Avaya IP Telephony BCM User Manual

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BCM Rls 6.0

IP Telephony

Task Based Guide

IP Telephony

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Table of Contents

IP Telephony...................................................................... 7

Overview .......................................................................................... 7

IP Telephones and VoIP Trunks ...................................................... 8

IP Telephones .................................................................................................... 8

VoIP Trunks ....................................................................................................... 8

Supporting Information ...................................................................................... 9

Key IP Telephony Concepts ............................................................................ 11

Remote Working Capability ............................................................................. 14

Required Information ..................................................................... 15

Flow Charts .................................................................................... 16

IP Telephone Configuration ............................................................................. 16

VoIP Gateway Configuration ........................................................................... 17

General Configuration .................................................... 18

Keycodes ....................................................................................... 18

Published IP Interface .................................................................... 19

Media Gateways ............................................................................ 22

Quality of Service (QoS) Settings .................................................. 22

DSCP Marking ................................................................................................. 23

DSCP Mapping ................................................................................................ 24

IP Telephones .................................................................. 26

DHCP Configuration ...................................................................... 26

DHCP Server - IP Terminal Options ................................................................ 26

Configuring the DHCP Address Ranges ......................................................... 29

Preparing Your System for IP Telephone Registration .................. 31

Registering the IP Phones to the System ...................................... 33

COLOR*SET .................................................................................................... 34

Configuring Telephone Settings ...................................................................... 34

IP Telephone Configuration Parameters – (On Phone‟s Display) ................... 38

Troubleshooting IP Telephones ....................................................................... 40

Deregistering IP Telephones ........................................................................... 41

Remote Worker Solution ................................ ................................ 43

Example Scenario and Configuration Overview .............................................. 43

BCM Configuration........................................................................................... 44

Router Configuration ........................................................................................ 49

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Configuring the Remote IP Phone ................................................................... 49

Remote Worker Security Considerations ........................................................ 49

2050 IP Softphone ......................................................................... 49

Licensing .......................................................................................................... 50

Minimum PC Requirements ............................................................................. 51

Supported Operating Systems ......................................................................... 51

USB Audio Kit .................................................................................................. 52

Installing the 2050 IP Softphone ...................................................................... 52

Configuring the 2050 IP Softphone.................................................................. 59

Licensing the i2050 Using the BCM HTTP Server Method ............................. 63

Registering the 2050 IP Softphone .................................................................. 67

Using the 2050 IP Softphone ........................................................................... 70

IP Terminal Features ..................................................................... 76

Feature List ...................................................................................................... 76

Feature List IP Set Usage ................................................................................ 78

Key Labels ....................................................................................................... 78

Hot Desking ..................................................................................................... 79

Keeping Call Forward Settings when IP Phones are Disconnected ................ 81

VoIP Gateways ................................................................ 83

Configuring the Local Gateway Settings ........................................ 83

IP Trunks .......................................................................................................... 84

H.323 Settings ................................................................................................. 85

SIP Settings ..................................................................................................... 88

H323 & SIP Media Parameters ...................................................... 89

H323 Media Parameters .................................................................................. 90

SIP Media Parameters ..................................................................................... 92

Private SIP Specific Configuration ................................................. 94

SIP Proxy ......................................................................................................... 94

SIP URI Map .................................................................................................... 96

SIP Authentication ........................................................................................... 97

SIP Trunk Settings ......................................................................................... 100

Public SIP Trunk Configuration .................................................... 102

Importing an ITSP Template .......................................................................... 102

Creating an ITSP Account ............................................................................. 106

Checking the Public IP Address .................................................................... 117

Configuring a SIP Public Route ..................................................................... 121

Remote Gateways (Routing Table) .............................................. 123

H.323 Routing Tables .................................................................................... 123

SIP Routing Tables ........................................................................................ 126

VoIP Trunk Call Routing Summary ................................................................ 129

Tandem Switching Example ........................................................ 130

Set-up Procedures for BCM with PSTN Connection ..................................... 130

Set-up Procedures for BCM with no PSTN Connection ................................ 134

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Additional Information .................................................. 139

1100 Series VPN Client Termination ........................................... 139

Supported Phones ......................................................................................... 139

Supported Main Office Routers ..................................................................... 139

VPN IP Phone Licensing ............................................................................... 140

VPN IP Phone Provisioning ........................................................................... 140

VPN Router Configuration ............................................................................. 140

Manually Configuring the IP Phone with the VPN Settings ........................... 141

Avaya Documentation Links ........................................ 145

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Overview

IP Telephony is the technology of transmitting voice conversations over a data network infrastructure using IP (Internet Protocol). IP Telephony is the ability to make a phone call using an IP based device, optionally via gateways such as the Business Communications Manager or using Internet Telephony Service Providers (ITSPs). This convergence of voice, video, and data enhances our ability to collaborate with tools such as video conferencing and other data related facilities.

Business Communications Manager (BCM) with Voice over IP (VoIP) provides several business critical advantages:

 Cost Savings. IP networks can be significantly less expensive to

operate and maintain than traditional networks. The simplified network infrastructure of an Internet Telephony solution cuts costs by connecting IP telephones over your LAN and eliminates the need for dual cabling. IP Telephony can also provide “internal” dialling capability on site-to-site calls via global four-digit dialling plans.

 Portability and flexibility. Employees can be more productive

because they are no longer confined by geographic location. IP telephones work anywhere on the network, even over a remote connection. Network deployments and reconfigurations are simplified, and service can be extended to remote sites and home offices over cost-effective IP links.

 Simplicity and consistency. Customers can centrally manage the IP

Telephony infrastructure from a central point via the Element Manager application. The ability to network existing PBXs using IP can bring new benefits to a business. For example, the ability to consolidate voicemail onto a single system, or to fewer systems, making it easier for voice mail users to network.

 Compatibility. IP Telephony is supported over a wide variety of

transport technologies. A user can gain access to just about any business system through a Digital Line, a LAN, frame relay, asynchronous transfer mode, SONET or wireless connection.

 Scalability. A future-proof, flexible, and safe solution, combined with

high reliability, allows a company to focus on customer needs, not network problems.

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Note: All IP Clients require licence seats enabling on the BCM to allow registration and functionality. The 2050 IP Softphone requires additional per seat licensing, as does the 1100 series VPN feature. The Remote Worker Solution (NAT traversal) also requires licensing, on a system-wide rather than per seat basis.

IP Telephones and VoIP Trunks

This guide describes two similar applications for IP telephony on the BCM system: IP telephones and VoIP trunks. These applications can be used separately or together as a network voice/data solution.

IP Telephones

IP telephones offer the functionality of regular telephones, but do not require a hardwire connection to the BCM. Instead, they must be plugged into an IP network that is connected to the LAN or WAN card (BCM50(b)e only) on the BCM.

Calls made from IP telephones through the BCM can pass over VoIP trunks or across a Public Switched Telephone Network (PSTN).

Avaya provides a range of IP telephones. The „i-series‟ telephones are hardwired to the system, in the case of the i2001, i2002, i2004, i2007 as well as the newer 1110, 1120E, 1140E, 1210, 1220, 1230 and the i2033 IP conference phone, or are accessed through your desktop or laptop computer as in the case of the IP Softphone 2050.

VoIP Trunks

VoIP trunks (Lines) allow voice signals to travel across IP networks. A gateway within the BCM converts the voice signal into IP packets, which are then transmitted through the IP network. The device at the other end reassembles the packets into a voice signal. NetMeeting is one of the H.323 protocol trunk devices that the BCM system supports.

H.323 is a standard for packet based multimedia communications systems. H.323 is widely used as the standard for IP telephony and allows for the voice packets to traverse an IP network. It was designed for multimedia communication over IP networks, including audio, video, and data conferencing. The most widely deployed use of H.323 is "Voice over IP" followed by "Videoconferencing".

SIP Session Initiation Protocol is text based application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. It can be used to create two or multiparty VoIP telephone calls. Name Translation and User Location is utilised where SIP translates an address to a name and thus reaches the called party at any location.

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Note: VoIP trunks are enabled via keycodes. The number of licence seats applied determines the maximum number of simultaneous calls via VoIP trunks.

Supporting Information

The following sections contain information the might be useful when considering network design and integration of BCM VoIP functionality into the network.

SIP Trunk Authentication

Ensures that only gateways with valid credentials can place calls to the BCM and that BCM can provide valid credentials on outgoing calls when challenges take place.

DNS (Domain Name Service)

DNS can be used to locate SIP servers. This means that customers do not need to know the IP addresses of remote servers and can use domain name entries instead.

SIP Proxy Failover

Enables use of multiple SIP Proxies without relying on DNS query method with multiple entries.

SIP REFER

Standards based method for handling incoming SIP REFER messages to support Call Transfer requests in a SIP network environment.

G.711 Fax Support

Option to use G.711 when placing calls from fax machines.

IP Network

The network administrator should be able to advise you about the network setup and how the BCM fits into the network.

WAN

A Wide Area Network (WAN) is a communications network that covers a wide geographic area, such as a state or country. If you want to deploy IP telephones that will be connected to a LAN outside of the LAN that the BCM is installed on, you must ensure the BCM has access to a network device that has a WAN connection. This includes ensuring that you obtain IP addresses and routing information that allows the remote telephones to find the BCM, and vice versa.

LAN

A Local Area Network (LAN) is a communications network that serves users within a confined geographical area. For BCM, a LAN is any IP network

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connected to a LAN Interface on the BCM system. Often, the LAN can include a router that forms a connection to the Internet.

Public Switched Telephone Network

The PSTN can play an important role in IP telephony communications. In many installations, the PSTN forms a fallback route. If a call across a VoIP trunk does not have adequate voice quality, the call can be routed across the PSTN instead, either on public lines or on a dedicated ISDN connection between the two systems. The BCM also serves as a gateway to the PSTN for all voice traffic on the system.

Gatekeeper

A gatekeeper tracks IP addresses of specified devices, and provides authorisation for making and accepting calls for these devices. A gatekeeper is not required for the BCM system, but can be useful on networks with a large number of devices.

A gatekeeper controls all H.323 clients (endpoints like MS Netmeeting) in its zone. Its primary function is to address translation between alias addresses and IP addresses. This way you can call "Fred" instead of knowing which IP address he currently works on. VoIP gateways can register at the gatekeeper and the gatekeeper finds the right gateway to use to call a specific number.

For example in the diagram below digital telephone A wants to call IP telephone B, which is attached to BCM B, over a network that is under the control of a gatekeeper. Digital telephone A sends a request to the gatekeeper. The gatekeeper provides Digital telephone A with the information it needs to contact BCM B over the network. BCM B then passes the call to IP telephone B.

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Below is a diagram showing an example of a VoIP Network.

IP Telephony

Key IP Telephony Concepts

In traditional telephony, the voice path between two telephones is circuit switched. This means that the digital connection between the two telephones is dedicated to the call. The voice quality is usually excellent, since there is no other signal to interfere.

In IP telephony, voice quality between IP telephones can vary significantly from call to call and time of day. When two IP telephones are on a call, each IP telephone encodes the speech at the handset microphone into small data packets called frames. The system sends the frames across the IP network to the other telephone, where the frames are decoded and played at the handset receiver. If some of the frames get lost while in transit, or are delayed too long, the receiving telephone experiences poor voice quality.

Codecs

The algorithm used to compress and decompress voice is embedded in a software entity called a codec (COde-DECode). Two popular Codecs are G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per second (kbps) while G.729 samples at a far lower rate of 8 kbps. Voice quality is

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better when using a G.711 CODEC, but more network bandwidth is used to exchange the voice frames between the telephones.

If you experience poor voice quality, and suspect it is due to heavy network traffic, you can get better voice quality by configuring the IP telephone to use a G.729 CODEC.

The BCM supports these codecs:

 G.729  G.723  G.729 with VAD (Voice Activity Detection - the transmission of "silent

packets" over the network)

 G.723 with VAD  G.711-uLaw  G.711-aLaw

BCM allows for CODEC renegotiation. This means that two sets and/or trunks using dissimilar CODEC settings, when initiating the VoIP call, would negotiate and decide which CODEC to use. In earlier BCM software levels, differing CODECS would have meant that the call would be dropped.

Jitter Buffer

Voice frames are transmitted at a fixed rate, because the time interval between frames is constant. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames. This is called jitter, and degrades the perceived voice quality. To minimize this problem, configure the IP telephone with a jitter buffer for arriving frames.

This is how the jitter buffer works - Assume a jitter buffer setting of five frames:

 The IP telephone firmware places the first five arriving frames in the

jitter buffer.

 When frame six arrives, the IP telephone firmware places it in the

buffer, and sends frame one to the handset speaker.

 When frame seven arrives, the IP telephone buffers it, and sends frame

two to the handset speaker.

 The net effect of using a jitter buffer is that the arriving packets are

delayed slightly in order to ensure a constant rate of arriving frames at the handset speaker.

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The below diagram shows a Jitter Buffer example assuming a jitter buffer setting of five frames:

Possible jitter buffer settings and corresponding voice packet latency (delay) for the BCM system IP telephones are:

 None  Small (G.711/G.729: 0.05 seconds)  Medium (G.711/G.729: 0.09 seconds)  Large (G.711/G.729: 0.15 seconds)

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QoS Routing

The process of prioritizing data frames is referred to as Quality of Service (QoS) routing.

The BCM system supports QoS routing, when it is integrated with other Avaya routing solutions. The BCM system can also be configured to monitor QoS so that the system reverts to a circuit-switched line if a suitable QoS cannot be guaranteed.

VoIP packets can also be “marked” using DSCP, with the aim of prioritising

these packets through the network.

Remote Working Capability

The latest release of BCM offers the option of being able to use an IP Telephone in remote locations, as it were a phone on the local system. The Remote Worker solution only requires standard routers and networking capability to perform this function. If necessary, the IP telephone can be moved to various locations as required, as long as there is network access to the BCM.

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A typical example of the Remote Worker solution would be a home worker who wishes to connect an IP telephone to the main office BCM, using their standard home router and the internet. The office BCM would be connected to the internet via a router which has a static public IP address, and forwards the IP telephone‟s data/voice traffic to the BCM (and vice-versa).

Alternatively, if extra security is required for the data/voice traffic, a VPN connection can be initiated via the 1120 and 1140 IP telephones. This requires enhanced IP phone configuration, and a VPN router at the main office hosting the BCM.

Required Information

Before configuring IP Telephony, the following information will need to be confirmed:

 Which interface will be used for the Published IP address?  Is there a Gatekeeper connected to the BCM, if so, what is the IP

address of the Gatekeeper and the Alias name for the BCM?

 If there is no Gatekeeper, what are the IP addresses of the remote

Gateways and what are the telephony destination digits required to dial those systems?

 What password will be used for IP Phone registration?  Are there any routers that should be referenced as part of the VoIP

configuration? These may be used to provide WAN access for example.

 If using the Remote Worker or 1100 series VPN solutions, what is the

public IP address of the router connecting the BCM to the Internet/WAN network.

 What telephony configuration is required for IP Telephony?  Will DHCP be required for the IP Phones, and if so, will the BCM be set

up to provide IP Addresses to the phones?

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Will the BCM be used to issue IP Addresses to the IP phones?

Ensure that the required keycodes are applied to the BCM: refer to the Keycodes section of this guide.

Set the BCM‟s IP Address that the IP phones will

Refer to the DHCP Configuration section of this guide.

Will the 2050 IP Softphone be used?

Refer to the 2050 IP Softphone section of this guide.

IP Phones have been configured for use.

Yes

Set the BCM up to allow IP phones to register:: refer to the Preparing Your system for IP Telephone

Registration section of this guide.

Flow Charts

Use the following flow charts to determine which sections of this guide to use.

IP Telephone Configuration

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VoIP Gateway Configuration

Will SIP be used over the VoIP trunks?

Determine how incoming and outgoing calls will be handled: refer to the Configuring the Local Gateway Settings section of this guide.

Check the H323 and/or SIP Media Parameters: refer to the H323 & SIP Media Parameters section of this guide.

Refer to the Private SIP Specific Configuration section of this guide.

If not using a Gatekeeper on the network, manually configure the Remote Gateways: refer to the Remote Gateways (Routing Table) section of this guide.

Yes

Will the SIP trunks be private to another system, or public to an ITSP?

Refer to the Public SIP Trunk Configuration section of this guide.

Private

Public to ITSP

IP Telephony

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General Configuration

The BCM supports the following IP telephony protocols: UNISTIM, H.323 and SIP.

 The IP telephones use UNISTIM.  The Symbol NetVision and NetVision Data telephones use H.323+.  VoIP Trunks can use either H.323 or SIP (defined on a per gateway

basis)

The applications that control these protocols on the BCM provide an invisible interface between the IP telephones and the digital voice processing controls on the BCM.

Keycodes

The first part of configuration for IP Telephony is ensuring that the required keycodes have been purchased and are entered.

1. In Element Manager, select the Configuration tab and then open the System folder. Select the Keycodes link and the keycodes that have been entered will be displayed.

2. Three keycode types are available, depending on your requirements:  VoIP (H.323) or SIP GW Trunks: two trunk protocols for networking

between compatible telephone systems. The number of trunk licence seats enables determines the maximum number of VoIP calls that can be placed over VoIP trunks. SIP GW trunks will be required to connect to ITSPs.

 IP Clients: The number of IP Client licence seats determines the

number of IP Phones and Software IP Phones that can be registered against the BCM.

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Note: The exception to this rule is when registering telephones to be used Remote Worker sets. Please refer to the Remote Worker Solution section of this guide for instructions on S1/S2 assignment for this feature.

Note: The Published IP Address is the address that LAN CTE should also register against. For further information, refer to the LAN CTE Guide.

 Remote Worker: A single keycode unlocks the Remote Worker

solution

Published IP Interface

The Published IP Interface is the IP Address that IP Telephones need to register against as well as the address that VoIP gateways need to be “pointed” to. You have the choice of selecting the Customer LAN (refer to the Configuring the LAN IP Address section of the System Start Up Guide) or any VLAN IP Addresses (refer to the VLANs Guide) that are configured on the BCM in the IP Subsystem section of Element Manager.

The Published IP Address must be set as the S1 IP (or S2 IP if the BCM will be used as a “backup” registration BCM) when configuring IP phones for registration.

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Use the following procedure to check or set the Published IP Address.

1. From the Configuration tab, open the System folder and select IP Subsystem. Click on the General Settings tab.

2. If checking the existing Published IP Address for IP phone registration purposes, view the read-only field.

3. If changing the setting, from the Published IP Interface drop-down list, select the Customer LAN or any of the VLANs configured on the BCM.

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4. A warning box will appear stating that all Voice over IP applications will be restarted. This may result in VoIP calls being dropped. Click OK to continue.

5. If changed, the new setting will be displayed,

6. Changing the Published IP Interface setting also has the effect of changing the S1 Primary Terminal Proxy Server IP Addresses (S1 & S2) in the DHCP Server IP Terminal DHCP Options screen (refer to the DHCP Server - IP Terminal Options section of this guide for further information).

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Attribute

Value

Description

Echo cancellation

<drop-down menu> Enabled w/NLP Enabled Disabled

Enable or disable echo cancellation for your system. Default: Enabled w/NLP (check with your internet system administrator before changing this) Echo Cancellation selects what type of echo cancellation is used on calls that go through a Media Gateway. NLP refers to Non-Linear Processing.

T.38 UDP redundancy

If T.38 fax is enabled on the system, this setting defines how many times the message is resent during a transmission, to avoid errors caused by lost T.38 messages.

Note: If any network hardware handling network traffic does not support DSCP, the packets will not be prioritised by that hardware, and will be treated on an equal basis to non–prioritised packets.

Media Gateways

Certain types of IP communications pass through Media Gateways on the BCM. You can control the performance of these communications by adjusting the parameters for echo-cancellation and UDP Redundancy.

The Media Gateways panel allows you to set basic parameters that control IP telephony.

1. Open the Resources folder and highlight Media Gateways. The Media Gateways screen will be displayed on the right. Configure the Parameters as described in the following table.

Media Gateways Settings

Quality of Service (QoS) Settings

The BCM can be configured to mark voice related data packets using the Differentiated Services Code Point (DSCP) feature, so that they have priority over other packets on the network. Prioritised packets pass through network hardware supporting the DSCP feature, ahead of lower priority packets. This has obvious benefits for real time applications such as Voice over IP.

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Note: Only configure BCM QoS if you have a plan of what types of packets are prioritised on the network, and the levels (class of service) of priority for those packet types.

The following types of data packets can be prioritised:

 VoIP Signalling (SIP, H.323, and Unistim)  Voice Media  T.38 Fax Media (SIP or H.323)

DSCP Marking

Use the following procedure to set the QoS values for VoIP Signalling, Voice Media, and Fax Media packets.

1. In Element Manager, select the Configuration tab. Open the Data Services folder, and click on QoS.

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Note: Avaya Automatic QoS should only be used if there are other devices on the network that support this feature.

2. In the DSCP Marking tab, select either to use Avaya Automatic QoS settings or select the values for each of VoIP Signalling, Voice Media, or Fax Media.

3. A value of CUSTOM can also be selected from the drop-down lists, which will enable a customisable ToS (Terms of Service value) to be entered.

DSCP Mapping

In this area DSCP values are assigned to various service classes. The service classes determine the priority level of the DSCP value.

The available Service Classes are (in order of priority):

 Critical  Network  Premium  Platinum  Gold  Silver  Bronze  Standard

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Therefore, a packet carrying a DSCP value associated with the Critical class will have the highest priority (assuming the default VLAN P Bit Mapping settings are not changed).

1. Click on the DSCP Mapping tab. If you want to assign a different service class to a DSCP value, double-click in the corresponding Avaya Service Code field and select the class from the drop-down list.

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IP Telephony

IP Telephones

IP telephones offer the functionality of regular telephones, but do not require a hardwire connection to the BCM. Instead, they must be plugged into an IP network which is connected to the BCM.

Calls made from IP telephones through the BCM can pass over VoIP (H.323 or SIP) trunks or across Public Switched Telephone Network (PSTN) lines.

Avaya provides two types of IP telephones. The IP telephones are wired to an IP network using Ethernet in the case of the IP telephones, or are accessed through your desktop or laptop computer, as in the case of the 2050 IP Softphone.

IP telephones can be configured to the network by the end user or by the administrator. If the end user is configuring the telephone, the administrator must provide the user with the required parameters.

DHCP Configuration

Refer to the following sections if the BCM will be used as the DHCP server for the IP phones.

DHCP Server - IP Terminal Options

If the BCM is configured to pass on DHCP details to IP phones using either the “Enabled – IP Phones Only” or “Enabled – All Devices” options in DHCP Server General Settings, then the BCM should be configured to supply the Primary (S1) and Secondary (S2) Terminal Proxy Server IP Addresses that the IP Phones should register against.

If the BCM will not be passing on DHCP information to IP Phones, then the IP Terminal DHCP Options do not require configuring.

Again, if you have configured the Published IP Interface in the Published IP Interface section, the S1 and S2 will be already set to the Published IP Address. However, you may wish to check these settings.

Use the following procedure to check or change the IP Terminal DHCP Options.

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1. From Configuration tab open the Data Services folder and select DHCP Server. Click on the General Settings tab. Check to see if the BCM is configured to provide DHCP information to IP Phones.

2. If either Enabled – IP Phones Only or Enabled – All Devices is selected, then continue with configuring the IP Terminal DHCP Options.

3. Click on the IP Terminal Options tab.

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Attribute

Value

Description

Primary Terminal Proxy Server (S1)

IP Address

The IP address of the Proxy Server for IP phones. This should be set to the BCMs Published IP Address.

Port

<drop-down list>

Select the appropriate port: BCM SRG Meridian 1/Succession 1000 Centrex/SL-100 Other

Port number

The port number on the terminal through which IP phones connect.

Action

The initial action code for the IP telephone. Retry count

The delay before an IP phone retries connecting to the proxy server.

Secondary Terminal Proxy Server (S2)

IP address

The IP address of the Proxy Server for IP phones. This should be set to the BCMs Published IP Address, or a backup BCM to register against.

Port

<drop-down list>

Select the appropriate port: BCM SRG Meridian 1/Succession 1000 Centrex/SL-100 Other

Port number

The port number on the terminal through which IP phones connect.

Action

The initial action code for the IP telephone Retry count

The delay before an IP phone retries connecting to the proxy server.

VLAN

VLAN identifiers (commadelimited)

Specify the Virtual LAN (VLAN) ID numbers that are given to the IP telephones. If you want DHCP to automatically assign VLAN IDs to the IP telephones, enter the VLAN IDs in the following format: VLAN-A:id1, id3,…,idn. Where: VLAN-A – is an identifier that tells the IP telephone that this message is a VLAN discovery message. Id1, id2,…idn – are the VLAN ID numbers that DHCP can assign to the IP telephones. You can have up to 4 (BCM50) or 8 (BCM450) VLAN ID numbers listed. The VLAN ID numbers must be a number from 1 to 4094. For example, if you wanted to use VLAN IDs 1100, 1200, 1300 and 1400, you would enter the following string in this box: VLAN-A:1100, 1200, 1300, 1400. If you do not want DHCP to automatically assign VLAN IDs to the telephones, enter VLAN-A:none, in this text box. Note1: The Avaya IP Terminal VLAN ID string, must be terminated with a period (.). Note2: If you do not know the VLAN ID, contact your network administrator. Note3: For information about how to setup a VLAN, refer to the user

4. Ensure that the IP address is set correctly for the Primary and Secondary Terminal Proxy Servers. Again, these addresses will be used during the IP Phone registration process. Also, ensure that the Port is set to BCM. This will automatically set the Port number field to

7000.

5. Configure all other fields as required.

IP Terminal DHCP Options

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Attribute

Value

Description

documentation that came with your VLAN compatible switch, as well as the VLAN Guide..

Avaya WLAN Handset Settings

TFTP Server

IP Address

Enter the IP Address of the TFTP server that is used for providing firmware to the WLAN handsets and the 2245 IP Telephony Manager

WLAN IP Telephony Manager 2245

IP Address

Enter the IP Address WLAN IP Telephony Manager 2245

Note: Consult with the network administrator to determine a suitable range of addresses, co-ordinating with the existing network design. For example, it may be necessary to set up an Address Range for VLANs that host the IP telephones. For more information on configuring VLANs, please refer to the

VLANs Guide.

Configuring the DHCP Address Ranges

If the BCM is configured to pass on DHCP information to IP Phones, you should configure a suitable range of addresses to assign to the IP Phones.

1. In the Configuration panel, open the Data Services folder and select DHCP Server.

2. Click on the Address Ranges tab.

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3. If there aren‟t any Address Ranges configured, click on the Add button.

4. Enter the start IP address in the From IP Address field. Enter the end IP address of the range in the To IP address field. In the Default Gateway field, enter the IP Address of the network default gateway. This may be the BCM S1 address in some situations. Click OK to submit the settings.

5. The new address range will be displayed.

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Preparing Your System for IP Telephone Registration

Before you can register an IP telephone to the BCM, you must activate terminal registration on the BCM.

1. Open the Resources folder and select the Telephony Resources link and then select the IP Sets Module Type.

2. Select the IP Terminal Global Settings tab and select the Enable Registration tick box.

3. If you want the installers to use a single password to configure and register the telephone, select the Enable global registration password check box, and then enter a numeric password (the password will have to be entered on the IP Phone keypad) in the Global password field.

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Attribute

Value

Description

Enable registration

Select to allow new IP clients to register with the system. Warning: Remember clear this check box when you have finished registering the new telephones.

Enable global registration password

If you want to require the installer to enter a password when IP telephones are configured and registered to the system, select this check box. If this field is left blank, the IP Phone installer may be prompted to enter the User ID = 738662 and Password = 266344..

Global password

<10 alphanumeric>

Default: bcmi (2264)

If the Enable global registration password check box is selected, enter the password the installer will enter on the IP telephone to connect to the system.

Auto-assign DNs

If selected, the system assigns an available DN as an IP terminal requests registration. It does not prompt the installer to enter a set DN.

Note: For this feature to work, Registration must be selected and Password must be blank.

If not selected, the installer receives a prompt to enter the assigned DN during the programming session.

Play DTMF Tone

Allows DTMF tones to be sent via VoIP calls.

Advertisement /Logo

Any information in this field appears on the display of all IP telephones. For example, your company name or slogan.

Default codec

Auto G.711-aLaw G.711-uLaw G.723 G.729 G.729 + VAD G.723 + VAD

If the IP telephone has not been configured with a preferred codec, choose a specific codec that the IP telephone will use when it connects to the system. If you choose Auto, the IP telephone selects the codec. If you are unsure about applying a specific codec, ask your network administrator for guidance.

Default jitter buffer

None Auto Small Medium Large

Choose one of these settings to change the default jitter buffer size: None: Minimal latency, best for short-haul networks with good bandwidth. Auto: The system will dynamically adjust the size. Small: The system will adjust the buffer size, depending on CODEC type and number of frames per packet to introduce a 60-millisecond delay. Medium: 120-millisecond delay Large: 180-millisecond delay

G.729 payload size (ms)

10, 20, 30, 40, 50, 60 Default: 30

Set the maximum required payload size, per codec, for the IP telephone calls sent over H.323 trunks. Note: Payload size can also be set for IP trunks G.723 payload size (ms)

G.711 payload size (ms)

10, 20, 30, 40, 50, 60 Default: 20

4. To automatically assign a DN to the phone being registered, select the Auto-assign DNs option.

5. Configure all other options as required.

Note: Turn Enable registration and Auto-assign DNs off when the telephones are registered. Leaving your IP registration open and unprotected by a password can pose a security risk.

IP Terminal Global Settings

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Attribute

Value

Description

Support Remote Worker

Tick this box to enable the Remote Worker feature. For full information on this feature, refer to the Remote Worker Solution section of this guide.

Discovered Public Address

Read-only field. Displays the public IP address of the router the BCM is connected to, if discovered via the STUN protocol. Refer to the Remote Worker Solution section of this guide for more information.

Provisioned Public Address

Read-only field. Displays the public IP address of the router the BCM is connected to, if manually entered. Refer to the Remote Worker

Solution section of this guide for more information.

Registering the IP Phones to the System

How you configure the telephones depends on whether DHCP is active on the network. When registering the IP Phones, you have the option of selecting the DHCP setting most appropriate to the network:

 DHCP (Full): The DHCP server will provide the following information to

the IP Phones:

o IP Address & Subnet Mask o Default Gateway o S1 & S2 Addresses o Port Number, Action, & Retry Count o VLAN ID

Only use DHCP (Full) if the BCM is supplying the DHCP information to the IP Phones, or the network DHCP server can be configured to supply this information.

 DHCP (Partial): The DHCP Server will provide the following

information to the IP Phones:

o IP Address & Subnet Mask o Default Gateway

The rest of the required information will have to be entered manually. DHCP (Partial) is used in situations where the BCM is not acting as the DHCP server to the phones, but another device is. This can also be used in scenarios where the IP Phone is on a remote network.

 DHCP (Off): All information will have to be entered manually during the

registration process. Use this in situations where there isn‟t a DHCP

server on the network, or you simply want to configure the settings manually.

When the telephone registers, it downloads the information from the system IP Telephony record to the telephone configuration record. This can include a new firmware download, which occurs automatically. If new firmware downloads, the telephone display indicates the event.

Once registration has completed, you do not need to go through the registration process again, unless you deregister the terminal.

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Display Keys

COLOR*SET

If booting up a new phone for the first time, you may be immediately prompted to enter a password. If this is the case, enter COLOR*SET (26567*738) followed by OK. You can then proceed with the registration process.

Configuring Telephone Settings

If you are not automatically registered to the BCM, you can configure your telephone settings to allow you to access a BCM on the network. You will also need to perform these steps if your IP telephone is not connected to the same LAN that the BCM is connected to.

Access the configuration parameters using the method described for the model of phone, and then configure the parameters to enable phone registration.

Accessing the Configuration Parameters – i2001, i2002, i2004

1. Restart the telephone by disconnecting the power, then reconnecting the power. After about four seconds, the top light flashes and the text Avaya appears on the screen.

2. When the greeting appears, quickly press the four display keys, one at a time, from left to right. These keys are located directly under the display. These keys must be pressed one after the other within 1.5 seconds or the telephone will not go into configuration mode.

3. If the display shows EAP Enable you have successfully accessed the configuration parameters. Proceed with configuring the parameters to enable phone registration.

Note: Use OK to access the next menu item.

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Display Keys

Accessing the Configuration Parameters – i2033

1. Restart the telephone by disconnecting the power, then reconnecting the power. After about 15 to 20 seconds, the top light flashes and the text Avaya appears on the screen.

2. When the greeting appears, quickly press the three display keys, one at a time, from left to right. These keys are located directly under the display. These keys must be pressed one after the other within 1.5 seconds or the telephone will not go into configuration mode.

3. If the display shows EAP Enable you have successfully accessed the configuration parameters. Proceed with configuring the parameters to enable phone registration.

Note: Use OK to access the next menu item.

Accessing the Configuration Parameters – i2007

1. Restart the telephone by disconnecting the power, then reconnecting the power. After about four seconds, the top light flashes and the text Avaya appears on the screen.

2. When the phone has started, press the Tool icon once.

3. Select Network Configuration from the menu.

4. If the display shows EAP Enable you have successfully accessed the configuration parameters. Proceed with configuring the parameters to enable phone registration.

Note: Navigation is performed by the navigation cluster at the bottom of the phone. You can also use the pointing device as the screen is touch sensitive.

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Services Key

Accessing the Configuration Parameters – 1110, 1120e, 1140e

1. Restart the telephone by disconnecting the power, then reconnecting the power. After about 15 to 20 seconds, the top light flashes and the text Avaya appears on the bottom left of the screen.

2. Wait a further 15 – 20 seconds. Press the Services ( ) key twice. A menu will display.

3. Select Network Configuration, either by pressing the associated keypad number, or by using the navigation cluster.

4. If the display shows EAP Enable you have successfully accessed the configuration parameters. Proceed with configuring the parameters to enable phone registration.

Note: Navigation is performed by the navigation cluster in the center of the phone. The central button is the Enter or OK key.

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Services Key

Accessing the Configuration Parameters – 1210, 1220, 1230

1. Restart the telephone by disconnecting the power, then reconnecting the power. After about 15 to 20 seconds, the top light flashes and the text Avaya appears on the bottom left of the screen.

2. Wait a further 15 – 20 seconds. Press the Services ( ) key twice. A menu will display.

3. Select Network Configuration, either by pressing the associated keypad number, or by using the navigation cluster.

4. If the display shows EAP Enable you have successfully accessed the configuration parameters. Proceed with configuring the parameters to enable phone registration.

Note: Navigation is performed by the navigation cluster in the center of the phone. The central button is the Enter or OK key.

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Note: The below options may differ slightly on certain phone models.

Field

Value

Description

DHCP

0 or 1

Enter 0 if not using a DHCP server to dispense IP addresses. Enter 1 if using a DHCP server. If you choose to use the Full DHCP server option rather than allocating static IP addresses for the IP telephones, skip the remainder of this section.

DHCP Partial

0 or 1

Only appears if DHCP is selected. Enter 0 for Full DHCP or 1 for Partial DHCP.

SET IP

The set IP must be a valid and unused IP address on the network that the telephone is connected to. (refer to Network Administrator)

NETMASK

This is the subnet mask. This setting is critical for locating the system you want to connect to. (refer to Network Administrator)

DEF GW

Default Gateway on the network (i.e., the nearest router to the telephone. The router for IP address W.X.Y.Z is usually at W.X.Y.1). If there are no routers between the telephone and the BCM network adaptor to which it is connected, (for example a direct HUB connection), then enter the Published IP address of the BCM as the DEF GW. If the IP telephone is not connected directly to the Published IP address network adaptor, set the DEF GW to the IP address of the network adaptor of the router the telephone is connected to. (refer to Network Administrator)

S1 IP

This is the Published IP address of the first BCM that you want to register the telephone to. (refer to Network Administrator)

S1 PORT

Default: 7000

This is the port the telephone will use to access this BCM.

S1 ACTION

Default: 1

S1 RETRY COUNT

Set this to the number of times you want the telephone to retry the connection to the BCM.

S2 IP

This is the Published IP address of the second BCM that you want to register the telephone to. It can also be the same as the S1 setting. (refer to Network Administrator)

S2 PORT

Default: 7000

This is the port the telephone will use to access this BCM.

S2 ACTION

Default: 1

S2 RETRY COUNT

Set this to the number of times you want the telephone to retry the connection to the BCM.

VLAN

0: No VLAN 1: Manual VLAN 2: Automatically discover VLAN using DHCP

If you have DHCP set to yes, you can select number 2 if you want the system to find the VLAN port assigned to the telephone. If you do not have DHCP, or if you want to set the VLAN port number manually, select number 1. If VLANs are not used on your network, select 0.

Cfg XAS?

0: No (default) 1: Yes

If you want to enable connection to a Net6 service provider server, choose 1. You are then prompted for an IP address for the server.

IP Telephone Configuration Parameters – (On Phone’s Display)

Note: Only the settings below are required to allow the IP Telephone to be registered. Accept the defaults for all other settings.

Note: To enter a full stop () when specifying an IP Address or Subnet Mask, use the  key on the dialpad.

When you have entered all the configuration information, the telephone attempts to connect to the BCM. The message Locating Server appears on the display. If the connection is successful, the message changes to

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Connecting to Server after about 15 seconds. Initialisation may take several minutes. Do not disturb the telephone during this time.

Once the telephone connects to the server, the display shows the DN number and a date display. Alternatively, if the Auto Assign DNs option is disabled (refer to the Preparing Your System for IP Telephone Registration section of this guide) you will be prompted to enter a DN for the telephone.

Note: You will be prompted to enter a password. Enter the registration password (i.e. the Global Registration Password described in the Preparing Your System for IP Telephone Registration section of this guide) and press the OK soft key. Alternatively, if the Global Registration Password is not enabled, you may be prompted to enter the following information: Registration: SETNNA = 738662 Password: CONFIG = 266344

Note: Each of the IP Telephones can be configured with the same settings as a standard digital handset. With this in mind, each needs to be assigned Lines and / or Line pool access granted. For information on these settings, please refer to the Telephony Services Guide.

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Message

Description and solution

SERVER: NO PORTS LEFT

The system has run out of ports (license seats). This message remains on the display until a port becomes available and the telephone is powered down and then up. To obtain more ports, you can apply additional IP Client keycodes.

Invalid Server Address

The S1 is incorrectly configured with the IP address of a system network adapter other than the published IP address.

IP Address conflict

The telephone detected that a device on the network is currently using the IP address allocated to the telephone.

Registration Disabled

The Registration on the system is set to OFF.

SERVER UNREACHABLE. RESTARTING

Check that you have entered the correct Netmask and gateway IP addresses. If the settings are correct, contact your system administrator.

NEW SET

The telephone has not been connected to the system before, and must be registered.

Problem

Suggested solution or cause

Telephone does not connect to system

If an IP telephone does not display the text Connecting to

server within two minutes after power up, the telephone did not

establish communications with the system. Double-check the IP configuration of the telephone and the IP connectivity to the system (cables, hubs, and so on).

Slow connection between the handset and the system

If the connection between the IP client and the system is slow (ISDN, dialup modem), change the preferred CODEC for the telephone from G.711 to G.729.

One-way or no speech paths

Signaling between the IP telephones and the system uses the system port 7000. However, voice packets are exchanged using the default RTP ports 28000 through 28255 at the BCM, and ports 51000 through 51200 at the IP telephones. If these ports are blocked by the firewall or NAT, you will experience one-way or no-way speech paths.

Change the contrast level

When an IP telephone is connected for the first time, the contrast level is set to the default setting of 1. Use FEATURE *7 and the

UP or DOWN key to adjust the contrast.

Block individual IP sets from dialling outside the system.

If you want to block one or more IP telephones from calling outside the system, use Restriction filters, and assign them to the telephones you want to block. Restriction filters are set up under

Configuration > Telephony > Call Security > Restriction Filters.

Troubleshooting IP Telephones

If a problem is encountered when IP phone attempts to register with the BCM you may see a number of messages appear on the telephones display. These are outlines as follows:

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Deregistering IP Telephones

You can deregister selected IP telephones from the system, and force the telephone to go through the registration process again. You can access the deregister button from two locations:

1. Select the Configuration tab and open the Resources folder then

select Telephony Resources.

2. Select the IP Sets bus (Configured Device column) and click on the IP

Terminal Details tab. Select the required DN, and click on the Deregister button.

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3. Alternatively open the Telephony folder, the Sets folder and highlight

Active Sets. Select the DN you wish to deregister.

4. Click the Capabilities and Preferences tab, followed by the IP

Terminal Details tab in the lower Details part of the screen. Then click the Deregister button.

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Remote Worker Solution

The Remote Worker solution provides an option for home workers, or BCM users operating on the outside of the BCM‟s network, to connect an IP Phone to the BCM. This solution does not require a VPN, and uses NAT to redirect IP Phone traffic between the connecting networks.

As the Remote Worker solution does not use a VPN (Virtual Private Network), the traffic is not encrypted, although the proprietary binary format is a form of simple encryption.

Example Scenario and Configuration Overview

Detailed below is a simple form of the Remote Worker solution. A BCM user has a home network, and wishes to connect their IP Phone to the office BCM via the internet.

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The following configuration steps are required for the above scenario:

1. The BCM has to be configured with the office router as the Default

Gateway and with the router‟s public IP Address as the Provisioned

Public Address to ensure that network traffic to the remote worker phone is correctly addressed. Additionally, the necessary entitlements of Remote Worker keycode, Support Remote Worker and Enable Registration options are required to ensure the remote phone can register and function on the BCM. The port ranges listed above are configured as default.

2. Next, the office router requires NAT/PAT configuration so that the

desired traffic types (IP Phone signalling and media (voice traffic)) are routed correctly to and from the BCM. In conjunction with NAT/PAT configuration, the Firewall should allow the same ports opening otherwise traffic destined for those ports will be blocked.

3. When the previous steps have been performed the IP Phone will be

able to register on the BCM, using the office router‟s public address as

the primary (S1) and secondary (S2) registration server addresses.

BCM Configuration

1. Launch Element Manager and connect to the BCM.

2. First, check that the Remote Worker keycode has been applied to the

BCM. In the Configuration tab, open the System folder, click on Keycodes and search for the Remote Worker item.

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3. If Remote Worker is not listed in the Feature Licenses table, contact

your keycode supplier for a keycode file containing this feature and apply the file to the BCM by clicking on the Load Keycode File… button.

4. Check that the BCM‟s Published IP Address and Default Gateway are

configured correctly. Under the System folder, click on IP Subsystem. The Default Gateway should be the LAN address of the office router (in this scenario). Also, the Published IP Address should be accessible from the router.

5. These settings should have been configured as part of the System

Start Up process. If they require changing, refer to the Configuring the LAN IP Address section of the System Start Up Guide.

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6. The Public IP Address of the router now needs to be configured on the

BCM. Under the System folder, click on IP Subsystem. In the Public Network area click on the Modify button.

7. You can choose to manually enter the public address of the router to

be used in the Remote Worker solution in the Provisioned Public Address field,

or tick the Address Discovery Flag to attempt to automatically discover the router public IP address using Stun. To do this, enter the Stun Server Address in the available field.

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8. For either method, click OK when the appropriate details have been

entered. Either the Provisioned Public Address or Discovered Public Address will be displayed, depending on which Discovery Setting method was used.

9. Next, the IP Telephony settings require configuration. Open the

Resources folder, click on the Telephony Resources folder and select IP Sets.

IP Telephony

10. In the Details area in the lower part of the screen, tick the Support

Remote Worker checkbox. Without this option enabled, remote workers will not be able to connect to the BCM. (You will notice the Provisioned/Discovered Public Address information as configured previously.) Click OK on the resulting WARNING screen (refer to the Remote Worker Security Considerations section for information on securing the system whilst the Support Remote Worker option is enabled).

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Note: It is always good practice to disable registration (un-tick the Enable registration checkbox) when known IP phones have been registered. This

prevents unauthorised phones from registering on the BCM, and using the system fraudulently.

11. Ensure that the general IP Terminal Registration details are configured

to allow IP Phones to register. Please refer to the Preparing Your System for IP Telephone Registration section of this guide for full details.

12. Lastly, check that the signalling and RTP over UDP port ranges are

entered on the BCM. Open the Resources folder and click on Port Ranges. The corresponding values should be used in the router configuration. The default values for a BCM50 are shown below. A BCM450 would have the RTP over UDP ranges of 30000 – 30999.

13. The BCM is now configured for the Remote Worker feature.

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Note: The S1 and S2 addresses entered during the registration process should be the public address of the router the BCM is connected to (e.g.

217.35.6.35 in the scenario described earlier).

Router Configuration

The office router (in this scenario) will require NAT/PAT configuration to route the remote worker IP phone signalling and media traffic to and from the BCM. Also, corresponding firewall configuration will be required to allow the signalling and media to reach the BCM, and return to the public network.

As previously described the ports that require NAT/PAT and firewall configuration are as follows:

 7000 – 7002  30000 – 30099 (BCM50)  30000 – 30999 (BCM450)

Configuring the Remote IP Phone

The IP should be registered as described in the Registering IP Phones to the System section of this guide.

Remote Worker Security Considerations

Enabling the Remote Worker feature can leave the BCM vulnerable to fraudulent use by unauthorised parties. If certain settings are left in their default state and the public IP address of the router is known, external IP phones could be registered against the BCM and fraudulent use of BCM facilities would occur.

To prevent against such fraudulent use, ensure the following security steps are taken:

 Ensure any accounts that have telset programming privileges have

their passwords changed, and that the passwords are changed on a regular basis. This will help prevent system resources being assigned to unauthorised remote sets. Refer to the User Management Guide for details on account management.

 Change the default Global Password used for registering the set.  After authorised sets have registered, disable the Enable Registration

option.

2050 IP Softphone

The 2050 IP Softphone (also referred to as the i2050) allows you to use a computer equipped with a USB headset to function as an IP terminal on the BCM system. The 2050 IP Softphone uses the computer IP network connection to connect to the BCM. Designed to look and feel like the desktop 1140 IP phone, there are also two additional compact skins, available in black and silver.

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The 2050 IP Softphone is an IP Telephony application that allows you to make calls over the LAN and WAN from your computer. The Software Phone provides classic telephony services, a local telephone Directory, easy access to Voice Mail, Caller ID information and multiple telephone lines or line appearances.

Now included with the 2050 IP Softphone are incoming and disconnect call popups, and a software Expansion Module which emulates an i2004 Key Expansion Module with 54 Keys. Calls arriving on keys on the software Expansion Module do not support incoming and disconnect popups.

The installation files for the 2050 IP Softphone are contained on a CD, which can be obtained from your BCM supplier, or from www.avaya.com.

Licensing

Each 2050 IP Softphone will require a keycode license seat on the BCM (refer to the Keycodes section of this guide). Additionally, the 2050 IP Softphone itself should be licensed, which can be achieved via one of a number of methods:

1. Using the BCM HTTP server

2. Node-Locked Licensing

3. A Licensing Server

The licensing process detailed in this guide will be the BCM HTTP server.

BCM HTTP Server Licensing

This is perhaps the simplest method of licensing the 2050 IP Softphone. License files are served from the BCM to the 2050 Softphone, unlocking the i2050 and enabling full functionality. License files are specific to each installation of the i2050.

Application of the license via the BCM HTTP server method consists of the following steps:

1. Install the i2050 on a PC.

2. Obtain the i2050 hardware ID. Your keycode supplier will need this

information.

3. Obtain the license files from your keycode supplier.

4. Upload the license files to the BCM HTTP server.

5. Set the Provisiong Server Protocolfield to HTTP and the URL to the

location of the BCM.

6. Restart the i2050. It will search for the licensing information on the

BCM and install the license, allowing the i2050 to connect to the BCM.

Full steps will be detailed in the Licensing the i2050 Using the BCM HTTP

Server Method section of this guide. Node-Locked Licensing

Node-locked licenses are specific to each i2050 installed on a specific PC. Once the licensing file is installed on the PC, the license is valid until the i2050

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is uninstalled. This mechanism negates the need for a Licensing Server to be installed on the network.

Application of the license via the Node-Locked method consists of the following steps:

1. Install the i2050 on a PC.

2. Obtain the i2050 hardware ID. Your keycode supplier will need this

information.

3. Obtain the license files from your keycode supplier.

4. Place the .license files in the default location of a TFTP server.

5. Set the TFTP server address in the i2050 Provisioning Server IP

Address field.

6. Restart the i2050. It will search for the .cfg and keycode information

and install the license, allowing the i2050 to connect to the BCM.

For further information concerning the Node-Locked Licensing method, please consult the Avaya document IP Phones Fundamentals (NN43001-368).

Licensing Server

A Licensing server can be installed on a networked PC, which will allow a certain amount of i2050s to connect to and function with the BCM. This method does not require a license to be generated for each i2050 on the KRS. Instead, a number of seats can be purchased and applied to the Licensing Server, which will then control the number of i2050s installations that can connect to the BCM.

If an i2050 is licensed via the Licensing Server method, the i2050 uses a heartbeat mechanism to validate the license every 2 mins. If the heartbeat is

lost, i.e. the i2050 can‟t connect to the server, the i2050 will try to reconnect 5

times and if the connection cannot be re-established then the i2050 will lose its licence and hence its connection to the BCM. Therefore, if using a Licensing Server it is imperative that the PC on which it is installed is available at all times.

For further information concerning the Licensing Server method, please consult the Avaya document IP Phones Fundamentals (NN43001-368).

Minimum PC Requirements

 Pentium® Pro 200 MHz  256 MB memory or higher  36 MB free hard-drive space (all languages)  USB port  Monitor settings: 16 bit High Colour; 800x600 resolution or higher

Supported Operating Systems

 Windows XP SP3  Windows Vista SP2 (32-bit)

 Windows 7 (32-bit)

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Note: Please ensure that you have the latest version of the 2050 IP Softphone. Earlier versions may not support the BCM HTTP Server licensing method.

USB Audio Kit

Operation of the 2050 IP Softphone requires the use of the Avaya USB Audio Kit or a Bluetooth headset (Bluetooth Power Class 2 profiles). The USB Audio Kit provides a high quality predictable audio interface, which is highly optimised for telephony applications. The USB Audio kit allows the 2050 IP Softphone to have an absolute and predictable loss and level plan implementation, which is necessary to meet TIA-810, FCC part 68 and its international equivalents as well as the ADA requirements for the hearing impaired. With the USB Audio kit, the 2050 IP Softphone can achieve performance rivalling or surpassing that of hardware telephones.

The USB Audio Kit is fully compliant with version 1.1 of the USB Device Specification and Windows Plug & Play specifications. It is fully compatible with suspend and resume functions for effective use in battery operated laptops.

Installing the 2050 IP Softphone

1. Insert the 2050 IP Softphone CD into the CD-ROM drive of your

computer. The install wizard starts.

2. Alternatively download the 2050 IP Softphone from www.avaya.com

and run the install/setup file.

3. The Choose Setup Language selection box will be displayed. From

the dropdown list select required language and click OK.

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4. Once the files have loaded the Install wizard screen will appear, click

Next.

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5. Once you have read the licence agreement select the I accept the

terms in the licence agreement button. Click Next.

6. The next screen displays the default file location; though it is possible

to change the location if required by clicking on the Browse button. Click Next.

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7. You can now select or deselect the languages to be installed that can

be chosen when using the i2050. Make your selections and click Next.

8. Choose which Start Menu folder location you would like to launch to

2050 IP Softphone form, or accept the default location. Click Next.

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9. Select which shortcuts you require for the 2050, and click Next.

10. Once all of the options needed to install have been selected, the

Ready to Install screen will appear. Click Install.

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11. On completion click Install.

12. After a few moments you will be prompted to select a language for the

i2050 prompts and dialogs. The selectable options relate to the languages selected/deselected earlier. Choose a language and click Next.

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13. Choose a theme for the main interface and click Next

14. Setup is now complete. Click Finish.

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15. Once installation is complete, you will need to run the 2050 IP

Softphone Settings utility to assign a server address and to configure audio peripherals. See the Configuring the 2050 IP Softphone section of this guide.

16. If you have been supplied with the USB Audio Kit, plug that into the

USB port of your PC/laptop now. Once it has been connected, you can select it as your audio device for 2050 usage.

Configuring the 2050 IP Softphone

Use the following procedure to configure the 2050 IP Softphone to connect to the BCM.

1. On the Computer, click the Start button and then select Programs\

Avaya\2050 IP Softphone, and click on 2050 IP Softphone settings.

Or, if the IP Softphone has already been launched, click on the Avaya logo, open the File menu and select Settings…

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2. The IP Software Phone 2050 Settings utility will now be launched.

3. Click on the Server option to configure how the Softphone will connect

to the BCM:

a. If your site uses DHCP: Select the Automatic (DHCP) option.

Using DHCP is the default method of locating the call server. If DHCP is used, no further configuration is required.

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Attribute

Description

Headset or Handset Microphone

Select the microphone used for making calls. Select USB Audio Device.

Headset or Handset Speaker

Select the speaker used for making calls. Select USB Audio Device.

Handsfree/ Paging/Ringing Microphone

This is the microphone which is used when the handsfree device is selected in the interface. This selection normally should match the Avaya USB Audio Kit which enumerates as a USB Audio Device

Handsfree/ Paging/Ringing Speaker

This is the speaker which is used when the handset free device is selected in the interface. It is also the speaker which is used to play ring tone and the device pages are directed. This selection normally should match the PC's speakers. This allows ring tone and pages to be heard over the PC speakers rather than on the headset

b. If you want to specify the server location manually, clear the

Automatic (DHCP) option. Select the Server type you wish to configure: Primary, Secondary or Application.

i. Enter the IP address of the server.

ii. Enter the Name of the server. c. Select the Server Type as BCM d. Ensure the Port number = 7000

4. Enter the number of Retries. If the initial connection fails, the 2050 will attempt to re-connect the number of times indicated by Retries,

5. Then select either the OK or Apply button to confirm the configuration.

6. Select the tab for Sound Devices, and make sure the Microphone/Speaker fields are configured for the USB headset kit (if using). Then select either the OK or Apply button to confirm the configuration.

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Attribute

Description

Version

Shows the version of the USB Headset Adapter. Note: If the USB Headset Adapter is not recognized or has a version number lower than 2.0 the other features in this table are greyed out and unavailable.

Headset Type

Select the type of headset that you have connected to the USB Headset Adapter. Due to differences in headset construction, you may not get optimal audio performance when using a headset that does not appear on the list. For optimal performance, always use one of the headsets that appears on the Headset Type drop list.

Manual Override

Select one of the available cadences to enable the Manual Override feature. When Manual Override is enabled, you can manually turn on the external lamp using the 2050 IP Softphone Smart Functions button on the USB Headset Adapter. For more information about the 2050 IP Softphone Smart Functions button, refer to the 2050 IP Softphone Help. Select None to disable the Manual Override feature.

Headset Disconnect,

Select one of the available cadences if you want the external lamp to indicate when the headset is disconnected from the USB Headset Adapter. Select None if you do not want the external lamp to indicate when the headset is disconnected.

Active Call

Select one of the available cadences if you want the external lamp to indicate when there is an active call on the IP Softphone 2050. If the USB Headset Adapter is selected as the Ringing Speaker, the external lamp also indicates when there is a call ringing on the IP Softphone 2050. Select None if you do not want the external lamp to indicate when there is an active call. Note: If you select a cadence for Active Call, the external lamp also turns on or flashes when another application uses the audio channel for the USB Headset Adapter.

Message Waiting

Select one of the available cadences if you want the external lamp to indicate when the 2050 IP Softphone message waiting light is on. The 2050 IP Softphone message waiting light normally indicates when there is a message waiting. However, most systems also turn on or flash the message waiting light when the 2050 IP Softphone is ringing.

7. Further details regarding USB headset configuration can be viewed from the USB Headset link.

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Attribute

Description

Select None if you do not want the external lamp to indicate when the message waiting light is on. Use backlight Select this check box to enable the backlight for the USB Headset Adapter buttons. Clear this check box to disable the backlight for the USB Headset Adapter buttons. Note: When you enable the backlight, you can use the state of the backlight to quickly determine if the 2050 IP Softphone is running. When the backlight is on, the 2050 IP Softphone is running. When the backlight is off, the 2050 IP Softphone is not running.

Configure Smart Functions

Click this button to set the options that are available when you press the Smart Functions button on the USB Headset Adapter.

Note: The External Lamp is an optional component. It normally is not included with the USB Headset Adapter, and must be ordered separately. The external lamp also is known as an “In-Use Indicator” lamp.

8. The 2050 IP Softphone will now require a licence. Refer to either the Licensing the i2050 Using the BCM HTTP Server Method section of this guide, or the Avaya document IP Phones Fundamentals (NN43001-368) for Licensing Server or Node-Locked licensing methods, dependant on which method is available for your system.

Licensing the i2050 Using the BCM HTTP Server Method

This method requires licenses to be generated on a per i2050 installation basis.

1. In the 2050 IP Softphone Settings window, click on the Hardware ID option, and make a note of the ID displayed. Send this to your keycode supplier and request the licensing files.

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2. Once you have obtained the licensing files they will need to be uploaded to the BCM. In the Element Manager Configuration tab, navigate to Resources, Telephony Resources, and click on IP Sets.

3. In the IP Terminal Global Settings tab of the Details area, click on the Upload button at the bottom of the screen.

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4. In the Import files window, click on Browse to locate the licensing files obtained from your keycode supplier.

5. Select all the i2050 licensing files obtained from your keycode supplier, and click on Select files.

6. Click OK to upload the files.

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7. The files will be displayed in the IP clients configuration files area.

8. The i2050 now needs to be configured to search for the files on the BCM. In the 2050 IP Softphone Settings window, click on the Server option.

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9. In the Provisioning Server field ensure the Protocol is set to HTTP and that the URL field contains the location of the BCM. Click OK when complete.

10. Continue with the Registering the 2050 IP Softphone section of this guide.

Registering the 2050 IP Softphone

Use the following procedure to register your 2050 IP Softphone with the BCM.

1. Start the 2050 IP Softphone. The i2050 will attempt to find the licensing information from the configured location. If licensing is successful, the registration process can continue.

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2. If a password prompt appears on the 2050 IP Softphone display, enter the registration password (i.e. the Global Password described in the Preparing your system for IP telephone registration section of this guide) and press the OK soft key. You will need to use the dialpad on the Softphone to enter the password.

3. Alternatively, you may be prompted to enter the following information:

a. Registration: SETNNA = 738662 b. Password: CONFIG = 266344

4. If a DN prompt appears on the 2050 IP Softphone display, enter the DN you want assigned to this telephone, and press the OK soft key. Otherwise, Auto-assign DNs will have been enabled in Element Manager, and therefore the DN will automatically be applied.

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5. After the registration is complete, you do not need to go through the registration steps described above, unless you deregister the terminal.

Note: The 2050 IP Softphone Telephone can be configured as a standard Digital handset. With this in mind, Lines and/or Line pool access require configuration. For more information on these settings, please refer to the Telephony Services Guide.

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Using the 2050 IP Softphone

The default presentation of the 2050 Software phone is operational. In this mode the user can operate most features available from the 1140e IP Telephone.

Calls can be answered or made by pressing the green headset button. In this mode the call server will select the line to answer or engage. The user can also hang-up, hold, retrieve from hold, mute, adjust volume and access network services such as voice mail.

The Number Pad provides a graphic keypad to dial numbers with a mouse. A number can also be dialed by using the computer keyboard.

The display shows up to six line or feature keys provisioned for the set by the BCM. The status of each line key on the display is illustrated by a graphic icon (idle, ringing, connected, etc.). The line is labeled based on its BCM provisioning information.

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Below is a diagram showing the key components of the i2050 interface.

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Additional options allow access to other features and functions.

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The i2050 can also function in the System Tray of the Windows desktop.

Software Expansion Module

The 2050 IP Softphone provides a Software Expansion Module in case extra feature or autodial buttons are required. An extra 54 buttons are available, and can be configured by using the usual button programming features via the interface, or under Element Manager programming (Telephony, Sets, Active Sets, Capabilities and Preferences tab, CAP/KIM Button Programming tab).

To monitor lines, the Software Expansion Module should be configured as a CAP Assignment in Element Manager under Telephony, Global Settings, CAP Assignment.

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To display the Software Expansion Module, click on the logo, navigate to View, and then select Expansion Module.

The Software Expansion Module will load, and display the buttons‟

functions/features as programmed. Use the scroll bar to view and locate all the buttons.

Incoming Call and Disconnect Popup

Calls ringing on the 2050 IP Softphone now generate a popup window containing basic call information and call control options. This feature is especially useful if the i2050 is minimised or operational in the System Tray.

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The Popup window behaviour is determined in the Notifications area of the 2050 IP Softphone Settings options.

Incoming calls generate a call popup window in the lower right corner of youir windows desktop. The call can be answered from the popup window, or the i2050 interface can be launched by clicking on the Open button.

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When the call is ended by either party, the Call Disconnected display will be shown. Again, this popup window will display if the i2050 is minimised or operating tin the Windows System Tray.

IP Terminal Features

The IP telephony sets and the 2050 IP Softphone can access the same telephone features available on standard TDM sets, with the exception of Voice Call.

In addition, the IP telephones have three additional IP-specific features:

 Feature List: allows specification of the features that appear in the

Features List on the IP phones.

 Key Labels: this feature allows labels for programmed buttons on the IP

phones to be specified.

 Hot Desking: a user can use assume control of an IP phone in a

different location as if they were using their own phone at their usual workplace.

Feature List

You can add and modify the features that display on the IP telephone feature list, which is accessed through the Services button or by using FEATURE *900.

The Feature Codes Guide provides a complete list of BCM Features and index codes.

1. In the Element Manager, open the Configuration tab, followed by Telephony, then Global Settings, and click on IP Terminal Features.

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2. Click on the Features list tab. This will now display the list of features already configured (12 features are assigned as default).

3. If you want to add a new feature to the list, click Add. Enter the name of the feature and the associated feature code.

4. Feature codes can be deleted from the list, or the order changed by selecting the feature and clicking Up or Down.

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Feature List IP Set Usage

The Feature List settings will appear on the handset.

1. On the IP handset, enter FEATURE *900 or press the Services button.

2. Use the Page+ and Page- display keys to scroll to the feature you want.

3. Use the navigation keys to move through the selections on the menu, and when having made the choice, press Select.

Key Labels

This feature enables the labeling of buttons programmed on the IP phones. For example, if you have a button programmed to F904, the button on the display can be labeled as CC Login, CC In/Out etc.

1. In the Element Manager, open the Configuration tab, followed by Telephony, then Global Settings, and click on IP Terminal Features.

2. Click on the Key Labels tab.

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3. Double-click in the Key Label field that you want to re-label, and enter a new description. Press the tab key to accept the change.

Hot Desking

The Hot Desking feature allows a user to divert calls and signals from one IP telephone to another. For example, if a user is temporarily working in another office, they can retain their telephone number by hot desking their usual telephone to the IP telephone in their temporary office.

Hot desking can be accessed using FEATURE *999 on the telephone to which the traffic will be diverted. The user can also evoke this feature from the Services key menu, where it is defaulted as the first item on the list.

Hot desking must be allowed on the originating telephone and you need to specify a password. These settings are found under the ADMIN key within the hot desking feature. Hot desking is invoked through the DIVERT key within the hot desking feature.

If the originating telephone does not have hot desking allowed, the user will receive a Not Allowed prompt, indicating that the telephone is not available for hot desking. This prompt also occurs if the originating telephone is on a call when the diversion command was issued.

Once hot desking occurs between two IP telephones, no activity is allowed on the originating telephone, except to cancel hot desking. The display on the originating telephone indicates where it has been diverted. On the diversion telephone, the key displays will reflect the displays from the originating telephone.

Call forwarding to voice mail continues as normal. Voice mail can be accessed from the active IP telephone, as if it were the originating telephone.

When hot desking is cancelled, this can be performed from either telephone, the displays for each telephone return to normal. If you forget the password, hot desking can only be cancelled from the originating set.

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Note: When you cancel hot desking, ensure that the telephone is on-hook. If you have just hung up, wait 10 seconds before attempting to cancel hot desking.

Use the following procedure to set up a password and activate the feature on the originating IP handset:

1. Enter FEATURE *999.

2. Press ADMIN.

3. Enter a new password, or change an existing password, and press OK.

4. Confirm the password, and press OK.

5. Allow/disallow hot desking, as required by pressing CHANGE.

6. Press QUIT to exit.

Using hot desking:

1. At the telephone you will be using to answer diverted calls, enter FEATURE *999 or access the hot desking feature by pressing the services key and selecting from the feature display list.

2. Press the soft key under the displayed DIVERT.

3. Enter the DN (extension number) of the telephone you want to divert to this telephone.

4. Enter the password of the diverted telephone.

The buttons on your telephone will mimic the buttons on the diverted set. The diverted telephone indicates that it has been diverted, and it cannot be used until hot desking is cancelled.

Cancel hot desking

You can cancel hot desking from either telephone. Ensure that the telephone is on-hook before canceling hot desking.

From the diverted telephone, press the soft key under the display of a CANCEL prompt.

OR, on the live telephone:

1. Access FEATURE *999 or access the hot desking feature by pressing the services key and selecting from the feature display list.

2. Enter the password of the diverted telephone.

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3. Press the soft key under the display of a CANCEL prompt.

Keeping Call Forward Settings when IP Phones are Disconnected

IP Phones can easily be relocated from one place to another. This will involve them being disconnected from the BCM. Similarly, the 2050 IP Softphone will be disconnected from the BCM due to its host PC/laptop being rebooted or shutdown.

The Keep DN Alive feature allows any configured call forward rules to apply, even when the set/Softphone is disconnected. This means that calls can still be routed to voicemail even when the IP DN is disconnected.

Use the following procedure to configure set the Keep DN Alive feature.

1. In Element Manager Configuration tab, navigate to Telephony, Sets, Active Sets and select the IP phone you want to configure.

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2. Select the Capabilities and Preferences tab, followed by the IP Terminals tab in the details section.

3. Select or de-select the Keep DN Alive checkbox to enable or disable this feature. Enabling this feature will ensure that Call Forward rules will still apply, even when the IP phone is disconnected from the BCM.

4. It is also possible to reset the Hotdesking password, force a firmware download, and deregister the DN from this area.

5. Codecs can also be specified on an individual DN basis, overriding the general IP phone codec settings for specific situations.

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VoIP Gateways

With a VoIP trunk, you can establish communications between a BCM and a remote system across an IP network.

The BCM system supports SIP and H.323 trunk protocols. Both types of trunks support connections to other BCMs, a central call server such as Succession 1000/M, and trunk-based applications. SIP trunks support connections to ITSPs for enhanced call routing capability.

SIP trunks and H.323 trunks are assigned to a single Pool, and the routing decision to route calls via H.323 or SIP is made based on the routing modes of the two services (Direct/Gatekeeper/Proxy) and the combined routing table.

If the BCM will only use one of the trunk protocols then only configure the associated tabs, i.e. if the BCM will only utilise H.323 then the SIP-specific settings do not require configuration.

Configuring the Local Gateway Settings

The VoIP trunk access point at each system is called a gateway. The gateway to your system, the local gateway, determines how incoming and outgoing calls will be handled.

The local gateway parameters define how the BCM allows call signalling information to be directed through VoIP trunks. Call signalling establishes and disconnects the calls.

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Field

Value

Description

Forward redirected OLI

If you select the check box, the OLI of an internal telephone is

forwarded over the VoIP trunk when a call is transferred to an external number over the private VoIP network. If not selected, the system forwards only the CLID of the transferred call.

Remote capability MWI

If you select the check box, the system sends the telephone name without going calls to the network.

Send name display

This setting must coordinate with the functionality of the remote system that hosts the remote voice mail.

Ignore inband DTMF in RTP

If you select the check box, the BCM ignores audible in-band DTMF tones received over VoIP trunks after the BCM connects to the remote end of a locally hosted contact center application or to a locally hosted CallPilot application, such as auto attendant, voice mail, or IVR.

IP Trunks

These are general settings that relate to both H.323 and SIP trunks.

1. Open the Resources folder and followed by the IP Trunks folder. Click the General option and then select the IP Trunks Settings tab.

2. Click on IP Trunk Settings and configure as required.

IP Trunks Settings

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Field

Value

Description

Telephony Settings

Fallback to circuitswitched

Enabled-All Enabled-TDM Disabled

Your choice determines how the system will handle calls if the IP network cannot be used.

 Enabled-All: All calls are rerouted over specified PSTN

trunks lines.

 Enabled-TDM: All TDM (digital telephones) voice calls will

be rerouted over specified PSTN trunks lines.

 Disabled: Calls will not be rerouted.

Default: Enabled-All

Note: Enabled-TDM-only enables fallback for calls originating on digital telephones. This is useful if your IP telephones are connected remotely, on the public side of the BCM network, because PSTN fallback is unlikely to result in better quality of service in that scenario

MCDN protocol

None SL1 CSE

Both these protocols require a keycode. SL1: use this protocol only for BCM 2.5 systems CSE: Use this protocol for BCM 3.0 and newer systems. This protocol supports Meridian 1 IPT. Otherwise, use None.

Gatekeeper digits

<0-9>

If dialed digits match gatekeeper digits, the call is routed via H323 protocol. If the digits do not match, the call is routed via SIP protocol.

Gatekeeper

If selected, all dialed digits match gatekeeper digits and VoIP calls

H.323 Settings

1. Open the Resources folder and followed by the IP Trunks folder. Click the H323 Trunking option.

2. Select the Settings tab and configure the H323 Settings as required.

H323 Settings

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Field

Value

Description

wildcard

will be routed through the gatekeeper

Normal Route Fallback To

None Prime set

Select None or Prime set. If Prime set is selected and the outgoing IP trunk leg of the call in a tandem scenario cannot be completed, the call will terminate on the prime set for the line. Default: None

Configuration (click on the Modify button to configure)

*Call signalling

Direct Gatekeeper Resolved Gatekeeper Routed Gatekeeper Routed no RAS

Direct: call signalling information is passed directly between endpoints. The remote gateway table in the Element Manager defines a destination code (digits) for each remote system to direct the calls for that system to route. In each system, the IP Terminals and H.323 Terminals records map IP addresses to specific telephones. Gatekeeper Resolved: all call signalling occurs directly between H.323 endpoints. This means that the gatekeeper resolves the phone numbers into IP addresses, but the gatekeeper is not involved in call signalling. Gatekeeper Routed: uses a gatekeeper for call setup and control. In this method, call signalling is directed through the gatekeeper. Gatekeeper Routed no RAS: Use this setting for a NetCentrex gatekeeper. With this setting, the system routes all calls through the gatekeeper but does not use any of the gatekeeper Registration and Admission Services (RAS).

Enable H245 tunnelling

If Enabled, the VoIP Gateway tunnels H.245 messages within H.225. The VoIP Gateway service must be restarted for a change to take effect. Default: Disabled.

Primary Gatekeeper IP

If Gatekeeper Routed, Gatekeeper Resolved or Gatekeeper Routed no RAS are selected under Call Signalling, type the IP address of the machine that is running the gatekeeper.

Backup Gatekeeper(s)

<IP address>,

NetCentrex gatekeeper does not support RAS, therefore, any backup gatekeepers must be entered in this field. Note: Gatekeepers that use RAS can provide a list of backup gatekeepers for the end point to use in the event of the primary gatekeeper failure.

If Gatekeeper Routed, Gatekeeper Resolved, or Gatekeeper Routed no RAS are selected under Call Signaling, enter one or more alias names for the gateway

Alias Names

Alias names are comma delimited, and may be one of the following types: E.164 - numeric identifier containing a digit in the range 0-9. Identified by the keyword TEL: Example: the BCM is assigned an E.164 and an H323 Identifier: Alias Names: TEL:76, NAME:bcm10.avaya.com

NPI-TON - also referred to as a PartyNumber alias. Similar to E164 except that the keyword indicates the NPI (numbering plan identification), as well as the TON (type of number). Identified by one of the following keywords: PUB (Public Unknown Number); PRI (Private Unknown Number); UDP (Private Level 1 Regional Number (UDP)); CDP (Private Local Number (CDP)).

H.323Identifier - alphanumeric strings representing names, e-mail addresses, etc. Identified by the keyword NAME:

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Field

Value

Description

Example: The BCM is assigned a public dialed number prefix of 76, a private CDP number of 45, and an H323 Identifier alias: Alias Names: PUB:76, CDP:45, NAME:bcm10.avaya.com

H.225 (Q.931) CallingPartyNumber (NetCentrex gatekeeper) - The NetCentrex gatekeeper uses the H.225(Q.931) CallingPartyNumber to resolve the call originator for billing purposes. This number must then contain a unique prefix, or location code that is unique across all endpoints that are using the NetCentrex gatekeeper. Identified by the keyword src:. Example for private networks: CDP alias = src:<DN>; UDP alias = src:<LOC><DN>. Example for public network: src:<public OLI>

Note: E164 or NPI-TON alias types are commonly used since they fit into dialling plans. A BCM alias list should not mix these types. Also, the type of alias used should be consistent with the dialling plan configuration. Use the same alias naming method on all BCMs within a network.

Configuration note:

Network note: If your private network contains a Meridian 1-IPT, you cannot use Radvision for a gatekeeper.

Modify Call Signaling Settings

Call signaling port

0-65535

Default: 1720 This field allows you to set non-standard call signaling port for VoIP applications that require special ports. 0 = The first available port is used. Ensure that you do not select a port that has been assigned elsewhere in the BCM. To ensure the port is not in use, run netstat-a from the command line.

RAS port

0-65535

Default: 0 This field allows you to set a non-standard Registration and Admission (RAS) port for VoIP applications that require special ports. 0 = The first available port is used. Ensure that you do not select a port that has been assigned elsewhere in the BCM. To ensure the port is not in use, run netstat-a from the command line.

Registration TTL (s)

Default: 60 seconds

This TimeToLive parameter specifies the intervals when the VoIP gateway sends KeepAlive signals to the gatekeeper. The gatekeeper can override this timer and send its own TimeToLive period.

Gatekeeper TTL (s)

The actual time used by the gatekeeper for the registration process. Status

<read-only>

Indicates if the device is online.

Modify

Click to modify the parameters. Note: All active H.323 calls are dropped if these settings are changed.

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Field

Value

Description

Telephony Settings

Fallback to circuitswitched

Enabled-All Enabled-TDM Disabled

Your choice determines how the system will handle calls if the IP network cannot be used. Enabled-All: All calls will be rerouted over specified PSTN trunks lines. Enabled-TDM: All TDM (digital telephones) voice calls will be rerouted over specified PSTN trunks lines. Disabled: Calls will not be rerouted.

Default: Enabled-All

Dynamic Payload

96 - 127 Default: 120

Set to 0 to disable RFC2833 functionality.

SIP Settings

Local Domain

Local domain of the SIP network.

Call signaling port

The listening port for the BCM. Note: FEPS (Functional Endpoint Proxy Server) must be restarted if this value is changed. Default: 5060 . Select Modify to change the Call Signalling Port

SIP Settings

1. Open the Resources folder and followed by the IP Trunks folder. Click the SIP Trunking option.

2. Click on the Global Settings tab and configure the SIP Settings as required.

SIP Settings

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Field

Value

Description

RTP Keep Alives

Scope

None RTP RTP-RTCP

This setting should be used if the BCM is behind a NAT Router. The available options are: None: RTP keep-alives are disabled. RTP: If selected, keep-alive parameters are displayed. If initial keep-alives are enabled, the BCM will send an RTP packet when a dialog is established. RTP-RTCP: If selected, keep-alive parameters are displayed. If initial keep-alives are enabled, the BCM will send an RTP packet and an RTCP packet when a dialog is established.

Status

<read-only>

Indicates the status of the gateway.

H323 & SIP Media Parameters

The H323 and SIP Media Parameters tabs determine a number of local system settings. These values need to be coordinated with the other systems on the network to ensure that all features work consistently across the network. Media parameters include setting:

• The order of preferred codecs

• Voice activity detection

• Jitter buffer size

• Codec payload size

• IP fax transmission availability on the network

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Field

Value

Description

Preferred Codecs

None G.711-uLaw G.711-aLaw G.729 G.723

Select the Codecs in the order in which you want the system to attempt to use them. Performance note: Codecs on all networked BCMs must be consistent to ensure that interacting features such as Transfer and Conference work correctly. Systems running BCM 3.5 or newer software allow codec negotiation and renegotiation to accommodate inconsistencies in Codec settings over VoIP trunks.

Settings

H323 Media Parameters

The H323 Media Parameters tab controls codec settings for H323 trunks.

1. Open the Resources folder and followed by the IP Trunks folder. Click the H323 Trunking option. Select the Media Parameters tab

2. Configure the H323 Settings as required.

H323 Media Parameters

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Field

Value

Description

Enable Voice Activity Detection

Voice activity detection, also known as silence suppression identifies periods of silence in a conversation, and stops sending IP speech packets during those periods. In a typical telephone conversation, most of the conversation is half-duplex, meaning that one person is speaking while the other is listening. If voice activity detection is enabled, no voice packets are sent from the listener end. This greatly reduces bandwidth requirements. G.723.1 and G.729 support voice activity detection. G.711 does not support voice activity detection. Performance note: Voice activity detection on all networked BCMs and IPT systems (VAD setting on IPT systems) must be consistent to ensure that interacting features such as Transfer and Conference work correctly. As well, the Payload size on the IPT must be set to 30ms. Default: Disabled

Jitter buffer

Auto None Small Medium Large

Select the size of jitter buffer you want to allow for your system. Default: Auto

G.729 payload size (ms)

10, 20, 30, 40, 50, 60 Default: 30

Set the maximum required payload size, per codec, for the VoIP calls sent over H.323 trunks. Note: Payload size can also be set for IP telephones.

G.723 payload size (ms)

G.711 payload size (ms

10, 20, 30, 40, 50, 60 Default: 30

Incremental payload size

When enabled, the system advertises a variable payload size (40, 30, 20, 10 ms)

Enable T.38 fax

Enabled: The system supports T.38 fax over IP. Disabled: The system does not support T.38 fax over IP

Caution: Operations note: Fax tones that broadcast through a telephone speaker may disrupt calls at other telephones using VoIP trunks in the vicinity of the fax machine. Here are some suggestions to minimize the possibility of your VoIP calls being dropped because of fax tone interference: Locate fax machine away from other telephones. Turn the speaker volume on the fax machine to the lowest level, or off, if that option is available.

Force G.711 for 3.1k Audio

When enabled, the system forces the VoIP trunk to use the G.711 codec for 3.1k audio signals such as modem or TTY machines. Note: This setting can also be used for fax machines if T.38 fax is not enabled on the trunk

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Field

Value

Description

Preferred Codecs

None G.711uLaw G.711aLaw G.729 G.723

Select the Codecs in the order in which you want the system to attempt to use them. Performance note: Codecs on all networked BCMs should be consistent to ensure that interacting features such as Transfer and Conference work correctly. Note: The G.723 protocol can be used between IP endpoints

Field

Value

Description

Settings

SIP Media Parameters

SIP trunks are administered separately from H.323 trunks. It is common for H.323 and SIP trunks to both exist on the same system; however, each has different network segments.

1. Open the Resources folder and followed by the IP Trunks folder. Click the SIP Trunking option. Select the Media Parameters tab.

2. Configure the SIP Settings as required.

SIP Media Parameters Settings

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Enable Voice Activity Detection

The voice activity detection (silence suppression) identifies periods of silence in a conversation, and stops sending IP speech packets during those periods. In a typical telephone conversation, most of the conversation is half-duplex, meaning that one person is speaking while the other is listening. Voice activity detection is enabled, no voice packets are sent from the listener end. This greatly reduces bandwidth requirements. G.723.1 and G.729 support silence suppression. G.711 does not support silence suppression. Performance note: voice activity detection on all networked BCMs and IPT systems (VAD setting on IPT systems) must be consistent to ensure that interacting features such as Transfer and Conference work correctly. Default: Disabled

Jitter Buffer

Auto None Small Medium Large

Select the size of jitter buffer you want to allow for your system. G.729 Payload Size (ms)

10, 20, 30, 40, 50, 60

Set the desired payload size, per codec, for the VoIP calls sent over SIP trunks. Note: Payload size can also be set for IP telephones.

G.723 Payload Size (ms)

G.711 Payload Size (ms)

10, 20, 30, 40, 50, 60 Default: 30

Fax Transport

<drop down list> T.38 G.711 Default: T.38

T.38: The system exclusively supports T.38 fax over IP. G.711: The system exclusively supports G.711 fax over IP.

Force G.711 for 3.1k Audio

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Provide inband ringback

This setting affects in-bound SIP trunk calls. If you select the check box, the BCM attempts to stream ringback, tones, or announcements in-band to the caller using RTP. This setting results in in-band ringback. It can be useful in tandem scenarios to transfer DTMF if the final leg in the tandem connects to an IVR that plays announcements before connecting the call. Attention: Fax tones that broadcast through a telephone speaker may disrupt calls at other telephones using VoIP trunks in the vicinity of the fax machine. Here are some suggestions to minimize the possibility of your VoIP calls being dropped because of fax tone interference: Locate the fax machine away from other telephones. Turn the speaker volume on the fax machine to the lowest level, or off, if that option is available.

Private SIP Specific Configuration

The following sections relate specifically to SIP configuration over private domains.

SIP Proxy

Allows the routing of calls through a configured SIP Proxy. The SIP Proxy‟s domain and Outbound Proxy Tables can be configured as outlined below.

1. Open the Resources folder and followed by the IP Trunks folder. Click the SIP Trunking option. Select the Private tab.

2. Select the Proxy tab and configure the Private SIP Proxy Settings as required.

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Private SIP Proxy Settings

Field

Value

Description

SIP Proxy

Domain

The name of the SIP Domain. This attribute is mandatory

Route all calls using proxy

<check box> Default: unchecked

If unchecked, the system first checks the routing table before routing all SIP calls. If checked, the system uses the SIP Proxy for all SIP calls.

MCDN Protocol

None CSE Default: None

Use CSE for interop with other devices (BCM or CS1K).

Optional IP Address for legacy routing

IP Address

Format 0.0.0.0 <7-24>

This attribute is optional. The system uses the IP Address and Port to route the message if the Outbound Proxy is not configured. The IP Address and Port are used in message headers. If supplied, the IP Address is used in the "maddr=" section of message headers The system uses these attributes for interop with NRS.

Port

<numeric> Default: 0

This attribute is optional. If the port is 0, the system uses the well-known SIP port 5060. Otherwise, the system uses the port you enter here.

Outbound Proxy Table

Name

The Name must be unique. If the name you enter is a Fully Qualified Domain Name, DNS resolves the address and the IP address can be left empty.

IP Address

Format 0.0.0.0 <7-24>

If you specify the IP Address, this address is used directly (the system does not use the Name attribute and does not invoke DNS). If you leave this attribute empty, the system uses the Name attribute.

Port

<0-65535> Default: 0

If you leave Port as 0, the system uses the well-known SIP port

5060. Otherwise, the system uses the Port number you specify here.

IP Telephony

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Loadbalancing Weight

<0-10> Default: 1

Enter the load-balancing weight. The system uses this attribute to distribute calls among the outbound proxies.

Keep alive

None OPTIONS Default: None

This attribute helps the system determine if an Outbound proxy device is responding. If you select None, the system does not ping the device, assuming the device is always active. If you select OPTIONS, the system sends a periodic OPTIONS message to the Outbound Proxy. If the proxy fails to respond, the system skips over it until it responds again

SIP Domain Names

Value

Description

e.164 / National

national.e164

String to use in phone context to identify numbering plan type

e.164 / Subscriber

subscriber.e164

String to use in phone context to identify numbering plan type

e.164 / Unknown

unknown.e164

String to use in phone context to identify numbering plan type

SIP URI Map

Use the SIP URI Map to configure the sub-domain name associated with each SIP URI (Session Initiated Protocol Uniform Resource Identifier). These strings must be coordinated with the other nodes in the network.

These fields correspond to Public Network, Private Network, and Routing settings of the Configuration > Telephony > Dialing Plan section of Element Manager.

1. Open the Resources folder and followed by the IP Trunks folder. Click the SIP Trunking option. Select the Private tab.

2. Click on the URI Map tab and configure the URI Map settings as required.

Private SIP URI Map Settings

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SIP Domain Names

Value

Description

e.164 / Special

special.e164

String to use in phone context to identify numbering plan type

Private / UDP

UDP

String to use in phone context to identify numbering plan type

Private / CDP

CDP

String to use in phone context to identify numbering plan type

Private / Special

special.private

String to use in phone context to identify numbering plan type

Private / Unknown

unknown.private

String to use in phone context to identify numbering plan type

Private / Subscriber

Subscriber.private

String to use in phone context to identify numbering plan type

Unknown / Unknown

unknown

String to use in phone context to identify numbering plan type

SIP Authentication

These settings ensure that only the gateways that have been authenticated i.e have valid credentials, can place calls to the BCM. If challenged, the BCM can also provide its own valid credentials on outgoing calls.

1. Open the Resources folder and followed by the IP Trunks folder. Click the SIP Trunking option. Select the Private tab.

2. Click on the Authentication tab and configure the SIP Authentication settings as required.

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Field

Value

Description

User Accounts

Description

An optional description of the user account.

Domain

Remote domain name of the service. Can be either FQDN or an IP address.

Parent

If selected, indicates that the user account is a parent account. Child accounts are mapped to individual sets.

CLID

If the account is a parent account, this field is empty. If it is a child account, you can enter CLID information to be displayed for this account in this field.

SIP Username

Provided to the administrator from the service provider.

Auth Username

The authentication username used in authentication challenges. This parameter is provided by the SIP service provider. The authentication username can be different than the SIP username.

Auth Password

The authentication password.

CLID Override

Overrides the Caller ID parameter for the account. If not configured, the Caller ID of the account is used.

Display name Override

Overrides the Display Name in From Header parameter for the account. If not configured,

Private SIP Authentication Settings

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Field

Value

Description

the Display Name in From Header of the account is used.

PAI CLID Override

Overrides the Caller ID in P-AssertedIdentity parameter for the account. If not configured, the PAI CLID of the account is used.

PAI Display name Override

Overrides the Display Name in PAI parameter for the account. If not configured, the PAI Display name of the account is used.

Contact Override

Used in cases where the SIP trunking service provider constructs R-URI for outgoing calls based on user part of contact header in SIP registration requests. Since RUri in incoming SIP trunk calls is used to determine received digits to match them to target lines, this parameter can be useful to control received digits for incoming calls.

Maddr in Contact

Select the check box to include maddr in contact for this account. When selected, this overrides the System Wide settings for Maddr in the Private SIP settings tab.

Local Domain Override

This field overrides the system wide local SIP domain for outgoing calls associated with the SIP user account.

Registration

Local SIP Authentication

Local Authentication

<check box> Default: unchecked

Checked: The BCM authenticates all incoming calls. Unchecked: The system does not authenticate incoming calls.

Quality of Protection

Authentication only Authentication and Integrity Default: Authentication only

"Authentication only" results in authentication username/password encryption. "Authentication and Integrity" adds a whole message integrity check. Note: This option adds to security but may impact NAT/firewall integration.

401 Reason

<alphanumeric> Default: Unauthorized

This character string is sent out in authentication challenges.

Local Accounts

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Field

Value

Description

User Id

The administrator supplies each remote domain with a unique User ID/Password. If the local system challenges incoming calls, the remote system must provide the User ID/Password combination.

Password

The administrator supplies each remote domain with a unique User ID/Password. If the local system challenges incoming calls, the remote system must provide the User ID/Password combination.

Description

Description of remote domain.

Remote Account Fields

Realm

Remote domain name.

User ID

User ID and Password are supplied by remote domain. Local system responds with User ID/Password if outgoing call is challenged by remote domain.

Password

User ID and Password are supplied by remote domain. Local system responds with User ID/Password if outgoing call is challenged by remote domain.

Description

Description of remote domain.

SIP Trunk Settings

These are general settings that relate to Private SIP trunks.

1. Open the Resources folder and followed by the IP Trunks folder. Click the SIP Trunking option. Select the Private tab.

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Avaya IP Telephony BCM User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual