Nortel Communication Server 1000
Main Office Configuration Guide
for SRG 50
NN43001-307
.
Download from Www.Somanuals.com. All Manuals Search And Download.
3
Revision history
December 2007
Standard 02.02. This document is up-issued to support CS 1000 Release
5.0 for SRG 50 Release 3.0. This document includes SIP Trunks
configuration at the main office.
August 2007
Standard 02.01. This document is up-issued to support CS 1000 Release
5.0 for SRG 50 Release 3.0.
June 2007
Standard 01.02. This document is up-issued to remove the Nortel Networks
Confidential statement.
May 2007
Standard 01.01. This document is up-issued to support Communication
Server 1000 Release 5.0. This document contains information previously
contained in the following legacy document, now retired: (553-3001-207).
This document is up-issued to include updated information due to CR
October 2006
Standard 3.00. This document is up-issued to support SRG 50 Release
2.0 for CS 1000 Release 4.5.
January 2006
Standard 2.00. This document is up-issued for CR Q01202736, with
information on reconfiguring Call Server alarm notification levels if
necessary when configuring Adaptive Network Bandwidth Management.
August 2005
Standard 1.00. This document is a new document to support
Communication Server 1000 Release 4.5.
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
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4
Revision history
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
Download from Www.Somanuals.com. All Manuals Search And Download.
5
Contents
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
Download from Www.Somanuals.com. All Manuals Search And Download.
6
Contents
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
Download from Www.Somanuals.com. All Manuals Search And Download.
Contents
7
Procedures
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
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8
Contents
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
Download from Www.Somanuals.com. All Manuals Search And Download.
9
New in this release
The CS 1000 Main Office Configuration for SRG50 (NN43001-307) for CS
1000 Release 5.0 includes support for SRG 50 Release 3.0.
The following sections detail what is new in CS 1000 Main Office
Configuration for SRG50 (NN43001-307) for CS 1000 Release 5.0 .
Other
CS 1000 Main Office Configuration for SRG50 (NN43001-307) includes
the following changes:
•
•
•
•
•
Replaced instances of OTM 2.2 with TM 3.1.
Updated Element Manager with enhancements.
Added support for new IP Phones.
Removed instances of CS 1000S and Small Systems.
Removed instances of Terminal Numbers (TN) in "c u" format. Only TN
in "l s c u" format are supported.
Subject
This document describes the CS 1000 Main Office Configuration for
SRG50 (NN43001-307) for software Release 3.0 for CS 1000 Release 5.0.
Information in this document complements information found in documents
in the Communication Server 1000 documentation suite. For information
about how to configure the SRG 50, see SRG50 Configuration Guide
Documentation > Communication Servers > BCM.
Intended audiences
This document is intended for individuals responsible for configuring the
main office for Survivable Remote Gateway for organizations using CS
1000 systems.
Related information
This section lists information sources that relate to this document.
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
Copyright © 2005-2007, Nortel Networks
.
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10 New in this release
NTPs
The following NTPs are referenced in this document:
•
•
Converging the Data Network with VoIP (NN43001-260)
Electronic Switched Network Reference—Signaling and Transmission
(NN43001-280)
•
•
•
•
•
•
•
•
•
•
•
•
Dialing Plans Reference (NN43001-283)
Signaling Server Installation and Commissioning (NN43001-312)
IP Peer Networking Installation and Commissioning (NN43001-313)
Branch Office Installation and Commissioning (NN43001-314)
Telephony Manager 3.1 System Administration (NN43050-601)
Software Input Output Administration (NN43001-611)
Emergency Services Access Fundamentals (NN43001-613)
Element Manager System Reference Administration (NN43001-632)
IP Line Fundamentals (NN43100-500)
ISDN Primary Rate Interface Fundamentals (NN43001-569)
Basic Network Feature Fundamentals (NN43001-579)
Communication Server 1000M and Meridian 1 Small System Planning
and Engineering (NN43011-220)
•
Communication Server 1000M and Meridian 1 Large System Planning
and Engineering (NN43021-220)
•
•
•
Communication Server 1000E Planning and Engineering (NN43041-220)
Software Input Output Reference—Maintenance (NN43001-711)
SRG50 Configuration Guide (NN40140-500)
Online
To access Nortel documentation online, click the Technical Documentation
link under Support & Training on the Nortel home page:
CD-ROM
To obtain Nortel documentation on CD-ROM, contact your Nortel customer
representative.
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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.
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11
Description
Contents
This section contains information about the following topics:
•
•
•
•
•
•
•
•
Survivable Remote Gateway
The Survivable Remote Gateway (SRG) extends the desktop feature and
user interface of the CS 1000 to remote IP branch office users and gives
them full access to the same applications as the main site. CallPilot, Contact
Center Management Server (CCMS), and other central applications are
shared by remote users to deliver state-of-the-art features and functionality
to small remote offices.
SRG 50 Release 2.0 provides the following:
•
•
extends the supported number of survivable IP users from 32 to 80
extends support for the IP Phone 1120E, IP Phone 1140E, IP Audio
Conference Phone 2033, and WLAN 2212
IP Phones.
•
•
supports H.323 and SIP Trunking to the CS 1000 main office
supports analog devices, such as fax machines and terminals but are
limited in number and limited to basic access
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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12 Description
SRG 50 Release 3.0 provides the following:
•
extends support for the IP Phone 1110
IP Phones.
•
•
evolves the SIP trunk to support a standard SIP Trunk interface
supports On-site Notification
Table 1
Supported software at the branch office
IP branch office
solution
Feature description
Survivable
users
Server
support
up to 90
SRG 1.0
Succession 3.0
CS 1000
Release 4.0
CS 1000
VoIP and Application Gateway
Local Mode = Basic telephony
features
Release 4.5
up to 32
SRG 50 Release 1.0
Succession 3.0
CS 1000
Release 4.0
CS 1000
VoIP and Application Gateway
Local Mode = Basic telephony
features
A more cost effective small branch
office solution.
Release 4.5
Provides H.323 trunking.
For more information, see CS
1000 Main Office Configuration
Guide for SRG 50 (553-3001-207).
up to 90
SRG200/400 Release
1.5
Succession 3.0
CS 1000
Release 4.0
CS 1000
Release 4.5
CS 1000
VoIP and Application Gateway
Local Mode = Basic telephony
features
Feature Parity with SRG 50,
new OS, and extended IP Phone
support.
Release 5.0
Provides H.323 trunking.
For more information, see Main
Office Configuration Guide for
SRG 200/400 Release 1.5
(NN43001-308).
up to 80
SRG 50 Release 2.0
Succession 3.0
CS 1000
Release 4.0
CS 1000
Release 4.5
CS 1000
Release 5.0
VoIP and Application Gateway
Local Mode = Basic telephony
features
Extends IP Phone support and
survivable IP users from 32 to 80.
Provides H.323 and SIP trunking.
For more information, see CS
1000 Main Office Configuration
Guide for SRG 50 (553-3001-207).
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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.
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Survivable Remote Gateway 13
IP branch office
solution
Feature description
Survivable
users
Server
support
up to 80
SRG 50 Release 3.0
CS 1000
Release 5.0
VoIP and Application Gateway
Local Mode = Basic telephony
features
Extends IP Phone support to
include the IP Phone 1110.
Supports On Site Notification for
E-911 calls.
Provides H.323 and SIP trunking.
up to 400
MG 1000B
MG 1000E
Succession 3.0
CS 1000
Release 4.0
CS 1000
Release 4.5
CS 1000
Release 5.0
100% CS 1000 feature and
application redundancy in
survivable mode. Designed and
positioned for larger IP branch
offices.
Provides H.323 and SIP trunking.
up to 400
CS 1000
Release 5.0
Provides survivability with the
addition of Call Processor Pentium
Mobile (CP PM).
The SRG is implemented on a BCM 50 platform and is connected to a CS
1000 at the main office through Virtual Trunks over a reliable IP WAN access
facility. This configuration allows the call processing for the IP Phones at
the SRG site to be centralized at the main office. The Call Server at the
main office provides the call processing for the IP Phones registered to
both the main office and branch offices. The SRG provides call processing
functionality to phones in local mode and local analog devices. The SRG
supports business continuity and call failover through digital and analog
trunk access to the local Public Switched Telephone Network (PSTN).
In order for devices in the CS 1000 network to access analog devices at
the SRG or to access the PSTN at the SRG, virtual trunks are used over
the LAN/WAN.
If the main office fails to function, or if there is a network/WAN outage, the
SRG automatically switches to Local mode and provides basic telephony
service to the phones located at the branch office. This enables the IP
Phones to survive the outage between the branch office and the main office.
To ensure proper operation of the SRG solution it must be configured to
support a common dialing plan with the CS 1000 main office. Any other
configuration is not guaranteed to work reliably. Since the Call Server and
the SRG handle dialing slightly differently, ensure that any settings you use
for the main office that need to interact with the SRG, can be accommodated
by the SRG call processing.
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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.
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14 Description
Figure 1 "SRG network" (page 14) shows the networking among the main
office, SRG, and IP Phones.
Figure 1
SRG network
Main office hardware description
The main office must be one of the following systems:
CS 1000E
CS 1000M Cabinet
•
•
•
•
•
•
CS 1000M Chassis
CS 1000M HG
CS 1000M SG
CS 1000M MG
Throughout this document, references to CS 1000 systems encompass
all CS 1000 system types.
The diagrams throughout this documentation show a CS 1000E main
office. All of the systems appearing in the list perform identical main office
functions as far as the SRG is concerned. For information about the SRG,
see SRG50 Configuration Guide (NN40140-500).
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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Main office hardware description 15
Signaling Server
The following Signaling Servers are available for CS 1000 Release 5.0 :
•
•
•
•
ISP1100
HP-DL320-G4
IBM-X306m
Common Processor Pentium Mobile (CP PM)
The Signaling Server is required at the main office only. It provides the
following functions:
•
Terminal Proxy Server (TPS)
The TPS provides a connection from the IP Phones to the Call Server
and a connection from a Virtual Trunk to the Call Server.
•
•
•
H.323 Gateway (Virtual Trunk)
SIP Gateway (Virtual Trunk)
CS 1000 Element Manager Web Server and Network Routing Service
(NRS)
•
NRS, consisting of:
— SIP Redirect Server NRS
— H.323 Gatekeeper
— Network Connection Service (NCS)
•
Personal Directory
A second Signaling Server can be used to provide redundancy in the case
of a failure in the primary Signaling Server at the main office.
A similar function to the Signaling Server is used at the SRG when the
phones are in local mode.
The Signaling Server supports en bloc signaling which is standard on the
Signaling Server.
For more information about the Signaling Server, see Signaling Server
Installation and Commissioning (NN43001-312). For more information
about H.323 and overlap signaling, see IP Peer Networking Installation
and Commissioning (NN43001-313).
Network Routing Services
The Network Routing Service (NRS) application provides network-based
•
routing, combining the following into a single application:
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
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16 Description
•
•
H.323 Gatekeeper— provides central dialing plan management and
routing for H.323-based endpoints and gateways.
SIP Redirect Server NRS — provides central dialing plan management
and routing for SIP-based endpoints and gateways. SIP Trunks are
used for Voice packet traffic alone.
•
•
•
NRS Database— stores the central dialing plan in XML format for the
H.323 Gatekeeper, and the SIP Redirect Server. The H.323 Gatekeeper
and the SIP Redirect Server accesses this common endpoint and
gateway database.
Network Connect Server (NCS) — used only for Media Gateway
Controller (MGC) based MG 1000B, SRG, Geographic Redundancy,
and Network-wide Virtual Office solutions. The NCS allows the Line
TPS (LTPS) to query the NRS.
NRS Manager web interface— the NRS provides its own web interface
to configure the H.323 Gatekeeper, SIP Redirect Server, and the NCS.
The NRS application provides routing services to H.323 devices and
SIP-compliant devices. The H.323 Gatekeeper can be configured to
support H.323 routing services, while the SIP Redirect Server NRS can be
configured to support SIP routing services. The H.323 Gatekeeper and the
SIP Redirect Server NRS can reside on the same Signaling Server.
Each system in an IP Peer network must register to the NRS. The NRS
software identifies the IP addresses of systems based on the network-wide
numbering plan. NRS registration eliminates the need for manual
configuration of IP addresses and numbering plan information at every site.
When configuring the NRS it is necessary to enable the NCS. Ensure that
the check box “Network Connection Server enabled” is checked in the NRS
configuration window of CS 1000 Element Manager.
For information about configuring the NRS, see IP Peer Networking
Installation and Commissioning (NN43001-313).
Supported IP Phones
for each software release.
Table 2
IP Phone support
IP Phone
Release 1.0
Release 2.0
Release 3.0
IP Phone 2001
Supported
Supported
Supported
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
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Main office requirements 17
IP Phone 2002
Supported
Supported
Supported
Supported
Supported
Supported
Supported
Supported
Supported
Supported
Supported
IP Phone 2004
Supported
IP Phone 2007
Supported
IP Audio Conference Phone 2033
IP Softphone 2050
IP Phone 1110
Supported
Supported
Not supported
Not supported
Not supported
Not supported
Supported
Not supported
Supported
Supported
Supported
Supported
Not supported
Supported
Supported
Supported
Supported
Supported
Not supported
Not supported
IP Phone 1120E
IP Phone 1140E
Supported
IP Phone 1150E
Not supported
Supported
Mobile Voice Client (MVC) 2050
Analog (500/2500-type) telephones Supported
Supported
WLAN Handset 2210
WLAN Handset 2211
WLAN Handset 2212
WLAN Handset 6120
WLAN Handset 6140
Supported
Supported
Supported
Supported
Not supported
Not supported
Not supported
Supported
Not supported
Not supported
Throughout this document, the IP Phones are referred to collectively as
IP Phones.
Main office requirements
The branch office running SRG Release 3.0 requires the following at the
main office:
•
CS 1000 hardware, running CS 1000 Release 4.0, CS 1000 Release
4.5, orCS 1000 Release 5.0 .
•
Configure at least one of the following packages for IP Peer Networking:
— H.323 Virtual Trunk (H323_VTRK) package 399
— SIP Gateway and Converged Desktop Package (SIP) package 406
•
•
The main office must have a software Service Level of 2 or higher to
work with the branch office.
Ensure that you have ordered enough IP user and Virtual Trunk licenses
at the main office to support the SRG 50 or the capacity of your branch
office.
The two different IP user licenses at the main office are:
— Basic IP License for the IP Phone 2001, IP Audio Conference Phone
2033, and IP Phone 1110
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Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
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18 Description
— IP User License for the IP Phone 2002, IP Phone 2004, IP Phone
2007, IP Phone 1120E, IP Phone 1140E, IP Softphone 2050, Mobile
Voice Client (MVC) 2050, WLAN Handset 2210, WLAN Handset
2211, and WLAN Handset 2212
The main office requires the following software packages to support
the specified Basic Network features. See Basic Network Feature
Fundamentals (NN43001-579) for more information about these features.
•
Network Call Back Queuing (MCBQ) package 38. This package is
required for SRG IP Phones to invoke any queuing feature or ringback
when free.
•
Network Speed Call (NSC) package 39. This package is required for
SRG IP Phones to invoke the Network Speed Call feature.
The main office requires the following software packages to support the
specified ISDN Primary Rate Interface features. See ISDN Primary Rate
Interface Fundamentals (NN43001-569) for more information about these
features.
•
Network Attendant Service (NAS) package 159. This package is
required for analog (500/2500-type) telephones in the branch office
to access attendant services when the attendant is configured on the
main office.
•
Network Message Services (NMS) package 175. This package is
required for analog (500/2500-type) telephones in the branch office to
share the voice mail system in the main office. For any configurations
using centralized CallPilot on the main office with one or more branch
offices in separate time zones, the NMS package is required at the main
office for the branch IP Phones.
Optional features to enhance SRG functionality
•
Network Alternate Route Selection (NARS) package 58. See Basic
Network Feature Fundamentals (NN43001-579).
•
Overlap Signaling (OVLP) package 184. This package is optional; it is
required for overlap signaling. It is packaged with H.323 Virtual Trunk
(H323_VTRK) package 399.
•
•
Emergency Services Access (ESA) package 329. This package is
optional; it is required only to receive 911/ESA features in North
American and some Caribbean and Latin American (CALA) markets.
See Emergency Services Access Fundamentals (NN43001-613).
Virtual Office (VIRTUAL_OFFICE) package 382. This package is
optional; it is required only for Virtual Office functionality.
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
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Normal Mode and Local Mode overview 19
•
Network Signaling (NSIG) package 37. This package is optional for
SRG IP Phones to access set-based Network Class of Service (NCOS)
features.
•
•
Adaptive Network Bandwidth Management package 407.
Alternative Call Routing for Network Bandwidth Management.
For software and hardware requirements for SRG, see SRG50 Configuration
Guide (NN40140-500).
Normal Mode and Local Mode overview
Normal Mode and Local Mode overview provides a description of the
following sections:
•
•
•
•
•
•
Normal Mode
Local Mode
Survivability
Recovery to Normal Mode
Local Mode operation
Virtual trunks
Normal Mode
IP Phones that are physically located at the SRG but are registered with the
main office are operating in Normal Mode. In Normal Mode, the main office
provides centralized call processing to all applications transparently to all
IP Phones at the Branch Office. All IP Phones at the Branch, in Normal
Mode, are registered to the main office TPS and are controlled by the Call
Server at the main office.
Users of the SRG IP Phones receive the features, applications, key
layout, and tones of the main office Call Server. This provides feature and
application transparency between the branch office and the main office.
Local Mode
Users at the branch office may be in Local Mode, or survivable mode for
two different reasons:
1. IP Phone has just booted up.
2. IP Phone cannot communicate to the main office because of a WAN
failure or a failure of the main office components.
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NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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20 Description
ATTENTION
When a telephone or trunk in the main office calls an SRG IP Phone that has
switched to Local Mode due to WAN failure, the call is treated according to the
main office call redirection configuration (such as forwarding to voice mail or
continuous ringback).
In the event that the IP Phones at the branch office lose the connection to
the main office CS 1000 call server for any reason (WAN failure, main office
call server failure, main office Signaling Server failure), the SRG 50 reverts
to Local Mode automatically. Essentially, when VoIP connectivity is lost,
each IP Phone loses its Reliable UDP (RUDP) connection with the main
office Terminal Proxy Server (TPS). The IP Phones at the branch office
reboot and reregister to the SRG 50, placing them in Local Mode.
After this occurs, the IP Phones displays an indication on the display area
that the set is in Local Mode of operation. This display is configurable by
installers to meet local language and usage norms.
In Local Mode, the IP users connected at the branch office are under the
control of the SRG 50 call services. As such, the normal main office call
server features are not available. The SRG 50 offers a basic feature set
when in Local Mode which allows IP Phones to continue to make and
receive calls internally within the branch office and over the provisioned
local PSTN interfaces. Basic services, such as transfer, last number redial,
and single key access through the PSTN to a centralized voice messaging
system are supported. Local PSTN access and local Emergency Services
access is also supported. No local applications or Business Communication
Manager features are supported in Local Mode operation.
Analog devices continue to be under the control of the SRG 50 system. It
is the intent of Local Mode to provide continued access to the PSTN for
critical calls and emergency services.
In Local Mode, since the SRG 50 handles all call processing, calls
between two IP phones at the SRG 50 are handled locally as a simple
station-to-station call. When an IP Phone initiates a local PSTN call, the
SRG 50 routes the call to a trunk that is connected to the local PSTN.
Incoming DID calls are also handled by the SRG 50 and terminated on
the appropriate IP Phone.
In the event of a WAN failure, in Local Mode, the IP Phones do not have
access to the main office network over the VoIP trunks. If the appropriate
alternate routes are configured, calls will be routed to the main office or
other branch offices using the available PSTN trunks.
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Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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Normal Mode and Local Mode overview 21
While in Local Mode, the SRG 50 system continues to monitor for a main
office CS 1000 heartbeat signal, and once detected, automatically redirects
phones on an individual basis back to Normal Mode of operation. If a
call is active, the SRG waits until the call is completed before redirecting
the phones; calls in progress are not interrupted. This switch-over occurs
almost immediately once the SRG determines that an individual phone
can be redirected. This reinstates the CS 1000 normal user interface and
feature set for the IP Phone user, on a user by user basis.
The SRG 50 system implements the same interface used by the MG 1000B
system to interact with the main office CS 1000 system. This allows the
main office to identify attached clients and the local PSTN as branch office
entities, enabling proper operation of dial plans and E911 access.
In Local Mode, devices that are physically located at the branch office, that
are controlled by the local system and receive a basic telephony feature
set, provide business continuity for the branch office during the WAN or
system failure. The SRG supports a main office heartbeat or reliable UDP
signaling which automatically reregisters users once WAN or system failure
has recovered.
For information about the features supported in Local Mode, see SRG50
Configuration Guide (NN40140-500).
Survivability
SRG is specifically designed to provide automatic survivability against WAN
failure, main office Call Server failure, main office Signaling Server failure,
and Gatekeeper failure.
SRG supports the Geographic Redundancy feature. For further information
about Geographic Redundancy, see System Redundancy Fundamentals
(NN43001-507).
In the event of a WAN failure, the SRG IP Phones lose communication
with the main office. This causes the SRG IP Phones to reset and register
with the SRG. The IP Phones then operate in Local Mode, providing
basic telephony services delivered by the local SRG system. For further
information about services and features supported on the SRG, see SRG50
Configuration Guide (NN40140-500).
If the main office Call Server fails and call processing services are provided
by an Alternate Call Server, the SRG IP Phones reset and reregister with
the Alternate Call Server and receive call processing services from it. If no
Alternate Call Server is available, the SRG IP Phones go to Local Mode
while the SRG attempts to find an Alternate Call Server by way of the NCS.
Nortel Communication Server 1000
Main Office Configuration Guide for SRG 50
NN43001-307 02.02 Standard
Release 5.0 3 December 2007
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22 Description
If the main office Signaling Server fails and an Alternate Signaling Server
is available, the SRG IP Phones reset and reregister with the SRG. The
SRG will then query the NCS for the Alternate Signaling Server IP address.
The SRG will redirect the IP Phone to the Alternate Signaling Server and
continue to receive call processing services from the main office Call Server.
If no Alternate Signaling Server is available, the SRG IP Phones reset and
register with the SRG in Local Mode.
When an IP Phone at the SRG first boots up, the IP Phone attempts
to communicate with the SRG. After communication with the SRG is
established, the SRG redirects the IP Phone to the main office. When
the SRG IP Phone attempts to register with the main office, the SRG first
queries the Primary NCS for the main office Virtual Trunk node IP address
to redirect the IP Phone. If the Primary NCS is down or unreachable, the
SRG queries the Alternate NRS (H.323 Gatekeeper/SIP Redirect Server),
if one is specified. If it receives a positive response, the SRG IP Phone is
redirected to the specified main office. Otherwise, if neither a Primary or
an Alternate NRS (H.323 Gatekeeper/SIP Redirect Server) is available,
the SRG IP Phone remains in Local Mode, and receives call processing
services from the SRG until communication can be reestablished.
SRG IP Phones in Normal Mode remain registered with the main office
if the Primary NCS fails and no Alternate NCS is available. They can
call any main office telephone or IP Phones in Normal Mode in other
branch offices. However, they cannot call any SRG analog (500/2500-type)
telephones or any external numbers through the SRG trunks because an
H.323 Gatekeeper/SIP proxy server, which could route call properly in case
of an NRS failure, is not available.
Recovery to Normal Mode
After communication is reestablished with the main office call server, all IP
Phones at the branch office that are in Local Mode automatically redirect
and reregister to the main office and return to Normal Mode operation.
IP Phones that were busy at the time communication was reestablished
complete the call in Local Mode, and then reregister with the main office
after the call is complete.
Local Mode operation
When an SRG IP Phone is in Local Mode, the user has full access to
the services configured at the SRG (analog devices or analog or digital
trunks) and to other IP Phones registered to the SRG. In Local Mode,
the IP Phones can make local calls to other IP Phones and other analog
(500/2500-type) telephones at the branch office. They can also be used to
make outgoing PSTN calls and receive incoming calls as usual. SRG IP
Phones can access the main office IP Phones or other branches by routing
through the local PSTN.
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Normal Mode and Local Mode overview 23
Testing the phone in Local Mode
From Normal Mode, the branch user has the option of going to Local Mode
manually using the Test Local Mode feature, or when the telephone is
power-cycled. The test can be performed by the user at any time and
does not require a password. This test is invoked from any IP Phone at
the branch office.
Nortel recommends testing Local Mode operation after changing the
provisioning for a telephone on the SRG.
To ensure that users do not forget to resume Normal Mode operation, the
SRG redirects the telephone to the main office to return the telephone to
Normal mode. This occurs if the telephone remains registered to the SRG
in Test Local Mode for ten minutes (default setting). Alternatively, the user
can press the Quit key on the phone to return to Normal Mode.
For further information about Local Mode functionality for SRG, see SRG50
Configuration Guide (NN40140-500).
Virtual Trunks
In order for endpoints in the CS 1000 network to access endpoints in local
mode at the SRG or to access the PSTN at the SRG, Virtual Trunks are
used over the LAN/WAN.
Virtual Trunks are software components that provide the trunking features
of the Meridian Customer-Defined Network (MCDN) feature set. Access
to PSTN digital or analog trunks at the branch office occurs through the
MCDN Virtual Trunk.
Virtual Trunks are sometimes referred to as SIP or H.323 Virtual Trunks. In
the SRG50 Configuration Guide (NN40140-500), Virtual Trunks are referred
to as IP Trunks.
For more information about Virtual Trunks, see IP Peer Networking
Installation and Commissioning (NN43001-313).
IP Phone calls
When an IP Phone calls another IP Phone, each telephone receives the
address of the other to exchange media directly between the telephones.
When in Normal Mode, an SRG IP Phone calling a main office IP Phone
does not require any trunking to set up the call. However, LAN/WAN
bandwidth is used to provide a media path for the call. For more information
on Direct IP media path functionality, see IP Peer Networking Installation
and Commissioning (NN43001-313).
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24 Description
Bandwidth Management Overview
For a complete overview of Bandwidth Management, see the Converging
the Data Network with VoIP (NN43001-260) and for details on configuration,
Network Bandwidth Management
Network Bandwidth Management allows for a limit to be placed on the
amount of interzone bandwidth allowed between IP Phones in Normal Mode
at the SRG and the rest of the CS 1000 network.
As well, it allows for the selection of interzone bandwidth codecs for calls
between the IP Phones in Normal Mode and the rest of the CS 1000
network.
Adaptive Network Bandwidth Management
Adaptive Network Bandwidth Management allows the system to dynamically
react to Quality of Service (QoS) degradation and take corrective action.
Network Bandwidth Management Zones
A zone is a collection of IP Phones that:
•
•
•
•
share similar IP bandwidth restrictions
are geographically close to one another
are all in the same time zone
are all in the same PSTN dialing plan
The Network Bandwidth Management Zone is made up of the VPNI and
the zone. The VPNI of the main office and all the SRG associated with it
must be the same.
Each SRG must have its own unique zone number and configured in the
main office Call Server and the SRG.
ATTENTION
Throughout this document, the term zone is defined as a Bandwidth Management
Time of Day
Because the SRG IP Phones, in Normal Mode, receive their clock
information from the main office, which may be located in a different time
zone, the main office must be able to provide a different time of day for
these phones.
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Bandwidth Management Overview 25
The time zone of the SRG is configured with the SRG zone at the main
office. The time zone adjusts the main office time for display at the SRG.
SRG phones then display the correct time of the SRG, rather than that of
the main office. For any configurations using centralized Call Pilot on the
main office with one or more branch offices in separate time zones, the
NMS package is required at the main office for the branch IP Phones.
SRG IP Phone to local PSTN calls
When an SRG IP Phone in Normal Mode dials a local PSTN number, the
call is processed by the main office Call Server. The dialed digits are
modified according to the dialing plan information configured in the zone for
the SRG IP Phone.
The call is configured to be routed over the Virtual Trunk to the branch
office. The SRG then tandems the call to the local PSTN. Likewise, long
distance calls can also be configured.
If you use one Access Code for both local and long distance calls, and that
Access Code is associated with a branch office zone, all calls (local and
long distance) are routed through the SRG
IP Phone to analog (500/2500-type) telephone calls
When an IP Phone in Normal Mode at the SRG calls an analog
(500/2500-type) telephone of the same SRG, the call is processed at the
main office Call Server. A Virtual Trunk route is selected according to the
digits dialed. The call is routed over a Virtual Trunk to the branch office. The
SRG processes the incoming Virtual Trunk call and terminates it to the
local analog (500/2500-type) telephone. Since this is a call between IP and
circuit-switched devices, a DSP resource on a Media Card is allocated and
connected to the analog (500/2500-type) telephone. The IP address of the
DSP resource is returned to the main office Call Server so a direct media
path between the IP Phone and the DSP resource can be set up when the
call is established. See IP Peer Networking Installation and Commissioning
(NN43001-313) for details.
Conference calls
When an SRG user initiates a conference call, the conference facilities of
the main office are used. This means that in a conference among three
SRG users, the LAN/WAN bandwidth of three media paths is used. The
calls are controlled by the main office, except in Local Mode. In Local Mode,
SRG users do not have access to conferencing.
Networking consideration
A fault condition can occur if IP Phones use a different route to the main
office than that used by the SRG.
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26 Description
CAUTION
Service Interruption
If the network is planned so that IP Phones use a different route
to the main office than that used by the SRG, a fault condition
can occur. When the SRG can reach the main office but the IP
Phone cannot ping the main office due to a network outage, an
IP Phone registration can force the telephone into a cycle of
registering locally, being redirected to the main office, rebooting,
and then registering locally again. When this cycle occurs, further
diagnose the network outage.
Capacity
Each CS 1000 main office can support up to 255 branch offices, which can
be made up of any combination SRG and MGC based MG 1000B. SRG 50
Release 2.0 and later supports up to 80 survivable IP users. However,
since all IP Phones register with the main office, the governing factor is the
maximum number of IP Phones that can be supported at the main office.
This means the total number of IP Phones in all offices can be no greater
than the capacity of the main office. See one of the following documents to
determine the total number of phones your system can support:
•
•
Communication Server 1000E Planning and Engineering (NN43041-220)
Communication Server 1000M and Meridian 1 Large System Planning
and Engineering (NN43021-220)
Virtual Trunks capacity
The SRG capacity to support a number of simultaneous calls depends on
the specific codec type used and the available bandwidth.
If both the intrazone and interzone codes are configured as Best Quality
(G.711), the SRG supports up to 24 Virtual Trunks (H.323 or SIP),
otherwise, only 15 Virtual Trunks (H.323 or SIP) are supported.
In Normal Mode, the codec selection used is controlled by specific
programming of the CS 1000.
In Local Mode, if the WAN has failed, Virtual Trunks between the SRG
and CS 1000 cannot be established. However, the SRG will continue to
convert calls from IP terminals for communication through the PSTN. Nortel
recommends you use G.711 codec.
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Branch office dialing plan 27
Branch office dialing plan
Since IP Phone users can be located at a branch office equipped with an
SRG, the routing of calls to the local gateway is important (especially when
toll charges apply to calls made from the central Call Server that controls
the telephone). The administrator can configure digit manipulation through
zone attributes for IP Phones to select a main office or branch office that
provides PSTN access local to the destination of the call.
Calls from the PSTN to users within the network can be routed with the
various ESN numbering plan configurations.
To access local PSTN resources, outgoing calls can be routed using ESN
as well as zone parameters that enable digit insertion. The zone parameters
force calls made by an SRG user to be routed to the desired local PSTN
facilities.
ATTENTION
Outgoing calls can include local and, optionally, long distance calls.
Nortel recommends that the Branch User ID (BUID) be the same at the
branch office as the DN at the main office. A BUID has a maximum of 15
digits. Under the recommended Coordinated Dialing Plan (CDP), the BUID
can be an extension (for example, 4567). Under the Uniform Dialing Plan
(UDP), the BUID is the user main office DN, the Location Code (LOC), plus
the Access Code (for example, 6 343-5555). The main office DN must be
The SRG only supports only one dialing plan option at a time. CDP and
UDP dialing plan options cannot be configured at the same time in the
same system.
For more information about dialing plans and configuration, see "Dialing
office dialing plan, see CS 1000 Main Office Configuration for SRG50
(NN43001-307).
ESN Access Codes
ESN data is configured with two Access Codes, called AC1 and AC2.
AC1 normally applies to long distance calls, whether placed on or off the
customer’s private network (for example, dialing 6). AC2 normally applies to
local calls (for example, 9). For more information, see Electronic Switched
Network Reference—Signaling and Transmission (NN43001-280).
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28 Description
Music on Hold
For SRG users in Normal Mode, the main office provides music to the user
if Music on Hold is provisioned. The use of the G.729A/AB codec between
the main office and the branch office can impact the music quality.
ATTENTION
G.723 codec is not supported on SRG 50.
Branch office and SRG 50 terminology
configuration-related terms and contexts where branch office and SRG
50 terminology differ.
Table 3
Branch office and SRG 50 terminology
Term or context
dialing plan
routing
Branch office
SRG 50
on-net/off-net dialing
distant steering codes (DSC),
Private/Public network dialing
call routing, destination codes, line
Trunk steering codes (TSC), Local pool access codes
steering codes (LSC)
Digit manipulation table
dial-out digits (routing)
scheduled call routing
alternate routing
selection
Facility Restriction Level (FRL)
Type of number
CDP/UDP/TNDN
CDP/UDP/no equivalent
Private
Numbering Plan ID
ISDN/Telephony
(E.164),Private, Telephony
(E.163), Telex, (F.69), Data
(X.121), National Standard
User ID
Trunks
BUID
BUID
bandwidth management zone
public exchange
virtual trunk
Zone ID
PSTN
IP trunk
access codes (SRG 50: 7 = system trunk access
destination codes)
7 = not assigned
8 = Basic Alternate Route Selection 8 = not assigned
(BARS)/Network Alternate Route
Selection (NARS)
9 = line pool A access code
9 = public exchange access
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Limitations 29
Term or context
Branch office
SRG 50
Network Class of Service (NCOS)
DN
telephone numbers
(internal, not PSTN)
DN
Limitations
The following is a list of limitations for SRG 50 Release 3.0:
•
When an IP Phone is in Local Mode, the SRG 50 does not provide all
the features as those provided by the CS 1000 main office. In Local
Mode, the SRG provides basic features, basic call handling, and basic
routing capabilities only.
•
When an IP Phone is in Local Mode, the SRG 50 does not support IP
Phone Key Expansion Module or Expansion Module for IP Phone 1100
Series.
•
•
You cannot configure the BUID and MOTN using the IP Phone.
Configure the BUID and MOTN using SRG Element Manager.
The SRG and the CS 1000 are configured separately. There is no
single management paradigm or application to update both the CS 1000
and the SRG. Use Element Manager to configure the SRG, and use
standard configuration tools to configure the CS 1000.
•
•
Virtual Office Login is not supported in Local Mode.
Language, Volume, and Contrast settings in the SRG are not
synchronized with the CS 1000 settings which causes a potential
mismatch in settings between Normal Mode and Local Mode.
•
•
Language options available on the CS 1000 may not be available on
the SRG.
For the CS 1000 Release 5.0 Alternate Routing for Network Bandwidth
Management feature, the SRG does not support an automatic
redirection of IP trunk calls through the PSTN when such calls are
blocked by the CS 1000 due to bandwidth availability.
•
•
Multiple ESDN is not supported.
VLAN tagging is not supported. However, VLAN tagging is achieved
by using an external router.
•
•
Active Call Failover is not supported.
SIP trunks are used only for voice packet traffic alone. H.323 trunking
is used for main office and Gatekeeper/NRS discovery, polling of WAN
link, as well as voice traffic.
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30 Description
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31
Setting up the main office
Contents
This section contains the following topics:
•
•
•
•
•
•
Introduction
This section describes the following information required to configure the
main office:
•
•
•
•
•
SRG information required by the main office
Main office information required by the SRG
Zone parameters
IP Phone passwords and parameters
Branch office IP Phone configuration
For more information on main office configuration, see IP Peer Networking
Installation and Commissioning (NN43001-313).
SRG information required by the main office
The main office administrator must gather information about the SRG
system. The following information is required:
•
an inventory of IP Phones that will be installed on the SRG so the
administrator knows what type of telephone to assign to each main
office terminal record
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32 Setting up the main office
•
•
information which allows the administrator to create an NCS (H.323
Gatekeeper or SIP Redirect Server) entry for the SRG
if using advanced routing, such as tandem dialing between systems,
local PSTN number for the SRG and the internal SRG routing codes
that will allow the main office to connect to the SRG and to tandem over
the SRG PSTN lines, is required
(page 32) to record the information before setting up the SRG on the main
office server.
Table 4
SRG information required for the main office configuration
SRG parameters
SRG public IP address
H.323 ID (required for requests to NCS)
Each H.323 ID in the node should match SIP endpoint name
for this system in pure SIP environment.
List of types and number of IP Phones
Telephone types are hard-coded to the Terminal Numbers
(TN) and the main office. Therefore, install the same type of IP
Phones to the coordinating record on the SRG.
PSTN number to dial into the SRG (in local mode)
Destination codes (steering codes) to route the main office calls
to the SRG and out through the SRG PSTN lines
IP Ports that affect SRG traffic with the main office and have
been assigned firewall filters
For further information on port configuration, see Converging
the Data Network with VoIP (NN43001-260) or SRG50
Configuration Guide (NN40140-500).
Main office information required by the SRG
The main office administrator must supply numerous main office settings to
the SRG installer so that the SRG can be efficiently configured. In addition,
the main office administrator needs to supply the following information:
•
•
•
a list of the terminal record numbers (TN)
a list of BUID (Prime DN)
if using advanced routing, such as tandem dialing between systems,
main office routing (steering) codes, are required
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Main office information required by the SRG 33
main office information required by the SRG.
Table 5
Main office interoperation information
Main office components
Information about this system
Main office IP network information:
Main office call server type
S1000 (default)
Primary network connect server address
Alternate network connect server
Network Connect server port
Trunk/telephony preferred codecs and jitter
buffers listed in order of preference
NRS (H.323 Gatekeeper/SIP Redirect Server)
requirements
Indicate if the SRG needs to manually assign
ports with firewall filters.
Telephony programming:
DN length, DN (TN) range
Numbering plan ID
Private (default)
Type of number
SRG 50 only supports CDP and UDP dialing
plans. Nortel recommends that the SRG use
CDP.
The SRG supports only one dialing plan option
at a time. CDP and UDP dialing plan options
cannot be configured at the same time in the
same system.
Node ID
When the SRG is down the phones use S2
settings to register with the main office.
Virtual Private Network ID (VPNI)
Zone ID and dialing string information
requirements
Main office dial-up number (for PSTN calls to the
main office in Local Mode)
Access code to reach the main office PSTN
through VoIP trunks
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34 Setting up the main office
Main office components
Information about this system
Zone dialing:
•
•
ZDP appended to SRG IP Phone PSTN
dialing strings to redirect the call to SRG
PSTN
Any steering codes (destination codes) that
must be mirrored by SRG programming
IP Phone configuration:
MOTN/BUID list, including which type of IP
Phone is assigned to each number.
Make note of the leading number, as SRG uses
this as the DN range for CDP dialing. If the DCP
access code is more than one digit, the second
digit number must also be used to further define
the DN range.
Current IP Phone firmware version
Is a VLAN configured on the network?
Zone parameters
Zone parameters must be configured at both the main office Call Server
and the SRG. The main office procedure is similar to an IP Peer Network
configuration with the branch office-specific configuration outlined in this
chapter.
Zone parameters are defined at the main office in LD 117 and are applied to
IP Phones in LD 11.
ESN and SRG zones.
Procedure 1
Configuring ESN and SRG zones
Step Action
ATTENTION
Before and after an upgrade, perform a data dump (using LD 43 EDD or through
Element Manager) on the Call Serve or on the MGC to back up existing data.
1
Configure the Home Location Code (HLOC) and the Virtual Private
Network Identifier (VPNI).
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Zone parameters 35
Table 6
Configure Customer Data Home Location Code and Virtual Private Network Identifier
Prompt
REQ:
Response
CHG
Description
Changing existing data
TYPE:
CUST
NET
ISDN and ESN Networking options
Customer number
0-99
Range for Large Systems
…
…
…
CLID
YES
Allow Calling Line Identification option
CLID entry to be configured
Home Location code (ESN) (3-7 digits)
Integrated Services Digital Network
-ENTRY
--HLOC
ISDN
xx
100-9999999
YES
-VPNI
(0)-16383
Virtual Private Network Identifier for Bandwidth
Management feature
X = Disables feature
1-16383 = Enables feature
<cr> = No Change
2
Configure the zone properties for IP Telephony bandwidth
management. Use LD 117 or Element Manager. See IP Peer
Networking Installation and Commissioning (NN43001-313).
The branch office zone number and zone bandwidth management
parameters at the main office must match the corresponding branch
office zone number and zone bandwidth management parameters
at the branch office.
ATTENTION
Zone 0, the default zone, must not be configured as a branch office zone.
Network Bandwidth Management does not support zone 0. If zone 0 is
configured as an branch office zone, the Bandwidth Management feature
is not activated.
3
Define the zone parameters for the branch office. Use LD 117
or Element Manager. See IP Peer Networking Installation and
Commissioning (NN43001-313).
LD 117 Define zone parameters for the branch office
Description
Command
CHG ZBRN <Zone> <yes|no>
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36 Setting up the main office
Description
Define a zone as a branch office zone.
Command
CHG ZDST <Zone> <yes|no> <StartMonth> <StartWeek> <StartDay> <StartHour> <EndMonth>
<EndWeek> <EndDay> <EndHour>
If the branch office observes Daylight Savings Time (DST), these parameters
specify the start and end of DST. During DST, the clock automatically advances
one hour forward.
CHG ZTDF <Zone> <TimeDifferencefromMainOffice>
Specified in minutes, the time difference between main office and branch office
when both are not in DST.
CHG ZDES <Zone> <ZoneDescription
A name to render data display more meaningful.
4
Enable the features for the branch office zone in LD 11.
LD 117 Enable features for an SRG zone
Description
Command
ENL ZBR <zone> ALL
Enables features for branch office <zone>.
—End—
Configuring zone parameters using CS 1000 Element Manager
Use Element Manager to configure the branch office specific zone
properties and time difference.
1. Select IP Network > Zones in Element Manager navigator.
Figure 2
Zone List web page
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Zone parameters 37
2. Select the zone to be configured and configure the following properties.
•
Basic Property and Bandwidth Management (see Figure 3 "Zone
•
Time Difference and Daylight Saving Time Property (see Figure 4
Figure 3
Zone Basic Property and Bandwidth Management web page
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38 Setting up the main office
Figure 4
Zone Time Difference and Time web page
Zone parameters must be configured on the main office and the branch
office. For information on configuring zones, see "Bandwidth Management"
Branch office IP Phone configuration at the main office
After the branch office zones and passwords are provisioned, provision the
branch office IP Phones at the main office. These can be provisioned using
ATTENTION
There is no automatic data synchronization between the main office Call Server
and SRG. The technician must provision the telephone on both the Call Server
and the SRG.
Branch office IP Phone configuration using Telephony Manager 3.1
At the main office, Telephony Manager 3.1 can be used to configure branch
office IP Phones. Use Telephone Pages to configure the telephones to
include the following:
•
•
Terminal Type
TN
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Branch office IP Phone configuration at the main office 39
•
•
•
Customer Number
Branch Office Zone
Prime DN corresponding to the BUID
See Telephony Manager 3.1 System Administration (NN43050-601) for
details.
Branch office IP Phone configuration using LD 11
using LD 11" (page 39) at the main office to configure branch office IP
Phones.
Procedure 2
Configuring branch office IP Phones at the main office using LD 11
Step Action
1
2
Configure the branch office zones and dialing plan. See Procedure
Configure the following telephone data in LD 11:
•
•
•
•
•
Terminal type
Customer Number
TN
Zone
Prime DN to correspond to BUID
LD 11 Provision Branch User and SCPW at the main office
Prompt
REQ:
Response
NEW CHG
a…a
Description
Add new data, or change existing data.
Terminal type.
TYPE:
Type ? for a list of possible responses.
Customer number as defined in LD 15.
Zone number to which the IP Phone belongs.
CUST
ZONE
xx
0-255
The zone prompt applies only when the TYPE is 2001P2,
2002P1, 2002P2, 2004P1, 2004P2, 2050PC, 2007, 1110,
1120, 1140, 2210, 2211, 2212
Zone number is not checked against LD 117.
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40 Setting up the main office
Prompt
…
Response
…
Description
…
SCPW
xxxx
Station Control Password
Must equal Station Control Password Length (SCPL) as
defined in LD 15. Not prompted if SCPL = 0. Precede with
X to delete.
—End—
SIP IP Trunks configuration at the main office
In order for the SRG 50 to act as a SIP endpoint and to use the SIP Trunks
for call signaling with the CS 1000, you must configure SIP Trunks between
the SRG 50 branch office and the CS 1000 Release 5.5 main office.
Configuring SIP IP Trunks
Step Action
1
From the Element Manager navigator, click IP Network > Nodes:
Servers, Media Cards.
The Node Configuration window appears.
2
3
4
Click the Edit button associated with the node to be updated.
Click the plus (+) sign beside Signaling Server Properties.
From the Enable IP Peer Gateway (Virtual Trunks TPS) list, select
SIP only.
5
6
Enter the CS 1000 domain name in the SIP Domain Name field.
Enter the SIP Port number in the Local SIP TCP UDP Port to
Listen to field.
7
Enter the Signaling Server name in the SIP Gateway Endpoint
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SIP IP Trunks configuration at the main office 41
Figure 5
SIP Trunk configuration in Element Manager
8
Click Save and Transfer.
The Save and Transfer window appears.
9
Click OK.
10
11
12
13
Log on to Network Routing Service (NRS) Manager.
Select the Configuration tab.
From the H.323 Support list, select H.323 not supported.
Select the Network Connection Server enabled check box. See
Figure 6
SIP Trunk configuration in NRS
14
15
Select Save.
Select Configuration > Gateway Endpoints.
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42 Setting up the main office
The Gateways Endpoints window appears.
The SRG registers as a static SIP endpoint. See Figure 7 "Gateways
Figure 7
Gateways Endpoints window in NRS
—End—
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NN43001-307 02.02 Standard
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43
Dialing Plan configuration
Contents
This section contains the following topics:
•
•
•
•
•
•
•
•
Overview
This section provides an overview of dialing plan programming on the SRG
and the main office.
When a number is dialed, the Call Server determines whether the called
number is internal or external to the branch office. If internal or off-net, the
system terminates the call on the appropriate terminal. If external or on-net,
the system routes the call using one of the supported dialing plans.
On-net dialing plan
The SRG only supports only one dialing plan option at a time. CDP and
UDP dialing plan options cannot be configured at the same time in the
same system.
The SRG supports the following dialing plans:
•
Coordinated Dialing Plan (CDP) – BUID is the same as the Directory
Number (DN)
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44 Dialing Plan configuration
•
Uniform Dialing Plan (UDP) – Location code is added to the DN for
the BUID
ATTENTION
Nortel recommends that the SRG use CDP.
CDP Terminal Numbers (TN) can be activated on the other systems if the
user moves and wants to retain their phone number. SRG does not support
Transferable Directory Numbers (TNDN) due to differences in dialing plans
and the small range of DN available on the SRG.
For specific examples for CDP and UDP dialing plans, see "Dialing plan
Once the call is sent over the IP network, the call is routed to the SRG,
which uses the NRS (H.323 Gatekeeper/SIP Redirect Server) to route the
call. The NRS (H.323 Gatekeeper/SIP Redirect Server) translates the
address form a telephone number to an IP address, and authorizes the call.
Specific dialing plan configuration is required for IP Phones to properly
select a main office or a branch office that provides access to the PSTN for
the originating IP Phone. A common configuration might be:
•
•
•
SRG users select the SRG PSTN for local calls.
Main office users select the main office PSTN for local calls.
All users select either the main office or SRG PSTN for long-distance
calls to minimize toll charges.
•
calls configured to minimize toll charges.
However, this configuration represents only one way that the dialing plan
could be configured. PSTN calls can be routed according to the point of
origin (main office or branch office) and/or the desired destination, and
can select trunks at the main office, branch office, or other branch offices
as required. Therefore, the user can route calls to gateways that minimize
long-distance costs, minimize bandwidth usage, or meet other criteria.
Nortel recommends that customers use Coordinated Dialing Plan (CDP)
between the main office and its branch offices since it enables all users,
at the main office or the branch office, to call each other using just an
extension number. CDP enables consistent dialing between the main office
and SRG IP Phones and devices.
For more information, see Dialing Plans Reference (NN43001-283).
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SIP/H.323 zones 45
Off-net dialing plan
When dialing to the PSTN, the Call Server determines that the call
destination is off-net by analyzing the digits that must be preconfigured at
major Call Servers in the network.
If routed over a Virtual Trunk, a request is sent to the NRS to determine
the location of public E.164 numbers. The NRS is configured with a list
of potential alternate routes that can be used to reach a certain dialed
number. Each route is configured with a unique route cost to determine
the least-cost route.
The NRS replies with the address information for E.164 numbers. It also
provides a list of alternative SIP or H.323 endpoints, sorted by cost. If a
terminating endpoint resource is busy when a call attempt is made, the
originating endpoint tries the next alternative. If no alternative is available
over the IP network, the originating endpoint steps to the next entry on its
route list, which could be a TIE or PSTN alternate route.
Routing calls
SRG user call to an SRG PSTN
The SRG user telephone is registered at the main office. The SRG user
telephones are physically located at the branch office, so routing of local
PSTN calls back to the branch office is essential, even if they are registered
with the main office.
Branch office behavior of the SRG user telephones at the main office is
configured by setting branch office zone characteristics through LD 117 at
the main office.
SRG PSTN to an SRG telephone (DID call)
If the DN is valid and can terminate, call termination at the branch office is
treated differently for IP Phones and non-IP Phones, as follows:
•
IP Phones—If the telephone is registered to the SRG (Local Mode),
the call is terminated locally. If the telephone is not registered to the
SRG (Normal Mode), the call is routed through a Virtual Trunk to the
main office.
•
Non-IP Phones—Calls are terminated locally (within the branch office).
SIP/H.323 zones
In a SIP/H.323 network, each NRS controls one SIP/H.323 zone. Each
zone can consist of many SIP/H.323 endpoints. If a call terminates beyond
the call originator zone, the SIP Redirect Server or H.323 Gatekeeper of the
called party zone provides the endpoint information to set up the connection.
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46 Dialing Plan configuration
It is possible to divide a system into several zones. It is also possible to
divide a customer within a system into different zones. It is more common to
assign one zone to one system and one customer.
Zone-based digit manipulation
For SRG users in Normal Mode, it may be desirable to provide routing that
is different from that provided to main office users. For example, it may be
desirable to route certain calls directly to the SRG PSTN trunk, rather than
receive the same routing as non-SRG users in the main office.
To achieve this, the Zone Access Code Behavior (ZACB) and Zone Digit
Prefix (ZDP) properties of the branch office zone are used to add digits to the
digits dialed by the SRG user. The resulting digit string is then used to route
the call. The net effect of this is that an SRG user’s and a main office user’s
call can be routed differently, even though the dialed digits were the same.
For example, if 1 87654321 is dialed, where 1 is the Access Code, then:
•
•
for a main office user, the call is routed based on the dialed digits.
for an SRG user, the digits undergo zone-based digit manipulation (such
as inserting 101), and the call is routed based on the new manipulated
digit string (in this example 1 101 87654321).
By performing this zone-based digit manipulation, calls from main office
users and SRG users undergo different routing. Some applications are:
•
•
•
•
routing all SRG user calls to the SRG PSTN trunk
routing SRG user local calls to the SRG PSTN trunk
routing all SRG user calls to the main office PSTN trunk
routing SRG user long-distance calls to the main office PSTN trunk
Special considerations apply in the case where a single Access Code
is used for both on-net and off-net calls, especially when UDP is used.
Routing of on-net and off-net calls is normally different. The Call Server
ESN Special Number provisioning and Gatekeeper Numbering Plan Entry
provisioning should be used to provide this different routing.
In the case where a single Access Code is not shared, that is, where
one Access Code is exclusively used for UDP on-net dialing, standard
procedures should be used. See Dialing Plans Reference (NN43001-283).
For a given branch office, there may be more than one zone defined at the
main office. Therefore, different SRG users may receive different routing
treatments.
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Zone-based digit manipulation 47
The combination of zone-based digit manipulation and CS 1000 routing
capabilities can be used to achieve many other routing outcomes for SRG
user calls.
Calling Line ID composition
Digital manipulation is commonly used for digit insertion and deletion. It
is also used for call type conversion before out-pulsing the digits to the
Virtual Trunk.
The IP Special Number (ISPN) parameter in the ESN data block ensures the
Calling Line ID (CLID) is formed correctly when a call-type is converted from
its original type (such as International, National, or SPN) to CDP/UDP/SPN
format. Conversion to CDP/UDP/SPN format ensures that the call-type
stays in the Private/Special Number domain.
The ISPN parameter is configured in LD 86. By default, it is set to NO.
If ISPN is NO, the CLID is formed based on the CTYP parameter of the DMI
data block, and INST digits are inserted.
If ISPN is YES, the CLID is formed based on the call-type before digit
manipulation. INST digits are inserted, and the CLID is considered an IP
Special Number. The call-type before digit manipulation is determined as
follows:
•
If the call-type before digit manipulation is SPN (Special Number), it
is converted to a value corresponding to the CLTP parameter in the
Table 7
Mapping between from CLTP parameter in SPN block to call-type before
digit manipulation
CLTP parameter
LOCL
Call-type before digit manipulation
Local PSTN
NATL
National PSTN
INTL
International PSTN
•
If the call-type before digit manipulation is not SPN (Special Number), it
is not changed.
CLID verification
Use the CLIDVER prompt in LD 20 to verify that the CLID has been properly
composed and configured. This command simulates a call, without actually
making the call, and generates a report of the properties of the call.
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48 Dialing Plan configuration
Configuring the dialing plan for PSTN access to SRG users in
Normal Mode
Preparing to configure the dialing plan
Before configuring the dialing plan for PSTN access to SRG users in Normal
Mode, you must complete the following steps:
•
At the main office, configure the Virtual Trunk to enable calls originating
on SRG IP Phones in Normal Mode to reach the branch office. See
IP Peer Networking Installation and Commissioning (NN43001-313)
for details.
•
•
At the main office, configure trunks for access to the PSTN.
At the branch office, configure the Virtual Trunk to enable calls originating
on SRG IP Phones in Normal Mode to reach the branch office. See
IP Peer Networking Installation and Commissioning (NN43001-313)
for details.
•
•
At the branch office, configure trunks for access to the PSTN.
At the main office, configure the branch office zone properties in LD
117, excluding the ZACB and ZDP properties. See IP Peer Networking
Installation and Commissioning (NN43001-313) for details.
•
•
At the main office, configure the routing for PSTN access.
At the branch office, configure the routing to enable calls made from
TDM or IP Phones in Local Mode to access the PSTN.
•
Configure IP Phones with the same zone number at both the main office
and the branch office. Nortel also recommends that the Prime DN be
the same at both the main and the branch offices. If different DN are
configured, the dial-in numbers change when the branch office is in
Local Mode.
•
Assign unique individual DN as Branch User Identities (BUID) to
Automatic Call Distribution (ACD) telephones.
Configuring the dialing plan
The steps to configure the dialing plan for SRG PSTN access are:
•
At the main office—see Procedure 3 "Configuring the main office" (page
1. Configure the ZACB property for the branch office zone.
2. Configure the ZDP property for the branch office zone.
3. Configure the Route List Index.
4. Configure the ESN Special Number and Digit Manipulation.
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 49
•
•
Configure the NRS—see Procedure 4 "Configuring the NRS database"
1. Access NRS Manager.
2. Select an endpoint.
3. Configure the Numbering Plan Entry for the branch office.
At the branch office—see Procedure 5 "Configuring the branch office"
1. Configure the Route List Index.
2. Configure ESN.
These steps can be done using overlays, as described in this section, or in
Element Manager and NRS Manager. See IP Peer Networking Installation
and Commissioning (NN43001-313) for more details.
Procedure 3
Configuring the main office
Step Action
1
Configure the ZACB property for the branch office zone.
LD 117 Define the zone Access Code handling for the branch office zone
Description
CHG ZACB <zone> [ALL]|[<AC1|AC2> <AC1|AC2>]
Define the Access Codes used to modify local or long-distance
Command
calls in the branch office to force all branch office calls to be
routed to the MG 1000B PSTN.
The ZACB and ZDP properties are used to configure the digit
manipulation behavior of the branch office zone (see step 2).
The ZACB property specifies which calls undergo digit manipulation.
The attribute can be configured in the following ways:
•
CHG ZACB <zone>
In this configuration, dialing AC1 or AC2 does not trigger digit
manipulation. SRG user calls are treated exactly the same as
those for main office users.
•
CHG ZACB <zone> ALL
In this configuration, calls dialed with AC1 and calls dialed with
AC2 undergo zone-based digit manipulation. All SRG user calls
can then be routed to the SRG PSTN.
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50 Dialing Plan configuration
For example, assume that AC1 = 1, AC2 = 2, and ZDP = 101.
If an SRG user dials 1 87654321, ZDP is inserted in the dialed
digits to form a digit string of 1 101 87654321. If an SRG user
dials 2 87654321, ZDP is inserted in the dialed digits to form a
digit string of 2 101 87654321.
•
CHG ZACB <zone> AC1 AC2
In this configuration, only calls dialed with AC1 undergo
zone-based digit manipulation. All SRG user calls dialed with
AC1 can then be routed to the SRG PSTN.
For example, assume that AC1 = 1, AC2 = 2, and ZDP = 101.
If an SRG user dials 1 87654321, ZDP is inserted in the dialed
digits to form a digit string of 2 101 87654321. If an SRG user
dials 2 87654321, zone-based digit manipulation does not occur
and the digit string remains unchanged.
•
CHG ZACB <zone> AC2 AC2
In this configuration, only calls dialed with AC2 undergo
zone-based digit manipulation. All SRG user calls dialed with
AC2 can then be routed to the SRG PSTN.
For example, assume that AC1 = 1, AC2 = 2, and ZDP = 101. If
an SRG user dials 1 87654321, zone-based digit manipulation
does not occur and the digit string remains unchanged. If an
SRG user dials 2 87654321, ZDP is inserted in the dialed digits
to form a digit string of 2 101 87654321.
As part of the ZACB configuration, you can also change the dialed
Access Code, so if you dial AC2 it can be changed to AC1, or
vice versa. This provides more flexibility in the main office NARS
configurations. Normally, you do not need to change the Access
Code.
The Access Code dialed by the user is used internally by the Call
Server. It is not sent as part of the outpulsed digits (to the NRS or
to the trunks).
If a specified Access Code is used for both local and long-distance
dialing, then both types of calls will receive the specified routing.
2
Configure the ZDB property for the branch office zone in the main
office. See IP Peer Networking Installation and Commissioning
(NN43001-313).
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 51
LD 117 Define the zone digit manipulation for the branch office zone at the main office
Description
CHG ZDP <zone> <DialingCode1> <DialingCode2> <DialingCode3>
Define the dialing plan for the branch office zone, where
Command
DialingCode1, DialingCode2, and DialingCode3 are inserted into
the dialed digits between the Access Code and the remainder of
the dialed number.
The ZDP and ZACB (step 1) properties are used to configure the
digit manipulation behavior of the branch office zone.
The ZDP property is inserted between the Access Code specified
in the ZACB command and the dialed digits. This zone-based digit
manipulation allows the main office Call Server and the network
NRS to distinguish the SRG user calls from the main office user
calls, and route them accordingly. The digit manipulation occurs
before any digit processing in the main office Call Server or NRS.
ATTENTION
If DialingCode1, DialingCode2, or DialingCode3 are already present in
the dialed digits, then they will not be re-inserted.
Nortel recommends that the ZDP attribute for each branch office
zone be set to a unique non-dialable number within the dialing plan
(for example 1019 or 999). This unique non-dialable number can
then be used, when configuring the main office ESN Special Number
and the NRS (H.323 Gatekeeper) (Procedure 4 "Configuring the
NRS database" (page 57)), to route the calls to the branch office for
connection to the local PSTN.
For example, assume AC1 = 1, AC2 = 2, ZACB = AC1 AC1, and
ZDP = 101.
If an branch office user dials 1 87654321, zone digit manipulation
occurs because AC1 was dialed and ZACB = AC1 AC1. ZDP is
inserted in the dialed digits to form a digit string of 1 101 87654321.
The call is routed differently than with the digits 1 87654321. ESN
configuration at the main office Call Server (step 4) routes the call
to the NRS because it recognizes 101 87654321 after the Access
Code rather than 87654321. The Access Code (1) is not included
in the digit string that is sent to the NRS. The NRS recognizes 101
at the front of the digit string and routes the call to the destination
SRG. At the branch office, the ESN Special Number is configured
(step 2) to remove 101 from the digit string and route the call based
on the digits 87654321.
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52 Dialing Plan configuration
Nortel recommends that the ZDP attribute for each branch office
zone be set to a unique non-dialable number within the dialing plan
(for example 1019 or 999). This unique non-dialable number can
then be used, when configuring the main office ESN Special Number
the NRS (H.323 Gatekeeper/SIP Redirect Server) (Procedure 4
"Configuring the NRS database" (page 57)), to route the calls to the
branch office for connection to the local PSTN.
For example, assume AC1 = 1, AC2 = 2, ZACB = AC1 AC1, and
ZDP = 101.
If an branch office user dials 1 87654321, zone digit manipulation
occurs because AC1 was dialed and ZACB = AC1 AC1. ZDP is
inserted in the dialed digits to form a digit string of 1 101 87654321.
The call is routed differently than with the digits 1 87654321. ESN
configuration at the main office Call Server (step 4) routes the call
to the NRS because it recognizes 101 87654321 after the Access
Code rather than 87654321. The Access Code (1) is not included
in the digit string that is sent to the NRS. The NRS recognizes 101
at the front of the digit string and routes the call to the destination
SRG. At the branch office, the ESN Special Number is configured
(step 2) to remove 101 from the digit string and route the call based
on the digits 87654321.
If an branch office user dials 2 87654321, zone-based digit
manipulation does not occur because AC2 was dialed and ZACB =
AC1 AC1. The digit string remains unchanged 2 101 87654321. The
main office routes the call using ESN configuration and the dialed
digits.
3
Configure the Route List Index at the main office.
After configuring zone-based digit manipulation, a specialized route
for the call must be configured. To select a trunk to route calls, a
Route List Index (RLI) must be configured in the Route List Block
(RLB). The RLI uses the route number for the Virtual Trunk to route
calls to the NRS. A Digit Manipulation Index (DMI) is associated with
the RLI to allow manipulation of the digits to be outpulsed. For this
application, at the main office, the DMI is used to update the call
type of the off-net calls to the Special Number (SPN) to make sure
the number stays in the Private/Special Number domain.
a. Configure the DMI in LD 86 with the DGT feature.
LD 86 Configure the Digit Manipulation Index at the main office
Prompt
Response
Description
REQ
NEW
Add new data.
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 53
Prompt
CUST
FEAT
DMI
Response
Description
xx
Customer number as defined in LD 15.
Digit manipulation data block
Digit Manipulation Index numbers
DGT
1-999
The maximum number of Digit Manipulation tables is defined at
the MXDM prompt in LD 86.
DEL
(0)-19
Number of leading digits to be deleted, usually 0 at the main
office.
INST
ISPN
x…x
Insert. Up to 31 leading digits can be inserted, usually none at
the main office. Default is none.
IP Special Number
For off-net calls
For on-net calls
(YES)
NO
CTYP
Call type to be used by the call. This call type must be
recognized by the NRS and far-end switch. This is critical for
correct CLID behavior.
If ISPN=NO, the CLID is based on this field. If ISPN=YES, the
CLID is based on the call type before digit manipulation.
SPN
LOC
For off-net calls (ISPN=YES)
For on-net calls (ISPN=NO)
b. Configure the RLI in LD 86 with the RLB feature.
LD 86 Configure Route List Index.
Prompt
REQ
Response
NEW
Description
Add new data.
CUST
FEAT
RLI
xx
Customer number as defined in LD 15.
Route List data block
RLB
Route List Index to be accessed
CDP and BARS
0-127
0-255
0-999
0-63
X
NARS
FNP
ENTR
Entry number for NARS/BARS Route List
Precede with x to remove
Local Termination entry
Route number of the Virtual Trunk as provisioned in LD 16.
LTER
ROUT
NO
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54 Dialing Plan configuration
Prompt
Response
Description
0-511
0-127
Range for Large Systems
Range for MG 1000B
…
DMI
1-999
Digit Manipulation Index number as defined in LD 86, FEAT =
DGT (see step a).
For example, assume that the Virtual Trunk is on route 10, and
the Customer number is 0:
>LD 86
REQ NEW
CUST 0
FEAT DGT
DMI 10
DEL
INST
ISPN YES
CTYP NATL
REQ NEW
CUST 0
FEAT RLB
RLI 10
ENTR 0
LTER NO
ROUT 10
...
DMI
...
4
Configure ESN Special Number and Digit Manipulation.
LD 90 Configure ESN Special Number and Digit Manipulation
Prompt
REQ
Response
NEW
Description
Add new data.
CUST
FEAT
TRAN
xx
Customer number as defined in LD 15
Network translation tables
Translator
NET
AC1
AC2
SPN
Access Code 1 (NARS/BARS)
Access Code 2 (NARS)
Special code translation data block
TYPE
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 55
Prompt
Response
Description
SPN
x…x
Special Number translation
Enter the SPN digits in groups of 3 or 4 digits, separated by a
space (for example, xxxx xxx xxxx). The SPN can be up to 19
digits long.
The maximum number of groups allowed is 5.
Flexible Length
FLEN
(0)-24
0-999
The number of digits the system expects to receive before
accessing a trunk and outpulsing these digits.
…
- RLI
- CLTP
Route List Index configured in LD 86 (see step 3)
Type of call that is defined by the special number.
Local PSTN
LOCL
NATL
INTL
National PSTN
International PSTN
—End—
After configuring the zone-based digit manipulation (step 1 and step 2)
and specialized route (step 3), the route must be associated with the ESN
Special Number. The main office ESN Special Number configuration is
based on new digits inserted by zone-based digit manipulation. The digits
are processed based on the Access Code, AC1 or AC2, that was dialed.
For off-net calls the following should be considered:
•
If all calls that have undergone Zone-based digit manipulation are to be
routed by the NRS, one SPN must be provisioned for each call type to
route calls to the NRS based on the ZDP.
•
If some calls are to be routed by the NRS, and others by the main office
Call Server, multiple SPN should be provisioned to route calls based on
the ZDP value and one or more dialed digits. Each SPN can then use
a different RLI if required.
For example, assume ZDP = 101. It is possible to provision multiple
SPN (1011, 1012, 1013, 1014, 1015, 1016, 1017, 1018, 1019, and
1010) to route calls based on the ZDP value plus the first dialed digit.
However, it may not be necessary to provision all SPN combinations.
For example, if calls dialed with a first digit of 3 after the Access Code
are invalid, then SPN 1013 does not need to be provisioned.
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56 Dialing Plan configuration
•
Be careful when choosing how many dialed digits to include in the
SPN. If one of the dialed digits is included in the SPN (that is, ZDP +
one dialed digit), a maximum of ten SPN must be configured for each
branch office. Similarly if two dialed digits are included in the SPN (ZDP
+ two dialed digits), a maximum of 100 SPN must be configured for each
branch office. For each additional dialed digit included in the SPN, the
maximum number of SPN that must be provisioned for each branch
office is increased by a factor of ten.
If a single Access Code that undergoes Zone-based digit manipulation is
used for both on-net and off-net calls, then separate DMI and SPN must be
provisioned to correctly route these calls. The SPN must correctly identify
the routing to be used, and its CLTP field must set the call type correctly. A
DMI, associated with this SPN, is used to make sure the number stays in
the Private/Special Number domain.
ESN Special Numbers are configured in LD 90. Respond to the prompts as
follows:
•
•
TRAN — Enter the Access Code.
TYPE — Enter SPN for this configuration, as the ZDP value configured
in step 3 is usually a unique non-dialable number.
•
SPN — Enter the ZDP value plus enough digits to distinguish the type
of number, such as national, international, or local. There must be
enough SPN entries to route all valid dialed numbers (see the example
in this section).
•
•
FLEN — Enter the number of digits that are expected for the call type.
RLI — Enter the RLI configured in LD 86 in step b. The RLI routes the
call to the NRS with the correct type of number.
•
•
CLTP — Enter the type of call defined by this Special Number: local
(LOCL), national (NATL), or international (INTL).
For example, assume the following:
AC1 = 1, ZACB = AC1 AC1, and ZDP = 101
Customer number = 0
Long-distance calls start with 1, have 11 digits, and use RLI = 10 and
DMI = 10.
Local calls start with 5 or 6, are seven digits long, and use RLI = 30
and DMI = 30.
ATTENTION
RLI and DMI values do not have to be the same, but for clarity, it may be
useful to set them the same.
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 57
>LD 90
REQ NEW
CUST 0
FEAT NET
TRAN AC1
TYPE SPN
SPN 1011
FLEN 14
11 digits for long-distance + 3 digits for ZDP
7 digits for long-distance + 3 digits for ZDP
7 digits for long-distance + 3 digits for ZDP
…
RLI 10
CLTP NATL
…
SPN 1015
FLEN 10
…
RLI 30
CLTP LOCL
…
SPN 1016
FLEN 10
RLI 30
CLTP LOCL
…
After configuring main office routing to the NRS, the NRS database must
be provisioned to identify the desired endpoint for the calls. This procedure
configures the NRS database with the inserted digits specified by the
zone-based digit manipulation configuration.
Instead of configuring the NRS database, you can configure a route in the
main office to directly route the call (see step 4).
This procedure provides information specific to the configuration of the
NRS database for this application. See IP Peer Networking Installation and
Commissioning (NN43001-313) for complete details on configuring the NRS.
Procedure 4
Configuring the NRS database
Step Action
1
2
Click the Configuration tab in NRS.
Click set Standby DB view to work in the standby (inactive)
database.
3
Select Routing entries from the navigation menu on the left-hand
side of the Network Routing Service window.
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58 Dialing Plan configuration
Figure 8
NRS Routing Entries window with no endpoint selected
4
5
6
Choose the appropriate Service Domain, L1 Domain, and L0
Domain from the corresponding drop-down menus.
Click on Look up to open a window with a lookup path for gateway
endpoints.
"NRS Lookup path for gateway endpoints" (page 58)), and click on
the endpoint at the branch office.
The Routing Entries window then displays a list of routing entries
Figure 9
NRS Lookup path for gateway endpoints
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 59
Figure 10
NRS Routing Entries window for selected endpoint
7
Click Add in the Routing Entries window to add a routing entry.
The Add Routing Entry window opens, as shown in Figure 11 "Add
Figure 11
Add Routing Entry
8
Configure the numbering plan entries for the branch office. This
is usually set to the unique non-dialable number that identifies the
branch office, as configured in the ZDP property of the branch office
zone in LD 117 at the main office (see step 2).
The type of number configured in the NRS should be set to match
the type of number as configured in the main office.
If some calls are to be routed differently from others, it is possible to
provision the multiple Numbering Plan Entries in the NRS to achieve
this.
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60 Dialing Plan configuration
For example, if ZDP = 101, it is possible to provision multiple
Numbering Plan Entries (101, 1011, and so on) to route calls based
on the ZDP value or the ZDP value plus some of the dialed digits.
Unlike on the Call Server, if the ZDP plus additional digits are
used to identify routing it is not necessary to provision all of the
combinations. For example, if calls with digit strings starting with
1011 are to be routed differently from those starting with 101x
(where x is a digit other than 1), then only 101 and 1011 need to be
provisioned as numbering plan entries on the NRS.
—End—
Procedure 5
Configuring the branch office
Step Action
1
Configure the Route List Index at the branch office.
After the call arrives at the branch office, a route must be provisioned
to handle the call. In order to be able to select a trunk to route calls,
a Route List Index (RLI) must be configured in the Route List Block
(RLB). The RLI uses the route number for PSTN trunk to route calls
to the PSTN. A Digit Manipulation Index (DMI) can be associated
with the RLI to allow manipulation of the digits to be outpulsed. For
this application, the DMI is used to remove the ZDP digits that were
inserted in the dialed digits at the main office. The DMI is also used
to convert the call type back correctly according to the incoming
SPN pattern.
a. Configure the DMI in LD 86 with the DGT feature.
LD 86 Configure Digit Manipulation Index at the branch office
Prompt
REQ
Response
NEW
Description
Add new data.
CUST
FEAT
DMI
xx
Customer number as defined in LD 15
Digit manipulation data block
DGT
1-999
Digit Manipulation Index numbers
The maximum number of Digit Manipulation tables is defined by
prompt MXDM in LD 86.
DEL
(0)-19
Number of leading digits to be deleted.
This would normally be configured to remove the unique
non-dialable number that identifies the branch office, configured
in the ZDP property of the branch office zone in LD 117 at the
main office (step 2).
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 61
Prompt
ISPN
Response
Description
NO
IP Special Number
INST
x…x
Insert. Up to 31 leading digits can be inserted.
CTYP
Call type used by the call. The far-end switch must recognize
this call type.
INTL
NPA
NXX
LOC
SPN
International
National
UDP
Local PSTN
Special Number
b. Configure the RLI in LD 86 with the RLB feature.
LD 86 Configure Route List Index
Prompt
REQ
Response
NEW
xx
Description
Add new data.
CUST
FEAT
RLI
Customer number as defined in LD 15
Route List data block
RLB
Route List Index to be accessed
CDP and BARS
0-127
0-255
0-999
0-63
X
NARS
FNP
ENTR
Entry number for NARS/BARS Route List
Precede with x to remove.
Local Termination entry
Route number of the Virtual Trunk as provisioned in LD 16.
Range for Large Systems
Range for MB 1000B
LTER
ROUT
NO
0-511
0-127
…
DMI
1-999
Digit Manipulation Index number as defined in LD 86, FEAT =
DGT (step a).
For example, assume that the PSTN trunk is on route 18 and the
Customer number = 0.
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62 Dialing Plan configuration
>LD
86
REQ NEW
CUST 0
FEAT DGT
DMI 18
DEL
3
Set to remove ZDP added in the main office
INST
CTYP LOC Set according to associated SPN pattern
REQ NEW
CUST 0
FEAT RLB
RLI 18
ENTR 0
LTER NO
ROUT 18
...
DMI 18
...
2
Configure ESN Special Number and Digit Manipulation.
LD 90 Configure ESN Special Number and Digit Manipulation
Prompt
REQ
Response
NEW
Description
Add new data.
CUST
FEAT
TRAN
xx
Customer number as defined in LD 15.
Network translation tables
Translator – Access Code 1 (NARS/BARS)
NET
Because the call is incoming to the branch office, AC1 is
triggered if INAC = YES in the Route Data Block for the Virtual
Trunk in LD 16 and the INTL call type is associated with AC1 in
NET_DATA of the Customer Data Block in LD 15.
AC1
SPN
x…x
TYPE
Special code translation data block
Special Number translation
Enter the SPN digits in groups of 3 or 4 digits, separated by a
space (for example, xxxx xxx xxxx). The SPN can be up to 19
digits long.
The maximum number of groups allowed is 5.
Flexible Length
- FLEN
(0)-24
The number of digits the system expects to receive before
accessing a trunk and outpulsing these digits.
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Configuring the dialing plan for PSTN access to SRG users in Normal Mode 63
Prompt
…
Response
Description
- RLI
0-999
Route List Index configured in LD 86 (see step 1)
After configuring the specialized route for calls that have been routed
to the branch office by the NRS, the route must be associated with
the ESN Special Number.
The branch office receives the manipulated number as an incoming
call, indicating that the ZDP value added at the main office is at the
beginning of the number. The branch office ESN configuration must
ensure that the extra digits (the ZDP value) are deleted by using a
proper DMI. The call then terminates at the PSTN connection.
ATTENTION
The DMI configured in LD 86 in step 1 is used to remove the digits that
were inserted in the dialed number at the main office.
For example, assume ZDP at the main office = 101, Customer
number = 0, and the RLI for the PSTN trunk = 18.
LD > 90
REQ NEW
CUST 0
FEAT NET
TRAN AC1
TYPE SPN
SPN 1011
FLEN 0
…
RLI 18
—End—
Configuring the dialing plan using CS 1000 Element Manager
From Element Manager, configure the branch office–specific zone dialing
plan and Access Codes. From the navigator, select IP Network >
Zones. From the Zones window in Element Manager, select the Branch
Office Dialing Plan and Access Codes option, and enter the necessary
information. See Figure 12 "Zone Dialing Plan and Access Codes" (page
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64 Dialing Plan configuration
Figure 12
Zone Dialing Plan and Access Codes
Testing PSTN access using an SRG IP Phone
Procedure 6
Testing PSTN access using an SRG IP Phone
Step Action
1
2
From an SRG IP Phone in Local Mode:
a. Make a local PSTN call.
b. Make a long-distance call.
The calls must be routed according to the branch office ESN
configuration.
From an SRG IP Phone in Normal Mode:
a. Make a call to the local PSTN.
b. Make a long-distance call.
—End—
The calls must be routed according to the ESN configuration that was
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Dialing plan examples 65
For calls that tandem over the Virtual Trunk to the branch office and go
out to the PSTN trunk(s) in the branch office, the following configuration
problems can occur:
•
The call can receive overflow tones. Use L D 96 to view the digits sent
to the Virtual Trunk (ENL MSGO {dch#}).
•
If the digits look correct at the main office, the NRS might not be
properly configured. If the NRS rejects the call, a diagnostic message
is displayed on the Signaling Server console.
•
If the call makes it to the correct branch office (check that it is not going
to the wrong node if the NRS is configured incorrectly) the branch office
is probably rejecting it because it does not know the digit string. Use
LD 96 to view the digits (ENL MSGI {dch#}).
Dialing plan examples
This section describes the following dialing plans:
•
•
Coordinated Dialing Plan (CDP)
Uniform Dialing Plan (UDP)
Coordinated Dialing Plan
The following section provides three options for creating a CDP dialing
configuration.
Overview
Dialing plans between the SRG and the main office need to be coordinated
to ensure seamless dialing between the systems. The option you choose will
determine how the user dials the other system or the SRG IP telephones.
•
Option 1: DN ranges in the main office and SRG are unique, and DNs
for SRG IP Phones are the same in both Normal and Local mode. This
is the recommended configuration to support seamless dialing on both
•
•
Option 2: DN ranges in the main office and SRG overlap, and DNs for
SRG IP Phones are the same in both Normal and Local mode. See
Option 3: DN of SRG IP Phones and DN in the main office overlap in
Call scenarios
Call scenarios fall into the following categories:
•
•
Common call scenarios occur in all CDP calls, regardless of which
option is used.
Unique call scenarios occur only within certain CDP options.
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66 Dialing Plan configuration
This section describes the common call scenarios. The unique call
scenarios are described with the configuration of the corresponding option,
Normal Mode: Main office telephone calls an analog phone at the
(page 66) shows how the call proceeds.
Figure 13
Normal Mode: Main office telephone calls an analog phone at the SRG
Normal Mode: Main office telephone calls a branch IP Phone The call
is recognized as a main office number, and the call is directed to the SRG
IP telephone using internal routing at the main office.
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Dialing plan examples 67
Normal Mode: Main office telephone makes a call over the PSTN
through the SRG Routing is configured so the destination code of the
the SRG" (page 67) shows how the call proceeds.
Figure 14
Normal Mode: Main office telephone makes a call over the PSTN through
the SRG
Normal Mode: SRG IP Phone makes a call over the PSTN Zone
management at the main office recognizes that an SRG IP Phone in Normal
Mode is dialing the PSTN. Figure 15 "Normal Mode: SRG IP Phone makes
a call over the PSTN" (page 68) shows how the call proceeds.
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68 Dialing Plan configuration
Figure 15
Normal Mode: SRG IP Phone makes a call over the PSTN
A telephone registered to the SRG calls another telephone registered
to the SRG The SRG routes the call internally.
Local Mode: SRG telephone calls an SRG IP Phone The call is handled
by the SRG and is sent directly to the SRG IP Phone.
Local Mode: SRG telephone calls a main office telephone In this case,
the WAN or NRS is not accessible. Figure 16 "Local Mode: SRG telephone
calls a main office telephone" (page 69) shows how the call proceeds.
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Dialing plan examples 69
The user must have configured the fallback route appropriately. See the
SRG50 Configuration Guide (NN40140-500) for further information.
Figure 16
Local Mode: SRG telephone calls a main office telephone
Local Mode: Main office telephone calls an SRG IP Phone The
call is treated according to main office redirection configuration, such as
forwarding to voice mail or continuous ringback.
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70 Dialing Plan configuration
Option 1
DN ranges in the main office and SRG are unique; DNs for SRG IP
Phone are the same in Normal and Local Mode
This is the recommended CDP configuration to offer seamless dialing.
In this configuration, the user dials the same DN for SRG IP Phones in
either Normal or Local Mode. The DN for SRG IP Phones are configured to
be the same on both the SRG and main office. This allows seamless dialing
from both the SRG and main office. However, in this configuration, the DN
range for telephones registered at the SRG is unique from the DN range for
telephones registered at the main office.
The advantage of this configuration is that the system manages the routing
for the SRG IP Phones, so users in the SRG and main office do not have to
be aware of whether the SRG is in Normal Mode.
Figure 17
CDP Option 1
Call scenarios Common call scenarios for this CDP option are listed
unique to this CDP option:
•
An SRG analog telephone registered to the SRG calls a telephone
registered at the main office that can also be an SRG IP Phone in
Normal Mode.
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Dialing plan examples 71
SRG analog phone calls an SRG IP Phone and a main office IP Phone
registered to the main office (Normal Mode).
office IP Phone " (page 73) shows the WAN is down. An SRG analog
phone calls an SRG IP Phone and a main office IP Phone registered
to the SRG (Local Mode).
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72 Dialing Plan configuration
Figure 18
Calls to an SRG IP Phone and a main office IP Phone registered to the main
office
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Dialing plan examples 73
Figure 19
Calls to an SRG analog phone, SRG IP Phone, and a main office IP Phone
Configuration To configure the main office:
•
Configure the ESN Control Block for CDP in LD 86.
>LD 86
REQ NEW
CUST 0
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74 Dialing Plan configuration
FEAT ESN
CDP YES
MXSC 50
NCDP 4
DLTN YES
•
Configure the CDP Distant Steering Code (DSC) in LD 87.
> LD 87
REQ NEW
CUST 0
FEAT CDP
TYPE DSC
DSC 50
FLEN 4
RLI 12
To configure the NRS (H.323 Gatekeeper/SIP Redirect Server):
•
•
•
Create CDP Domain: MO_BO_CDP.
Create H.323/SIP endpoints: MO, BO.
Create Numbering Plan entries in CDP Domain:
— Add 40 for endpoint BO.
— Add 30 for endpoint MO.
— Add 42 for endpoint MO.
For information about configuring H.323/SIP Redirect Server, see IP Peer
Networking Installation and Commissioning (NN43001-313).
To configure the SRG:
•
•
•
Configure DN and BUID as the same number on each of the redirected
IP Phones. For example, DN/BUID = 42XX.
Set the main office VoIP Trunk Access code to 3. For example, main
office VoIP trunk access code = 3.
Set the destination code for the VoIP trunk to 30 (retain all digits) or 34
(remove first digit). For example, BUID dialout = 342XX.
The VoIP route destination codes 30 (no digits dropped) and 34 (1 digit
dropped) route any call that starts with 30 or 34 out of the system over
the VoIP trunk to the main office.
The main office access code length is still 0.
•
Assign the telephones registered to the SRG (IP Phones or analog
[500/2500-type]) telephones to a different range, such as 40XX. See the
NRS configuration above.
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Dialing plan examples 75
The users in both the main office and the SRG dial only the DN for all
telephones in the main office and the SRG in both Normal Mode and
Local Mode.
For more information on configuring the main office and NRS, see Branch
Office Installation and Commissioning (NN43001-314) and IP Peer
Networking Installation and Commissioning (NN43001-313). For more
information on configuring the SRG, see SRG50 Configuration Guide
(NN40140-500).
Option 2
DN ranges in the main office and SRG overlap; DNs for SRG IP Phones
are the same in Normal and Local Mode
In this configuration, the SRG DN overlap with the main office DN. However,
since SRG does not support Vacant Number Routing (VNR), a user
registered to the SRG must dial a destination code before the main office
DN to call a main office telephone.
To call an SRG IP Phone in either Normal or Local Mode, SRG and main
office users need to dial only the DN for the SRG IP Phone. SRG IP Phone
calls are forwarded with the main office Private Network ID/destination code
appended to the BUID, which allows the call to flow to the VoIP trunks for
the main office.
This configuration is not a true CDP dialing plan. A destination code is
added by the system to properly direct the SRG IP Phone calls, since the
start digits of the DN are not unique for SRG and main office users. Users
dialing a telephone registered at the main office must dial a destination code
before the main office DN. This plan allows all systems on the network to
appear to be available within a range of numbers.
Since the SRG DN range is limited to about 200 DN, this configuration
only works if SRG dialing to the main office is limited to the redirected IP
Phones and to a small number of main office telephones, such as to a
central attendant and voice mail lines.
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76 Dialing Plan configuration
Figure 20
CDP Option 2
Call scenarios Common call scenarios for this CDP option are listed
unique to this CDP option:
•
Normal Mode: An SRG analog phone calls an SRG IP Phone and a
main office IP Phone registered to the main office.
The WAN is up. SRG analog phone calls an SRG IP Phone and a main
office IP Phone registered to the main Office (Normal Mode). SeeFigure
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Dialing plan examples 77
Figure 21
SRG analog phone calls an SRG IP Phone and a main office IP Phone
registered to the main office
•
Local Mode: SRG IP Phones are registered to the SRG.
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78 Dialing Plan configuration
In this scenario, the WAN and the NCS are working. However, the
SRG IP Phones are redirected to the SRG and are in Local Mode (Call
Forward All Calls is inactive). The following occur:
— Telephones registered at the SRG dial local DNs (see the common
— SRG calls to the main office use VoIP routing. The WAN is down.
SRG analog phone calls an SRG IP Phone and a main office IP
Phone registered to the SRG (Local Mode) See Figure 22 "SRG
— Main office calls to SRG IP Phones in Local Mode cannot complete
because the NRS cannot resolve the numbering.
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Dialing plan examples 79
Figure 22
SRG analog phone calls an SRG IP Phone and a main office IP Phone
Configuration To configure the main office:
•
Configure the ESN Control Block for CDP in LD 86.
> LD 86
REQ NEW
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80 Dialing Plan configuration
CUST 0
FEAT ESN
CDP YES
MXSC 50
NCDP 4
DLTN YES
•
Configure the CDP Distant Steering Code (DSC) in LD 87.
> LD 87
REQ NEW
CUST 0
FEAT CDP
TYPE DSC
DSC 50
FLEN 4
RLI 12
To configure the NRS (H.323 Gatekeeper/SIP Redirect Server):
•
•
•
Create CDP Domain: MO_BO_CDP.
Create H.323 and SIP endpoints: MO, BO.
Create Numbering Plan entries in CDP Domain:
— Add 30 for endpoint BO.
— Add 32 for endpoint MO.
For information about configuring H.323 Gatekeeper/SIP Redirect Server,
see IP Peer Networking Installation and Commissioning (NN43001-313).
To configure the SRG:
•
•
•
Configure DN and BUID as the same number on each of the redirected
IP Phones. For example, DN/BUID = 32XX.
Set the main office VoIP Trunk Access code to 6. For example, main
office VoIP trunk access code = 6.
Set the destination code for the VoIP trunk to 6, the same value as the
access code. For example, BUID dialout = 632XX.
The main office access code length is still 0.
•
Assign the telephones registered to the SRG (IP Phones or analog
[500/2500-type] telephones) to a different range, such as 30XX, than
the telephones registered to the main office.
SRG users must dial the destination code before the DN when making
a call to a telephone in the main office, whether they are in Normal or
Local Mode. When calling another IP Phone in the SRG, SRG users dial
only the DN, whether they are in Normal or Local Mode. The main office
uses VNR to route SRG DN to the SRG in both Normal and Local Mode.
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Dialing plan examples 81
For more information on configuring the main office and NRS, see Branch
Office Installation and Commissioning (NN43001-314) and IP Peer
Networking Installation and Commissioning (NN43001-313). For more
information on configuring the SRG, see SRG50 Configuration Guide
(NN40140-500).
Option 3
DNs of SRG IP Phones and DNs in the main office overlap in Normal
Mode, but are unique in Local Mode
In this CDP configuration, each node on the network has unique leading
digits that is included in the DN range. The unique leading digits indicate
the private network code for the system.
This configuration allows seamless dialing for users registered at the SRG,
but main office users must dial a different DN to call SRG IP Phones in
Normal and Local mode. Therefore, SRG IP Phones have DNs and BUIDS
that do not match.
starting with 4 on the SRG to accommodate the SRG Private Network Code.
On the main office, the SRG IP Phones are given a DN (BUID) starting
with 3, the main office Private Network Code. The NRS is programmed
to recognize that 3X numbers go to the main office and that 4X numbers
go to the SRG.
In Normal mode, when a call is directed into the SRG, or from a telephone
registered at the SRG, to the SRG IP Phone in Normal mode, the SRG
system translates the SRG IP telephone DN (4XXX) to the main office BUID
(3XXX) so that the call can route correctly through the main office VoIP
trunk. Users registered at the main office dial the main office DN (3XXX) for
the SRG IP Phone.
In Local mode, the users registered to the SRG still dial the SRG IP Phone
DN (4XXX). The main office users can not call the SRG IP Phone by dialing
the main office DN for the telephone (3XXX) because the NRS cannot route
the call to the SRG. If the main office user dials the SRG IP Phone DN
(4XXX), the call goes through.
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82 Dialing Plan configuration
Figure 23
CDP Option 3
Call scenarios Common call scenarios for this CDP option are listed
unique to this CDP option:
•
Normal Mode: An SRG analog phone calls an SRG IP Phone and a
main office IP Phone registered to the main office.
In this scenario, the telephone registered to the SRG can either dial the
SRG DN or the main office DN for the SRG IP Phone. In Local Mode,
the SRG IP telephone is reached only with the SRG DN.
In Normal Mode, the display on the IP Phone displays the main office
DN (3xxx) for the IP Phone. In Local Mode, the SRG DN (4xxx) is
displayed. The WAN is up: SRG analog phone calls an SRG IP Phone
and a main office IP Phone registered to the main office (Normal Mode).
IP Phone registered to the main office" (page 83) shows this scenario.
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Dialing plan examples 83
Figure 24
SRG analog phone calls an SRG IP Phone and a main office IP Phone
registered to the main office
• Local Mode: SRG IP Phones are registered at the SRG.
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84 Dialing Plan configuration
In this scenario, the WAN and the NCS are working. If the main office user
dials the SRG DN (42xx) to call the IP Phone, the call goes through.
Configuration To configure the main office:
•
Configure the ESN Control Block for CDP in LD 86.
> LD 86
REQ NEW
CUST 0
FEAT ESN
CDP YES
MXSC 50
NCDP 4
DLTN YES
•
Configure the CDP Distant Steering Code (DSC) in LD 87.
> LD 87
REQ NEW
CUST 0
FEAT CDP
TYPE DSC
DSC 4
FLEN 4
RLI 12
To configure the NRS (H.323 Gatekeeper/SIP Redirect Server):
•
•
•
Create CDP Domain: MO_BO_CDP.
Create H.323/SIP endpoints: MO, BO.
Create Numbering Plan entries in CDP Domain:
— Add 4 for endpoint BO.
— Add 30 for endpoint MO.
For information about configuring H.323 Gatekeeper/SIP Redirect Server,
see IP Peer Networking Installation and Commissioning (NN43001-313).
To configure the SRG:
•
Set the BUID to the same number that was assigned for the TN by the
main office.
•
•
Set the main office VoIP Trunk Access code to 0.
Do not assign a value to the main office trunk access code field.
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Dialing plan examples 85
For more information on configuring the main office and NRS, see Branch
Office Installation and Commissioning (NN43001-314) and IP Peer
Networking Installation and Commissioning (NN43001-313). For more
information on configuring the SRG, see SRG50 Configuration Guide
(NN40140-500).
Uniform Dialing Plan
Overview
Figure 25 "UDP using location codes" (page 85) shows an example of a
Uniform Dialing Plan (UDP) using location codes (Access Code + LOC
+ DN) configuration.
In this type of dialing plan, the DNs on the SRG do not need to be different
from the BUID, since the location code (LOC) defines the unique node
characteristic. Therefore, in this example:
•
The SRG IP Phone has DN 3002 and BUID 3002. (The system adds
the routing code and LOC code to the BUID).
•
•
•
The local telephone has a DN of 3101.
The main office has a telephone configured as TN 3001.
On the main office, the AC1 steering code for the SRG is 6 and the
LOC is 504.
•
On the SRG, the destination code for the main is 6 and the LOC is 501.
Figure 25
UDP using location codes
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86 Dialing Plan configuration
Call scenarios
This section describes how calls interact between the SRG and main office
with UDP.
Calling from main office to the SRG and SRG PSTN, in Normal
mode In this scenario, a telephone registered at the main office calls a
telephone registered to the SRG, or makes a call over the PSTN through the
in Normal Mode" (page 86) shows this scenario.
Figure 26
Calling from the main office to the SRG and SRG PSTN, in Normal Mode
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Dialing plan examples 87
Calling from the SRG to the main office, in Normal Mode In this
scenario, a telephone registered at the SRG calls an SRG IP Phone and a
main office IP Phone registered to the main office. The WAN is up. SRG
analog phone calls an SRG IP Phone and a main office IP Phone registered
to the main office (Normal Mode). Figure 27 "SRG analog phone calls an
(page 88)shows this scenario.
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Figure 27
SRG analog phone calls an SRG IP Phone and a main office IP Phone
registered to the main office
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Dialing plan examples 89
Calling in Local Mode In this scenario, the IP Phones at the SRG
are in Local Mode because the WAN is down. The SRG IP telephones
are reregistered to the SRG and call forward BUID is inactive on these
telephones. These IP Phones are registered at the SRG, and call forward
BUID is inactive on these telephones.
The inset shows a main office call to SRG telephones. The user must dial
the SRG DN for the IP telephone (6002 instead of 3002). In this case, the
user dialing is different in the following ways:
•
•
DN 3001 can call DN 3002 by dialing 65043002, instead of 3002.
DN 3101 can call DN 3002 by dialing 3002, instead of 65013002 dialed
in Normal Mode.
•
•
DN 3002 can call DN 3001 by dialing 65013001, instead of 3001 dialed
in Normal Mode.
DN 3002 can call DN 3101 by dialing 3101 instead of 65043101 dialed
in Normal Mode.
The WAN is down. SRG analog phone calls an IP Phone and a main office
(page 90) shows a call from the SRG to an SRG IP Phone and a main
office IP Phone registered at the SRG.
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90 Dialing Plan configuration
Figure 28
SRG analog phone calls an IP Phone and a main office IP Phone registered
to the SRG
Configuration examples
The following configurations are based on the examples provided in
this section. For further information, see Branch Office Installation and
Commissioning (NN43001-314).
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Dialing plan examples 91
To configure the main office:
•
•
•
Configure the ESN Control Block for UDP in LD 86.
> LD 86
REQ NEW
CUST 0
FEAT ESN
AC1 16
Configure Digit Manipulation (DGT) in LD 86.
> LD 86
REQ NEW
FEAT DGT
DMI 6
DEL 3
Configure the UDP Location Code (LOC) in LD 90.
> LD 90
REQ NEW
FEAT NET
TRAN AC1
TYPE LOC
LOC 504
FLEN 7
RLI 12
LDN 0
•
•
Configure the UDP HLOC in LD 90.
FEAT NET
TRAN AC1
TYPE HLOC
HLOC 501
DMI 6
Configure the HLOC in the Customer Data Block in LD 15.
> LD 15
REQ CHG
TYPE CDB
NET_DATA YES
ISDN YES
CLID YES
ENTRY <xx>
HLOC 501
•
Configure the Virtual Trunk route in LD 16.
> LD 16
REQ NEW
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92 Dialing Plan configuration
TYPE RDB
CUST 00
ROUT 120
DES VTRKNODE51
TKTP TIE
VTRK YES
ZONE 101
NODE 51
PCID H323
ISDN YES
MODE ISLD
DCH 12
IFC SL1
INAC YES
To configure the NRS (H.323 Gatekeeper/SIP Redirect Server):
•
•
Create H.323/SIP endpoints: MO, BO.
Create Numbering Plan entries:
— Choose type UDP-LOC.
— Add 504 for endpoint BO.
— Add 501 for endpoint MO.
For information about configuring H.323 Gatekeeper/SIP Redirect Server,
see IP Peer Networking Installation and Commissioning (NN43001-313).
To configure the SRG:
•
•
Create route and destination code to main office.
In the main office screen:
— Set the type of number to ESN LOC.
— The VoIP trunk access code field is empty.
— Set the main office Access Code Length to 1.
You can also include the LOC as the dial out when you configure the
route for the VoIP line pool. This allows users to dial fewer numbers. For
example, if 501 is configured as the dialout, and 6 is the destination
code, the user could dial 6+<main office DN>. Once the system
identifies the route (VoIP trunks) and drops the 6, it adds the LOC in
front of the DN and dials <LOC>+<DN>. In the case of redirected IP
Phones, the BUID is <destination code>+DN. The main office Access
code length, in this circumstance, is set to 1.
•
Dialing plan:
— Set Type to UDP.
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Dialing plan examples 93
— Set LOC to 504.
•
Set the BUID on the IP Phones to <VoIP trunk destination code> +
<LOC> + <DN>.
For more information on configuring the main office and NRS, see Branch
Office Installation and Commissioning (NN43001-314) and IP Peer
Networking Installation and Commissioning (NN43001-313). For more
information on configuring the SRG, see SRG50 Configuration Guide
(NN40140-500).
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95
Bandwidth Management
Contents
This section contains information on the following topics:
•
•
•
•
•
•
•
•
Introduction
CS 1000 supports Bandwidth Management on a network-wide basis so that
voice quality can be managed between multiple Call Servers.
Bandwidth management allows for codec selection and bandwidth
limitations to be placed on calls, depending on whether the calls are
intrazone or interzone.
Adaptive Network Bandwidth Management is an enhancement of Bandwidth
Management in which Quality of Service (QoS) metrics are used to
automatically lower available bandwidth.
Once all bandwidth is used, any additional calls are blocked or rerouted.
Keep this in mind when designing and implementing Network Bandwidth
Management
Codec negotiation
Codec refers to the voice coding and compression algorithm used by DSP.
Each codec has different QoS and compression properties.
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96 Bandwidth Management
IP Peer Networking supports the per-call selection of codec standards,
based on the type of call (interzone or intrazone). IP Peer Networking
supports the following codecs (with supported payload sizes in parentheses,
with the default value in bold):
•
•
•
•
G.711 A/mu-law (10 ms, 20 ms, and 30 ms)
G.729 A (10 ms, 20 ms, 30 ms, 40 ms, and 50 ms)
G.729 AB (10 ms, 20 ms, 30 ms, 40 ms, and 50 ms)
G.723.1 (30 ms) (though it can limit the number of DSP channels
available)
SRG 50 does not support G.723 codec.
T.38 for fax
•
ATTENTION
The G.XXX series of codecs are standards defined by the International
Telecommunications Union (ITU).
By default, the G.711 codec must be supported at both ends of a call.
Codec configuration is performed for each node and is independent of the
signaling gateway that is used on the node.
The payload size on the CS 1000 must be set to 30 msec in order to work
with the SRG.
IP Peer Networking performs codec negotiation by providing a list of codecs
that the devices can support. Use CS 1000 Element Manager to configure
the list of codec capabilities. See IP Peer Networking Installation and
Commissioning (NN43001-313) for instructions on configuring codecs.
The codec preference sequence sent over SIP/H.323 depends on the
bandwidth policy selected for the Virtual Trunk zone and the involved
telephones. For “Best Quality”, the list is sorted from best to worst voice
quality. For Best Bandwidth, the list is sorted from best to worst bandwidth
usage.
The G.711 codec delivers “toll quality” audio at 64 kbit/s. This codec is
optimal for speech quality, as it has the smallest delay and is resilient to
channel errors. However, the G.711 codec uses the largest bandwidth.
The G.729A codec provides near toll quality voice at a low delay. The
G.729A codec uses compression at 8 kbit/s. The G.729AB codec also
uses compression at 8 kbit/s.
The G.723.1 codec provides the greatest compression.
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Codec negotiation 97
SRG 50 does not support G.723 codec.
ATTENTION
If the payload sizes are set higher than the default values (for example, to support
a third-party gateway), then the local IP calls are affected by higher latency.
This is because the codec configuration applies to both IP Peer calls and local
IP (IP Line) calls.
G.711 A-law and mu-law interworking
In case the far end uses a different Pulse Code Modulation (PCM) encoding
law for its G.711 codec, systems that are configured as G.711 A-law also
include G.711 mu-law on their codec preferences list. Systems configured
as G.711 mu-law include G.711 A-law as their last choice. Therefore,
encoding law conversion is performed between systems with different laws.
Bandwidth management and codecs
Bandwidth management defines which codecs are used for intrazone calls
and interzone calls.
Bandwidth management enables administrators to define codec preferences
for IP Phone to IP Phone calls controlled by the same CS 1000 system in
the same zone. These calls are known as intrazone calls. This is different
than the codec preferences for calls between an IP Phone on the CS 1000
system to a Virtual Trunk (potentially an IP Phone on another CS 1000
system) or calls to IP Phones in another zone. These calls are known as
interzone calls.
For example, you may prefer high quality speech (G.711) over high
bandwidth within one system, and lower quality speech (G.729AB plus
Voice Activity Detection [VAD]) over lower bandwidth to a Virtual Trunk.
Such a mechanism can be useful when a system is on the same LAN as
the IP Phones it controls, but the other systems are on a different LAN
(connected through a WAN).
The Virtual Trunk usage of bandwidth zones is different than IP Phone
bandwidth usage. For Virtual Trunks, a zone number is configured in the
Route Data Block (RDB) (LD 16). The zone number determines codec
selection for interzone and intrazone calls (that is, Best Bandwidth or
Best Quality). See IP Peer Networking Installation and Commissioning
(NN43001-313) for information on configuring the RDB zone.
Bandwidth usage for Virtual Trunks is accumulated in its zone in order
to block calls that exceed the bandwidth availability in a specific zone.
However, the amount of bandwidth that is required to complete a given
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98 Bandwidth Management
call is not known until both call endpoints have negotiated which codec to
use. The bandwidth used for calculating the usage of a Virtual Trunk call
is determined by the preferred codec of the device that connects to the
Virtual Trunk. If the device is an IP Phone, the bandwidth calculations use
the preferred codec of the IP Phone, based on the codec policy defined for
the zones involved (that is, Best Bandwidth or Best Quality). Likewise, the
bandwidth calculations use the preferred codec of the Voice Gateway Media
Card for connections between a circuit-switched device (for example, a PRI
trunk) and a Virtual Trunk.
Codec selection
For every Virtual Trunk call, a codec must be selected before the media
path can be opened. When a call is set up or modified (that is, media
redirection), one of two processes occurs:
•
The terminating node selects a common codec and sends the selected
codec to the originating node.
•
The codec selection occurs on both nodes.
Each node has two codec lists: its own list and the far end’s list. In order to
select the same codec on both nodes, it is essential to use the same codec
selection algorithm on both nodes. Before the codec selection occurs, the
following conditions are met:
•
Each codec list contains more than one payload size for a given codec
type (it depends on the codec configuration). Payload size must be set
to 30 msec for proper functionality between the CS 1000 and the SRG.
•
Each codec list is sorted by order of preference (the first codec in the
near end’s list is the near end’s most preferred codec, the first codec in
the far end’s list is the far end’s preferred codec).
Codec selection algorithms
When the codec lists meet the above conditions, one of the following codec
selection algorithms selects the codec to be used:
•
•
•
H.323 Master/Slave algorithm
SIP Offer/Answer model
Best Bandwidth codec selection algorithm
H.323 Master/Slave algorithm
In the case of a Virtual Trunk call between Nortel and third-party equipment,
the H.323 Master/Slave algorithm is used.
The codec selection algorithm proposed by the H.323 standard involves a
Master/Slave negotiation. This is initiated each time two nodes exchange
their capabilities (TCS message). The Master/Slave information decides
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Codec negotiation 99
that one node is Master and the other node is Slave. The outcome of the
Master/Slave negotiation is not known in advance; it is a random result. One
node could be Master then Slave (or vice versa) during the same call.
Algorithm details The H.323 Master/Slave algorithm operates in the
following manner:
•
The Master node uses its own codec list as the preferred one and finds
a common codec in the far end’s list. In other words, the Master gets the
first codec in its list (for example, C1), checks in the far end’s list if it is a
common codec; if it is, C1 is the selected codec. Otherwise, it gets the
second codec in its list and verifies it against the far end, and so on.
•
The Slave node uses the far end’s list as the preferred one and finds
in its own list the common codec.
Issues caused by the H.323 Master/Slave algorithm The issues
caused by the Master/Slave algorithm are due to the random nature of the
Master/Slave information. In other words, one cannot predetermine the
codec that is used during a Virtual Trunk call.
The following are the issues associated with the H.323 Master/Slave
algorithm:
•
•
•
After an on-hold and off-hold scenario (which triggers Master/Slave
negotiation), the codec used for the restored call might be different than
the one used before on-hold, because the Master/Slave information
could have been changed.
When using Fast Start codec selection, a call from Telephone 1 (node1)
to Telephone 2 (node2) can use a different codec than a call from
Telephone 2 (node2) to Telephone 1 (node1), because the terminating
end is always Master.
For tandem calls, the Master/Slave information is not relevant. The
Master/Slave information is designed for use between two nodes only,
not between three or more nodes. It makes the codec selection for
tandem calls more complex and inefficient.
To solve the issues, another codec selection algorithm, not based on the
unpredictable Master/Slave information, is needed. Since any change to the
Master/Slave algorithm implies a change to the H.323 standard, the new
codec algorithm is used for Virtual Trunk calls between Nortel equipment.
SIP Offer/Answer model
The SIP codec negotiation is based on the Offer/Answer model with Session
Description Protocol (SDP).
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100 Bandwidth Management
The following three cases of codec negotiation are supported:
•
The calling user agent sends an SDP offer with its codec list in the
INVITE message with a sendrecv attribute. In this case, the called user
agent selects one codec and sends the selected codec in an SDP
answer. The SDP answer is included in the 200 OK message (which is
the response to the INVITE) with the sendrecv attribute.
This is the preferred method of operation.
•
•
The calling user agent sends an SDP offer with its codec list in the
INVITE message with a sendrecv attribute. The called user agent
returns more than one codec in the SDP answer. In the case that many
codecs are included in the response, the calling user agent picks the
first compatible codec from the called user agent’s list, and sends a new
SDP offer with a single codec to lock it in.
If the SDP of the calling user agent is not present in the INVITE
message, then the called user agent sends its codec list in an SDP
offer in the 200 OK message, with the sendrecv attribute. The calling
user agent selects one codec and sends the selected codec in an SDP
answer inside the ACK message, with sendrecv attribute.
For more information on this algorithm, refer to RFC 3264 – An
Offer/Answer Model with the Session Description Protocol (SDP).
Best Bandwidth codec selection algorithm
The “Best Bandwidth” codec selection algorithm solves the issues caused
by the H.323 Master/Slave algorithm. The “Best Bandwidth” algorithm
selects one common codec based on two codec lists. Every time the
selection is done with the same two lists, the selected codec is the same.
The “Best Bandwidth” codec decision is based on the codec type only,
it does not take into account the fact that some codecs, while generally
using less bandwidth, can consume more bandwidth than others at certain
payload sizes.
Algorithm details The selected codec is the type considered as the
best bandwidth codec type. To know whether one codec type has better
Table 8
Codec types
G.711 A-law
G.711 A-law
G.711 mu-law
G.729 A
G.711 mu-law
G.711 mu-law
G.711 mu-law
G.729 A
G.729 A
G.729 A
G.729 A
G.729 A
G. 729 AB
G. 729 AB
G. 729 AB
G. 729 AB
G. 723.1
G. 723.1
G. 723.1
G.729 A
G.711 A-law
G.711 mu-law
G.729 A
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G. 729 AB
G. 723.1
G. 729 AB
G. 723.1
G. 729 AB
G. 723.1
G. 729
AB
G. 729 AB
G. 729 AB
G. 729 AB
G. 723.1
G.729 A
SRG 50 does not support G.723 codec.
Interoperability between CS 1000 and SRG
The SRG is designed to interoperate with CS 1000 in a manner similar to
MG 1000B but with a limitation with respect to codec selection policy. Calls
between branch IP Phones and branch analog phones are based on the
interzone policy rather than the intrazone policy defined in the CS 1000
main office. The zone table is updated based on the intrazone policy.
The net result of this limitation is that calls between branch IP Phone
users and the branch PSTN or between the IP Phones and branch analog
phones will always use a Best Bandwidth codec. However, the calls will
be accounted for as Best Quality. This may impact the perception of call
quality in this scenario, but it will not result in early call blocking. There is
no impact to codec selection or bandwidth usage tracking for calls that
require WAN bandwidth.
Configuring Bandwidth Management parameters
The following sections describe how to configure Bandwidth Management
in a CS 1000 network. Nortel recommends that you read the Bandwidth
Management section in Converging the Data Network with VoIP
(NN43001-260) before using the following configuration information.
Zones
Bandwidth Management Zones are configured for each endpoint on a Call
Server. The Network Bandwidth Zone number determines if a call is an
intrazone call or an interzone call. Once that is determined, the proper
codec and bandwidth limit is applied to the call.
All of the endpoints on one Call Server are configured with Zone number to
identify all of the endpoints as being in a unique geographic location in the
network. In addition, Virtual Trunks are configured with a Zone number that
is different from the endpoint Zone numbers in the Call Server.
Configuration rules
There are four configuration rules for Bandwidth Management, as follows:
1. Each Call Server in the network must be configured with a unique VPNI,
with the only exception noted in point 2, next.
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102 Bandwidth Management
2. Branch office (MG 1000B and SRG) Call Servers must be configured
with the same VPNI as that of the main office Call Server with which
they register.
3. Nortel recommends that all the endpoints on a Call Server (IP Phones
and Voice Gateway Media Cards) be configured with the same Zone
number.
4. Virtual Trunks must be configured with a different Zone number than
the endpoints.
Network Planning
Before configuring Bandwidth Management in a CS 1000 network, follow
these steps:
Step Action
1
2
Choose unique VPNI for all Call Servers in the network.
Choose unique Bandwidth Zone numbers for all Call Servers in
the network. These are used when configuring the endpoints
(telephones and gateways) on the Call Server.
3
Choose unique Bandwidth Zone numbers for the Virtual Trunks in
the network.
4
5
Choose the codecs that will be enabled on each Call Server.
Identify what the interzone codec strategy will be (BB-Best
Bandwidth or BQ-Best Quality) for each zone in the network.
6
7
8
9
Identify what the intrazone codec strategy will be (BB-Best
Bandwidth or BQ-Best Quality) for each zone in the network.
Calculate the bandwidth available for intrazone calls for each zone
in the network.
Calculate the bandwidth available for interzone calls for each zone
in the network.
Calculate the bandwidth available for intrazone calls
—End—
Enabling codecs
In Element Manager, select the codecs that will be enabled for calls on the
Call Server, and define the associated parameters, such as payload size.
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Configuring Bandwidth Management parameters 103
codec or configure a new one in the VGW and IP phone codec profile
Peer Networking Installation and Commissioning (NN43001-313) for full
instructions on configuring a codec.
Figure 29
Configuring a codec
Configuring Bandwidth Management
The steps to configure Bandwidth Management on the Call Server are as
follows:
Step Action
1
2
Define a VPNI number in LD 15.
Configure the Bandwidth Management parameters for each zone on
the Call Server using either Element Manager (see "Configuration
using CS 1000 Element Manager" (page 104)) or LD 117 (see
•
Call Server zones that will be used for endpoints (telephones
and gateways) with the following properties:
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— Intrazone Preferred Strategy = Best Quality (BQ)
— Intrazone Bandwidth = default (1000000)
— Interzone Preferred Strategy = Best Bandwidth (BB)
— Interzone Bandwidth = maximum bandwidth usage allowed
between peer Call Servers
•
Call Server zones that will be used for Virtual Trunks with the
following properties:
— Intrazone Preferred Strategy = Best Quality (BQ)
— Intrazone Bandwidth = default (1000000)
— Interzone Preferred Strategy = Best Quality (BQ)
— Interzone Bandwidth = default (1000000)
3
Configure the IP Phone, DSP and Virtual Trunk data with the
corresponding zone numbers.
For example, for an IP Phone 2004 telephone in zone 8:
LD 11
REQ NEW
TYPE 2004
...
ZONE 8
...
—End—
Configuration using CS 1000 Element Manager
Zones are configured from the Zones web page, shown in Figure 30 "Zones
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Figure 30
Zones web
Configuration using LD 117
A new Bandwidth Management zone is configured in LD 117 using the
NEW ZONE command. An existing zone can be modified using the CHG
ZONE command.
LD 117 Configure a new or existing Bandwidth Management zone.
Description
Command
NEW | CHG ZONE <zoneNumber> [<intraZoneBandwidth> <intraZoneStrategy>
<interZoneBandwidth> <interZoneStrategy> <zoneIntent> <zoneResourceType>]
Configure a new zone (NEW) or change (CHG) an existing zone, where:
•
•
zoneNumber = 0-255
intraZoneBandwidth = Available intrazone bandwidth (Kbit/s); Nortel
recommends this value be set to the maximum value.
•
•
intraZoneStrategy = BB (Best Bandwidth) or BQ (Best Quality); Nortel
recommends this value be set to BQ.
interZoneBandwidth =
— For Call Server zone = Maximum bandwidth usage (in Kbit/s)
allowed between peer Call Servers
— For Virtual Trunk zones = 1000000 (Kbit/s)
•
•
interZoneStrategy = BB (Best Bandwidth) or BQ (Best Quality); Nortel
recommends this value be set to BB to conserve interzone bandwidth.
zoneIntent = type of zone, where:
— MO = Main Office (Call Server) zone
— BMG = Branch Media Gateway (for branch office zones)
— VTRK = Virtual Trunk zone
•
zoneResourceType = resource intrazone preferred strategy, where:
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Description
Command
— shared = shared DSP channels (default)
— private = private DSP channels
In CS 1000 Release 5.0, the zones that were described with BMG
designator stay with BMG one, all the other zones are provided with
MO designator. It is possible to update ZoneIntent using CHG ZONE
command.
Maintenance commands
Maintenance commands can be run from Element Manager or LD 117.
Maintenance using Element Manager
The PRT INTRAZONE and PRT INTERZONE commands are available in
Element Manager from the Zones web page, shown in Figure 30 "Zones
Procedure 7
Printing intrazone and interzone statistics for a zone
Step Action
1
Select IP Networks > Zones from the navigator.
The Zones web page opens.
2
Click Maintenance Commands for Zones (LD 117).
The Maintenance Commands for Zones web page opens, as
(page 107). This page lists all the configured zones.
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Figure 31
Maintenance Commands for Zones web page
3
Do one of the following:
•
To display intrazone statistics:
1. Select Print Interzone Statistics (PRT INTERZONE) from
the Action drop-down list.
2. Select a zone from the Zone Number drop-down list, by
doing of the following:
•
•
Select ALL to print statistics for all zones.
Select a specific zone number to display statistics for a
specific zone.
•
To display interzone statistics:
1. Select Print Intrazone Statistics per Local Zone (PRT
INTRAZONE) from the Action drop-down list.
2. Select a zone from the Near End Zone Number drop-down
list, by doing of the following:
•
•
Select ALL to print statistics for all zones.
Select a specific zone number to display statistics for a
specific zone.
4
Click Submit.
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108 Bandwidth Management
The Maintenance Commands for Zones web page reopens,
displaying the statistics for the specified zone or zones. A blank
field indicates that statistic is either not available or not applicable to
that zone.
an example of intrazone statistics.Figure 33 "Element Manager
interzone statistics" (page 108) shows an example of interzone
statistics.
Figure 32
Element Manager intrazone statistics
Figure 33
Element Manager interzone statistics
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—End—
Maintenance using LD 117
Use the PRT INTRAZONE or PRT INTERZONE commands in LD 117 to
view the intrazone or interzone statistics for specified zones.
ATTENTION
Do not use the PRT ZONE command — it has been replaced by the PRT
INTRAZONE and PRT INTERZONE commands.
LD 117 Print zone statistics.
Description
Command
PRT INTRAZONE [<zone>]
Print intrazone statistics for the identified zones, where:
zone = ALL or 0-255
•
The output of this command displays the following information:
•
•
•
•
•
•
•
Zone
Type = PRIVATE/SHARED
Strategy = BB/BQ
zoneIntent = MO/VTRK/BMG
Bandwidth = number of Kbps
Usage = number of Kbps
Peak = %
PRT INTERZONE [nearZone>] [<nearVPNI>] [<farZone>] [<farVPNI>]
Print interzone statistics for the specific VPNI zone; where:
•
nearZone = ALL or 0-255
The output of this command displays the following information:
•
•
Zone number = 0-255
Zone VPNI = 1-16283
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110 Bandwidth Management
Description
Command
•
•
•
Type= PRIVATE/SHARED
Strategy = BB/BQ
ZoneIntent = MO/VTRK
Adaptive Network Bandwidth Management
CS 1000 Release 4.5 supports the Adaptive Network Bandwidth
Management feature.
Description
The Adaptive Network Bandwidth Management feature enhances the
performance of Voice over Internet Protocol (VoIP) networks based
on real-time interaction. It provides the means to automatically adjust
bandwidth limits and take corrective action in response to Quality of Service
(QoS) feedback. This dynamic bandwidth adjustment maintains a high level
of voice quality during network degradation.
The Adaptive Network Bandwidth Management feature dynamically adapts
to QoS in the network and reduces the bandwidth available for interzone
calls if QoS degrades. Typically, each Call Server in the network has a zone
assigned to it. The Call Server keeps track of the bandwidth being used
between its own zone and zones belonging to other Call Servers. If the QoS
degrades between the Call Server’s zone and a particular zone belonging
to another Call Server, the available bandwidth is reduced automatically
between those two zones. When the QoS between the two zones improves,
then the bandwidth limit is allowed to return to normal.
When an IP Phone encounters degradation of the network, it informs the
Call Server through various QoS alarms. These QoS alarms (packet loss,
jitter, delay, and, for phase 2 IP Phones, R value) get reported to the Call
Server. Depending upon the rate of the incoming alarms and the value of the
alarms, the Call Server reduces the available bandwidth available to make
new calls. The Call Server will lower/limit the number of new calls allowed,
based on the available bandwidth. This prevents excessive calls being
placed on a network with limited bandwidth (resulting in poor voice quality).
Once the adjusted (lowered) bandwidth reaches its full capacity, new calls
are either routed to an alternate route (if available) using Network Alternate
Routing Service (NARS) , or Alternative Routing for Network Bandwidth
Management feature, or new calls are blocked. For further information about
the Alternative Routing for Network Bandwidth Management feature, see
Branch Office Installation and Commissioning (NN43001-314). The Call
Server continues to monitor the network throughout the network degradation
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Adaptive Network Bandwidth Management 111
period. When the degradation is removed or the performance of the network
improves, the allowable bandwidth returns to provisioned levels and the Call
Server gradually starts allowing new calls.
Essentially, Adaptive Network Bandwidth Management provides a fallback to
PSTN on QoS degradation for new calls. As a result, bandwidth is managed
and quality measured between all the zones across the entire network, and
when necessary corrective action is taken. Due to the real-time interaction
with the network, less maintenance is required for the network since the
system reacts automatically to network conditions.
With Adaptive Network Bandwidth Management, it is not necessary to
provision bandwidth parameters between every zone in the network. Rather,
the Call Server automatically learns of new zones in the network and applies
Adaptive Network Bandwidth Management to these new zones as required.
Therefore, as new Call Servers are added to the network, it is not necessary
to re-provision all the other Call Servers on the network to take into account
this new Call Server. Conversely, when Call Servers are removed from the
network, the remaining Call Servers age out the old Call Server information
and therefore, provide only up to date bandwidth information.
This feature operates between all IP Peer CS 1000 systems, including the
Media Gateway 1000B and Survivable Remote Gateway 50.
Call scenario
A call is requested from a telephone in VPNI 1/Zone 2 on Call Server A to a
telephone in VPNI 3/Zone 3 on Call Server B. Both zones have Adaptive
Network Bandwidth Management enabled.
1. Call Server A contacts the Network Redirect Server to obtain the
address of Call Server B.
2. Call Server A sends a call setup message to Call Server B, identifying
the calling telephone’s VPNI and zone.
3. Call Server B determines if there is sufficient bandwidth for the call, and
sends back the VPNI and zone of the called telephone.
4. Call Server A checks its bandwidth table to determine if there is sufficient
bandwidth available for the call from Call Server A to Call Server B.
5. If Call Server A determines there is enough bandwidth available, the call
is established; otherwise, alternate treatment is provided in the form
of blocking or rerouting the call.
Both Call Server A and Call Server B must consult their own bandwidth
tables to determine if there is enough bandwidth for the call to proceed.
(page 112) shows this scenario.
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112 Bandwidth Management
Figure 34
Call Progress with Adaptive Network Bandwidth Management
Zone bandwidth management and Adaptive Network Bandwidth
Management
Using Element Manager or the Command Line Interface (CLI), previously
configured zones (except Zone 0) can have the Adaptive Network Bandwidth
Management feature turned on or off. Once turned on, alarm threshold
levels and the QoS coefficients can be adjusted from the default values.
Adaptive Network Bandwidth Management cannot be enabled for Zone 0.
When Adaptive Network Bandwidth Management is enabled for a particular
zone on the Call Server, the zone appears in the zone table. The zone table
can be displayed using Element Manager or LD 117. When a call is made
from the configured zone to another zone, the bandwidth used appears in
the zone table.
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When a call is made from a zone with Adaptive Network Bandwidth
Management enabled, to a third party gateway, which has no zone, then the
zone of the Virtual Trunk (VTRK) is used and appears in the zone table.
Figure 35
Adaptive Network Bandwidth Management graph
When a Call Server receives a QoS alarm, the two zones that originated
the alarm are determined. Using this information, the Call Server reduces
the bandwidth limit between the two zones. This zone-to-zone bandwidth
limit (in effect at any particular time) is known as the Sliding Maximum
Bandwidth Limit and is a percentage of the Configured Interzone bandwidth
limit. The Sliding Maximum value is displayed using the prt interzone
command. The QoS Factor % is also displayed and is the percentage of
the Sliding Maximum versus the configured allowable bandwidth. The Call
Server checks the Network Bandwidth zone management tables for the
originating and terminating zones of the new call to determine the available
bandwidth for the call.
For more information about alarms, see Software Input Output
Reference—System Messages (NN43001-712).
When feedback indicates a significant QoS change in a zone, the Call
Server reduces the available bandwidth (Sliding Maximum Bandwidth Limit)
in the zone until the QoS reaches a satisfactory level. Once satisfactory
QoS is reached, the bandwidth is slowly raised until either the full bandwidth
is available or until QoS degrades again. Bandwidth changes can be
configured to be gradual (to reduce rapid swings and variations) or rapid.
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Multiple Appearance Directory Numbers (MADN) can exist on different
zones. Calls to an MADN are handled the same as other IP Phone calls,
and are subject to the same bandwidth limitations.
New SNMP alarms are provided to monitor the system. When the
bandwidth limit between zones is reduced below configured levels, an alarm
is raised. A Warning alarm and an Unacceptable alarm, each corresponding
to a drop below a configured threshold, are used. When the bandwidth
returns to normal, the alarm is cleared. If the bandwidth limit reaches
zero, an additional Unacceptable alarm is raised. These alarms allow the
system administrator to monitor the system and take corrective action
when required.
Adaptive Network Bandwidth Management configuration
parameters
Packet Loss (pl), Jitter (j) and Delay (d) measurements, along with the R
factor (r) in IP Phone 200x Phase II telephones, are used to calculate the
QoS level for the zones. The coefficients for these QoS measurements
— packet loss (Cpl), jitter (Cj), delay (Cd), and the R factor (Cr) — can
be configured and are used to calculate the rate of bandwidth change.
Increasing them from their default values causes the Sliding Maximum to
decrease faster in response to the specific QoS alarm.
The QoS Coefficient (CQoS) can be varied from its default value. Increasing
this value causes the Sliding Maximum to change more rapidly in response
to QoS alarms. However, making this value too large will result in loss
of overall bandwidth, as shown in Figure 36 "Effect of the default CQos
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Adaptive Network Bandwidth Management 115
Figure 36
Effect of the default CQos Coefficient
Figure 37
Effect of a higher CQoS Coefficient
Other configurable coefficients used in the calculation are the QoS
Coefficient (CQoS), QoS Response Time Increase (ZQRT), and QoS
Response Time Interval (ZQRTI). CQoS, Cr, Cd, Cpl, and Cj control the
rate of bandwidth decrease, while ZQRT and ZQRTI control the rate of
bandwidth increase.
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116 Bandwidth Management
The Call Admission Control (CAC) Validity Time Interval (CACVT) is used to
control the length of time that records from a Call Server are saved in the
Bandwidth Management table. If there have not been any calls between two
Call Servers within the configured time, the Call Server is removed from the
table. For example, if Call Server A has Call Server B in the table, and no
call has been placed between A and B for the CACVT time, then Call Server
A removes all Call Server B records in the table.
Limitations
Virtual Office IP Phones are not subject to bandwidth limitations. They
may not have the correct zone information configured. They can also be
controlled by a Call Server that is not responsible for the particular zone.
Thus, bandwidth management is not possible for these phones.
Feature packaging
The Adaptive Network Bandwidth Management feature requires the
following packages:
•
QoS Enhanced Reporting (PVQM) package 401
ATTENTION
Package 401, QoS Enhanced Reporting (PVQM), is required if the R value
from the Phase II IP Phones is to be reported and used in the calculations.
•
Call Admission Control (CAC) package 407
Configuration rules
The configuration rules for Adaptive Network Bandwidth Management are
as follows:
•
•
•
Each main office Call Server in a network must have a unique non-zero
VPNI.
All branch offices associated with a particular main office must have the
same VPNI as the main office Call Server.
All IP Phones (other than specific IP Softphone 2050) and DSP
endpoints on a Call Server must be configured for the same zone.
•
•
IP Softphone 2050 being used remotely must be configured for Zone 0.
Branch office systems must tandem all calls through the main office Call
Server to allow bandwidth monitoring and control. In this case, the media
path is direct between the branch office and any point in the network.
•
Trunk Route Optimization (TRO) must be disabled between the main
office Call Server and the SRG. In this case, the media path is direct
between the branch office and any point in the network.
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•
Adaptive Network Bandwidth Management parameters are configured
on the main office only and must not be configured at the branch offices.
Configuring Adaptive Network Bandwidth Management
The following is a summary of the tasks necessary to configure Adaptive
Network Bandwidth Management in the network.
1. Enable the Call Admission Control (CAC) package.
2. Configure CAC in Element Manager or LD 117:
a. Configure the VPNI on the main office and branch offices.
b. Configure both the main office and branch office zones at the main
office.
c. Configure the branch office zone on the SRG.
d. Configure the interzone and intrazone bandwidth limits at the main
office and SRG.
e. Enable Adaptive Network Bandwidth Management for the zones
on the main office Call Server.
f. If required, alter the Adaptive Network Bandwidth Management
parameters in keeping with the information in "Advanced
3. Tandem the outbound branch office calls by configuring the NRS.
4. Tandem the inbound branch office calls by creating a dialing plan which
routes all calls destined for the branch office through the main office.
Advanced Configuration Notes
1. The default values for Cpl, Cj, Cd, Cr and CQos can be increased to
increase the response time for Sliding Maximum changes. However,
increasing them can cause the Sliding Maximum to temporarily decrease
to a lower value then necessary, resulting in the needless blocking of
interzone calls.
2. Increasing the value of ZQRT will increase the speed at which the
Sliding Maximum increases. The same effect can be achieved by
decreasing ZQRTI. However, changing these values can cause the
Sliding maximum to oscillate until the network degradation is removed.
Configuration using Element Manager
Element Manager can be used to enable and configure the feature.
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118 Bandwidth Management
The zone must exist before it can be configured for Adaptive Network
Bandwidth Management. See IP Peer Networking Installation and
Commissioning (NN43001-313) for instruction on how to create and
configure basic properties of the zone.
To configure the Adaptive Network Bandwidth Management feature, select a
click Adaptive Network Bandwidth Management and CAC. The Adaptive
Network Bandwidth Management and CAC web page opens, as shown
ATTENTION
Do not configure Adaptive Networks Bandwidth Management for Zone 0 or Virtual
Trunk zones.
Figure 38
Adaptive Network Bandwidth Management and CAC web page
If the Adaptive Network Bandwidth Management feature is enabled using
the Enable Call Admission Control feature (ZCAC) check box, then the
other parameters can be adjusted as required.
and CAC web page, the field definitions, and their LD 117 command
equivalent.
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Adaptive Network Bandwidth Management 119
Table 9
Adaptive Network Bandwidth Management and CAC fields
LD 117
Field Title
Field Definition
equivalents
Enable Call Admission
Control Feature (CAC)
Control the CAC feature for the zone
ENL ZCAC
DIS ZCAC
•
•
Enable (check box selected)
disable (clear the check box)
QoS Response Time
Increase (ZQRT)
Bandwidth limit increment, as a percentage of
the QoS factor for the zone
CHG ZQRT
CHG ZQRTI
QoS Response Time
Interval (ZQRTI)
Time (in minutes) between bandwidth limit
increments
Warning Alarm Threshold
(ZQWAT)
A QoS value, which is lower than this value, but CHG ZQWAT
higher than the Critical (Unacceptable) Alarm
Threshold, triggers a Major Alarm.
Critical Alarm Threshold
(ZQUAT)
A QoS value, which is lower than this value,
triggers an Unacceptable (Critical) Alarm.
CHG ZQUAT
R Alarm Coefficient (CR)
The R (Cr) coefficient is used to calculate the
QoS value for the zone.
CHG CR
Packet Loss Alarm
Coefficient (CPL)
The Packet Loss (Cpl) coefficient is used to
calculate the QoS value for the zone.
CHG CPL
CHG CD
Delay Alarm Coefficient
(CD)
The Delay (Cd) coefficient is used to calculate
the QoS value for the zone.
Jitter Alarm Coefficient (CJ) The Jitter (Cj) coefficient is used to calculate the CHG CJ
QoS value for the zone.
Coefficient of QoS (CQoS)
The Coefficient of QoS (CQoS) is used to
calculate the overall QoS value for the zone.
CHG CQOS
Recent Validity Time Interval Amount of time (in hours) for zone-to-zone record CHG CACVT
(CACVT)
validity. Once this interval expires, records for
unused zones are purged from the tables.
Configuration using Command Line Interface
You can also configure the Adaptive Network Bandwidth Management
feature using LD 117.
LD 117 Configure Adaptive Network Bandwidth Management
Description
CHG CACVT <Zone> <Interval>
Command
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120 Bandwidth Management
Description
Configure the zone-to-zone record validity time interval, where:
Command
•
•
Zone = 1-255
Interval = 1-(48)-255
CHG CD <Zone> <Cd>
Change the Cd coefficient in the formula that determines how quickly an alarm
reduces the Sliding Maximum bandwidth for the identified zone, where:
•
Zone = 1-255
•
Cd = Cd coefficient = 1-(50)-100
CHG CPL <Zone> <Cpl>
Change the Cpl coefficient in the formula that determines how quickly an alarm
reduces the Sliding Maximum bandwidth for the identified zone, where:
•
Zone = 1-255
•
Cpl = Cpl coefficient = 1-(50)-100
CHG CJ <Zone> <Jitter>
Change the Cj coefficient in the formula that determines how quickly an alarm
reduces the Sliding Maximum bandwidth for the identified zone, where:
•
•
Zone = 1-255
Jitter = Jitter coefficient = 1-(50)-100
CHG CQOS <Zone> <QoS>
Change the QoS coefficient in the formula that determines how quickly an alarm
reduces the Sliding Maximum bandwidth for the identified zone, where:
•
Zone = 1-255
•
QoS = QoS coefficient = 1-(50)-100
CHG CR <Zone> <Cr>
Change the Cr coefficient in the formula that determines how quickly an alarm
reduces the Sliding Maximum bandwidth for the identified zone, where:
•
•
Zone = 1-255
Cr = Cr coefficient = 1-(50)-100
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Adaptive Network Bandwidth Management 121
Description
Command
CHG ZONE <zoneNumber> <intraZoneBandwidth> <intraZoneStrategy>
<interZoneBandwidth> <interZoneStrategy> [<zoneIntent> <zoneResourceType>]
Change the parameters of an existing zone, where:
•
•
•
zoneNumber = 1-255
intraZoneBandwidth = 1000000 (Mbit/s)
intraZoneStrategy = intrazone preferred strategy
— Best Quality = BQ
— Best Bandwidth = BB
•
•
interZoneBandwidth = 1000000 (Mbit/s)
interZoneStrategy = intrazone preferred strategy
— Best Quality = BQ
— Best Bandwidth = BB
•
•
zoneIntent = type of zone, where:
— MO = Main Office zone
— BMG = Branch Media Gateway (branch office) zone
— VTRK = Virtual Trunk zone
zoneResourceType = resource intrazone preferred strategy
— shared DSP channels (default) = shared
— private DSP channels = private
In CS 1000 Release 5.0 , the zones that were described with BMG designator
stay with BMG one, all the other zones are provided with MO designator. It is
possible to update ZoneIntent using the CHG ZONE command.
CHG ZQRT <Zone> <Incr>
Change ZQRT, which is Response time increase by percentage. It is used to
determine the increase to the Sliding Maximum for the identified zone, where:
•
•
Zone = 1-255
Incr = increase value in percentage = 1-(10)-100
CHG ZQRTI <Zone> <Interval>
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122 Bandwidth Management
Description
Command
Change the QoS Response Time Interval while alarms are not coming, in order
to increase the Sliding Maximum for the identified zone, where:
•
•
Zone = 1-255
Interval = interval in minutes = 1-(5)-120
CHG ZQUAT <Zone> <Thres>
Change the QoS Unacceptable Alarm Threshold value for the identified zone,
where:
•
•
Zone = 1-255
Thres = threshold value = 1-(75)-99
When the zone-to-zone QoS value drops below the threshold value, the alarm
is presented. When the zone-to-zone QoS value is greater than this threshold
value, this alarm is presented as being deactivated. This value must be below
the value of ZQWAT.
CHG ZQWAT <Zone> <Thres>
Change the QoS Warning Alarm Threshold value for the identified zone, where:
•
•
Zone = 1-255
Thres = threshold value = 1-(85)-99
When the zone-to-zone QoS value drops below the threshold value, the alarm
is presented. When the zone-to-zone QoS value is greater than this threshold
value, this alarm is presented as being deactivated. The value for ZQWAT must
be higher than the value of ZQUAT.
NEW ZONE <zoneNumber> [<intraZoneBandwidth> <intraZoneStrategy> <interZoneBandwidth>
<interZoneStrategy> <zoneIntent> <zoneResourceType>]
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Adaptive Network Bandwidth Management 123
Description
Command
•
•
•
•
•
zoneNumber = 1-255
intraZoneBandwidth = 1000000 (Mbit/s)
intraZoneStrategy = BQ (Best Quality)
interZoneBandwidth = 1000000 (Mbit/s)
interZoneStrategy = intrazone preferred strategy
— Best Quality = BQ
— Best Bandwidth = BB
•
•
zoneIntent = type of zone, where:
— MO = Main Office zone
— BMG = Branch Media Gateway (branch office) zone
— VTRK = Virtual Trunk zone
zoneResourceType = resource intrazone preferred strategy
— shared DSP channels (default) = shared
— private DSP channels = private
DIS ZCAC <Zone>
Disables the Call Admission Control (CAC) feature for the specified zone, where:
•
Zone = 1-255
This disables the feature on a zone-by-zone basis.
ENL ZCAC <Zone>
Enables the Call Admission Control (CAC) feature for the specified zone, where:
•
Zone = 1-255
This enables the feature on a zone-by-zone basis.
Maintenance commands
The Adaptive Network Bandwidth Management feature can be maintained
using Element Manager or LD 117.
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124 Bandwidth Management
Maintenance using Element Manager
The CAC parameters, intrazone statistics, and interzone statistics for one
of more zones are available in Element Manager from the Zones web
or interzone statistics, use Procedure 7 "Printing intrazone and interzone
Procedure 8
Displaying CAC parameters for one or more zones
Step Action
1
2
Select IP Network > Zones from the navigator.
Click Maintenance Commands for Zones (LD 117).
The Maintenance Commands for Zones web page appears,
as shown in Figure 31 "Maintenance Commands for Zones web
page" (page 107). This page lists all the configured zones and their
intrazone statistics by default.
3
4
Select Print Adaptive Network Bandwidth Management and CAC
Parameters (PRT ZCAC) from the Action drop-down list.
Select a zone from the Zone Number drop-down list, by doing one
of the following:
•
•
Select ALL to print statistics for all zones.
Select a specific zone number to display statistics for a specific
zone.
5
Click Submit.
The Maintenance Commands for Zones web page reopens,
displaying the statistics for the specified zone or zones. A blank
field indicates that statistic is either not available or not applicable to
that zone.
an example of the CAC parameters.
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Adaptive Network Bandwidth Management 125
Figure 39
Element Manager CAC parameters
—End—
Maintenance using LD 117
The same information can be displayed using commands in LD 117.
LD 117 Display Adaptive Network Bandwidth Management information
Description
Command
CLR CACR <nearZone> [<nearVPNI>] [<farZone>] [<farVPNI>]
Clear zone-to-zone record for near (VPNI-Zone) for far (VPNI-Zone), where:
•
•
•
•
nearZone = 0-255
nearVPNI = 1-16383
farZone = 0-255
farVPNI = 1-16383
PRT INTRAZONE [<zone>]
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126 Bandwidth Management
Command
Description
Print intrazone statistics for the identified zones, where:
•
zone = ALL or 1-255
The output of this command displays the following information:
•
•
•
•
•
•
•
•
Zone
State = ENL/DIS
Type = PRIVATE/SHARED
Strategy = BB/BQ
MO/VTRK/BMG = zoneIntent
Bandwidth = Kbps
Usage = Kbps
Peak = %
example of the output for this command.
PRT INTERZONE [<nearZone>] [<nearVPNI>] [<farZone>] [<farVPNI>]
Print interzone statistics for the specific VPNI zone; where:
•
•
•
•
nearZone = ALL or 0-255
nearVPNI = 1-16383
farZone = 0-255
farVPNI = 1-16383
The output of this command displays the following information:
•
•
•
•
•
•
Near end Zone
Near end VPNI
Far end Zone
Far end VPNI
State = ENL/DIS
Type = PRIVATE/SHARED
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Adaptive Network Bandwidth Management 127
Command
Description
•
•
•
•
•
•
•
•
•
Strategy = BB/BQ
MO/VTRK/BMG = zoneIntent
QoS factor = %
Bandwidth configured = Kbps
Sliding max = Kbps
Usage = Kbps
Peak = %
Call = Cph
Alarm = Aph
The report rows are grouped as:
•
•
First row = summary bandwidth usage per near zone
Next rows = bandwidth usage per near (VPNI- Zone) and far (VPNI - Zone)
example of the output for this command.
PRT ZCAC [<zone>]
Print CAC parameters for the specified zone, or for all zones, where:
•
zone = ALL or 0-255
The output of this command displays the following information:
•
•
•
•
•
•
•
•
Local ZONE = 0-255
State = ENL/DIS
CR = 1-100
CPL = 1-100
CD = 1-100
CJ = 1-100
CQOS = 1-100
ZQRT = 1-100
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128 Bandwidth Management
Command
Description
•
•
•
•
ZQRTI = 10-120
ZQUAT = 1-99
ZQWAT =1-99
CACVT = 1-255
PRT INTRAZONE command" (page 128) shows an example of the output
INTERZONE command" (page 129) shows an example of the output of
the PRT INTERZONE command.
Figure 40
Sample output for PRT INTRAZONE command
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Tandem Bandwidth Management overview 129
Figure 41
Sample output for PRT INTERZONE command
Tandem Bandwidth Management overview
In order for the main office to correctly keep track of all the bandwidth being
used to and from a branch office the call must be tandemed through the
main office. When calls are tandemed through the main office only the
signaling is tandemed, the actual voice bandwidth travels directly between
the source and destination.
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130 Bandwidth Management
Bandwidth utilization for the branch office is tracked at the main office and
can be displayed in LD 117 using the PRT INTERZONE command. In order
to provide the correct bandwidth utilization to the main office Call Server,
when a branch office is calling another node in the network, the calls must
be tandemed through the main office Call Server in both the inbound and
outbound direction.
Entering the main office Gateway endpoint identifier in the Tandem Endpoint
field for each branch office gateways configured on the NRS only provides
tandeming in the outbound direction from each branch office (from branch
office to main office).
In order to tandem calls through the main office in the inbound direction
(from main office to branch office), one must make use of the dialing plan
capabilities of the CS 1000 to first route the call to the main office. The main
office prepends a prefix to the dialed number and the number is routed
to the branch office.
Tandeming all branch office calls through the main office allows the main
office to keep track of the bandwidth being used at each branch office.
Application
This feature applies to the branch office and the Adaptive Bandwidth
Management feature. Specifically, it applies to calls made to and from the
branch office from either telephones registered locally at the branch office
(digital, analog [500/2500-type] telephones, and IP Phones) or trunks at
the branch office to another node in the network. It does not apply when
using branch office IP Phones that are registered with the main office (for
example, Normal Mode).
Patch Number
A patch is required on the main office in order to invoke this feature. The
patch number is MPLR 20259.
Dialing Plan Overview
Depending upon the type of dialing plan used in the network (Coordinated
Dialing Plan [CDP], or Uniform Dialing Plan [UDP] or a combination of both)
the general idea is to have all calls that are terminating at a branch office
first dial a number that will get routed to the main office associated with
that branch office. The main office recognizes this number as belonging to
the branch office and appends a tandem prefix to this number using Digit
Manipulation Index (DMI). The main office then routes the call to the branch
office while accounting for the additional bandwidth used.
office" (page 131) for an example of a tandem call.
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Dialing Plan Overview 131
Figure 42
A call between two branch offices tandems through the main office
Figure 43 "General legend" (page 132) shows a general legend for the
figures in the following section.
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132 Bandwidth Management
Figure 43
General legend
Network using Uniform Dialing Plan
The following section provides general network configuration for a network
using UDP only.
two or more main offices with their branch offices, within a larger network.
Callers within each main office/branch office region use UDP to place calls
between systems. Callers also use UDP to place calls across the IP network
to the other main office(s) and its (their) branch offices.
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Network using Uniform Dialing Plan 133
In a typical network, a full region uses a single Home Location Code
(HLOC). However, it is also possible, where the number of users requires it,
to have two or more codes, although using one for the main office and one
for each branch office is unlikely at best.
Figure 44
Scenario 1: UDP throughout the network
Common details
In general, if an HLOC is shared between two or more systems, the
provisioning at the main office gets more complex, unless all branch offices
share HLOC with the main office. That is, if the main office has two or more
HLOC, and one or more of these (but not necessarily the same one) is used
by every branch office, then provisioning is relatively straight forward.
the network configuration and the steps that a call takes during its setup.
Table 10
Configuration details for the general case
Call progress
steps
Configuration detail and call progress
during call setup
Region
1, 2, 3
1, 2, 3
1, 2, 3
UDP used for all calls within the region.
UDP used for region to region calls.
Prefixes for branch offices for regular calls are required for
all branch offices. May have additional prefixes for E-911
calls, if required, or may share prefixes.
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134 Bandwidth Management
Call progress
steps
Configuration detail and call progress
during call setup
Region
1
1
All branch offices are provisioned at the NRS to route all
outbound calls (from the branch office) through the main
office. (NRS tandem configuration).
1
1
2
3
4
Main office sends all UDP calls to destinations that are not
its own branch office to the NRS with unchanged dialled
digits.
Main office sends all UDP calls to destinations that are its
own branch office to the NRS with a specific gateway prefix
in front of the dialled digits.
1
All branch offices delete the prefix and any LOC codes, and
terminate the calls. May be to a local set or to a trunk.
2,3
Similar call setup steps take places for calls within region 2
and 3.
Differences when every branch office HLOC is shared with the main
office
configuration when the branch office HLOC is shared with the main office.
Table 11
Provisioning details for this case
Region
Provisioning detail
1
Provisioning on the main office requires parsing to only normal LOC
identification and HLOC deletion.
1
LOC values that are on branch offices may be provisioned as
extended LOC (> 3 digit codes).
1
The DMI for the branch office LOC inserts a gateway routing prefix in
front of the number.
2,3
Similar configuration, as above, applies to regions 2 and 3.
Call between two branch offices associated with the same main office
The following scenarios describe calls between two branch offices that
belong to the same main office. the different scenarios described below
vary in the manner in which the HLOC is architected; branch offices have
same HLOC as the main office, branch offices have a different HLOC than
the main office and so on.
Every branch office HLOC is shared with the main office
In the following example, the HLOC of all the branch offices and the HLOC
of the main office are all the same. See Figure 45 "Call flow for Scenario 1 -
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Network using Uniform Dialing Plan 135
Figure 45
Call flow for Scenario 1 - local call
1. The branch office user dials 6-395-3456. The system transmits
395-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 395-3456 to the main office.
3. The main office determines that this is LOC 39534, to another branch
office, with gateway routing prefix 552. The system inserts the prefix and
transmits 552-395-3456 to the NRS. The NRS checks its provisioning,
and determines that all calls to prefix 552 are to be sent to branch office
A2; it directs the call to the branch office.
4. The main office sends the call to 552-395-3456 to the branch office. The
branch office deletes the prefix and the HLOC, and rings set 3456.
No branch office HLOC is shared with the main office, but can be shared
with another branch office
In this example, the HLOC of the branch offices are the same but the HLOC
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136 Bandwidth Management
Figure 46
Call flow for Scenario 1 - local call
1. The branch office user dials 6-395-3456. The system transmits
395-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 395-3456 to the main office.
3. The main office determines that this is LOC 39534 to another branch
office, with gateway routing prefix 552. The system inserts the prefix and
transmits 552-395-3456 to the NRS. The NRS checks its provisioning,
and determines that all calls to prefix 552 are to be sent to branch office
A2; it directs the call to the branch office.
4. The main office sends the call to 552-395-3456 to the branch office. The
branch office deletes the prefix and the HLOC and rings set 3456.
No branch office HLOC is shared with the main office or another branch
office
In this example, the HLOC is unique between all the branch offices and the
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Network using Uniform Dialing Plan 137
Figure 47
Call flow for Scenario 1- local call
1. The branch office user dials 6-395-3456. The system transmits
399-3456 to the branch office user dials 6-399-3456. NRS. The NRS
checks its provisioning, and determines that all calls are to be sent to
the main office; it directs the call to the main office.
2. The branch office sends the call to 399-3456 to the main office.
3. The main office determines that this is to another branch office,
with office prefix 552. The system inserts the prefix and transmits
552-399-3456 to the NRS. The NRS checks its provisioning, and
determines that all calls to prefix 552 are to be sent to branch office A2;
it directs the call to the branch office.
4. The main office sends the call to 552-399-3456 to the branch office. The
branch office deletes the prefix and the HLOC, and rings set 3456.
Call between branch offices associated with different main office
The following scenarios describe calls between two branch offices that
belong to different main offices. Note that the different scenarios described
below vary in the manner in which the HLOC is architected; branch offices
have same HLOC as the main office, branch offices have a different HLOC
than the main office and so on.
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138 Bandwidth Management
Every branch office HLOC is shared with the main office
this example, the branch office and the main office share the same HLOC.
Figure 48
Call to a remote branch office on the originator side
1. The branch office user dials 6-444-3456. The system transmits
444-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 444-3456 to the main office.
3. The main office determines that this is to another main office. The
system transmits 444-3456 to the NRS. The NRS checks its provisioning,
and determines that this call goes to main office B.
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Network using Uniform Dialing Plan 139
Figure 49
Call to remote branch office on the destination side
1. Main office B determines that this is to LOC 44434, which is a local
branch office with prefix 225. The system transmits 225-444-3456 to the
NRS. The NRS checks its provisioning, and determines that this call
goes to branch office B1.
2. The main office sends the call to 225-444-3456 to the branch office. The
branch office deletes the prefix, discovers the call is to its HLOC 444,
deletes the HLOC, and rings set 3456.
No branch office HLOC is shared with the main office, but can be
shared with another branch office
Call to remote branch office on the originator sideshows the first half of the
call (originator side of the call). Figure 51 "Call to remote branch office on
destination side" (page 141)shows the second half of the call (destination
side of the call).
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140 Bandwidth Management
Figure 50
Call to remote branch office on the originator side
1. The branch office user dials 6-444-3456. The system transmits
444-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 444-3456 to the main office.
3. The main office determines that this is to another main office. The
system transmits 444-3456 to the NRS. The NRS checks its provisioning,
and determines that this call goes to main office B.
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Network using Uniform Dialing Plan 141
Figure 51
Call to remote branch office on destination side
1. Main office B determines that this LOC plus digits is to a local branch
office with prefix 225. (If sharing this LOC with another branch office,
the extended LOC is 44434.) The system transmits 225-444-3456 to
the NRS. The NRS checks its provisioning, and determines that this call
goes to branch office B1.
2. The main office sends the call to 225-444-3456 to the branch office. The
branch office deletes the prefix, and the HLOC, and rings set 3456.
No branch office HLOC is shared with the main office or another branch
office
The following example shows a call between two branch offices. In this
example, the HLOC is unique between the main office and branch office.
second half of the call (destination side of the call).
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142 Bandwidth Management
Figure 52
Call to remote branch office on the originator side
1. The branch office user dials 6-444-3456. The system transmits
444-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 444-3456 to the main office.
3. The main office determines that this is to another main office. The
system transmits 444-3456 to the NRS. NRS checks its provisioning,
and determines that this call goes to main office B.
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Network using Uniform Dialing Plan 143
Figure 53
Call to remote branch office on destination side
1. Main office B determines that LOC 444 is to a local branch office with
prefix 225. The system transmits 225-444-3456 to the NRS. The NRS
checks its provisioning, and determines that this call goes to branch
office B1.
2. The main office sends the call to 225-444-3456 to the branch office.
The branch office deletes the prefix, discovers the call is to its HLOC,
deletes the HLOC, and rings set 3456.
Summary of provisioning procedures for Tandem Bandwidth
Management
Step Action
1
2
3
Enter the main office Gateway endpoint identifier in the Tandem
Endpoint field for each branch office gateway configured on the
NRS. This provides tandeming for outbound calls from a branch
office through its main office. See Step 1.
Plan the gateway routing prefixes, if not already done. At least one
prefix is needed per branch office, although any branch offices that
have a prefix for ESA 911 calls does not necessarily require another.
(These prefixes will be SPN - Special Number - entries.) See Step 2.
Provision the NRS to send all calls to a LOC without a gateway
routing prefix to the main office of that LOC, or to the main office
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144 Bandwidth Management
which provides service for the branch office using the LOC. See
Step 3.
4
5
Provision the NRS to send all calls to a LOC with a gateway routing
prefix to the branch office directly. Using the gateway routing prefix
and the Type of Number of SPN, the entries can be differentiated
from the normal LOC easily. See Step 4.
Provision the main office with the DGT table DMI to insert the
prefixes and set the Type of Number correctly. Create RLB RLI
entries to use these DMI for the VTRK route(s). One RLI per branch
office will be the minimum requirement. Note that calls from remote
systems will typically have the HLOC prefix, so this is defined here.
See Step 5.
6
Provision the main office with CDP DSC (mapped by the RLI into
Location Codes) sufficient to uniquely identify all of its branch offices
(using extended location codes, if required); use the RLI index
defined for each branch office as the RLI value of the LOC definition.
This is the route to the branch office. See Step 6.
7
8
9
Provision the main office and branch office with a home location
code (HLOC) or multiple codes to terminate all calls that should
terminate on this system. See Step 7.
Provision the main office to send all other LOC to the IP network
without prefixes. These are going to a remote main office. See
Step 8.
Provision the branch office with a terminating RLI with a DMI to
delete the LOC prefixes. See Step 9.
—End—
Provisioning Example of Tandem Bandwidth Management
Figure 54 "Provisioning example" (page 145) shows an example of the
network configuration.
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Network using Uniform Dialing Plan 145
Figure 54
Provisioning example
Procedure 9
Provisioning Tandem Bandwidth Management
Step Action
1
Enter the main office Gateway endpoint identifier in the Tandem
Endpoint field for each branch office GW configured on the NRS.
This provides tandeming for outbound calls from a branch office
through its main office.
(page 146) shows the tandem endpoint configuration in Element
Manager.
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146 Bandwidth Management
Figure 55
Tandem endpoint configuration in Element Manager
2
Plan the gateway routing prefixes. At least one prefix is needed per
branch office, although any branch offices that have a prefix for ESA
911 calls does not necessarily require another. (These prefixes will
be SPN - Special Number - entries if you are using ESA 911. In the
example these are LOC codes because network 911 is not being
used.) In our example the Branch office prefixes are 741 (branch
office B) and 742 (branch office A).
3
4
Provision the NRS to send all calls to a LOC without a gateway
routing prefix to the main office of that LOC, or to the main office
which provides service for the branch office using the LOC. In our
example the NRS is provisioned with 841 (for main office B) and 842
(for main office A).
Provision the NRS to send all calls to a LOC with a gateway routing
prefix to the branch office directly. Using the gateway routing prefix
and the Type of Number as used (LOC or SPN), the entries can be
differentiated from the normal LOC easily.
In our example the NRS is provisioned with 741-841 at branch office
B and 742-842 for branch office A.
5
Provision the main office with the DGT table DMI to insert the
prefixes and set the Type of Number correctly. Create RLB RLI
entries to use these DMI for the VTRK route(s). One RLI per branch
office will be the minimum requirement. Note that calls from remote
systems will typically have the HLOC prefix, so this is defined here
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Network using Uniform Dialing Plan 147
office B" (page 147) lists main office B DMI and RLI provisioning.
Table 12
Main office B DMI and RLI provisioning for calls in branch office B
Create a DMI
LD 86
Create an RLI
LD 86
REQ new
CUST 0
REQ new
CUST 0
FEAT rlb
RLI 50
FEAT dgt
DMI 50
DEL 0
ENTR 0
LTER no
ROUT 71
DMI 50
ISPN no
INST 741841
CTYP loc
6
Provision the main office with CDP DSC (mapped by the RLI into
Location Codes) sufficient to uniquely identify all of its branch offices
(using extended location codes, if required); use the RLI index
defined for each branch office as the RLI value of the LOC definition.
This is the route to the branch office.
Table 13
Main office B LOC provisioning for LOC 741 841
Create a CDP mapped to the LOC:
LD 87
REQ NEW
CUST 0
FEAT CDP
TYPE DSC
DSC 4030
FLEN 4
RLI 50
Create a CDP mapped to the LOC:
LD 87
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148 Bandwidth Management
7
Provision the main office and branch office with a home location
code (HLOC) or multiple codes to terminate all calls that should
terminate on this system.
office and branch office HLOC provisioning.
Table 14
Main office B and branch office B
Create a DMI
LD 86
Create an HLOC
LD 90
REQ new
CUST 0
FEAT dgt
DMI 61
REQ new
CUST 0
FEAT net
TRAN ac1
TYPE hloc
HLOC 841
DMI 61
DEL 3
ISPN no
Repeat the above for all the main offices and branch offices.
8
Provision the main office to send all other LOC to the IP network
without prefixes. These are going to a remote main office.
office system " (page 148) lists main office B LOC provisioning for
LOC to remote main office. The Main Office A is LOC 842.
Table 15
Main office B LOC provisioning for LOC to remote main office system
Create an RLI
LD 86
Create a LOC
LD 90
REQ new
CUST 0
REQ NEW
CUST 0
FEAT rlb
RLI 51
FEAT NET
TRAN AC1
TYPE LOC
LOC 842
FLEN 7
ENTR 0
LTER no
ROUT 71
RLI 51
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Network using Uniform Dialing Plan 149
office systems " (page 149) lists main office A LOC provisioning for
LOC to the remote office. The Main office B is LOC 841
Table 16
Main office A LOC provisioning for LOC to remote main office systems
Create an RLI
LD 86
Create a LOC
LD 90
REQ new
CUST 0
REQ NEW
CUST 0
FEAT rlb
RLI 71
FEAT NET
TRAN AC1
TYPE LOC
LOC 841
FLEN 7
ENTR 0
LTER no
ROUT 75
RLI 71
9
Provision the branch office with a terminating RLI with a DMI to
delete the LOC prefixes.
Table 17
Branch office terminating RLI provisioning
Create a DMI
LD 86
Create an HLOC
LD 90
REQ new
CUST 0
FEAT dgt
DMI 61
REQ NEW
CUST 0
FEAT net
TRAN ac1
TYPE hloc
HLOC 741
DMI 61
DEL 6
ISPN no
—End—
Network using mixed Coordinated Dialing Plan and Uniform Dialing Plan
The following section provides general details of the network setup. shows
an example of a mixed network configuration.
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150 Bandwidth Management
Figure 56
UDP between main offices and CDP inside the main office region
details for a mixed network.
Table 18
Provisioning details for this case
Region
1, 2, 3
Provisioning detail
CDP used for all calls within the region.
UDP used for region to region calls.
1, 2, 3
1, 2, 3
Prefixes for branch offices for regular calls not required. May still
have prefixes for E-911 calls, if required.
1
1
All branch offices are provisioned at the NRS to route all calls
through the main office.
Main office sends all UDP calls to destinations that are not its
own branch office to the NRS with unchanged dialled digits.
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Network using Uniform Dialing Plan 151
Region
Provisioning detail
1
Main office sends all UDP calls to destinations that are its own
branch office to the NRS after deleting the HLOC and converting
to CDP.
2,3
Similar configuration, as above, applies to regions 2 and 3.
Call between two local branch offices
Figure 57 "Local call dials CDP" (page 151) shows the NRS Configuration
web page in Element Manager.
Figure 57
Local call dials CDP
1. The branch office user dials 3456 (CDP). The system transmits 3456 to
the NRS. The NRS checks its provisioning, and determines that all calls
are to be sent to the main office; it directs the call to the main office.
2. The branch office sends the call to 3456 to the main office.
3. The main office determines that this is to another branch office. The
system transmits 3456 to the NRS. The NRS checks its provisioning,
and determines that all calls to 3456 in this CDP domain are to be sent
to branch office A2; it directs the call to the branch office.
4. The main office sends the call to 3456 to the branch office. The branch
office rings set 3456.
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152 Bandwidth Management
Abnormal case - calls originating using UDP, but terminating
using CDP
Figure 58
Call flow for Scenario 2 - local call dial UDP
1. The branch office user dials 6-395-3456. The system transmits
395-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 395-3456 to the main office.
3. The main office determines that this is to another branch office, using
CDP. The system deletes the HLOC and transmits 3456 to the NRS.
The NRS checks its provisioning, and determines that all calls to 3456
from this CDP region are to be sent to branch office A2; it directs the call
to the branch office.
4. The main office sends the call to 3456 to the branch office. The branch
office rings set 3456.
Call between branch offices associated with different main offices
(originator side)" (page 153), the first half of the call is shown (the originator
side of the call).
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Network using Uniform Dialing Plan 153
Figure 59
Call flow for Scenario 2 - local call to remote branch office (originator side)
1. The branch office user dials 6-444-3456. The system transmits
444-3456 to the NRS. The NRS checks its provisioning, and determines
that all calls are to be sent to the main office; it directs the call to the
main office.
2. The branch office sends the call to 444-3456 to the main office. The
main office determines that this is to another main office. The system
transmits 444-3456 to the NRS. The NRS checks its provisioning, and
determines that this call goes to main office B.
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154 Bandwidth Management
Figure 60
Call to remote branch office on the destination side
1. The main office B deletes the HLOC, and determines that this is to a
local branch office. The system transmits 3456 to the NRS. The NRS
checks its provisioning, and determines that for this CDP region this call
goes to branch office B1.
2. The main office sends the call to 3456 to the branch office. The branch
office rings set 3456.
Network using Coordinated Dialing Plan
The following section provides general details of network setup.
Figure 61 "Full CDP network" (page 155) shows a full CDP network
configuration.
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Network using Coordinated Dialing Plan 155
Figure 61
Full CDP network
details for a full CDP network.
Table 19
Provisioning details for this case
Region
1, 2, 3
Provisioning detail
CDP used for all calls within the region.
CDP used for region to region calls.
1, 2, 3
1, 2, 3
All CDP numbers must be sufficiently long to allow unique
termination of the calls. That is, every main office/branch office
region requires its own LSC to ensure that all numbers are
unique.
1, 2, 3
Prefixes for branch offices for regular calls are required. May
have additional prefixes for E-911 calls, if required, or may share
prefixes.
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156 Bandwidth Management
Region
Provisioning detail
1
All branch offices are provisioned at the NRS to route all calls
through the main office.
1
1
Main office sends all CDP calls to destinations that are not its
own branch office to the NRS with unchanged dialled digits.
Main office sends all CDP calls to destinations that are its own
branch office to the NRS with a specific gateway prefix in front of
the dialled digits.
1
All branch offices delete the prefix and terminate the calls. May
be to a local set or to a trunk.
2,3
Similar configuration, as above, applies to regions 2 and 3.
Call between two local branch offices
Figure 62 "Call flow for Scenario 3 - local call" (page 156) shows the call
flow of a call between two local branch offices.
Figure 62
Call flow for Scenario 3 - local call
1. The branch office user dials 43456. The system transmits 43456 to the
NRS. The NRS checks its provisioning, and determines that all calls are
to be sent to the main office; it directs the call to the main office.
2. The branch office sends the call to 43456 to the main office.
3. The main office determines that this is to another branch office, with
office prefix 552. The system inserts the prefix and transmits 552-43456
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Network using Coordinated Dialing Plan 157
to the NRS. The NRS checks its provisioning, and determines that all
calls to prefix 552 are to be sent to branch office A2; it directs the call
to the branch office.
4. The main office sends the call to 552-43456 to the branch office. The
branch office deletes the prefix and LSC 4, and rings set 3456.
Call between branch offices associated with different main offices
(originator side)" (page 157), the first half of the call is shown (originator
side of the call).
Figure 63
Call flow for Scenario 3 - calls to remote branch office (originator side)
1. The branch office user dials 53456. The system transmits 53456 to the
NRS. The NRS checks its provisioning, and determines that all calls are
to be sent to the main office; it directs the call to the main office.
2. The branch office sends the call to 53456 to the main office.
3. The main office determines that this is to another main office. The
system transmits 53456 to the NRS. The NRS checks its provisioning,
and determines that this call goes to main office B.
(destination side)" (page 158), the second half of the call is shown
(destination side of the call).
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158 Bandwidth Management
Figure 64
Call flow for Scenario 3- calls to remote branch office (destination side)
1. Main office B determines that this is to a local branch office with prefix
225. The system transmits 225-53456 to the NRS. The NRS checks its
provisioning, and determines that this call goes to branch office B1.
2. The main office sends the call to 225-53456 to the branch office. The
branch office deletes the prefix and LSC, and rings set 3456.
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159
Alternative Call Routing for Network
Bandwidth Management
Contents
This section contains information on the following topics:
•
•
•
•
•
•
•
•
•
Description
Alternative Call Routing for Network Bandwidth Management (NBWM) is
available for CS 1000 Release 4.5 and CS 1000 Release 5.0 .
Alternative Call Routing for Network Bandwidth Management (NBWM)
allows a station-to-station call (that is, a call that does not use a trunk)
between a branch office and main office to overflow to traditional routes.
Overflow can occur if there is insufficient interzone bandwidth available
to carry the call or if the Quality of Service (QoS) has degraded to
unacceptable levels. The feature also applies to station-to-station calls
from one branch office to another branch office, provided both stations are
registered to the same main office.
(page 160) shows a diagram of a typical network where Alternative Call
Routing for NBWM can be used. It illustrates the sequence of events that
are explained in the information following the diagram.
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160 Alternative Call Routing for Network Bandwidth Management
Event number 1 pertains to the originally dialed call that experiences low
bandwidth or unacceptable QoS conditions. Event number 2 pertains to
the alternately routed call.
Figure 65
Example of Alternative Call Routing for NBWM in operation
There are multiple choices of alternate routes provided for the overflowed
calls. Network administrators who do not want calls to be blocked, but
have a limited amount of bandwidth available, want to overflow calls
to conventional trunks, (Public Switched Telephone Network [PSTN] or
TIE/Meridian Customer Defined Network [MCDN]). This feature allows
calls to be routed by overflowing them, trading off the capital cost of WAN
bandwidth against the incremental cost of overflowed calls.
Nortel recommends that this feature be used with DID (Direct Inward Dial)
numbers. This allows calls that are rerouted over the PSTN to ring the
intended telephone directly. It is possible to use this feature without having
DID, so that when the call gets rerouted over the PSTN the call reaches an
attendant console or a specific telephone.
When there is insufficient bandwidth for a station-to-station call, the
Alternative Call Routing for NBWM feature uses a trunk for a call which
would not normally use a trunk.
Prior to the introduction of the Alternative Call Routing for NBWM feature,
there was no alternate routing mechanism for the following types of
station-to-station calls:
•
•
Branch office calls to or from the main office
Branch office calls to or from another branch office controlled by the
same main office
ATTENTION
The term branch office refers to Media Gateway 1000B (MG 1000B) and
Survivable Remote Gateway (SRG) systems in this document.
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Description 161
This feature enables alternate routing to occur for branch office users
registered to the main office, when they place interzone, station-to-station
calls to main office stations or branch office stations.
The Alternative Call Routing for NBWM feature can be triggered by
operation of either of the following features:
•
•
NBWM
Adaptive Network Bandwidth Management
For more information on Network Bandwidth Management and Adaptive
Network Bandwidth Management, see IP Peer Networking Installation and
Commissioning (NN43001-313).
The Alternative Call Routing for NBWM feature can be implemented to
operate in one of the following two ways:
•
Bandwidth insufficiency: Alternately routes station-to-station IP network
calls that encounter lack of bandwidth or poor QoS in the originating or
terminating zone as the calls are being established
•
All Calls: Alternately routes station-to-station IP network calls,
regardless of available bandwidth
Configure one of these two modes of operation (Bandwidth insufficiency or
All Calls) on a zone basis.
The Alternative Call Routing for NBWM feature deals only with interzone
calls. This feature is configurable for main office and branch office zones,
not virtual trunk zones.
It is possible to configure this feature to reroute all calls to MCDN or PSTN
routes and still maintain the main office-branch office architecture.
ALTPrefix
When the Alternative Call Routing for NBWM feature is invoked, the dialed
number is modified with a prefix (called the ALTPrefix) inserted at the
beginning of the digit string. The ALTPrefix can have a maximum of seven
digits.
The system handles the call with the ALTPrefix inserted as if the user had
dialed the digits.
The ALTPrefix applies to the zone in which the call originates. There is one
ALTPrefix for each zone that requires alternate routing.
All Call Servers must be able to translate the ALTPrefixes for all zones that
require Alternative Call Routing for NBWM.
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162 Alternative Call Routing for Network Bandwidth Management
How Alternative Call Routing for NBWM works
When a user dials a station-to-station call between two different zones (that
is, the calling and called telephones are not located in the same geographic
area) and the bandwidth limit has been reached, then the Alternative Call
Routing feature is invoked.
The following steps occur in Alternative Call Routing scenarios.
Call from branch office telephone in Normal Mode
1. The interzone station-to-station call is attempted. There is insufficient
bandwidth available to make the call.
2. The main office inserts the ALTPrefix associated with the originator’s
zone (the branch office in this case) before the dialed digits.
3. The main office uses Coordinated Dialing Plan (CDP) or Vacant Number
Routing (VNR) to route the call to a virtual trunk.
4. A request is made to the NRS in order to determine the endpoint. The
NRS returns the branch office address as the endpoint.
5. The call is routed on the virtual trunk to the branch office.
6. The branch office treats the ALTPrefix as a steering code. The
branch office determines the Route List Index (RLI) and uses a
Digit Manipulation Index (DMI) to change the dialed number into a
PSTN/MCDN format number.
7. The call is routed over the PSTN/MCDN trunks to the telephone at the
main office or other branch office.
Call from main office telephone to branch office telephone
1. The interzone station-to-station call is attempted. There is insufficient
bandwidth available to make the call.
2. The main office inserts the ALTPrefix associated with the originator’s
zone (the main office in this case) before the dialed digits.
3. The main office uses CDP and recognizes the ALTPrefix as a steering
code.
4. The main office determines the RLI and uses a DMI to change the
dialed number into a PSTN/MCDN format number.
5. The call is routed over the PSTN/MCDN to the telephone at the branch
office.
With the ALTPrefix digits inserted, the resulting number must be one that
CDP or VNR software can process as a steering code.
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Description 163
For SRG, the main office manipulates the digits in the call into a format
appropriate for routing from the SRG on PSTN/MCDN trunks.
The steering code allows the call to be translated and referenced to a Route
List Index from which the system chooses the alternate route. The digit
manipulation capability of CDP allows digits (such as the ALTPrefix) to be
deleted and it can insert digits so the resulting number is appropriate for
the alternate route choice selected (PSTN or MCDN). For example, you can
manipulate the digits for PSTN routes so that DID destinations are dialable.
Dialing plans
Consider the many ways that calls can be dialed in a network. A
station-to-station call can be dialed using the following:
•
•
Directory Number (DN)
Coordinated Dialing Plan DN (starting with a Local Steering Code or
Distant Steering Code)
•
•
Uniform Dialing Plan DN (starting with a Location Code or Home
Location Code)
Transferable DN (TNDN) or Group Dialing Plan DN
Main offices and branch offices must be able to translate the calls, after the
ALTPrefix is inserted, using CDP or VNR. Take into account the format of
the calls with an ALTPrefix inserted prior to the rest of the digits.
Before deciding whether to program the steering codes as Distant Steering
Codes (DSC) or Trunk Steering Codes (TSC), take into account that the
system counts the digits in calls that start with a DSC.
Without Flexible Numbering Plan (FNP), DSC calls must have the same
number of digits as specified in LD 86 ESN (the NCDP prompt). With FNP,
fewer digits are acceptable, if the FLEN prompt is programmed correctly.
The maximum length of a CDP DN is seven digits (if DNXP is not equipped),
and ten digits, if DNXP is equipped. For more information see Dialing Plans
Reference (NN43001-283).
Calls preceded by the ALTPrefix can be handled by the Vacant Number
Routing (VNR) feature. Use Flexible Numbering Plan (FNP) software for
this type of routing.
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164 Alternative Call Routing for Network Bandwidth Management
Examples of Alternative Call Routing for NBWM in operation
A call from a branch office telephone to a main office telephone
a main office telephone" (page 164) shows an example of an alternately
routed call between two CS 1000 systems. The system shown on the left
is a main office. The system shown on the right is a branch office. Both
systems have access to the PSTN. The Wide Area Network (WAN) is shown
in the center with a gatekeeper/Network Routing Service (NRS) that can be
co-resident with a Signaling Server. IP Phones are also shown associated
with each system. Telephone A belongs to the main office. Telephone B
is a telephone at the branch office registered to the main office. There are
two bandwidth zones indicated by the dashed line (main office zone - MO
zone and branch office zone - BO2 zone). Telephone A and the Media
Gateway belong to the MO zone. The branch office and Telephone B belong
to the BO2 zone.
Figure 66
Call between a branch office telephone in Normal Mode and a main office
telephone
Telephone B calls Telephone A (the user dials DN 5262). In this example,
the call alternately routes to the PSTN. (The PSTN telephone number of
Telephone A is 613-966-5262.)
Configuration required Configure the following:
At the main office:
•
Use one of the dialing plan software packages to process the call, such
as CDP or NARS.
•
Provision an ALTPrefix for the BO2 zone.
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Description 165
•
Provision the ALTPrefix as a Steering Code at the main office. Build a
Route List Index (RLI) so these calls go to the virtual trunk or configure
the Vacant Number Routing feature to do the same.
At the branch office:
•
Provision the ALTPrefix as a steering code. Build a Route List Index and
DMI associated with this steering code that alternately routes these calls
to the PSTN with the correct digits.
At the NRS:
•
Create a CDP entry for the ALTPrefix in the branch office gateway.
Sequence of events in the call The information that follows refers to
NBWM" (page 166) illustrates the steps that occur in an alternately routed
station-to-station call. The diagram assumes that alternately routed calls
originate on Voice Gateway Media Cards at the branch office.
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166 Alternative Call Routing for Network Bandwidth Management
Figure 67
Digits dialed and outpulsed with Alternative Call Routing for NBWM
1. IP Phone User B in the BO2 zone dials IP Phone A in the main office
zone (DN 5262) (action labeled 1 in Figure 66 "Call between a branch
2. The main office Call Server determines there is insufficient bandwidth
between the two zones.
3. The Call Server inserts the ALTPrefix configured for the BO2 zone prior
to IP Phone A’s telephone number, dialed by the user.
Example: ALTPrefix assigned is 222. The dialed number becomes 222
5262.
4. The call is routed to the virtual trunk by CDP or VNR. The virtual trunk
sends a request to the NRS for address resolution. The digit string sent
to the NRS contains the ALTPrefix. The NRS returns the IP address of
the branch office endpoint to the virtual trunk.
5. The virtual trunk places the call to the branch office.
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Description 167
6. The branch office receives the call and recognizes the first part of the
number as a Steering Code. The call is steered to an RLI. The DMI
manipulates the number into a PSTN number and the branch office
outpulses the digits to the Central Office (CO) serving the branch office.
(This may be the same CO as the one serving the main office.) If the
alternate route has MCDN trunks in the BO2 zone, the call is outpulsed
on one of these trunks, after the branch office uses Digit Manipulation
(action labeled 2 in Figure 66 "Call between a branch office telephone in
2225 is programmed as a Steering Code at the branch office. Calls
starting with this Steering Code are handled by an RLI with a PSTN
trunk route as an entry. If the Public format number for the destination
telephone is a DID number, then the Digit Manipulation Index associated
with the PSTN route must:
•
•
delete 3 digits (remove the ALTPrefix 222)
insert 1613966 (in order to compose the DID number of the
destination telephone, which is 16139665262)
Program other Steering Codes to route calls to other locations properly
(for example, program Steering Code 2226 to route calls to another
branch office where 15063486XXX would have to be outpulsed). See
the section called "A call from a branch office telephone to another
7. Call comes into IP Phone A from the PSTN or MCDN trunks in the main
office zone (action labeled 3 in ) Figure 66 "Call between a branch office
A call from a main office telephone to a branch office telephone
telephone" (page 168) shows an alternately routed call between two CS
1000 systems. The system shown on the left is a main office. The system
shown on the right is a branch office. Both systems have access to the
PSTN. The Wide Area Network (WAN) is shown in the center with a
gatekeeper/Network Routing Service (NRS) that can be co-resident with a
Signaling Server. IP Phones are also shown associated with each system.
Telephone A belongs to the main office. Telephone B is a telephone at the
branch office registered to the main office. There are two bandwidth zones
indicated by the dashed line (main office zone [MO zone] and branch office
zone [BO2 zone]). Telephone A and the Media Gateway belong to the MO
zone. Telephones B and C and the branch office belong to the BO2 zone.
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168 Alternative Call Routing for Network Bandwidth Management
Figure 68
Call between a main office telephone and a branch office telephone
Telephone A calls Telephone B (the user dials DN 3121). In this example,
the call alternately routes to the PSTN. (The PSTN telephone number of
Telephone B is 613-966-3121.)
Configuration required Configure the following:
At the main office:
•
Use one of the dialing plan software packages to process the call, such
as CDP or NARS.
•
•
Provision an ALTPrefix for the MO zone.
Provision the ALTPrefix as a Steering Code at the main office. Build a
Route List Index (RLI) so these calls go to the virtual trunk or configure
the Vacant Number Routing feature to do the same.
•
•
Provision the ALTPrefix as a Steering Code. Build a Route List Index
(RLI). Translate these calls to go to this RLI which contains PSTN or
MCDN trunks.
Set up a Digit Manipulation Index (DMI) to outpulse the correct digits to
the branch office telephone.
At the branch office:
No provisioning required.
•
At the NRS:
•
No provisioning required.
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Description 169
Sequence of events in the call The information that follows refers to
Figure 69
Digits dialed and outpulsed with Alternative Call Routing for NBWM
1. IP Phone User A in the MO zone dials IP Phone B in the BO2 zone
(DN 3121) (action labeled 1 in Figure 68 "Call between a main office
2. The main office Call Server determines there is insufficient bandwidth
between the two zones.
3. The Call Server inserts the ALTPrefix configured for the MO zone
prior to IP Phone B’s telephone number. The rebuilt number contains
the ALTPrefix.
Example: ALTPrefix assigned is 777. The dialed number becomes 777
3121.
4. The call is routed by CDP or VNR to the outgoing route using an RLI.
Example: 7773 is a steering code, translated to go to an RLI where
there is one entry, a PSTN trunk route.
5. The main office manipulates the digits and the call is routed to the PSTN
in the MO zone (action labeled 2 in Figure 68 "Call between a main
Example: Use DMI to delete and insert digits on the PSTN trunk route.
•
delete 3 digits (remove the ALTPrefix 777)
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170 Alternative Call Routing for Network Bandwidth Management
•
insert 966 (in order to compose the DID number of the destination
telephone which is 9663121, a local call.)
6. The call terminates on IP Phone B from the PSTN in the BO2 zone
(action labeled 3 in Figure 68 "Call between a main office telephone and
A call from a branch office telephone to another branch office
telephone
office telephone" (page 171) shows an alternately routed call between three
systems: one main office and two branch offices.
All three systems are configured with the same Virtual Private Network
Identifier (VPNI). All systems have virtual trunks.
The Wide Area Network (WAN) is shown with a gatekeeper/Network Routing
Service (NRS) that can be co-resident with one of the Signaling Servers.
Telephone F belongs to branch office 1 in Normal Mode and telephone B
is a branch office 2 telephone in Normal Mode.
There are three bandwidth zones indicated by the dashed lines (main office
zone [MO zone], branch office zone [BO2 zone], and branch office 1 zone
[BO1 zone]). Telephone F and branch office 1 belong to the BO1 zone.
Telephone B and the other branch office belong to the BO2 zone.
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Description 171
Figure 70
Call between a branch office telephone and another branch office telephone
Telephone B in the BO2 zone calls Telephone F in the BO1 zone (the user
dials 6020). In this example, the call alternately routes to the PSTN. (The
PSTN telephone number of Telephone F is 506-348-6020.)
Configuration required Configure the following:
At the main office:
•
Use one of the dialing plan software packages to process the call, such
as CDP or NARS.
•
•
Provision an ALTPrefix for the BO2 zone.
Provision the ALTPrefix as a Steering Code at the main office. Build a
Route List Index (RLI) so these calls go to the virtual trunk or configure
the Vacant Number Routing feature to do the same.
At the branch office:
•
Provision the ALTPrefix as a steering code. Build a Route List Index and
DMI associated with this steering code that alternately routes these
calls to the PSTN.
At the NRS:
•
Create a CDP entry for the ALTPrefix in the branch office gateway.
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172 Alternative Call Routing for Network Bandwidth Management
Sequence of events in the call The information that follows refers to
Figure 71
Digits dialed and outpulsed with Alternative Call Routing for NBWM
1. The user of IP Phone B in the BO2 zone dials IP Phone F in the BO1
zone (DN 6020) (action labeled 1 in ) Figure 70 "Call between a branch
2. The main office Call Server determines there is insufficient bandwidth
between the BO2 and BO1 zones.
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Description 173
3. The Call Server inserts the ALTPrefix configured for the BO2 zone prior
to IP Phone F’s telephone number, dialed by the user.
Example: ALTPrefix assigned is 222. The dialed number becomes 222
6020.
4. The call is routed to the virtual trunk by CDP or VNR. The virtual trunk
sends a request to the NRS for address resolution. The digit string sent
to the NRS contains the ALTPrefix. The NRS returns the IP address of
the branch office endpoint to the virtual trunk.
5. The virtual trunk places the call to the branch office.
The branch office receives the call and recognizes the first part of the
number as a Steering Code. The call is steered to an RLI. The DMI
manipulates the number into a PSTN number and the branch office
outpulses the digits to the Central Office (CO) serving the branch office.
(This may be the same CO as the one serving the main office.) If the
alternate route has MCDN trunks in the BO2 zone, the call is outpulsed
on one of these trunks, after the branch office uses Digit Manipulation
(action labeled 2 in Figure 70 "Call between a branch office telephone
Example: The digit string 2226 is programmed as a Distant Steering
Code (DSC) at the branch office. Calls starting with this DSC are
handled by an RLI with PSTN trunks as an entry. If the Public format
number for the destination telephone is a DID number, then the Digit
Manipulation Index associated with the PSTN route must:
•
•
delete 3 digits (remove the ALTPrefix 222)
insert 1506348 (in order to compose the DID number of the
destination telephone, which is 15063486020)
6. The call comes into IP Phone F from the PSTN in the BO1 zone (action
labeled 3 in Figure 70 "Call between a branch office telephone and
Alternative Call Routing for NBWM feature used in All Calls mode
The situations illustrated in Figure 66 "Call between a branch office
another branch office telephone" (page 171) can also apply to the operation
of the feature when it is active all the time, regardless of available bandwidth.
For an example of this, the information below applies to Figure 66 "Call
telephone" (page 164) when all calls are alternately routed.
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174 Alternative Call Routing for Network Bandwidth Management
The configuration and provisioning for the All Calls mode is exactly the
same as the regular Alternative Call Routing for NBWM feature except
that the Alternate Routing for All Calls option is selected in overlay 117
or Element Manager.
Operating parameters
This feature applies to all CS 1000 systems.
This feature applies to station-to-station, interzone calls. The call can be
between IP Phones or a TDM telephone at the main office and an IP Phone
at the branch office.
This feature is configurable for main office and branch office zones, not
virtual trunk zones.
The operation of the Alternative Call Routing for NBWM feature depends on
the proper configuration of the Network Bandwidth Management feature. If
Network Bandwidth Management encounters insufficient bandwidth for an
interzone call, the Alternative Call Routing for NBWM feature attempts to
re-route the call through the PSTN or TIE/MCDN route, if Alternative Call
Routing is configured for the originating zone.
Configure the Voice Gateway bandwidth zone and the IP Phone bandwidth
zone with the same number on each Call Server.
For alternately routed calls that originate at the branch office, the network
administrator must decide whether to program the main office to use
conventional PSTN or TIE/MCDN routes that physically terminate at the
branch office or to use conventional trunks at the main office. Outpulsed
digits on the PSTN trunks must be in the format required by the Central
Office serving the chosen system. For example, if the two systems are
served by the same Central Office, PSTN calls can use the trunks at the
main office. However, if the two systems are located at a great distance
from each other, it may make sense to program the main office to use trunks
at the branch office for alternately routed calls.
If there is a location with multiple NXX codes for DID users, it is possible
to translate and outpulse calls properly as long as each NXX has a unique
DN range. Non-DID users can be reached through the attendant console,
by deleting the ALTPrefix and the DN dialed and inserting the digits in the
Listed Directory Number (LDN).
When calls are rerouted to use the PSTN instead of the station-to-station
IP network, there may be a loss of feature functionality normally available
for station-to-station calls. Features that are not available over the PSTN
are not available to the user.
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Feature interactions 175
This feature does not apply to virtual trunk calls. It only applies to
station-to-station calls, and routes them over the PSTN or MCDN network.
Virtual Trunk calls already have this feature and can be alternately routed
using traditional methods (such as NARS), which is outside the scope of
this feature.
Alternative Call Routing for NBWM does not apply to users who are
registered in local mode to the Branch Office.
Calls that are in an ACD queue cannot be alternately routed by this feature.
These calls remain in the ACD queue until an ACD agent and sufficient
bandwidth are available.
Music on Hold is not affected by this feature.
A QoS0038 message prints out when insufficient bandwidth is detected
between two zones. A QoS0039 message prints out when the Alternative
Call Routing for NBWM feature is invoked.
If a user at a branch office attempts to make a conference call, Alternative
Call Routing for NBWM is not invoked.
Feature interactions
Call Redirections
The Alternative Call Routing for NBWM feature works with Call Transfer,
Call Forward All Calls, and Conference. Redirection operates as if the user
dialed the ALTPrefix manually.
Multiple Appearance Directory Number
If telephone endpoints sharing Multiple Appearance DN (MADN) are
configured in different zones, then this feature handles the call in the same
way that Network Bandwidth Management does. The zone number of the
telephone that is prime for the DN is the only zone considered.
Network Bandwidth Management
The Alternative Call Routing for NBWM feature does not detect insufficient
bandwidth. It reacts to insufficient bandwidth detected by the Network
Bandwidth Management and Adaptive Network Bandwidth Management
features. For more information on these two features, see IP Peer
Networking Installation and Commissioning (NN43001-313).
Network Class of Service
The telephone calling must have an NCOS assigned that allows the call
to use the alternate route.
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176 Alternative Call Routing for Network Bandwidth Management
Network Routing Service (NRS)
Both the main office and branch office must be registered on the Nortel NRS
for tandem routing to work for main office to branch office tandeming.
Trunk Route Optimization (TRO)
Disable Trunk Route Optimization between the branch office (or SRG) and
the main office in order to allow the tandeming required for the feature to
work.
Virtual Office
It is not possible to determine the real Bandwidth Zone of a telephone
logged in as a Virtual Office telephone.
Feature packaging
The Alternative Call Routing for NBWM feature requires the following
software package:
•
Coordinated Dialing Plan (CDP) package 59
The following software package is optional:
Flexible Numbering Plan (FNP) package 160
•
Feature implementation using Command Line Interface
Task summary list
The following is a summary of the tasks in this section:
Enable Alternative Call Routing for NBWM for a particular zone
Configure Alternate Prefix number for a particular zone and the All Calls
option, if desired.
Print Alternate Prefix number for a particular zone.
LD 117
Description
Command
ENL ZALT <zone>
Enable Alternative Call Routing for NBWM, where:
<zone>
Input zone number (0-255).
Configure the branch office zone using LD 117 at the main
office.
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Feature implementation using Command Line Interface 177
LD 117
Description
Command
CHG ZALT <zone> <ALTPrefix> [<re-route all calls>]
Change ALTPrefix number for zone, where:
<zone>
Input zone number (0 to 255).
Configure the branch office zone using LD 117 at the main
office.
<ALTPrefix>
A digit string, of up to 7 digits, added to the start of the
dialed number, if the call will not be routed through the
WAN (due to lack of bandwidth, poor QoS, or feature is
configured for all calls).
[<re-route all calls>]
Allow or Deny Alternative Call Routing for all calls, where:
•
•
(NO) = deny
YES = allow
LD 117
Description
Command
PRT ZALT <zone>
Print the ALTPrefix assigned to a particular zone and if the feature operates for all calls from that
zone, where:
<zone>
Input zone number (0 to 255).
If you do not input a zone number, the system prints the
information for all configured zones.
Sample printout
Alarms
Suppression
#
ZALT
ENL
ENL
DIS
Alternate Prefix
All Calls
YES
Time
10
11
12
100
101
102
50
0
YES
1000
NO
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178 Alternative Call Routing for Network Bandwidth Management
Feature implementation using Element Manager
Zone configuration
Use Element Manager to configure the specific zone properties.
1. Access the Zones web page to select Alternate Routing for Calls
between IP Stations.
2. Access the Alternate Routing for Calls between IP Stations web
page to configure the feature.
Accessing the Zones web page
Procedure 10
Accessing the Zones web page
Step Action
1
From the navigator, click IP Network > Zones.
Figure 72
Zones
2
3
Under Configuration click beside the Zone, or click the Zone you
are programming, to expand the choices under it.
Click Alternate Routing for Calls between IP Stations.
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Feature implementation using Element Manager 179
A web page appears that displays the configuration parameters
related to the Alternative Call Routing for NBWM feature. See Figure
Figure 73
Access the Alternate Routing for Calls between IP Stations
The Zone Number for the zone you selected in step 3 is displayed.
4
5
6
7
8
Select the Enable Alternate Routing feature (ENL_ZALT) check
box to enable the Alternative Call Routing for NBWM feature.
Enter a maximum of 7 digits in Alternate Routing Prefix Digits
(ALTPrefix).
Select the Re-route for All Calls (ALL_CALLS) check box to
enable the feature for all calls.
Enter a maximum of 4 digits in Alarm Suppression Time Period
(ZAST).
Click Submit to enter the data.
—End—
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180 Alternative Call Routing for Network Bandwidth Management
Diagnostics
Command Line Interface diagnostics
LD 117 Obtain status for a particular zone
Description
Command
STAT ZALT <zone>
Display Alternative Call Routing Status, where:
zone = bandwidth zone
If you do not enter a value in the zone field, you obtain a printout of the status of all configured zones.
The printout for all configured zones appears as follows:
#
Alternate Routing Status
10
11
12
ENL
DIS
DIS
LD 117 Obtain status for a branch office zone
Description
Command
STAT ZBR <zone>
Show status of specified branch office zone, where:
zone = bandwidth zone
If you do not enter a value in the zone field, you obtain a printout of the status of all configured
branch office zones.
The printout for all configured branch office zones appears as follows:
#
State
ENL
DIS
Flags
Des
10
11
12
TIM
BVW
TOR
LOC ALT
ESA ALT
DIS
Des entries are codes you assign to each branch office for your own records.
Element Manager diagnostics
Print zone Alternative Call Routing information
The print capability allows the following parameters to be viewed for each
zone:
•
•
enable/disable status of the Alternative Call Routing for NBWM feature
ALTPrefix digits
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Diagnostics 181
•
enable/disable status of the All Calls option
Procedure 11
Printing zone ALTPrefix
Step Action
1
From the navigator, click System > Maintenance. The Maintenance
web page appears. See Figure 74 "Maintenance web page" (page
Figure 74
Maintenance web page
2
Click Select by Functionality. The Select by Functionality web
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182 Alternative Call Routing for Network Bandwidth Management
Figure 75
Select by Functionality
3
Click Zone diagnostics. The Maintenance Commands for Zones
Figure 76
Maintenance Commands for Zones
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Diagnostics 183
4
Select Print Zone Alternate Prefix Information (PRT ZALT) from
the drop-down list beside Action. See
Figure 77
Maintenance Commands for Zones Action list
5
6
Select ALL or a zone number from the Zone Number drop-down list.
Click Submit.
—End—
Show the status of the Alternative Call Routing for NBWM feature
The Show Status functionality displays the enable/disable status of the
Alternative Call Routing for NBWM feature. The configured zones and the
status of the feature for each zone can be listed.
Procedure 12
Show Status
Step Action
1
From the navigator, click System > Maintenance. The Maintenance
web page appears. See Figure 74 "Maintenance web page" (page
2
3
Click Select by Functionality. The Select by Functionality web
Click Zone diagnostics. The Maintenance Commands for Zones
web page appears.
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184 Alternative Call Routing for Network Bandwidth Management
4
Select Show Alternate Routing Status (STAT ZALT) from the
Figure 78
Show Alternative Routing Status
5
6
Select ALL or a zone number from the Zone Number drop-down list.
Click Submit to enter the data
The display updates with the data associated with that zone.
—End—
Maintenance
Command Line Interface
LD 117 Enable and disable Alternative Call Routing for NBWM for a zone
Description
Command
ENL ZALT <zone>
Enable Alternative Call Routing for Bandwidth Management
for zone specified.
DIS ZALT <zone>
Disable Alternative Call Routing for Bandwidth Management
for zone specified.
The system responds with ok, if operation is successful.
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Maintenance 185
Description
Command
ENL ZBR <zone> [ALL] [LOC][ESA][TIM][ALT] Enable features for the branch office zone. If you do
not input features, then all are enabled.
DIS ZBR [ALL] [LOC][ESA][TIM][ALT] Disable features for the branch office zone. If you do not
input features, then all are disabled.
The system responds with ok, if operation is successful.
LD 117 Change and Print Zone Alarm Suppression Time Interval
Description
CHG ZAST <zone> <Alarm Suppression Interval>
Change Suppression Time Interval for QoS alarms related
Command
to Alternative Call Routing for NBWM feature for zone
specified. Time Interval is measured in seconds from the
time the last alarm was printed.
Default is 0. Range is 0 - 3600 seconds.
PRT ZAST <zone>
Print Suppression Time Interval for QoS alarms related
to Alternative Call Routing for NBWM feature for zone
specified. No entry for zone number results in printout for
all zones.
Sample printout:
#
Alarms Suppression Time
10
11
12
50
0
1000
Where # is the column header for zone number.
Element Manager maintenance
Enable branch office behavior for a zone
You can configure the Alternative Call Routing for NBWM feature while you
configure the zone’s branch office behavior.
Procedure 13
Enabling behavior at a zone
Step Action
1
From the navigator, click System > Maintenance. The Maintenance
web page appears. See Figure 74 "Maintenance web page" (page
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186 Alternative Call Routing for Network Bandwidth Management
2
3
4
Click Select by Functionality. The Select by Functionality web
Click Zone diagnostics. The Maintenance Commands for Zones
web page appears .
Select Enable a Zone’s Branch Office Behavior from the
Figure 79
Enable branch office behavior for a zone
5
6
7
Select the zone number you want to configure from the Zone
Number drop-down list.
Select the check box beside the Alternate Routing for Branch
option to enable the feature.
Click Submit. The display updates with new configuration data for
the zone you specified.
—End—
Suppress alarms
You can suppress alarms QoS0038 and QoS0039 for a configurable amount
of time (0 - 3600 seconds).
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Feature operation 187
Procedure 14
Suppress Alternative Call Routing for NBWM alarms
Step Action
1
Click IP Network > Zones.
The Zones web page appears.
2
3
4
Click beside the Zone, or click the Zone you are programming, to
expand the choices under it.
Enter digits in Alarm Suppression Time Period (ZAST). See
Click Submit to enter the data.
Figure 80
Alarm Suppression Time Period (ZAST)
—End—
Feature operation
No specific operating procedures are required to use this feature
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188 Alternative Call Routing for Network Bandwidth Management
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189
Emergency Services configuration
Contents
This section contains the following topics:
•
•
•
•
•
•
•
•
Overview
Support for access to emergency services by branch users in Normal Mode
is configured at the main office.
The key difference between the main office user and the branch user is the
route selected for the emergency call. An emergency call must be handed
off to the PSTN over a trunk at the central office that is geographically
closest to the caller—this means that there is normally an emergency trunk
in the main office, and one in each of the branch offices. An emergency call
originating from an SRG IP Phone must route from the main office Call
Server to the SRG so that the call can be sent on the SRG PSTN Trunks.
In Normal Mode, an IP Phone must have a Virtual Trunk available and
configured between the main office and branch office in order to complete
an emergency services call.
ATTENTION
Do not route ESA calls to a node that has no direct ESA trunks.
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190 Emergency Services configuration
Nortel recommends using the Emergency Services Access (ESA) feature.
This is the preferred method in North America, the Caribbean and Latin
America (CALA), and in those countries that are members of the European
Union (EU). ESA provides specific features and capabilities required by
legislation in these jurisdictions.
The ESA feature provides the following advantages:
•
•
•
•
•
•
recognizes special Emergency Service Directory Number (ESDN)
overrides calling restrictions
provides optional assignment of ESA CLID per DN
provides optional selection of a special emergency route
provides optional routing digits (for NRS resolution)
provides optional assignment of an Emergency Location Identification
Number (ELIN)
•
provides On Site Notification (OSN) through an external tool, which traps
the emergency call event and records an alarm when an emergency
calls are placed at the branch office.
For more information about ESA, see Emergency Services Access
Fundamentals (NN43001-613).
The main office Call Server forwards emergency services calls to the SRG
using a virtual trunk.
Emergency Services Access
The Emergency Services Access (ESA) configuration specifies the digit
sequence (a DN) that the user dials to start an emergency call, known as
the Emergency Services Directory Number (ESDN). There can only be one
ESA configuration per customer and thus only one ESDN per customer,
which means that all telephones on the same network must be in the same
numbering plan.
With all sites using the same ESDN, a conflict occurs in the NRS because
the same ESDN may need to route to different gateways. The conflict is
resolved by using a routing digit for each site that the main office adds as it
routes the call. The suggested routing digit is the ESN home location code
of the SRG, or alternately, the Numbering Plan Area (NPA) code of the SRG
if there is not more than one Call Server in the NPA. Virtually any unique
digit string (maximum 15 digits) can be used as a routing digit, because the
call is sent to the NRS as a Private/Special Number (SPN). In the NRS,
SPN have their own separate numbering plan.
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Emergency Services Access 191
The Automatic Number Identification (ANI) data sent to the Public Safety
Answering Point (PSAP) identifies the location of the caller. In some
constituencies, legislation requires one ANI (DID) per fixed area, so
the physical location of the emergency can be approximated based on
the telephone number delivered to the PSAP. The ESA feature has a
comprehensive scheme that can be used to convert an extension into an
appropriate DID.
If the branch office is relatively small, it can be easier to use a single ANI
number for the branch office. For more information on this command, see
Software Input Output Reference—Maintenance (NN43001-711).
Routing Emergency Services Access (ESA) calls
ATTENTION
Do not route ESA calls to a node that has no direct ESA trunks.
Ideally, route ESA calls directly over Central Office (CO) trunks to the Public
Safety Answering Point (PSAP). In those cases where this routing is not
possible, do not route ESA calls to nodes that have no direct ESA trunks.
The implications of routing calls to nodes without direct ESA trunks are as
follows:
•
At the node without the direct ESA trunks, the node cannot route the
ESA call directly to the PSAP. Instead, that node must re-route the call
to another node. This re-routing is an unnecessary use of resources.
•
If the node is a CS 1000E node, the only tandem trunks are IP Peer
trunks. There is no way to specify the appropriate rerouting digits (that
is, Prepend Digits) to reroute the ESA call to another node with direct
ESA trunks.
Therefore, if unable to route ESA calls directly to the PSAP, the next best
practice is to route ESA calls to nodes with direct ESA trunks.
Emergency call routing
A Call Server can provide service to IP phones across multiple emergency
jurisdictions. This can also occur with traditional non-IP equipment in the
form of remote peripheral equipment (for example, Carrier Remote, Fiber
Remote).
An emergency call should be handled by the designated means for the
phone location (for example, local security desk or local PSAP). The
emergency call should be routed to a service at the current location of the
phone.
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192 Emergency Services configuration
Configuring ESA for the branch office
For ESA, the main office Call Server forwards the call to the branch office
for termination. Calls are redirected over a Virtual Trunk using the NRS. The
NRS routes the calls using a special number, referred to in this section
as the ESA Special Number.
ESA must be configured and tested on the main office Call Server and the
SRG to differentiate between emergency calls originating from IP Phones at
each location and calls originating on trunks, which refers to the forwarded
emergency call that the SRG receives from the main office for an IP Phone
in Normal Mode.
Use the following steps to configure ESA for emergency access at each
location:
At the main office:
Step Action
1
2
Determine the dialing plan (for example, numbering plan) for ESA
calls.
Configure the main office emergency trunk (CAMA or PRI).
For EMEA, the following trunks are supported:
•
•
•
•
•
BRIE (Basic Rate Interface–ETSI based)
PRI (Primary Rate Interface per EURO ISDN)
QSIG on PRI
DPNSS
IP tandem trunks on ISDN
3
4
5
6
Configure the Virtual Trunk at the main office.
Configure ESN at the main office.
Configure ESA at the main office.
Configure the SRG zone on the main office.
Configure a zone for each branch office which is used in conjunction
with ESA parameters to route an emergency call to the SRG.
7
Test ESDN using a main office telephone.
Configure the SRG emergency trunk (CAMA or PRI).
Configure the Virtual Trunk at the MG 1000B.
Configure ESN at the branch office.
8
9
10
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Emergency Services Access 193
11
12
Configure ESA at the branch office.
Configure the branch office zone on the branch office.
The branch office zone is required for bandwidth management but
does not require ESA parameters.
13
14
15
Configure the ESN SPN on the branch office.
Configure the NRS for the ESA Special Number used.
Test ESDN using an analog (500/2500-type) telephone located at
the branch office.
16
Test ESDN using an SRG IP Phone in Normal Mode and in Local
Mode.
—End—
At the SRG, or branch office:
Step Action
1
2
3
4
5
Configure the SRG emergency trunk (CAMA or PRI).
Configure the Virtual Trunk at the MG 1000B.
Configure ESN at the branch office.
Configure ESA at the branch office.
Configure the branch office zone on the branch office.
The branch office zone is required for bandwidth management but
does not require ESA parameters.
6
7
8
Configure the ESN SPN on the branch office.
Configure the NRS for the ESA Special Number used.
Test ESDN using an analog (500/2500-type) telephone located at
the branch office.
9
Test ESDN using an SRG IP Phone in Normal Mode and in Local
Mode.
—End—
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194 Emergency Services configuration
Reregistering to minimally configured branch office
A branch user in Local Mode but who is not physically at the branch can get
incorrect emergency service handling.
If the SRG is not provisioned with knowledge of all the ERL in the enterprise,
one of two scenarios occurs when an IP Phone reregisters to the branch
(either by VO ESA redirection or by fallback to Local Mode):
•
If the local TN is provisioned as Manual Update, then the phone inherits
the static location data. The static location data probably indicates basic
ESA processing (per LD 24) if this is a small branch.
•
If the local TN is provisioned as Auto Update, then cached location data
in the phone is rejected if undefined locally, and unknown location values
(ERL = 0, ECL = 0, LocDesc = Unknown) are assigned. Unknown
location indicates default (basic) emergency processing (per LD 24),
which is acceptable for a small branch. A system message is also
generated to indicate that the phone location data was actually unknown
and defaults were used, but emergency calls should be handled
correctly.
Minimally configured branches (without LIS support) can be configured as
manual update.
Routing configuration for ESA calls on SRG 50
Use the following steps to configure routing for ESA calls for the SRG 50:
Step Action
1
2
3
4
Build a destination code corresponding to the ESA SPN for the
branch office.
Configure the destination code to absorb the leading digits for the
SPN, leaving just the ESDN.
Configure the destination code to use a public route to the PSTN
trunks.
Ensure the Remote access package (00 to 15 under Call Security)
assigned to the VoIP trunks has the appropriate Line Pool
Access/Bloc for PRI.
5
Ensure there is a Public Prefix of 911 with a length of 3 to match to
outgoing digits. This eliminates any delay. As soon as the 3 digits
are collected, the call is sent.
—End—
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Emergency Services Access 195
Determining the dialing plan for ESA calls
In many jurisdictions of the United States and Canada, the emergency
number must be “911”. The call processor cannot have a DN that conflicts
with these digits, but since “9” is often used for NARS AC2 (the local call
Access Code), this is not usually a problem.
ESA for international deployment must support the standard emergency
number 112 and any emergency numbers in use prior to the EU directive.
In general, ESA calls should leave the network through a trunk at the
branch office where the originating telephone is located. To enable this,
it is necessary for telephones at each branch office to supply a unique
identifying prefix to the NRS when the ESA calls are being routed so that
the NRS can select a distinct route for each branch office. This prefix can
be configured with the zone data for the SRG telephones. The provisioning
of this prefix is an enhancement for branch office.
While a variety of numbering schemes are available, Nortel recommends
that customers use 0 + the ESN location code of the SRG + ESDN, where
ESDN is:
•
•
for North America and CALA—911
for members of the European Union—112 and any other emergency
numbers in use prior to the EU directive
This number, referred to here as the ESA Special Number, is configured as
a special number (SPN) in the NRS so that the Virtual Trunk routes the call
to the branch office.
Use Element Manager or the Command Line Interface for the following
procedure. See IP Peer Networking Installation and Commissioning
(NN43001-313) for details.
Procedure 15
Configuring the main office
Step Action
1
2
Configure the main office emergency trunk (CAMA or PRI).
Configure either analog CAMA or digital PRI to correctly signal the
call identification.
ESA overrides all restrictions. Configure the trunk with restrictions
so that other features cannot access the trunk.
Configure the Virtual Trunk using the procedure from IP Peer
Networking Installation and Commissioning (NN43001-313).
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196 Emergency Services configuration
The Virtual Trunk must be configured to enable emergency calls
originating from SRG IP Phones registered at the main office to
reach the branch office.
3
4
Configure ESN.
ESA uses a route number rather than ESN route list index. However,
ESN is required at the branch office.
Configure Emergency Services Access (ESA) in LD 24.
LD 24 Configure Emergency Services Access
Prompt
REQ:
Response
NEW CHG
ESA
Description
Add new data, or change existing data.
Emergency Services Access data block
Customer number as defined in LD 15
TYPE:
CUST
xx
ESDN
xxxx
Emergency Services DN (for example, 911). Up to four digits
are accepted.
ESRT
ESA route number
0-511
0-127
Range for Large Systems
Range for MG 1000B
DDGT
DFCL
x…x
Directing Digits (for CAMA Trunks)(for example, 1, 11, or 911).
Up to four digits are accepted.
x…x
Default ESA Calling Number. The input must be the following
lengths:
•
•
On a system that is not FNP equipped, 8 or 11 digits are
accepted if the first digit of the input is ’1’; otherwise the input
must be 7 or 10 digits.
On a system that is FNP equipped, up to 16 digits are
allowed.
OSDN
x…x
On-Site Notification station DN. The input must be a valid single
appearance internal DN.
You configure OSDN to alert the local security personnel about an
emergency call in progress. Leave the ESA route number blank
to make test calls without using any trunk resources. If the route
number has been configured, remove it by entering “x” at the prompt.
Nortel recommends that the system administrator arrange a test call
with the Public Services Access Point (PSAP).
5
Test ESDN using a main office telephone to confirm that main office
calls exit the main office trunks. The ESA Configuration Audit feature
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Emergency Services Access 197
provides CLID Verification (CLIDVER) reports that determine how an
emergency call is routed, without actually routing the call. Use LD
20 to generate a CLIDVER report.
Table 20
LD 20 Generate a CLIDVER report
Prompt
REQ:
Response
Description
Print
PRT
TYPE
CLIDVER
(DN) TN
CLID Verification
SORTBY
The output/report is sorted
based on this flag.
If the response is DN, the
overlay prompts the user to
enter the DN, and the output
is sorted by the DN. If the
response is TN, the overlay
prompts the user to enter the
TN and the output is sorted by
the TN.
ESA_ONLY
(YES) NO
Flag used to decide if the report
should contain information for
ESA call type only or for all call
types.
If the ESA package is restricted,
this input prompt does not
appear. The report contains
non-ESA data only.
SHORT
TN
(YES) NO
lscu
Flag used to decide if the output
report should be a Short report
of a Detail report.
Terminal Number
Format for Large System,
where l = loop, s = shelf, c =
card, and u = unit
CUST
DN
xx
Customer number as defined in
LD 15.
x…x
Directory Number. If no value is
entered, the report includes all
supported Directory Numbers.
DATE
PAGE
dd mmm yyy
(NO) YES
Date
Data printed on a per page
basis.
DES
d…d
Designator
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198 Emergency Services configuration
For more information about CLIDVER reports, see Emergency
Services Access Fundamentals (NN43001-613).
6
Configure the branch office zone on the main office.
a. Configure the branch office zone ESA dialing information in LD
117.
LD 117 Configure branch office zone ESA route
Description
CHG ZESA <Zone><ESA Route #><AC><ESA Prefix><ESA Locator>
Defines the ESA parameters for the branch office zone, where:
Command
•
Zone = Zone number for the branch office.
ESA Route # = Virtual Trunk route to SRG.
•
•
AC = Access Code to add to dialed digits. If no AC is required, enter
AC0 in place of AC1 or AC2.
•
ESA Prefix = Digit string added to start of ESDN. This is a unique
prefix in the NRS. Nortel recommends that users use 0 + ESN location
code of the branch office node. An example for location code 725
would be: 0725.
•
ESA Locator = Direct Inward Dial telephone number sent as part of
ANI for use by the PSAP to locate the source of the call.
b. Enable the branch office zone ESA in LD 117.
ENL ZBR <Zone> ESA
7
Configure the ESA Special Number at the main office.
Configure the ESA Special Number in the NRS. Using NRS,
configure the ESA Special Number defined for the branch office
zone. See IP Peer Networking Installation and Commissioning
(NN43001-313).
Nortel recommends that customers use “0” + the ESN Location code
+ ESDN. An example for location code 725 would be 0725911. The
zero is recommended to prevent a collision in the ESN data with
the HLOC entry.
8
Do the following:
a. In LD 86, configure Emergency Service Access Digit
Manipulation for AC + ESDN dialing to allow recognition of the
ESDN even if AC1 or AC2 is used.
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Emergency Services Access 199
LD 86
Prompt
REQ
Response
NEW CHG
xx
Description
Add, Change
CUST
FEAT
…
Customer number as defined in LD 15
DGT
Digit Manipulation
…
…
DMI
1- 999
Digit Manipulation Table numbers.
ATTENTION
Do not use Digit Manipulation Table 0, as it results in the
incorrect call termination treatment.
b. Configure the system to trap the ESDN within the AC1 and AC2
translation tables to reprocess the ESDN locally.
LD 86 Configure the system to trap the ESDN and reprocess it locally
Prompt
REQ
Response
NEW CHG
xx
Description
Add, Change
CUST
FEAT
…
Customer number as defined in LD 15
Route List Block
RLB
…
…
RLI
xxx
Route List Index to be accessed
Entry number for NARS/BARS Route List
ENTR
LTER
xxx
YES
Local Termination entry. This allows the AC + ESDN call to be
recognized as an Emergency Services Access call.
DMI
1 - 999
Digit Manipulation Table.
Use the table configured in LD 86. This allows the digits after
the AC to remain in the call register as a called number.
ATTENTION
Do not use Digit Manipulation Table 0, as it results in the
incorrect call termination treatment.
c. Configure Emergency Service Access call recognition for AC +
ESDN dialing in LD 90.
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200 Emergency Services configuration
LD 90
Prompt
REQ
Response
NEW CHG
xx
Description
Add, Change
CUST
FEAT
TRAN
TYPE
- SPN
Customer number as defined in LD 15
Network translation tables
NET
aaa
SPN
911
AC + ESDN is recognized as an Emergency Service Access call.
Use the number configured for ESDN.
Route List Index.
112
- RLI
xxx
Use Route List Index configured in LD 86.
—End—
Procedure 16
Configuring the branch office zone
Step Action
1
Configure the branch office zone on the branch office.
In the branch office, only the zone number and bandwidth/codec
selection is configured.
Use the same zone number between the branch office and main
office. The main office configuration (Procedure 15 "Configuring the
main office" (page 195), step 6) provides the branch office zone
characteristics (local time, local dialing, and ESA).
2
Configure the routing tables on the SRG.
The SRG must recognize the incoming digits on the Virtual Trunk and
remove all but the ESDN. The call is routed to a local termination.
—End—
Emergency Services for Virtual Office
The E911 Virtual Office feature allows Virtual Office users, whether they are
logged in or logged out of Virtual Office to place an emergency (E911) call
to the correct Public Safety Answering Point (PSAP) for their geographical
location.
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Configuring the NRS for ESA SPN 201
The use of the terms Normal Mode and Local Mode apply to SRG branch
user only.
Emergency Services while logged in to Virtual Office
The E911 Virtual Office feature recognizes when a user dials an ESDN and
it forces the Virtual Office IP Phone to log out of Normal Mode (into Local
Mode) in order to place the emergency call directly from the branch office
location to the PSAP.
Emergency Services while logged out of Virtual Office
If 911 is dialed while logged out of Virtual Office the LTPS redirects the 911
call to the local 911 service (PSAP), not the remote Call Server 911 service.
The Call Server is provisioned with Emergency Services Access Terminal
Numbers (ESTN). The ESTN is used to register the IP Phone with the Call
Server. The logged out IP Phone can make ESA calls only.
For more information on emergency services for Virtual Office, see
Emergency Services Access Fundamentals (NN43001-613) and Branch
Office Installation and Commissioning (NN43001-314).
On-Site Notification
The ESA On-Site Notification (OSN) function notifies local security
personnel when an emergency call occurs. When an emergency call is
placed at the branch office, an external tool traps the notification and
records an alarm. This applies to IP Phones that the main office returns in
local mode when an emergency call is made, as well as locally connected
analog (500/2500-type) telephones.
A LAN port must be enabled on the SRG to support the external tool. For
more information, see SRG50 Configuration Guide (NN40140-500).
Configuring the NRS for ESA SPN
The NRS must be configured for the ESA Special Number (SPN). The
NRS uses the ESA SPN to route the emergency call from the main office
to the branch office.
Nortel recommends that a consistent pattern be followed for all ESA calls.
For example, use 0 + ESN Location code of the branch office node + the
ESDN. An example for location code 725 would be: 0725911. The zero is
recommended to prevent a collision in the ESN data with the HLOC entry.
For more information, see IP Peer Networking Installation and
Commissioning (NN43001-313).
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202 Emergency Services configuration
Testing the ESDN number
test the ESDN number from any telephone in the branch office.
Procedure 17
Testing ESDN using an SRG telephone
Step Action
1
For IP Phones:
a. Dial the ESDN on an SRG IP Phone in Local Mode.
The calls must go out on the emergency trunk(s) in the branch
office.
b. Dial the ESDN on an SRG IP Phone in Normal Mode.
The calls must tandem over the Virtual Trunk to the branch office
and go out on the emergency trunk(s) in the branch office. The
following configuration problems can occur:
•
The call can receive overflow tones. Use LD 96 to view the
digits sent to the Virtual Trunk (ENL MSGO dch#).
•
If the digits look correct on the main office, the NRS might
not be properly configured. If the NRS rejects the call, a
diagnostic message is displayed on the NRS console.
•
If the call makes it to the correct branch office (check that
it is not going to the wrong node if the NRS is configured
incorrectly), the branch office is probably rejecting it because
it does not know the digit string.
2
For analog (500/2500-type) telephones, dial the ESDN on an SRG
analog (500/2500-type) telephone.
The calls must go out on the emergency trunk(s) in the branch office.
—End—
Configuring ESA using Element Manager
To configure Emergency Services Access in Element Manager, see Element
Manager System Reference Administration (NN43001-632).
Emergency Service using Special Numbers (SPN)
Determining the dialing plan for emergency access calls is critical.
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Emergency Service using Special Numbers (SPN) 203
In many jurisdictions, the emergency number is a fixed number (for example,
112 or 999). The main office Call Server or SRG cannot have a DN that
conflicts with these digits.
Access to Emergency Service using SPN should be configured in the
following circumstances:
•
When the Emergency Service number at the branch office is different
from that at the main office.
•
When there is more than one number used for accessing Emergency
Service; for example, when there are different numbers for Police, Fire,
and Ambulance services.
•
In markets where the ESA feature is not available (outside of North
America, CALA, and EMEA).
To configure Emergency Service using SPN, follow the process outlined in
"Dialing Plan configuration" (page 43). If SRG PSTN access is correctly
configured, Emergency Service from the branch office will already be
present.
Branch office access to Emergency Service using SPN must be configured
and tested the main office Call Server and the SRG to differentiate between
emergency calls originating from IP Phones at each location and emergency
calls originating on trunks.
The special handling provided by ESA is not available in this scenario, such
as OSN and zone-based routing.
For information on emergency services for Virtual Office, see Emergency
Services Access Fundamentals (NN43001-613).
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Enhanced UNIStim Firmware Download
Contents
This section contains the following topics:
•
•
Description
This section applies to the main office and the following IP Phones:
•
•
•
•
•
•
•
•
•
IP Phone 2001
IP Phone 2002
IP Phone 2004
IP Phone 2007
IP Audio Conference Phone 2033
IP Phone 1110
IP Phone 1120E
IP Phone 1140E
IP Phone 1150E
It does not apply to the IP Softphone 2050 and WLAN
2210/2211/2212/6120/6140.
The redirected IP Phones at the SRG 50 are under the control of the main
office Call Server for the majority of the deployment (Normal Mode). Users
of the SRG IP Phones receive the features, key layout, and tones of the main
office Call Server. Therefore, the version of the IP Phone firmware must
align with the requirements of the CS 1000. When an IP Phone requires
firmware upgrade, the CS 1000 uses the umsUpgradeAll command, or
variant, to redirect the IP Phone back to the SRG 50 for upgrading.
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For CS 1000 Release 4.5 and later, if the required firmware file does not
exist on the SRG 50, or the version of the file is incorrect, the SRG 50
initiates an FTP session to the TPS for the IP Phone to retrieve the required
file. The SRG 50 upgrades the IP Phone and redirects the IP Phone back
to the CS 1000.
For SRG 50 Release 2.0 and later, if the required firmware does not exist
on the SRG 50, or the version is incorrect, the SRG 50 initiates an FTP
session to the TPS for the IP Phone to retrieve the required file. The SRG
50 upgrades the IP Phone and redirects the IP Phone back to the CS 1000.
For SRG 50 Release 1.0, the SRG 50 must be patched to the proper
firmware level with patch number BCM50.90. For CS 1000 Release
4.0, ensure MPLR21148 is installed on the Signaling Server. Firmware
download does not occur when IP Phones register to the TPS by a Virtual
Office Login or branch office redirection to the main office. Instead, SRG
IP Phones are redirected back to the SRG TPS for firmware files upgrade.
This redirection occurs only if the umsUpgradeAll command is issued from
the main office TPS, and the current firmware files are missing.
For CS 1000 Release 4.0, ensure MPLR21148 is installed on the Signaling
Server.
Firmware download does not occur when IP Phones register to the TPS by
a Virtual Office Login or branch office redirection to the main office. Instead,
SRG IP Phones are redirected back to the SRG TPS for firmware files
upgrade. This redirection occurs only if the umsUpgradeAll command is
issued from the main office TPS, and the current firmware files are missing.
If an IP Phone is in use when the umsUpgradeAll command is issued, the
call is not interrupted. Its firmware version is checked against the main
office TPS firmware policy, and if there is no match, the IP Phone is flagged,
then redirected to the MG 1000B TPS when the call is completed. The
umsUpgradeAll command has no immediate impact on IP Phones that
are logged in or out by Virtual Office. However, the firmware files may be
upgraded, if required, when the Virtual Office session is terminated.
For information on Enhanced UNIStim Firmware, see IP Line Fundamentals
(NN43100-500).
Firmware upgrade
firmware. For information about upgrading IP Phone firmware, see IP Line
Fundamentals (NN43100-500).
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Firmware upgrade 207
Procedure 18
Upgrading firmware
Step Action
1
2
At the Main office, upgrade IP Phone firmware on the Signaling
Server. For instructions, see Signaling Server Installation and
Commissioning (NN43001-312).
Issue the CLI command umsUpgradeAll at the main office. IP
Phones at the Main office and branch office are upgraded as
necessary.
—End—
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209
Appendix A
Media Redirection Scenarios
In addition to basic call scenarios, Network Bandwidth Management also
supports the following media redirection scenarios:
•
Scenario 1: Codec switches correctly during media redirection. See
•
Scenario 2: Call transfer works correctly with IP Phones:
— Scenario 2.1: Call Transfer from an SRG IP Phone in Normal Mode
— Scenario 2.2: Call Transfer from main office IP Phone to an SRG IP
•
Scenario 3: Conference Call works correctly with a branch office:
— Scenario 3.1: Conference call between branch office and main
— Scenario 3.2: Conference call between main office and branch
The zone table is examined using the commands in LD 117. See Software
Input Output Reference—Maintenance (NN43001-711) for more information
on these commands.
In these scenarios, consult the zone table at the main office for accurate
bandwidth usage information.
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210 Appendix A Media Redirection Scenarios
Table 21
Codec switches correctly during media redirection
Event
1
Result
An incoming Direct Inward Dial (DID) call
to branch office uses IP Peer to reach the
symposium controller Control Directory
Number (CDN) in the main office.
The external caller hears music and
announcements with a G.729 codec. Bandwidth
usage in the main office indicates the call is an
interzone call.
The external caller is connected to an Automatic
Call Distribution (ACD) agent with a G.711 codec.
Bandwidth usage in the main office indicates the
call is an intrazone call. The ACD agent is an
SRG IP Phone registered to the main office.
2
The call is released.
The zone table indicates the bandwidth usage for
the call is removed correctly on the main office
Call Server and in the branch office.
Table 22
Call transfer from SRG IP Phone in Normal Mode to main office IP Phone
Event
Result
1
An SRG TDM telephone calls an IP Phone
registered to the main office.
A speech path is established between the SRG
TDM telephone and the IP Phone registered to the
main office. The zone table indicates intrazone
bandwidth usage.
2
The SRG IP Phone registered to the main
The SRG TDM telephone is put on hold. A speech
office initiates a call transfer to a main office path is established between the SRG IP Phone
IP Phone.
registered to the main office and the main office
IP Phone. The zone table indicates interzone
bandwidth usage.
3
4
The Call Transfer key on the SRG IP Phone A speech path is established between the SRG
registered to the main office is pressed to
complete the call transfer.
TDM telephone and the main office IP Phone. The
zone table indicates interzone bandwidth usage.
The call is released.
The zone table indicates bandwidth usage for the
call is unreserved correctly.
Table 23
Call transfer from main office IP Phone to SRG IP Phone in Normal Mode
Event
Result
1
An SRG TDM telephone calls a main office
IP Phone.
A speech path is established between the SRG
TDM telephone and the main office IP Phone. The
zone table indicates interzone bandwidth usage.
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Appendix A Media Redirection Scenarios 211
Event
Result
2
The main office IP Phone initiates a call
transfer to an SRG IP Phone registered to
the main office.
The SRG TDM telephone is put on hold. A speech
path is established between the main office IP
Phone and the SRG IP Phone registered to the
main office. The zone table indicates interzone
bandwidth usage.
3
4
A speech path is established between the IP
Phone registered to the main office and the
SRG TDM telephone. The zone table indicates
intrazone bandwidth usage.
The Call Transfer key on the main office
IP Phone is pressed to complete the call
transfer.
The call is released.
The zone table indicates bandwidth usage for the
call is unreserved correctly.
Table 24
Conference call between branch office and main office, initiated by SRG IP Phone in Normal
Mode
Event
Result
1
An SRG TDM telephone calls an SRG IP
Phone registered to the main office.
A speech path is established between the SRG
TDM telephone and the SRG IP Phone registered
to the main office. The zone table indicates
intrazone bandwidth usage.
2
The SRG IP Phone registered to the main
office initiates a conference call to a main
office IP Phone.
The SRG TDM telephone is put on hold. A
speech path is established between the SRG
IP Phone registered to the main office and the
main office IP Phone. The zone table indicates
interzone bandwidth usage.
3
The Conference key on the SRG IP Phone
registered to the main office is pressed to
complete the conference call.
Speech paths are established among the SRG
TDM telephone, the SRG IP Phone registered
to the main office, and the main office IP Phone.
The zone table indicates interzone and intrazone
bandwidth usage.
4
5
The SRG TDM telephone releases the call.
The call is released.
A speech path is established between the main
office IP Phone and the SRG IP Phone registered
to the main office. The zone table indicates
interzone bandwidth usage.
The zone table indicates bandwidth usage for the
call is unreserved correctly.
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212 Appendix A Media Redirection Scenarios
Table 25
Conference call between main office and branch office, initiated by main office IP Phone
Event
Result
1
2
An SRG TDM telephone calls a main office
IP Phone.
A speech path is established between the SRG
TDM telephone and the main office IP Phone.
The zone table indicates interzone bandwidth
usage.
The main office IP Phone initiates a
conference call to an SRG IP Phone
registered to the main office.
The SRG TDM telephone is put on hold. A
speech path is established between the main
office IP Phone and the SRG IP Phone registered
to the main office. The zone table indicates
interzone bandwidth usage.
3
Speech paths are established among the SRG
TDM telephone, the SRG IP Phone registered
to the main office, and the main office IP Phone.
The zone table indicates interzone and intrazone
bandwidth usage.
The Conference key on the main office
IP Phone is pressed to complete the
conference call.
4
5
The SRG TDM telephone releases the call.
The call is released.
A speech path is established between the SRG
IP Phone registered to the main office and the
main office IP Phone. The zone table indicates
interzone bandwidth usage.
The zone table indicates bandwidth usage for the
call is unreserved correctly.
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213
List of terms
Branch office
An SRG that is remote from the main office. The SRG provides telephony
services using the main office servers (for Normal Mode) or local system
services when the SRG loses IP communication with the main office (Local
Mode).
CDP
Coordinated Dialing Plan. Under the recommended Coordinated Dialing
Plan, the Branch User ID can be an extension (for example, 4567). For
more information about CDP, see Dialing Plans Reference (NN43001-283).
dialing plan
Each system uses a specific numbering configuration (dialing plan) that
determines how calls will be handled over a private or public network.
DSP
Digital Signal Processing, which refers to manipulating analog information,
such as sound or photographs that have been converted into a digital form.
DSP also implies the use of a data compression technique.
When used as a noun, DSP stands for Digital Signaling Processor, a special
type of coprocessor designed for performing the mathematics involved in
DSP. Most DSP are programmable, which means that they can be used for
manipulating different types of information, including sound, images, and
video.
ESA
Emergency Services Access is a feature that places a customer in
compliance with federal legislation that requires the Private 911 type of
functionality provided by ESA. Please note, however, that the ESA feature
is also generally useful for users who are not subject to legislation, and
is broad enough to be used in different countries. For example, it will be
appreciated by any customer who wants to route emergency calls in a
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214 List of terms
special manner, or who wants to be notified when a telephone user makes
an emergency call. It would also appeal to a customer who wishes to have
ESA calls answered on-site,
on the business premises, rather than being forwarded to the Public
Services Answering Point (PSAP). See Emergency Services Access
Fundamentals (NN43001-613) for complete information.
Gatekeeper
The Gatekeeper is a separate application on an IP network that directs
IP traffic to all the systems on the network. Parameters for both the main
office and SRG must be assigned to all gatekeepers active on the network.
If the Gatekeeper is down, the SRG attempts to connect to the Alternate
Gatekeeper, if there is one. If the Alternate Gatekeeper is down as well, or
there is no Alternate Gatekeeper, the SRG IP Phones remain registered
with the main office but calls cannot be sent to the SRG.
gateway
In networking, a combination of hardware and software that links two
different types of networks. Gateways between e-mail systems, for example,
enable users on different e-mail systems to exchange messages.
H.323
A standard approved by the International Telecommunication Union (ITU)
that defines how audiovisual conferences data is transmitted across
networks. In theory, H.323 enables users to participate in the same
conference even though they are using different video conferencing
applications. Although most video conferencing vendors have announced
that their products conform to H.323, it is too early to say whether such
adherence actually results in interoperability.
IP
Abbreviation of Internet Protocol, pronounced as two separate letters. IP
specifies the format of packets, also called datagrams, and the addressing
scheme. Most networks combine IP with a higher-level protocol called
Transport Control Protocol (TCP), which establishes a virtual connection
between a destination and a source.
IP by itself is something like the postal system. It enables you to address a
package and drop it in the system, but there’s no direct link between you
and the recipient. TCP/IP, on the other hand, establishes a connection
between two hosts so that they can send messages back and forth for
a period of time.
LAN
Local Area Network.
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List of terms 215
Local Mode
The SRG is in Local Mode when:
•
•
•
The IP Phones are first installed and not yet reregistered with the main
office
The SRG cannot communicate with the main office and the IP Phones
are reregistered with the SRG
A user deliberately puts the IP Phone in the Test Local Mode condition.
Main office
The CS 1000 system that has been programmed to accept redirection of the
SRG IP Phones and provide call service for the SRG in Normal Mode.
NCS
Network Connection Service. It provides a TPS interface to the NRS,
allowing the TPS to query the NRS using the UNIStim protocol. It is required
to support the main office, branch office, Virtual Office, and Geographic
Redundancy features.
Normal Mode
The SRG is in Normal Mode when the IP Phones on the SRG are correctly
redirected to the main office Call Server.
NRS
Network Routing Service. The software application where all systems in the
network are registered. The NRS consists of the H.323 Gatekeeper and the
Network Connection Service (NCS).
PSTN
Public Switched Telephone Network. The international telephone system
based on copper wires carrying analog voice data. This is in contrast to
newer telephone networks based on digital technologies.
Telephone service carried by the PSTN is often called plain old telephone
service (POTS).
QoS
Quality of Service, a networking term that specifies a guaranteed throughput
level. One of the biggest advantages of ATM over competing technologies,
such as Frame Relay and Fast Ethernet, is that ATM supports QoS levels.
This enables ATM providers to guarantee to their customers that end-to-end
latency does not exceed a specified level.
There are several methods to provide QoS, as follows:
•
high bandwidth
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216 List of terms
•
•
•
•
•
packet classification
DiffServ
IP fragmentation
traffic shaping
use of the platform’s queuing mechanisms
routing
The process of selecting the correct path for packets transmitted between IP
networks by using software-based algorithms. Each packet is processed by
the algorithm to determine its destination.
SRG 50
Survivable Remote Gateway 50. This describes the equipment used to
create an IP branch office with a CS 1000 system acting as the main office.
The base system for SRG is a Business Communication Manager running
BCM 3.6 software.
TPS
UDP
IP Phone Terminal Proxy Server. This server controls the connection of
IP Phones. It resides on the Signaling Server with an emergency backup
on the Voice Gateway Media Card.
Uniform Dialing Plan. Each location within the network is assigned a
Location Code, and each telephone has a Directory Number that is unique
within the network. Under the UDP, the SRG must include the location code
in the Branch User ID (BUID).
VoIP
WAN
Voice over IP trunk. This IP pathway between two system IP voice gateways
allows the system to exchange telephone calls over the Internet.
Wide Area Network. A computer network that spans a relatively large
geographical area. Typically, a WAN consists of two or more local area
networks (LAN).
Computers connected to a wide area network are often connected through
public networks, such as the telephone system. They can also be connected
through leased lines or satellites. The largest WAN in existence is the
Internet.
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List of terms 217
ZDP
Zone Digit Prefix. This is the number that the main office appends to a local
SRG PSTN call dialed from an SRG IP Phone in Normal Mode. This number
differentiates the call from a main office PSTN call dialed by the main office
telephones. The ZDP routes the call through VoIP trunk to the SRG.
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218
Index
B
G
Gatekeeper 214
Gateway 214
C
Capacity 26
CDP 213
H
H.323 214
Configuring ESA using Element
Manager 202
Configuring the dialing plan using Element
Manager 63
Configuring zone parameters using Element
Manager 36
I
IP 214
L
M
D
Main office information required by the
SRG 32, 32
Dialing plan configuration for SRG PSTN
access 48
Dialing plan configuration in Normal
Mode 48
DSP 213
E
N
Emergency Service using Special
Numbers 202
NCS 215
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Index 219
NRS 215
SRG PSTN to an SRG telephone (DID
call) 45
O
Optional features to enhance SRG
functionality 18
T
P
PSTN 215
TPS 216
Q
QoS 215
U
UDP 216
R
Routing 216
V
VLAN 216
S
VoIP 216
SIP Redirect Server Network Routing
Service 15
Z
ZDP 217
SRG 216
SRG information required by the main
office 31, 31
Zones 24, 24
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220 Index
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Nortel Communication Server 1000
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Copyright © 2005-2007, Nortel Networks
All Rights Reserved.
Publication: NN43001-307
Document status: Standard
Document version: 02.02
Document date: 3 December 2007
Sourced in Canada.
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.
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