Clearspan Enterprise Guide

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ENTERPRISE SOLUTIONS GUIDE 2739-001

Release 14.0

Document Revision History

Release Version Reason for Change Date Author

14.0 1 Aastra Intecom Rebranded April 26, 2007 D.Woelfle

CLEARSPAN ENTERPRISE SOLUTIONS GUIDE AASTRA - 2739-001

2007 AASTRA NTECOM PRELIMINARY PAGE 2 OF 55

Copyright Notice

Copyright © 2007 Aastra USA Inc.

All rights reserved.

Any technical documentation that is made available by Aastra USA Inc. is proprietary and confidential and is considered the copyrighted work of Aastra USA Inc.

This publication is for distribution under Aastra USA Inc. non-disclosure agreement only. No part of this publication may be duplicated without the express written permission of Aastra USA Inc., 2811 Internet Blvd, Frisco, TX 75034.

Aastra USA Inc. reserves the right to make changes without prior notice.

Trademarks

Aastra® is a registered trademark of Aastra Technologies, Ltd.

BroadWorks® is a registered trademark of BroadSoft, Inc.

Printed in the United States of America.



Table of Contents

1 Purpose and Audience..........................................................................................9

2 Overview ................................................................................................................9 2.1 What is Clearspan? ........................................................................................9 2.2 Why choose Clearspan?................................................................................9 2.3 What is in the Clearspan Enterprise Solution? .......................................... 11 2.4 Key Business Applications.......................................................................... 12

3 Standard Deployment Models.............................................................................13 3.1 Single Site Deployment Model.................................................................... 13 3.2 Multi-Site Deployment Model...................................................................... 15

4 Key User Applications .........................................................................................20 4.1 Clearspan Communicator Family ............................................................... 20

4.1.1 Clearspan Communicator ................................................................. 20 4.1.2 Clearspan Communicator - Multimedia............................................ 21

4.2 Clearspan Assistant Family......................................................................... 22 4.2.1 BroadWorks Assistant – Enterprise.................................................. 22 4.2.2 Clearspan Assistant - Mobile ............................................................ 24

4.3 Personal Web Portal.................................................................................... 25 4.4 Unified Messaging ....................................................................................... 27 4.5 Contact Centers........................................................................................... 28 4.6 Auto Attendants ........................................................................................... 30

4.6.1 Clearspan Receptionist ..................................................................... 30 4.7 Web Conferencing....................................................................................... 32 4.8 Clearspan Deployment Studio.................................................................... 34

5 Integrating with Legacy Equipment.....................................................................34 5.1 PBX Solution (SIP Trunking)....................................................................... 34

5.1.1 Network Layout.................................................................................. 34 5.1.2 Services.............................................................................................. 35 5.1.3 Services for PBXs without DID ......................................................... 36

6 Clearspan Service Delivery Platform ..................................................................39 6.1 Network Elements ....................................................................................... 39

6.1.1 Application Server.............................................................................. 39 6.1.2 Network Server .................................................................................. 40 6.1.3 Media Server...................................................................................... 41 6.1.4 Web Server ........................................................................................ 42 6.1.5 Conference Server............................................................................. 43

6.2 Call Scenarios.............................................................................................. 44 6.2.1 Registration ........................................................................................ 44 6.2.2 Call Origination................................................................................... 44



6.2.3 Call Termination................................................................................. 44 6.2.4 Entry / Exit Point................................................................................. 45 6.2.5 Media Resources............................................................................... 45

7 Access Devices ...................................................................................................45 7.1 Access Gateways........................................................................................ 46 7.2 SIP Phones.................................................................................................. 46 7.3 Soft clients.................................................................................................... 46

8 Network Gateways ..............................................................................................46

9 Network Infrastructure Components ...................................................................47 9.1 LAN Requirements ...................................................................................... 47 9.2 Using DHCP................................................................................................. 49 9.3 Using DNS ................................................................................................... 49 9.4 Using TFTP.................................................................................................. 50

10 Operations, Administration, Maintenance and Provisioning .......................50 10.1 Configuration and Provisioning ................................................................. 50

10.1.1 Initial System Configuration............................................................. 51 10.1.2 Creating an Enterprise..................................................................... 53 10.1.3 Creating a Group ............................................................................. 54 10.1.4 Configuring Translations and Routing ............................................ 55 10.1.5 Bulk Provisioning ............................................................................. 55



Figure 1 Fully Integrated Communication Suite .................................................................................. 10 Figure 2 Enterprise Solution ................................................................................................................... 12 Figure 3 PSTN Connectivity ................................................................................................................... 14 Figure 4 Clearspan interface to PSTN ................................................................................................... 15 Figure 5 Distributed Call Control Model................................................................................................ 16 Figure 6 Clearspan Call Control Model.................................................................................................. 17 Figure 7 Clearspan Multi-site Deployment Model................................................................................ 18 Figure 8 Clearspan Communicator Multimedia Menu ........................................................................ 21 Figure 9 Clearspan Call Center - Agent................................................................................................. 23 Figure 10 Clearspan Call Center - Agent Menu.................................................................................... 23 Figure 11 Clearspan CommPilot Personal Portal................................................................................ 25 Figure 12 Clearspan Call Manager ......................................................................................................... 26 Figure 13 Clearspan Contact Center Menu........................................................................................... 28 Figure 14 Clearspan Contact Center...................................................................................................... 29 Figure 16 Clearspan Conference............................................................................................................ 33 Figure 17 Clearspan Conference Call Menu......................................................................................... 33 Figure 18 Clearspan Network Layout .................................................................................................... 35 Figure 19 Application Servers................................................................................................................. 40 Figure 20 Network Server ........................................................................................................................ 41 Figure 21 Media Server............................................................................................................................. 42 Figure 22 Web Server ............................................................................................................................... 43 Figure 23 Conference Server .................................................................................................................. 43





Part I – Solution Overview



1 Purpose and Audience

The Clearspan Enterprise Solutions guide is intended for enterprise telecom administrators or those who support enterprise communications networks (system administrators and system integrators). This document describes how Clearspan is deployed as a converged communications platform in the enterprise. It is designed to advise implementers about the advantages, processes and requirements of deploying centralized voice services architectures.

At the core of the Clearspan solution is BroadWorks, an open, standards-based, highly resilient and easily managed software suite, capable of elegantly scaling to millions of users. Deployed on industry standard server platforms, the solution delivers lower total cost of ownership (TCO) by lowing implementation, operating, support and maintenance costs. IT and Telecom managers will enjoy an easy-to-provision and easy-to-manage solution that is extensible with many 3rd party applications, such as CRM and Microsoft LCS.

In addition to offering a converged environment, Clearspan addresses an additional set of problems inherent in traditional networks. Using the web, Clearspan provides a high servicing component between the enterprise administrator and the user. User control and configuration is provided via a standard web portal that augments basic phone functionality to make existing services (for example, conference calling and call forwarding) easier to use, and new services easier to deploy. For example, a user can selectively forward calls from an office phone to a cell phone with a few clicks on a web page or wireless PDA.

The Clearspan platform operates on a standards-based, modular architecture that uses common protocols (such as SIP), open interfaces, and scalable, industry-standard hardware. The open environment enables enterprises, as well as third-party service developers, to rapidly introduce new features and launch new applications.

2 Overview

2.1 What is Clearspan? Clearspan is an enhanced unified communications system that allows large enterprises to offer advanced communications applications over a Voice over IP (VoIP) network. VoIP networks consist of network elements that provide packet conversion, call control, and applications. Clearspan provides call control and enhanced services for enterprise users. It is developed on a standards-based, modular architecture utilizing common protocols, open interfaces, and scalable, industry-standard hardware.

2.2 Why choose Clearspan? There are many compelling reasons why an enterprise should choose Clearspan IP-PBX communications platform:

A Fully Integrated Communications Suite

Clearspan provides a complete suite of communications applications. It offers not only classic PBX-like services, but also a rich set of enhanced services including unified messaging, interactive voice response, business conferencing and contact center. These services are integrated into a single platform with a common web-enabled end-user



interface, eliminating the need to deploy and integrate dedicated communications platforms for each application.

Unified MessagingVendor C

IVR - Vendor B

N-Way ConferencingVendor D

• Hardware-based multi-vendor solution• Proprietary vendor hardware• Custom platform for each application• Poor user experience• Proprietary graphical user interfaces• Limited extensibility• Proprietary legacy CPE

PBX - Vendor A

Unified MessagingVendor C

IVR - Vendor B

N-Way ConferencingVendor D

• Hardware-based multi-vendor solution• Proprietary vendor hardware• Custom platform for each application• Poor user experience• Proprietary graphical user interfaces• Limited extensibility• Proprietary legacy CPE

PBX - Vendor A

• Software-based solution• Off-the-shelf hardware• Single platform for all applications• Integrated user experience• Web-based access• Extensible service platform• Standard SIP-based CPE

• Software-based solution• Off-the-shelf hardware• Single platform for all applications• Integrated user experience• Web-based access• Extensible service platform• Standard SIP-based CPE

Figure 1 Fully Integrated Communication Suite

A Scalable Enterprise Communications Solution

Clearspan is a flexible unified communicatoins platform designed to meet the demanding requirements of today’s large enterprises. At its core is a highly scalable call processing architecture capable of serving enterprises as small as a thousand users, as well as the ability to grow to a network serving hundreds of thousands of users without sacrificing ease of management or reliability.

Single Site or Multi-Site Deployment Models

With Clearspan, service delivery is not bound to a physical site. Users can be deployed in a simple single site network, or distributed across a multi-site enterprise without impact to service functionality, capacity or performance. Clearspan can be deployed at a single site, yet support users distributed across hundreds or even thousands of branch locations. To better manage calls across a large multi-site enterprise, Clearspan includes a variety of powerful site-based call routing and virtual private networking policies.

Disaster Proof Reliability

All Clearspan network elements can be deployed across geographically distinct locations throughout the corporate communications network. This allows enterprises to deploy a fully redundant data center in two geographically separate sites. In the event that a major disaster disables the primary data center, end-users will still have full access to their service profile through the redundant data center. With Clearspan, enterprises are provided with carrier level five 9’s reliability.

An Open, Standards Based Platform



Clearspan does not use any proprietary interfaces. When interfacing with access equipment, Clearspan uses the Session Initiation Protocol (SIP). This allows enterprises to choose from a broad range of standards-based phones, terminals, gateways and peripherals that best meet their budget and end-user requirements. Finally enterprises can choose a communications solution without fear of “vendor lock-in”.

Clearspan is also an open services platform, offering a rich set of application programming interfaces that enable real-time computer telephony integration with enterprise applications. Unlike legacy communications platforms, these interfaces leverage simple XML-based transaction models allowing enterprises to rapidly integrate both real-time call control and service management applications.

2.3 What is in the Clearspan Enterprise Solution? At its foundation, a Clearspan based enterprise unified communications solution includes the service delivery platform built around BroadSoft BroadWorks. This platform is comprised of four separate network elements: the network server, the application server, the media server and the web server. If conferencing is required, a conferencing server is also included. These elements work together to deliver core call control, enhanced applications and service logic. Typically Clearspan is deployed in the enterprise data center. For high availability and disaster recovery solutions, the network elements can be deployed across two physically separate data centers without impact to performance, manageability or scalability.



Figure 2 Enterprise Solution

To deliver these services to the end user, Aastra also incorporate session border controllers, access devices and network gateways. End-user access devices include access gateways, SIP phones and soft clients. Network gateways include PSTN gateways using either FXO, CAS, or PRI based interfaces.

Aastra has assembled a set of “enterprise configuration kits” that itemize vendor-specific packages of access and network equipment that can be reliably deployed with Clearspan. Each kit is tested and validated so that it integrates seamlessly, providing a complete communications solution.

2.4 Key Business Applications The flexibility of the Clearspan solution, coupled with Aastra’s technology and partners, enables enterprises to address a wide range of applications. Along with the following



applications, enterprises may easily develop their own applications, features and

them a cross

gration with Outlook, click-to-dial, voice and

ditional user

y to

reserving legacy PBX investments and slowly migrating all locations to VoIP.

seacend

3 Stan ent Models

integrations with a commercially available software development environment.

The applications most often considered are:

Converged Communications – maximize end-user productivity by providingfully integrated communications environment, enabling smooth interaction aindividuals and groups. With its tight intevideo, unified messaging, call center and conferencing, Aastra provides a ubiquitous communication experience, independent of the endpoint device used.

Fixed/Mobile Convergence (FMC) - Whether users are using mobile or stationary phones, they share the same experience. Because the features are separated from the endpoints, Aastra can easily integrate any type of device, without adlicensing expense. With Aastra’s FMC solution, there is no reason to involve the cellular carrier as all features supplied to the mobile client are independent of carrier used.

Business (SIP) Trunking - allows enterprises to provide “IP Trunking” connectivitexisting PBXs. Enterprises may deploy next-generation trunking and overlay featureswhile p

Aa tra provides a best-in-class set of integrated features and end-user clients that anchor h of these applications. Users experience a unified service regardless of location or point used.

dard Deploym

Clearspan supports a variety of deployment models – from single site deployments to deployments.

3.1 Sing site deployment. In this example, a

large single building enterprise has deployed Clearspan to offer all voice services in the andard LAN configuration with one or more wiring closets

Acc

ets on floors that have

access gateway serves to convert standard analog phones to

Ser

that houses all of the enterprise

PSTN Connectivity

geographically dispersed multi-site

le Site Deployment Model The picture below depicts a typical Clearspan single

network. The building has a ston each floor. In each wiring closet are rack mounted Fast-Ethernet (100Base-T) layer 2 access switches – enough to serve all stations on the floor.

ess Device Connectivity

Not all desktops have to have SIP phones; legacy analog equipment is also supported. Inthis example, a line access gateway is deployed in wiring closanalog handsets. The line SIP signaling and RTP media traffic over the IP network. The line gateway connects to the local Fast Ethernet switch in much the same way as an IP phone. If only a few analog phones exist, an Integrated Access Device (IAD) may also be used.

Each wiring closet is interconnected to the core enterprise LAN using Gigabit Ethernet where traffic is backhauled to the service infrastructure.

vice Infrastructure Deployed in Centralized Server Room

A fully redundant Clearspan system is deployed in a datacenter with a high-capacity IP connection to the core LAN. Often this is the same room data applications.



Also in the datacenter are redundant PSTN gateways. The gateways connect to the local PSTN switch using standard TDM interfaces – typically one or more T1s in North America,or E1s outside of No

rth America. The PSTN gateways connect to a layer-2 switch using

rnet or Gigabit Ethernet connection. Clearspan controls all signaling to N gateways.

either a Fast Etheand from the PST

Figure 3 PSTN Connectivity



Clearspan Manages All Call Control

The picture below shows the system interfaces for a typical outgoing call to the PSTN. In this example, the user employs a web-based click-to-dial application to originate the call. .

outbound call. Once Clearspan has established the call, RTP media is exchanged directly between the SIP pho

3.2 MultIssu

tages in large multi-site deployments. Many traditional PBX solutions suffer from a number of issues when deployed in multi-site

re usually very large enterprises often as big as

lability – when a traditional PBX system has reached capacity, another

• s – some traditional PBX solutions are not designed to

•

• Availability – some traditional PBX solutions have very weak redundancy solutions – especially when applied to a multi-site enterprise. Usually,

Figure 4 Clearspan interface to PSTN

It is important to note that all call control is handled directly by Clearspan. Signaling with the SIP phones is carried over the IP network using the standards-based Session Initiation Protocol (SIP). Clearspan also uses SIP to signal with the PSTN gateways to setup the

ne and the gateway.

i-Site Deployment Model es with Large Multi-Site PBX Solutions

Although Clearspan easily supports large single-site enterprises, it is often chosen over a classic PBX or IP-PBX solution for its advan

enterprises:

• Capacity – multi-site enterprises aa 100,000 users. Some traditional PBX solutions can typically handle up to 2500 users per system. Once this capacity limit has been reached, another system must be deployed.

• Scasystem must be deployed in the network. Adding multiple call control elements to a voice network makes routing and translations rules complicated – and often it requires deploying a full mesh network between all the PBX systems.

Feature Limitationseamlessly deliver features to users across multiple sites. Often features acrossmultiple sites have limitations or simply are not supported.

Manageability – legacy PBX solutions are hard to manage as they often lack a centralized multi-system management interface.



redundancy requires spares for each line card, so a fully redundant PBX solwould require separate spares per site. This is an extremely expensive solution, and is also difficult to inventory and manage.

ution

Disadva

This appcharactelimitationConside 00 sites as depicted in the diagram below:

The solution looked more like a network of 33 disparate enterprises, than one large integrated solution.

Clea pan is Ideal for Multi-Site Enterprises

The s how the same retail enterprise described above would be served

• Cost – the cost of deploying and managing multiple PBX systems in a distributedmulti-site network is high.

ntages of a Distributed Call Control Model

roach of adding PBX systems to the network in order to meet capacity can be rized as a “distributed call control model”. Often these systems have geographic s that require the PBX system to be deployed close to the devices that it controls. r a retail enterprise case study consisting of 16

Figure 5 Distributed Call Control Model

In this example, a popular IP-PBX solution was analyzed. To meet the requirements, the solution required 33 separate call control data centers, each with their own distinct PBX system. Each of the 33 PBX systems then needed to be interconnected and provisioned with routing and translation logic about each of the other 32 PBX systems in the network.

rs

Clearspan was designed to address all of these multi-site issues not only for classic PBX services, but also for the complete suite of voice applications including conferencing, unified messaging, IVR and contact center services. This makes Clearspan an ideal choice for large multi-site enterprises.

Clearspan can be characterized as having a logically centralized call control model. diagram below depictusing Clearspan:



Figure 6 Clearspan Call Control Model

With Clearspan, the number and complexity of sites may be greatly reduced. Centralicall control enabled this enterprise to share a geographically redundant platform, re

zed cing

maintenance costs, support costs and total cost of ownership.

Unprecedented Capacity and Scalability

ional PBX solutions,

du

In this deployment, all 1600 retail sites and hundreds of thousands of employees have their services delivered through a single pair of Clearspan servers in geographically redundant data centers. This is possible because, unlike traditClearspan has no limitations on the number of sites or the number of subscribers in each site. The diagram below takes a closer look at a typical multi-site Clearspan deployment model:



Figure 7 Clearspan Multi-site Deployment Model

Voice Enabled Wide Area IP Network

Traditional multi-site PBX solutions require dedicated trunks between each site. These solutions often require a fully meshed set of interconnects in order to optimize call routing. Because Clearspan is a voice over IP solution, it can leverage the existing core wide area IP network that already interconnects enterprise sites for data applications, as shown above. It is important to note that the quality of service over the IP network must be managed, as it will be used for both call signaling and media traffic between sites. Precedence must be given to voice traffic over data traffic entering the network.

Geographic Data Center Redundancy

Voice service is a critical business application –enterprises cannot function if the voice network is down. If all call control is provided from a single data center on the enterprise network, there is a risk of total network outage in a disaster scenario. To meet enterprise disaster recovery requirements, Clearspan supports geographic redundancy. Instead of installing all network elements in one dedicated data center, Clearspan allows the redundant elements to be installed in a separate geographically redundant enterprise data center. Although these network elements are installed across two physically separate sites, they are still managed as one logical system – configuration and provisioning changes are automatically replicated between data centers. This approach provides an enterprise with carrier grade five 9’s reliability.



Optimized PSTN Inter-working and Expense

Interconnections to the PSTN are an expensive recurring cost to the enterprise. In a multi-site enterprise, it is important to minimize the number of direct connections to the PSTN and leverage the wide area network for intra-site calls whenever possible. Clearspan will always route calls over the core WAN, utilizing VoIP whenever possible.

Clearspan also has routing policies that allow enterprises to leverage their PSTN gateway deployments. For instance, in the diagram above, Branch Office A has a PSTN gateway in local calling area 1 (LCA-1). This PSTN gateway is not only used for local calls by users in Branch Office A, but also Branch Office B. Similarly, Branch Office D in local calling area 2 (LCA-2) also serves users in Branch Office C.

To optimize hop-off costs to the PSTN, Clearspan can choose the best gateway route for a given caller and called number. For instance, if a user in Branch Office D (LCA 2) calls a number that is LCA 1, Clearspan will route the call over the WAN and through the PSTN gateway that is hosted in Branch Office A, where it will be delivered as a local call. This provides for least cost routing of public calls over the private network.

Often a large enterprise will negotiate a reduced long distance rate with an inter-exchange carrier. The rate will apply to all calls made out of the PSTN gateway hosted in the primary data center (headquarters) or an alternative backup data center (division office). Clearspan can be provisioned with routing policies that will select the PSTN gateways in the data centers for all long distance calls from any of the enterprise sites.

Site Independent Feature Functionality

Unlike some traditional PBX solutions, Clearspan services are not bound by physical site boundaries. In the network above, users in one site can belong to service groups with users in any of the other sites. For instance, a user in Branch Office B can belong to a hunt group with users in Branch Office C and users in Headquarters. The same user can act as an agent with one or more call centers in the network – regardless of where the call center pilot number is hosted. Relocating users from one site to another is a simple web based operation. This capability allows enterprises to reduce employee redundancy and improve responsiveness.

Smart Sites – Remote Survivability Solutions

Because Clearspan is based on the Session Initiation Protocol (SIP) standard (a peer to peer protocol) as opposed to a master slave protocol like MGCP or SCCP, sites can be deployed with remote survivability support. These “smart sites” are able to detect a WAN outage. When an outage occurs, the local access equipment provides a set of basic services including: intra-site extension dialing, call hold/retrieve, call transfer, and 3-way calling. If a PSTN gateway is available on the site, then incoming and outgoing PSTN calls can also be delivered.

Remote Worker Support

Clearspan allows remote workers or telecommuters to have access to their complete set of voice services while working from home or on the road. Clearspan supports both “broadband” and “narrowband” remote worker solutions.

With a broadband connection, users get service directly through their IP phone or soft phone connected to the corporate network over a VPN connection. With a narrowband connection, users get service through their local PSTN connection. Incoming calls are forwarded to their “remote-office” number, while outgoing calls are initiated through a web



based call back mechanism. Both solutions allow enterprises to manage outgoing calling plans and centralize billing for all remote users.

Ease of Management

One of the biggest advantages of Clearspan over some traditional PBX solutions is manageability. While other PBX solutions often require routing policies and translations to be replicated across multiple systems, routing policies and translations in a Clearspan network only need to be provisioned once.

Clearspan also incorporates a multi-level web portal for user and service management. This portal allows enterprises to administer company wide group services, define sites, and manage resources such as devices, phone numbers, and services across the corporation. The department level portal allows delegation of user moves/adds/changes (MACs) to a designated department administrator. Enterprise administrators have the flexibility to manage users across all sites remotely, or delegate user-management responsibility to individual site administrators.

Finally, the user portal empowers end-users to take control of their service profile through the web. Users can select their incoming and outgoing calling preferences directly on the web. From classic services like call forwarding, call rejection, and 3-way calling, to enhanced services like remote-office, voice mail and business conferencing, the web portal makes it easy for end users to access their services anywhere on the corporate network.

4 Key User Applications

All Clearspan features run on the same platform, use a common interface, and can be added to the network at anytime. The sections below briefly describe a few of the key applications provided by Clearspan. For a complete description of all of the features and their functionality, refer to the latest “Clearspan Product Overview” for details. All of these features are software based solutions and require no extra hardware to deploy.

4.1 Clearspan Communicator Family The Clearspan Communicator family enables end users to convert their desktop computer into a fully functional phone. This client offers advantages over other client softphones as a result of the integration with the Clearspan system platform and tailored features-functionality to Clearspan applications.

4.1.1 Clearspan Communicator

The Communicator client is an audio-only client that provides a feature-rich desktop interface to make and receive calls as well as execute enhanced features. The Communicator has access to Clearspan’ many advanced VoIP features including screening and messaging/voice mail integration.

As an integrated device with the Clearspan platform, enterprise administrator are able to auto provision the client, provide automatic updates, and maintain version control seamlessly through the Application Server. The Communicator is capable of supporting G.729a protocol. The Communicator allows users to utilize the client as a primary or secondary phone device. Clearspan supports this via server features that enable both devices to register to the same user account and directory number (DN)/direct-inward dialing (DID) number.



The Communicator is available in English, Spanish, and Simplified Chinese, in addition to having multi-language support through the Clearspan platform.

4.1.2 Clearspan Communicator - Multimedia

The Clearspan Communicator - Multimedia provides all the capabilities of the audio-only Communicator and adds high quality multimedia transmission. With the Communicator - Multimedia, a user’s desktop computer can function as a full-feature multimedia communications device.

Figure 8 Clearspan Communicator Multimedia Menu

While Clearspan integration offers considerable value for the audio-only client, the benefits are even further expanded with the addition of video. First, the video codec negotiation is somewhat more complicated than audio-only codec negotiation. Clearspan pre-integration limits the issues associated with video codec negotiation, simplifying deployment and reducing support calls and customer issues. Second, pre-integration with the video client enables separate routing of voice streams over traditional handsets while the video signal is connected to the video client. Thus, users can enjoy features of their desk handsets such as a high-fidelity speakers and microphone, along with the multimedia experience available from the desktop client.



4.2 Clearspan Assistant Family The Assistant family is a suite of products that helps, or “assists”, end users to manage, control, and configure their phone. In a compact and extremely useful user interface, Clearspan Assistant provides users with an easy way to answer, transfer, and dial calls, search enterprise, personal, or Outlook contacts directories, and quickly enable/disable call settings such as Call Forwarding, Do Not Disturb, and Simultaneous Ringing with a click of a mouse.

4.2.1 BroadWorks Assistant – Enterprise

The BroadWorks Assistant is a feature-rich, integrated toolbar that enables users to make and accept telephone calls, and change telephone settings, from within Internet Explorer and Outlook. The toolbar delivers the following features and benefits to users:

Complete call control capabilities, including making, receiving, transferring and holding calls

Right-click to dial on highlighted phone numbers within web pages

Easily change telephony service settings such as Simultaneous Ringing, Call Forwarding All, Call Forwarding No Answer, Call Forwarding Busy, Do Not Disturb, Remote Office, VM User, Logged-in Status

Add Microsoft Outlook contact vCards for new incoming calls

Outbound click-to-dial from Outlook contacts

Group, enterprise and LDAP directory browsing

Call notifications with caller ID from Clearspan

See who is calling based on a contact list in various Clearspan directories

View call history

User help functions available during any operation with an HTML user guide built into the application



Figure 9 Clearspan Call Center - Agent

Figure 10 Clearspan Call Center - Agent Menu



4.2.2 Clearspan Assistant - Mobile

Clearspan Assistant - Mobile is a feature-rich, easy-to-use mobile application that is fully integrated with the Clearspan platform. It is an integrated Nokia E-series mobile handset application that enables users to originate Clearspan-routed telephone calls, modify their telephone settings, view their corporate directory, and have access to their call logs. Clearspan Assistant - Mobile is a carrier class, lightweight mobile application for everyday users that delivers the following features and benefits to users:

Provides access to a corporate directory

Addresses the needs of corporate, small business and small office/home office users

Receives voice mail notifications

Easily change telephony service settings such as Simultaneous Ringing, Call Forwarding All, Call Forwarding No Answer, Call Forwarding Busy, Do Not Disturb, and Remote Office

View call history

Place calls from a mobile phone but have it appear like as if the user is calling from their office line

Manage only one phone number and voice mail account

Create international profiles



4.3 Personal Web Portal The Personal web portal provides individual users with the ability to configure and manage a host of traditional and advanced telephony services. Each user is empowered with the control and flexibility to easily configure their services to meet their unique needs. Users no longer have to remember any star codes or complex procedures to configure their services, as is often the case with legacy systems. Rather, Clearspan improves personal productivity by leveraging the web to make services understandable and actually useable.

Users can customize their services to follow them anywhere, whether at work, at home or on the road. For example, the Call Notify service (shown below) enables users to indicate which incoming calls they want to be notified of, and during which hours of the week. They can also choose to have their notifications sent to their mobile phone or e-mail address.

Figure 11 Clearspan CommPilot Personal Portal

After logging in to the personal web portal with his/her user identity and password, a user can activate, deactivate, and modify the parameters of his/her own services, including: Call Forwarding (Always, Busy, No Answer, Selective), Simultaneous Ring, Do Not Disturb, Selective Call Acceptance and Rejection, Anonymous Call Rejection, Priority and Distinctive Ringing, Calling Line ID Blocking, Voice Messaging Notification, Voice Messaging to E-mail, Personalized Name Recording, and Remote Office. The left navigation area lists services and features available to the user.

The Web Portal updates features and configurations in real-time, providing utility and convenience. Convenience is important for features that are frequently updated by users such as the call-forwarding suite. The web interface also provides a means for making sophisticated entries and configurations, very difficult to replicate through a standard voice portal. For example, in addition to specifying phone numbers for service treatments, Clearspan enables users to provide SIP URLs, feature access codes, and/or speed codes. Selective call treatments can be defined for parameters that include line ID status such as PRIVATE, and UNAVAILABLE, along with the more common IDs such as ANONYMOUS.

The following features can be provided with the CommPilot Call Manager:

Dial/Redial (click-to-dial) – Enables a user to enter and dial a number, dial directly from a drop-down Phone List (Group, Personal, or Call Log) or from Outlook or LDAP tabs, or click the Redial button.



Transfer – Enables a user to redirect a ringing, active, or held call to another number or directly to voice mail. Before transferring the caller, the user can choose to consult with the third party first or establish a three-way consultation.

Send to VM – Enables a user to transfer calls to voice mail.

Talk – Enables a user who is already engaged in call to answer another waiting call. When available, Calling Line ID is displayed with caller’s name and number.

Hold – Enables a user to place an existing call on hold for an extended period of time, and then retrieve the call to resume conversation. While the calling party is held, the user can choose to make a consultation call to another party.

Conference – Enables a user to establish a three-way call involving two other parties.

Hang Up – Enables a user to disconnect a call that has been answered.

Hot Links – Buttons are provided to enable a user to turn on/off frequently-used services such as Call Forwarding Always and Do Not Disturb. Alternatively, if CommPilot Express has been configured, the user can change their CommPilot Express status (for example, Available, Busy, Unavailable) by choosing from a drop-down list.

System Buttons – Buttons are provided to enable a user to send an e-mail for technical support, get context-sensitive online help, and configure the CommPilot Call Manager.

Figure 12 Clearspan Call Manager

The Call Manager is used to augment the functions of a standard phone. It provides an alternative user interface for enhanced services, replacing flash-hook and star codes. It is not intended to replace the phone or be a stand-alone substitute for basic call functions. Calls can be initiated and manipulated using the phone or the web interface. The audio is transported through the phone, not via the user’s computer.



4.4 Unified Messaging Clearspan messaging provides all of the features of a traditional voice messaging solution, plus:

• Message delivery to any specified email account.

• Message waiting notification delivered to the phone and to any specified mail or Short Message Service (SMS) account (e.g. cell phone).

• Integration of messaging capabilities with Clearspan services (call back, transfer, find me / follow me services, escape to extension, voice portal, etc.).

Incoming calls to the user are sent to voice mail upon reaching a busy or no-answer condition. The caller is then played a greeting. There can be different greetings for busy and no-answer conditions and all greetings can be partially or fully customized by the user:

• Default busy greeting

• Default busy greeting with name

• Custom busy greeting

• Default no-answer greeting

• Default no-answer greeting with name

• Custom no-answer greeting

The caller can then leave a message or press 0 to transfer to an attendant. The attendant is configurable by the user and can be any valid phone number. If the caller leaves a message, he/she has access to the following functions:

• Long message warning tone

• Set the message status to urgent and/or confidential

• Review the message and erase, record it again or deposit it

• Users can also configure their voice mail service to serve other phones, like a cell phone. With this capability, users can forward any phone to the CommPilot Voice Portal phone number and have the calls be sent directly to their mailbox greeting. This powerful application allows employees to use a single Clearspan voice mail system for their office phone and mobile phone – they do not need to manage multiple voice mail accounts separately.

Voice messages are stored on standard email servers (POP3, IMAP4, Microsoft Exchange Server) as WAV audio files attached to emails.

Users can retrieve their emails from their main location, from a third-party location or from any standard email client. When retrieving emails from their location, users simply dial the CommPilot Voice Portal phone number (or extension). The system prompts the user for their passcode. After entering the passcode, the user is informed of the mailbox status (how many urgent, new, expired, and saved messages) and can review the messages through a menu. While reviewing the messages, users can play the message envelope, jump to next or previous message, skip ahead, skip back, pause, repeat, erase, save, reply, call back, forward, compose and send to a user or a distribution list.

When retrieving emails from an email client, the user simply configures the client to collect email from the email server where the messages are stored. Messages are retrieved as WAV attachments to emails and can be listened to with standard audio software. Messages received as email can be manipulated like any other email (stored, forwarded, replied to, etc.). Some enterprises prefer to keep “carbon copies” of all voice mail



messages received, for regulatory or security reasons – this can be done easily on Clearspan with just a few clicks.

When the user receives new messages, they can be notified by standard message waiting indication mechanism (stutter dial tone and message waiting lamp). Users can also request a notification to be emailed to a specific location, like a cell phone, when a voice message is received.

4.5 Contact Centers Clearspan provides support for basic contact centers, allowing business agents to receive incoming calls from a central phone number. Using this service, a business can establish technical assistance lines, customer support numbers, or order-taking centers. Multiple contact centers can be supported per business. Incoming calls to a contact center are presented to the next available agent.

The Contact Center is based on Automatic Call Distribution (ACD) service to provide a complete, business-ready application. Hence, contact centers inherit all of the characteristics of the Hunt Group service and are also provided with sophisticated call-handling features like queuing, music on hold, etc.

Figure 13 Clearspan Contact Center Menu

Clearspan expands the capabilities of legacy contact centers by delivering a true remote agent solution, allowing call center agents to be geographically distributed. Therefore agents can attend calls from home, a satellite office, or any other location served by Clearspan in a transparent fashion



.

Figure 14 Clearspan Contact Center

Clearspan Contact Center functionality can be combined with other Clearspan Call Completion services to ensure that all incoming calls are serviced expeditiously under any network condition and at anytime.

Voice mail – If there are no agents to handle an incoming call or the call goes unanswered for a specified amount of time, the call can be forwarded to a contact center voice mailbox.

Night service – Calls received after-hours or on non-business days receive a service menu of options allowing a caller to leave a voice message or transfer to an emergency number.

Multiple call distribution policies – Incoming calls are handled according to the selected policy, which include uniform call distribution, linear hunt group, circular hunt group, and simultaneous ringing.

Call queuing – When all call center agents are busy, incoming calls can be queued until they can be presented to an available agent.

Queue escape – Callers who are queued can press a key to be sent directly to the contact center voice mailbox instead of waiting for an available agent.

Overflow – When a contact center cannot accept any more calls, incoming calls can be forwarded to an overflow phone number.

Statistics – Statistics are generated for each contact center group and can be viewed by the group administrator via the web portal.

Service integration – Any Clearspan personal service can be assigned to a contact center phone number to customize the contact center group. This includes services such as Call Forwarding, Call Notification, Call Screening, and Voice Messaging.

Queue flushing – When all agents in the contact center group log out, queued calls are automatically sent to the contact center group voice mailbox.



Outlook contact integration – vCards from the agent’s Outlook or Exchange contact database pop up for incoming calls.

Agent log in/log off – Agents can log in and out from the group so that calls are only presented to agents who are on duty.

Screen pop-ups – Incoming calls pop up on a web screen showing information associated with the incoming call. A group-specific URL is accessed for each call.

Configurable Music on Hold – The queued callers are provided with music or advertisements between periodic announcements.

4.6 Auto Attendants The Auto Attendant provides enterprises with a powerful and flexible tool to field inbound calls and deliver them to the intended destination through interactions with the caller. The Clearspan Auto Attendant is an integral part of the product offering and does not require any third-party equipment.

The Auto Attendant is reached by dialing an associated phone number or an extension. Once connected to the Auto Attendant, the caller is played a greeting that provides a menu of options to complete call routing.

The menu, which is configurable by an enterprise administrator, can provide up to nine options to the caller, including:

One-key dialing – The caller presses a pre-defined DTMF key to reach a particular phone number or extension within the group. This option is also used to build multilevel IVR menus.

Operator dialing – The caller presses a pre-defined DTMF key to reach an operator.

Name dialing – The caller spells the name of the intended party using the numerical DTMF keypad. Upon identifying a unique match, the caller is played the name of the called party and is then transferred.

Extension dialing – The caller enters the extension of the intended party through the numerical DTMF keypad. Upon collecting the full extension, the caller is played the name of the called party and is then transferred.

The moves, adds, and changes for users in a group are automatically available for the Auto Attendant name dialing and extension dialing functions. Access to users currently in the group is always available.

The Auto Attendant uses the multi-location enterprise capabilities of the Clearspan platform to transparently support geographically distributed groups. The Auto Attendant supports users with a Direct Inward Dialing (DID) number as well as users without an external public directory number. These users originate calls as usual and the Auto Attendant allows them to receive external calls. Calls made to the Auto Attendant use the routing capabilities described above to terminate calls to the appropriate user.

This support provides greater flexibility for a group administrator to create and delete users and in many cases reduces the costs associated with obtaining DID numbers. Clearspan can also offer speech-based Auto Attendants, which use a speech IVR instead of a DTMF-based IVR (“call Joe on his mobile”).

4.6.1 Clearspan Receptionist

The Clearspan Receptionist is a carrier class IP telephony attendant console for use by receptionists, or telephone attendants, who manage and screen inbound calls for enterprises. The BroadWorks Receptionist is a feature-rich desktop application that is fully



integrated with the Clearspan platform. The application delivers the following benefits to users:

Easy-to-use design that follows the natural work “flow” of a call from the top to the bottom of the screen.

Intuitive business processes, as only “valid” options are presented to the attendant.

Professional call handling as critical information is available in real time.

Accurate delivery of messages via a one-step process when people are unavailable.

The BroadWorks Receptionist can be used under a number of different operational scenarios. These scenarios include:

After Hours – Allows operators to automate switching from day to night mode.

Call Center Queue – Allows operators to monitor and control calls in a Call Center queue, and to manage their availability status.

Hoteling – Allows multiple part-time operators to share a single log-in sequence when they change shifts.

Low Traffic – Single receptionist answering one or more dedicated main line numbers.

High Traffic – More than one attendant console managing multiple dedicated main line numbers.

Network Attendant Console – Geographically dispersed operators supporting each other in an enterprise configuration.

Multi-tenanted Offices – One or more operator answering calls on behalf of different organizations.

Optional Voice Mail Transfer – Operator has the added ability to transfer calls to voice mail for contacts in a group/enterprise that are busy or unavailable.



4.7 Web Conferencing Clearspan supports a variety of business conferencing services, including meet-me conferencing, reservationless conferencing, automated dial-out conferencing, collaboration tools, and more.

Web Conferencing enables the set up, use, and monitoring of n-way conferences via a web interface. Both internal and external participants can use a conference bridge once it has been set up. A single

The Conferencing service includes the following features:

Figure 15 Clearspan Receptionist Menu

Audio and web conferencing

Scheduled, recurring, reservation-less, and ad-hoc

Meet-me dial-in numbers

Web collaboration

Share Microsoft PowerPoint, Excel, and Word files (extensions are automatically checked by Clearspan for validity)

Secure SSL and password protection

Web browser viewable, no client required

Optional click-to-download of web presentations

Moderator control

Dial-out capability



Mute, hold, drop, and add participants

DTMF and web portal interfaces

Roll call (service can record and announce participants’ names as they join)

In-call functions

Hand raising (muted participants can request the conference leader to un-mute them temporarily)

Optional leader

PIM integration

Automated e-mail invitations and Outlook calendar entries

Reporting

Web-based reporting

Department and project codes

Recording

Recording and playback of individual conferences (the CDR can account for recording duration)

Access code generation

Automatic, pre-assigned, or user-defined

Figure 16 Clearspan Conference

Figure 17 Clearspan Conference Call Menu



4.8 Clearspan Deployment Studio The Deployment Studio facilitates the customization, deployment, and management of Clearspan Assistant, Assistant - Enterprise, Receptionist, and Call Center - Agent/Supervisor desktop client software by enterprise providers. Providers can configure a variety of client settings, such as branding, networking, localization, and additional pre-defined settings, for Assistant, Assistant - Enterprise, Receptionist, and Call Center. The Deployment Studio is required to deploy all Clearspan clients.

Enterprise providers proceed through an intuitive GUI to fully customize their client applications to suit their specific requirements.

5 Integrating with Legacy Equipment

This section describes solutions for integrating Clearspan with existing enterprise voice networks. This is often required in multi-site enterprises with a large investment in legacy voice equipment. The services provided by Clearspan complement the services provided by the existing solution. Equipment on the enterprise site communicates with Clearspan in the enterprise data center. Solutions are described for enterprises with existing Private Branch Exchanges (PBXs).

5.1 PBX Solution (SIP Trunking) A Private Branch Exchange (PBX) is a private telephone system that provides on-premises dial service and connections to the PSTN. A PBX offers call control features such as call waiting, call transfer and conferencing. Most PBXs can interface with other communications applications such as auto attendant, voicemail, automatic call distribution and call accounting. Many PBXs use proprietary phones, since a significant portion of a PBX sale involves handsets.

PBXs typically serve medium to large enterprises locations. Typically, each user has a dedicated line. For outgoing calls, the PBX switches calls from a user line to a trunk connected to the PSTN. For incoming calls, a PBX switches calls from a trunk connected to the PSTN to a user line. PBXs provide both personal and group based services. They are popular in enterprises where employees require direct dial access.

5.1.1 Network Layout

A PBX gateway is used to inter-work a legacy PBX with Clearspan. The gateway provides trunk interfaces for connectivity with the PBX and Ethernet interfaces for connectivity with the LAN. The PSTN interfaces are via CAS or PRI trunks. On the Clearspan system, the address of the gateway connected to the PBX is setup as a routing element within a Voice VPN enterprise group.



Figure 18 Clearspan Network Layout

As shown, the PBX connects to the GW via CAS or PRI trunks. The gateway is connected to the corporate LAN via one or more Ethernet interfaces.

5.1.2 Services

To illustrate how the Clearspan services can be used to complement the services provided by a legacy PBX, an example of a typical enterprise is used. In this example, an enterprise has a PBX serving telephones and/or fax machines with a T1 PRI trunk connected to a GW.

In this example, calls can be delivered to the GW with or without direct inward dialing (DID). If DID is used, then individual phone numbers are assigned to each user. This allows services to be provided on a user basis. If DID is not used, then there is a main phone number for the PBX. Clearspan delivers all incoming calls to the main number, which is associated with an attendant or auto attendant. The attendant then delivers the call to individual lines connected to the PBX. Note, most PBXs typically support both DID and non-DID access simultaneously. Some users will use DID and some will not.

5.1.2.1 Services for PBXs with DID Some of the basic personal services provided by Clearspan are also provided by the PBX, such as call transfer, call forwarding, extension dialing, etc. Clearspan complements these basic services, providing additional enhanced personal services, including:

Service Usage Anonymous Call Rejection Deny anonymous incoming calls before reaching the PBX.



Service Usage Call Forwarding Selective Forward certain incoming calls before reaching the PBX. Call Notify Log all incoming calls before reaching the PBX. CommPilot Call Manager Provide for web based call control functions. CommPilot Express Customized incoming call management. LDAP Directory Integration Integrate with enterprise directory. Outlook Integration Integrate with Outlook or Exchange Server. Remote Office Provide support for tele-workers and remote offices. Selective Call Acceptance Accept only certain incoming calls before reaching the PBX. Selective Call Rejection Deny only certain incoming calls before reaching the PBX. Simultaneous Ring Ring the PBX user in addition to multiple external numbers. Voice Messaging Hosted, centralized voice messaging.

Group services can also be used with PBXs with DID support. Group services that are typically assigned include:

Service Usage Account and Authorization Codes

Track and/or authorize all outgoing calls made from the PBX.

Attendant Console Provide hook status for off network calls to and from the PBX. Auto Attendant Provide automated attendants for PBX users. Call Centers Provide call centers for PBX users. Calling Group Identity Delivery

Provide the enterprise identity, instead of the individual line identity, for all outgoing calls made from the PBX.

Hunt Groups Provide hunt groups for PBX users.

Incoming Calling Plan Block certain types of incoming calls made to the PBX. Outgoing Calling Plan Block certain types of outgoing calls made from the PBX. Voice Portal Provide a dial in interface for service interaction and management.

5.1.3 Services for PBXs without DID

When DID is not available, services are provided on an enterprise wide basis (main phone number), not on a per user basis. Clearspan provides hosted, enhanced services that complement those provided by the PBX, including:

Service Usage Call Forward No Answer Forward unanswered calls to an auto attendant or enterprise voice

mailbox. Call Forwarding Selective Calls received after hours or on the weekend are forwarded to an auto

attendant, enterprise voice mailbox or remote location. Call Notify Calls to the main number are logged via email notifications. Calling Line Identity Delivery Provide caller number and name for incoming calls. Selective Call Rejection Nuisance calls to the main number are blocked. Voice Messaging Provide an enterprise voice mailbox or individual voice mailboxes. This

works in a similar manner to a KTS. See KTS Solution for more details.

Group services can also be used with PBXs without DID support. Group services that are typically assigned include:



Service Usage

Account and Authorization Codes

Track and/or authorize all outgoing calls made from the PBX.

Calling Group Identity Delivery

Provide the enterprise identity, instead of the individual line identity, for all outgoing calls made from the PBX.

Outgoing Calling Plan Prevent certain types of outgoing calls made from the PBX.



Part II – Solution Components



6 Clearspan Service Delivery Platform

This section describes the Clearspan platform in more detail. It provides specifics on each Clearspan server as well as the preferred hardware platforms. In addition, it provides call flows showing how the components interact.

6.1 Network Elements Clearspan is composed of the following servers:

Application Server – provides call processing, service logic, endpoint management and service management

Network Server – provides policy based translations and routing, subscriber location directory and media server selection

Media Server – provides specialized media resources, including media recording, media playback, DTMF digit detection, three way conferencing and media relaying

Web Server –supports Clearspan system web interfaces and additional external interfaces. The Web Server supports the Personal Web Portal and Call Manager via HTTP, as well as the BroadWorks Open Client Interface for the Assistant and Receptionist applications.

Conference Server (optional) – provides enhanced audio and web conferencing.

PM Server – monitors the health, performance and reliability of the Clearspan system.

Client Management System (CMS) – provides the central point to control the functions associated with the configuration, deployment, and activation of user profiles and templates for endpoints or clients.

Together, these components enable a wide array of personal, group, and network services including PBX equivalent services, messaging, attendants, interactive voice response, call distribution, site networking, conferencing and desktop integration. Each server is described in more detail below.

6.1.1 Application Server

The Application Server is a service delivery platform responsible for the execution and management of enhanced personal and group services. Application Server functions include management of network traffic, handling of signaling interfaces, and logical execution and management of services. The Application Server comprises a database, the ServiceOS™ abstraction layer, and protocol stacks.

The Application Server supports the Session Initiation Protocol (SIP) as the communication protocol with VoIP equipment, including:

Access Equipment – SIP Phones, Analog Line Gateways, WiFi Phones, Soft Clients

Network Equipment – PSTN Gateways, PBX Gateways, Session Border Controllers, Edge Elements

Other Servers – Network Servers, Media Servers, Conferencing Servers, Third Party Servers

The Application Server provides secure access to subscribers and services for management, administration, provisioning, and configuration. The associated web portals can be customized for different customer groups, based on the subscribed-to services of



those groups. With various levels of security and accessibility (system provider, enterprise administrator, group administrators, department and end-user), secure access is granted not only to the system, but also to the correct level of privileges. Different users are directed to the applicable screens with proper security and real-time information. Information is retrieved from the database and displayed in real time.

Figure 19 Application Servers

Application Servers are deployed in pairs, where one servers acts as the primary server and the other as the secondary server. Under normal operation, the primary Application Server will manage all call processing and endpoint management. User modifications, additions and deletions are replicated across the database of both the primary and secondary Application Server, making all service and user profiles available on both servers. Database replication is done in real-time, as additions and modifications are made. Thus, both the primary and secondary servers have access to exactly the same database.

In the event that the primary server fails, or is simply inaccessible from one or more endpoints in the network, then those endpoints will be able to route their calls through the secondary server. The failover time is the time it takes an endpoint to retry after a non-response to the secondary server. This is typically engineered to be less than 1 second. This type of redundancy may be geographically distributed; protecting against server failures as well as against IP networking failures (e.g. router and circuit failures).

Application Servers are hosted on IBM servers, running the Linux operating system. Each server connects to the VoIP network using 10/100BaseT Ethernet interfaces. Multiple interfaces are available supporting a wide variety of IP network configurations. No specialized hardware is required.

6.1.2 Network Server

The Network Server enables system providers to centrally manage network-related applications within their network. This includes public translations and routing capabilities such as least-cost routing, as well as enterprise-focused network applications such as voice virtual private networks (VPNs).



The Network Server allows construction of massive next-generation voice networks by assisting with the scaling of IP telephony networks and offloading subscriber-specific routing functions, allowing softswitches to focus on core call state management for trunking media gateways. The Network Server also acts as a platform for network-based enterprise applications such as private dial plans, and supports passing of non-numerical characters like * and # to the network so they can be used to trigger functionality in other networks.

The Network Server optimizes network resource utilization by providing the capability to selectively route calls to geographically dispersed resources, thus maximizing network bandwidth utilization. For example, the Network Server can manage Media Servers as a single network-wide pool of resources and, at the same time, select an appropriate Media Server for the location of the requesting subscriber. The Network Server also performs a central role in supporting Clearspan redundancy capabilities.

Provisioning new subscribers can occur via synchronization of the group and user data between Application Servers and Network Servers. The Network Server functions with the Application Server as shown in the following figure:

Network Devices

SIP

SIP

SubscriberLocation

Translations& Routing

Figure 20 Network Server

The policy database provides public and private (that is, enterprise) policies, to manage a system provider’s network services. It also maintains subscriber location information to track where subscribers are hosted within the network. The policy database contains translation processing and a routing engine that is driven by a flexible policy approach. Dialing plans, call typing, route selection, and network services configuration are policy-driven and can be updated “on-the-fly”, including the introduction of new policies within the network. Operators can configure the precedence of their routing policies to match requirements.

The ServiceOS manages the sessions, which are the network connections associated with a user or network element.

6.1.3 Media Server

The Media Server functions together with an Application Server (AS) and enables enhanced services, such as Auto Attendants, Video Auto Attendants, Meet-Me, Conferencing, Multimedia Messaging, video advertisements in Call Center queues, play tones and treatments to callers. The Media Server supports these services for wireline,



wireless, and/or converged networks, offering enterprises enhance productivity for end-users. Media Servers can be geographically distributed, which can minimize call latency and bandwidth utilization.

RTPRTCP

S I P

NS

SIP

HTTPAS

Figure 21 Media Server

In a stand-alone architecture, SIP and HTTP are used as standard protocols for communication between the Application Server, Network Server, and the Media Servers. The Application Server requests resources from Media Servers and controls the behavior of those resources. Third-party media servers can also be deployed and controlled via a standard SIP interface.

Enterprise providers can choose from multiple codecs that are supported on the Media Servers. Configurable classes of multimedia service can also be assigned to enterprise providers, groups, and individual users, restricting callers to the Codecs in their assigned set. Enterprise providers can opt for lower bit-rate Codecs to increase the number of simultaneous calls that can be provided on an access link to end users. G.729 is one of the available Codecs supported.

In addition, the Clearspan system can be configured to route calls differently depending on the Codecs they use, eliminating network elements that do not support included media, or prioritizing network elements that offer better support for included media.

6.1.4 Web Server

System providers deploy Web Server farms to support Clearspan system web interfaces and additional external interfaces. The Web Server supports the CommPilot Interface using HTTP, and the Open Client Interface. System providers can choose to use either of these interfaces or both concurrently. The following figure shows the functions within the Web Server:



•

End Users andAdministrators

OCI

SubscriberLocation

DataAccess

LOC

OCI

HTTP

Web Portals,Applicationsand Plug Ins

Figure 22 Web Server

6.1.4.1 Open Client Interface The Open Client Interface (OCI) is an open interface that enables third-party applications to leverage Clearspan call control and provisioning functionality. This interface operates using both XML and HTTP/SOAP. Thus, Clearspan makes it easy for developers to create new applications to augment existing Clearspan functionality. For example, the OCI queries the Network Server to find the Application Server with the requested data such as a user’s mobile number or department.

6.1.5 Conference Server

The Conferencing Server is a specialized media resource that provides a complete set of enhanced business conferencing features, including web-based presentation and collaboration. A single pool of Conference Servers can support multiple Application Servers in a Clearspan system. The Conferencing Server functions as shown in the following figure:

•

Access and NetworkDevices

RTP

HTTP

SIP

DataRetrieval

ResourceControl

Figure 23 Conference Server

The Conferencing Server is fully integrated with the Clearspan back office and operates on standard hardware, thereby allowing for cost-effective scaling with the rest of the platform.



6.2 Call Scenarios This section describes how the components of Clearspan work together to deliver a call from and to a user.

6.2.1 Registration

When an access device, such as an IP phone or analog gateway, is powered on, it sends a registration message to the Application Server. This message indicates the user and their associated address. Clearspan stores this information in its location database. It is used when calls are received for a user, so the call is delivered to the appropriate device.

6.2.2 Call Origination

When a user originates a call, a signaling message is delivered to the Application Server to which the endpoint is registered. The Application Server provides call control and service logic execution. This includes services such as call screening and extension dialing.

If the call is to another user within the same group then a signaling message is sent to the destination user’s endpoint, and the device is alerted. If the call is not to another user within the group, a signaling message is sent to the Network Server. The Network Server computes a destination route using the user’s phone number and group along with the dialed number.

The Network Server computes a list of destination gateways, and returns this list to the Application Server. The Application Server sends a signaling message to the first gateway in the list. If the gateway accepts the call, then call setup is complete. If the gateway denies the call, then the next gateway in the list is tried until successful call setup is achieved.

6.2.3 Call Termination

When someone external to the group calls a user, the PSTN gateway delivers a signaling message to the Network Server. The Network Server tracks which Application Server is currently serving the user, and returns this address to the PSTN gateway. The PSTN gateway then sends a signaling message to the hosting Application Server.

The Application Server receives the signaling message, and provides call control and signaling logic execution. This includes services such as call screening, call forwarding, find me/follow me and voice messaging. After the initial termination services are provided, a signaling message is sent to the user’s endpoint and call setup is completed.



6.2.4 Entry / Exit Point

The Network Server is responsible for managing routing information associated with a Clearspan system, and for managing location information for individual users. It acts as a single point of access for calls coming from outside of Clearspan via a PSTN or PBX gateway.

For example, when a call from the PSTN is destined to a user managed by Clearspan, the call will be routed from the PSTN to a PSTN gateway. The PSTN Gateway will then send a signaling message to the Network Server; the Network Server will then redirect the call to the Application Server that is hosting the called user. If the Network Server is hosting multiple Application Servers, it will redirect the call to the Application Server hosting the user with a phone number matching the called phone number.

Since the Network Server is a centralized point of contact, it is the key to providing scalability. Because the Network Server is a stateless, transaction-oriented server, it scales easily by adding more Network Servers. As a system grows, a provider can add Application Servers and Network Servers to increase system capacity. As the number of supported users grows, the provider must also add Media and Conference Servers, since the system will require more media resources.

6.2.5 Media Resources

When a service on the Application Server requires interaction with the user, the Application Server requests a media resource. It does this by sending a media resource request to the Network Server. The Network Server computes a list of available Media Servers, which are closest to the point of entry. This is computed by comparing the calling phone number with the phone number associated with the Media Server.

The Network Server returns a list of Media Servers to the Application Server. The Application Server then uses this list to request a media resource. If the first Media Server in the list is not available or does not have available resources, then the next server in the list is tried. This is done until a media resource is successfully allocated.

When a resource is allocated, the Application Server requests that the media path be reset so that the user or destination is directly connected to the media resource. Then, the particular service on the Application Server orders the media resource to perform a particular function. This can include things such as playing an announcement, recording audio, detecting DTMF digits or conferencing multiple parties together.

When the service completes, the Application Server requests that the media path be reset back to the original settings so that the user or destination is released from the media resource. Then, the media resource is released and returned to the queue of available resources.

7 Access Devices

Access devices provide terminal-access to the network by allowing an end-user to originate and terminate calls. In this solution, all access devices use SIP (the Session Initiation Protocol) as the common access signaling protocol. In addition to providing a call signaling path into the network, the access device is also responsible for converting the end-user analog voice transmission into a packet based voice over IP stream in real time. This voice over IP stream is commonly referred to as the RTP (Real-Time Transport Protocol) stream. In general, access devices can be further categorized into access gateways, IP phones and soft clients. Access devices should all support the G.711 CODEC for u-Law and a-Law, as well as the RFC 2833 standard for in-band DTMF transmission.



7.1 Access Gateways Access gateways allow standard analog phones to interface to the VoIP network. On one side they provide an FXS interface to a standard analog phone. On the other side they provide a 10/100BaseT interface to the VoIP LAN. Access gateways allow existing analog wiring to the desktop to be re-used in a VoIP network. Access gateways can be further sub-divided into “desktop gateways” and “rack-mounted gateways”.

Desktop-gateways are usually one to four port units that are intended to be deployed near the desktop or station (i.e. fax terminal). Rack mount gateways can support anywhere from 8 to 48 ports and are intended to be deployed in a wiring closet within the same enterprise site hosting the users – usually deployed along side other networking equipment such as hubs and switches.

Access gateways will typically support a number of local or “built-in” services. These include call-hold, retrieve, transfer, call-waiting (up to 2 call appearances), calling name/calling line id, and sometimes even 3-way calling. These basic built-in services are an important building block when designing remote site survivability solutions in the event that access to both of the redundant data centers is lost.

7.2 SIP Phones SIP Phones are digital handsets that interface directly to the VoIP network without requiring an intervening access gateway. SIP phones have a scalability advantage over access gateways as they require no extra rack space as new users are added to the network. SIP phones can be further categorized as basic and enhanced.

A basic SIP phone provides a core set of services similar to those provided in an analog gateway: hold, transfer, calling name/caller id, and usually 3-way calling. Basic SIP phones usually offer only one line and only provide up to two simultaneous call appearances. They should include a built-in Ethernet bridge so that users can easily share the port on their desktop with their phone and their PC. The bridge should support in-line power as well as VLAN and TOS bit tagging.

Enhanced SIP phones, such as Aastra’s 3- and 5-series, build on the capabilities of the basic SIP phone, but include a comparable handset experience to that found in traditional PBX offerings. Features like multiple call appearances and bridged line appearances are supported. An enhanced SIP phone also includes a built-in hands-free speaker phone. These phones typically include a large screen LCD display that allows access to directories, contacts and other hosted content through their micro-browser interface.

7.3 Soft clients Soft clients allow end-users to access the communications network using software on a general purpose desktop computer, laptop or personal digital assistant. These clients leverage the soundcards commonly found in computer platforms. Users can simply plug in a headset and a microphone to the system and use the software to originate and terminate calls. Features and quality tend to vary greatly from one vendor to another. Usually a soft client is deployed as a secondary device and used only when a standard IP or gateway-based handset is not available.

8 Network Gateways

PSTN gateways are used to inter-connect Clearspan with the PSTN. The gateway provides trunk interfaces for connectivity with the PSTN and Ethernet interfaces for connectivity with the LAN. The PSTN interfaces are via CAS or PRI trunks, or FXO analog lines. On the Clearspan system, the address of the gateway connected to the PSTN is



setup as a routing element. Public routing policies are used to route calls to the appropriate gateway.

For incoming calls from the PSTN to a Clearspan user, the following information is delivered:

Trunk Type Calling Party Called Party CAS Trunks Provided

(Based on Gateway or Connected Switch) Provided (User Identity = Number)

PRI Trunks Provided (Number + Optional Calling Name)

Provided (User Identity = Number)

FXO Lines Not Provided Not Provided (Must Utilize an Attendant or Auto Attendant)

For outgoing calls from a Clearspan user to the PSTN, the following information is delivered:

Trunk Type Calling Party Called Party CAS Trunks Provided

(Based on Gateway or Connected Switch) Provided (Normalized Destination Number)

PRI Trunks Provided (User Identity = Number)

Provided (Normalized Destination Number)

FXO Lines Not Provided Provided (Normalized Destination Number)

PSTN gateways can be placed at any site within an enterprise. Using the sophisticated translations and routing policies used by the Network Server, calls from users within an enterprise can utilize any PSTN gateway at any site within the enterprise. Calls to users can be received from the PSTN gateway co-located with their site, or can be relayed on-network to a remote site. A variety of routing policies are available, based on the call origin, destination or call type, allowing a high degree of flexibility when configuring a system.

9 Network Infrastructure Components

9.1 LAN Requirements An obvious key element in the delivery of VoIP services over a data network is the architecture and performance related to that data network. The architectures and configurations available and diverse types of equipment deployed make it impractical to outline a recommended data network configuration. In general, some basic considerations must be given to how that IP network will perform. The enterprise must ensure that the network is able to meet quality considerations as VoIP traffic is very sensitive to various network characteristics. Characteristics to keep in mind are: Delay — The amount of time it takes a packet to move from one point in the network to another. VoIP can typically tolerate delays up to approximately 150ms before the quality of the call becomes unsatisfactory with less than 70ms providing for excellent voice quality. Jitter — The variation in delay over time. If the delay varies too widely (i.e., jitter is large) in a VoIP call, the call quality is degraded. The amount of jitter that can be tolerated in the network depends on the depth of the jitter buffer on the network elements in the voice path. The greater the jitter buffer, the more the network, in general, can reduce the effects of jitter.



Packet loss — The loss of packets along the voice path can severely degrade the quality of VoIP calls. VoIP networks should achieve packet loss of less than 1% for satisfactory voice quality, closer to zero provides for excellent voice quality. The enterprise must understand the services being offered that could affect overall network design (i.e. converged voice and data) while still allowing these metrics to be achieved. Mechanisms such as QoS—which is dependant on what is supported by the end points (i.e. IADs and/or Network Gateways)—will need to be employed. In addition, voice CODECs that are enabled in the network will play a part in network design as they impact the amount of bandwidth required, and some CODECs are more susceptible to network problems. 10/100 BaseT Switched Ethernet Infrastructure -- All VoIP endpoints should be connected to the LAN through layer 2 IP switches. This puts each endpoint onto a separate segment allowing maximum bandwidth at the edge of the network. The “hub and spoke” topology of a switched layer 2 network also provides excellent security against “snooping” attacks within the enterprise. Virtual LANs -- Ideally, VoIP endpoints should be deployed on separate physical LANs from data endpoints. This gives voice traffic and data traffic their own layer 2 broadcast and collision zones. This eliminates the possibility of large bursty data packets (like a file transfer application) from colliding with streaming voice packets on the local network segment. However, in some enterprises this may not be practical, as it would require rewiring every station with two Ethernet ports – one for voice and one for data. Fortunately, virtual LAN technology has been widely adopted by all mainstream layer 2 switching manufacturers. The accepted standard in the industry is IEEE 802.1q – commonly called “VLAN”. VLANs allow network administrators to easily group endpoints into logical LANs, regardless of their physical location. This also simplifies moving endpoints throughout the network. Note that when employing VLANs, it is important to make sure all network elements conform to the same VLAN standard. Traffic Classification -- Eventually all packets in a converged network must converge. Typically, this happens at a layer 3 switch or router. If all traffic passing through the router were treated equal, then voice over IP service would degrade during peak office hours as both data and voice contended for the same routing resources. Congestion at the router would cause both voice and data packets to be equally lost and delayed. Lost data packets can be easily retransmitted without the user noticing, but too many lost voice packets will become very noticeable to the user. Variably delayed voice packets will result in an annoying jitter in the speech. To avoid degradation of voice service, a router must be able to route voice packets with a different priority than it routes data packets. To do this, a number of traffic classification technologies have evolved. The predominant ones to consider are: Class of Service -- Also known as IEEE 802.1p, or CoS. This is a layer 2 traffic classification technology that extends and is dependant on the VLAN IEEE specification 802.1q. Using 802.1p to classify packets on VLANs allows network administrators to prioritize traffic between layer 2 switching elements. IP Precedence -- This is a layer 3 traffic classification technology that the Type of Service byte in the IP header to classify traffic. Layer 3 switches or routers can then use this classification to apply different routing priorities and queuing policies to traffic. DSCP -- Differentiated Service Code Points, or DiffServ is a layer 3 traffic classification technology. It uses the Type of Service byte in the IP header. Packets in a DiffServ network are classified on a per hop basis – as opposed to IP precedence which is intended to specify a network wide classification. Network wide classification is not practical in large networks managed by different groups or organization – which makes DiffServ a more practical choice for wide area network traffic classification. Regardless of



the traffic classification technology, it is important that packets are classified as close to the edge as possible. This means that traffic classification should ideally be performed at the VoIP endpoints. Traffic Prioritization -- Once packets have been classified, it is important that the network infrastructure transport the packets with the appropriate priority. This is achieved in routers and switches by defining different queues for passing traffic. Each queue is given a different routing priority and mapped to a corresponding class of traffic. Then a queuing algorithm determines how each packet is de-queued for routing. There are many queuing algorithms to choose from, but a common queuing algorithm for converged voice over IP networks is called “Low Latency Queuing”. This algorithm defines a strict priority queue for low latency traffic like voice. Voice traffic gets priority and all other traffic is distributed using a class-based weight fair queuing (CBWFQ) model. Inline Power -- Inline power delivers DC power to endpoints over the same physical CAT-5 wiring used for the layer 2 network. Not only does this simplify deployment of IP phones, it also allows network administrators to use centralized battery backup solutions to keep the voice network operational in the event of a power loss. It is important to choose a matching inline power technology. Inline power, often call Power over Ethernet (POE), is standardized through the IETF.

9.2 Using DHCP DHCP is used to dynamically allocate and assign IP addresses. DHCP allows you to move network devices from one subnet to another without administrative attention. If using DHCP, you can connect Aastra SIP IP phones to the network and become operational without having to manually assign an IP address and additional network parameters. Aastra SIP IP phones comply with the DHCP specifications documented in RFC 2131. By default, Aastra SIP IP phones are DHCP-enabled. Mandatory DHCP settings are:

• IP Address • Subnet Mask • Default Route • DNS Server • Domain

Optional settings are (one of the following) • DHCP option 150 (TFTP server IP address) • Standard DHCP option 66 (TFTP server name)

TFTP servers are used to provide specific and general configuration information for all IP phones on the network. Specifying TFTP server address via DHCP eliminates any manual phone based configuration. For more information, see the Using TFTP section.

9.3 Using DNS Clearspan, and Clearspan hosted endpoints, use DNS for the location of service resources. Service resources consist of:

• Proxy/Registrar Location

• Network Routing Policy Location

• Gateway Location



All network devices use DNS SRV (RFC 2782) to locate service nodes on the network. DNS SRV specifies resource location using predicable cost and weights for each service request. Request resolutions are based on protocol, and return weighted device locations that are cached by each device. Examples of weighted service locations include:

• Active/Hot Standby Application Servers

• Load Balanced Network Servers

• Load Balanced Gateway Interfaces

Implementing DNS SRV for voice services enables administrators to modify network addresses for entire networks, from a single location.

9.4 Using TFTP There are two configuration files that you can use to define a phone’s SIP parameters; the default configuration file and the phone-specific configuration file. These configuration files must be stored in the root directory of your TFTP server.

You can use the default configuration file to define values for SIP parameters that are common to all phones. You can use the phone-specific configuration file to define parameters that are specific to a phone. Phone-specific parameters should only be defined via a phone-specific configuration file or manually configured. Phone-specific parameters should not be defined in the default configuration file.

Configuration File Guidelines

When modifying the default configuration file and creating the phone-specific configuration files, adhere to the following guidelines and requirements:

• Except for parameters used to define the lines and users on a phone, all other SIP parameters can be defined in either the default configuration file or the phone-specific configuration file. However, for network control and maintenance purposes, we recommend that you define the parameters that you want to apply to all phones in the default configuration file.

• SIP parameters specified in the default configuration file will override those parameters stored in Flash memory. Parameters specified in a phone-specific configuration file will override those stored in Flash memory or specified in the default configuration file.

• The name of each phones' phone-specific configuration file is unique and is based on the MAC address of the phone.

10 Operations, Administration, Maintenance and Provisioning

10.1 Configuration and Provisioning Clearspan is managed through an intuitive web portal. It provides separate web portals for end-users, department administrators, site administrators, enterprise administrators, and system administrators. Each level of the management hierarchy can configure and manage their respective resources in real-time. All service activation and configuration changes are implemented immediately. Users and administrators are empowered with more control and responsiveness. The functions of each level are described below:

System Provider (System Administrator)



Monitors and maintains Clearspan

Access to all levels of administration

Sets and manages system parameters

Adds, modifies, or deletes enterprises, groups, and users

Enterprise Administrator

Adds, modifies, or deletes groups and users

Adds groups and group administrators

Provisions phone numbers and access resources

Provisions services/Customizes web branding

Group Administrator (Site Administrator)

Adds, modifies, or deletes users

Assigns phone numbers and access resources

Sets and manages services for group

Department Administrator

Adds, modifies, or deletes users

Sets and manages services for department

End User

Customizes individual phone services and features

Makes calls using the CommPilot Call Manager

In addition to the web portal, the system administrator also has access to the command line interface (CLI). The CLI allows access to system configuration functions, as well as monitoring and management functions.

10.1.1 Initial System Configuration

Upon initial installation, the following functions can be performed on the Application Server using the command line interface:

Application Server Aliases - Set all the names that identify the Application Server as a proxy or call agent, including the FQDN for the redundant pair

Network Server Identity - Set the Network Server address

Enable Media Server Selection – Enable to perform Media Server selection using the policies on the Network Server

Enable AS/NS Sync API – Enable to automatically propagate group and user phone number additions and deletions to Network Server

Enable Accounting – Enable to generate call detail records for each call, logging the records into a flat file in XML or CSV format with automatic FTP transfer to an external server

Configure SNMP – Add destination addresses for traps to access control list

Intra-Group/PSTN Routing – Set to route all inter-group calls to the Network Server

Voice Messaging SMTP/POP3 Setup – Set SMTP addresses for email dispatch, and POP3 address for email retrieval



System Domain – Set to the default system domain can be configured, which allows users belonging to the default domain to login with username only

SIP Realm – Set to use the domain as the realm for SIP authentication challenges

The following functions can be performed on the Application Server using the system provider web portal:

Administrators – Add, modify or remove system administrators or provisioning administrators (only required if there is a need for multiple system administrators)

Announcements and Tones – Select and upload the messages to be played for system announcements and the tones (only required if the defaults are not sufficient)

Password Rules – Define the password rules that users and administrators must follow to create or update passwords (only required if the defaults are not sufficient)

Web Branding – Modify the look and feel of the web portal for all administrators and users (only required if the default is not sufficient)

Services – Set the system parameters for services (only required if the defaults are not sufficient)

Group

Account/Authorization Codes - Select the messages to be played for Account/Authorization Code prompts and failures

Call Park - Select the messages to be played for Call Park prompts and failures

Customer Originated Trace - Select the messages to be played for Customer Originated Trace prompts and failures

Emergency Zones - Select the messages to be played for Emergency Zones call failures

Incoming Call Plan - Select the messages to be played for Incoming Call Plan prompts and failures

Instant Conferencing - Select the messages to be played for Instant Conferencing prompts and failures

LDAP Directory - Configure a system-wide LDAP Directory

Outgoing Call Plan - Select the messages to be played for Outgoing Call Plan prompts and failures

Voice Messaging - Configure the system-wide Voice Messaging settings

User

Anonymous Rejection - Select the messages to be played for Anonymous Rejection prompts and failures

Call Forwarding Always - Select the messages to be played for Call Forwarding Always prompts and failures

Call Forwarding Busy - Select the messages to be played for Call Forwarding Busy prompts and failures

Call Forwarding No Answer - Select the messages to be played for Call Forwarding No Answer prompts and failures

Call Notify - Select the messages to be played for Call Notify prompts and failures



Call Return - Select the messages to be played for Call Return prompts and failures

Do Not Disturb - Select the messages to be played for Do Not Disturb prompts and failures

Speed Call 8 - Select the messages to be played for Speed Call 8 prompts and failures

Speed Call 100 - Select the messages to be played for Speed Call 100 prompts and failures

Upon initial installation, the following functions can be performed on the Network Server using the command line interface:

Network Server Identity - Set the Network Server address and aliases

NNACL and LCA Files – Install Telcorida NNACL and LCA files, used for public routing

Public Policy Defaults – Verify and modify policies for call typing, call screening, subscriber location, near end routing, far end routing and service center routing

Media Server Selection – Verify and modify settings for media server selection

Upon initial installation, the following functions can be performed on the Media Server using the command line interface:

Signaling Interfaces – Configure the addresses used for signaling

RTP Interfaces – Configure the addresses used for media

Enable RFC2833 – Enable DTMF via In-band RTP packets

Codecs – Set the preferred codec for the media server

10.1.2 Creating an Enterprise

After the initial system configuration is performed, an enterprise must be created. The enterprise represents the entire organization to be served or a significant subset. Multiple enterprises may exist on a system. Each enterprise has a virtual system, which is completely independent of other enterprises.

The system administrator uses a wizard which requests the information necessary to create and populate a new enterprise. The administrator can add, modify, delete, and manage groups, resources and additional enterprise administrator accounts.

The following functions can be performed on the Application Server using the enterprise web portal:

Profile Information – Add or modify enterprise profile information (name, contact information, location, etc.)

Enterprise Administrator – Add, modify or remove enterprise administrators

Domains – Add domains associated with an enterprise (used to scope user names for login, for SIP authentication realms and for SIP URL identification and calling)

Phone Numbers – Authorize the list of phone numbers and/or number ranges which the enterprise is allowed to use

Services – Authorize the list of services and service quantities which the enterprise is allowed to use

Conference Ports – Allocate conference ports which the enterprise is allowed to use



Password Rules – Define the rules that users and administrators must follow to create and update passwords (only required if the default is not sufficient)

Web Branding – Modify the look and feel of the web portal for users and administrators (only required if the default is not sufficient)

10.1.3 Creating a Group

After an enterprise is created, a group can be created. The group represents a site or location of an enterprise. Multiple groups may exist per enterprise. Each group has a virtual system, which is completely independent of other groups.

The enterprise administrator uses a wizard which requests the information necessary to create and populate a new group. The group can add, modify, delete, and manage departments and users, manage group services and manage group administrators.

The following functions can be performed on the Application Server using the group web portal:

Profile Information – Add or modify group profile information (name, contact information, location, etc.)

Group Administrator – Add, modify or remove group administrators

Departments – Add, modify or remove departments within the group

Department Administrator – Add, modify or remove department administrators

Policies – Configure group level access and authorization policies

Devices – Add, modify or remove IP phones, Integrated Access Devices, analog line gateways, etc which the group is allowed to use

Assign Domains – Assign a domain from the enterprise to a group (used to scope user names for login, for SIP authentication realms and for SIP URL identification and calling)

Assign Phone Numbers – Assign the list of phone numbers and/or number ranges from the enterprise which the group is allowed to use

Assign Services – Assign the list of services and service quantities from the enterprise which the group is allowed to use

Assign Conference Ports – Assign conference ports from the enterprise which the group is allowed to use

New User Services Template - Add or remove services for the user template, which is applied when a new user is created

Calling Line ID - Set the name and/or number of the group for outgoing calls, which is used instead of a user's name and number (optional)

Common Phone List – Setup common phone numbers for the group phone list (optional)

Feature Access Codes - Specify feature access codes (also known as star codes) and feature code prefixes associated with the group's services (optional)

Password Rules - Define the password rules that users and administrators must follow to create and update passwords (optional)

Configure Device - Load or modify the default configuration file for an access device (optional)

Digit Collection - View or modify the digit map for the group (optional)



Extension Dialing - Configure extension dialing for the group (optional)

LDAP Directory - Configure the LDAP directory for the group (optional)

10.1.4 Configuring Translations and Routing

The translations and routing policies for an enterprise are configured. This includes the routing of private calls as well as public calls. The following functions can be performed on the Network Server using the command line interface and web portal:

Hosting Network Elements - Configure the application servers as hosting network elements (hosting users)

Routing Network Elements – Configure PBX and PSTN gateways as routing network elements (inter-connection points with other networks or devices)

Media Servers – Configure media servers list, used for selecting a resource when requested by the application server

Enterprise Entity – Setup enterprise entity for entire enterprise (corresponds to a specific enterprise)

Voice VPN – Setup voice VPN dialing between different sites, including sites with legacy PBXs (routing network elements)

Least Cost Routing – Setup far end hop-off routing (use gateway closest to destination for public calls) or enterprise gateway routing (use gateway closest to origin for public calls)

Service Center Routing – Setup translations and routing for emergency, operator and directory assistance calls

10.1.5 Bulk Provisioning

Bulk provisioning can be used with the command line interface. The CLI can parse and execute files containing commands, allowing a batch style processing mechanism. This is useful for creating templates for commonly used configurations.



Technology

Clearspan Enterprise Guide