The information in this document is subject to change without notice. The statements, configurations, technical data, and
recommendations in this document are believed to be accurate and reliable, but are presented without express or implied
warranty. Users must take full responsibility for their applications of any products specified in this document. The
information in this document is proprietary to Nortel Networks.
Trademarks
Nortel, the Nortel logo, and the Globemark are trademarks of Nortel Networks.
Microsoft, MS, MS-DOS, Windows, and Windows NT are registered trademarks of Microsoft Corporation.
The Bluetooth trademark and logos are owned by the Bluetooth SIG, Inc. and any use of such marks by Nortel Networks is
under license. Other trademarks are those of their respective owners.
All other trademarks and registered trademarks are the property of their respective owners.
Refer to the following topics for general BCM information:
•“About BCM”
•“Symbols and conventions used in this guide” on page 33
•“Related publications” on page 34
•“How to get Help” on page 39
About this guide
The BCM 4.0 Networking Configuration Guide describes how to install, configure, and maintain
the BCM200, BCM400, and BCM1000 hardware running Business Communications Manager 4.0
(BCM) 4.0 software.
29
Purpose
The concepts, operations, and tasks described in this guide relate to the hardware of the BCM
system. This guide provides task-based information on how to configure a BCM network.
Use Element Manager, Startup Profile, and Telset Administration to configure various BCM
parameters.
In brief, the information in this guide explains:
•public and private networking
•configuring trunks and lines
•media gateways
•loops
•IP settings
•dialing plans
Audience
The BCM 4.0 Networking Configuration Guide is directed to installers responsible for installing,
configuring, and maintaining BCM systems.
To use this guide, you must:
•be an authorized BCM installer/administrator within your organization
•know basic Nortel BCM terminology
•be knowledgeable about telephony and IP networking technology
BCM 4.0 Networking Configuration Guide
30Chapter 1 Getting started with BCM
Acronyms
The following is a list of acronyms used in this guide.
Table 1 Acronyms (Sheet 1 of 2)
AcronymDescription
AHauthentication header
ARPaddress resolution protocol
ARSautomatic route selection
ASManalog station module
ATAanalog terminal adapter
BCMBusiness Communications Manager
BRIbasic rate interface
CbCCall-by-call
CoSClass of Service
CLIDcalling line identification
CLIRcalling line information restriction
CRCcyclic redundancy check
CSUChannel Service Unit
DHCPDynamic host configuration protocol
DIDdirect inward dial
DISAdirect inward system access
DLCIdata link connection identifier
DNSdomain name server
DTMFdual tone multi-frequency
FEMfiber expansion module
FoIPfax over IP
MCDNMeridian customer defined networking
MCIDmalicious call identification
MSCmedia services card
NATnetwork address translation
OLIoutgoing line identification
ONNoutgoing name and number
OSPFopen shortest path first
PFSperfect forward security
PPPoEpoint to point over Ethernet
QoSquality of service
RIProuting information protocol
SIPsession initiated protocol
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Table 1 Acronyms (Sheet 2 of 2)
AcronymDescription
SRGsurvivable remote gateway
TATtrunk anti-tromboning
TTLtime to live
UDPuniversal dialing plan
VADvoice activity detection
VLANvirtual LAN
VoIPvoice over IP
Organization
This guide is organized for easy access to information that explains the concepts, operations, and
procedures associated with the BCM system.
About BCM
Chapter 1 Getting started with BCM31
The BCM system provides private network and telephony management capability to small and
medium-sized businesses.
The BCM system:
•integrates voice and data capabilities, VoIP gateway functions, and QoS data-routing features
into a single telephony system
•enables you to create and provide telephony applications for use in a business environment
BCM key hardware elements
BCM includes the following key elements:
•BCM200 main unit
•BCM400 main unit
•BCM1000 main unit
•BCM expansion unit (compatible with BCM400 main unit)
•BCM400 expansion gateway
•BCM media bay modules (MBM):
— 4x16
— ASM8, ASM8+
— BRIM
—CTM4, CTM8
— DDIM
— DSM16+, DSM32+
BCM 4.0 Networking Configuration Guide
32Chapter 1 Getting started with BCM
—DTM
—FEM
— GASM
—GATM4, GATM8
BCM features
BCM supports the complete range of IP telephony features offered by existing BCM products:
Note: You enable the following features by entering the appropriate keycodes (no
additional hardware is required).
•VoIP Gateway: Up to 12 VoIP trunks
•VoIP Telephony Clients: Up to 64 VoIP Telephony clients, supporting the range of Nortel
IP Phones.
BCM applications
BCM supports many applications provided on the existing BCM platforms.
Note: You enable the following features by entering the appropriate keycodes (no
additional hardware is required).
•Voice Messaging for standard voice mail and auto-attendant features
•Unified Messaging providing integrated voice mail management between voice mail and
common e-mail applications
•Fax Suite providing support for attached analog fax devices
•Voice Networking features
•LAN CTE (computer telephony engine)
•VEWAN (Voice Enabled WAN)
•IVR (Integrated Voice Response)
•IP Music
•Intelligent Contact Center
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Chapter 1 Getting started with BCM33
Symbols and conventions used in this guide
These symbols are used to highlight critical information for the BCM system:
Caution: Alerts you to conditions where you can damage the equipment.
Danger: Alerts you to conditions where you can get an electrical shock.
Warning: Alerts you to conditions where you can cause the system to fail or work
improperly.
Note: Alerts you to important information.
Tip: Alerts you to additional information that can help you perform a task.
Security Note: Indicates a point of system security where a default should be
changed, or where the administrator needs to make a decision about the level of
!
security required for the system.
Warning: Alerts you to ground yourself with an antistatic grounding strap
before performing the maintenance procedure.
Warning: Alerts you to remove the BCM main unit and expansion unit power
cords from the ac outlet before performing any maintenance procedure.
BCM 4.0 Networking Configuration Guide
34Chapter 1 Getting started with BCM
The following conventions and symbols are used to represent the Business Series Terminal display
and dialpad.
ConventionExampleUsed for
Word in a special font (shown in
the top line of the display)
Underlined word in capital letters
(shown in the bottom line of a
two-line display telephone)
Dialpad buttonsButtons you press on the dialpad to select a
Pswd:
PLAY
Command line prompts on display telephones.
Display option. Available on two line display
telephones
option on the display to proceed.
particular option.
. Press the button directly below the
The following text conventions are used in this guide to indicate the information described:
ConventionDescription
bold Courier
text
Indicates command names and options and text that you must enter.
Example: Use the
Example: Enter
info command.
show ip {alerts|routes}.
italic textIndicates book titles.
plain Courier
text
FEATURE
HOLD
Indicates command syntax and system output (for example, prompts
and system messages).
Example:
Set Trap Monitor Filters
Indicates that you press the button with the coordinating icon on
whichever set you are using.
RELEASE
Related publications
This section provides a list of additional documents referred to in this guide. There are two types
of publications: Technical Documents on page 34 and User Guides on page 36.
Technical Documents
BCM 4.0 System Overview (N0060607)
System Installation
BCM 3.x to BCM 4.0 Upgrade Guide (N0060597)
BCM 4.0 Installation Checklist and Quick Start Guide (N0060602)
BCM1000 BCM 3.7 Installation and Maintenance Guide (N0008587 01)
BCM 4.0 for BCM1000 Installation and Maintenance Guide Addendum (N0060603)
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Chapter 1 Getting started with BCM35
BCM200/400 BCM 4.0 Installation and Maintenance Guide (N0060612)
Keycode Installation Guide (N0060625)
BCM R2MFC Installation and Configuration Guide (N0027684)
BCM 4.0 Personal Call Manager User Guide (N0027256 02)
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Chapter 1 Getting started with BCM39
How to get Help
This section explains how to get help for Nortel products and services.
Getting Help from the Nortel Web site
The best source of support for Nortel products is the Nortel Support Web site:
http://www.nortel.com/support
This site enables customers to:
•download software and related tools
•download technical documents, release notes, and product bulletins
•sign up for automatic notification of new software and documentation
•search the Support Web site and Nortel Knowledge Base
•open and manage technical support cases
Getting Help over the phone from a Nortel Solutions Center
If you have a Nortel support contract and cannot find the information you require on the
Nortel Support Web site, you can get help over the phone from a Nortel Solutions Center.
In North America, call 1-800-4NORTEL (1-800-466-7835).
Outside North America, go to the Web site below and look up the phone number that applies
in your region:
http://www.nortel.com/callus
When you speak to the phone agent, you can reference an Express Routing Code (ERC) to more
quickly route your call to the appropriate support specialist. To locate the ERC for your product or
service, go to:
http://www.nortel.com/erc
Getting Help through a Nortel distributor or reseller
If you purchased a service contract for your Nortel product from a distributor or authorized
reseller, you can contact the technical support staff for that distributor or reseller.
BCM 4.0 Networking Configuration Guide
40Chapter 1 Getting started with BCM
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Chapter 2
System telephony networking overview
The system supports both public and private networking for telephony traffic.
•The public network is created by PSTN trunk connections from a Central Office terminating
on a telephone system such as the BCM.
•A private network is created when the system is connected through dedicated PSTN lines or
VoIP trunks to other systems. This system may take several forms. At the simplest level, your
system may be behind a private PBX, which connects directly to the Central Office. A more
complicated system may be a node in a network of systems of various types, where calls not
only terminate at the system, but calls may need to be passed through the system to other
nodes unconnected to the originating node.
Refer to the following topics:
•“Basic system configurations”
•“Private network parameters” on page 45
41
Basic system configurations
In the most basic application, your system can provide support for system telephones to make and
receive calls over public network (PSTN) lines.
Two basic system telephony configurations
The following provides a broad overview of the telephony setup for a PBX and a DID system.
PBX system
This setup is for a larger offices which have fewer CO lines than there are telephones. In this case
the lines are pooled, and the line pool access is assigned to all DNs. There may also be a
designated attendant with a telephone that has all lines individually assigned.
BCM 4.0 Networking Configuration Guide
42Chapter 2 System telephony networking overview
Figure 1 PBX system
All telephones
are assigned
access to the line
pool for outgoing
calls
Lines are assigned to a line pool
CO line 1
CO line 2
CO line 3
CO line 4
Receptionist
Assigned all lines/
appearance and
ring
Incoming calls
1A call comes in on a line.
2The receptionist answers the call and finds out who the call is for.
3The receptionist transfers the call to a specific telephone (DN).
4The person can pick up the call at that DN only.
Outgoing calls
1User selects the intercom button or dials a line pool access code, which selects a line in the line
pool.
2The user dials the outgoing telephone number.
DID system
This setup allows you to assign a dedicated phone number to each telephone. The CO assigns a list
of available numbers for each DID (Direct Inward Dial) line. You can change your DN range to
match these numbers, and you use target lines to match each number with a DN.
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Figure 2 DID system
Chapter 2 System telephony networking overview43
Target line mapped to
DN (4005)
Target line mapped to
DN (4006)
Target line mapped to
DN (4007)
Incoming calls
1DID trunks are assigned to be auto-answer.
Note: PRI lines are automatically set to auto-answer.
2All DNs are assigned target lines.
CO DID line
i.e. 769
with range of call
numbers (4005 to
4020)
Target line mapped to
DN (4008)
3A caller dials a system code and a DN. In the example shown above, it might be 769-4006.
4The call comes into the trunk, which answers and maps the call on the target line assigned to
the matching received digits.
5The DN assigned to that target line rings.
You can assign unanswered or busy telephones to Call Forward to another DN, such as a
designated attendant or a voice mail system.
Basic telephony routing
In a basic configuration, simple access codes (for example Line Pool Codes) are used to access the
PSTN network.
In a more complex configuration, more advanced destination codes are required to access multiple
PSTNs, private network resources, and remote nodes. Access to these resources enables advanced
features, such as tandem routing.
BCM 4.0 Networking Configuration Guide
44Chapter 2 System telephony networking overview
Tandem calling toa remote PSTN
A system connected to a private network that uses dedicated circuits or VoIP circuits can allow a
user to dial directly to many other users, on different nodes, using a coordinating dialing plan.
Using a private network saves on toll charges, and local charges, as fewer PSTN accesses are
required for internal and external calling. Several nodes located on one site initiate their external
local calls to a centralized BCM having a T1 termination to the PSTN. This type of configuration
avoids multiple PSTN terminations at other local nodes.
The same tandeming concepts can be applied to inbound calls. DID numbers dialed from the
PSTN can be processed and tandem routed out of the centralized system to the localized remote
nodes. See other details on Tandem routing “Creating tandem private networks” on page 50.
Figure 3 Tandem dialing through a BCM to/from a private network
In the above example, there are three types of callers.
Each type of caller has a specific method of accessing the other two systems.
Callers using BCM
These callers can:
•call directly to a specific telephone
•select an outgoing line to access a private network
•select an outgoing line to access features that are available on the private network
•select an outgoing central office line to access the public network
•use all of the BCM features
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Chapter 2 System telephony networking overview45
Callers in the public network
These callers use the public lines to:
•call directly to one or more BCM DNs
•call into BCM and select an outgoing TIE line to access a private network
•call into BCM and select an outgoing central office line to access the public network
•call into BCM and use remote features
Callers in the private network node
These callers use the private lines to:
•call directly to one or more BCM DNs
•call into BCM and select an outgoing TIE line to access other nodes in a private network
•call into BCM and select an outgoing central office line to access the public network
•call into BCM and use remote features
System numbering and dialing plans
All systems on a private network must coordinate dialing plans, to ensure that calls get directed to
the correct network node. As well, routing becomes more complex, especially if the system is not
an end node and must be configured to relay calls to nodes not directly connected to the system.
The type of dialing plan supported by the network determines whether each node also requires
unique DNs.
Private network parameters
The following provides an overview of the values in the system that affect private networking.
Private networking protocols
The BCM supports the following protocols for private networking:
•PRI: ETSI QSIG, Nortel Voice Networking (MCDN)
•DPNSS (UK only)
•BRI: ETSI QSIG
•T1: E&M
•VoIP trunks (with optional MCDN)
Note: Nortel Voice Networking (MCDN) is referred to as SL-1 in Element
Manager.
BCM 4.0 Networking Configuration Guide
46Chapter 2 System telephony networking overview
BCM systems can be networked together using T-1, PRI or VoIP trunks. PRI SL-1 lines and VoIP
trunks also offer the opportunity to use the MCDN protocol, which provides enhanced trunking
features and end-to-end user identification. If a Meridian 1 is part of the Nortel MCDN network,
the network can also provide centralized voice mail and auto attendant off the Meridian.
MCDN note: MCDN networking requires all nodes on the network to use a common Universal
Dialing Plan (UDP) or a Coordinated Dialing Plan (CDP).
Keycode requirements
Keycodes are required to activate the protocols that are used to create private networking,
including:
•VoIP Gateway keycodes
•an MCDN, DPNSS, or Q. Sig keycode, if you want to use a networking protocol between the
systems
You must purchase and install these keycodes before you can create a network. Consult with your
Nortel distributor to ensure you order the correct keycodes for the type of network you want to
create.
Remote access to the network
Authorized users can access TIE lines, central office lines, and BCM features from outside the
system. Remote users accessing a private network configured over a large geographical area can
avoid toll charges.
Note: You cannot program a DISA DN or Auto DN to a VoIP trunk, as they
act as auto-answer trunks from one private network to the next. However, you
can configure VoIP line pools with remote access packages so that callers can
access telephones or the local PSTN on remote nodes on a tandemed network
that use VoIP trunks between systems.
Lines used for networking
External (trunk) lines provide the physical connection between BCM and other systems in a
private or public network.
The BCM numbers physical lines from 061 to 238. Default numbering depends on the trunk
module positioning within the BCM.
VoIP trunks: Although a VoIP gateway does not use physical lines, it is easier to think of them
that way. Therefore, in the BCM, lines 001 to 060 are used for VoIP trunk functionality.
BCM networking configurations that use PRI lines, require specific DTM modules.
•DTMs configured for PRI are used for incoming and outgoing calls (two-way DID). Incoming
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calls are routed directly to a BCM DN that has a properly configured and assigned target line.
All outgoing calls made through PRI, are initiated using the destination codes.
Chapter 2 System telephony networking overview47
•DTMs configured for T1 can have digital lines configured as Groundstart, E&M, Loop,
or DID.
Target lines are virtual communication paths between trunks and telephones on the BCM system.
They are incoming lines only, and cannot be selected for outgoing calls or networking
applications. With target lines, you can concentrate incoming calls on fewer trunks. This type of
concentration is an advantage of DID lines. BCM target lines allow you to direct each DID number
to one or more telephones. VoIP trunks also require target lines to direct incoming traffic. Target
lines are numbered 241 to 492.
Telephones can be configured to have an appearance of analog lines or multiple appearances of
target lines.
Note: PRI B-channels cannot be assigned as line appearances. PRI
B-channels, or “trunks”, can only be configured into PRI line pools for
inbound routing through target lines with receive digits or outbound routing
through destination codes.
Types of private networks
There are several ways you can create private networks. Configuration can be based on such things
as cost of trunks, proximity of network nodes, size of the private network, and business
requirements for communications.
VoIP-based networking also requires an understanding of IP features such as codecs, jitter buffers,
Quality of Service (QoS) function, and silence suppression. Refer to “Silence suppression” on
page 703 for more information.
The services provided within networks is based on the type of trunks and the protocols assigned to
the trunks. All trunks within the network should be running the same protocols, to provide a
technically sound and stable network.
The following links are procedures to set up basic networks to advanced networks, using the
support protocols within BCM:
•“Routing-based networks using T1 E&M lines” on page 47
•“PRI networking using Call-by-Call services” on page 49
•“PRI SL-1/Q.Sig/DPNSS and VoIP trunk networking” on page 49
Routing-based networks using T1 E&M lines
By properly planning and programming routing tables and destination codes, an installer can
create a dialing plan where T1 E&M lines between BCM systems are available to other systems in
the network
Figure 4 shows a network of three BCM systems. Two remote systems connect to a central system.
BCM 4.0 Networking Configuration Guide
48Chapter 2 System telephony networking overview
Figure 4 Dialing plan for T1 E&M routing network
New York
Network # 2221
Received # 2221
Internal # 2221
Pool H
Pool M
T1 E&M
Toront o
Network # 6221
Received # 6221
Internal # 6221
Pool N
Pool B
T1 E&M
Santa Clara
Network # 4221
Received # 4221
Internal # 4221
Each system must be running BCM software. Each system must be equipped with target lines and
a DTM with at least one T1 E&M line.
The call appears on the auto answer line on the BCM in Santa Clara as 6-221. Because 6 is
programmed as a destination code for Toronto on the Santa Clara system, another call is placed
using route 002 from Santa Clara to Toronto. At the Toronto system, the digits 6-221 are
interpreted as a target line Private received number. The call now alerts at DN 6221 in Toronto.
Note: Network calls that use routes are subject to any restriction filters in
effect. If the telephone used to make a network call has an appearance of a line
used by the route, the call will move from the intercom button to the Line
button. The telephone used to make a network call must have access to the line
pool used by the route. Network calls are external calls, even though they are
dialed as if they were internal calls. Only the features and capabilities available
to external calls can be used.When programming a button to dial a Network
number automatically (autodial), network calls must be treated as external
numbers, even though they resemble internal telephone numbers. Routes
generally define the path between your BCM switch and another switch in
your network, not other individual telephones on that switch.
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Chapter 2 System telephony networking overview49
PRI networking using Call-by-Call services
The example shown in Figure 5 highlights the use of PRI Call-by-Call services. It shows two
offices of a company, one in New York and one in Toronto. Each office is equipped with a BCM
system and a PRI line. Each office has to handle incoming and outgoing calls to the public
network. In addition, employees at each office often have to call colleagues in the other office.
Refer to “Private networking: PRI Call-by-Call services” on page 383 for more information.
Figure 5 PRI networking using Call-by-Call Services
New York office
PRI
Central
Office
Central
Office
Toronto office
Network # 2221
Received # 2221
Internal # 2221
DID # 763-2221
BCM
PRI
Network # 6221
Received # 6221
Internal # 6221
DID # 562-6221
BCM
TIE Connection
Public Network
To reduce long distance costs, and to allow for a coordinated dialing plan between the offices,
private lines are used to handle inter-office traffic.
If call-by-call services were not used, each BCM system might have to be equipped with the
following trunks:
•12 T1 DID lines needed to handle peak incoming call traffic.
•eight T1 E&M lines needed to handle inter-office calls.
•eight lines needed to handle outgoing public calls
PRI SL-1/Q.Sig/DPNSS and VoIP trunk networking
PRI SL-1 trunks and VoIP trunks can be used to create private networks between BCM systems or
between BCM systems and larger call servers such as Meridian 1, Succession 1000/M, DMS100/
250 and CSE.
ETSI-QSIG and DPNSS private networking is configured very similarly, although network
features may be supported slightly differently due to local line and network requirements.
BCM 4.0 Networking Configuration Guide
50Chapter 2 System telephony networking overview
If the MCDN protocol is added to this type of private network, the network provides additional
network management features, as well as allowing centralized voice mail features to be available
to all nodes on the network.
The following sections describe the different aspects of SL-1 and MCDN private networking.
•“System dialing plans”
•“Creating tandem private networks”
•“Understanding MCDN network features” on page 53
•“Networking with ETSI QSIG” on page 56
•“Private networking with DPNSS” on page 65
The type of network you require depends on the equipment in the network, and how you want to
use the network.
•With MCDN, you can tie a set of BCM systems together with PRI SL-1(MCDN)/ETSI-QSIG,
DPNSS or VoIP trunks to create a tandem network. This type of network provides the
additional advantage of providing private line access to local PSTNs for all the nodes on the
network.
Note: A keycode is required to use SL-1(MCDN).
System dialing plans
Both these types of networks require similar setups for dialing plans and routing. Each node must
have a way to route external calls to the adjacent node or nodes. To do this, all nodes must have the
same Private DN length.
You use routing and a private dialing plan to control calls over the network. Each example in this
section describes the routing configurations that are required to support calls over the network.
Depending on the type of dialing plan you choose, each node must also have a unique location or
steering code so the calls can be correctly routed through the nodes of the network. MCDN
networks also require a Private Network ID, which is supplied by the Meridian network
administrator to define how the Meridian system identifies each node.
Creating tandem private networks
You can tie a number of BCM systems together with SL-1 lines. This tandem network provides
you with the benefits of end-to-end name display and toll-free calling over the SL-1 private link.
Each BCM becomes a node in the network. In this type of network, you must ensure that each
BCM system, known as a node of the network, is set up to route calls internally as well as to other
nodes on the system. This means each node must have a route to the immediately adjacent node,
and the correct codes to distribute the called numbers. Each node must have a unique identification
number, which is determined by the type of dialing plan chosen for the network.
As well, you can save costs by having a public network connection to only one or two nodes, and
routing external calls from other nodes out through the local PSTN, thus avoiding toll charges for
single calls.
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Chapter 2 System telephony networking overview51
VoIP note: You can also use VoIP trunks between some or all of the nodes. The setup is the same,
except that you need to create gateway records for each end of the trunk, and routing tables to
accommodate the gateway codes.
Routing for tandem networks
In tandem networks, each node needs to know how to route calls that do not terminate locally. To
do this, you set up routes for each connecting node by defining destination codes for each route.
If the node is also connected to the public network, the usual routing is required for that
connection.
The following tables show the routing tables for Node A and Node C for external and internal
terminating calls.
Note: The PRI and ETSI QSIG trunks are en bloc dialing lines, so all dialed
digits are collected before being dialed out.
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52Chapter 2 System telephony networking overview
Table 1 Node A destination code table, external termination
RouteAbsorb lengthDestination code (public DNs)
4 (PSTN)191604
3 (Node B)091403762 (Node B)
3 (Node B)091403765 (Node E)
4 (PSTN)19140376* (not internal network)
4 (PSTN)1914037* (not internal network)
4 (PSTN)191403* (not internal network)
4 (PSTN)19* (not internal network)
* This wild card represents a single digit.
Table 2 Node A destination code table, internal termination
RouteAbsorb lengthDestination code (private DNs)
3 (Node B)0392 (Node B)
3 (Node B)0395 (Node E)
5 (Node C)0393 (Node C)
5 (Node C)0394 (Node D)
5 (Node C)0396 (Node F)
Table 3 Node C destination code table, external termination
RouteAbsorb lengthDestination code (Public DNs)
3 (Node B)091613764 (Node D)
3 (Node B)091613766 (Node F)
4 (PSTN)19161376* (not internal network)
4 (PSTN)1916137* (not internal network)
4 (PSTN)191613* (not internal network)
4 (PSTN)19161* (not internal network)
4 (PSTN)1916* (not internal network)
4 (PSTN)191* (not internal network)
4 (PSTN)19 (not internal network)
* This wild card represents a single digit.
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Chapter 2 System telephony networking overview53
Table 4 Node C destination code table, internal termination
RouteAbsorb lengthDestination code (Private DNs)
3 (Node D)0394 (Node D)
3 (Node D)0396 (Node F)
5 (Node A)0391 (Node A)
5 (Node A)0392 (Node B)
5 (Node A)0395 (Node E)
Understanding MCDN network features
When you connect your BCM systems through PRI-SL-1/ETSI QSIG/DPNSS or VoIP trunks, and
activate the MCDN protocol, your network provides a number of network call features. You can
use this protocol to network other BCM systems, such as the tandem system shown in the previous
section, Norstar systems, Meridian 1 systems, Succession systems, DMS 100 systems or CSE
systems.
Table 5 lists the MCDN features that are provided by all SL-1/VoIP networks where MCDN is
active. The features affect call redirection and trunking functions.
Centralize trunking“ISDN Call Connection Limitation” on page 54 (ICCL)
“Trunk Route Optimization” on page 55 (TRO)
“Trunk Anti-tromboning” on page 55 (TAT)
Network Call Redirection Information
NCRI (Network Call Redirection Information) builds on the following BCM features:
•External Call Forward
•Call Transfer
•Call Forward
NCRI adds the ability to redirect a call across an MCDN network using Call Forward (All Calls,
No Answer, Busy) and Call Transfer features. The call destination also receives the necessary
redirection information. This feature allows the system to automatically redirect calls from within
a BCM system to the mail system, such as Meridian Mail, which resides outside the BCM system
on the Meridian 1.
Figure 6 shows an example of this situation, where user A calls user B on the same BCM. If user B
is busy or not answering, the call automatically gets transferred to a Meridian Mail number
(user C) across an MCDN link between the BCM system and the Meridian 1 system where the
mailboxes are set up.
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54Chapter 2 System telephony networking overview
Figure 6 Network call redirection path
BCM
SL-1 MCDN
Telephone A
Original call
Telephone B
Call forwarded to Meridian Mail
Meridian 1
with Meridian Mail
ISDN Call Connection Limitation
The ICCL (ISDN Call Connection Limitation) feature piggybacks on the call initiation request and
acts as a check at transit PBX points to prevent misconfigured routes or calls with errors from
blocking channels. Also refer to “ISDN overview” on page 709.
This feature adds a transit/tandem counter to a call setup message. This counter is compared at
each transit PBX with a value programmed into the transit PBX, in a range from 0 to 31. If the call
setup counter is higher than the PBX value, the call will be blocked at the PBX system and cleared
back to the network. This prevents calls from creating loops that tie up lines.
Figure 7 demonstrates how a call might loop through a network if the system is not set up
with ICCL.
Figure 7 Call loop on system without ICCL
BCM
BCM
Telephone A
BCM
Meridian 1
Meridian 1
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Chapter 2 System telephony networking overview55
Trunk Route Optimization
Trunk Route Optimization (TRO) finds the most direct route through the network to send a call
between nodes. This function occurs during the initial alerting phase of a call.
BCM configurations:
•Under Configuration > Dialing Plan > Private Network, select the check box beside TRO.
•Configure call routing for all optimal routes.
•Configure call forward (All Calls, No Answer, Busy) or Selective Line Redirection to use the
optimal routes.
This feature avoids the following situation: A call originating from a BCM system may be
networked to a Meridian system, which, in turn, is networked to another Meridian system, which is
the destination for the call. If the call routes through the first Meridian (M1) to reach the second
Meridian (M2), two trunks are required for the call. An optimal choice is a straight connection to
M2. This finds these connections and overrides the less-efficient setup.
Figure 8 shows two call paths. The first route, through the Meridian, demonstrates how a call
might route if TRO is not active. The second route, that bypasses the Meridian, demonstrates how
TRO selects the optimum routing for a call.
Figure 8 Call paths with and without TRO
BCM
Telephone A
Trunk Anti-tromboning
Trunk Anti-Tromboning (TAT) is a call-reroute feature that works to find better routes during a
transfer of an active call. This feature acts to prevent unnecessary tandeming and tromboning of
trunks.
Meridian 1
PRI SL-1
Original call (no TRO)
Forwarded call (no TRO)
Call path with TRO
Telephone B
Meridian 1PRI SL-1
PRI SL-1
Telephone C
Note: This feature is not applicable for alerting calls.
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56Chapter 2 System telephony networking overview
Figure 9 shows how TAT reduces the line requirements. The solid line shows Telephone A calling
Telephone B and being transferred over an additional PRI line to Telephone C. With TAT active,
the same call is transferred to Telephone C over the same PRI line.
Figure 9 Call paths with and without TAT
BCM
Telephone ATelephone BTelephone C
Forwarded call (no TAT)
Forwarded call (using TAT)
Networking with ETSI QSIG
(ETSI QSIG applies only to international systems equipped with a DTM or BRIM. It is not
supported on VoIP)
ETSI QSIG is the European standard signaling protocol for multi-vendor peer-to-peer
communications between PBX systems and/or central offices.
BCM
Other information in this section: “ETSI Euro network services” on page 57.
Figure 10 illustrates an ETSI QSIG network. Note that this is exactly the same setup as that shown
in the MCDN section for North America. The hardware programming for ETSI QSIG is described
in Table 6. All other configurations are the same as those shown in the MCDN section for North
America.
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Figure 10 ETSI QSIG networking
Chapter 2 System telephony networking overview57
BCM West end branch
Network # 2221
Received # 2221
Internal # 2221
PRI/BRI ETSI QSIG
Central
Office
PBX
Network # 6221
Received # 6221
Internal # 6221
PRI/BRI
ETSI QSIG
BCM East end branch
DN 4221
PRI (public
protocol)
Table 6 lists the settings for some of the hardware parameters for ETSI QSIG networking example
shown in Figure 10.
Table 6 Hardware programming for branch offices
West End office:
Hardware
programming
DTM/BRIM PRI/BRIHardware
ProtocolETSI QSIGProtocolETSI QSIG
BchanSeqAscend
(PRI only)
ClockSrcPrimaryClockSrcPrimary
ETSI Euro network services
If your system has ETSI ISDN BRI/PRI lines, you can activate the malicious call identification
(MCID) and Network Diversion features. Advice of charge-end call (AOCE) is active if your
service provider has activated that service on the line.
East End office:
DTM/BRIMPRI/BRI
programming
BchanSeqAscend
(PRI only)
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58Chapter 2 System telephony networking overview
When the features are activated, users can:
•display a call charge
•redirect calls over the ETSI ISDN BRI/PRI line to the outside network
•tag malicious calls
Advice of Charge-End of Call (AOCE) — AOCE is a supplementary service available from your
service provider on ETSI ISDN BRI/PRI links. This feature allows the BCM user to view the
charges for an outgoing call once the call completes. This information is also reported to the Call
Detail Reporting Application. The information can be provided in currency or charging units,
depending on how the feature is set up by your service provider.
To invoke the feature, the user presses FEATURE 818.
DPNSS 1 services
The Digital Private Network Signaling System (DPNSS 1) is a networking protocol enhancement
that extends the private networking capabilities of existing BCM systems. It is designed to offer
greater centralized functionality for operators, giving them access to BCM features over multiple
combined networks.
Note: The DPNSS feature is dependent on which region is loaded on your
system at startup and requires that a software keycode was entered to enable
the feature. The feature also requires a DTM-based connection.
Refer to the following topics:
•“DPNSS 1 capabilities”
•“DPNSS 1 features” on page 59
•“Private networking with DPNSS” on page 65
DPNSS 1 allows a BCM local node, acting as a terminating node, to communicate with other
PBXs over the network. For example, corporate offices separated geographically can be linked
over DPNSS 1 to other BCM nodes, bypassing the restrictions of the PSTNs to which they may be
connected. This allows connected BCM nodes to function like a private network, with all features
of BCM accessible.
Note: BCM DPNSS 1 works as a terminating node only. BCM to BCM
DPNSS is not supported.
DPNSS 1 features can be used on any BCM telephone. On most BCM telephones, you must use
specific keys and/or enter a number code to access the features.
DPNSS 1 capabilities
A single BCM node, acting as a terminating node on the network, supports the following
capabilities over DPNSS 1 lines:
•Direct Dial Inward (DDI) for incoming calls.
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Chapter 2 System telephony networking overview59
•Originating Line Identification (OLI) for incoming and outgoing calls:
— For incoming calls, the Calling Line Identification (CLI/CLID) information is displayed to
the user on telephones with line display. This must be configured in programming.
— For outgoing calls, the directory number of the originating party is sent out as OLI.
•Terminal Line Identification (TLI) for incoming and outgoing calls. Referred to as Called Line
Identification.
•Selective Line Redirect (SLR) and External Call Forward (ECF) implemented on calls
between DPNSS 1, and BRI/PRI, DASS2, and analog lines.
•These remote access features are supported on DPNSS: DDI, line pool access code,
destination codes, and remote page feature codes.
Note: Keycodes are required to enable DPNSS 1.
DPNSS to Embark connections
DPNSS lines connected to an Embark switch perform call redirection/diversion using the Call
Forward feature to create a tandem link back to the switch. Since this is different from other
switches, you must select the type of switch DPNSS will be connecting to when you do module
programming.
Before you program Call Forwarding, ensure that:
•Both real channels and virtual channels are provisioned.
•Destination or line pool codes are programmed for the DPNSS to Embark link.
Also, during programming for Call Forward No Answer and Call Forward on Busy, when you
enter the Forward to: digits, the system performs a validation check with the switch on the
number. (Configuration > Telephony > Sets > Active Sets > Line Access> Properties tab)
DPNSS 1 features
The following features are available and can be programmed over DPNSS lines:
•“Three party service” on page 60
•Diversion (“Using the diversion feature” on page 61)
•Redirection (“Using the Redirection feature” on page 62)
•“Executive intrusion” on page 62
•“Call Offer” on page 63
•“Route optimization” on page 64
•“Loop avoidance” on page 65
•Message Waiting Indication
The following parameters can be configured for DPNNS 1 lines:
•Line type
•Prime set
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60Chapter 2 System telephony networking overview
•CLID set
•Auto privacy
•Answer mode
•Auxiliary ringer
•Full autohold
Some features are transparent to the user, but must be programmed to be activated. Others are
available for end-user programming at the telephone. Details about these features are given below.
Three party service
The Three Party Service allows a user, usually an operator, to take three-party conference by
calling two other parties from one telephone. Once the connection is made, the controlling party
can hang up, leaving the other two connected. They can even put one party on hold, and talk to the
other party.
If the original call is inbound to the BCM, any subsequent three-party conference will be
terminated when the BCM user drops out of the call.
For example, PSTN user A calls a BCM user.
After the call is established, the BCM user initiates a three-party conference with another PSTN
user. If the BCM user hangs up, the conference is closed and all parties are dropped from the call.
Following are the two rules:
Rule 1 - if the call is an inbound call, the BCM user can create a conference, but if they drop out,
the calls are all dropped.
Rule 2 - if the BCM initiates outbound calls to 2 parties, conferences them, the BCM user can drop
out leaving them connected.
Note: BCM does not support Hold over the DPNSS link itself. This means
that the conferenced party on the distant end of the network cannot place a
Three Party Service call on Hold.
This feature is designed to allow operators to assist in the connection of calls from one main
location.
Making a conference call
To initiate or disconnect from a conference call on a BCM system over DPNSS 1, use the
procedure described in the BCM 4.0 Device Configuration Guide (N0060600)
Note: Three Party Service is supported on model 7000 telephones, but in a
receive-only fashion. These telephone types cannot initiate Three Party
Service. For more information about these telephone types, see the BCM 4.0 Telephony Device Installation Guide (N0060609).
.
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(model 7000 phones, supported in Europe only.)
Chapter 2 System telephony networking overview61
Using the diversion feature
Diversion is a DPNSS 1 feature for BCM that allows users to forward their calls to a third party on
the DPNSS 1 network. This feature is similar to Call Forward on BCM, but takes advantage of the
broader capabilities of DPNSS.
There are five variations of Diversion: Call Diversion Immediate, Call Diversion On Busy, Call
Diversion On No Reply, Bypass Call Diversion, and Follow-me Diversion. These variations are
described below:
•Diversion Immediate diverts all calls to an alternate telephone. This function is programmed
by the user at their telephone.
•Diversion On Busy diverts all calls to an alternate telephone when a telephone is busy. This
feature is programmed in the Element Manager.
•Diversion On No Reply diverts calls that go unanswered after a specified amount of time. This
feature is programmed in the Element Manager.
•Bypass Call Diversion overrides all call forward features active on a telephone over a DPNSS
line. An incoming call to the telephone will not be forwarded; instead, the telephone will
continue to ring as if call forward were not active. This feature is used to force a call to be
answered at that location. Bypass Call Diversion is a receive-only feature on BCM, and cannot
be used from a BCM telephone.
•Follow-me Diversion is also a receive-only feature. It allows the call forwarded destination to
remotely change the BCM call forwarding programming (Call Forward All Calls [CFAC]
feature) to a different telephone.
Note: BCM CFAC must be active and the destination set/PBX system must
support the feature.
For example, user A forwards all calls to telephone B, a temporary office. Later, user A moves
on to location C. The user does not have to be at telephone A to forward calls to location C.
Using telephone B and Follow-me Diversion, the user can forward calls from A to location C.
Follow-me diversion can be cancelled from the forwarded location.
•Diversion on Busy and Diversion on No Reply cannot be cancelled from the forwarded
telephone. These are programmable only by an installer and not by the user.
•If multiple telephones are programmed to take a call, the first telephone to respond will act. All
other telephones responding are ignored. Therefore, if the first telephone to respond has
Diversion enabled, this feature will be invoked.
Restrictions by telephone type
•all variations supported on BCM digital and IP telephones
•ATA2/ASM8+—all variations supported on an ATA
•ISDN—all variations supported on ISDN telephones, except Diversion on Busy and CFWD
Busy
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62Chapter 2 System telephony networking overview
Setting Diversion
You set Diversion for DPNSS in the same way as Call Forward. You will need to enter the end DN
when prompted. You may also need to include the DPNSS 1 routing number.
Using the Redirection feature
Redirection is a DPNSS 1 feature similar to BCM Transfer Callback. Redirection lets a call
awaiting connection, or reconnection, be redirected by the originating party to an alternate
destination after a time-out period. Failed calls can also be redirected. Priority calls are not
redirected.
Note: The address to redirect depends on the history of the call. Calls that
have been transferred are redirected to the party that transferred them. In all
other cases, the address to redirect is the one registered at the PBX system
originating the redirection.
Note: BCM does not support the redirection of BCM originated calls, even
over DPNSS 1.
The Diversion on No Reply feature takes precedence over Redirection.
Restrictions by telephone type
•For telephones with a single line display, the number key (#) acts as MORE and the star key (*)
acts as VIEW
•ISDN—all variations supported on ISDN telephones
Setting redirection
The timer used for the network Callback feature is also used for redirection.
Executive intrusion
Executive Intrusion (EI) is a DPNSS 1 feature that allows an operator, or other calling party, to
intrude on a line when it is busy. An example of the use of this feature is to make an important
announcement when the recipient is on another call.
EI is similar in functionality to BCM Priority Call, but it is a receive-only feature on BCM
telephones. EI cannot be initiated from a BCM telephone. The person using this feature must be on
another PBX system on the DPNSS 1 network.
When EI is used to intrude on a call in progress, a three-way connection is established between the
originating party and the two parties on the call. The result is very much like a conference call.
When one of the three parties clears the line, the other two remain connected, and EI is terminated.
Restrictions by telephone type
•ATA2/ASM8+—supported
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Chapter 2 System telephony networking overview63
•ISDN—not supported
The telephone receiving the intrusion displays Intrusion Call. A warning indication tone
will sound after intrusion has taken place, and the standard conference call tone will sound every
20 seconds.
Intrusion levels
Whether or not a telephone will accept or reject an Executive Intrusion request depends on the
level of intrusion protection programmed. Each telephone (DN) has an Intrusion Capability Level
(ICL) and four Intrusion Protection Levels (IPL).
When the ICL of the intruding telephone is higher than the IPLs of both telephones on the active
call, EI occurs. Nortel recommends setting the IPLs of most BCM telephones to the default of
None, Low, or Medium.
Intrusion levels are described as follows:
•ICL: determines the ability of the attendant to intrude. As long as the ICL is higher than the
IPL of the wanted party, EI is allowed. Since EI is a receive-only feature, the ICL cannot be set
on BCM.
•IPL: determines the ability of the attendant to refuse intrusion. If the IPL is lower than the ICL
of the originating party, EI is allowed. For general purposes, Nortel recommends setting the
IPL to None, Low, or Medium, unless intrusion is not wanted.
Call Offer
Call Offer over DPNSS 1 allows a calling party to indicate to the wanted party that there is an
incoming call available, even though there is no answer button available to present the call on the
telephone. The intended recipient can ignore, accept, or decline the offered call. Call Offer is
useful in increasing the call-coverage capability of a BCM system, and helps to lift the network
processing load. It is a receive-only capability on BCM: incoming calls would be initiated at
another PBX system on the DPNSS 1 network.
An example of Call Offer in use is an operator or attendant who has a number of calls coming in at
once. The operator can call offer one call and move to the next without waiting for the first call to
be answered.
Call Offer Displays
When a Call Offer is made by the originating exchange, the target telephone displays a message,
and a tone is heard. When an offered call arrives on telephones with line display, the user sees
XX...X wtng if the calling party ID is available and CLID is enabled. If CLID is not available
or CLID is disabled, Line XXX waiting appears (the line name associated with the call). If
there are more than 11 digits in the incoming number, only the last 10 will display.
If Call Queuing is programmed for the system, the display shows Release Line XXX.
This is the line name of the highest-priority queued call if it is an offered call.
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64Chapter 2 System telephony networking overview
Restrictions by telephone type
•model 7000 telephone — associated LED or LCD flashes, and a tone is heard (model 7000
phones, supported in Europe only.)
•ATA2/ASM8+—Call Offer is supported as a Camp On feature, and a tone is heard
•ISDN—not supported
Note the following general conditions and restrictions:
•DND on busy must be selected (DN > Capabilities and Preferences > Capabilities tab) for a
telephone to accept Call Offer.
•If CF on busy is programmed for the telephone, Call Offer is not accepted.
•The target line for the telephone must be set to: If busy: busy tone, which is the default.
•Call Offer does not work if sent over Manual answer lines. It is recommended that the lines be
left at the default: Auto.
User actions
The party receiving a Call Offer has three choices:
•Ignore it. After a programmed time interval, the Offer request is removed.
•Reject it. If the user activates Do Not Disturb on Busy (DND) when the Call Offer request is
made, the request is removed from the telephone. The calling party is informed of the
rejection.
Note: A call cannot be offered to a telephone with DND active. The line
indicator for external incoming calls still flashes.
•Accept it. The Offer is accepted by releasing the active call.
Note: Forward on Busy takes priority over DND on Busy. Call Offer cannot
be accepted by putting an active call on hold.
Route optimization
Route Optimization is a DPNSS 1 feature for BCM that allows calls to follow the optimum route
between two end PBXs. This allows efficient use of network resources.
Route Optimization is initiated by the system and is transparent to the user. However, the user may
see a call switch from an appearance on the telephone to another appearance key or from an
intercom button to the appearance key or vice versa. This occurs when BCM receives a Route
Optimization request and initiates a new call to follow the optimal route.
If a telephone is active on a private line call, the Route Optimization call being established may go
on a public line. This will cause a loss of privacy on that line.
Data calls are rejected by Route Optimization in order to ensure the data transmission is not
affected.
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Chapter 2 System telephony networking overview65
Certain situations result in Route Optimization not taking place. For example, calls that are using
Hold, Parking or Camp features do not undergo Route Optimization, and if a Route Optimization
call undergoes Diversion, the Route Optimization is dropped.
Setting Route Optimization
There is no system programming required for the feature when BCM is working as a terminating
PBX system. However, BCM must have a private access code programmed that maps to a valid
destination code or line pool code on DPNSS lines. Further, Allow Redirect must be selected.
Loop avoidance
Errors in the configuration of a network may make it possible for a call to be misrouted, and arrive
at a PBX system through which it has already passed. This would continue, causing a loop which
would eventually use up all of the available channels. The Loop Avoidance service permits
counting of DPNSS 1 transit PBXs and rejecting a call when the count exceeds a predetermined
limit.
Private networking with DPNSS
(International only)
DPNSS supports the Universal Dialing Plan (UDP), an international standard for sending and
receiving private numbers over networks. The UDP requires that a dialing number include the
following:
•a Private Access Code, programmed into the system as part of the destination code table to
prevent conflicts with the internal numbering system. (Access Codes)
•a Home Location Code (HLC) assigned to each PBX system, and configured as part of the
destination code (a maximum of seven digits). For each HLC, a destination code must be
programmed in the system. (Configuration >Telephony > Dialing Plan > Private Networking)
•a Directory Number (DN) assigned to each extension as a line appearance. The DN appears as
the last string segment in a dialed number. In the number 244-1111, 1111 is the DN.
A typical Private Number, using a private access code and dialed from another site on the network,
appears below.
Private Access Code + Home Location Code+ Directory Number= Calling Party Number
6+ 848+ 2222= 6-848-2222
In this networking example, a private network is formed when several systems are connected
through a Meridian 1 and a terminating BCM system. Each site has its own HLC and a range of
DNs. Figure 11 illustrates this example.
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66Chapter 2 System telephony networking overview
Table 7 shows examples of the construction of numbers used when dialing within the example
network. Note that 6 is the Private Access code.
Table 7 Calling numbers required for DPNSS network example
Calling Site LOC/HLC
Site A 244244 1111Site B6 668 2222668 2222
Site B668668 2222Site D6 848 2222848 2222
Site D8482222Site D22292229
Site C496496 3333Public DN9 563 3245563 3245
Figure 11 DPNSS networking
Terminating
BCM Site A
DN # 111
LOC # 244
DPNSS
Calling Party
Number
Private
Network
Called SiteDialing String
DPNSS
BCM
Site C
DN # 3333
LOC # 496
Called Party
Number
DPNSSDPNSS
Meridian M1
LOC # 563
BCM
Site B
DN # 2222
LOC # 668
BCM Site D
DN # 2229
Extension 2222
LOC # 848
Calls are dialed and identified to the system as follows:
•To reach a telephone inside the Private Network, at the BCM site, the user dials the DN of
choice.
•To reach a telephone inside the Private Network, from another site, the user dials HLC + DN.
•To reach a telephone outside the Private Network, the user dials an Access Code + HLC + DN.
Each node has its own destination (dest) code, which includes the appropriate access and HLC
codes to route the call appropriately.
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Chapter 3
Telephony programming: Configuring call traffic
Telephony call traffic has a number of configuration requirements. Some configuration is common
to both incoming and outgoing traffic. Other settings are specific to the call direction.
In the case of private networking, call configuration becomes more complex, as remote systems
send calls over the private network to other nodes or to your system PSTN network and your local
PSTN handles calls directed to remote nodes through your system.
Line programming and number planning both play critical roles in controlling call traffic for your
system.
For incoming calls, you can have a central reception point, or you can specify target lines to one or
more telephones to receive directed calling.
You can arrange your telephones in Hunt groups, ringing groups, or call groups that use
system-wide call appearance (SWCA) assignments to share calls.
You can also configure lines for use by system users who call in from outside the system. You can
give them direct access to the system with an Auto DN, or you can configure the line so they hear
a stuttered dial tone, at which point they need to enter a password (CoS) to gain access
(DISA DN).
For outgoing calls, you can assign one or more intercom keys to directly link to a line pool or
prime line, or allow line pool access codes, destination codes, or internal system numbers to direct
the call. Telephones without intercom keys on the telephone have intercom keys assigned, but the
user must pick up the handset to access calls. In this case, the intercom key is an assigned DN.
For calls within the system, all telephones are virtually linked within the system. To call another
telephone inside the system, lift the handset and dial the local DN. In this case, the prime line has
to be set to intercom or none.
For calls going outside the system:
•If you assign the prime line to a line pool, all the lines in that line pool must be assigned to the
telephone. When you pick up the handset, the telephone automatically grabs the first available
line from the assigned line pool. In this configuration, you must ensure that the outgoing
number is allowed by the line pool.
•If you assign the prime line to an intercom button, when you press the intercom button you get
system dial tone. Then, you enter a line pool access code or a destination code to direct the
outgoing call to the appropriate line pool, where it exits the system on any available line in that
pool.
Figure 18 Configuring outgoing call traffic (Sheet 2 of 2)
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Chapter 4
Application resources panel
The application resources panel allows you to modify resources allocated to applications on the
BCM. While the panel tracks four types of resources, DSP resources are generally the only type of
resources that affect performance on the BCM. For more information on planning your application
resources, see “Determining the resources you require” on page 88.
Note: Do not change these settings unless you want to restrict resources.
The application resources panel consists of three tables and a panel:
•DS30 Allocation
•“Total Resources” on page 78
•“Reserved Resources” on page 79
•“Application Resource Reservations” on page 97
77
DS30 Allocation
The DS30 allocation determines how many IP telephones you can connect to your system. If you
have a system that does not use IP telephones, the number of signaling channels does not affect
your configuration. For more information, refer to “Changing the DS30 split” on page 98.
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78Chapter 4 Application resources panel
Figure 19 DS30 Allocation
Table 2 DS30 allocation
AttributeValueDescription
DS30 split<read-only>Number of signaling channels
Actions
Modify<button>Modify the resource allocation.
DS30 split2/6
3/5
OK<button>Confirm the selection. This may
Cancel<button>Cancel change.
available.
Select the resource allocation.
require a system reboot.
Total Resources
The total resources options show the maximum resources available for each type of resources.
Figure 20 Total Resources
Table 8 Total Resources (Sheet 1 of 2)
AttributeValueDescription
Signalling
channels
<read-only>The total number of signalling channels on the system.
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Table 8 Total Resources (Sheet 2 of 2)
AttributeValueDescription
Chapter 4 Application resources panel79
VDI
channels
Media
channels
DSP
resources
<read-only>The total number of VDI channels on the system.
<read-only>The total number of media channels on the system.
<read-only>The total number of DSP resources on the system.
Reserved Resources
The Reserved Resources options show the resources currently reserved or in use.
Figure 21 Reserved Resources
Table 9 Reserved Resources
AttributeValueDescription
Signalling
channels
VDI channels<read-only>The number of VDI channels in use on the system. This number can
Media
channels
DSP resources <read-only>The number of DSP resources in use on the system. This number
<read-only>The number of signalling channels in use on the system. This
number can change based on the values entered for applications,
and on the those applications currently in use.
change based on the values entered for applications, and on the
those applications currently in use.
<read-only>The number of media channels in use on the system. This number
can change based on the values entered for applications, and on the
those applications currently in use.
can change based on the values entered for applications, and on the
those applications currently in use.
The Application Resource Reservations table allow you to set minimum and maximum values for
each of six types of applications. The table contains 10 columns, eight of which are read-only. For
information on determining the appropriate values for each type of application, see “Rules for
managing the resources” on page 85.
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80Chapter 4 Application resources panel
Figure 22 Application Resource Reservations
Table 10 Total Resources
AttributeValueDescription
Application<read-only>The name of the application.
MinimumNumeric valueThe minimum number of resources reserved at all times for the
application. If a value of 2 is entered, the system will always reserve
enough resources for 2 instances of the application.
MaximumNumeric value, or the string
MAX
Licence<read-only>The number of licenses the system has activated for the application.
System Max.The maximum instances of an application the BCM can support.
Change
Pending
Sig. Ch.<read-only>The number of signalling channels reserved by the application. This
VDI Ch.<read-only>The number of VDI channels reserved by the application. This can
Media Ch.<read-only>The number of media channels reserved by the application. This
DSP<read-only>The number of DSP resources reserved by the application. This can
<read-only>If this box is checked, a change is pending to the system. Most
The maximum number of applications to allow. If the value is set to
MAX, the system will allow up to the system maximum, as long as
there are enough resources.
If the value is N/A, the application does not require licenses.
changes take effect immediately, but in some instances, a change
may wait until applications shut down. Details about changes
pending can be seen in the details panel.
can be changed by modifying the minimum and maximum values for
the application. If the field has a value of N/A, the application does
not require this type of resource.
be changed by modifying the minimum and maximum values for the
application. If the field has a value of N/A, the application does not
require this type of resource.
can be changed by modifying the minimum and maximum values for
the application. If the field has a value of N/A, the application does
not require this type of resource.
be changed by modifying the minimum and maximum values for the
application. If the field has a value of N/A, the application does not
require this type of resource.
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Details for application
The Details for Application panel changes whenever a different row is selected from the
Application Resource Reservations table. It reflects the current minimum and maximum limits, in
instances where changes do not happen immediately.
Table 11 Total Resources
AttributeValueDescription
Chapter 4 Application resources panel81
Current
minimum
assigned limit
Current
maximum
assigned limit
Notetext fieldEnter any notes regarding these limits.
<read-only>The current minimum assigned for an application.
<read-only>The current maximum assigned for an application.
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82Chapter 4 Application resources panel
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Chapter 5
Configuring application resources
The following describes how to set up the application resources controlled by the Media Services
Card (MSC), which is the control center for voice and data traffic in the BCM.
The following path indicates where to configure the resources in Element Manager:
Warning: Only system administrators should have access to these Element Manager
records. Changing settings can affect other parts of the system. You need to understand the
consequences of any changes before you make them. Some changes are NOT reversible.
Refer to the following topics:
•“Types of resources”
•“Rules for managing the resources” on page 85
•“Determining the resources you require” on page 88
•“Understanding the minimum and maximum values” on page 96
•“Changing the DS30 split” on page 98
83
Types of resources
Application resources are required for the following features:
•system functions
•voice mail, contact center, and IVR (Interactive Voice Response)
•Fax mail
•IP telephony trunks
•IP clients
•Dial-on-Demand (DoD) WAN and Backup ISDN WAN connections
When you configure the resources, you are configuring how BCM shares the resources between
these features.
There are several values that you must check when you are configuring resources:
•“Signaling channels” on page 84
•“Media channels” on page 84
•“DSP resources” on page 84
•“Voice bus paths” on page 84
•“Media gateways” on page 84
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84Chapter 5 Configuring application resources
Signaling channels
Signaling channels are the communication channels used to send control signals to and from the
MSC. You must have one signaling channel for each connected device and enabled feature port.
The number of signaling channels on your system determines the number of devices that you can
connect and feature ports that you can enable on your system. Signaling channels are also known
as D-channels.
Media channels
Media channels are the communication channels used to send voice and data information between
the devices and feature ports. Media channels are required only when a device or feature is sending
or receiving voice or data information. For this reason, the devices and feature ports can share
media channels.
The number of media channels you have determines how many devices and feature ports can
exchange voice and data information at the same time. Media channels are also known as
B-channels.
DSP resources
Digital Signal Processors (DSP) provide the voice processing functions on BCM. Voice
processing is required to convert voice information to and from digital format for voice mail,
Contact Center and IVR. Voice processing is also required to handle encoding and decoding of IP
telephony calls. The DSPs are located on the MS-PEC cards installed in your MSC.
The number of DSP resources you have determines the number of voice mail ports, contact center
ports, Fax mail ports, IVR ports, IVR Fax ports, WAN connections, and IP telephony calls that can
be active at the same time.
Voice bus paths
The voice bus paths are the communication channels between the DSPs on the MS-PECs and the
master DSP on the MSC. One voice bus path is required for each voice processing task that is
operating on the DSPs.
There are 62 voice bus paths available on BCM.
Media gateways
Media gateways are logical connections that are a combination of DSP resources, media channels,
and voice bus paths that provide protocol translation between IP telephones and trunks and analog
and digital telephony devices.
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Chapter 5 Configuring application resources85
Rules for managing the resources
The following rules are provided to assist you in configuring your resources.
•“Signaling channel rules”
•“Media channel rules” on page 85
•“DSP resources rules” on page 86
•“Voice bus path” on page 87
•“Media gateways” on page 87
Signaling channel rules
Signaling channels determine how many IP telephones you can connect to your system. If you
have a system that does not use IP telephones, the number of signaling channels does not affect
your configuration.
•The total number signaling channels available depends on the DS30 split you have configured.
For information about how to view and change the DS30 split, refer to “Changing the DS30
split” on page 98.
If you have a 2/6 DS30 split, the total number of signaling channels is 64.
If you have a 3/5 DS30 split, the total number of signaling channels is 96.
•Management functions use six signaling channels.
•Dial-on-Demand ISDN WAN uses 27 signaling channels.
All 27 signaling channels are used, regardless of the number of WAN channels configured.
•Voice Mail requires one signaling channel for each voice mail port enabled. You can enable up
to 32 voice mail ports.
Both voice mail and contact center use Voice Mail ports.
•IP Telephony clients require one signaling channel for each IP telephone connected to the
system.
•IP Telephony trunks require one signaling channel.
Only one signaling channel is required regardless of the number of IP Telephony trunks
enabled.
•IVR requires 1 signaling channel for each enabled IVR port.
•Up to 24 ports enabled. Maximum of 32 ports between IVR and voice mail.
Media channel rules
The media channels are used to transport voice and data signals between devices.
•Management functions use five media channels. These five channels are reserved for
management functions and are always in use.
•Dial-on-Demand ISDN WAN uses 27 media channels.
All 27 media channels are used, regardless of the number of WAN channels configured. The
maximum number of WAN channels is 16.
•Voice Mail and contact center use one media channel for each active session.
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86Chapter 5 Configuring application resources
•A call between an IP telephone and a digital or analog telephone or a PSTN line uses a media
channel for the duration of the call.
•A call from a digital or analog telephone that uses an IP trunk uses a media channel for the
duration of the call.
•A call between two IP telephones on the same BCM uses a media channel during call setup.
After the call is established, the media channel is released.
•A call on an IP telephone using an IP trunk uses a media channel during call setup. After the
call is established, the media channel is released.
•IVR needs 1 media channel for each active session.
Since most of the devices do not use media channels all of the time, your system can have more
devices than there are media channels. However, to ensure you have sufficient system resources,
make sure the number of media channels you have exceeds your estimate of peak media channel
usage. The section below provides an example of how to estimate your peak media channel usage.
Example of how to estimate peak media channel usage
The example below is for a fictional company named CompanyABC. The numbers used are
strictly for this example. Actual numbers will vary, depending on the company. When you are
estimating your peak media channel usage, make sure you use numbers that reflect your business.
•CompanyABC has a BCM system with 96 telephones. Of these telephones, 48 are digital
telephones and 48 are IP telephones.
The percentage of IP telephones is 50% (48/96). This percentage is used to estimate how many
calls will be made between IP telephones and digital telephones.
•In CompanyABC, the users are typically on the telephone 15 minutes out of each hour, or 25%
of the time. During peak hours, the users are on the telephone 30 minutes, or 50% of the time.
Therefore, the peak usage of IP telephones is 24 (50% X 48 IP telephones).
•In CompanyABC, half of the calls are made to external destinations and half of the calls are
made within the BCM system. CompanyABC does not have IP trunks, so the calls from the IP
telephones to external destinations must use PSTN lines.
The peak number of IP telephone calls that use PSTN lines is 12.
(50% of calls external X 24 IP telephones during peak usage.)
•For internal calls, there is a 50% chance the call is made to a digital telephone.
The peak number of IP telephone calls to digital telephones is 6.
(50% of calls internal X 24 IP telephones peak usage X 50% number of digital telephones.)
•The peak media channel usage for IP telephony is 18.
(12 media channels for external calls and 6 for calls made to digital telephones.)
DSP resources rules
The number of DSP resources you have depends of the number and type of MS-PECs that are
installed.
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Chapter 5 Configuring application resources87
For the purposes of calculating DSP resources, we can estimate the relative power of each
configuration as follows:
•4 MS-PEC I 24 units
•2 MS-PEC III 64 units
•4 MS-PEC III128 units
The number of DSP resources you need depends on the features and type of codec you are using.
Refer to Appendix C, “Codec rates,” on page 721.
•Dial-on-Demand WAN uses 1 unit for each 64Kbit/s channel.
•Voice Mail, IVR, and contact center use 1 unit for each active session.
•Fax uses 6 units for each active fax channel.
•IP telephone or IP trunk using G.711 codec uses 1 unit.
•IP telephone or IP trunk using G.729 codec uses 3 units.
•IP telephone or IP trunk using G.723 codec uses 4 units.
Note: Some of the DSP resource units in the preceding list are rounded to the nearest
whole number. This is done to ease the calculation of the DSP resources you require. To
calculate more accurate DSP requirements, use the DSP resource units in shown in the
following table.
Table 12 DSP resource requirements
Feature or codecResource units on an MS-PEC IResource units on an MS-PEC III
G.72932.75
G.72344.2
Fax56
T. 3 8 I P F a x56
IVR Fax66
Voice bus path
There are 62 voice bus paths available on BCM.
•Voice mail and IVR use one voice bus path for each active session.
•Dial-on-Demand WAN uses one voice bus path for each 64Kbit/s channel that is active.
•IP telephones and IP trunks require one voice bus path when ever a media channel is required.
Media gateways
One media gateway is required for each call:
•from an IP telephone to an analog or digital telephone
•from an IP telephone using a PSTN line
•from an analog or digital telephone using an IP trunk
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88Chapter 5 Configuring application resources
Determining the resources you require
The following questions are designed to help determine how many resources you require. Based
on the answers to these questions, you can calculate the number of signaling channels, media
channels, voice bus paths, and DSP resource units you need. Use the table in “Evaluation” on
page 94 to determine the configurations.
Note: In the following questions, “peak periods” refers to the periods of time when there
is the highest overall activity. It is necessary to consider the resource requirements for
“peak periods” to determine if available voice bus paths and DSP resources meet your
resource requirements at all times.
ISDN WAN (Dial-up/Nailed-up)
As you answer the following questions, record your answers in the table in “Record of required
resources” on page 93.
1What is the maximum required WAN bandwidth?
The range is 0 to 1 Mbit/s (16 x 64 kbit/s) in 64 kbit/s increments.
If the answer is more than zero:
•Add 27 to the signaling channel count.
•Add 27 to the media channel count.
2What is the required WAN bandwidth during peak periods?
The range is 0 to the maximum bandwidth you entered in question 1.
For each 64 kbit/s of bandwidth:
•add 1 to the voice bus time slot count
•add 1 to the DSP resource unit count
Voice mail and CC
3What is the maximum number of voice mail ports required? Voice mail ports are used for
voice mail and Contact Center.
The range is 0 to 32 ports.
For each voice mail port:
•add 1 to the signaling channel count
•add 1 to the media channel count
4What is the number of Voice mail ports required during peak periods?
The range is 0 to the maximum number of ports selected in question 3.
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For each voice mail port
•add 1 to the voice bus path count
•add 1 to the DSP resource unit count
Chapter 5 Configuring application resources89
5How many fax tasks will be used during peak periods?
The range is 0 to 2.
For each fax task:
•add 6 to the DSP resource unit count
Note: The maximum number of voice ports shared between voice mail and IVR is 32.
The maximum number of ports shared between voice mail, IVR, and T.38 IP fax is
eight. There are only 2 fax ports.
Note: The fax DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
IVR and IVR Fax
6What is the maximum number of IVR ports required? IVR ports are used for interactive voice
response applications.
The range is 0 to 24 ports.
For each voice mail port:
•add 1 to the signaling channel count
•add 1 to the media channel count
7What is the number of IVR ports required during peak periods?
The range is 0 to the maximum number of ports selected in question 6.
For each voice mail port:
•add 1 to the voice bus path count
•add 1 to the DSP resource unit count
8How many fax tasks will be used during peak periods?
The range is 0 to max.
For each fax task:
•add 6 to the DSP resource unit count
Note: The maximum number of voice ports shared between voice mail and IVR is 32.
The maximum number of ports shared between voice mail, IVR, and T.38 IP Fax is
eight. There are only 2 fax ports.
Note: The fax DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
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90Chapter 5 Configuring application resources
IP telephones
9What is the maximum number of IP telephones required?
The range is 0 to 90 IP telephones.
For each IP telephone:
•add 1 to the signaling channel count
10 How many IP telephones will be calling an analog or digital telephone or using a PSTN trunk
during peak periods?
The range is 0 to the maximum number of IP telephones selected in question 9.
For each IP telephone:
•add 1 to the media channel count
•add 1 to the voice bus path count
11 How many IP telephones specified in question 10 will be using the G.711 codec?
The range is 0 to the maximum number of IP telephones selected in question 10.
For each IP telephone:
•add 1 to the DSP resource unit count
12 How many IP telephones specified in question 10 will be using the G.729 codec?
The range is 0 to the maximum number of IP telephones selected in question 11.
For each IP telephone:
•add 3 to the DSP resource unit count
Note: The G.729 DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
13 How many IP telephones specified in question 10 will be using the G.723 codec?
The range is 0 to the maximum number of IP telephones selected in question 11.
For each IP telephone:
•add 4 to the DSP resource unit count
Note: The G.723 DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
IP Trunks
14 What is the maximum number of IP trunks required?
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The range is 0 to 60 IP trunks.
Chapter 5 Configuring application resources91
If the number is greater than zero IP trunks:
•add 1 to the signaling channel count
15 How many analog or digital telephones (not IP telephones) will use IP trunks during peak
periods?
The range is 0 to the maximum number of IP trunks selected in question 14.
For each IP trunk:
•add 1 to the voice bus path count
•add 1 to the media channel count
16 How many IP trunks specified in question 15 will be using the G.711 codec?
The range is 0 to the maximum number of IP trunks selected in question 15.
For each IP trunk:
•add 1 to the DSP resource unit count
17 How many IP trunks specified in question 16 will be using the G.729 codec?
The range is 0 to the maximum number of IP trunks selected in question 16.
For each IP trunk:
•add 3 to the DSP resource unit count
Note: The G.729 DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
18 How many IP trunks specified in question 16 will be using the G.723 codec?
The range is 0 to the maximum number of IP trunks selected in question 16.
For each IP trunk:
•add 4 to the DSP resource unit count
Note: The G.723 DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
19 How many T.38 fax tasks will be used during peak periods?
The range is 0 to 8.
For each fax task:
•add 6 to the DSP resource unit count
Note: The maximum number of ports shared between voice mail, IVR, and T.38 IP
Fax is eight. There are only 2 fax ports.
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92Chapter 5 Configuring application resources
Note: The fax DSP resource unit count is rounded to ease calculations. For a more
accurate DSP resource unit count, refer to the table in “DSP resources rules” on
page 86.
Note: If the source or destination of the T.38 IP Fax can be fax mail or IVR, the fax
message requires two fax tasks (12 units). One fax task handles the IP Fax portion of
the transmission, and the other task handles the IVR or fax mail portion of the
transmission.
Note: To use T.38 Fax, you must have 2 or 4 MS-PEC III installed in your MSC card.
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Chapter 5 Configuring application resources93
Record of required resources
Use Table 13 to record the resources you require for your BCM system. To determine the
resources that you require, answer the questions in “Determining the resources you require” on
page 88.
Table 13 Required resources
Question
1. WAN------
2. Peak WAN------
3. VM/ACD------
4. IVR
5. Peak VM/ACD------
6. Peak FAX---------
7. Peak IVR
8. IVR FAX
9. IP Sets---------
10. Peak IP Sets------
11. IP Sets G711---------
12. IP Sets G729---------
13. IP Sets G723---------
14. IP Trunks---------
15. Peak IP Trunks------
16. IP Trunks G.711---------
17. IP Trunks G.729---------
18. IP Trunks G.723---------
19. IP Trunks T.38 Fax---------
To t al s
Answer
Signaling
channels
Media
channels
Voice bus
paths
DSP resource
units
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94Chapter 5 Configuring application resources
Evaluation
After you answer the questions and calculate the four totals, use the following rules to determine
the required BCM configuration.
Table 14 Evaluation of required BCM configuration
ResourceNumber required Required configuration
58 or less2/6 DS30 split
Signaling channel count
Media channel count
Voice bus path count
DSP resource units
59 to 903/5 DS30 split
91 or moreexceeds BCM capacity
58 or less2/6 DS30 split
59 to 903/5 DS30 split
91 or moreexceeds BCM capacity
62 or lesswithin BCM capacity
63 or moreexceeds BCM capacity
1 to 244 MS PEC I
1 to 642 MS PEC III
65 to 1284 MS PEC III
129 or moreexceeds BCM capacity
Note: If your system requires more resources than are available on your MS-PEC
configuration, you can upgrade your MS-PECs. For information about how to upgrade
your MS-PECs, refer to the BCM200/400 4.0 Installation and Maintenance Guide
(N0060612).
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Chapter 5 Configuring application resources95
Example of a BCM configuration
The following two tables provide examples of required configurations.
Table 15 Example of required configuration
Signaling
QuestionAnswer
1. WAN512 kbit/s (8)2727------
2. Peak WAN512 kbit/s (8)------88
3. VM/ACD888------
4. IVR
5. Peak VM/ACD6------66
6. Peak IVR
7. Peak FAX1---------6
8. IVR FAX
9. IP Sets2424---------
10. Peak IP Sets12---1212---
11. IP Sets G7116---------6
12. IP Sets G7294---------12
13. IP Sets G7232---------8
14. IP Trunks321---------
15. Peak IP Trunks20---2020---
16. IP Trunks G.71112---------12
17. IP Trunks G.7296---------18
18. IP Trunks G.7232---------8
19. IP Trunks T.38 Fax---------
Totals---60674684
channels
Media
channels
Voice bus
paths
DSP resource
units
Table 16 Evaluation for the example of required configuration
ResourceNumber required Recommended configuration
Signaling channel count603/5 DS30 split
Media channel count673/5 DS30 split
Voice bus path count46within BCM capacity
DSP resource units844 MS-PEC III
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96Chapter 5 Configuring application resources
Understanding the minimum and maximum values
The MSC Configuration allows you to determine how the resources are assigned on your BCM.
In some BCM systems, the total number of features and devices that require resources exceeds the
number of resources that are available. To address this issue, BCM allows you to share the
resources. By changing minimum and maximum values for each component you can fine tune this
sharing.
Minimum
The minimum value is the number of resources that are always assigned to a component. You use
this number to ensure a base level of service for a specific component. For example, to ensure that
at least four people can be using voice mail at all times, you would enter four as a minimum value
for the Voice Port component.
The resources that are not assigned using the minimum values are shared by the components. If a
component needs additional resources, it can use some of the shared resources to provide service
during the busy period. This method of sharing resources allows your BCM system to adapt to the
changing demands for services.
Maximum
The maximum value is the maximum number of resources that a component can use. You use this
number to ensure a single component does not consume all of the shared resources.
The MSC configuration you choose greatly affects the performance of your BCM system. Make
sure you consider the needs of your users, including peak usage times, when selecting the
minimum and maximum values. Table 17 describes the advantages and disadvantages of changing
these values.
Table 17 Advantages and Disadvantages of Minimum and Maximum values (Sheet 1 of 2)
ValueAdvantageDisadvantage
Increasing
Minimum Value
Decreasing
Minimum Value
Increasing
Maximum Value
Increases the guaranteed level of service for a
component. The DSP resources you assign as a
Minimum are always available to the users of
this component.
More DSP resources are available to share with
other components. When there is a large pool of
shared DSP resources, BCM more readily
adapts to changing component use.
Allows this component to use more of the
shared DSP resources during times of peak use.
This allows more people to use this component
at the same time.
Decreases the flexibility of DSP resource
sharing. DSP resources that are assigned to the
Minimum value are not shared with other
components. If you set the Minimum level too
high, other components may not be available
due to a lack of available DSP resources.
Lower guaranteed level of service for this
component. If the Minimum value is too low, it is
possible that some users will not be able to
access this component when other components
are in heavy use.
During times of peak use, this component may
consume all of the shared resources. This may
cause other components to be unavailable to
users.
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Chapter 5 Configuring application resources97
Table 17 Advantages and Disadvantages of Minimum and Maximum values (Sheet 2 of 2)
ValueAdvantageDisadvantage
Decreasing
Maximum Value
Prevents this component from using so many of
the shared DSP resources, that other
components are unavailable.
Limits the number of people that can use this
component even if sufficient DSP resources are
available.
Application Resource Reservations
Table 18 describes each component on the MSC card.
Table 18 MSC custom configuration parameters (Sheet 1 of 2)
ComponentDescription
IP ClientsIP Clients are Nortel IP telephones.
DSP resources are required only when the IP telephone is in use (for example, to make a call,
receive a call, listen to voice mail).
For information about how to configure IP clients, refer to the BCM 4.0 Telephony Device Installation Guide (N0060609)
Note: The codec (G.711, G.723 or G.729) you are using for the IP Client affects how many IP
clients you can use on your system.
IP TrunksIP Trunks are communication channels that BCM uses to send and receive IP telephony calls
using the Public Data Network. You can use IP trunks to connect your BCM system to:
•another BCM system
•a Meridian 1 IPT system
For information about how to configure IP trunks, refer to the BCM 4.0 Telephony Device
Installation Guide (N0060609).
Note: The codec (G.711, G.723 or G.729) you are using for the IP Trunk affects how many IP
Trunks you can use on your system.
Media GatewaysMedia Gateways provide the connection between IP telephony devices (IP trunks, and Nortel IP
Voice Mail +
contact center
FaxFax ports are communication channels that connect a fax machine to the BCM.
WANWAN channels are dial-up ISDN WAN connections.
telephones) and normal telephony devices (PSTN lines; 74XX, 7316E, 7316, 7208s, 7100, 7000
digital phones; analog telephones etc.).
Voice Mail and contact center ports are communication channels that connect users to the
CallPilot Voice Mail and Contact Center Software.
DSP resources are required only when a user connects to voice mail or Contact Center. This
includes callers hearing greetings, callers leaving messages, and users accessing their
mailboxes.
The minimum value for Voice Mail and Contact Center Ports must be 2 or higher, unless you want
to disable CallPilot Voice Mail and Contact Center Software.
The maximum value for Voice Mail and Contact Center Ports must be 2 or higher, unless you want
to disable CallPilot voice mail and Contact Center Software.
To disable CallPilot voice mail and Contact Center Software, change the minimum and maximum
values for Voice Mail and Contact Center Ports to zero.
Fax mail ports are communication channels that connect a fax machine to a fax mailbox or a user
to a Fax-on-Demand mailbox.
IVR fax ports are communication channels that connect a fax machine to IVR functions.
T.38 IP Fax ports are communication channels that connect to a fax machine that is using an IP
trunk.
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98Chapter 5 Configuring application resources
Table 18 MSC custom configuration parameters (Sheet 2 of 2)
ComponentDescription
IVR PortsIVR ports are communication channels that connect users to the IVR Software.
DSP resources are required only when a user connects to IVR. This includes callers hearing
greetings, callers leaving messages, and users accessing their mailboxes.
The minimum value for IVR Ports must be 2 or higher, unless you want to disable IVR.
The maximum value for IVR Ports must be 2 or higher, unless you want to disable IVR.
To disable IVR, change the minimum and maximum values for IVR Ports to zero.
CTE PortsCTE ports are communication channels that connect CTE applications to the BCM. An example
of a CTE application is BCM Personal Call Manager.
Changing the DS30 split
A DS30 bus is a group of 32 signaling channel and 32 media channels. The DS30 split determines
how these channels are assigned on BCM.
You have a choice of a 2/6 or a 3/5 split. If you choose a 2/6 split, two DS30 buses are assigned to
the MSC and six are assigned to the Media Bay Modules. If you choose a 3/5 split, three DS30
buses are assigned to the MSC and five are assigned to the Media Bay Modules.
The split you choose is determined by the number of signaling channels you require for
applications such as voice mail, IVR, IP trunks, IP telephones, and dialup ISDN WAN
connections. If you need 58 signaling channels or less for these applications, use a 2/6 DS30 split.
If you need 59 signaling channels or more, use a 3/5 DS30 split.
If your signaling channel requirements change, for example you want to increase the number of IP
telephones, you can change from a 2/6 setting to a 3/5 setting without losing data. All new records
added after the update will reflect the new default settings. To determine what the channel
requirements are, refer to “Determining the resources you require” on page 88.
Warning: Ensure that the system is idle before you perform this procedure. You must
restart the system after you have changed the setting.
Note: Ensure you have a current backup before you perform this procedure.
Note: You must ensure that your system has adequate DSP resources to support an
increase in voice processing traffic. To determine if you have enough DSP resources, refer
to “Determining the resources you require” on page 88. If you need to add MS-PEC IIIs,
refer to the BCM200/400 4.0 Installation and Maintenance Guide (N0060612) for
installation instructions. Refer to the BCM sales catalogue for part numbers and ordering
instructions.
N0060606N0060606
Chapter 5 Configuring application resources99
Warning: If you choose to change the DS30 split of your system after you configure your
system, you could risk losing data for both the core system and optional applications.
Make sure you understand the implications of the changes before you go forward with this
procedure.