HCNP-UC-IHUCS V2.8 Training Material(2016.2.1)

677
 Huawei Unified Communications Certification Training  HCNP-UC-IHUCS  Implementing Huawei Unified Communications Solution Huawei Te ch nol ogies Co., Ltd  

Transcript of HCNP-UC-IHUCS V2.8 Training Material(2016.2.1)

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Huawei Unified Communications Certification Training

HCNP-UC-IHUCS Implementing Huawei Unified

Communications Solution

Huawei Technologies Co., Ltd

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Copyright © Huawei Technologies Co., Ltd. 2016 All rights reserved.

No pa rt of t his do cument may be r eproduced or t ransmitted i n a ny form o r by a ny meanswithout prior written consent of Huawei Technologies Co., Ltd.

Trademarks and Permissions

and other H uawei t rademarks ar e t rademarks of H uawei T echnologies C o., L td. A ll o thertrademarks and trade names mentioned in this document are the property of their respective holders.

Notice

The information in t his do cument is s ubject to change w ithout not ice. E very e ffort ha s be enmade in t he pr eparation o f t his do cument to en sure accur acy o f t he co ntents, but a ll s tatements,information, a nd recommendations in t his do cument do not c onstitute the w arranty o f a ny k ind,express or implied.

Huawei Certification

HCNP-UC-IHUCS

Implementing Huawei Unified Communications Solution

Edition V2.8

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Huawei Certification System

Relying on its unparalleled technical advantages and professional training system, Huawei

Certification is committed to delivering practical, professional, and hierarchical certification programs to customers based on different customer requirements.

Considering the characteristics of ICT technologies and customers' requirements, HuaweiCertification provides the following certification programs:

Huawei Certified Network Associate - Unified Communications Introducing Huawei UnifiedCommunications Administration (HCNA-UC-IHUCA) is intended for networking engineers whohave associate-level knowledge and skills of Huawei enterprise-specific UC solutions. It focuses ontypical solutions, component functions; service flows, basic configurations, and identifies enterprise

UC experts who are capable of system operation, administration, and maintenance (OAM).

Huawei Certified Network Professional - Unified Communications (HCNP-UC-IHUCS) is intendedfor networking engineers who have professional-level knowledge and skills of Huaweienterprise-specific UC solutions. It focuses on UC solutions planning and design as well as thetechnical capabilities of UC solutions in large- and medium-sized enterprises. Engineers withHCNP-UC certification are capable of UC system design and OAM.

Huawei Certified Network Professional - Unified Communications (HCIE-UC) is intended fornetworking engineers who have expert-level knowledge and skills of Huawei enterprise-specific

UC solutions. It focuses on UC solutions maintenance and troubleshooting as well as the technicalcapabilities of UC solutions in large enterprises. Engineers with HCIE-UC certification are capableof UC system design and troubleshooting.

Huawei authentication enables you to acquire the required industry knowledge and keep pace withthe most advanced technologies.

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Preface

Introduction

HCNP-UC-IHUCS Implementing Huawei Unified Communications Solution is designed to a ssist those who are preparing to attend the HCNP-UC-IHUCS exam, or want to learn the general principles of unified communications andthe applications in Huawei unified communications system.

Prerequisites

In order to learn the content more effectively, readers should have the basic knowledge listed below

• A general familiarity with communication theory;• HCNA-UC Certificated or acquired equal capability.

Content

This book includes 5 modules, starting from eSpace UC solution overview, elaborates eSpace UC solution applicationscenarios with detailed configuration and maintenance guide.

Mudule 1 i ntroduces Huawei eSpace UC s olution, i ncluding a ll p roducts, typical app lication scenarios, networking principles and deployment.

Mudule 2 describes the deployment and configuration of eSpace UC service, i ncluding the related installation of UCservers, user registration with the BMU, t he basic voice call service of eSpace Desktop, the instant messaging, voiceconferencing, rich media and the multi-media conferencing service.

Mudule 3 introduces how to deploy medium or large scale VoIP network for enterprise along with reliable networkingsolutions. Usually, U1900 can function as central or local node. AR voice router, a new product in eSpace UC solution,can be used as local node.

Mudule 4 explains t he eSpace c onsole s ystem a nd eSpace a udio recorder, providing readers value-added services t omeet personalized requirement.

Mudule 5 provides gui delines f or eSpace UC s ystem m aintenance an d t roubleshooting. By analyzing typical troublecases, readers can have a better understanding of whole system working principles.

Book guides the reader grasp technology in the eSpace UC solution step by step; also readers can read your interestingchapter according to yourself.

Referenced document• eSpace UC Product Documentation• eSpace U1900 Series Unified Gateway Produc Documentation• AR120&AR150&AR160&AR200&AR500&AR510&AR1200&AR2200&AR3200&AR3600 Product

Documentation

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Huawei Unified Communicatios Referenced Icon

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Contents

1 eSpace UC Solution Overview…………………………………………………………………1

1.1 eSpace UC Solution Overview………………………………………………………………………4

1.2 eSpace UC Product Portofolio……………………………………………………………………33

1.3 eSpace UC Application Scenarios…………………………………………………………………502 eSpace UC Solution Network………………………………………………………………58

2.1 eSpace UC Solution Network Overview…………………………………………………………………61

2.2 Networking Principles……………………………………………………………………………………63

2.3 Typical Application Scenarios……………………………………………………………………………69

2.4 Licenses…………………………………………………………………………………………………79

3 ECS Application Deployment ……………………………………………………………84

3.1 ECS Deployment Plan………………………………………………………………………………87

3.2 ECS Installation………………………………………………………………………………………101

3.3 ECS Service Configuration…………………………………………………………………………113

4 ECS Basic Services Configuration …………………………………………………………118

4.1 UC Account Configuration…………………………………………………………………………121

4.2 UC Terminal Configuration…………………………………………………………………………145

4.3 Managing an Enterprise Address Book………………………………………………………………155

4.3 Presence and IM Service…………………………………………………………………………162

4.4 User Self-Service………………………………………………………………………………………168

5 ECS Advanced Services Configuration ……………………………………………………177

5.1 Multimedia Conference Service…………………………………………………………………180

5.2 Multimedia Conference Configuration…………………………………………………………188

5.3 Using the Conference Service……………………………………………………………………196

6 eSpace IPT Reliable Solution Network……………………………………………………207

6.1 Reliable Network Introduction…………………………………………………………………………210

6.2 Dual Redundancy Recovery……………………………………………………………………………214

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6.3 Remote Disaster Recovery………………………………………………………………………………238

7 Multi-Node Centralized Network………………………………………………………241

7.1 Network Scheme and Principle of Multi-Node Centralized Network…………………………………244

7.2 Typical Network Scenario Description and Data Configuration………………………………2558 AR Voice System Introduction ………………………………………………………………284

8.1 AR Voice Product Introduction………………………………………………………………287

8.2 AR Voice Hardware Introduction…………………………………………………………………293

8.3 AR Voice Basic Concepts…………………………………………………………………………306

9 VRP Introduction and Operation ……………………………………………………………315

10 AR Voice Basic Operation and Configuration (Local Branch Scenario) …………………341

10.1 Networking Principle……………………………………………………………………………344

10.2 Data Configuration…………………………………………………………………………………354

11 Multi-Node Distributed Network ……………………………………………………387

11.1 Peer-to-Peer Mode Network……………………………………………………………………………390

11.2 Convergent Mode Network……………………………………………………………………………395

12 Toll Fraud Prevention ………………………………………………………………………404

12.1 Toll Fraud Scenarios……………………………………………………………………………………407

12.2 Preventing Calls from Being Transferred Through a Trunk and the VU………………………410

12.3 Preventing Registration or Login of Unauthorized Users………………………………………417

12.4 Taking Other Prevention Measures……………………………………………………………………419

13 Extranet Access of UC Terminals ………………………………………………………424

13.1 Extranet Access Solution Overview………………………………………………………………427

13.2 Networking Principle………………………………………………………………………………437

13.3 Data Configuration…………………………………………………………………………………441

14 eSpace Console System ……………………………………………………………………455

14.1 Console Networking…………………………………………………………………………………458

14.2 Console Scenario……………………………………………………………………………………461

14.3 Console Configuration………………………………………………………………………………463

15 eSpace Audio Recorder ………………………………………………………………477

15.1 Recording System Overview………………………………………………………………………480

15.2 Application Scenario………………………………………………………………………………492

15.3 Service Configuration and Operation……………………………………………………………498

16 eSpace UC Routine Maintenance ………………………………………………………538

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16.1 Precautions for Routine Maintenance………………………………………………………541

16.2 Backup and Recovery…………………………………………………………………………555

16.3 CDR and Billing………………………………………………………………………………570

16.4 Maintenance Operations on the AR………………………………………………………573

17 AR Voice System Maintenance and Troubleshooting ………………………………578

17.1 Status Query………………………………………………………………………………………581

17.2 Signaling Tracing……………………………………………………………………………………597

17.3 Debugging…………………………………………………………………………………………603

17.4 Statistics……………………………………………………………………………………………608

18 eSpace UC System Troubleshooting ………………………………………………………618

18.1 Troubleshooting Overview………………………………………………………………………62118.2 Collecting Faults Information……………………………………………………………………627

18.3 Typical Fault Analysis………………………………………………………………………………635

18.4 Obtaining Technical Support……………………………………………………………………658

Appendix . Glossary …………………………………………………………………………663

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eSpace UC solution is a leading unified communications solution developed by Huawei forenterprises. With the SIP-based open platform design, eSpace UC solution providesdiversified services, allows access for multiple terminals, and features high reliability,simplified deployment, and easy integration.

eSpace UC solution is ideal for small office home office (SOHO), and small, medium, andlarge enterprises. To meet increasing collaboration office needs, eSpace UC solutionconverges voice, data, and video, and uses a variety of terminals (such as fixed-line phone,handset, computer, and tablet) to provide enterprise users with numerous functions,including Voice over IP (VoIP), collaborative application, and mobile office. Enterprise userscan securely and conveniently access the enterprise service platform anytime, anywhere,using any device, dramatically enhancing office efficiency and user experience.

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Supplementary services also includes :

do-not-disturb (DND)

distinctive ring tone

emergency call

call pickup

call park

and ringback tone (RBT)

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A voice conference is a call session made among three or more users. Depending on theconference convening mode, voice conferences can be classified into scheduledconferences and instant conferences.

Scheduled conference : A convener schedules a conference beforehand using a web-basedself-service system. If the web-based self-service system is unavailable, the convener asksthe enterprise administrator to schedule a conference. When the conference time arrives,participants can join the conference in any of the following three ways:

Individual dial-in Participants dial the conference access code and enter theconference ID and password as prompted to join the conference.

Moderator convening The moderator joins a conference and performs operations onthe phone to invite participants to join.

System convening The system automatically invites participants to the conference.

Instance conference : The moderator does not schedule a conference beforehand. Instead,he or she directly dials the conference access code on a terminal and then invitesparticipants to join when prompted.

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The automatic switchboard service is also known as the interactive voice response (IVR)service. After a number is set as the automatic switchboard number and the automaticswitchboard service is enabled in an enterprise, all incoming calls are automaticallyforwarded to the automatic switchboard. The switchboard then plays announcements,prompting the calling party to perform operations accordingly. By default, the automaticswitchboard prompts the calling party to dial an extension number.

Enterprise administrators can customize switchboard announcements and multi-level IVRannouncements and configure the system to play varied announcements at different timeperiods of a day. The time periods can be configured by year, month, date, week, starttime, end time, and holiday.

The system supports a maximum of 256 automatic switchboards and a maximum of threelevels of IVR.

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In an enterprise, employees in the same field can be added to a hunt group, with thegroup access code configured. When a user dials the group access code, the phones ofthe sign-in group members ring simultaneously, cyclically, or sequentially according to theconfigured routing policy. The hunt group service enables employees in the same field towork together for more efficient request processing.

A user can be added to multiple hunt groups to handle different service requirements.

When a call is connected to a hunt group through the hunt group access code, personalservices (such as call forwarding and DND) for the hunt group members, if enabled, will beunavailable.

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Application Scenario

In an enterprise, employees in a department (for example, IT support department)have been added to a hunt group with a unified access code, and simultaneousringing has been configured for the hunt group. During the office hours, all groupmembers sign in to the hunt group. When a user dials the access code of this huntgroup, the phones of all employees in this hunt group ring at the same time. Afteran employee answers the call, the phones of the other employees stop ringing.

If sequential ringing is configured, the phone of the desired employee (whose phoneis expected to ring first according to the configured routing policy) rings first. If thedesired employee cannot answer the call, the call is diverted to the next expectednumber.

When no employee in the hunt group answers the call, the call is forwarded to theforward-to number configured for the hunt group access code. For example, thehunt group has configured call transfer to the voice mailbox upon ringing timeout.During the break, if a user dials the hunt group access code, no one answers thecall. When ringing times out, the system asks the user to leave a message. After theuser leaves a message, the MWIs on the phones of all members in the hunt groupare lit up. When the group members go back to work and find that the MWI is on, amember presses the corresponding button on the phone, enters the password, andlistens to the message. Then, the MWIs on the other phones are turned off.

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When a user dials the access code of a call group, phones of all members in the group ringsimultaneously. Members can pick up phones to join a meeting.

By default, the instant hotline service is enabled for call group members who can talk withspecific outer-group users after picking up phones. Members in a call group must beconfigured to the agent role. The call group service applies to highway scenarios.

When an outer-group user dials the group access code, all phones in the call group ring.After picking up their phones, call group members can directly talk with the outer-groupuser.

Call group specifications:

Item U1980/U1981 U1960 U1911

Maximum number of call groups 256 100 10

Maximum number of users in acall group

10 10 10

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Typical Scenarios

[Scenario 1] An agent calls a call group

When an outer-group agent dials the group access code 83000, all phones(including 83001, 83002, and 83003) in the call group ring. Any member whopicks up a phone can talk with the agent. Phones that are not picked upcontinue ringing.

[Scenario 2] A call group member calls an agent

When call group member 83001 picks up a phone, the system automaticallycalls the outer-group agent and member 83001 hears a ringback tone.Member 83001 can talk with the outer-group agent who answers the call.

During the call between member 83001 and the outer-group agent, thesystem automatically calls the outer-group agent if member 83002 picks up aphone. Member 83001 hears a ringback tone and waits in the call queue. Theouter-group agent can check calls in the queue.

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Multiple states

Online, busy, away, calling, in conference, and so on

Outlook presence plug-in

The status of receivers and senders is displayed in the email address bar.

Presence function for third parties

The presence interface can be invoked by third-party applications, for example, athird-party OA system.

Function Description

UC soft terminals display the terminal status and information of enterprise users.Users can set their terminal status for others to learn about their working status.Users can query the status and information of a contact in real time and select anappropriate communication mode according to the contact status.

The contact status can be online, away, busy, offline, and DND, and the informationabout the contact includes the signature, profile picture, and phone number. Thesystem provides default auto replies when users are away, and also allows users tocustomize auto replies.The contact status changes are synchronized to the terminalsof other users in real time.

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Application Scenario

If a user wants to go somewhere else, the user can set his or her UC soft terminalstatus to Away and the auto reply to "I am leaving for a while. I will reply to youlater." Others can learn about the status of this user and will contact this user lateror select an appropriate communication mode to reach this user. If others send IMsto this user even when finding that this user is away, they will receive the auto replypreset by this user.

If a user is busy and does not want others to disturb him or her, he or she can sethis or her terminal status to Busy. Others can learn about the status of this user and

will contact this user later if things are not urgent.

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Corporate Directory

Directory creation and maintenance

An enterprise administrator creates department structures and adds employeeinformation on the BMU for the enterprise. The employee information includesthe employee name, department, mobile and other phone numbers, position,fax number, address, and postal code.The administrator can rearrange sub-departments under a department. After the rearrangement, sub-departmentsare displayed in a rearranged order on UC soft terminals.The administrator canmodify the information about an employee on the BMU when the employee

information is changed.Directory search

Enterprise users can search the corporate directory for other users within theenterprise on their UC soft terminals. Directory search by multiple keywords issupported.

Directory access rights control

The administrator can set different user levels (a maximum of 20 user levels)

for accessing the corporate directory. Users at a higher level can viewinformation about others at the same or a lower level, but users at a lowerlevel cannot view information about others at a higher level.

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P2P IM

The P2P IM services allow two users to send IMs, transfer files, and send SMSmessages with each other.

Temporary Group IM

An enterprise user can select multiple contacts to create a temporary group oneSpace Desktop. Contacts in the temporary group can send IMs in the group's IMwindow.

Contact Group IM

An enterprise user can create and manage contact groups. A contact group membercan send IMs in the contact group IM window, and all the other members in thegroup can receive the IMs.

P2P File Transfer

Files can be transferred between eSpace Desktops. A user can drag a file to the IMwindow of eSpace Desktop for file transfer.

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In IP phone linkage, a UC account corresponds to a linkage number for two terminals (UCsoft terminal and IP phone). The linked IP phone and eSpace Desktop must register withthe same U1900.

IP phone linkage combines the advantages of eSpace Desktop (operation convenience) andIP phone (HD audio) to deliver better communications experience for enterprise users.After linkage is enabled, a user can select his or her IP phone as the communication deviceon his or her eSpace Desktop. The user's IP phone and eSpace Desktop are then linked.The user can perform operations, such as placing, answering, rejecting calls, holding andresuming calls, diverting calls, initiating instant conferences, and joining scheduledconferences, on the eSpace Desktop for the linked IP phone. The IP phone carries audiostreams and responds according to the operations performed.

The linked IP phone and eSpace Desktop must register with the same U1900.

Retains user habits, ensures high-quality voice experience, brings large-screenadvantages into full play, simplifies operations, diversifies application scenarios, andenhances user experience.

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Multiple types of terminals join a meeting

When a person you want to invite to a meeting is not online, you can directly call hisor her other number, ensuring that the desired participant will join the meeting.

With a mouse click, you can easily select the terminal type for all participants to jointhe meeting.

When a camera is available, you can also start a video call.

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A multimedia conference is a combination of voice, video, and data conferences.

Conference scheduling: A convener schedules a conference beforehand using a web-basedself-service system. If the web-based self-service system is unavailable, the convener asksthe enterprise administrator to schedule a conference.

Ways of joining multimedia conferences:

1) Users can dial the conference access code and join a conference as prompted or clickthe link in the conference notification email.

2) Users accept the conference invitation sent by the conference moderator to join aconference.

3) eSpace Desktop users can click the conference access shortcut and perform relatedoperations as prompted to join a conference.

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During a conference, common participants can freely browse the document shared by thepresenter, free of the presenter's stop on the document.

Supported files include PPT, Word, Excel, PDF, text, jpeg, and bmp.

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Functions:

Users who log in to one client can view chat records left on the other client.

Solution:

Chat records are synchronized to the UC server, and can be reached across UCclients.

Customer benefits:

Chat records are stored via cloud, ensuring continuous communication between PCand mobile clients.

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Secure, real-time and convenient: more than a corporate directory.

Real-time contact presence display: determine a contact status instantly.Without local storage and maintenance, easily obtain latest immense amount of contactdata.

Encrypt Corporate directory data through AES256 with high information security.

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Supports voice clip, video clip, picture, photo, and doodle.

Provides mobile social experience and multimedia message sharing .

Wherever you are, you can send first-hand messages to others in a quick, accurate, andsecure manner.

Security measures include:

TLS/SRTP encryption

AES encryption

VPN channel encryption

Enterprise self-management

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Feature Definition

When the eSpace Mobile is offline, the server pushes message notifications to users'mobile phones when receiving a new message. This enables users to check newmessages on the eSpace Mobile in a timely manner, helping improve communicationefficiency.

Typical Scenarios

The server pushes message notifications to users' mobile phones in either of thefollowing scenarios:

The eSpace Mobile is exited but not logged out, and users do not log in toother terminals.

The eSpace Mobile process is terminated.

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NOTE:

In the mobile data network or Wi-Fi networks environment, the quality of voice callsis subject to the 3G or Wi-Fi network's quality of service (QoS) performance. Huaweicannot guarantee the voice quality in such environments.

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Lync 2013 plug-in can be installed on Lync 2013.

Lync provides the IM function, and other functions such as voice and conference areimplemented using Lync plug-in.

Lync plug-in provides the following functions:

Voice and video calls

Instant conference

Phone status synchronization

Service configuration

Historical record management

IP phone control

Directory search

Before using Lync plug-in, Huawei UC accounts must be available.

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Sametime 8.5 plug-in can be installed on Sametime 8.5.

Sametime provides the IM function, and other functions such as voice and conference areimplemented using Sametime plug-in.

Sametime plug-in provides the following functions:

Voice and video calls

Instant conference

Phone status synchronization

Service configuration

Historical record management

IP phone control

Directory search

Before using Sametime plug-in, Sametime UC accounts must be available.

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embed UC capabilities into enterprise applications.

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eSpace UC solution delivers consistent user experience from anywhere, at any time, onmultiple devices including IP phones, analog phones, fax machines, PCs, smartphones, andtablets.

An IP phone is a multimedia terminal that connects to an IP network through SIP.

An analog phone is a traditional desktop phone that converts vibrations intoelectrical frequency signals for transmission. It can connect to the system through asubscriber board on eSpace U1911/U1960/U1980/U1981 or through an eSpace IAD.

eSpace SoftConsole is a client used by contact center agents. Users can beconnected to eSpace SoftConsole through the enterprise switchboard. eSpaceSoftConsole provides a variety of functions for agents, including call transfer,incoming call queuing, barge, and forced release.

Users can enjoy diverse services provided by UC soft terminals (eSpace Desktop forPCs, eSpace Mobile for smartphones, and eSpace Mobile HD for tablets) anytimeanywhere. The services include IM, voice and video calling, and conferencing.

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An eSpace IAD is a voice over IP (VoIP) and fax over IP (FoIP) media access gateway. Itconnects analog phones or fax machines to an IP network for providing efficient andquality voice or fax services.

As the media access gateway of Voice over IP (VoIP) and Fax over IP (FoIP), the IAD isapplied to the Next Generation Network (NGN) or IP Multimedia Subsystem (IMS). The IADconverts analog voice data into IP packets and transmits data through the IP network.

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As the integrated access gateway, AR series products provide integrated capabilities,including voice, route, switch, and security, which reduces branch deployment costs.

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The eSpace U1900 series are switches used in Huawei IP Telephony (IPT) solutions. TheeSpace U1900 series provide professional voice over IP (VoIP) solutions that meetcommunication requirements of various enterprises.

The eSpace U1900 series gateway is the core switch control device used in Huawei IPTand eSpace UC solutions. It can be deployed on a hybrid network with IP phones, remoteanalog phones, and soft phones, providing high-quality voice and value-added services forenterprises customers, such as government departments, electric power companies, andfinancial companies.

3GE: Triple-network-port mode is supported. Different network ports process different

types of data (management data, service data, and call record data), and they havedifferent external IP addresses.

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Works with eSpace SoftConsole and eSpace U1911/U1960/U1980/U1981 to comprise theconsole system. An eSpace ConsoleServer can connect to a maximum of eight eSpaceU1911/U1960/U1980/U1981s. It supports two devices working in active/standby mode.

Can be deployed on the same host as eSpace SoftConsole.

Stores contact center agent information and authenticates agents when they log in fromeSpace SoftConsole.

Queues incoming calls, maintains such queues, and updates queue information to eachSoftConsole.

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SoftConsole

Provides basic call functions as phones do, such as placing, answering, and holdingcalls.

Provides agent functions, such as showing busy, showing idle, and forwarding calls.

ConsoleServer

Saves information about agents.

Authenticates agents during their login.

Queues incoming calls, maintains such queues, and updates queue information toeach SoftConsole.

Unified gateway

Handles SIP user registration of the SoftConsole.

Connects calls between agents and users.

BMU Provides SoftConsole parameters and the service configuration interface, and deliversservice configurations to the unified gateway.

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eSpace U1911/U1960/U1980/U1981 works with the CDRServer to generate call detailrecords (CDRs) for the Business Management Unit (BMU) or a third-party billing system toobtain.

eSpace U1911/U1960/U1980/U1981 generates CDRs for all users connected to it andsaves the CDRs to its CDR pool. The CDRServer obtains the CDRs from the CDR pool ofeSpace U1911/U1960/U1980/U1981, parses the CDRs to generate CDR files, and savethem in the local disk. A CDRServer can connect to a maximum of 255 eSpaceU1911/U1960/U1980/U1981.

The BMU obtains call records from the CDRServer through an FTPS interface, and

writes the call records to the database.The administrator can manage call records on the BMU, such as querying anddeleting call records.

Enterprise users can query their own call records on the BMU self-service platform.

The CDRServer allows a third-party billing system to obtain the CDRs through an FTPor FTPS interface.

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In eSpace UC solution, the eSpace Audio Recorder can record two-party calls andconferences.

The eSpace Audio Recorder uses the site recording mode, in which the eSpace AudioRecorder joins a call or conference as a participant to receive data for recording.

Recording a two-party call :

The eSpace Audio Recorder joins an ongoing two-party call to establish athree-party conference. The eSpace Audio Recorder receives SIP and RTPstreams in the conference for recording.

Recording a conference :

During a voice conference, the moderator can enable the recording function.The eSpace Audio Recorder joins the conference as a participant and recordsthe conference.

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eSpace UC solution supports centralized service management using the BMU and built-in webmanagement system of eSpace U1911/U1960/U1980/U1981.

The BMU is a service management system in eSpace UC solution. By connecting to multipleunified gateways, CDRServers, and ConsoleServers, the BMU enables users to manage a singleor a batch of numbers, accounts, and conferences as well as perform other self-service tasks.

Main Functions:

Number Management

Account management

Service configuration : Configure service rights for numbers in batches

Conference management : Configure conference access prefixes

Self-service

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The Local Maintenance Terminal (LMT) is a local configuration and maintenance system foreSpace U1911/U1960/U1980/U1981.

The LMT system connects to eSpace U1900 series using SSH/Telnet, and a maximum offour eSpace U1900 series can be connected at the same time.

Main management functions of the LMT system:

Configuration management : Provides a command navigation tree for dataconfiguration and query.

Alarm management : Supports real-time alarm query and provides alarm handlingsuggestions; supports alarm export.

Maintenance management : Supports functions such as data configuration, voice filemanagement, patch management, license management, and visualized VU scriptgeneration.

Signaling tracing : Traces and parses broadband and narrowband signaling.

Upgrade : Provides version and patch backup, version upgrade, and patchinstallation functions.

Log collection : Allows users to load operation logs, run logs, and debug logs.

Offline operation : Allows users to parse alarm files, log files, and signaling tracingfiles offline, and supports offline upgrade of data files.

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eSight centrally monitors NEs by managing their performance and alarms.

Topology Management

eSight provides a network topology to present NEs and their connections. With thehelp of the network topology, users can understand the hierarchical structure andrunning status of all NEs across the network. When a new NE is created, eSightautomatically discovers and adds it to the network topology. Alternatively, users canmanually search for new NEs and add them to the network topology.

Resource Management

eSight manages NEs by subnet. According to the physical locations of the devices,the administrators can put the devices into different subnets for management.

Performance Management

eSight collects the performance data of managed NEs and displays it in charts. Byreading these charts, users can understand potential network risks and takepreventative measures accordingly. eSight displays monitoring results in bar chartsor line charts to clearly show the performance trends. The administrators caneliminate potential risks by taking active measures before a performance counter

reaches its threshold.

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Alarm Management

eSight monitors network alarms in topology views, alarm panels, and alarm barcharts. It monitors network exceptions in real time so that the administrators cantake measures against the detected network exceptions in a timely manner torestore network operation. The administrators can set remote notification rules,masking rules, and tones for alarms based on the site requirements, which optimizesnetwork management.Once eSight detects an alarm, the administrators can reviewalarm details, acknowledge the alarm, locate the alarm, and then clear the alarm.

Configuration Management

eSight allows the administrators to remotely configure specific devices.

Device Panel Management

eSight provides an intuitive device simulation GUI (also known as device panel) foreSpace U1911/U1960/U1980/U1981, where the administrators can view board andindicator status.

Maintenance Management

eSight supports batch upgrade of eSpace IADs and IP phones, which is useful whena large number of devices need to be upgraded.

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Solution Highlights

All-IP communication, simple deployment, reducing the CAPEX.

IP PBX with a built-in web management system, reducing the OPEX.

Rich services, such as conference call, call transfer, and hunting group, improvingefficiency of daily communication.

Compatibility with legacy POTS phones and fax machines, delivering a high ROI.

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Solution Highlights

All-IP communication, diverse VoIP services for IP phones, POTS phones, and PCclients.

Large-capacity high-quality conference system, abundant resources, various accessmodes, excellent conference experience.

Convergent unified communications services, such as messaging, presence, group,and conferencing, enhancing communication efficiency.

PC client and IP phone linkage, call quality assurance, simple operation.

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Solution Highlights

Endpoint access security: 802.1x authentication, registration authentication.

Communication service security: TLS/SRTP encryption for call services; AESencryption for presence, group, and directory services

Management security: security management protocols such as HTTPS and SSH:rights- and domain-based management

IP network security: SBC for network isolation.

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Solution Highlights

High device reliability:Module redundancy:

main control board

media resource board

power supply

network port

Board plugging:

mixed insertion

hot plugging

Online patch:

hot patch

High system reliability:

IP PBX: two-node hot standby cluster/geographical redundancy.

ECS: two-node/cluster cluster, service servers supporting redundancy.

Branch gateway: Local regeneration/self-switching, PC client also supporting voiceregeneration.

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A single-node network is suitable for an enterprise with no more than 10,000 users thatdoes not have branches or has branches that do not require local regeneration. In such anetwork, the unified gateway and a variety of service servers are all deployed at theenterprise's equipment room, and all terminal users register with the unified gateway. Ifthe enterprise has small-sized branches, eSpace IADs can be deployed at branches andfunction as proxies for user registration to the unified gateway, or IP phones can bedeployed at branches and register with the unified gateway.

Networking Description

The unified gateway and a variety of service servers are deployed at the

headquarters.SIP terminals (including IP phones and SoftConsoles) register with the unifiedgateway. Analog phones and fax machines can directly connect to eSpaceU1911/U1960/U1980/U1981 through the ASI board thereon. They can also connectto eSpace IADs, and register with the unified gateway as SIP users through theconnected eSpace IADs that function as proxies for user registration.

The unified gateway can connect to the PSTN through trunks such as PRA, SS7,QSIG, R2, BRI, and AT0.

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Service servers, such as the UMS (for the unified messaging service), CDRServer (forthe CDR service), ConsoleServer (for the console service), and Audio Recorder (forthe recording service), can be deployed based on service requirements.

The ECS software can be deployed to obtain ECS value-added services such as IM,group, and multimedia conferencing. UC soft terminals send registration andauthentication requests to the eServer for registration and authentication.

The built-in VMS of the unified gateway and the UMS have overlapping services.Customers can deploy either the VMS or USM based on their voice mailbox servicerequirements.

eSight centrally manages eSpace U1911/U1960/U1980/U1981, eSpace IAD and IPPhone.

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A centralized call management network can be used for an enterprise with no more than10,000 users that has branches that require local regeneration. In such a network, theunified gateway functioning as the central node and a variety of service servers are alldeployed at an enterprise's equipment room, and all terminal users register with thecentral node in normal conditions. Unified gateways functioning as branch nodes aredeployed at the branches for local outgoing calls and local regeneration.

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A distributed call management network in peer-to-peer mode can be used for enterpriseswith 10,000-30,000 users. In such a network, two or three nodes each deploy a unifiedgateway, which provides call services for the corresponding node. Different nodes areconnected using SIP trunks, through which inter-node calls are routed.

Networking Description

Each node has a unified gateway deployed. The unified gateway at a node is a voicegateway, which is responsible for user registration, intra-office users, and local PSTNcalls at that node. Different nodes are connected using SIP trunks, through whichinter-node calls are routed. The unified gateways can be eSpace U1911, U1960,

U1980, or U1981.IP phones, SoftConsole users (SIP users), UC soft terminal users (SIP users), andeSpace IADs (on behalf of analog phones and fax machines) at a node register withthe unified gateway at the node.

The unified gateways at different nodes can connect to the PSTN using trunks suchas PRA, SS7, QSIG, R2, BRI, and AT0 for local outgoing calls.

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Enterprises with more than 30,000 users can use a distributed call management networkwhere four or more unified gateways are convergent.

Networking Description

Each node has a unified gateway deployed. The unified gateway at a node is a voicegateway, which is responsible for user registration, intra-office users, and local PSTNcalls at that node. A voice gateway can be eSpace U1911, U1960, U1980, orU1981.

IP phones, SoftConsole users (SIP users), UC soft terminal users (SIP users), andeSpace IADs (on behalf of analog phones and fax machines) at a node register withthe voice gateway at the node.

The voice gateways at different nodes can connect to the PSTN using trunks such asPRA, SS7, QSIG, R2, BRI, and AT0 for local outgoing calls.

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Voice Communication

The unified gateway supports basic voice communication, including: Intra-officecommunication Intra-office users under the U1980 unified gateway can use varioussupported narrowband terminals to make voice calls to each other.

Narrowband trunk-based communication The U1980 unified gateway connects tothe PSTN or TDM PBX through digital trunks or analog trunk such as PRA, R2, SS7(ISUP/TUP), Q Signaling (QSIG). SS7 is short for signaling system No. 7, ISUP forintegrated services digital network user part, and TUP for telephone user part. Userscan make voice calls and receive incoming calls based on narrowband trunks.

Broadband trunk-based communication The U1980 unified gateway uses the SIPtrunk to connect to the IP PBX, softswitch, or IMS. Voice calls can be made betweenintra-office users and the users under other devices.

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Instant MessagingP2P IM The P2P IM services allow two users to send IMs, transfer files, and sendSMS messages with each other

Temporary Group IM An enterprise user can select multiple contacts to create atemporary group on eSpace Desktop. Contacts in the temporary group can send IMsin the group's IM window.Contact Group IM An enterprise user can create and manage contact groups. Acontact group member can send IMs in the contact group IM window, and all theother members in the group can receive the IMs.

P2P File Transfer Files can be transferred between eSpace Desktops. A user candrag a file to the IM window of eSpace Desktop for file transfer.

Group ServiceContact Group A user can create a contact group on his or her UC soft terminal,and invite others to this group. The user who creates the group is the groupadministrator. Others can accept or reject the invitation.Temporary Group A user can create a temporary group and invite relatedpersonnel to the group for discussion. A user in the temporary group can sendgroup IMs, view the status of others in the group, and start instant conferences withall group members.

Presence

UC soft terminals (eSpace Desktop, eSpace Mobile, and eSpace Mobile HD) displaythe terminal status and information of enterprise users. Users can set their terminalstatus for others to learn about their working status.

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Multimedia Conferencing

A multimedia conference is a combination of voice, video, and data conferences.

Conference scheduling: A convener schedules a conference beforehand using aweb-based self-service system. If the web-based self-service system is unavailable,the convener asks the enterprise administrator to schedule a conference.

Ways of joining multimedia conferences: 1) Users can dial the conference accesscode and join a conference as prompted or click the link in the conferencenotification email. 2) Users accept the conference invitation sent by the conferencemoderator to join a conference. 3) eSpace Desktop users can click the conferenceaccess shortcut and perform related operations as prompted to join a conference.

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Two-Node Cluster and Clustering

eSpace UC solution supports two-node clusters for core gateways, some serviceservers, and some databases. When the active node in a two-node cluster fails dueto a hardware or network fault, the standby node automatically takes over services.In addition, eSpace UC solution supports clustering of eServers, MAAs, MeetingMSs, and recording servers.

Local Regeneration

When the active and standby nodes are both faulty or network connectivity is lostbetween the headquarters and branches, users at the branches fail to register withthe headquarter gateways. At that time, the branch nodes can use the localregeneration function to provide basic telephony services for local IP and analogphone users.

Users' dialing rules remain unchanged, but inter-branch calling using short numbersis unavailable. To call an internal user at another branch, others can dial the PSTNnumber of this user for connecting to this user through the PSTN.

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SBC An enterprise may have multiple networks that are isolated from each other, and UCterminals and the UC system (that is, UC servers) of the enterprise may be located ondifferent networks. As a result, UC terminals cannot directly connect to the UC system.When this occurs, you can use the SBC to function as the signaling proxy and mediaaggregation gateway to enable VoIP and data communication among the isolatednetworks.

The SBC mainly fulfills the following functions: Signaling proxy Works as the terminalregistration proxy gateway to register UC terminals with the SIP server.

Media aggregation Aggregates media for establishing sessions between two UC

terminals or between a UC terminal and the UC system on two isolated networks.Network Address Translation (NAT) traversal Provides the NAT traversal functionwhen SIP messages need to be traversed by NAT (only the ACME SBC provides thefunction).

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eSpace U1911/U1960/U1980/U1981 generates CDRs for all users connected to it andsaves the CDRs to its CDR pool. The CDRServer obtains the CDRs from the CDR pool ofeSpace U1911/U1960/U1980/U1981, parses the CDRs to generate CDR files, and savethem in the local disk. A CDRServer can connect to a maximum of 255 eSpaceU1911/U1960/U1980/U1981s.

The BMU obtains call records from the CDRServer through an FTPS interface, andwrites the call records to the database.

The administrator can manage call records on the BMU, such as querying anddeleting call records.

Enterprise users can query their own call records on the BMU self-service platform.

The CDRServer allows a third-party billing system to obtain the CDRs through an FTPor FTPS interface.

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The eSpace Audio Recorder uses the site recording mode, in which the eSpace AudioRecorder joins a call or conference as a participant to receive data for recording.

Recording a two-party call The eSpace Audio Recorder joins an ongoing two-partycall to establish a three-party conference. The eSpace Audio Recorder receives SIPand RTP streams in the conference for recording

Recording a conference During a voice conference, the moderator can enable therecording function. The eSpace Audio Recorder joins the conference as a participantand records the conference

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U1911/U1960/U1980/U1981 support dual host redundancy and remote disaster

recovery(A/S mode).

ECS advanced edition support the two-node cluster and clustering.

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When the total users accounts number be in the range 1000~10000

eSpace UC users is 0

If the corporate directory records≤3000 : We can deploy the DB server on thesame host as the BMU;

If 3000<corporate directory records≤30000 : We must deploy the DB serverindependently.

eSpace UC users be in the range 0~1000(mobile clients ≤ 300)

If the corporate directory records≤3000: We can deploy the BMU server on

the same host as the eServer;

If 3000<corporate directory records≤30000 : We must deploy the BMU serverindependently.

eSpace UC users be in the range 1000~10000(mobile clients ≤ 3000)

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The number of concurrent register mobile clients(mobile user, PAD user) is Total UC usersnumber*30% by default.

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When the eSpace UC users is 0, we can deploy the eSight server like following:

It is deployed on the same host as the CDR server.

If no CDR server or console server, eSight is deployed independently.

When the eSpace UC users less than 1000

Meeting server

It support 100 concurrent multimedia conferences, when MS deployed on thesame host as the eServer.

If the concurrent multimedia conferences more than 100, MS need deployedindependently. Single MS support 800 concurrent multimedia conferences. MScluster support 8 MS, so the total concurrent multimedia conferences is 6400.

BMU/APPAgnet server

If corporate directory records≤3000: deployed on the same host as theeServer.

If 3000<corporate directory records≤30000 : deployed independently.

eSight server

It is deployed on the same host as the CDR server.

If no CDR server or console server, eSight is deployed independently.

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When the eSpace UC users is 0, we can deploy the DB server like following:

It is deployed on the same host as the BMU server.

If no CDR server or console server, eSight is deployed independently.

When the eSpace UC users less than 1000

Meeting server

It support 100 concurrent multimedia conferences, when MS deployed on thesame host as the eServer.

If the concurrent multimedia conferences more than 100, MS need deployedindependently. Single MS support 800 concurrent multimedia conferences. MScluster support 8 MS, so the total concurrent multimedia conferences is 6400.

BMU/APPAgnet server

If corporate directory records≤3000: deployed on the same host as theeServer.

If 3000<corporate directory records≤30000: deployed independently.

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When the eSpace UC users less than 10000

Meeting server

It support 400 concurrent multimedia conferences, when MS deployed on thesame host as the eServer.

If the concurrent multimedia conferences more than 400, MS need deployedindependently. Single MS support 800 concurrent multimedia conferences. MScluster support 8 MS, so the total concurrent multimedia conferences is 6400.

When the eSpace UC users less than 50000

Meeting server

It support 400 concurrent multimedia conferences, when MS deployed on thesame host as the eServer.

If the concurrent multimedia conferences more than 400, MS need deployedindependently. Single MS support 800 concurrent multimedia conferences. MScluster support 8 MS, so the total concurrent multimedia conferences is 6400.

MAA server

If 3000 < mobile users ≤10000: deployed MAA server independently.

If 10000 < mobile users ≤15000: deployed a two -node cluster of the MAAservers.

In advanced edition, one S2600T and 12 Disk Unit will be needed for eServer and DB.

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Server port

Indicates the port used by the BMU to connect to the unified gateway.

The default port used to connect to U1900 series unified gateways is 8000.

Run the command show system bmu information to see the result.

Server password

Indicates the password for the BMU to access the unified gateway.

Set this parameter to the password that is used by the admin user to enter theconfig mode,default is Change_Me.

Area

Set this parameter based on your site scenario.

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IsJoint

Indicates whether a SIP number is linked.

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If you want to register SIP numbers during the UC account registration on the BMU,configure a number allocation policy first.

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Number allocation policy

If you have not added SIP numbers, select an existing number allocation policy andenter the SIP number in Service number. Then, the system generates the UC accountand the mapping SIP number.

If you have added SIP numbers, you do not have to select a number allocation policyhere.

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1. Double-click the eSpace Desktop shortcut icon .2. Enter the account number and password.

3. Click in the lower right corner of the main window and click Server address. Click theIntranet tab. On the tab page that is displayed, set the server IP address and port numberand click Save.

4. Set Server Address to the eServer IP address. Use the default port number 8011.

5. Click Login.

NOTE: The password is specified by the administrator or automatically generated by thesystem during account registration. You can obtain the password from the administratoror in the email.

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A multimedia conference is a combination of voice, video, and data conferences.

Conference scheduling: A convener schedules a conference beforehand using a web-basedself-service system. If the web-based self-service system is unavailable, the convener asksthe enterprise administrator to schedule a conference.

3 Ways of joining multimedia conferences:

Users can dial the conference access code and join a conference as prompted orclick the link in the conference notification email.

Users accept the conference invitation sent by the conference moderator to join aconference.

eSpace Desktop users can click the conference access shortcut and perform relatedoperations as prompted to join a conference.

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The presenter can create a questionnaire. The other participants can complete thequestionnaires on their own UC soft terminals.

A maximum of 20 questionnaires can be created.

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Do not select Calling line identification restriction when you enable instant conferencing. Ifit is selected, the user cannot use multimedia conferencing.

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eSpace U1911/U1960/U1980/U1981 supports networking for remote disaster recovery (orcalled geographic redundancy). Midsize and large enterprises with multiple branches canuse such networking for better reliability. When devices in one place are faulty, devices inanother place take over services to ensure service continuity.

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Active/Standby redundancy Two unified gateways are deployed at the headquarters as the active and standbynodes. The active and standby nodes use a heartbeat mechanism to constantlycheck each other's status.

NOTE:The active and standby nodes must use the same unified gateway model.

When the active and standby nodes are running correctly, all users register with the activenode. The active node processes all user requests and synchronizes user registration andconfiguration data to the standby node in real time.

When the active node fails, the standby node takes over all services from the active nodeand processes all user requests.

If the trunk gateway is not deployed, the active and standby nodes both connect to thePSTN. When the active node fails, configurations are required on the PSTN to set up trunkchannels with the standby node, and the standby node then routes calls to the PSTNthrough trunks.

All service servers support active/standby switching of the gateways. This means these

service servers automatically connect to the standby node for service processing after aswitching from the active node to the standby node.

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The switching has the following impacts: New services cannot be started for ongoing calls,for example, call hold.

Ongoing voice conferences are interrupted.

CDRs cannot be generated for ongoing calls or conferences.

Ongoing message services are interrupted.

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Queues in the console system become invalid.eSpace SoftConsole disconnects from the ConsoleServer.

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The disaster recovery mechanism is described as follows:Two central nodes are deployed in two places, and each central node has oneunified gateway deployed. The two unified gateways work in active/standby mode,known as an active node and a standby node (the active and standby nodes mustuse the same gateway model). The active and standby nodes use a heartbeatmechanism to constantly check each other's status. The service servers are deployedat the central node where the active unified gateway resides.

When the active and standby nodes are running correctly, all users register with the

active node. The active node processes all user requests and synchronizes data tothe standby node in real time. At least 1 Mbit/s bandwidth must be reserved fordata synchronization, and the round trip time (RTT) must be shorter than 80 ms.

When the active node fails, the standby node takes over all services from the activenode and processes all user requests.

A trunk gateway can be deployed for connecting to the PSTN. If the trunk gatewayis not deployed, the active and standby nodes both connect to the PSTN. When theactive and standby nodes are running correctly, the active node routes calls to thePSTN through trunks. When the active node fails, the standby node routes calls tothe PSTN through trunks.

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Networking DescriptionThe unified gateway at the headquarters is the central node, which can be eSpaceU1911, U1960, U1980, or U1981. The central node supports two unified gatewaysworking in active/standby mode, known as the active node and standby node.

The unified gateway at a branch is the branch node, which can be eSpace U1911,U1960, U1980, or U1981.

The central node and branch nodes all connect to the PSTN using trunks such asPRA, SS7, QSIG, R2, BRI, and AT0. Two connection modes are available:

A trunk gateway is deployed at the headquarters for connecting to the PSTNand routes call to the PSTN using trunks. The trunk gateway connects to thecentral node using SIP trunks.

A trunk gateway is not deployed at the headquarters, but the central nodedirectly connects to the PSTN. If two unified gateways working inactive/standby mode are deployed as the central node, the active node andstandby node each connect to the PSTN using trunks.

The central node and branch nodes are connected using SIP trunks. They alsouse a heartbeat mechanism to constantly check each other's status.

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When the central node is correctly connected to the branch nodes,All users at the headquarters and branches register with the central node.

Services are uniformly configured and managed on the central node, and thecentral node synchronizes data to branch nodes in real time.

All calls are processed by the central node. When a branch user places anoutgoing PSTN call, the central node analyzes the call and routes the call tothe local branch node, which then routes the call outside of the enterprise.

When the central node is faulty or disconnects from a branch node, users at thebranch register with the branch node, and the branch node processes servicerequests from users at the branch. This is known as local regeneration.

Basic call functions for analog users at the branch node are not affected.

IP phones at the branch fail to register with the central node, and thenautomatically switch to the branch node for registration. Basic call functionsfor SIP users are not affected.

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Local Regeneration at Branch side

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Notice:Local call prefix 9 can be configured only after international toll call prefix 900 andnational toll call prefix 90 are configured.

When a user at local node dials prefix 9 (outgoing call prefix) and a called number,the call is routed to the active node. The active node reroutes the call to local nodeaccording to the call source (Digits of called numbers are deleted on local node .Users at local node belong to call source 1.).

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Automatical creation:The office route selection code/ office no / sip trunk are automatically createdprimaryserver and standby server are configured.

The sip trunk DefDNPrimarySrv and the office route selection code 1021 andoffice no 1021 to primaryserver are automatically created when primaryserver isadded!

The sip trunk DefDNStandbySrv and the office route selection code 1022 andoffice no 1022 to standbyserver are automatically created when standbyserver isadded!

The repeatroute 1021 bounded up with office selection code 1022 to standbyserveris automatically added! This repeatroute is associated to office selection code 1021.

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Notice Assume that MTU board is from slot 0 of trunk gateway. And 0 port is connected toPSTN.

config add board slot 0 type mtu officeno 0

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Automatical creation:The office route selection code/ office no / sip trunk are automatically createdprimaryserver and standby server are configured.

The sip trunk DefDNPrimarySrv and the office route selection code 1021 andoffice no 1021 to primaryserver are automatically created when primaryserver isadded!

The sip trunk DefDNStandbySrv and the office route selection code 1022 andoffice no 1022 to standbyserver are automatically created when standbyserver isadded!

The repeatroute 1021 bounded up with office selection code 1022 to standbyserveris automatically added! This repeatroute is associated to office selection code 1021.

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Notice Assume that ASI board is inserted in slot 1 of trunk gateway..

config add board slot 0 type asi

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Automatical data are created after configuring networking data:config add repeatroute no 1021 failedtype callfailed officeselectcode 1022cldpredeal no

config add office selectcode no 1021 repeatroute yes repeatrouteno 1021

config add office selectcode no 1022 repeatroute no

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Notice:Assume that MTU board is from slot 0 of local node. And port 0 is connected toPSTN.

config add board slot 0 type mtu officeno 0

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Notice:Local call prefix 9 can be configured only after international toll call prefix 900 andnational toll call prefix 90 are configured.

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It is recommended to modify the sip registration cycleEnter the “advanced> SIP signaling”, modify the “sip registration cycle” to 300s.

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There are some non-voice function in the AR product. If you want to learn more about thenon-voice function, please refer to our product document.

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AR is the products of enterprise voice-data integration solution AR series enterpriserouters (ARs) include AR150, AR160, AR200, AR510, AR1200, AR2200, and AR3200. Theyare the next-generation routing and gateway devices, which provide the routing,switching, wireless, voice, and security functions.

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VRP Versatile Routing Platform

Versatile Routing Platform (VRP) is a network Operating System (OS) used by datacommunication products of Huawei. As the core engine of the software for Huaweiproducts such as all series of routers, Ethernet switch, and service gateway, VRPprovides a uniform user interface and management interface, including the real -timeOS core, IP soft-forwarding engine, route processing and configurationmanagement planes. It provides the function of control plane, and specifies thestandard for forwarding plane interface, implementing the interaction between theforwarding plane of products and the control plane of VRP. At the same time, VRPprovides network layer interface to interconnect the link layer to the network layer.

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To provide voice services for trunk, analog or digital board is required.

To provide voice services for POTS users on AR1200, AR2200 ,and AR3200 series routers,4FXS/1FXO board is required.

To provide voice services for ISDN users on AR1200, AR2200 ,and AR3200 series routers,2BST board is required.

ProductSeries

Model Supporting VoiceCommunication

Remarks

AR150 AR157VW and AR158EVW -

AR160 AR169FVW,AR169FGVW-L -

AR200 AR201VW-P, AR207V, AR207VW, andAR207V-P -

AR1200 AR1220V ,AR1220VW -

AR2200 All models of the AR2200, excludingAR2201 and AR2202

•The AR220X and AR2220Lmust be equipped with thehardware that supports voice

communication.•The DSP module is required.AR3200 All models of the AR3200

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Slot: A position defined by a upper guide rail and the corresponding lower guide rail in aframe. A slot houses a board.

Slots can be merged into one, but one slot can not be split.

The number of the new merged slot equals to the larger one of the former slots.

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SRU: Service and Router Unit.

The SRU integrates the control and management functions and provides the control plane,management plane, and switching plane for the system.

Control plane: provides functions such as protocol processing, service processing,route calculation, forwarding control, service scheduling, traffic statistics, andsystem security.

Management plane: provides functions such as system monitoring, environmentmonitoring, log and alarm processing, system loading, and system upgrade.

Switching plane: provides high-speed and non-blocking data channels to implementservice switching between service modules.

Ethernet LAN Interface Card

Provides Ethernet access and support Layer 2 and Layer 3 line-speed switching anddevice management.

WAN Interface Card

Provide WAN Access .

There are some AR cards with non-voice function. If you want to learn more about thenon-voice function, please refer to our product document.

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The SRU40, SRU60, and SRU80 panels are identical except for having different silkscreens.

The SRU200, and SRU400 panels are identical except for having different silkscreens.

Function andFeature Description

Basic functions Stores information such as system configuration data, startup files,upgrade software, and system logs.

Voice functions SRU40, SRU60 and SRU80 provide voice functions.NOTE: SRU200 and SRU400 do not provide voice functions.

Control module

Functions as the control and management plane for the system,

implementing protocol processing, route calculation, forwardingcontrol, system management, and system security.

Switchingmodule

Functions as the service switching plane and provides high-speedservice channels to implement service switching, including voiceswitching, data switching, and conversion between voice and dataservices.

Power module Provides power for the SRU.

Clock module Provides synchronous clock signals for voice cards and local voiceswitching.

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Mapping between the SRU and software versions

Note:

Y: supported

N: not supported

Card Name AR2240 AR3260 AR3670SRU40 Y Y NSRU60 Y Y NSRU80 Y Y N

SRU100 Y Y NSRU200 Y Y NSRU400 Y Y NSRUX5 N N Y

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Card Name CPU Function

SRU40 600 MHZ CPU with eightkernels

Does not support traffic management.

SRU60 600 MHZ CPU with eight

kernels

Does not support traffic management.

SRU80 750 MHZ CPU with 12 kernels Supports traffic management.

SRU200 1.2 GHz CPU with 32 kernels Supports traffic management, two 10GE SFP+interfaces, and four GE combo interfaces.

SRU400 1.2 GHz CPU with 32 kernels Supports traffic management, two 10GE SFP+interfaces, and four GE combo interfaces.

SRUX5 2.2 GHz CPU with 6 kernels Supports traffic management.

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Number Indicator olor Description

1 STAT

Green Slow blinking: The system is running properly.Fast blinking: The system is powering on orrestarting.

Red Steady on: A fault that affects services hasoccurred and cannot be rectifiedautomatically. The fault needs to be rectifiedmanually.

Off Off: The software is not running or is beingreset.

2 ACT (interface statusindicator)

Green Stead on: The FXO channel is being occupiedby a call.

Off: The FXO channel is idle.

3 ACT (interface statusindicator)

Green Stead on: The FXS channel is being occupiedby a call.

Off: The FXS channel is idle.

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Number Indicator olor Description

1 STAT

green Slow blinking: The system is running properly.Fast blinking: The system is powering on orrestarting.

red Steady on: A fault that affects services has occurredand cannot be rectified automatically. The fault needsto be rectified manually.

off Off: The software is not running or is being reset.

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Number Indicator Color Description

1 STAT

Green Slow blinking: The system is runningproperly.Fast blinking: The system is powering on orrestarting.

Red Steady on: A fault that affects services hasoccurred and cannot be rectifiedautomatically. The fault needs to be rectifiedmanually.

Off Off: The software is not running or is beingreset.

2 ACT (interfacestatus indicator)

Green Stead on: The FXO channel is being occupied

by a call.Off: The FXO channel is idle.

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Number Indicator Color Description

1 STAT

Green Slow blinking: The system is running properly.Fast blinking: The system is powering on orrestarting.

Red Steady on: A fault that affects services hasoccurred and cannot be rectified automatically.The fault needs to be rectified manually.

Off Off: The software is not running or is being reset.

2

ACT(interfacestatusindicator)

Green Stead on: The ISDN channel has been activated.

Off: The ISDN channel in inactive.

3 B1/B2Green Blinking: The B1/B2 channel is used.

Off: The B1/B2 channel is idle.

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Functions

Sends, receives, and processes E1/T1 data traffic.

Provides CE1/T1 access.

Provides the ISDN PRI function.

Number Indicator Color Description

1 STAT

Green Slow blinking: The system software is running normally.Fast blinking: The card is loading the system software or isresetting.

Red Steady on: A fault that affects services has occurred. Thefault cannot be rectified automatically and needs to berectified manually.

Off The software is not running or the card is resetting.

2 LINK/ACT

Green Steady on: A link has been established on the interface.

Yellow Blinking: The interface is transmitting and receiving data.

Off No link is established on the interface.

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NOTICE:

Only the DSP module supported by the AR router can be inserted into the DSPDIMM slot. The DDR3 memory board or unsupported DIMMs cannot be inserted;otherwise, the AR router may be damaged or does not function properly.

NOTE:

Y: supported.

N: not supported.

Card Name AR1200 Series AR2204 AR2220 AR2240 AR3260

16/32/64/128-channelDSPmodule

N YNOTE:V200R003C00and laterversions

YNOTE:V200R001C01and laterversions

YNOTE:V200R001C01and laterversions

YNOTE:V200R001C01and laterversions

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An PBX checks the validity of a dialed number and connects or rejects the call based onthe call prefix. A call can be connected correctly only if the call matches the correct callprefix. Therefore, a correct call prefix configuration is the key to the call service. A callmust match at least one call prefix

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Short Number and Long Number

A short number is an internal number of an enterprise, and a long number isallocated by the carrier to an enterprise. Internal users of an enterprise can dial theshort numbers of each other. External users need to dial the long number of theenterprise.

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Call-right

all

ddd

idd

inter

Local

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Answer : B

AR is the products of enterprise voice-data integration solution AR series enterpriserouters (ARs) include AR150, AR160, AR200, AR510, AR1200, AR2200, and AR3200. Theyare the next-generation routing and gateway devices, which provide the routing,switching, wireless, voice, and security functions.

The license is necessary. If there are only SIP users, the voice cards are not needed , but ifthere are POTS users and Trunk for outgoing ,the cards are needed.

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Note:

To simplify the problem description, this course uses Telnet as an example todescribe the related technologies. Telnet and STelnet can be used to log in to thedevice. Using Telnet or STelnetv1 has potential security risks. STelnetv2 isrecommended.

To simplify the problem description, this course uses FTP as an example to describethe related technologies. The device can transfer files through FTP, TFTP, andSFTPv1. Using FTP and TFTP or SFTPv1 has potential security risks. SFTPv2 isrecommended.

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Versatile Routing Platform (VRP) is a network Operating System (OS) used by datacommunication products of Huawei. As the core engine of the software for Huawei productssuch as all series of routers, Ethernet switch, and service gateway, VRP provides a uniformuser interface and management interface, including the real-time OS core, IP soft-forwardingengine, route processing and configuration management planes. It provides the function ofcontrol plane, and specifies the standard for forwarding plane interface, implementing theinteraction between the forwarding plane of products and the control plane of VRP. At thesame time, VRP provides network layer interface to interconnect the link layer to thenetwork layer.

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Download Putty: www.putty.org

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This login password can be modified under the user-interface con 0 view, using command:set authentication password.

Under this view, command authentication-mode aaa is used to set user + passwordauthentication.

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The router's commands are classified into a maximum of 16 levels.

By default, the commands are classified into four levels:

Visit level. The commands include diagnostic tools, such as the ping and tracertcommands, and the commands that are used to access a remote device such asTelnet.

Monitoring level. The commands are used for system maintenance, such as displaycommands.

Configuration level. The commands are used for service configuration, includingrouting commands and commands at each network layer, to provide networkservices to users.

Management level. The commands are used for basic system operations, includingfile system, FTP, Trivial File Transfer Protocol (TFTP), Xmodem download, andconfiguration file switching commands, slave board control commands, usermanagement commands, command level configuration commands, systemparameter configuration commands, and debugging commands.

If refined right management is required, command levels can be extended to 0 to 15.

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Command line interfaces are classified into command line views. All commands must beexecuted in command line views. You can run a command only after you enter itscommand view.

# Connect to the switch. If the switch uses default settings, you enter the user view.

<Huawei>

# Enter system-view and press Enter to enter the system view.

<Huawei> system-view

[Huawei]

# Enter aaa in the system view to enter the AAA view.

[Huawei] aaa

[Huawei-aaa]

Note: The default sysname is Huawei. The prompt indicates the current view. For example,<> indicates the user view and [] indicates all other views except user view.

Some commands can be executed in multiple views, but they have different functions afterbeing executed in different views.

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There are thousands of commands in VRP system, but users do not need to rememberthem all. But users must be familiar with some of the commonly used key commands.With the help system of VRP, users can complete configuration efficiently.

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"aaa" and "password" are keywords. "AAA authentication" and "Authentication throughthe password of a user terminal interface" describe the keywords respectively.

<cr> indicates that there is no keyword or parameter in this position. You can press Enterto run this command.

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INTEGER<1-35791> describes the value range of the parameter. The value of FTP timeout(in minutes) briefly describes the function of this parameter.

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Enter the first several letters of a key word in the command and then press Tab to displaya complete key word. A complete keyword is displayed only if the partial string of lettersuniquely identifies a specific key word. If they do not identify a specific key work,continuing to press Tab will display different key words. You can select the needed keyword.

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Error Message Example:

<Huawei>display xyz

^

Error: Unrecognized command found at '^' position.

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The sysname command sets the device host name.

The undo sysname command restores the default device host name.

By default, the device host name is Huawei. But for some equipments, the defaulthost name may be Quidway.

Changing the host name affects the prompt of the command interface. For example,if the host name is Huawei, the prompt of the user interface is <Huawei>.

The display current-configuration command displays the currently running configuration.

The command does not display the default configuration.

If the output is more than 24 lines(default value), the output will be displayed morethan one page. Use <Enter> to display the next line, and use <Space> to display thenext page.

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The display version command displays the version of the device.

You can use the display version command to view the version of the device todetermine whether the device needs to be upgraded.

And the uptime may indicate whether the device rebooted recently.

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The display device command displays the type and status of the components on a device.

You can use this command to check the working status of components on a device,ensuring that the device is working normally.

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The user-interface command displays one or multiple user interface views.

VTY Virtual Type Terminal user interface Manages and controls users that log into the device using Telnet or SSH.

The first one is VTY 0, the second one is VTY 1, and so forth.

The authentication-mode command configures the authentication mode for accessing theuser interface.

After you set the authentication mode for accessing a user interface to password,run the set authentication password command to configure an authenticationpassword. Keep the password safe. You need to enter the password when loggingin to the device. The levels of commands accessible to a user depends on the levelconfigured for the user interface to which the user logs in.

The set authentication password command configures a local authentication password.

Different products may require different password strength.

The user privilege command configures the user level.

By default, users who log in to the device using the console interface are atlevel 15 and other users are at level 0.

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When the authentication mode is set to aaa, the authentication password is deleted at thesame time. Users are required to enter the login user name and password to log in to thedevice. After login, the level of the commands the user can run depends on the level of thelocal user specified in AAA configuration.

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The interface command displays the interface view or sub-interface view.

Specifies the type and number of an interface. The interface type and number canbe closely next to each other or separated by a space character.

The ip address command configures an IP address for an interface.

The mask can be a mask in full format or the length of mask.

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The display this command displays the running configuration in the current view.

After the configurations are complete in a certain view, runthe display this command to check the current configurations.

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Login in to the AR with Putty tool.

If the authentication mode is aaa (not password), the user name is required first, and thenthe password.

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1. CDE.

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Networking Description

The unified gateway at the headquarters is the central node, which can be eSpaceU1911, U1960, U1900, or U1981. The central node supports two unified gatewaysworking in active/standby mode, known as the active node and standby node.

The unified gateway at a branch is the branch node, which can be eSpace U1911,U1960, U1900, or U1981.

The central node and branch nodes all connect to the PSTN using trunks such asPRA, SS7, QSIG, R2, BRI, and AT0. Two connection modes are available:

A trunk gateway is deployed at the headquarters for connecting to the PSTNand routes call to the PSTN using trunks. The trunk gateway connects to thecentral node using SIP trunks.

A trunk gateway is not deployed at the headquarters, but the central nodedirectly connects to the PSTN. If two unified gateways working inactive/standby mode are deployed as the central node, the active node andstandby node each connect to the PSTN using trunks.

The central node and branch nodes are connected using SIP trunks. They also use a

heartbeat mechanism to constantly check each other's status.

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When the central node is correctly connected to the branch nodes,

All users at the headquarters and branches register with the central node.

Services are uniformly configured and managed on the central node, and thecentral node synchronizes data to branch nodes in real time.

All calls are processed by the central node. When a branch user places anoutgoing PSTN call, the central node analyzes the call and routes the call tothe local branch node, which then routes the call outside of the enterprise.

When the central node is faulty or disconnects from a branch node, users at thebranch register with the branch node, and the branch node processes servicerequests from users at the branch. This is known as Local-survival.

Basic call functions for analog users at the branch node are not affected.

IP phones at the branch fail to register with the central node, and thenautomatically switch to the branch node for registration. Basic call functionsfor SIP users are not affected.

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All calls are processed by the central node.

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When configuring numbers on the primary node, you do not need to synchronize POTSphone numbers from the local node; however, you must configure a gateway ID (specifiedby localgateway id) for the numbers for authentication. If you add numbers of POTS usersbelonging to the local node on the primary node, you must set the number type to SIP;however, these numbers are configured as the POTS numbers with the proxy registrationfunction enabled on the local node. As a result, different types are configured for the samenumbers on the primary node and local node. Therefore, do not enable synchronization forPOTS numbers belonging to the local node on the primary node.

Registration process:

A POTS number corresponds to a specific port number on the user board on the localnode.

The local node uses a SIP registration proxy to register all POTS users with the primarynode.

On the primary node, registration IP addresses of the POTS numbers belonging to thelocal node are displayed as the IP address of the local node.

Call process:

When a POTS user makes a call as the calling party, the local node initiates a call forthe POTS user.

When a POTS user receives a call as the called party, the primary node learns that theIP address corresponding to the called number is the IP address of the local node, andforwards the call to the local node. The local node queries the port numbercorresponding to the called number on the user board and connects the call.

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When the primary and local nodes are disconnected:

The local nodes automatically enable local survival and become independent IP PBXsfor SIP users to register with.

The status of users on local nodes is displayed as FAULT.

The basic call function is not affected for POTS users on local nodes.

When detecting that the primary node is disconnected, the local IAD automaticallysets the local node as the SIP server. The basic call function is not affected for SIPusers on the local node.

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Networking Description

The unified gateway at the headquarters is the central node, which can be eSpaceU1911, U1960, U1900, or U1981. The central node supports two unified gatewaysworking in active/standby mode, known as the active node and standby node.

The unified gateway at a branch is the branch node, which can be eSpace U1911,U1960, U1900, or U1981.

The central node and branch nodes all connect to the PSTN using trunks such asPRA, SS7, QSIG, R2, BRI, and AT0. Two connection modes are available:

A trunk gateway is deployed at the headquarters for connecting to the PSTNand routes call to the PSTN using trunks. The trunk gateway connects to thecentral node using SIP trunks.

A trunk gateway is not deployed at the headquarters, but the central nodedirectly connects to the PSTN. If two unified gateways working inactive/standby mode are deployed as the central node, the active node andstandby node each connect to the PSTN using trunks.

The central node and branch nodes are connected using SIP trunks. They also use a

heartbeat mechanism to constantly check each other's status.

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NE: Network Element

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Local call prefix 9 can be configured only after international toll call prefix 900 andnational toll call prefix 90 are configured.

config add prefix dn 900 callcategory basic callattribute idd cldpredeal no officeselectcode 0 minlen 0 maxlen 32 uselongcli yes.

config add prefix dn 90 callcategory basic callattribute ddd cldpredeal no officeselectcode 0 minlen 0 maxlen 32 uselongcli yes

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Configure call sources according to the data plan to implement the following requirements:

When a user at the central node dials 9 (outgoing call prefix) and a called number, thecall is routed through the central node. Users at the central node belong to the defaultcall source 0.

When a user at the central node dials 9 (outgoing call prefix) and a called number, thecall is outgoing to PSTN through the local node.

So we should configure two call sources for users at central node and local node.

5XXX belongs to the call source 0 it is default ,no need to be configured );

6XXX belongs to the call source 1;

Configure reroute analysis.

Configure the binding between call source and office route. When a user in call source1 dials outgoing call prefix 9 to make an outgoing call ,the call is rerouted to officeroute 1 (local node 1).

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NE: Network Element.

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Configuring Ethernert IP addresses, media IP address pools, and signaling IP address poolsfor interfaces on the PBX, so that the PBX can connect to the IP network.

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SIP server is an important entity in the SIP protocol architecture. The PBX can function asthe SIP server to accept registration information of SIP users, save the information in theaddress information database, and manage and maintain users' registration information.

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Before configuring the Automatic Switchboard, Pls refer to AR Access Router ProductsDocument to finish the configuration of IVR.

“ vu-service-name” is the “service name” configured in IVR.

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When a POTS user registers with the central node through the AR local agent, configurethe header field on the AR, enabling the unified gateway at the central node to identifythe POTS user.

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Assume that the local regeneration BIN channel authentication password is a12345678which is the same as the authentication password for adding the local node AR on theU1900 central node.

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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 TLStransmission.

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1. D.

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A distributed call management network in peer-to-peer mode can be used for enterpriseswith 10,000-30,000 users. In such a network, two or three nodes each deploy a unifiedgateway, which provides call services for the corresponding node. Different nodes areconnected using SIP trunks, through which inter-node calls are routed.

This network applies only to the U1911/U1960/U1980/U1981.

Networking Description

Each node has a unified gateway deployed. The unified gateway at a node is a voicegateway, which is responsible for user registration, intra-office users, and local PSTN

calls at that node. Different nodes are connected using SIP trunks, through whichinter-node calls are routed.

IP phones, SoftConsole users (SIP users), UC soft terminal users (SIP users), andeSpace IADs (on behalf of analog phones and fax machines) at a node register withthe unified gateway at the node.

The unified gateways at different nodes can connect to the PSTN using trunks suchas PRA, SS7, QSIG, R2, BRI, and AT0 for local outgoing calls.

The BMU, eSight, and other related service servers are deployed at the headquarters

for centrally processing services.Service servers, such as the UMS (for the unified messaging service), CDRServer (forthe CDR service), ConsoleServer (for the console service), and Audio Recorder (forthe recording service), can be deployed based on service requirements.

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This topic assumes the following number segments as the planned ones in the currentnetwork scenario:

Node 1: 80000 89999

Node 2: 60000 69999

Node 3: 70000 79999

In order to implement voice communication between different nodes, we need to configthe office route, prefix and sip trunk to other node on every node.

When the user 80001 dial 70001, the node 1 do the number analysis and route thecall to node 2, and the node 2 connect it’s internal user 70001.

When the user 80001 dial 60001, the node 1 do the number analysis and route thecall to node 3, and the node 3 connect it’s internal user 60001.

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Enterprises with more than 30,000 users can use a distributed call management network wherefour or more unified gateways are convergent.

This network applies only to the U1911/U1960/U1980/U1981.

Networking Description

Each node has a unified gateway deployed. The unified gateway at a node is a voicegateway, which is responsible for user registration, intra-office users, and local PSTN calls atthat node.

IP phones, SoftConsole users (SIP users), UC soft terminal users (SIP users), and IADs (onbehalf of analog phones and fax machines) at a node register with the voice gateway at thenode.

The voice gateways at different nodes can connect to the PSTN using trunks such as PRA,SS7, QSIG, R2, BRI, and AT0 for local outgoing calls.

The convergence gateways route inter-node calls. Different convergence gateways areconnected using SIP trunks, and each convergence gateway connects to different voicegateways also using SIP trunks. A convergence gateway can be U1980 or U1981.

Multiple convergence gateways can be deployed in active-standby or load balancing mode.

The BMU, eSight, and other related service servers are deployed at the headquarters for

centrally processing services.Service servers, such as the UMS (for the unified messaging service), CDRServer (for the CDRservice), ConsoleServer (for the console service), and Audio Recorder (for the recordingservice), can be deployed based on service requirements.

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Service RequirementsUsers at different local nodes can call each other using "area code+short numbers".

Users within a local node can call each other using short numbers or "areacode+short numbers".

Dialing area code+short number: The area code here is the local area code,and the calling method is the same as that used for calling users at the othernodes.

Dialing a short number: Users within a local node can call each other bydirectly dialing the short numbers.

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Note:The above tables not contains the data plan in the node 2 and 3, please plan thedata in node 2 and 3 refer node 1;

Before config the local prefix in the U1900, the U1900 force you to add the DDDand IDD prefix first, use it to prevent toll fraud, and the DDD and IDD prefix wedidn’t contain in the above table.

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In this network scenario, user numbers at each node are in format of "area code+shortnumber".

To implement inter-node voice communication, configure required parameters at eachlocal node and the convergent node. Specifically, at each local node, configure officeroutes, outgoing prefixes, and SIP trunks to the convergent node; at the convergent node,also configure these parameters to each local node.

When user 01080001 dials 02170001, local node 1 analyzes the number andaccording to the analysis result (021 is the call prefix to the convergent node), routes

the call to the convergent node. The convergent node analyzes the number andaccording to the analysis result (021 is the call prefix to local node 2), routes the callto local node 2. Then, local node 2 connects the call to user 05718004.

When user 01080001 dials 02560001, local node 1 analyzes the number andaccording to the analysis result (025 is the call prefix to the convergent node), routesthe call to the convergent node. The convergent node analyzes the number andaccording to the analysis result (025 is the call prefix to local node 3), routes the callto local node 3. Then, local node 3 connects the call to user 02560001.

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Toll Fraud Scenarios

Transfer by the automatic switchboard

As shown by line (1) in Figure, outer-office user B dials the automatic switchboard number ofthe company with U1900 unified gateway A deployed, and dials user C's number when hearingan announcement or a two-stage dialing tone. If the call is connected by the automaticswitchboard, user B is charged only for a local call, while the company must pay the toll callfee.

Transfer through a trunk

As shown by line (2) in Figure, outer-office user C dials the incoming prefix configured onU1900 unified gateway A, the outgoing prefix for making calls from U1900 unified gateway Ato U1900 unified gateway B, and a toll number in sequence to make a toll call. For example, if0 is the incoming prefix configured on U1900 unified gateway A, and 00 is the outgoing prefixfor making calls from U1900 unified gateway A to U1900 unified gateway B, user C can dial0+00+123456789 to make a call to toll number 123456789.

Unauthorized registration

As shown by line (3) in Figure 1, an authorized user who is registered with U1900 unifiedgateway B by impersonating a registered user makes a call to outer-office user D. The call iscompletely free of charge to the unauthorized user.

Misuse of call forwarding services

As shown by line (4) in Figure 1, user B illegally obtains the administrator account andpassword for logging to U1900 unified gateway A or user A's self-service account andpassword, and enables the call forwarding unconditional (CFU) service for user A with user C'snumber configured as the forwarded-to number. When user B makes a call to user A, the call isforwarded to user C regardless of user A's status. User B pays only the local call fee to getconnected to user C, while the company must pay the toll call fee.

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You can configure multiple outgoing call rights for a trunk. By default, a trunk has thelocal call rights. If you want to forbid the outgoing call rights for trunk tandem callsthrough a trunk, you are advised to configure only the inter call right for the trunk so thatan outer-office user who makes an incoming call through the trunk cannot make a call toanother outer-office user. To allow an outer-office user who makes an incoming callthrough the trunk to make a local call, you can use the default setting. To allow an outer-office user who makes an incoming call through the trunk to make a national orinternational toll call, you must add the ddd or idd call right to the trunk.

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For example, to allow users to dial only extension numbers whose length does not exceed4, you must change the extension number length to 4 in the automatic switchboard script.

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Notice:

The default user name and password of the self-help service are User Name andChange_Me . To ensure password security, change the password periodically.

The SIP parameter of the Defense Against Brute Force Registration:

Regfailedcnt: Maximum number of registration failures allowed within within 1minute.

value type/range:1~50

Default:5

Regfreezeinterval freezen interval when user fail registration times exceed theregfailedcnt value.

value type/range:0~3600,sec

Default:300

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In this configuration example, the following requirement needs to be met: Only users withIP addresses within the range of 10.10.10.10 to 10.10.10.20 are allowed to log in to theU1900 unified gateway.

config add trust loginip startip 10.10.10.10 endip 10.10.10.20

config trust loginip switch on

Notice:

If the whitelist function is enabled but no IP address is added to a whitelist or the

network between the device and the IP addresses in the whitelist is unavailable, youwill fail to log in to the device through the IP network. In this case, log in to thedevice using the serial port, run the config trust loginip switch off command todisable the whitelist function, modify the whitelisted IP addresses as required, andenable the whitelist function again.

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Deploying the CDR Server

You can query call fraud information on the CDR server when a fraud occurs.

NOTE:

The show system iddlimitnum command can be used to query the configurationand lock status. If the international toll call rights are locked, the administrator canrun the unlock system iddlimit command to unlock the rights.

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An enterprise deploys a session border controller (SBC) to bridge the enterprise intranet,demilitarized zone (DMZ), and extranet. Enterprise users on the extranet can register withthe unified gateway on the enterprise intranet through the SBC (as a proxy). Aftersuccessful registration, enterprise users can work efficiently on the extranet as they do onthe intranet.

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The SBC mainly fulfills the following functions:

Signaling proxy Works as the terminal registration proxy gateway to register UCterminals with the SIP server.

Media aggregation Aggregates media for establishing sessions between two UCterminals or between a UC terminal and the UC system on two isolated networks.

Network Address Translation (NAT) traversal Provides the NAT traversal functionwhen SIP messages need to be traversed by NAT (only the ACME SBC provides thefunction).

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The following describes the authentication and registration principles of a UC terminal onthe extranet:

eSpace Desktop sends an authentication request to the eServer through the firewall.

The eServer sends the SBC's external IP address corresponding to the SIP server(which is the U1900 series unified gateway) back to eSpace Desktop through thefirewall.

eSpace Mobile or eSpace Mobile HD sends an authentication request to the MAA inthe DMZ through the firewall.

The MAA sends the SBC's external IP address corresponding to the SIP server backto eSpace Mobile or eSpace Mobile HD through the firewall.

A UC terminal sends a registration request to the SBC's external IP address.

The SBC, through the external SIP proxy registration port, forwards the registrationrequest to the internal SIP server for registration.

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Prerequisites

You have configured the basic network settings for the SBC.

You have obtained the password for logging in to the SBC.

Background

When the SBC and the firewall are deployed in parallel, the intranet and extranet IPaddresses need to be configured for the SBC.

When the SBC is deployed behind the firewall, only the intranet (DMZ) IP addressneeds to be configured for the SBC.

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eSpace Desktop, eSpace Mobile, or eSpace Mobile HD on the extranet sends a registrationrequest to the firewall's external IP address. The registration request contains the portnumber and the terminal's IP address. The firewall transparently transmits the registrationrequest to the service server on the intranet. Upon receiving the request, the service serverdetermines the area to which the UC terminal belongs according to the UC terminal's IPaddress, queries the server IP address of the corresponding service in the area, and returnsthe IP address to the UC terminal.

The BMU determines the area to which a terminal belongs according to the IP addressreported by the terminal. If a terminal does not belong to any area, the policy for thedefault area is used.

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Functions of devices in a console system:

SoftConsole

Provides basic call functions as phones do, such as placing, answering, and holding

calls.Provides agent functions, such as showing busy, showing idle, and forwarding calls.

ConsoleServer

Saves information about agents.

Authenticates agents during their login.

Queues incoming calls, maintains such queues, and updates queue information toeach SoftConsole.

Unified gateway

Handles SIP user registration of the SoftConsole.

Connects calls between agents and users.

If a user attempts to log in to the SoftConsole, the ConsoleServer first authenticates theuser. After the user is successfully authenticated, the user is registered with the unifiedgateway.

When a user dials a SoftConsole group access code, the unified gateway routes the call toan idle agent in the group who will then answer the call. If all agents in the group arebusy, the system queues incoming calls. When one or more agents become available, thesystem routes the calls to the idle agent(s) following the "first-come, first-served" policy.

Notice

A SoftConsole group access code is a hotline number provided by an enterprise, bydialing which users can be routed to agents.

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Enterprise A releases the SoftConsole hotline 28887890 which is used to process internaland external calls.

Enterprise A poses the following requirements on the SoftConsole:

A SoftConsole group, including three agents, must be configured.

The supervisor can forcibly release and barge into agents' calls.

Data Plan

Agent Requirement AgentNumber

AgentSoftConsole

Number

Agent UserName/Passw

ord

Agent Tom •Agents and the supervisorare in the same SoftConsolegroup.•Only the supervisor isassigned the barge andforced-release service rights.

81001 81001 Tom/m12345

Agent John 81002 81002 John/n12345

SupervisorLucy

81003 81003 Lucy/y12345

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The default user name and password for the ConsoleServer administrator are admin andhuawei123 .

eSpace Console System

Parameter DescriptionUnified Gatewaylistening

Use the default value 9501 .

SoftConsolelistening port

Use the default value 9500 .NOTE: If you do not use the default port but customize one, ensure thatyou will use this customized one when configuring the interworkingbetween the SoftConsole and the ConsoleServer.

Synchronize time Select it to ensure that the waiting time of incoming calls on theSoftConsole is consistent with the mapping record on the unifiedgateway.NOTE: When you select this option, the operating system time of theconsole server is not affected.

every The console server reads the unified gateway time every 64 seconds.[Default Value] 64

SNTP Server IP Enter the IP address of the SNTP server.

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Parameter DescriptionUser lock Set the maximum number of consecutive incorrect password inputs. The

value ranges from 0 to 10. The default value is 3.If the number of consecutive incorrect password inputs reaches themaximum, the user account will be locked for a specified period of time

(value range: 0-30 minutes; default value: 5 minutes).The system automatically unlocks the user account after a specified periodof time. Then, you can enter the account password again.NOTE: When a user account is locked, the administrator can manually unlockthe user account. To unlock a user account, select the user account on theMaintain User Info page and click Unlock .

Use AppAgent •The SoftConsole connects to the BMU through the AppAgent. It obtains thecorporate directory from the BMU and enables the Night Service, Searchrecord, DND, Wake Up, Reset, and Call Rights services. To connect to theBMU, select this option.•To disconnect from the BMU, clear this option.

HTTP The SoftConsole uses the HTTP or HTTPS protocol to access the AppAgent.NOTE: Using HTTP to transfer files may bring some security risks becauseHTTP is an insecure protocol. It is recommended that you use the HTTPSprotocol.

HTTPS

AppAgent IPaddress

Set it to the IP address of the AppAgent.

Port •HTTP: 80•HTTPS: 8443

User name Set the AppAgent user name and password on the BMU.The default user name is AppAgent , and initial password is 1qaz@WSX .NOTE: Log in to the BMU as the admin user, choose Functions > eSpaceMaintenance .Here, Service ID and Password are the AppAgent user name and password.

Password

Use FTP •If this parameter is selected, recording files of agents are stored on aspecified server.•If this parameter is deselected, recording files of agents are stored on theserver where the SoftConsole is located. The storage directory is theSoftConsole installation directory.

FTP Use the FTP or FTPS protocol to store recording files of the SoftConsole on aspecified server.NOTE: Using FTP to transfer files may bring some security risks because FTPis an insecure protocol. It is recommended that you use the FTPS protocol.

FTPS

FTP IP address Set it to the IP address of the server where recording files are stored.Port Use the default value 21 .User name Set the user name and password for FTP or FTPS service.

The user name and password are configured during FTP or FTPS serviceinstallation.

Password

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If active and standby unified gateways exist, add IP addresses of the active and standbyunified gateways. In the Unified Gateway IP Address area, right-click a configured unifiedgateway IP address and set the unified gateway as the active or standby unified gateway.

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Agent informationParameter Description

User type Different user types have different console rights.•Administrator: Can view the status of and configure services for all agentswithin an enterprise domain.•Supervisor: Can view the status of and configure services for agents only in thesame group as the supervisor.•Operator: Can view the status of agents only in the same group as the operator.•Console Server Administrator: Can log in to the ConsoleServer to manageSoftConsole accounts. This type of user cannot log in to the SoftConsole.Select a user type based on the console service data plan.

User name Enter the user name for logging in to the SoftConsole.

Password

Unified Enter the IP address of the unified gateway where the SoftConsole group resides.

Agent group Enter the ID of the SoftConsole group to which the agent belongs.

Bound EID Select the agent number, 81003 for example, when the equipment ID (EID) is thesame as the user number. If the EID is different from the user number, obtainthe EID from the user number plan table and select this EID.

RecordPermission

Set whether to enable the recording function for the agent. The recordingfunction is provided by the SoftConsole.The media engin of the agent's PC records each call between the agent and usersfor the convenience of the query. The record files are saved on the agent's PC orthe FTP/FTPS server. The storage policy and time are determined by the agent.The system will remind users that the call will be recorded.

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Parameters for connecting the SoftConsole to the ConsoleServerParameter Description

Server IP Enter the ConsoleServer IP address.

Server port Enter the value of SoftConsole listening port specifiedin Configuring the Interconnection Between the ConsoleServerand the Unified Gateway . The default value 9500 isrecommended.

Local IP Use the default value Auto .

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Using the BMU server, agent leaders can perform particular service management andconfigurations.

Call rights configuration An administrator can configure intra-office numbers' callrights, including: intra-office call, local call, national call, and international call.

Setting Do not disturb (DND) An administrator can configure the DND service forintra-office users.

Wake-up service An administrator can configure the wake-up service or VIP wake-up service for a specified number based on the time segment, and collect wake-uprecords.

One-click restoration All services configured for a specified number can berestored to the initial settings using one key.

Night service After the night service is set, all calls to the Console system duringthe specified time segments will be forwarded to a preset night service extension.

Call record querying An administrator can query CDR information about specifiednumbers. Detailed CDR information includes the charge details, summary, callingnumbers, number of calls, call duration, call time, called numbers, and call types.

User information modification An administrator can modify user informationdetails in the corporate directory.

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Unified gateway: Provides number registration and call connection functions, exchangesSIP signaling with the SIP Server for sending requests to start and stop recording, andsends RTP/SRTP streams to the recording system.

DataBase: Stores recording server parameters and recording data.

SIP Server: Exchanges signaling with the unified gateway for receiving requests to startand stop recording, and sends the requests to the recording server.

IMP: Centrally manages recording servers and configures recording server parameters.

Recording server: Receives RTP/SRTP streams sent from the site, analyzes recordinginformation based on the requests sent by the SIP Server, and generates recording filesand data.

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If automatic recording is enabled for a user, the system automatically recordsconversations of this user.

If recording on demand is enabled for a user, the user can press the recording button torecord conversations.

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If automatic recording is enabled for a user, the system automatically recordsconversations of this user.

If recording on demand is enabled for a user, the user can press the recording button torecord conversations.

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The database, Log Server, and HWLog IMP can be installed on the same server.

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Here the "single" refers to the IMP server, the database server machine, without the use of "eSpaceAudio Recorder V100R001C02SPC300 IP and SQL database floating cluster configuration guide 04"deployment of cluster server, recording can be single or single above.

eSpace U1900 Series Gateway

Interacts with the SIP Server through signaling, sends call information and requests for tarting andstopping the recording process to the SIP Server, and sends RTP /SRPT streams to the Log Servercluster.

SIP Server

SIP Server interact with eSpace U2900 Series Gateway , send requests to the Log Servercluster forstarting or stopping the recording process based on the Log Server load to implement load balancing

in the Log Server cluster .

Log Server Colony

Receives RTP/SRTP streams from eSpace U2900 Series Gateway using multiple Log Servers, startsrecording and stops recording the message recording information based on the SIP Server, andgenerates and stores recording files and recording data. A single Log Server failure does not affect therecording of the entire cluster .

Database

Stores Log Server parameters and recordings.

HWLog Integrated Management Platform (IMP)

Manages and configures the Log Server parameters.

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Here the “dual” refers to the IMP server, database server system, need to use the “eSpace AudioRecorder V100R001C02SPC300 IP and SQL database floating cluster configuration guide 04 ” deployment of cluster environment, the recording server can be two or more than two.

eSpace U1900 Series Gateway

Interacts with the SIP Server through signaling, sends call information and requests for tarting andstopping the recording process to the SIP Server, and sends RTP /SRPT streams to the Log Servercluster.

SIP Server

SIP Server interact with eSpace U2900 Series Gateway , send requests to the Log Servercluster forstarting or stopping the recording process based on the Log Server load to implement load balancing

in the Log Server cluster .

Log Server Colony

Receives RTP/SRTP streams from eSpace U2900 Series Gateway using multiple Log Servers, startsrecording and stops recording the message recording information based on the SIP Server, andgenerates and stores recording files and recording data. A single Log Server failure does not affect therecording of the entire cluster .

Database

Stores Log Server parameters and recordings.

HWLog Integrated Management Platform (IMP)

Manages and configures the Log Server parameters.

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the external virtual IP address refers to the cluster cases need to enter the IP address, if asingle deployment, enter the IMP server is the physical IP address.

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If the software has been installed before, you need to restart the computer and chooseStart >All Programs > Remove HWLogIMP to remove the software, and then do asfollowing steps to reinstall it.

Click Install to begin installation components, pop-up certificate password input box,enter the password which is set at the step of configuring HWLog IMP.

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The Voice Management Center (VMC) manages the entire eSpace Audio Recordersystem.

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Parameter Description

Database Database used to store the Log Serverinformation.The default database is Microsoft SQLServer 2008 .

Server Cluster deployment, please fill in thedatabase wherethe cluster database's virtual IP address,the default forthe cluster's virtual IP address; singledeployment, no

foreign virtual IP address, please fill in theserver ofthe physical address of the IP database.

Port Port number for accessing the database.The default port number is 1433 .

Description VMC server information.

User User name for logging in to the database.

Password Password for logging in to the database.

Create Database if there doesn't exist Select this check box to automaticallycreate a VMCdatabase if the database does not exist.

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Parameter Description

Number of Channels According to current dongle information andnumber of channels to add. A channel correspondsto an extension.

Use the same parameter Configured extension whose parameters you wantto use for new channels.This function allows you to quickly copy theconfigurations to new channels.

Extension starts at Extension start number to be

entered when you add channel

Extension start number to be entered when you

add channelsfor a voice server for the first time.Only digits are allowed. The extension numbersincrease inascending order.

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On the eSpace Audio Recorder page that is displayed, choose Basic Function > Logger Management in the navigation tree on the left.

Double-click the VMC server under the organization ( Huawei Technologies Co., Ltd. ).The VMC parameter list is displayed in the lower right corner of the page window.

Host

SIP server IP address.

Sip Extension

Sip Access code, the value must be consistent with the value of recording accesscode in the U1900 parameter.

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The outgoing call prefix is also the recording system access code and is configured on theunified gateway side. It is recommended that the prefix be different from other prefixes.

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The recording instruction IDs are fixed. Do not change them.

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In startdtmf 4000 , 4000 indicates the DTMF code for starting recording.In enddtmf 4001 , 4001 indicates the DTMF code for stopping recording.

The recording access code must be consistent with the value of SIP access code in theaudio recording system.

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To allow a user to record conferences held on the BMU, assign the conference recordingrights to the user on the BMU.

Log in to the BMU as the admin user.

Choose UC Account > Role , modify service rights for the user, and select Meetingrecording .

Click OK.

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. To assign the call recording service rights to an eSpace 7900 series IP phone, perform thefollowing steps:

Log in to the web management page of an eSpace 7900 series IP phone as theadmin user.

Choose Advanced > Service Rights Management .

Set Authorized for Call Recording to Yes .

Click Save .

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Routine maintenance serves the following purposes:

Discovering alarms generated by the equipment or existing defects of the equipmentin time, and accordingly taking proper measures. This ensures the stability of theequipment and lowers the rate of failure.

Recognizing abnormal cases that occur during the running of the charging systemand handling it in time. This reduces economic loss due to loss of bills.

Grasping the running statuses of both equipment and network in real time andidentifying the running tendency of both equipment and network. This improves theefficiency of maintenance staff in handling emergencies.

Carrying out periodic checks, backup, tests and cleaning for identifying the defectsin the equipment such as, natural aging, function failure, and deterioration ofperformance; and taking proper measures to eliminate these defects.

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Item Action Expected Result

Firepreventioncondition

Check the power cable andconnector of the powermodule.

No hidden danger of fire exists.Firefighting devices are in goodcondition.

Dust-proofcondition

Check the external surfaces ofthe device, the computer, thefloor and the desktop.

Each position is dust-free.

Theft proofcondition

Check the doors, windows,and theft-proof iron net.

All the facilities are in goodcondition.

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Alarms on LMT can be categorized into:

Critical

Major

Minor

Warning

Uncertain

Cleared

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If trunk circuit is in ISOLATE state, Check the following items:

Run the config cancel isolate board/card command to cancel the isolation of theboard.

Verify that the board where the trunk circuit is located is normal.Verify that the connection of the E1 trunk line is correct. That is, verify that the local-end TX is connected to the peer-end RX, and the local-end RX is connected to thepeer-end TX.

If PRA links is in fault state, Check the following items:

The board where the PRA links resides is normal.

The E1 trunk line is connected correctly (the local-end TX is connected to the peer-

end RX, the local-end RX to the peer-end TX).The PRA links is configured correctly (if the local-end link is user, the peer-end link isnetwork; if the local-end link is network, the peer-end link is user).

If subscriber is in fault state, Check the following items:

If a subscriber is in BLOCKED state, run the unblock subscriber command to unblockthe subscriber.

If a subscriber is in LOCKED state, check whether the subscriber does not dial any

number after hook-off or the subscriber does not hang up when a call ends. If thesubscriber does not dial any number after hook-off or the subscriber does not hangup when a call ends, hang up, and then check whether the fault is rectified. If thefault is caused because of other reasons, check the circuit to determine whether ashort circuit occurs.

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Monthly maintenance refers to the tasks carried out by the maintenance staff every month.It covers the tasks on shelf, terminal system, and spare parts.

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The temperature in the warehouse is 0 ° C–45 ° C.

The relative humidity in the warehouse is 5% –95%, without condensation.

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Yearly maintenance refers to the tasks carried out by the maintenance staff every year.

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To back up the data, do as follows:

Step 1: Use the save command to save the current configuration and service data to theFLASH memory.

Step 2: When the system prompts "Saving flash in the background finished!" (it meansthe saving is finished), use the config upload file data command to back up the datain the FLASH memory to the load directory on the TFTP server.

Step 3: After successful backup, check that the data.bin file is contained in the loaddirectory on the TFTP server. The data.bin file is the backup file of the database.

Step 4: Rename the data.bin file, back it up for restoring the data when necessary.

To restore the data of the specified time, load the backed-up data file to the host. Torestore the data, do as follows:

Step 1: Store the data file to be imported in the load directory on the TFTP server. Thedata file need to be named data.bin. If not, you need to rename it. Otherwise, youwill fail to import the data.

Step 2: Use the config download file data command to load the data to the FLASHmemory.

Step 3: After running the config download file data command successfully, run thereboot command to reboot the system so that the restored data takes effect.

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Ensure that the hostip has been configured as the IP address of TFTP server and thedirectory is rightly set.

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In the eSpace UC solution, the U1900 series unified gateways works with the CDRServer togenerate call detail records (CDRs) for the Business Management Unit (BMU) or a third-party billing system to obtain. The CDR system consists of the unified gateway andCDRServer. The U1900 series unified gateways generates CDRs for all users connected to itand saves the CDRs to its CDR pool. The CDRServer obtains the CDRs, parses the CDRs togenerate CDR files, and saves them to the local disk.

Functions of devices in a CDR system:

Unified gateway: Generates CDRs for all users connected to it and saves the CDRs toits CDR pool. The unified gateway periodically pushes CDRs to the CDRServer in anunsolicited manner or upon receiving the CDRServer's request.

CDRServer: Parses the obtained CDRs and stores them as binary files ( .bill) to aspecific location on the CDRServer. The CDRServer can proactively send a request tothe unified gateway for obtaining CDRs after CDR transfer is enabled under CDRConsole > Call record transfer switch of the CDRServer.

BMU: Obtains call records from the CDRServer through an FTPS interface, and writesthe CDRs into the database. The administrator can manage CDRs on the BMU, suchas querying and deleting CDRs. Enterprise users can query their own CDRs on theBMU self-service platform.

Third-party billing system: Obtains CDR files from the CDRServer through an FTP orFTPS interface.

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For detailed configuration steps, please refer to the corresponding lab guide .

Key configuration points

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Network Type Configuration on the Unified GatewayConfiguration on

the CDRServer

Single-nodeEnable the function of generating CDR files and configure theCDRServer IP address on the unified gateway at a single node.

Configure IP addressof all unifiedgateways.

Multi-nodecentralized

Enable the function of generating CDR files and configure theCDRServer IP address on the unified gateway at each node.•In normal network conditions, numbers of all branches areregistered with the central node, the central node routes allcalls, and the central node generates CDRs and send them tothe CDRServer. Unified gateways at branch nodes do notgenerate CDRs.•In case of local regeneration, a branch node disconnects fromthe central node. The unified gateway at the branch nodegenerates CDRs and buffers them in their own CDR pool. Afterthe network recovers, the unified gateway at the branch nodesends the buffered CDRs to the CDRServer.

Multi-nodedistributed(peer-to-peer)

Enable the function of generating CDR files and configure theCDRServer IP address on the unified gateway at each node.

Multi-nodedistributed(convergent)

Enable the function of generating CDR files and configure theCDRServer IP address on the unified gateway at each node(excluding the convergent gateway).

Configure IP addressof all unifiedgateways (excludingthe convergentgateway).

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In the third column X/Y, X indicates the number of used resources, and Y indicates thetotal number of requested resources.

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SlotID: ID of the slot where the SRU is inserted. The value is as follows:

AR207V, AR1220, and AR2220: 0AR2240: 11

AR3260: 15

DSP-Index: index number of the DSP chip. The value is 0, 1, or 2.

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When we initial an outgoing call go through AT0 trunk failed, we can use the commandto query the AT0 interface status to analysis.

Check the parameter " PTPSrvState ", it means the interface status.

Fault

Confirm the physical telephone line is connect to the FXO port;Confirm the physical telephone line works well, we can connect the

line to analog phone to test. Normal : It means the AT0 interface and telephone line works well, and if

you also can’t make a call use the AT0 trunk, please check the configuration

about callprefix and confirm if there is a outgoing callprefix bound to theAT0 trunk.

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When we initial an outgoing call go through E1 PRI trunk failed, we can use thecommand to query the E1 PRI interface status to analysis.

Check the parameter " PTPSrvState ", it means the E1 PRI physical line connectionstatus

Line abnormal

Confirm the physical connectivity of the E1 line; Confirm the transmit connectoron one E1 line connect to receive connector on the other E1 line.

Normal : It means physical connectivity of the E1 line works well.

Check the parameter " CTPDState ", it means the E1 PRI Layer 2 connectionstatus.

Release

It means both sides of the connection are configured as the same role, user ornet. Use the following command to modify it:

trunk-group name [ dss1-net | dss1-user | qsig-net | qsig-user ]

Establish : It means the layer 2 established. And if you also can’t make a call

use the PRI trunk, please check the configuration about callprefix andconfirm if there is a outgoing callprefix bound to the PRI trunk. If theCTPDState is none, means not configure the E1 PRI trunk, please use thefollowing command to configure:

trunk-group name [ dss1-net | dss1-user | qsig-net | qsig-user ]

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If the sip user register to AR failed, we can check it use the following ways:

Confirm the network between sip phone and AR device is reachable;Check the register information(sip server ip, port, and domain name) configured onthe sip phone the same as the configuration on the AR device;

Make sure that you already reset the sipserver service after configure it on the ARdevice;

Confirm the SIPUE account and password the same as the configuration on the ARdevice.

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When we initial an outgoing call go through SIP trunk failed, we can use the commandto query the SIP trunk group status to analysis.

Check the parameter "SIP trunk group state“

Wait ack

Check peer sip trunk ip is reachable;Confirm the peer sip trunk ip and port is correct and reset the sip

trunk group after configure it.Normal: It means the sip trunk group works well, and if you also can’t makea call use the sip trunk group, please check the configuration about callprefix

and confirm if there is a outgoing callprefix bound to the sip trunk group.

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When we initial an outgoing call go through SIP AT0 trunk failed, we can query the SIPtrunk group and SIP AT0 trunk status to analysis.

1. Check SIP trunk group status, if normally, then please check the SIP AT0 trunkstatus;

2. Use the command " displace voice trunk-sipat0 (brief) " to query SIP AT0 trunkstatus, see the parameter " Register status " or "Reg Status “

Wait ack

Check peer sip trunk ip is reachable.

Confirm the peer sip trunk ip and port is correct and reset the siptrunk group after configure it.Normal: It means the sip trunk group works well, and if you also can’t makea call use the sip trunk group, please check the configuration about callprefixand confirm if there is a outgoing callprefix bound to the sip trunk group.

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[message-type ]: There are messages exchanged between modules and packets receivedand sent from external devices.

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Emergency faults have a great impact, are of great urgency, and cause great direct orindirect loss.

An emergency fault occurs if any of the following symptoms appear:

Most or all users in an office cannot use certain call functions properly. For example,users cannot answer calls properly.

Most or all users in an office cannot use the SMS function.

Users cannot log in to eSpace terminals.

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Understanding the troubleshooting process helps you quickly locate and handle faults.Statistics show that most faults have only one fault source. Therefore, accurately locatingthe fault source is the key to handle a fault.

Procedure Description

Analyze the faultsymptom.

Analyze the fault symptom to determine the fault type andseverity, and rectify the fault based on the troubleshootingprocedure.Rectify the fault

by referring totroubleshootingcases.

Performemergencymaintenance.

•If a fault is an emergency fault, rectify the fault based on theemergency maintenance process.•If no emergency maintenance guide is provided for the product,contact technical support.

Contact technicalsupport.

If the fault persists after attempting the methods described in thisdocument, refer to the fault information collection guidance tocollect all onsite fault information and contact technical support.

Rectify the faultunder remote or

onsite guidance.

Rectify the fault under remote or onsite guidance of technicalsupport engineers. NOTE: If technical support engineers are not

onsite, rectify the fault under their remote assistance.

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Port 2020 is the default port of the CDRServer, enabling the communication between theCDRServer and U1900. The U1900 report call data to the CDRServer through port 2020.

If the CDRServer uses another port, check the port occupation based on actual situation.

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The BMU synchronizes time from the U1900 at an interval. After changing the system timeof U1900, you need to restart the BMU so that the time takes effect for scheduledmeetings immediately.

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Number

3WC: Three-Way Calling

A

AC: Alternative Current

AAC-LD: Advanced Audio Coding with Low Delay

ACD: Automatic Call Distributor

AG: Access Gateway

AN: Access Network

AR: Access Router

ARP: Address Resolution ProtocolB

B2BUA: Back-to-Back User Agent

BEST: Branch Exchange for Survivable Telephony

BHCA: Busy Hour Call Attempts

BHCC: Busy Hour Completed Call

BLF: Busy Lamp Field

BMU: Business Management Unit

BRA: Basic Rate Access

BS: Billing System

C

CAS: Channel Associated Signaling

CC: Conference Call

CDMA: Code Division Multiple Access

CDR: Call Details Records

CFB: Call Forwarding Busy

CFNR: Call Forwarding No Reply

CFO: Call Forwarding Offline

CFU: Call Forwarding Unconditional

CTVMNR: Call Transfer to Voice Mailbox on No Reply

CTVMU: Call Transfer to Voice Mailbox Unconditional

Appendix I. Glossary

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CTVMB: Call Transfer to Voice Message on Busy

CH: Call Hold

CLI: Command Line Interface

CLIP: Calling Line Identification Presentation

CLIR: Calling Line Identification Restriction

CLIRO: Calling Line Identification Restriction Override

CNIP: Calling Name Identification Presentation

CNIR: Calling/Connected Name Identification Restriction

CODEC: Code And Decode

COLP: Connected Line Identification PresentationCONP: Connected Name Identification Presentation

CRC: Cyclic Redundancy Check

CT: Call Transfer

D

DC: Direct Current

DID: Direct-Inward-Dialing

DDI: Direct-Dialing-In

DND: Do-Not-Disturb

DSP: Digital Signal Processor

DSS1: Digital Subscriber Signaling No.1

DTMF: Dual Tone Multiple Frequency

E

EMC: Electromagnetic Compatibility

EMS: Element Management System

ESD: ElectroStatic Discharge

F

FXS: Foreign eXchange Station

FXO: Foreign eXchange Office

G

GUI: Graphical User Interface

Appendix I. Glossary

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I

IAD: Integrated Access Device

IPCC: IP Contact Center

IPT: IP Telephony

ISDN: Integrated Services Digital Network

IVR: Interactive Voice Response

L

LAN: Local Area Network

LDAP: Lightweight Directory Access Protocol

LE: Local ExchangeLMT: Local Maintenance Terminal

M

MAS: Mobile Agent Server

MGC: Media Gateway Controller

MGW/MG: Media Gateway

MoH: Music on Hold

N

NGN: Next Generation Network

O

OMU: Operation Maintenance Unit

ONLY: One Number Link You

P

PBX: Private Branch Exchange

PCM: Pulse Code Modulation

PGND: Protection Ground

POTS: Plain Old Telephone Service

PRA: Primary Rate Adaptation

PRI: Primary Rate Interface

PSTN: Public Switched Telephone Network

Appendix I. Glossary