An IUA link set is added.

Step 5 Run:
iua-link linkno linksetno localport { localipaddr | local-addr-name local-addr-name-value } remoteport
remoteipaddr1 [ remoteipaddr2 remoteipaddr2 | priority priority | jointly-work-with jointly-work-with ] *
An IUA link is added.

Step 6 Run:
quit
Exit the voice view.

Step 7 Run:
H248aguser slotid/subcard/portid [agid agid [ terminalid terminalid linksetno linksetno interfaceid
interfaceid ] * ]

NetEngine AR
The ISDN user is created and the user view is displayed.
Step 8 Run:
uni-report flag
The flag for reporting a UNI fault is set.
Step 9 Run:
auto-resume-limit limit
The maximum number of automatic recovery attempts from deterioration is set.
Step 10 Run:
power-dialer limit
The detection of malicious calling users is enabled.
----End

Run the display voice h248aguser bra command to verify the ISDN BRA user
configuration.
Run the display voice h248aguser pra command to verify the ISDN PRA user
configuration.

This topic describes how to set physical voice interface parameters and H.248 AG
4.9.1 (Optional) Setting Parameters for an FXS Interface

This section describes how to set parameters for an FXS interface.
Context
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
device, including physical attributes and electrical attributes.
Before setting parameters for an FXS interface, complete the following task:
Procedure
Step 1 Run:
system-view

NetEngine AR
Step 2 Run:
voice
Step 3 Run:
port fxs slotid/subcardid/portid
The FXS interface view is displayed.
Step 4 Set physical attributes for the FXS interface.

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

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

● 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.

● Run:
electric current current-value
The current is set.

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

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

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

Run the display voice port fxs [ slotid/subcardid/portid ] command to verify the
FXS interface configuration.
4.9.2 Setting Parameters for a BRA Interface
Context
A basic rate access (BRA) interface connects to an ISDN phone. On the device, you
can enable the BRA interface Layer 2 monitoring, remote power supply, and alarm
functions, and set the working mode and Layer 1 activation mode on a BRA
interface.

NetEngine AR
Before setting parameters for a BRA interface, complete the following task:
Procedure
Step 1 Run:
system-view

Step 2 Run:
set workmode slot slot-id bri bri-voice { nt-mode | te-mode }
The working mode of the 2BST interface card is configured.
NOTE
When the BRA interface connects to ISDN phones, set the working mode of the 2BST interface
card to nt-mode. When the device is configured to work in H.248 AG mode, the BRA interface
can only connect to ISDN phones.
2BST interface card. If you enter Y, the system resets the 2BST interface card to make the
configuration take effect. Otherwise, run the reset slot command to restart the 2BST interface
card.
Step 3 Run:
voice

Step 4 Run:
port bra slotid/subcardid/portid
The BRA interface view is displayed.

l2-monitor enable
BRA interface Layer 2 monitoring is enabled.

work-mode { p2p | p2mp }
The working mode is set.

remote-power enable
Remote power supply is enabled.

alarm enable
The alarm function is enabled.
----End

NetEngine AR

Run the display voice port bra [ slotid/subcardid/portid ] command to verify the
BRA interface configuration.
4.9.3 Setting Parameters for a VE1 Interface

This section describes how to set parameters for a voice E1 (VE1) interface.
Context
When the device works in H.248 AG mode, the VE1 interface is used often to
connect to PBXs. The PBX can connect to the H.248 AG through the PRA trunk
(bound to a VE1 interface). On the device, you can enable the CRC4 check, VE1
interface Layer 2 monitoring, and VE1 interface pulse code modulation (PCM)
alarm functions, and set the CRC alarm threshold and VE1 interface signaling
mode on a VE1 interface.
NOTE
VE1 interface refers to the E1 interface in voice mode.
Before setting parameters for a VE1 interface, complete the following task:
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.
NOTE

E1 voice card. If you enter Y, the system resets the E1 voice card to make the configuration take
effect. Otherwise, run the reset slot command to restart the E1 voice card.
Step 3 Run:
voice
Step 4 Run:
port ve1 slotid/subcardid/portid
The VE1 interface view is displayed.

NetEngine AR
Step 5 Run:
display voice port ve1 [ state slotid/subcardid/portid | slotid/subcardid/portid ]
The signaling mode of the VE1 interface is displayed. If the signaling mode is not
CCS, run the signal ccs command to change the signaling mode of the VE1
interface to CCS
NOTE
In H.248 AG mode, the signaling mode of the VE1 interface must be CCS.

access-mode { digital | direct }
The access digital section mode is set.

crc4 enable
The CRC4 check is enabled.

crc-alarm-threshold { es es-threshold | cses cses-threshold | dm dm- threshold }
The CRC alarm threshold is set.

l2-monitor enable
VE1 interface Layer 2 monitoring is enabled.

pcm-alarm
The PCM alarm function is enabled.

impedance impedance-value
The impedance of the VE1 interface is set.
----End

Run the display voice port ve1 [ slotid/subcardid/portid ] command to verify the
VE1 interface configuration.
4.9.4 (Optional) Setting System Parameters
Context
The AR G3 router working in SIP AG mode can exchange information with a
softswitch device using SIP to implement call services. Different countries and
regions use different voice parameter standards; therefore, set voice parameters
on the SIP AG in accordance with local standards.

NetEngine AR
Before setting system parameters, complete the following task:
Procedure
4.9.4.1 Setting Upper and Lower Thresholds for Hookflash Pressing

Parameters
Context
The upper and lower thresholds for hookflash pressing determine the hookflash
duration. 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.
The softswitch provides services after the hookflash button is pressed. The
following describes the common scenarios after a hookflash press:
● Hookflash pressed by a called party: If the called party UserA wants to
transfer an incoming call to UserB, UserA can press the hookflash and dial the
number of UserB.
● Hookflash pressed by a calling party: UserA calls UserB. UserB answers the
call and talks with UserA. UserA can press the hookflash and dial the number
of UserC after hearing a special dial tone.
After UserA presses the hookflash:
● If UserC is busy, UserA can press the hookflash and talk with UserB.
● If UserC does not respond for a long period of time, UserA can press the
hookflash and talk with UserB.
● If the phone of UserC rings, UserA hangs up and UserB hears the ringback
tone. UserC picks up the phone and talks with UserB.
● 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.

NetEngine AR
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
4.9.4.2 Setting the MWI Mode
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 }
The MWI mode is set.
----End
4.9.4.3 Configuring the Codec Mode for Voice Services
Context
You can configure the G.711 codec mode for voice services so that the user device
can work in compliance with the local standard G.711, also known as Pulse Code
Modulation (PCM), is a commonly used waveform codec. G.711 defines two main
compression algorithms, the µ-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. μ-law encoding takes a 15-bit signed linear
audio sample as input.

NetEngine AR
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
4.9.4.4 Setting Ringing Parameters
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 a device to meet local standards.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run the following commands based on user requirements.
● Run:
ring frequency { 16hz | 25hz | 50hz }
The frequency of the ringing current is set.
By default, the frequency of the ringing current is 25 Hz.
● Run the ring voltage voltage-value command to set the ringing voltage.
By default, the ringing voltage is 45 Vrms.
● Run:
user-defined-ring ring-index { first-ring first-ring-period | first-interval first-interval | second-ring
second-ring-period | second-interval second-interval | third-ring third-ring-period | third-interval
third-interval }*
The cadence ratio of a specified user-defined ring type is set.
● Run:
stop-initial-ring { enable | disable }
The initial ringing function is enabled or disable.

NetEngine AR
By default, this function is enabled, which indicates that the initial ring is
stopped.
----End
4.9.4.5 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
4.9.4.6 Setting CLIP Parameters in Offhook State
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

● 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.

NetEngine AR
● 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.
● Run:
clip offhook dtas-duration dtas-dur-value
The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.
● Run:
clip offhook dtas-level dtas-level
The level of the DT-AS in offhook state is configured.

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

● Run:
clip offhook mark-signal-bit mark-signal-bit
The number of bits of the FSK synchronization mask in offhook state is set.
----End
4.9.4.7 Setting CLIP Parameters in Onhook State
Context
When the CLIP service is registered, CLIP parameters in onhook state need to be
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

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

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

● Run:

NetEngine AR
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.
● Run:
clip onhook dtas-level dtas-level
The level of the DT-AS in onhook state is configured.

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

● Run:
clip onhook mark-signal-bit marksignal-bit
The number of bits of the FSK synchronization mask in onhook state is set.
----End
4.9.4.8 Loading the Voice Prompt File and Voice Service Logic File
Context
A prompt tone is the voice prompt heard by the calling and called users when the
calling user initiates a call. For example, when the calling user calls the called user,
the calling user hears the prompt tone indicating that the called user is busy. A
voice prompt file stores all voice prompt tones. The voice service logic file defines
the service exchange process.
The system software has the default voice prompt file and voice service logic file.
Voice prompt tone rules and services (including three-way conversation, three-
party service, teleconference, call transfer, and call waiting) have different usage
scenarios in different countries. You can customize the voice prompt voice and
voice service logic file according to service requirements.
Procedure
Step 1 Make the voice prompt file and voice service logic file and generate a file with file
name extension .res or .cc.
NOTE
For details about generating the voice prompt file and voice service logic file with file name
extension .res or .cc, please contact technical support personnel.
You make the voice prompt file and voice service logic file into the same file or two different
files with file name extension .res or .cc and then load the file to the device.
Step 2 Upload or download the voice prompt file and voice service logic file to the
device's memory.
Step 3 Load the voice prompt file and voice service logic file. You can load the voice
prompt file and voice service logic file using either of the following methods.
a. Run the load voice-package filename command to load the voice
prompt file/voice service logic file.

NetEngine AR
NOTE
When you use this method to load the voice prompt file and voice service logic file,
the files take effect immediately after being loaded and the voice prompt file and
voice service logic file for next startup are modified.
a. Run the startup voice-package filename command to load the voice
prompt file.
NOTE
To make the current voice prompt file and voice service logic file continue to take
effect and the new voice prompt file and voice service logic file take effect during
next startup, use this method.
b. Run the reboot to restart the device.
----End
4.9.4.9 Configuring a Media Port
Context
Different media ports may be used in different scenarios, so media port number
range needs to be set.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
media-port start start-port-value end end-port-value
The media port number range is set.
By default, the system allocates a media port number range.
----End
4.9.4.10 Verifying the System Configuration
Procedure
● Run the display voice configuration command to verify the voice
configuration.
● Run the display voice user-defined-ring [ring-index ] command to check
user-defined ring information.

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● Run the display voice clip command to verify the CLIP parameter settings.
● Run the display startup command to check the loaded voice prompt file and
voice service logic file.
● Run the display voice media-bandwidth-control command to verify the CAC
configuration on the SIP AG and uplink bandwidth occupied by voice data.
----End
4.9.5 (Optional) Setting DSP Parameters
Context
form.
voice communication quality, the DSP needs to process voice signals.
Before setting DSP parameters, complete the following task:
Procedure
4.9.5.1 Enabling Echo Cancellation on a DSP Channel
Context
A user may hear the user's echo in the phone receiver in a conversation. If the
delay exceeds 25 ms, the voice quality deteriorates.
Users can enable echo cancellation on a DSP channel to eliminate the echo and
improve voice quality.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

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

By default, echo cancellation is disabled on the DSP channel.
----End
4.9.5.2 Enabling PLC 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.

By default, the PLC function is disabled.
----End
4.9.5.3 Enabling Silence Compression on a DSP Channel
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

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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.
By default, silence compression is disabled on the DSP channel.
----End
4.9.5.4 Setting Fax Parameters
Context
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.

● Run:
fax redundancy-t4 redundancy-t4-value

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● Run:

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

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

● Run:
V8 negotiation is enabled.
----End
4.9.5.5 Setting Jitter Buffer Parameters
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.
● Run:
jitter-buffer adapt-jb-threshold adapt-jb-threshold
The alarm threshold of the dynamic jitter buffer is set.

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

● Run:
jitter-buffer init-fixed-jb normal-fixed-jb-value

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The initial value of the static jitter buffer is set.

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

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

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

● Run:
jitter-buffer min-fixed-jb min-fixed-jb-value
The minimum value of the static jitter buffer is set.
----End
4.9.5.6 Setting the Payload Type Value
Context
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.

● Run:

● Run:

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● Run:

● Run:

● Run:
The NTE payload type value of a DSP channel is set.

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

● Run:
payload-type vbd vbd-value
The VBD payload type value of a DSP channel is set.
----End
4.9.5.7 Setting 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. This section describes how to set RTP control protocol (RTCP)
parameters.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
dsp-attribute

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

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

● Run:
rtcp vqm enable
The Voice Quality Monitor (VQM) function is enabled.
----End
4.9.5.8 Setting 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
Step 3 Run:
dsp-attribute

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

● 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

NetEngine AR
4.9.5.9 Configuring 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.
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.

● Run:
loop-back loopback-type channel
The loopback function is configured on a DSP channel.
● Run:
DSP channels are prohibited.
----End
4.9.5.10 Verifying the DSP Configuration
Context
After the DSP configuration is complete, you can verify the DSP parameter
settings including the fax parameter settings and jitter buffer parameter settings.
Procedure
● Run the display voice dsp-attribute command to verify the DSP
configuration.

NetEngine AR
● Run the display voice dsp statistic command to check the DSP statistics.
● 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.
----End
4.9.6 (Optional) H.248 Common Configuration

Users can configure H.248 common services based on the service requirements.
4.9.6.1 Setting RFC 2198 Parameters
Context
You can configure RFC 2198 upper packet loss limit and number of redundant
packets.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
rfc2198 smart-parameters section-value { up-threshold up-threshold-value | redundant-num redundant-
num-value }*
RFC 2198 parameters are set.
----End
Result
Run the display voice rfc2198 smart-parameters command to view the RFC 2198
configurations.
4.9.6.2 Setting FAX/Modem Parameters
Context
An MG supports the fax and modem services. Related parameters must be
configured to use either service. If both services are required, the common
parameters of the two services must be configured.
Procedure
Step 1 Run:
system-view

NetEngine AR

Step 2 Run:
voice

Step 3 Run:
fax-modem fax { is-port+2 portvalue | transmission-mode mode | flow flowvalue }
Fax parameters are set.

Step 4 Run:
fax-modem modem { eventmode eventmode | tranmode tranmode }
Modem parameters are set.

Step 5 Run:
fax-modem common { negomode negomode-value | packet-interval-10ms packet-interval-10ms-value |
transevent transevent-value | vbd-codec vbd-codec-value | vbd-pt-type vbd-pt-type-value }
Common parameters are set.
----End
Result
Run the display voice fax-modem command to view the MG configurations.
4.9.6.3 Configuring a TID Profile
Context
You can customize a TID profile.
Procedure
Step 1 Run:
system-view

Step 2 Run:
voice

Step 3 Run:
tid-template templateid format formatstring para-list string name namestring
A custom TID profile is created.
----End
Result
Run the display voice tid-template [ index ] command to view the information
about the custom TID profile.

NetEngine AR
4.9.6.4 Configuring a Local Digitmap
Context
After configuring a local digitmap, you can map telephone numbers based on the
digitmap.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
local-digitmap name {bodyvalue | append bodyvalue }
The local digitmap is created.
----End
Result
Run the display voice local-digitmap [ digitmapname ] command to view the
local digitmap.
4.9.7 Configuration Examples for the H.248 AG

This section provides configuration examples for the H.248 AG.
4.9.7.1 Example for Configuring the Voice Service
The H.248 AG allocates the terminal ID to the enterprise, users are registered with
the softswitch based on the terminal prefix and terminal ID, and the softswitch
locates users and connects calls based on the terminal prefix and terminal ID.
Figure 4-9 shows the networking.
The configuration roadmap is as follows:
● Configure media IP address 1 of 192.168.1.2 for the MG interface.

● Configure the signaling address of 192.168.1.2 and port number of 2949 for
the H.248 interface.
● Configure the static IP address of 192.168.1.110 and port number of 2944 for
the active MGC.

NetEngine AR
Figure 4-9 Networking of the voice service

Softswitch
User C
H.248AG
Port 1/0/0 Port 1/0/1
User A User B
Data Plan
The data plan here is used for reference. The actual configuration is determined
based on the user device and softswitch.
Table 4-2 Data Plan

Terminal Prefix Ter User Name Remarks
min
ati
on
ID
A 0-3 1/0/0-1/0/3 Softswitch user
NOTICE
The terminal prefix and terminal ID of the user must be the same as those on the
softswitch; otherwise, the H.248 user may fail to be configured. Here, the terminal
ID is the same as that on the softswitch.
Procedure
Step 1 Configure the device to work in H.248 AG mode.
[Huawei] voice
[huawei-voice] service-mode h248ag
<Huawei> save
<Huawei>reboot

NetEngine AR
Step 2 # Set the Ethernet IP address of interface0/0/0 to 192.168.1.2, and add IP address
192.168.1.2 to media and signaling IP address pools.
[Huawei] voice
Step 3 Configure an H.248 AG interface.

[Huawei-voice] h248ag 1
[Huawei-voice-h248ag-1] mg-media-ip1 192.168.1.2
[Huawei-voice-h248ag-1] mg-signal-ip 192.168.1.2 mgport 2949
[Huawei-voice-h248ag-1] primary-mgc-address1 static 192.168.1.110 2944
[Huawei-voice-h248ag-1] reset coldstart
[Huawei-voice-h248ag-1] quit
Step 4 Configure an H.248 user.

[Huawei-voice] h248aguser 1/0/0 agid 1 terminalid 0
[Huawei-voice] save

Item Result
Intra-office call User A can dial user B and

communicate.
Inter-office call User A can dial user C and

communicate.
Incoming call User C can dial user A and

communicate.
----End
Configuration Files
#
ip address 192.168.200.155 255.255.255.0
#
voice
#
h248ag 1
mg-signal-ip 192.168.1.2 mgport 2949
primary-mgc-address1 static 192.168.1.110 2944
mg-media-ip1 192.168.1.2
#
h248aguser 1/0/0 agid 1 terminalid 0

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CLI-based Configuration Guide -

HUAWEI TECHNOLOGIES CO., LTD.

Trademarks and Permissions

Huawei Technologies Co., Ltd.

Issue 07 (2021-12-01) Copyright © Huawei Technologies Co., Ltd. i

1 About This Document.............................................................................................................1

Issue 07 (2021-12-01) Copyright © Huawei Technologies Co., Ltd. ii

2.10.4 Configuring a Pre-routing Number Change Plan................................................................................................ 67

Issue 07 (2021-12-01) Copyright © Huawei Technologies Co., Ltd. iii

2.14.3.5 Configuring Call Transfer........................................................................................................................................ 190

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2.14.4 Configuring the Service Password.......................................................................................................................... 287

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3 SIP AG Configuration......................................................................................................... 450

Issue 07 (2021-12-01) Copyright © Huawei Technologies Co., Ltd. vi

3.10.6.3 Enabling Silence Compression on a DSP Channel......................................................................................... 497

4 H.248 AG Configuration.................................................................................................... 504

Issue 07 (2021-12-01) Copyright © Huawei Technologies Co., Ltd. vii

4.9.5.4 Setting Fax Parameters.............................................................................................................................................. 542

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1 About This Document

Indicates a hazard with a high level of risk which, if not

Indicates a hazard with a medium level of risk which, if

Indicates a hazard with a low level of risk which, if not

Indicates a potentially hazardous situation which, if not

Issue 07 (2021-12-01) Copyright © Huawei Technologies Co., Ltd. 1

Supplements the important information in the main text.

Boldface The keywords of a command line are in boldface.

Italic Command arguments are in italics.

[] Items (keywords or arguments) in brackets [ ] are

{ x | y | ... } Optional items are grouped in braces and separated

[ x | y | ... ] Optional items are grouped in brackets and

{ x | y | ... }* Optional items are grouped in braces and separated

[ x | y | ... ]* Optional items are grouped in brackets and

&<1-n> The parameter before the & sign can be repeated 1

# A line starting with the # sign is comments.

Interface Numbering Conventions

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Reference Standards and Protocols

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● In this document, public IP addresses may be used in feature introduction and

Mappings Between Product Software Versions and NMS Versions

AR Product eSight iManager U2000 NCE-IP

V300R019C00 V300R010C00 V200R018C60 -

V300R019C10 V300R010C00 - V100R019C10

V300R019C11 V300R010C00 - V100R020C00

V300R019C13 V300R010C00 - V100R020C10

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2.1 Overview of PBX

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2.11 Configuring Trunk Groups

2.1 Overview of PBX

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Figure 2-1 Encrypting transmission channels using VPN

2.1.1 Voice Communication Function

Basic Concept of PBX

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2.1.2 Concepts Related to PBX