Nortel Networks Network Router SRG50 User Manual

SRG50 Configuration Guide  
BCM50 3.0  
Survivable Remote Gateway  
Document Status: Standard  
Document Number: NN40140-500  
Document Version: 02.03  
Date: June 2008  
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Task List  
SRG50 Configuration Guide  
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NN40140-500  
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Contents  
SRG50 Configuration Guide  
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NN40140-500  
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SRG50 Configuration Guide  
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NN40140-500  
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9
Chapter 1  
Getting started  
About this guide  
The SRG50 Configuration Guide describes how to install, configure, and maintain the Survivable  
Remote Gateway (SRG) 50 Release 3.0.  
The SRG50 is positioned as a cost-effective Small IP Branch Office solution for CS 1000 Main  
office systems. The SRG50 offers business continuity and public switched telephone network  
(PSTN) failover for voice over IP (VoIP) networks. An SRG provides transparent operation,  
feature and application parity with a main office call server while in normal operating mode. If  
connectivity with the call server or wide area network (WAN) is lost, the normal mode sets revert  
back to local mode. The SRG takes ownership of call control for the local sets automatically and  
provides internal communications as well as external connectivity to the PSTN.  
SRG50 supports H323 and SIP Trunking and up to 80 survivable IP users with a single SRG  
application authorization code. It is provided as a cost-effective VoIP business continuity solution  
for small branch offices. The SRG50 supports CS 1000 Release 4.0, 4.5, 5.0, and 5.5.  
Audience  
The SRG50 Configuration Guide is intended for two audiences:  
the individuals responsible for engineering the SRG50 site and installing the BCM50,  
configuring it for operation as an SRG50, and connecting it to the network  
the individuals responsible for post-installation system administration and  
maintenance.  
The SRG50 site engineer and installer must be familiar with BCM50 hardware and software, and  
IP telephony and VoIP trunk configuration on the BCM50.  
Acronyms  
The following is a list of acronyms used in this guide.  
Table 1 Acronyms used in this guide (Sheet 1 of 2)  
Acronym  
Description  
ACR  
Alternative call routing  
ANBWM  
ASM  
Adaptive network bandwidth management  
Analog station module  
ATA  
Analog terminal adapter  
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10 Chapter 1 Getting started  
Table 1 Acronyms used in this guide (Sheet 2 of 2)  
Acronym  
Description  
BARS  
BUID  
CDP  
DDIM  
DN  
Basic alternate route selection  
Branch user ID  
Coordinated dialing plan  
Digital drop and insert mux  
Directory number  
DSC  
DSM  
DTM  
ESA  
Distant steering codes  
Digital station module  
Digital trunk module  
Emergency services access  
Emergency services DN  
Facility restriction level  
Global analog trunk module  
Key expansion module  
Keycode retrieval system  
Local area network  
ESDN  
FRL  
GATM  
KEM  
KRS  
LAN  
LSC  
Local steering codes  
MCDN  
MOTN  
MVC  
NARS  
NBWM  
NCS  
NRS  
PSTN  
QoS  
Meridian customer defined network  
Main office terminal number  
Mobile voice client  
Network alternate route selection  
Network bandwidth management  
Network connection server  
Network routing service  
Public switched telephone network  
Quality of service  
SPN  
SRG  
TAT  
Special number  
Survivable remote gateway  
Trunk anti-tromboning  
Trunk route optimization  
Trunk steering codes  
TRO  
TSC  
UDP  
VNR  
VoIP  
VPN  
VPNI  
WAN  
ZDP  
Uniform dialing plan  
Vacant number routing  
Voice over internet protocol  
Virtual private network  
Virtual private network ID  
Wide area network  
Zone digit prefix  
NN40140-500  
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Chapter 1 Getting started 11  
Symbols and conventions used in this guide  
These symbols highlight critical information for the SRG system.  
Caution: Alerts you to conditions where you can damage the equipment.  
Danger: Alerts you to conditions where you can get an electrical shock.  
Warning: Alerts you to conditions where you can cause the system to fail or  
work improperly.  
Note: Alerts you to important information.  
Tip: Alerts you to additional information that can help you perform a task.  
Security Note: Indicates a point of system security where you can change a  
default or where the administrator must decide on the level of security required  
!
for the system.  
Warning: Alerts you to ground yourself with an antistatic grounding strap  
before performing the maintenance procedure.  
Warning: Alerts you to remove the main unit and expansion unit power cords  
from the AC outlet before performing any maintenance procedure.  
These conventions and symbols represent the Business Series Terminal display and dialpad.  
Convention  
Example  
Used for  
Word in a special font (shown in  
the top line of the display)  
Command line prompts on display telephones.  
Pswd:  
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12 Chapter 1 Getting started  
Convention  
Example  
Used for  
Underlined word in capital letters  
(shown in the bottom line of a  
two-line display telephone)  
Display options on two-line display telephones.  
Press the button directly below the option on the  
display to proceed.  
PLAY  
Dialpad buttons  
Buttons you press on the dialpad to select a  
particular option.  
£
These text conventions are used in this guide to indicate the information described.  
Convention  
Description  
bold Courier  
text  
Indicates command names, options, and text that you must enter.  
Example: Use the infocommand.  
Example: Enter show ip{alerts|routes}.  
italic text  
Indicates book titles.  
plain Courier  
text  
Indicates command syntax and system output (for example, prompts  
and system messages).  
Example: Set Trap Monitor Filters  
FEATURE  
HOLD  
RELEASE  
Indicates that you press the button with the corresponding icon on the  
set you are using.  
Related publications  
This section provides a list of additional documents referred to in this guide.  
Administration Guide (NN40020-600)  
Device Configuration Guide (NN40020-300)  
Installation Checklist and Quick Start Guide (NN40020-308)  
Installation and Maintenance Guide (NN40020-302)  
Main Office Configuration Guide for SRG50 (NN43001-307)  
Networking Configuration Guide (NN40020-603)  
Telephony Device Installation Guide (NN40020-309)  
NN40140-500  
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Chapter 1 Getting started 13  
How to get help  
This section explains how to get help for Nortel products and services.  
Getting Help from the Nortel Web site  
The best way to get technical support for Nortel products is from the Nortel Technical Support  
Web site:  
This site provides quick access to software, documentation, bulletins, and tools to address issues  
with Nortel products. More specifically, the site enables you to:  
download software, documentation, and product bulletins  
search the Technical Support Web site and the Nortel Knowledge Base for answers to  
technical issues  
sign up for automatic notification of new software and documentation for Nortel  
equipment  
open and manage technical support cases  
Getting Help over the phone from a Nortel Solutions Center  
If you don’t find the information you require on the Nortel Technical Support Web site, and have a  
Nortel support contract, you can also get help over the phone from a Nortel Solutions Center.  
In North America, call 1-800-4NORTEL (1-800-466-7835).  
Outside North America, go to the following Web site to obtain the phone number for your region:  
Getting Help from a specialist by using an Express Routing Code  
To access some Nortel Technical Solutions Centers, you can use an Express Routing Code (ERC)  
to quickly route your call to a specialist in your Nortel product or service. To locate the ERC for  
your product or service, go to:  
Getting Help through a Nortel distributor or reseller  
If you purchased a service contract for your Nortel product from a distributor or authorized  
reseller, contact the technical support staff for that distributor or reseller.  
SRG50 Configuration Guide  
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NN40140-500  
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15  
Chapter 2  
SRG50 overview  
The SRG50 is a software application that leverages the BCM50 platform. It is optimized to  
provide feature transparency to the main office call server and to act as a survival remote gateway  
in a CS 1000 IP branch office environment.  
SRG50 supports up to 80 survivable IP users with a single SRG application authorization code.  
SRG50 Release 3.0 operates with CS 1000 running Release 4.0, 4.5, 5.0, and 5.5.  
Configure SRG50 with Business Element Manager (Element Manager) or Network Configuration  
Manager (NCM). For detailed information about managing SRG50 with NCM, see your NCM  
documentation.  
For a summary of differences between SRG versions, see the following table.  
Table 2 Summary of supported features on the different SRG versions (Sheet 1 of 2)  
Features  
SRG 1.5 on BCM 200/ SRG50 1.0  
400  
SRG50 2.0  
SRG50 3.0  
Platform  
Available on BCM 200/ Available on standard Available on standard Available on standard  
400 hardware.  
BCM50 hardware.  
BCM50 and BCM50b  
hardware only.  
BCM50 and BCM50b  
hardware only.  
SRG 1.5 is not  
SRG50 1.0 is not  
available on BCM1000 available on BCM50a  
SRG50 2.0 is not  
SRG50 3.0 is not  
hardware.  
or BCM50e hardware. available on BCM50a, available on BCM50a,  
BCM50ba, BCM50e,  
and BCM50be  
hardware.  
BCM50ba, BCM50e,  
and BCM50be  
hardware.  
Call Servers  
Trunking  
CS1000 Release 3.0,  
4.0, and 4.5  
CS1000 Release 3.0,  
4.0, and 4.5  
CS1000 Release 4.0,  
4.5, and 5.0  
CS1000 Release  
4.0,4.5, and 5.0  
H323 is supported.  
H323 is supported.  
When CS1000 is  
supported by SRG,  
both H323 and SIP  
trunks are supported  
between them.  
H323 and SIP trunking  
are supported.  
IP clients  
IP Phones 2001, 2002 IP Phones 2001, 2002 IP Phones 2001, 2002 IP Phones 2001, 2002  
(phase 1 and 2), 2004 (phase 1 and 2), 2004 (phase 1 and 2), 2004 (phase 1 and 2), 2004  
(phase 1 and 2), 2007. (phase 1 and 2), 2007. (phase 1 and 2), 2007. (phase 1 and 2), 2007.  
IP KEM, IP Phone  
2050 series.  
IP KEM, IP Phone  
2050 (v1).  
IP KEM, IP Phone  
2050 series.  
IP KEM, IP Phone  
2050 series.  
IP Phone 1110, 1120, IP Phone 2033  
IP Phone 1120, 1140, IP Phone 1110, 1120,  
1140, IP Phone 2033  
(polycom), WLAN  
2210/2211/2212.  
(polycom), 2122  
IP Phone 2033  
1140, IP Phone 2033  
(polycom), WLAN  
2210/2211/2212.  
(polycom), WLAN  
2210/2211/2212.  
Number of IP  
clients  
90  
32  
80  
80  
supported  
NetIQ  
Not supported  
Not supported  
Not supported  
Not applicable  
Supported  
Supported  
Supported  
Supported  
PVQM  
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16 Chapter 2 SRG50 overview  
Table 2 Summary of supported features on the different SRG versions (Sheet 2 of 2)  
Features  
SRG 1.5 on BCM 200/ SRG50 1.0  
400  
SRG50 2.0  
SRG50 3.0  
ESA  
Supported  
Supported  
Supported  
Supported.  
On Site Notification  
(OSN) of E911 alarms  
supported for local  
users through a third  
party tool.  
Vo911  
Supported  
Supported  
Supported  
Supported  
Supported  
Supported  
Supported.  
Supported  
Enhanced  
firmware  
download  
Analog station The number supported Two analog station  
Two analog station  
modules are  
supported.  
Two analog station  
modules are  
supported.  
support  
depends on the  
hardware.  
modules are  
supported.  
For BCM200: two  
analog station modules nodes= 2*8 =16)  
are supported.  
number of analog  
number of analog  
nodes= 2*8 =16)  
number of analog  
nodes= 2*8 =16)  
number of analog  
nodes= 2*8 =16.  
For BCM400: Four  
analog station modules  
are supported.  
number of analog  
nodes= 4*8 =32)  
SRG50 creation  
An SRG50 is created by applying the appropriate SRG keycode to a functional BCM50 system.  
SRG50 is only supported on the BCM50 and BCM50b main units. Integrated router versions of  
the BCM50 (BCM50a, BCM50e, BCM50ba, and BCM50be) do not support the SRG50  
application.  
The Installation Checklist and Quick Start Guide is provided on the SRG50 Documentation CD  
that is shipped with your SRG50 system. Instructions in that guide are referenced in the following  
procedures. Also, the relevant BCM50 default IP addresses, user names, and passwords are  
excerpted from that guide and provided for your reference in the table BCM50 default IP addresses  
Table 3 BCM50 default IP addresses  
Port  
IP address  
Subnet mask  
OAM port (see Note)  
10.10.11.1  
255.255.255.252  
255.255.255.0  
BCM50 LAN (no router)  
192.168.1.2  
Note: DHCP is enabled on the OAM port and assigns the following IP address: 10.10.11.2  
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Chapter 2 SRG50 overview 17  
Table 4 BCM50 default user names and passwords  
Tool  
User ID| User Name  
Password  
Element Manager  
nnadmin  
nnadmin  
PlsChgMe!  
PlsChgMe!  
Onbox main web page  
(http:// [IP address]  
SRG50 keycode activation  
To create an SRG50, use Element Manager to activate the SRG keycode on a BCM50 system  
(BCM50 or BCM50b main unit).  
To activate the SRG keycode  
1
2
Locate the SRG authorization codes supplied with your product.  
Open Element Manager. For information about installing and opening Element Manager, see  
the Installation Checklist and Quick Start Guide (NN40020-308).  
3
With Element Manager, connect to the BCM system that you want to convert to an SRG. For  
information about connecting to a BCM system, see the Installation Checklist and Quick Start  
Guide (NN40020-308).  
4
5
Navigate to the Keycodes panel (Configuration > System > Keycodes).  
Click Connect to Nortel Keycode Retrieval system to obtain the keycode file for your  
system from the Nortel Keycode Retrieval System (KRS).  
For more information about keycodes, see the Keycode Installation Guide (NN40010-301).  
6
In the KRS, generate the keycode file for your system and save it on your management  
computer.  
Make sure the SRG feature is included in your keycode as well as any other features you  
require for your system. To use the on-site notification for Emergency Services, you need to  
include a LAN CTE port in your keycode.  
7
8
9
In Element Manager, return to the keycodes panel.  
Click Load File.  
Browse to the location on your management computer containing the keycode file for this  
system.  
10 Select the keycode file, and then click Open.  
The keycode file is applied.  
11 Reboot your system to complete the creation of the SRG.  
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18 Chapter 2 SRG50 overview  
To reboot the system  
1
2
In Element Manager, navigate to the Reset panel (Administration > Utilities > Reset).  
Click Reboot BCM50 System.  
To verify that the SRG has been successfully created  
1
In Element Manager, navigate to the Keycodes panel (Configuration > System > Keycodes).  
In the Feature licenses table, verify that the status of the SRG keycode is ACTIVE.  
Open the Resources folder (Configuration > Resources).  
2
Verify that there is a Survivable Remote Gateway panel.  
SRG50 and BCM50 features comparison  
The table Comparison of BCM50 and SRG50 on page 18 compares SRG50 and BCM50 features.  
Table 5 Comparison of BCM50 and SRG50 (Sheet 1 of 2)  
Item  
BCM50  
SRG50  
MBMs  
See the  
Recommended:  
Installation and  
Maintenance  
Guide  
ASM8+ (8 port Analog Station Module); DTM  
(Digital Trunk Module - 24 lines on either T1  
or E1 or PRI); BRI (4 line BRI S/T Module);  
GATM4 (Global Analog Trunk MBM - 4 port);  
GATM8 (Global Analog Trunk MBM - 8 port)  
ADID4 (4 analog lines)  
(NN40020-302)  
ADID8 (8 analog lines)  
G4x16 (16 digital ports and 4 analog trunks);  
G8x16 (16 digital ports and 8 analog trunks)  
Supported for ATA connections:  
DSM16 (Digital Station Module - 16 ports);  
DSM32 (Digital Station Module - 32 ports);  
4x16 Combo (16 digital ports, 4 analog  
trunks and 1 analog station);  
Does not support:  
DDIM (Digital Drop and Insert Mux)  
FEM (Fiber Expansion Module)  
DECT (Digital Enhanced Cordless  
Telecommunications)  
Digital telephone sets  
FCAPS  
Yes  
Yes  
No  
Yes, extended to include SRG-specific  
alarms and keycodes  
Network Configuration Manager  
Telset Administration  
Yes  
Yes  
N/A  
N/A  
Yes  
No  
CS 1000 Geographic Redundancy  
CS 1000 Network Bandwidth Management  
Yes  
Yes  
NN40140-500  
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Chapter 2 SRG50 overview 19  
Table 5 Comparison of BCM50 and SRG50 (Sheet 2 of 2)  
Item  
BCM50  
SRG50  
CS 1000 Adaptive Network Bandwidth Management N/A  
Yes  
Yes  
Yes  
CS 1000 Alternative Call Routing  
N/A  
N/A  
N/A  
CS 1000 Emergency Services Access  
Firmware Download from main office call server  
Yes (CS 1000 Release 4.0, 4.5, 5.0, and 5.5)  
(CS 1000 Release 4.0 requires patch  
MPLR22418)  
SRG-specific features for interaction with a main  
office call server, including: Heartbeat detection of  
WAN recovery; IP telephone redirection to main  
office in Normal Mode; Local Mode IP telephone  
interface; H.323 Gateway to PSTN under control of  
main office call server  
N/A  
Yes  
Supported devices  
The SRG50 Release 3.0 supports:  
IP Phones 1110, 1120E, and 1140E  
IP Phone 2001, 2002, 2004, 2007, and 2033  
IP Phone Key Expansion Module (KEM) – The IP Phone KEM is supported on an  
SRG with normal mode IP Phones. It does not function with local mode or test local  
mode IP Phones.  
IP Softphone 2050 v1/v2 and Mobile Voice Client (MVC) 2050  
WLAN Handsets 2210, 2211, and 2212  
WLAN Handsets 6120 and 6140  
WLAN Handsets 1210, 1220, and 1230  
Analog (500/2500 type) telephones  
Analog devices such as fax machines  
Note: Throughout this document, the IP Phones in this list are referred to collectively as  
IP Phones.  
The SRG50 is positioned primarily to support IP telephones and clients. However, analog devices  
can be supported using analog station modules (ASM), or by using an analog terminal adapter  
(ATA2) in conjunction with a digital station module (DSM). The SRG50 does not support digital  
or ISDN telephones.  
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20 Chapter 2 SRG50 overview  
SRG50 terminology  
The table SRG50 terminology on page 20 identifies SRG terms that may be unfamiliar to main  
office installers. They are provided to facilitate communications between SRG and main office  
personnel. In the table, the Element Manager path where the term appears is provided for reference  
and may not represent every appearance of the term.  
Table 6 SRG50 terminology (Sheet 1 of 2)  
Term  
Description  
Port  
For telephony configuration (Configuration > Telephony), a port is an internal number that  
identifies a physical termination point for a telephone set or a physical trunk.  
For the configuration of resources (Configuration > Resources) and data services  
(Configuration > Data Services), port is used in the context of the TCP/IP protocol suite.  
IP Terminal  
Sets  
IP telephone  
Configuration > Resources > Telephony Resources > IP & App Sets  
Can refer to actual telephones, or to the directory number (DN) assigned to the port to  
which a particular telephone is connected.  
Telephone  
Configuration > Resources > Telephony Resources > IP & App Sets  
Mapping DN to Telephone  
Configuration > Telephony > Sets  
DN  
Configuration > Telephony > Lines > Target Lines > Target Lines table > Control Set and  
Prime Set columns  
Trunks  
Trunks refer to external facilities that are connected to the SRG and provide incoming and  
outgoing communication paths. Paths can be physical (examples: loop; PRI; T1) or virtual  
(VoIP trunks).  
Configuration > Resources  
Loop trunk  
Lines  
An analog loop (FXO) that connects to the PSTN: a POTS line.  
A line is the generic term used for all communication paths, both internal and external.  
Configuration > Telephony > Lines  
Physical trunks.  
Physical Lines  
VoIP Lines  
Configuration > Telephony > Lines > Active Physical Lines  
(Lines 061 to 124)  
VoIP trunks.  
Configuration > Telephony > Lines > Active VoIP Lines  
(Lines 001 to 024)  
NN40140-500  
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Chapter 2 SRG50 overview 21  
Table 6 SRG50 terminology (Sheet 2 of 2)  
Term  
Description  
Target Lines  
Target lines are internal, virtual paths between trunks and telephones for incoming calls  
(only). They provide flexibility in the way trunks and telephones can be associated: target  
lines can be used to direct an incoming call to one or more telephones, direct one or  
more trunks to one phone, or direct several trunks (in a line pool) to one or more phones.  
Target lines are assigned to DNs. A target line triggers ringing voltage to the telephone(s)  
connected to the port(s) associated with the DN(s) that the target line is assigned to. (For  
example, if a unique target line is assigned to each DN, only one telephone rings when  
the DN is called. If several DNs are assigned to one target line, calling any of the DNs  
ring all of the associated phones.)  
Target lines are required for auto-answer trunks. Because VoIP lines are set internally to  
auto-answer, target lines are required for SRG operation.  
Element Manager provides two methods for assigning target lines to DNs.  
1) Configuration > Telephony > Sets > All DNs > All DNs table > Details for DN subpanel  
> Line Assignment tab  
or  
2) Configuration > Telephony > Lines > Target Lines > Target Lines table > Details for  
Line subpanel > Assigned DNs tab  
The first method provides a convenient way to assign the target line to the DN when the  
DN record is configured. The second method provides fields that allow incoming digit  
strings to be mapped to the DN.  
(Lines 125 to 268)  
For more information about target lines, see the Networking Configuration Guide  
(NN40020-603).  
Line pool  
Several of the same type of trunk configured as one group: a trunk group.  
Coordination with the main office call server  
Configuration of the SRG branch office requires datafill at both the SRG and the main office call  
server. Main office configuration drives SRG configuration, and Nortel recommends that the main  
office activities be concluded before undertaking SRG configuration.  
For information, see CS 1000 considerations on page 33.  
SRG operating modes  
The SRG has two operating modes:  
Normal mode  
Local mode  
Normal mode  
In normal mode (see the figure Normal mode on page 22), the SRG is connected to the main office  
call server over a WAN using VoIP trunks. From the perspective of the main office, the SRG is a  
branch office.  
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22 Chapter 2 SRG50 overview  
Figure 1 Normal mode  
Main office  
Branch office  
SRG  
IP phones  
IP phones  
VoIP connection over WAN  
Software  
phones  
WLAN  
handsets  
Normal mode  
IP telephones connected at the SRG are registered with the main office call server and are under  
main office control. They operate as branch user sets and have access to all telephony services and  
features that the call server offers to IP telephones connected directly to the main office.  
When a branch user set initiates a local PSTN call, the main office sets up the call using the VoIP  
trunks, which establishes a local media path. Emergency Services Access calls are similarly routed  
to the SRG PSTN. The telephone is redirected to local mode and the SRG initiates a local PSTN  
call to 911. For main office callers, the SRG acts as a VoIP-PSTN gateway during normal mode.  
When call forwarding has been configured, incoming PSTN calls to the branch user set are  
forwarded over VoIP trunks (either H.323 or SIP) to the main office, which terminates the call at  
the branch user. Similarly, calls from analog telephones connected to the SRG to the branch user  
set are forwarded to the main office over VoIP trunks, which then terminates the call at the branch  
user. Calls from the branch user set to the analog telephones at the SRG are routed over the VoIP  
trunks to terminate at the analog telephone. In all these call scenarios, only signaling messages go  
through the VoIP trunk. The media path is set up directly between the branch user set and the voice  
gateway at the SRG. This means that these calls do not use any WAN bandwidth between the main  
office and the branch office after calls are established.  
When a branch user IP telephone calls a main office IP telephone and vice versa, the call is a  
simple station-to-station call within the main office call server. Since the branch user IP telephone  
is physically remote from the call server, the media path goes through the WAN connection  
between the main office and the SRG, and thus uses WAN bandwidth, as demanded by the codec  
used in the call.  
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Chapter 2 SRG50 overview 23  
Local mode  
In the event of a WAN failure or the call server at the main office becomes unavailable, the  
IP Phones in normal mode revert to local mode automatically. In local mode, the IP users  
connected to the SRG are under the control of the SRG. When in local mode, main office call  
features are not available to users attached to the SRG. The SRG offers a set of basic features for  
the IP telephones, including access to the local PSTN, dialing emergency service numbers, and  
calling local extensions. For a complete list of local mode features, see Features in local mode on  
Note: The IP Phone KEM is supported on an SRG with normal mode IP Phones.  
It does not function with local mode or test local mode IP Phones.  
Figure 2 Local mode  
Main office  
Branch office  
SRG  
IP phones  
IP phones  
WAN connectivity lost,  
PSTN provides fallback  
Software  
phones  
SRG polling  
WLAN  
handsets  
Local mode  
The SRG handles all call processing. Calls between two IP telephones at the SRG are handled  
locally as a simple station-to-station call. When an IP telephone initiates a local PSTN call, the  
SRG routes the call to a trunk that is connected to the local PSTN. Incoming DID calls are also  
handled by the SRG and terminated on the appropriate IP telephone set.  
In local mode, the IP telephones do not have access to the main office network over the VoIP  
trunks. If alternate routes are configured, then calls can be made to the main office or other branch  
offices using the available PSTN trunks.  
Several situations, described below, can cause the IP phone to be in local mode.  
Initial registration  
When the IP telephone is installed, it first registers with the SRG, and is in local mode. When the  
SRG configuration at the main office and the SRG is complete the IP telephone is redirected to the  
main office, where it registers as a branch user and changes from local mode to normal mode.  
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24 Chapter 2 SRG50 overview  
Automatic registration with the main office  
When configured as a branch office user set, an IP telephone at the SRG automatically attempts to  
register with the main office when:  
The phone is in local mode because of loss of connectivity with the main office, and  
the SRG is redirecting it back to the main office because connectivity has been  
The phone is in local mode because Test Local Mode was invoked and the timer has  
expired or the Exit button is pressed.  
The phone is in local mode, the main office is a CS 1000, and this is the first time that  
the phone has been redirected to the main office.  
The IP telephone can fail to register with the main office for several reasons. These are detailed in  
Loss of WAN or VoIP connectivity  
The WAN or VoIP connectivity between the main office and the SRG can become unavailable if,  
for example, router failure occurs, the main office becomes unavailable, a WAN failure occurs, or  
the VoIP trunks reach capacity. When VoIP connectivity is lost, each IP telephone loses its  
connection with the main office terminal proxy server. The IP telephones reboot and reregister at  
the SRG, placing them in local mode. If enabled, call forwarding to the main office is  
automatically cancelled.  
The IP telephones remain under the control of the SRG until VoIP connectivity is confirmed.  
When confirmation is received, the IP telephones are automatically redirected to the main office;  
redirection requires no user intervention. If the telephone is busy at the time that connectivity is  
reestablished, the SRG redirects the phone when it is free.  
Test Local Mode  
Test Local Mode is a facility that allows the IP telephone to be redirected back to the SRG when it  
is in normal mode. This forces the IP telephone to go into local mode and allows the telephone  
user or system administrator to test local mode operation without taking down the VoIP trunks to  
the main office. Implementation of Test Local Mode depends on the main office call server. For  
SRG installation and configuration summary  
The SRG50 Configuration Guide provides information specific to configuring a BCM50 as an  
SRG. Information pertaining to generic BCM50 practices and procedures is provided in the  
BCM50 documentation suite.  
Generally, SRG50 activities leverage an installer’s general knowledge of BCM50 activities.  
However, Nortel recommends that the BCM50/SRG50 site engineer and installer familiarize  
themselves with SRG-specific requirements before starting any installation activities.  
NN40140-500  
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Chapter 2 SRG50 overview 25  
for installing and configuring an SRG50. The procedures in this document assume that the  
following activities have been completed:  
The BCM50, including media bay modules, cabling, telephones, and peripherals, have  
been installed.  
BCM50 administration has been set up.  
The basic parameters of the BCM50 have been configured.  
CS1000 main office system has been installed and configured to support SRG.  
The SRG has been connected to the LAN (as required) and WAN (to the main office).  
System functionality has been tested to this point.  
Attached devices have been installed and configured (for information about  
configuring IP Phones, see IP telephones setup and configuration on page 53).  
Non-SRG-specific networking and device configuration has been completed (for  
information about configuring the network, see Setting up the private VoIP network on  
page 65).  
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26 Chapter 2 SRG50 overview  
Process map for installing and configuring an SRG  
Figure 3 Process map for installing and configuring an SRG  
The main office is installed and running and the  
information to support SRG has been configured.  
Install and configure attached  
configuring the IP telephones. See  
page 79 when setting up analog  
trunks and devices.  
You have reviewed the SRG and BCM50  
system requirements and documentation.  
Locate the BCM50 that will be  
converted to the SRG.  
Install the physical components,  
including media bay modules,  
cabling, telephones, and  
peripherals, as per the Installation  
and Maintenance Guide.  
Set up the IP telephones for SRG  
operation, as per IP telephones  
Set up the private VoIP network, as  
Configure the basic parameters of the  
BCM50 as per the Installation  
Checklist and Quick Start Guide. (see  
the Installation and Maintenance  
Guide (NN40020-302) and the  
Configure the server-specific data for  
the main office call server (see the  
documentation for your call server).  
Administration Guide (NN40020-600)).  
Create the SRG (see SRG50 creation  
Register each telephone  
with the main office and the  
SRG.  
Complete the initial installation,  
connect the SRG to the LAN (as  
required) and WAN (to main office),  
and test the SRG system, as per the  
Installation and Maintenance Guide  
(NN40020-302).  
Troubleshoot any registration issues  
Test the telephones and analog  
equipment in normal mode.  
Set up SRG administration, and  
carry out non-SRG-specific  
networking and device configuration  
as per the Administration Guide  
(NN40020-600), the Networking  
Configuration Guide (NN40020-603),  
and the Device Configuration  
Guide (NN40020-300).  
Familiarize users with the services for  
NN40140-500  
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27  
Chapter 3  
Task summary  
The task summary offers a high level, chronological review of the tasks required to configure the  
SRG50. The paths (Xxxx > Yyyy > Zzzz) direct you to the appropriate Business Element Manager  
(Element Manager) panels.  
Foundation configuration  
Foundation configuration refers to configuration that is done as part of BCM50 foundation  
activities. The items identified here are significant for SRG operation and main office planning  
and installation.  
1
2
Configure the SRG IP address, net mask, and gateway.  
Configuration >System > IP Subsystem  
External Reference: Installation and Maintenance Guide (NN40020-302)  
Confirm the number of IP sets and VoIP trunks.  
Configuration > Resources > Application Resources  
The Licence column indicates the number of resources available.  
External Reference: Keycode Installation Guide (NN40010-301)  
Verify the global telephony settings.  
3
4
Configuration > Telephony > Global Settings  
External Reference: Device Configuration Guide (NN40020-300)  
Configure the Start DN (determined by the dialing plan).  
Administration > Utilities > Reset > Cold Reset Telephony Services button > Cold Reset  
Telephony dialog box > Start DN field  
Internal Reference: Basic parameters on page 67  
External Reference: Installation and Maintenance Guide (NN40020-302)  
Verify the DN length.  
5
i) For local calls between telephones on the SRG.  
Configuration > Telephony > Dialing Plan > General > Dialing Plan - General panel >  
Global Settings subpanel > DN length (intercom) field  
ii) For incoming calls from the PSTN  
Configuration > Telephony > Dialing Plan > Public Network > Dialing Plan - Public  
Network panel> Public Network Settings subpanel > Public Received number length  
field  
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28 Chapter 3 Task summary  
iii) For calls coming in from the private network  
Configuration > Telephony > Dialing Plan > Private Network > Dialing Plan - Private  
Network panel > Private Network Settings subpanel > Private Received number length  
field  
and  
Configuration > Telephony > Dialing Plan > Private Network > Dialing Plan - Private  
Network panel > Private Network Settings subpanel > Private DN length field  
(Private DN length is used for DPNSS applications only. See the Networking Configuration  
Guide (NN40020-603).)  
Internal Reference: Basic parameters on page 67  
External Reference: Networking Configuration Guide (NN40020-603)  
6
7
Verify the line pool assignment of VoIP trunks.  
In the default configuration, the VoIP trunks are assigned to line pool BlocA. Instructions in  
the SRG50 Configuration Guide assume that the default configuration has been maintained.  
Configuration > Telephony > Lines > Active VoIP Lines > Active VoIP Lines table > Line  
Type column > Line Type field  
External Reference: Networking Configuration Guide (NN40020-603)  
The SRG supports four analog loop trunks on the main unit*. Verify the line pool assignment  
of these trunks.  
In the default configuration, these trunks are assigned to line pool A. Instructions in the  
SRG50 Configuration Guide assume that the default configuration has been maintained.  
Configuration > Telephony > Lines > Active Physical Lines > Active Physical Lines table  
> Line Type column > Line Type field  
External Reference: Networking Configuration Guide (NN40020-603)  
* Category 1 countries  
IP telephone configuration  
1
Configure the registration password.  
Configuration > Resources > Telephony Resources > IP Sets row > Details for Module  
subpanel > IP Terminal Global Settings tab  
2
Configure the local mode indication (Advertisement/Logo).  
Configuration > Resources > Telephony Resources > IP Sets row > Details for Module  
subpanel > IP Terminal Global Settings tab  
NN40140-500  
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Chapter 3 Task summary 29  
3
Configure the IP telephone codec and jitter settings.  
Configuration > Resources > Telephony Resources > IP Sets row > Details for Module  
subpanel > IP Terminal Global Settings tab  
Configure the telephone (DN) records.  
4
5
Configuration > Telephony > Sets > All DNs  
Configure the received numbers.  
Configuration > Telephony > Lines > Target Lines  
External Reference: Networking Configuration Guide (NN40020-603)  
6
7
Decide on the call forwarding option.  
Configure the IP telephones.  
Dialing plan configuration  
1
Configure the private network type (CDP or UDP).  
Configuration > Telephony > Dialing Plan > Private Network > Dialing Plan - Private  
Network panel > Private Network Settings subpanel > Private network type  
External Reference: Networking Configuration Guide (NN40020-603)  
2
Enable MCDN TAT.  
Configuration > Telephony > Dialing Plan > Private Network > Dialing Plan - Private  
Network panel > MCDN subpanel  
External Reference: Networking Configuration Guide (NN40020-603)  
VoIP trunk configuration  
Your SRG VoIP trunks are either H323 trunks, SIP trunks, or a combination of both types.  
1
Configure routing table.  
Configuration > Resources > Telephony Resources > Modules panel > IP Trunks row >  
Routing Table.  
External Reference: Networking Configuration Guide (NN40020-603)  
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30 Chapter 3 Task summary  
2
Configure VoIP trunk QoS settings.  
Configuration > Resources > Telephony Resources > Modules panel > IP Trunks row >  
H323 Media Parameters tab or SIP Media Parameters tab  
External Reference: Networking Configuration Guide (NN40020-603)  
3
Enable or disable fallback.  
Configuration > Resources > Telephony Resources > Modules panel > IP Trunks row >  
H323 Settings tab or SIP Settings tab  
Internal References:  
External Reference: Networking Configuration Guide (NN40020-603)  
4
Configure H323 settings.  
Configuration > Resources > Telephony Resources > Modules panel > IP Trunks row >  
H323 Settings tab  
External Reference: Networking Configuration Guide (NN40020-603)  
5
6
Configure SIP settings.  
Configuration > Resources > Telephony Resources > Modules panel > IP Trunks row >  
SIP Settings tab  
External Reference: Networking Configuration Guide (NN40020-603)  
Assign VoIP trunks to a line pool (if default configuration has not been maintained).  
Configuration > Telephony > Lines > Active VoIP Lines  
Internal Reference:Line pools on page 74  
External Reference: Networking Configuration Guide (NN40020-603)  
Note: In configuring SIP trunks, the SRG BCM shall register itself with the configured H323ID with NRS  
in CS1000. It is with this ID, the SRG will get MOTPS IP address from NCS.  
7
Assign PSTN trunks to a line pool (if default configuration has not been maintained).  
Configuration > Telephony > Lines > Active Physical Lines  
Internal Reference:Line pools on page 74  
External Reference: Networking Configuration Guide (NN40020-603)  
8
Assign remote access packages to the VoIP trunks.  
Configuration > Telephony > Call Security > Remote Access Packages  
Internal Reference: SRG PSTN access on page 76  
External Reference: Networking Configuration Guide (NN40020-603)  
NN40140-500  
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Chapter 3 Task summary 31  
Call routing configuration  
1
Decide on the fallback scheme.  
Configure the outgoing routes (VoIP and PSTN fallback).  
Configuration > Telephony > Dialing Plan > Routing  
2
External Reference: Networking Configuration Guide (NN40020-603)  
Configure access to the SRG PSTN (for both local and tandem calls).  
Configuration > Telephony > Dialing Plan > Routing  
3
4
Internal Reference: SRG PSTN access on page 76  
External Reference: Networking Configuration Guide (NN40020-603)  
Configure for Network Bandwidth Management and Advanced Network Bandwidth  
Management.  
External Reference: Branch Office: Installation and Configuration (553-3001-214) and Data  
Networking for Voice over IP (553-3001-160)  
5
Configure for Alternative Call Routing.  
External Reference: What’s New for Communication Server 1000 Release 4.5 (553-3001-015)  
Redirection and call forward configuration  
1
Configure the main office settings.  
Configuration > Resources > Survivable Remote Gateway > S1000 Main Office Settings  
tab  
Configure the IP terminal settings.  
2
Configuration > Resources > Survivable Remote Gateway > S1000 IP Terminal Details  
tab  
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32 Chapter 3 Task summary  
NN40140-500  
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Chapter 4 CS 1000 considerations 33  
Chapter 4  
CS 1000 considerations  
A Survivable Remote Gateway (SRG) extends CS 1000 features from the main office and provides  
a business continuity solution to one or more remote SRG locations (branch offices).  
The Main Office Configuration Guide for SRG50 (NN43001-307) provides information specific to  
the configuration of an SRG50 on the CS 1000. This guide is included on the SRG50  
documentation CD for your reference. Access to other CS 1000 documentation may be required if  
personnel are not familiar with configuration of branch offices on the CS 1000.  
The following activities are specific to SRG50 configuration when the main office call server is a  
CS 1000:  
CS 1000 and SRG terminology comparison  
configuration-related terms and contexts of the CS 1000 and the SRG.  
Table 7 Comparison of CS 1000 and SRG terms and contexts (Sheet 1 of 2)  
Term or Context  
CS 1000  
SRG  
Dialing plan  
Type of number  
Numbers  
on-net / off-net dialing  
CDP / UDP / GDP / TNDN  
TN (terminal number)  
Private / Public network dialing  
CDP / UDP / no equivalent  
MOTN (main office terminal number)  
TN = MOTN. That is, the TN from the main office is entered on the SRG in the  
BUID (branch user ID)  
The CS 1000 BUID is entered on the  
panel on page 48) but there is no SRG  
equivalent for BUID.  
The dialable number of an IP telephone  
at the SRG when it is called from a  
phone located at the main office or  
another branch office.  
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34 Chapter 4 CS 1000 considerations  
Table 7 Comparison of CS 1000 and SRG terms and contexts (Sheet 2 of 2)  
Term or Context  
CS 1000  
SRG  
DN (directory number)  
DN (directory number)  
The dialable number of a telephone at The dialable number of a telephone at  
the main office when it is called from  
another phone at the main office.  
the SRG when it is called from another  
phone at the SRG.  
In the case of a CDP dialing plan, it is recommended that the BUID and the SRG  
DN be the same.  
In the case of a UDP dialing plan, the BUID has the form: <VOIP Trunk Access  
Code> + <LOC> + <DN>. In this case, it is recommended that the SRG DN be  
the same as <DN>.  
The dialable number of an IP telephone at the SRG, when dialed from another  
phone at the SRG, remains the same in both normal and local mode if the  
preceding recommendations are implemented.  
AC1  
VOIP Trunk Access Code (see  
Destination code for VoIP trunks (see  
74)  
AC1 = VOIP Trunk Access Code = Destination code for VoIP trunks  
Routing  
distant steering codes (DSC), trunk  
steering codes (TSC), local steering  
codes (LSC)  
call routing, destination codes, line pool  
access codes  
digit manipulation table  
dialout digits (routing)  
Private  
Numbering Plan ID  
ISDN/Telephony (E.164), Private,  
Telephony (E.163), Telex (F.69), Data  
(X.121), National Standard  
Access codes  
7 = system trunk access  
7 = not assigned  
8 = not assigned  
(SRG: Destination codes)  
8 = Basic Alternate Route Selection  
(BARS)/Network Alternate Route  
Selection (NARS)  
9 = public exchange access  
9 = line pool A access code  
scheduled call routing  
Zone ID  
Network Class of Service  
Facility Restriction Level (FRL)  
Network Bandwidth Management Zone ID  
Virtual Private Network ID (VPNI)  
CS 1000 Zone ID = SRG Zone ID  
ZDP = VPNI.  
That is, the CS 1000 Zone ID is entered on the SRG in the Zone ID field, and the  
ZDP is entered on the SRG in the VPNI field (see Bandwidth management on  
Trunks  
public exchange  
PSTN  
IP telephone password  
installer password  
global password  
The two passwords can be made the same. See Registration password on page  
NN40140-500  
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Chapter 4 CS 1000 considerations 35  
Normal and local mode overview  
Normal mode and local mode overview provides a description of the following sections:  
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.  
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
2
IP Phone may have just booted up.  
IP Phone cannot communicate to the main office because of a WAN failure or a failure of the  
main office components.  
Note: 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 CS1000  
call server for any reason (WAN failure, main office call server failure, main office Signaling  
Server failure), the SRG50 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 re-register to the  
SRG50, placing them in local mode.  
Once this has occurred, 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 SRG50 call  
services. As such, the normal main office call server features are not available. The SRG50 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  
SRG50 Configuration Guide  
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36 Chapter 4 CS 1000 considerations  
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 SRG50 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 SRG50 handles all call processing, calls between two IP phones at the  
SRG50 are handled locally as a simple station-to-station call. When an IP Phone initiates a local  
PSTN call, the SRG50 routes the call to a trunk that is connected to the local PSTN. Incoming DID  
calls are also handled by the SRG50 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.  
While in local mode, the SRG50 system continues to monitor for a main office CS1000 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 redirection occurs almost immediately. This  
reinstates the CS1000 normal user interface and feature set for the IP Phone user, on a user by user  
basis.  
The SRG50 system implements the same interface used by the MG1000B system to interact with  
the main office CS1000 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 which  
automatically reregisters users once WAN or system failure has recovered.  
Survivability  
SRG is specifically designed to provide automatic survivability against WAN failure, main office  
Call Server failure, main office Signaling Server failure, and NRS (gatekeeper) failure.  
SRG supports the Geographic Redundancy feature.  
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.  
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 network connection  
server (NCS).  
NN40140-500  
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Chapter 4 CS 1000 considerations 37  
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’s 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 NRS for the main office Virtual Trunk node IP address to redirect the IP Phone. If the  
Primary NRS is down or unreachable, the SRG queries the Alternate NRS, 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 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 NRS fails  
and no Alternate NRS is available. They can call any main office telephone or IP Phones in normal  
mode in other branch offices.  
SIP/ H323 trunks are used for Voice traffic alone. Proprietary TPSAR protocol is used for WAN  
link health check and for Main Office TPS address discovery through NRS.  
However, they cannot call any SRG analog (500/2500-type) telephones or any external numbers  
through the SRG trunks because the Virtual Trunks are not available. (SRG analog [500/  
2500-type] telephones are accessible if alternate routing is available through the PSTN.)  
For more information about virtual trunks for SRG50, see the Main Office Configuration for  
Survivable Remote Gateway 50 (NN43001-307).  
Recovery to normal mode  
Once communication is re-established 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.  
Testing the telephone in local mode  
From normal mode, the branch user has the option of going to local mode manually using the Test  
local mode feature. 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 from the set to return to normal mode.  
SRG50 Configuration Guide  
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38 Chapter 4 CS 1000 considerations  
Virtual trunk capacity  
The SRG supports a number of simultaneous calls depends on the specific codec type used.  
SRG50 supports 24 virtual IP trunks that are either H.323 only, SIP only, or a combination of the  
H.323 and SIP.  
In normal mode, the codec selection used is controlled by specific programming of the CS1000. In  
this case: SRG50 supports up to a maximum of 15 Virtual trunks unless both the intrazone and  
interzone codecs are configured as Best Quality (G.711) in which case, the maximum number of  
virtual trunks would be 24.  
In local mode, if the WAN has failed, there are no longer any virtual trunks available between the  
SRG50 and CS1000. However, the SRG50 will continue to convert calls from IP terminals for  
communication via the PSTN. In this case, if G.711 is used (recommended), the number of  
simultaneous calls from IP terminals to the PSTN supportable is a maximum of 24.  
For more information about virtual trunks for SRG50, see the Main Office Configuration for  
Survivable Remote Gateway 50 (NN43001-307).  
Vacant Number Routing (VNR)  
The SRG does not support Vacant Number Routing (VNR). Instead, the SRG uses Call Forward  
All Calls to emulate VNR for the IP telephones that are in normal mode. Call Forward All Calls is  
automatically cancelled when the phones revert to local mode.  
A single destination code and route (or a group of destination codes and routes) can be configured  
on the SRG to route all calls not terminated locally by the SRG. These calls are routed over the  
VoIP trunks. If the VoIP trunks become unavailable, the calls are routed to the proper location  
using PSTN fallback. This feature is similar to the VNR feature on the Media Gateway 1000B  
(MG1000B).  
Seamless dialing requires that the start digit of the DNs are unique for each system (coordinated  
dialing plan). If the start digit is the same on both systems, the local users on the SRG must dial a  
separate destination code before the main office DN.  
For details about dialing plan and routing configuration, see Setting up the private VoIP network  
Bandwidth management  
Three levels of bandwidth management are supported by the CS 1000:  
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Chapter 4 CS 1000 considerations 39  
Network Bandwidth Management (NBWM)  
The SRG interoperates with the Network Bandwidth Management (NBWM) feature in a manner  
similar to Media Gateway (MG) 1000B, though only G.711 and G.729 codecs are supported. At  
the SRG, a Virtual Private Network ID (VPNI) and Zone ID are entered with values defined by the  
ACR on page 40). The VPNI and Zone ID allow the CS 1000 to recognize that H.323 and SIP calls  
to and from the SRG are from a specific Bandwidth Management zone.  
NBWM allows bandwidth zones to be configured on a network basis so that codec selection and  
bandwidth allocation software can identify whether IP telephones or gateways are physically  
collocated (in the same bandwidth zone) even though they are controlled by different call servers.  
NBWM is used to define the codec selection policy and track bandwidth used for calls that  
traverse the WAN (interzone calls) and the LAN (intrazone calls). The bulk of configuration for  
NBWM is done at the main office.  
Adaptive Network Bandwidth Management (ADBWM)  
As with NBWM, only the VPNI and Zone ID are required at the SRG in order to implement he  
Adaptive Network Bandwidth Management (ADBWM) feature on the SRG (see Bandwidth  
ADBWM uses real-time interaction to enhance the performance of Voice over Internet Protocol  
(VoIP) networks. ADBWM adjusts bandwidth limits and takes corrective action in response to  
Quality of Service (QoS) feedback. This adjustment occurs dynamically, while calls are in  
progress. A call server with ADBWM uses VPNIs and Zone IDs to keep track of the bandwidth  
being used between its own zone and zones belonging to other call servers. If the interzone QoS  
degrades below an acceptable level, the available bandwidth is reduced automatically between the  
two zones. When the QoS between the two zones improves, the bandwidth limit is allowed to  
return to normal.  
Alternative Call Routing (ACR)  
Configuration for Alternative Call Routing (ACR) at the SRG includes datafilling the Virtual  
Private Network ID (VPNI) and Zone ID required by NBWM and ADBWM. However, additional  
configuration is required and depends on the type of trunking provisioned at the main office:  
Attendant service or DID trunks (see Bandwidth management configuration: NBWM, ADBWM,  
ACR for NBWM allows a station-to-station call (that is, a call that does not use a trunk) to  
overflow to traditional routes. Overflow can occur if there is insufficient interzone bandwidth  
available to carry the call, or if the QoS has degraded to unacceptable levels. The feature applies to  
station-to-station calls between a branch office and main office as well as from one branch office  
to another branch office, provided both stations are registered to the same main office.  
Network administrators who do not want calls to be blocked, yet have a limited amount of  
bandwidth available, can use ACR to overflow calls to conventional trunks (PSTN or Tie/MCDN).  
ACR allows calls to be routed by overflowing them, trading off the capital cost of WAN  
bandwidth against the incremental cost of overflowed calls.  
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40 Chapter 4 CS 1000 considerations  
Bandwidth management configuration: NBWM, ADBWM, and  
ACR  
To configure SRG for NBWM and ADBWM  
1
2
Obtain the Virtual Private Network ID and the Zone ID numbers configured at the main office.  
Use Element Manager to enter these numbers in the appropriate fields at Configuration >  
Telephony > Dialing Plan > Private Network > Dialing Plan - Private Network panel > VoIP  
Figure 4 Dialing Plan - Private Network panel, VoIP subpanel  
To configure Alternative Call Routing with attendant service  
1
For more information, see Outgoing calls configuration on page 74, and see also the  
Networking Configuration Guide (NN40020-603).  
2
Obtain the ALTPrefix for the SRG (configured at the main office).  
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Chapter 4 CS 1000 considerations 41  
3
Define a route to the main office Attendant over the PSTN.  
a
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing) and select the Routes tab.  
b
c
d
Add a new route (for example, 997).  
Ensure that the DN Type is Public (Unknown).  
In the External Number field, enter the PSTN number of the main office Attendant  
telephone.  
e
Assign the PSTN line pool to the route (select the line pool from the Use Pool list; default  
is A).  
4
Add a destination code.  
a
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing) and select the Destination Codes tab.  
b
Add a new destination code.  
Use the ALTPrefix as the destination code.  
c
d
e
In the ALTPrefix Destination Code row, select the Normal Route field.  
Enter the route for the Attendant telephone (997).  
In the adjacent Absorbed Length field, select All from the list.  
When the SRG receives the ALTPrefix+DN digits from the main office, it looks up the destination  
code table, finds a match for the ALTPrefix, dumps all the digits (ALTPrefix+DN), and dials the  
main office Attendant.  
To configure Alternative Call Routing with DID trunks  
Note: In order to support DID trunks, the MODN dialed by an SRG  
caller must match the DID digits.  
Note: In the following procedure, it is assumed that the MODN matches  
the XXXX portion of the DID’s NPA-NXX-XXXX; and that the  
ALTPrefix is 3 digits.  
1
For more information, see Outgoing calls configuration on page 74 and the Networking  
Configuration Guide (NN40020-603).  
2
3
Obtain the ALTPrefix for the SRG (configured at the main office).  
Define a route for the NPA-NXXX portion of the main office DID numbers.  
a
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing) and select the Routes tab.  
b
Add a new route (for example, 996).  
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42 Chapter 4 CS 1000 considerations  
c
d
Ensure that the DN Type is Public (Unknown).  
In the External Number field, enter the NPA-NXXX of the DID trunks that serve the  
main office.  
e
Assign the PSTN line pool to the route (select the line pool from the Use Pool list; default  
is A).  
4
Add a destination code.  
a
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing) and select the Destination Codes tab.  
b
Add a new destination code.  
Use the ALTPrefix as the destination code.  
c
d
e
On the ALTPrefix Destination Code row, select the Normal Route field.  
Enter the route for the route added above (996).  
In the adjacent Absorbed Length field, select 3 from the list.  
When the SRG receives the ALTPrefix+DN digits from the main office, it looks up the destination  
code table, finds a match for the ALTPrefix, dumps the 3- digit ALTPrefix, appends the DN to the  
External Number and dials the External Number+DN.  
Emergency Services Access (ESA) configuration  
The Main Office Configuration for Survivable Remote Gateway 50 (NN43001-307) guide covers  
the procedures for configuring Emergency Services Access on both the SRG and the CS 1000. The  
information here expands on the SRG procedure in that guide.  
There are two methods of ESA support:  
When a user dials 911, the telephone is redirected to the Branch Office and connects to  
the SRG in local mode. The Main Office indicates the reason for redirection (as e911)  
in the DRAM. The SRG dials 911 across the PSTN. The telephone remains in local  
mode after the 911 call and is redirected to the Main Office after 10 minutes.  
On-site notification for 911 — An external tool records an alarm when 911 calls are  
made at the branch office. This is applicable to both telephones that are in local mode  
due to the activation of a 911 call as well as locally connected analog sets. The  
external tool connects to the SRG through LAN CTE.  
Note: This procedure applies only to redirected IP telephones when the  
SRG is in normal mode.  
For IP telephones in local mode, and for other telephones at the SRG, see  
the Networking Configuration Guide (NN40020-603) for configuring  
emergency services.  
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Chapter 4 CS 1000 considerations 43  
To configure Emergency Services Access  
1
Verify that a remote access package has been assigned to the VoIP trunks (see Remote Access  
2
3
Obtain the ESA Special Number (SPN).  
In Element Manager, access the Dialing Plan - Routing panel (Configuration > Telephony >  
Dialing Plan > Routing) and click the Destination Codes tab.  
4
5
Add a destination code corresponding to the ESA SPN for the SRG branch office.  
In the Destination Codes table, click the Absorbed Length field of the ESA SPN destination  
code. The numbers indicate the number of digits the SRG absorbs, from left to right.  
6
7
Select the number of digits to absorb so that just the Emergency Services DN (ESDN)  
remains.  
In the Destination Codes table, click the Normal Route field of the ESA SPN destination  
code. Enter a public route to the PSTN trunks.  
Note: The Normal Route field defaults to 000. Route 000 (click the  
Routes tab) is preconfigured to Use Pool A and cannot be changed. Pool  
In  
A is preconfigured for PSTN trunks in the default state. Hence, if the  
default state of Pool A has not been changed, leave the Normal Route  
field as 000.  
To check the state of Pool A, navigate to Configuration > Lines >  
Active Physical Lines. Pool A must be assigned to at least one Trunk  
Type that provides access to the PSTN.  
Do not configure Alternate Routes.  
8
9
Navigate to the Dialing Plan - Public Network panel (Configuration > Dialing Plan > Public  
Network).  
In the Public Network DN Lengths subpanel, verify that there is a DN Prefix of 911 with a  
DN Length of 3.  
If not, add the 911 DN Prefix. If required, double click the DN Length field and then change  
the value to 3.  
To configure on-site notification  
To use the on-site notification for Emergency Services, you need to include a LAN CTE port  
in your keycode.  
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CS 1000 information for the SRG  
In order to redirect IP telephones and forward calls to the main office (Call Forward All Calls  
feature), the SRG requires information about the main office network environment. This  
information is recorded through Element Manager on the S1000 Main Office Settings panel.  
S1000 Main Office Settings panel  
The table S1000 Main Office Settings on page 44 lists and describes each field of the S1000 Main  
Office Settings panel. Record the actual value in the Values column to facilitate configuration and  
provide a record of the datafill.  
Table 8 S1000 Main Office Settings (Sheet 1 of 3)  
Field  
Values  
Description  
Primary Network  
Connect Server Address  
<ip address>  
IP address of the primary NCS.  
Alternate Network  
Connector Server  
Address  
<ip address>  
IP address of the alternate NCS, if deployed. If not,  
enter the same address as for Primary Network  
Connect Server Address.  
Network Connect Server 16500 (default)  
Port  
Port on the SRG used to connect to the NCS.  
Range: 0 to 65535  
Heartbeat Protocol Port  
VOIP Trunk Access Code  
16501 (default)  
Port on the SRG that the SRG uses to monitor the  
status of the connection with the main office  
terminal proxy server (that is, to confirm  
connectivity with the main office)  
Range: 0 to 65535  
Access code for the main office VoIP trunk.  
Required for UDP dialing plan only. Ignored for  
CDP dialing plan, field can be left blank.  
VOIP Trunk Access Code = Destination code for  
VoIP trunks* = AC1**  
* Destination code for VoIP trunks is entered during  
configuration for advanced routing. see “Outgoing  
** For a UDP dialing plan, AC1 is the access code  
in the digit string <AC1> <LOC> <DN>  
Test Local Mode Timeout 10 minutes (default)  
Period that an IP telephone remains in local mode  
after being set in local mode manually. Telephone  
returns to normal mode automatically at the end of  
the time-out.  
Range:  
2 to 10 minutes  
Local mode can be invoked by the Test Local Mode  
button on the telephone or by command from the  
main office.  
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Chapter 4 CS 1000 considerations 45  
Table 8 S1000 Main Office Settings (Sheet 2 of 3)  
Field  
Values  
Description  
H323 ID  
SRG* (default)  
Gatekeeper setting that identifies the SRG.  
*This setting must be changed.  
This value must match the value in the Alias  
see the Networking Configuration names field of the Local IP gateway:  
Guide (NN40020-603) for naming  
conventions.  
Configuration > Resources > Telephony  
Resources > Modules panel > Module type  
column: select IP Trunks > Details for Module:  
Internal details panel > Local Gateway tab >  
Gatekeeper Support subpanel > Alias names  
The SRG acts as a SIP endpoint and obtains the  
Main Office TPS IP address from the NCS using  
the H323 ID.  
Configure this field as the same SIP endpoint name  
as you configure for the NRS endpoint name. This  
is a generic field used for both SIP and H.323  
virtual trunk configurations.  
Numbering Plan ID  
Unknown  
The type of numbering plan at the main office.  
ISDN/Telephony (E.164)  
Private  
Telephony (E.163)  
Telex (F.69)  
Data (X.121)  
National Standard  
Default: Private  
Type of Number  
Unknown  
The main office dialing plan.  
International Number  
National Number  
Special Number  
Subscriber Number  
ESN LOC (UDP)  
ESN CDP  
Ensure that the SRG private dialing plan is  
configured to match the selected value.  
ESN Special Number  
Default: ESN CDP (for CDP dialing  
plans) (BUID = DN)  
UDP dialing plans: select ESN  
LOC (UDP) (BUID = LOC+DN)  
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Table 8 S1000 Main Office Settings (Sheet 3 of 3)  
Field  
Values  
Description  
Node ID  
9999 (default)  
Automatically written to the IP telephone firmware  
when the IP telephone registers with the main  
office.  
Range: 0 to 9999  
Used to identify the node on the main office  
associated with the IP telephone DN.  
MO Access Code Length For CDP dialing plans: set to 0  
The number of digits to add to the BUID (DN) so  
the main office system can determine if the  
incoming call is valid.  
For UDP dialing plans: set to  
length of line pool access code or  
destination code in front of LOC.  
Range: 0 to 34  
To datafill the S1000 Main Office Settings panel  
1
In Element Manager, navigate to Configuration > Resources > Survivable Remote Gateway  
2
3
Select the S1000 Main Office Settings tab.  
Enter the information in the appropriate fields.  
Figure 5 S1000 Main Office Settings panel  
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Chapter 4 CS 1000 considerations 47  
IP telephones redirection  
Once an IP telephone at the SRG is configured (see IP telephones setup and configuration on page  
53), it automatically registers with the SRG (S1). To configure an IP telephone for redirection to  
the main office call server, SRG-specific datafill is required. The SRG-specific configuration  
includes:  
To configure SRG using SIP trunks, CS1K must accept the end point as an H323 and SIP entity  
and accept it as h323 and SIP endpoint. You do not need to enter BCM SRG50 configuration  
under Telephony Resources > IP Trunks > H323 Setting. The entire configuration for SIP trunking  
must be entered under SIP settings, SIP Proxy, SIP media parameters, SIP URI Maps and SIP  
Authentication.  
IP telephones numbers and models  
SRG DNs are assigned to IP telephones using Element Manager. Redirection to the main office  
requires configuration at the SRG to associate the SRG DN with the CS 1000 terminal number  
(TN) and the corresponding branch user ID (BUID).  
When the CS1000 TN is configured, the IP telephone model forms part of the record. At the SRG,  
the actual IP telephone configured to an SRG DN must be the same model that is configured in the  
TN record that is associated with the SRG DN.  
The table IP telephone numbers and models on page 47 provides a record of the CS 1000 TN and  
BUID, the SRG MOTN and DN, and the model of phone, to facilitate installation and provide a  
permanent record of the configuration.  
Table 9 IP telephone numbers and models (Sheet 1 of 2)  
TN (CS 1000)  
MOTN (SRG)  
(same number)  
BUID (same  
number at CS 1000 SRG DN  
and SRG)  
IP telephone model  
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48 Chapter 4 CS 1000 considerations  
Table 9 IP telephone numbers and models (Sheet 2 of 2)  
TN (CS 1000)  
MOTN (SRG)  
(same number)  
BUID (same  
number at CS 1000 SRG DN  
and SRG)  
IP telephone model  
S1000 IP Terminal Details panel  
Element Manager provides SRG-specific panels for recording the CS 1000 TN and BUID that are  
associated with a particular SRG DN. The table SRG S1000 IP Terminal Details fields on page 48  
lists and describes the fields on the S1000 IP Terminal Details panel.  
Table 10 SRG S1000 IP Terminal Details fields (Sheet 1 of 2)  
Field  
DN  
Values  
Description  
Read-only  
The SRG DN assigned to the telephone.  
The SRG DN must be configured before proceeding with the  
procedures that follow in this section. See Telephone (DN) records  
Hardware  
ID  
Read-only  
Read-only  
Hardware ID.  
Unique for each IP telephone.  
Status  
Current status of the telephone.  
See IP terminal details on page 84 (expand the field to read the entire  
status message).  
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Chapter 4 CS 1000 considerations 49  
Table 10 SRG S1000 IP Terminal Details fields (Sheet 2 of 2)  
Field  
Values  
Description  
Firmware  
Version  
Read-only  
Updated by the main office when a terminal is sent back to the SRG for  
firmware upgrade purposes. The field specifies the firmware version  
required by the main office.  
MOTN  
BUID  
XXX  
Required for telephone redirection.  
The field is the main office TN associated with the IP telephone.  
CDP network: <DN>  
Required for telephone redirection.  
The field represents the dialable number of an IP telephone at the SRG  
when it is called from a phone located at the main office or another  
branch office.  
UDP network: <VoIP access  
code> + <LOC> + <DN>  
The BUID at the SRG must be the same as the BUID at the main office.  
MO TPS  
Read-only  
This field echoes the address of the main office terminal proxy server  
when the IP telephone is redirected.  
Note: The SRG DNs must be configured before the following  
procedures can be undertaken. See Telephone (DN) records  
To enter the MOTN and BUID  
1
2
3
In Element Manager, navigate to Configuration > Resources > Survivable Remote  
Gateway.  
Select the S1000 IP Terminal Details tab (see the figure S1000 IP Terminal Details panel on  
page 50).  
Refer to the numbers and models recorded in the table IP telephone numbers and models on  
4
5
6
Select the required DN.  
Press the Modify button.  
Enter the MOTN and the BUID in the appropriate fields.  
To redirect the telephone to the main office call server  
1
In Element Manager, navigate to Configuration > Resources > Survivable Remote  
Gateway.  
2
Select the S1000 IP Terminal Details tab (see the figure S1000 IP Terminal Details panel on  
page 50).  
3
4
Select the DN of the telephone to be redirected.  
Press the Redirect Set button.  
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Figure 6 S1000 IP Terminal Details panel  
IP telephone settings  
For IP telephones that are redirected to the main office call server, incorporate the settings shown  
in the table Configuration settings for redirected IP Phones on page 50. The Device Configuration  
Guide (NN40020-300) and the Networking Configuration Guide (NN40020-603) provide detailed  
instructions for configuring IP telephones.  
Table 11 Configuration settings for redirected IP Phones  
Parameter  
Setting  
S1 IP  
SRG IP address  
S1 Port  
7300  
S1 Action  
S1 Retry Count  
S2 IP  
1
1
SRG IP address  
S2 Port  
7300  
S2 Action  
S2 Retry Count  
1
1
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Chapter 4 CS 1000 considerations 51  
Firmware upgrade  
The redirected IP telephones at the SRG are under the control of the main office call server for the  
majority of their deployment and receive all of their features in that context. Therefore, the version  
of IP set firmware must align with the requirements of the CS 1000.  
Supported firmware  
The table Supported IP clients and firmware versions on page 51 lists the IP clients and related  
firmware versions supported on the SRG50. The SRG50 column indicates the firmware versions  
included with the SRG software. The CS 1000 columns identify the version of firmware to use for  
specific releases.  
Table 12 Supported IP clients and firmware versions  
IP Client  
SRG50  
CS 1000  
Release 4.5  
Release 4.0  
Release 5.0  
Phase I: 2002, 2004  
B65 or greater  
D98 or greater  
B65 or greater  
D88 or greater  
B76 or greater  
B76 or greater  
0604DBG  
Phase II: 2001, 2002,  
2004  
0604DBG or  
0604DAS  
IP Phone 2007  
C22 or greater  
C22 or greater  
0621C4J or 0621C3J 0621C4J  
WLAN Handsets 2210/ Not embedded in  
Use the 2210/2211  
GA firmware load.  
97.059 or greater  
97.059 or greater  
2211  
SRG software  
WLAN Handset 2212  
Not embedded in  
SRG software  
97.039 or greater  
97.059 or greater  
97.059 or greater  
WLAN Handsets 6140/ Not embedded in  
6120  
SRG software  
WLAN Handsets 1210/  
1220/1230  
062AC57  
IP Phone 2033  
Not embedded in  
SRG software  
S12 or greater  
S12 or greater  
S12 or greater  
IP Phone 1110  
IP Phone 1120E  
IP Phone 1140E  
IP Softphone 2050  
MVC 2050  
0623C38  
0624C1B  
0625C1B  
0624C1E  
0624C1E  
0625C1E  
0623C4B or 0623C3C 0623C4D  
0624C4B or 0624C3C 0624C4D  
0625C4B or 0625C3C 0625C4D  
build 385 or greater build 385 or greater build 385 or greater  
build 126 or greater build 126 or greater build 126 or greater  
2.1.355 or greater  
2.1.202 or greater  
Firmware upgrade procedure  
When an IP telephone requires a firmware upgrade, the CS 1000 uses the umsUpgradeAll  
command, or variant, to redirect the telephone back to the SRG for upgrading. If the required file  
does not exist on the SRG, or its version is incorrect, the SRG initiates an FTP session to the TPS  
for that phone to retrieve the required file. The SRG upgrades the phone and redirects it back to the  
CS 1000. After the SRG receives new firmware from the Main Office for a specific IP telephone,  
all the IP telephones of the same type are loaded with the new firmware.  
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Chapter 5  
IP telephones setup and configuration  
IP telephone setup and DN configuration are described in detail in the IP Telephone Installation  
and Configuration Guide and the Networking Configuration Guide (NN40020-603), respectively.  
SRG-specific procedures and settings include:  
Registration password  
If a registration password is configured on the SRG, the IP telephone installer must enter the  
password before the telephone can be configured.  
To set the IP telephone registration password  
1
2
In Element Manager, navigate to Configuration > Resources > Telephony Resources.  
On the Modules panel, locate the Module type column and select the IP & Application Sets  
3
On the Details for Module: subpanel, select the IP Terminal Global Settings tab.  
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54 Chapter 5 IP telephones setup and configuration  
There are three fields that define the password registration process (see the table Password  
Enable registration, Enable global registration password, and Global password.  
Table 13 Password registration parameters  
Enable registration  
Must be selected to allow IP telephones to register.  
Enable global registration  
password  
Select if a password is going to be entered in the next  
field, Global password.  
If selected, the installer must enter the Global password  
(below) at the IP telephone before the telephone can be  
configured.  
Global password  
Enter the password.  
If no password is entered, or the Enable global  
registration password is not selected, no password is  
required to configure the IP telephone.  
Note: Nortel recommends that you synchronize this  
password with the CS 1000 password.  
Figure 7 Telephony Resources panel, IP and App Sets  
4
5
Select and enter the values to meet the password requirements of your installation.  
Set the Auto-assign DNs check box according to the requirements of your installation (if set,  
the SRG automatically assigns DNs, see the Device Configuration Guide (NN40020-300) for  
details).  
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Chapter 5 IP telephones setup and configuration 55  
Local mode indication  
When an IP telephone is in local mode, a message is displayed on the phone to indicate the local  
mode state to the user. The default setting is Local Mode.  
To change local mode indication  
1
2
In Element Manager, navigate to Configuration > Resources > Telephony Resources.  
On the Modules panel, locate the Module type column and select the IP Sets row (see the  
3
4
On the Details for Module subpanel, select the IP Terminal Global Settings tab.  
The Advertisement/Logo field specifies the message that provides local mode indication.  
Change as required.  
IP telephone codec and jitter settings  
When the IP telephones are operating in local mode, codec and jitter settings are set on a  
phone-by-phone basis. Configure the settings to meet the requirements of the local SRG  
environment. They do not have to be the same as the main office settings (in contrast to the QoS  
To enter codec and jitter settings for IP telephones in local mode  
1
2
In Element Manager, navigate to Configuration > Resources > Telephony Resources.  
On the Modules panel, locate the Module type column and select the IP Sets row (see the  
3
4
On the Details for Module subpanel, select the IP Terminal Global Settings tab.  
The fields related to QoS are on the right side of the panel.  
Enter the appropriate values.  
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56 Chapter 5 IP telephones setup and configuration  
Figure 8 Modules panel, IP & Application Sets  
Telephone (DN) records configuration  
DN records for IP telephones are configured through the All DNs panel (Configuration >  
Telephony > Sets > All DNs (see the figure All DNs panel on page 58). The Networking  
Configuration Guide (NN40020-603) and the Device Configuration Guide (NN40020-300)  
provide basic instructions for configuring DNs and IP telephones. The following instructions are in  
addition to these instructions and only apply to IP telephones that are to be redirected to a main  
office call server.  
Note: It is assumed that the line pools have been assigned. In the default  
configuration, VoIP trunks are assigned to line pool BlocA and the four  
PSTN trunks are assigned to line pool A. For more information, see  
To configure DN records for redirected IP telephones  
1
In Element Manager, navigate to the All DNs panel (Configuration > Telephony > Sets > All  
DNs).  
2
3
4
Select the Line Access tab.  
Identify the row of the DN record to be configured.  
Refer to the list of numbers and phone models recorded in the table (see CS 1000  
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Chapter 5 IP telephones setup and configuration 57  
5
From the Model list, select the model of telephone assigned to this DN.  
6
7
8
To support outgoing caller ID over the VoIP trunk, the Private OLI field (Priv. OLI) must be  
set to the DN.  
To support outgoing number display over the PSTN, enter the public access number for the  
telephone in the Public OLI field (Pub. OLI).  
Leave the Fwd All field blank to disable the Call Forward All Calls feature when the telephone  
is in local mode. In normal mode, the SRG can forward all calls to the main office call server  
Calls feature is automatically discontinued and the SRG routes calls to the SRG DNs.  
Call Forward All Calls does not require Allow redirect to be enabled (Allow redirect  
is found on the Capabilities tab of the Details for DN subpanel when the  
Capabilities and Preferences tab is selected).  
9
To assign specific PSTN lines to each telephone, add the line(s) (Details for DN details panel  
> Line Assignment tab). You would do this if, for example, you want one user to field all  
customer calls when the system is in local mode (see the figure All DNs panel on page 58).  
10 Ensure that the Appearance Type (Details for DN details panel > Line Assignment tab) is  
set to Ring only.  
11 Assign the VoIP and PSTN trunk line pools to the DN (Details for DN details panel > Line  
Pool Access tab).  
12 Assign a target line to the DN (Details for DN details panel > Line Pool Access tab). For a  
description of target lines, see SRG50 terminology on page 20.  
Note: At this point, you may want to configure the received numbers.  
Refer to Received numbers configuration on page 59 and then return to  
this procedure.  
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58 Chapter 5 IP telephones setup and configuration  
Figure 9 All DNs panel  
13 Select the Capabilities and Preferences tab.  
14 In the Intercom keys column, set Intercom Keys to 1 if required.  
15 On the Details for DN subpanel, select the Button Programing Table tab.  
16 Program the voice mail access button with the PSTN dialup for the main office voice message  
system.  
Model  
Button  
2001  
Message  
06  
2002  
2004  
08  
2007  
08  
2033  
08  
2050, 2050CE  
2210  
08  
08  
2211  
08  
2212  
08  
1110  
08  
1120E  
1140E  
08  
08  
NN40140-500  
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Chapter 5 IP telephones setup and configuration 59  
Received numbers configuration  
The Public Received number length and the Private Received number length (see Basic  
parameters on page 67) determine the number of digits that the SRG retains for call processing.  
The retained digits are mapped to the DN using fields provided on the Target Lines panel. (For  
more information on target lines, see SRG50 terminology on page 20).  
Note: If the retained digits are the same as the DN, the fields (Pub.  
Received # field and Priv. Received # field) can be left blank.  
To map received numbers to the DN  
1
In Element Manager, navigate to Configuration > Telephony > Lines > Target Lines (see  
2
3
Select the target line of the DN for which you want to configure the received numbers.  
Double-click the Pub. Received # field, and then enter the retained received digits for calls  
originating from the PSTN.  
4
Double-click the Priv. Received # field, and then enter the retained received digits for calls  
originating from the private network (that is, the VoIP trunks). This number is usually the same  
as the DN.  
Figure 10 Target Lines panel  
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60 Chapter 5 IP telephones setup and configuration  
DHCP settings configuration  
To configure DHCP settings for SRG operation  
1
2
3
4
5
6
In Element Manager, navigate to Configuration > Data Services > DHCP Server.  
Select the General Settings tab.  
From the DHCP Server is list, select Enabled - IP Phones Only.  
From the WINS node type list, select H-node.  
In the Default gateway field, enter an address that meets the requirements of the SRG50 LAN.  
In the Lease time field, enter a value that meets the requirements of your system.  
Leave all other fields under the General Settings tab blank.  
Select the IP Terminal DHCP Options tab.  
7
8
On the Primary Terminal Proxy Server (S1) and Secondary Terminal Proxy Server (S2)  
subpanels:  
a
b
c
In the IP address field, enter the IP address of the SRG.  
From the Port list, select SRG.  
In the Retry count field, enter the number of retries that the IP telephone is allowed to  
connect to the SRG before an event is generated (see IP terminal details on page 84).  
9
Select the Address Ranges tab and add a range of IP address to meet the requirements of your  
system.  
Call forwarding options  
There are two options for configuring call forwarding on the SRG:  
The target DN is determined by the BUID/MODN and VoIP Trunk access codes.  
In this case, the information required for call forwarding is entered using the  
SRG-specific panels of Element Manager (Configuration > Resources > Survivable  
Remote Gateway). For more information, see CS 1000 considerations on page 33.  
Note: Call forwarding is mandatory for CS 1000.  
The target DN is configured explicitly for each IP telephone.  
The DN is configured through Element Manager on the Telephony > Sets > Active  
Sets panel using the standard BCM50 procedure. Typically, the call forward number  
is the BUID. For more details, see the Device Configuration Guide (NN40020-300).  
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Chapter 5 IP telephones setup and configuration 61  
The disadvantage of the second option is that the installer must configure the target  
DN in two places: on the Active Sets panel and on the SRG-specific panels (the  
SRG-specific panels must be completed). The SRG looks at the SRG-specific panels  
first. It goes to the Active Sets panel only if the VOIP Trunk Access Code has not  
been configured.  
Configuration settings for redirected phones  
The Telephony Device Installation Guide and the Networking Configuration Guide provide  
detailed instructions for configuring IP telephones. For IP telephones that are redirected to the  
main office call server, incorporate the settings shown in the table Configuration settings for  
Table 14 Configuration settings for redirected IP telephones  
Parameter  
CS 1000  
S1 IP  
SRG IP address  
S1 Port  
7300  
S1 Action  
S1 Retry Count  
S2 IP  
1
1
SRG IP address  
S2 Port  
7300  
S2 Action  
S2 Retry Count  
1
1
For more information, see CS 1000 considerations on page 33.  
Test Local Mode  
An IP telephone operating in normal mode can be forced to redirect to the SRG. This allows the  
telephone user, and system administrator, to test local mode operation without taking down the  
VoIP trunk to the main office.  
To activate Test Local Mode  
1
2
From the telephone dialpad, press the Services key  
Press the Test Local Mode button.  
.
The telephone is now in Test Local Mode.  
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62 Chapter 5 IP telephones setup and configuration  
To exit Test Local Mode (and return to Normal Mode)  
Generally, you exit Test Local Mode by waiting for the feature to time out or by pressing the Exit  
key (  
). This key is active only when the telephone is in the local mode test.  
For the WLAN Handsets 2210/2211/2212, pressing the End key causes the phone to exit Test  
Local Mode.  
If the phone does not have an Exit key, you must wait until the test times out For more information  
about Test Local Mode Timeout, see S1000 Main Office Settings panel on page 44.  
Features in local mode  
In local mode, IP telephones at the SRG no longer have access to the full suite of main office  
applications. However, the SRG does provide a set of features that include connectivity with the  
local PSTN, access to Emergency Services, and the ability to call local extensions.  
The SRG also supports the following features in local mode:  
Hold  
Transfer (dedicated key on the 2002, 2004, and 2007 models)  
Call Forward No Answer/Busy (if the feature has been enabled on the DN:  
Configuration > Telephony > Sets > All DNs)  
Last Number Redial (dedicated key on the 2002/2004 models)  
Inbox Key (on 2002/2004 models)  
The user experience in local mode can be enhanced if certain global feature settings are  
coordinated with the main office so that the settings are the same at both the main office and the  
SRG. These feature settings are configured with the Feature Settings panel on the Element  
Manager interface (Configuration > Telephony > Global settings > Feature Settings). Feature  
settings that can be coordinated with the main office are:  
Background music (if it is provided for on-hold)  
On-hold  
This determines if a caller on hold hears tones, music, or nothing.  
Receiver volume  
Set to use the system volume, since IP users cannot use the feature code to set a  
default telephone volume.  
Delayed ring transfer  
If a transfer to an external number is not answered, you can indicate if the call will be  
dropped (Off) or transferred to the designated Prime telephone.  
Check the Transfer callback timeout. This setting defaults to After 4 rings. If you  
are using the Delayed ring transfer feature, turn Transfer callback timeout off if  
you want all unanswered transferred calls to ring at the Prime set (usually the system  
attendant). If you want the transferred call to ring at the telephone from which it was  
transferred first, set this field to a setting that is less than the setting for Delayed ring  
transfer.  
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Chapter 5 IP telephones setup and configuration 63  
Held line reminder  
If set to a time, determines period between when a call is put on hold and when a short  
tone sounds at the telephone to indicate the call is still on hold.  
Alarm set  
The alarm set feature is not supported on SRG.  
Language and Contrast  
Language and Contrast are DN-specific settings and are configured at Configuration  
> Telephony > Sets > All DNs > All DNs panel > Capabilities and Preferences tab  
> Details for DN subpanel > Preferences tab.  
Features not supported in local mode include: Hot Desking, Do Not Disturb, Page, Call Forward,  
Background Music, Call Park, Call Pickup, Speed Dial, and Conference.  
911 Emergency Services Support  
For IP telephones in local mode, and for other telephones at the SRG, the Networking  
Configuration Guide (NN40020-603) provides details for configuration of 911 emergency  
services.  
For redirected IP telephones in normal mode, the IP telephone is registered with the main office  
call server. Ensure that the main office call server is configured so that a 911 call from an IP  
telephone at the SRG is routed back to the SRG’s local PSTN. Emergency Services Access (ESA)  
configuration on page 42 includes a procedure for configuring the SRG for CS 1000 Emergency  
Services Access.  
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65  
Chapter 6  
Setting up the private VoIP network  
To provide SRG functionality and to take advantage of VoIP technology, a private VoIP network  
is required between the SRG and the main office. This chapter details the procedures for  
establishing appropriate WAN connections to enable a VoIP network between the main office and  
SRG branch locations. Before proceeding, ensure that IP networking from the SRG to the WAN,  
and from the main office call server to the WAN have been configured and tested (see the figure IP  
establishes the VoIP network (see the figure VoIP path for SRG operation on page 66).  
Figure 11 IP networking, SRG to WAN, main office to WAN  
Main office  
Branch office  
SRG  
IP phones  
IP phones  
IP network connections  
Software  
phones  
WLAN  
handsets  
IP Networking  
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66 Chapter 6 Setting up the private VoIP network  
Figure 12 VoIP path for SRG operation  
Main office  
Branch office  
SRG  
IP phones  
IP phones  
IP network connections  
VoIP trunks  
Software  
phones  
WLAN  
handsets  
IP Networking + VoIP trunks  
Generic procedures for setting up a private network on the SRG are covered in the Networking  
Configuration Guide (NN40020-603). Items to address when establishing the private VoIP  
network between the SRG and the main office are:  
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Chapter 6 Setting up the private VoIP network 67  
Basic parameters  
The table Basic parameters on page 67 provides a record of basic parameters that are significant  
for SRG operation. Typically, these parameters are specified as part of BCM50 foundation  
activities; in most cases, their configuration is not covered in the SRG50 Configuration Guide.  
Table 15 Basic parameters (Sheet 1 of 2)  
Parameter  
Value  
Context  
DN length  
Configured as part of BCM50 foundation configuration.  
There are four DN lengths to consider.  
1. Configuration > Telephony > Dialing Plan > General > Dialing  
Plan - General panel > Global Settings subpanel > DN length  
(intercom) field  
This is the internal DN length. That is, the length of DNs for calls  
between telephones on the SRG.  
2. Configuration > Telephony > Dialing Plan > Public Network >  
Dialing Plan - Public Network panel> Public Network Settings  
subpanel > Public Received number length field  
For calls originating from the PSTN, this establishes the number of  
digits the SRG retains.  
3. Configuration > Telephony > Dialing Plan > Private  
Network > Dialing Plan - Private Network panel > Private  
Network Settings subpanel > Private Received number length field  
For calls originating from a private network, this establishes the  
number of digits the SRG retains.  
4. Configuration > Telephony > Dialing Plan > Private  
Network > Dialing Plan - Private Network panel > Private  
Network Settings subpanel > Private DN length field  
Used for DPNSS applications only. See the Networking  
Configuration Guide (NN40020-603).  
DN range  
Configured as part of BCM50 foundation configuration.  
DN range of the SRG is set by the hardware configuration and  
keycodes. Actual numbering is contiguous from the Start DN:  
Administration > Utilities > Reset > Reset panel > Cold Reset  
Telephony Services button > Cold Reset Telephony dialog box >  
Start DN field  
Destination code  
for VoIP trunks  
Configured for advanced routing. See Outgoing calls configuration  
VoIP destination code = VOIP Trunk Access Code* = AC1**  
* VOIP Trunk Access Code is entered on the main office settings  
panel. See the server-specific chapters for details.  
** For a UDP dialing plan, AC1 is the access code in the digit string  
<AC1> <LOC> <DN>.  
SRG IP address  
Configured as part of BCM50 foundation configuration.  
Configuration > System > IP Subsystem > IP Subsystem panel >  
General Settings tab > IP Settings details subpanel  
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68 Chapter 6 Setting up the private VoIP network  
Table 15 Basic parameters (Sheet 2 of 2)  
Parameter  
Value  
Context  
SRG net mask  
Configured as part of BCM50 foundation configuration.  
Configuration > System > IP Subsystem > IP Subsystem panel >  
General Settings tab > IP Settings details subpanel  
IP address of  
SRG gateway  
Configured as part of BCM50 foundation configuration.  
Configuration > System > IP Subsystem > IP Subsystem panel >  
General Settings tab > IP Settings details subpanel  
VLAN  
If the SRG operates as part of a VLAN, obtain the required  
identifiers from the VLAN administrator.  
Configuration > Data Services > DHCP Server > DHCPServer  
panel > IP Terminal DHCP Options tab > VLAN Identifiers subpanel  
PSTN number for  
dialing into the  
main office from  
the SRG when in  
local mode  
Required to specify a PSTN fallback route. See Outgoing calls  
PSTN number for  
dialing into the  
SRG from the  
main office when  
in local mode  
Required for main office configuration.  
Private dialing plan  
For SRG operation, either a coordinated dialing plan (CDP) or a uniform dialing plan (UDP) can  
be configured. Nortel recommends CDP because it requires the least dialing manipulation between  
the SRG and the main office call server. The dialing plan choice also determines whether the DN  
on the SRG matches the BUID.  
Dialing plans between the SRG and the main office call server must be compatible. Private dialing  
plan configuration is described in detail in the Networking Configuration Guide (NN40020-603).  
The type of dialing plan, CDP or UDP, is determined by the main office configuration.  
The path to the Element Manager panel for setting up the dialing plan is Configuration >  
Telephony > Dialing Plan > Private Network > Dialing Plan - Private Network panel > Private  
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Chapter 6 Setting up the private VoIP network 69  
Figure 13 Dialing Plan — Private Network panel, Private Network Settings subpanel  
Meridian Customer Defined Network (MCDN)  
MCDN is automatically activated when the system is converted to SRG operation. To ensure that  
redirected IP telephones can transfer calls to the SRG local telephones, trunk anti-tromboning  
(TAT) must be enabled.  
To enable MCDN TAT  
Use Element Manager to enable TAT by selecting the TAT checkbox at Configuration >  
Telephony > Dialing Plan > Private Network > Dialing Plan - Private Network panel > MCDN  
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70 Chapter 6 Setting up the private VoIP network  
Figure 14 Dialing Plan — Private Network panel, MCDN subpanel  
QoS settings (codec, jitter buffer, and related items)  
Quality of Service (QoS) settings for the VoIP trunks at the SRG are determined by the main office  
settings; the SRG settings must match the main office. Use the table Main office QoS settings on  
page 70 to record the main office settings, to facilitate configuration, and to provide a record of the  
datafill.  
Table 16 Main office QoS settings (Sheet 1 of 2)  
Media parameter  
H.323 settings  
SIP settings  
Enable Voice Activity Detection  
Codec Preferences  
Silence compression (yes/no)  
Jitter buffer  
--  
G.729 payload size (ms)  
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Chapter 6 Setting up the private VoIP network 71  
Table 16 Main office QoS settings (Sheet 2 of 2)  
Media parameter  
H.323 settings  
SIP settings  
G.723 payload size (ms)  
Supported for IP telephones for local calls only. See IP  
G.711 payload size (ms)  
Incremental payload size (yes/no)  
T.38 fax (yes/no)  
--  
--  
Force G.711 for 3.1k audio (yes/no)  
Fax Transport  
--  
--  
--  
Provide in-band ringback  
In Element Manager, enter QoS settings for VoIP trunks through the Telephony Resources panel.  
To enter the QoS settings for VoIP trunks  
1
2
In Element Manager, navigate to Configuration > Resources > Telephony Resources.  
On the Modules panel, locate the Module type column and select the IP Trunks row (see the  
3
4
On the Details for Module subpanel, select the H323 Media Parameters tab or the SIP  
Media Parameters tab depending on which media parameters you want to configure.  
Refer to the main office QoS settings recorded in the table Main office QoS settings on page  
70 and enter the appropriate values.  
Figure 15 Modules panel, IP Trunks  
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72 Chapter 6 Setting up the private VoIP network  
Network security  
Firewall configuration for SRG is the same as for the BCM50 and is detailed in the Networking  
Configuration Guide (NN40020-603). Firewalls cannot be configured to allow VoIP pass through.  
Instead, the SRG supports IPsec tunnels to provide VoIP pass through. IPsec tunnels are also  
covered in the Networking Configuration Guide (NN40020-603).  
VoIP trunking configuration  
When the SRG is operating in normal mode, connectivity to the main office call server is over  
VoIP trunks. An SRG can support up to 60 VoIP trunks.  
Configuring VoIP trunks has three components:  
Fallback configuration  
For SRG operation, fallback and gatekeeper configuration is common to all VoIP trunks. Fallback  
is determined in conjunction with the QoS settings (see To enter the QoS settings for VoIP trunks  
Note: For calls originating from IP Sets, configure the payload size. Access the IP Sets configuration  
panel in Element Manager (Configuration > Resources > IP Sets > IP Terminal Global Settings.  
Change the Payload Size to 20ms.  
To enable fallback  
1
Access the IP Trunks configuration panel in Element Manager (Configuration > Resources >  
Telephony Resources > Modules panel > Module type column) and then select IP Trunks.  
2
3
On the Details for Module panel, select the H323 Settings tab or the SIP Settings tab.  
On the Telephony Settings subpanel, locate the Fallback to circuit-switched field and use the  
list to:  
a
b
c
select Enabled-All if you want calls to be able to fallback to PSTN trunks if connectivity  
to the main office is lost.  
select Enabled-TDM if you want all TDM calls to be able to fallback to PSTN trunks if  
connectivity to the main office is lost.  
select Disabled if fallback is not required.  
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Chapter 6 Setting up the private VoIP network 73  
Gatekeeper routing  
The gatekeeper routes the calls based on an internal numbering table. Ensure that the gatekeeper  
administrator has a list of the numbers that identify the SRG and the SRG PSTN.  
Examples:  
If the system is running with a CDP dialing plan and the SRG DN range is from 3000  
to 3199, the gatekeeper requires this information to establish that calls starting with 30  
and 31 are routed to the SRG.  
If the PSTN to which the SRG connects has a location code of 521, the gatekeeper  
must have a record of this code in the SRG list so that main office calls to the SRG  
PSTN can be routed correctly.  
If you require H323 trunks, you must select the Gatekeeper wildcard check box in the H323  
settings. If you do not select the Gatekeeper wildcard check box, then calls are SIP routed (default  
setting).  
When you select the Gatekeeper wildcard check box, all dialed digits match gatekeeper digits and  
VoIP calls are routed through the gatekeeper.  
For detailed information about H323 gatekeeper and SIP proxy, see the Networking Configuration  
Guide (NN40020-603).  
To configure gatekeeper settings for H323 trunks  
1
Access the IP Trunks configuration panel in Element Manager (Configuration > Resources >  
Telephony Resources > Modules panel > Module type column) select IP Trunks.  
2
3
4
5
On the Details for Module panel, select the H323 Settings tab.  
On the Configuration subpanel, click Modify.  
From the Call signaling list, select Gatekeeper Resolved or Gatekeeper Routed.  
In the Primary Gatekeeper IP field, enter the IP address for the Primary Network Routing  
Services Address*  
6
7
In the Backup Gatekeeper(s) field, enter the IP address for Alternate Network Routing  
Services Address  
In the Alias names field, enter Name: followed by the H.323 ID of the SRG (for naming  
conventions, see the Networking Configuration Guide (NN40020-603)).  
8
9
Click Ok.  
On the Telephony Setting subpanel, from the MCDN protocol list, select CSE.  
10 On the Telephony Settings subpanel, select the Gatekeeper wildcard check box.  
SIP configuration  
Access the IP Trunks configuration panel in Element Manager to configure the SIP trunks  
(Configuration > Resources > Telephony Resources > Modules panel > Module type column > IP  
Trunks, then select the SIP Settings tab).  
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74 Chapter 6 Setting up the private VoIP network  
When you set the value of the RFC2833 Dynamic Payload field, if the value is greater than 0, then  
this feature is enabled. If the feature is enabled then DTMF digits are sent in RTP packets. For  
CS1000 Release 4.5 set the value to 0.  
For detailed information on entering the values in this panel, see the Networking Configuration  
Guide (NN40020-603).  
Line pools  
Both VoIP trunks and PSTN trunks must be configured in separate line pools. In the default state,  
all VoIP trunks are assigned to line pool BlocA and all PSTN trunks are assigned to line pool A. It  
is not necessary to reassign the line pools.  
Instructions for configuring line pools is provided in the Networking Configuration Guide  
(NN40020-603).  
Call routing  
Call routing is covered in depth in the Networking Configuration Guide. The instructions in the  
SRG50 Configuration Guide are an abbreviation of that material, and only cover procedures that  
are specific to SRG operations; that is, for calls from redirected IP telephones. For more detailed  
information, see the Networking Configuration Guide.  
Note: The DNs for the main office telephones system are marked off by the vacant  
number routing feature. SRG does not support Vacant Number Routing (VNR).  
Instead, SRG uses Call Forward All Calls to emulate VNR for the SRG IP telephones that  
are in normal mode. When the telephones switch to local mode, Call Forward All Calls is  
cancelled for those telephones.  
A single destination code and route (or a group of destination codes and routes) can be set  
up on the SRG to route all the calls that are not terminated locally by the SRG. These calls  
are routed over the VOIP trunks. In the case where the VoIP trunks become unavailable, the  
calls can be routed to the proper location using PSTN fallback. This is similar to the VNR  
feature in CS 1000.  
Outgoing calls configuration  
To configure routing for outgoing calls  
1
Create a schedule.  
a
b
Access the Scheduled Services panel (Configuration > Telephony > Scheduled Services).  
Select a Schedule (Sched 4, for example).  
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Chapter 6 Setting up the private VoIP network 75  
c
Change the schedule name (optional).  
In this procedure, the name SRG is used as the name of the schedule.  
d
e
f
In the Services subpanel, select your schedule.  
In the Routing Svc row for your schedule, select Auto from the list.  
Select the Overflow checkbox for your schedule.  
g
Change the schedule time so that the schedule runs continuously (Start Time 00:00:00,  
Stop Time 23:59:59, MTWTFSS).  
2
Define a route for calls to the main office over the VoIP trunks and a route to the main office  
over the PSTN.  
a
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing).  
b
c
d
e
f
Select the Routes tab.  
Click Add to add a new route.  
In the Route field, enter a number for the new route (for example, 998).  
Click Ok.  
Ensure that the DN Type is Public (Unknown).  
g
Assign the VoIP line pool to the route (select the line pool from the Use Pool list; default is  
A).  
h
i
Click Add to add another new route.  
In the Route field, enter a number for the new route (for example, 999).  
Click Ok.  
j
k
l
Ensure that the DN Type is Public (Unknown).  
In the External Number field, enter the PSTN number of the main office.  
m Assign the PSTN line pool to the route (select the line pool from the Use Pool list; default  
is A).  
3
Add a destination code to provide access to the newly created routes. This code is used in both  
normal and local modes for dialing the main office from the SRG site.  
a
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing).  
b
c
d
e
f
Select the Destination Codes tab.  
Click Add to add a new destination code.  
In the Destination Code field, enter the new destination code (for example, 678).  
Click Ok.  
With the Destination Code row highlighted, select the SRG schedule from the Alternate  
Routes list (Alternate Routes for Destination Code details panel).  
g
In the First Route field, enter 998 (the VoIP route).  
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h
In the adjacent Absorbed Length field, select the number of digits to be absorbed. The  
Absorbed Length applies to the digits of the destination code only.  
i
j
In the Second Route field, enter 999 (the PSTN route).  
In the adjacent Absorbed Length field, select the number of digits to be absorbed.  
Depending on the dialing plan, the destination code is integrated with the DN or is dialed as a  
prefix to the DN. When a user calls the main office, the SRG examines the destination code to  
determine the routing. If the First Route, the VoIP trunks, is unavailable, the call is routed to the  
Second Route, the PSTN, and the External Number is called. Because Overflow was selected, if  
both the First Route and the Second Route are unavailable, the call is routed using the Normal  
Route specified in the Normal Route column of the Destination Codes table. Because Auto was  
selected, the routing occurs without manual intervention.  
SRG PSTN access  
Access to the SRG PSTN is required for:  
calls to the SRG PSTN from SRG telephones or redirected SRG IP telephones in local  
mode  
calls from the SRG PSTN to redirected SRG IP telephones in normal mode  
calls from main office telephones to the SRG PSTN, using the VoIP trunks  
To achieve this access, a remote access package for the VoIP trunks and a destination code for the  
PSTN must be configured.  
Remote Access Package for VoIP trunks  
The SRG views all calls coming in over the VoIP trunks as remote access calls, even though the  
VoIP pathway is a dedicated trunk to another closed system.  
To allow tandem dialing from the main office through the SRG PSTN, or to redirect SRG IP  
telephones to use the SRG local PSTN, you must specify a remote package that provides access to  
the PSTN line pool. This remote package is then assigned to each VoIP trunk.  
To configure remote access packages  
1
LocateSet up a remote access package for the PSTN line pool (Configuration > Telephony >  
Call Security > Remote Access Packages).  
2
Assign the package to each VoIP trunk (Configuration > Telephony > Lines > Active VoIP  
Lines > Trunk Type column > Details for Line panel > Restrictions tab > Use remote package  
field).  
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Chapter 6 Setting up the private VoIP network 77  
PSTN destination codes configuration  
To allow SRG telephones to dial out over the PSTN and to allow main office telephones to tandem  
out through the local SRG PSTN, you need to define a destination code that accesses the PSTN  
line pool without an External Number. Frequently, this code is 9, but it does not have to be.  
The following procedure provides a basic PSTN routing setup.  
To configure destination codes for the PSTN  
1
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing).  
2
Select the Routes tab. See the Networking Configuration Guide (NN40020-603) to:  
a
b
c
d
Add a new route (for the PSTN line pool).  
From the DN Type list, select Public.  
Leave the External Number field blank.  
Assign the PSTN line pool to the route (select the line pool from the Use Pool list).  
3
4
Access the Dialing Plan - Routing panel (Configuration > Telephony > Dialing Plan >  
Routing).  
Select the Destination Codes tab. See the Networking Configuration Guide (NN40020-603)  
to:  
a
Add the destination code to be used to access the local (SRG) PSTN.  
Users on both SRG and main office telephones dial this destination code to access the  
local (SRG) PSTN. If this code goes only to the SRG PSTN, enter 9 + Wild Card 1. This  
wild card allows any numbers not used by other 9-based destination codes.  
The default line pool access code for pool A is 9. Delete this access code  
before you attempt to create a destination code with 9.  
In normal mode, the destination code is forwarded from the main office to the SRG for  
SRG IP telephone calls that are connecting to the SRG PSTN.  
Note: For main office programming, this code is the offnet dialing code  
that the gatekeeper recognizes for routing to the SRG.  
At the main office, zone-based digit manipulation is used to add a Zone  
Digit Prefix (ZDP) to PSTN calls from SRG IP telephones. The ZDP  
allows the main office to differentiate between local PSTN calls made  
from main office telephones (to the main office PSTN) and PSTN calls  
made from SRG IP telephones (to the SRG PSTN). The main office  
administrator f supplies this ZDP with the prerequisite information.  
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b
Assign the Normal and SRG scheduled route for the two destination codes.  
Main office information  
The SRG requires information about the main office call server that is not needed for a BCM50.  
Element Manager accommodates this information with SRG-specific panels that are activated  
after the SRG50 keycode is applied. The information required for these panels is specific to the  
main office call server CS 1000 considerations on page 33.  
External attendant support  
The SRG can use the BCM50 Selective Line Redirection capability to provide an external  
attendant. If the attendant is located in the main office, there are two ways to maintain the  
attendant if the VoIP trunks become unavailable:  
1
2
A fallback (or Prime) DN at the SRG can be specified. Since this DN is likely to receive calls  
in a WAN failure scenario, it must be an IP telephone that can transfer the calls to the desired  
party. If the IP set is also a redirected IP set, there is a period of time where inbound calls are  
un-routable, until the IP set falls back to the SRG.  
A fallback route to the main office call server over the PSTN can be configured. At the main  
office, vacant number handling (such as routing to voice mail) can be applied.  
Note: The SRG does not have local attendant capability.  
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Chapter 7  
PSTN access and analog devices  
SRG-specific items relevant to PSTN trunks and analog devices include:  
PSTN access considerations  
Consider the following for PSTN access:  
PSTN access  
To provide access to the SRG PSTN when the SRG is in local mode, or to be able to  
set up tandem dialing from the main office through the SRG to the SRG PSTN, one or  
more PSTN trunks must be configured on the system. See the Networking  
Configuration Guide (NN40020-603).  
Tandem calls  
The SRG considers all calls coming in over the VoIP trunks as remote access calls,  
even though the VoIP pathway is a dedicated trunk to another closed system.  
To allow tandem dialing from the main office to the SRG PSTN, or to allow redirected  
SRG IP telephones to use the SRG local PSTN, a remote access package must be  
specified to provide access to the PSTN line pool. This procedure is covered in  
Manual- and auto-answer lines  
If the trunk is configured as a manual-answer line:  
Enter the line pool access code and the dial string for the main office attendant  
telephone in the Redirect to field (Configuration > Telephony > Lines > Active  
Physical Lines).  
If the line is an auto-answer line in normal mode, incoming call requests are  
automatically call forwarded to the main office. When the SRG IP telephones revert to  
local mode, the system discontinues Call Forward All Calls and calls are delivered  
directly to the SRG IP telephones at the SRG.  
Analog devices considerations  
Consider the following for analog devices:  
Basic operation  
Analog telephones and devices connected to the SRG always function as local  
telephones to the SRG. They can use the VoIP trunk to the main office using access  
codes or destination codes, if the VoIP trunk line pool is assigned to the device, but the  
main office does not have any settings or administration for these devices.  
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Access to system features  
Analog telephones do not have a Feature key. Instead, they use a Link (*) key to  
access system features. If you leave the analog telephone records at the default  
settings, these telephones have greater feature access on the SRG than the  
IP telephones in local mode. If you do not want different feature access on the analog  
telephones, turn the unwanted settings off as you program the telephone.  
To configure the DNs for analog devices  
1
In Element Manager, navigate to the Active Sets panel (Configuration > Telephony > Sets >  
Active Sets). See the figure Active Sets panel on page 80.  
2
Select the Line Access tab.  
Figure 16 Active Sets panel  
3
4
5
Identify the DNs for which the Model is Analog and align the settings for each with the  
information in the following steps.  
To support outgoing number display over the PSTN, enter the public access number for the  
telephone in the Public OLI field (Pub. OLI).  
To assign specific PSTN lines to each telephone, add the line(s) (Details for DN panel > Line  
Assignment tab). You would do this if, for example, you want one user to field all customer  
calls when the system is in local mode.  
6
Ensure that the Appearance Type (Details for DN panel > Line Assignment tab) is set to Ring  
only.  
7
8
Assign the target line to the telephone (Details for DN panel > Line Pool Access tab).  
Assign both the PSTN and VoIP trunk line pools to all telephones that are allowed to make  
calls over the PSTN or to the main office over the VoIP trunk.  
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Chapter 7 PSTN access and analog devices 81  
If you want the analog telephones to emulate local mode call functionality always, assign only  
the PSTN line pool to the analog devices.  
9
Select the Capabilities and Preferences tab.  
10 From the Details for DNs panel, select the Capabilities tab.  
11 From the Handsfree list, select None.  
12 Clear the HF Answerback check box.  
13 Clear the Paging check box.  
14 Select the Allow Redirect check box if you want the user to be able to call forward to the main  
office or redirect lines to the main office.  
15 From the Details for DNs panel, select the ATA Settings tab.  
16 From the ATA device list, select Telephone.  
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Chapter 8  
Troubleshooting  
Potential problems, probable causes, and suggested solutions for SRG-specific configuration and  
operating troubles are categorized under the following topics:  
IP telephone troubleshooting  
The table IP telephone troubleshooting on page 83 provides issues and solutions for IP Phone  
troubleshooting.  
Table 17 IP telephone troubleshooting (Sheet 1 of 2)  
Issue / Problem  
Probable Cause / Solution  
Telephone does not connect to  
system  
If an IP telephone does not display the text Connecting to server  
within two minutes after power up, the telephone was unable to establish  
communications with the SRG. Double check the IP configuration of the  
telephone and the IP connectivity to the SRG (cables, switches, and so on).  
Slow connection between the  
handset and the Business  
Communications Manager  
If the connection between the IP client and the SRG is slow (ISDN, dialup  
modem), change the preferred codec for the telephone from G.711 to G.729.  
Block individual IP sets from dialing If you want to block one or more IP telephones from calling outside the system,  
outside the system.  
use Restriction filters and assign them to the telephones you want to block.  
Restriction filters are set up under Configuration > Telephony > Call Security >  
Restriction Filters.  
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84 Chapter 8 Troubleshooting  
Table 17 IP telephone troubleshooting (Sheet 2 of 2)  
Issue / Problem  
Probable Cause / Solution  
One-way or no speech paths  
Signaling between the IP telephones and the SRG uses UDP port 7300.  
Voice packets are exchanged using the default RTP:  
Source port (output filters)/Destination port (input filters): 28000 through 28511  
for the VoIP gateway. Output filter Destination IP is set to ALL. Input filter  
Destination IP is the IP address of the SRG local gateway.  
Source port (output filters)/Destination port (input filters): 5200 - 5201 for the IP  
telephones. Output Destination IP is set to ALL. Input filter Destination IPs are  
the IP range for all IP telephones (behind the firewall) in Normal mode.  
Source port (output filters)/Destination port (input filters): 51000 - 51184 for the  
local mode IP sets. Destination port (output filter) and Source Port (input filter)  
are set to ALL. Output Destination IP is set to ALL. Input filter Destination IPs  
are the IP range for all IP telephones (behind the firewall) in Local mode.  
UniSTIM signals use specific source and destination ports:  
Output filters: Source port, 5000; Destination port, 4100, 5100, 7300. Output  
filter Destination IP is the IP address of the main office TPS.  
Input filters: Source port, 4100, 5100, 7300; Destination port, 5000.Input filter  
Destination IPs are the IP range for all IP telephones (behind the firewall) in  
Normal mode.  
If these ports are blocked by the firewall or NAT, you will experience one-way or  
no-way speech paths.  
Firewall note: If the firewall filter is set to Pass Outgoing and Block Incoming  
Except IP Phones, this allows only IP telephony registration traffic through, but  
blocks all other traffic, including H.323 calls on this interface. You must still  
specify an H.323 rule to allow IP call voice traffic.  
IP terminal details  
The table IP Terminal Details on page 84 summarizes the events that can be raised by the SRG.  
The events and details appear in Element Manager at Configuration > Telephony > Sets > Active  
Sets > IP Terminal Details.  
Periodic retries may result in the same condition being detected over and over again. In these cases  
the SRG state machine uses flags to indicate that a given event has been logged.  
Table 18 IP Terminal Details (Sheet 1 of 2)  
Call Server  
Details  
Event Id  
Severity  
Type  
Comments  
SRG Started  
57000  
57001  
Warning  
Warning  
Warning  
Warning  
All  
Indicates that the SRG process has started.  
Indicates that the SRG process has shut down.  
Test Feature  
SRG Shutdown  
All  
DN:XXX, Test Local Mode 57002  
All  
DN:XXX, Local Mode -  
Firmware is out of sync  
with Main Office Call  
Server.  
57003  
S1000  
Indicates that IP set FW on main office has  
been upgraded and the required FW version is  
available on the SRG  
DN:XXX, Local Mode - Set 57004  
Firmware Upgrade in  
Progress  
Warning  
S1000  
The firmware required by the main office is  
being upgraded to the set,  
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Chapter 8 Troubleshooting 85  
Table 18 IP Terminal Details (Sheet 2 of 2)  
Call Server  
Type  
Details  
Event Id  
Severity  
Comments  
DN:XXX, Normal Mode -  
Set Redirected to Main  
Office  
57005  
Warning  
All  
The set has been redirected to the main office.  
DN:XXX, Local Mode -  
Redirection Pending (Set  
on call)  
57006  
57007  
57008  
Warning  
Warning  
Warning  
Minor  
All  
The redirection of the set is pending as the set  
is on a call.  
DN:XXX, Local Mode -  
Firmware Upgrade  
Pending (Set on call)  
S1000  
All  
The firmware upgrade to the set is pending as  
the set is on a call.  
DN:XXX, Local Mode -  
Main Office Parameters  
Not Provisioned.  
The set is not provisioned to be redirected.  
Indicates configuration problem.  
Indicates configuration problem.  
Indicates configuration problem.  
DN:XXX, Invalid ID (1) - No 57250  
endpoint in Gatekeeper  
database  
S1000  
S1000  
S1000  
DN:XXX, Invalid ID (2) - ID 57251  
unknown within the Call  
Server  
Minor  
DN:XXX, Invalid ID (3) -  
Endpoint in Gatekeeper  
database is Originating  
Call Server  
57252  
Minor  
DN:XXX, Local Mode - Net 57253  
Connect Server  
Unreachable  
Major  
Major  
Major  
Critical  
S1000  
All  
Indicates either a configuration error, or a  
network connectivity error or the Net connect  
server has stopped.  
DN:XXX, Local Mode -  
Main Office TPS  
Unreachable  
57500  
57501  
57750  
Indicates either a configuration error, or a  
network connectivity error, or the MO TPS has  
stopped.  
DN:XXX, Local Mode -  
Firmware is not available  
on the SRG  
S1000  
All  
Indicates firmware required by the main office  
is not available in the SRG.  
SRG terminated  
unexpectedly.  
Indicates that a critical error caused the SRG  
process to terminate.  
Probable causes for redirection failure  
The IP telephone registration to the main office call server can fail due to improper configuration  
or lack of WAN connectivity. When a registration failure occurs, the error code and description is  
shown in the status field for the IP telephone in the IP Terminal Details field (see IP terminal  
details on page 84); the IP telephone remains registered with the SRG in local mode operation.  
Definitive causes for registration failure depend on the main office call server. These causes can  
include:  
The main office is unreachable.  
There is no endpoint configured for the user id or branch user id / TN combination.  
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The actual IP telephone set type at the SRG does not match MOTN set type at the  
main office.  
The user id is registered and not idle.  
The user id entry in the gatekeeper database points back to the originating node.  
Troubleshooting fallback to local mode  
If the system reverts to local mode and the problem is not the WAN link to the main office, check  
for:  
1
IP telephone firmware discrepancies  
The SRG supports automatic firmware updates (see CS 1000 considerations on page 33).  
However, the possibility exists that a non-network reversion to local mode is caused when the  
IP telephone firmware has been updated on the main office and not on the SRG.  
Check the IP Terminal Details tab (see IP terminal details on page 84) for this Status:  
Firmware is Out of Sync with the Main Office Call Server  
The preferred way of handling firmware upgrades is to install the patch onto the SRG first,  
then on the main office equipment.  
When the IP telephone firmware is updated on the main office, the main office redirects all  
SRG IP telephones back to the SRG for a firmware upgrade. If the SRG has already been  
patched with the new firmware, the telephone is upgraded when it registers with the SRG.  
Once the telephone has the new firmware, the system automatically allows the telephone to  
reregister with the main office. If the correct firmware cannot be applied, for example because  
the SRG has not been upgraded with the new firmware, the telephone is redirected back to the  
main office.  
2
Gatekeeper failure  
If an IP telephone fails to establish communication with the gatekeeper when it tries to register  
to the main office, the telephone remains registered to the SRG and stays in local mode.  
Troubleshoot the problem by checking the settings made when implementing the CS 1000  
changes, manually redirect the telephones (IP telephones manual redirection on page 86,  
below).  
IP telephones manual redirection  
To manually redirect an IP telephone to the main office  
1
2
Access the Configuration > Telephony > Sets > Active Sets > IP Terminal Details.  
Click the telephone listing that you want to redirect to normal mode.  
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Chapter 8 Troubleshooting 87  
3
Click the Status tab to view the Status field. If Status displays Up, the conversion was  
successful.  
a
If the IP terminal does not register correctly with the main office, refer back to the IP  
Terminal Status tab, Status field and review the message to determine where the problem  
b
If the conversion occurred correctly, perform basic telephony tests to ensure that the  
telephones are working as expected:  
— Make and receive calls.  
— Check feature access.  
— Check voice mail access  
For specific information about making calls and using features, see the feature guides for the  
main office application.  
Cannot clear alarms using an alarm set  
If you specify an alarm set (telephone) on the SRG through Element Manager, you cannot clear  
alarms using the alarm set. When the alarm set is in local mode, the soft keys are disabled and  
therefore alarms cannot be cleared.  
The Alarm Set feature is not supported on SRG.  
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Appendix A  
Telephone features in normal and local mode  
The information provided here is designed for distribution to telephone users at the SRG.  
The SRG50 supports the following:  
IP Phones 110, 1120E, and 1140E  
IP Phones 2001, 2002, 2004, 2007, and 2033  
IP Phone Key Expansion Module (KEM) – The IP Phone KEM is supported on an  
SRG with normal mode IP Phones. It does not function with local mode or test local  
mode IP Phones.  
IP Softphone 2050 v1/v2  
Mobile Voice Client (MVC) 2050  
WLAN Handsets 2210, 2211, and 2212  
WLAN Handsets 6120 and 6140  
WLAN Handsets 1210, 1220, and 1230  
Analog (500/2500 type) telephones  
Analog devices such as fax machines  
Normal mode  
In normal mode, IP telephones have the feature set that is supported by the main office. User cards  
are not supplied with the SRG because the feature set depends on the main office applications. If  
required, obtain user cards from the main office for normal mode features.  
Features available to analog and ISDN telephones are provided by the SRG and depend on the  
SRG applications. Consult the SRG system administrator for a complete description.  
A quick reference list to the default SRG features for an analog telephone are provided in ATA  
extension features on page 95. Consult the SRG administrator to determine if these features have  
been changed.  
Local mode  
In local mode, call control and features are provided by the SRG are processed by the SRG. Access  
to the main office is over PSTN lines; main office telephony features and applications are not  
available.  
If routing and destination codes are set up as suggested in Call routing on page 74, the dialing  
sequence for the main office is the same as in normal mode.  
For illustrations that show the default display settings for each type of IP telephone when the  
phone is in local mode, refer to:  
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Telephone features in normal and local mode  
IP Phone 1110 in Local mode  
Autodials voice mail through  
Activate Test in Local  
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Telephone features in normal and local mode 91  
IP Phone 1120E in Local mode  
Intercom/Line  
Local Mode  
Autodials voice mail through  
Activate Test in Local  
Exit/Quit key: only active  
if MO set to Test in local  
IP Phone 1140E in Local mode  
Intercom/Line  
Local Mode  
Autodials voice mail through PSTN  
Activate Test in Local mode  
Exit/Quit key: only active if MO set to  
Test in local mode  
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Telephone features in normal and local mode  
IP Phone 2001 in Local mode  
DN: 345  
Aug 6 9:43 am  
Local mode  
Programmable label  
Activate Test in Local  
Autodials voice mail through  
IP Phone 2002 in Local mode  
Transfer  
Lastnum  
345  
Intercom/Line  
Aug 6 9:43 am  
Local mode  
Programmable label  
Exit/Quit key: only active  
if MO set to Test in local  
mode  
Activate Test in Local  
mode  
Autodials voice mail through PSTN  
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Telephone features in normal and local mode 93  
IP Phone 2004 in Local mode  
Not active  
Transfer  
Lastnum  
Intercom/Line  
345  
Aug 6 9:43  
Local Mode  
Programmable label  
Activate Test in Local  
mode  
Exit/Quit key: only active if  
set to Test in local mode  
Autodials voice mail through PSTN  
IP Phone 2007 in Local mode  
Intercom/Line  
Local Mode  
Programmable label  
Autodials voice mail through  
Activate Test in Local  
Exit/Quit key: only active  
if MO set to Test in local  
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Telephone features in normal and local mode  
IP Phone 2033 in Local mode  
Autodials voice mail through  
Local Mode  
Activate Test in Local  
Programmable label  
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Telephone features in normal and local mode 95  
IP 2050 Softphone in Local Mode  
Exit/Quit key:  
only active if  
MO set to Test  
in local mode  
Programmable label  
Not active  
Intercom/Line  
Activate Test in local  
mode  
Autodials voice mail through PSTN  
ATA extension features  
Analog telephones can be connected to the system through analog station modules or by installing  
an Analog Terminal Adapter (ATA) between the telephone and a digital station module. These  
telephones have only basic button configurations, so instead of using the feature key, press the link  
key (*) to invoke features on the system. For specific key sequences, see the table Link key  
Table 19 Link key sequences (Sheet 1 of 2)  
Feature  
Activate  
Cancel  
Feature  
Activate  
Cancel  
Alternate line  
LINK 2  
Privacy control  
Link  
LINK *83  
LINK *71  
Call Forward (local system)  
LINK *4  
LINK #4  
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96  
Telephone features in normal and local mode  
Table 19 Link key sequences (Sheet 2 of 2)  
Feature  
Activate  
Cancel  
Feature  
Activate  
Cancel  
Call Forward  
(external system)  
LINK *4  
<dialed #>  
LINK 2  
LINK #4  
Pause  
LINK *78  
LINK *72  
LINK *2  
Timed release  
Ring Again  
LINK #2  
Call parking  
LINK *74  
LINK *76  
LINK *75  
Saved Number  
Redial  
LINK *67  
Call pick-up (Directed)  
Call pick-up (Group)  
Tones  
LINK *809  
LINK  
#809  
Call Queuing  
LINK *801  
LINK *82  
LINK *3  
LINK *79  
LINK 2  
Transfer  
LINK *70  
LINK *800  
LINK *66  
Camp-on  
Trunk Answer  
Voice Call  
Conference call  
Hold Call (Exclusive)  
Hold Call (Public)  
Last Number Redial  
Page - Intercom  
Voice messaging - Internal  
LINK *5  
Access mailbox  
LINK *981  
LINK *980  
LINK *61 and  
zone (0 to 6)  
Leave a message  
Page - External  
Page - All  
LINK *62  
LINK *63 and  
zone (0 to 6)  
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97  
Glossary  
BDP  
Both Dialing Plans.  
A dialing plan option that is supported on the main office only. The SRG supports  
CDP or UDP only. If the main office is running BDP, the SRG zone must be  
configured to allow either CDP or UDP, not both.  
branch office  
A system that is remote from the main office but provides telephony services using  
the main office servers. When a branch office is a survivable remote gateway,  
telephony services are provided by the branch office if communication with the main  
office is lost.  
call routing  
CDP  
Coding that is configured on a system to ensure that outgoing calls are directed to the  
correct trunks and incoming calls are directed to the correct device(s) on the system.  
(see also: dialing plan)  
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, consult the main  
office documentation that covers dialing plans.  
dialing plan  
FXO  
Each system uses a specific numbering configuration (dialing plan) that determines  
how calls will be handled over a private or public network. (see also: call routing)  
Foreign eXchange Office: an interface that connects to the PSTN central office and is  
the interface offered on a standard telephone. Example: RJ-11 connector that allows  
analog connection to the central office.  
gatekeeper  
The gatekeeper is an IP network application that directs IP traffic to all the systems on  
the network. Parameters for both the main office and the SRG must be assigned to all  
gatekeepers 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 also down, or  
there is no alternate gatekeeper, the SRG IP telephones remain registered with the  
main office, but calls cannot be sent to the SRG.  
gateway  
H.323  
The IP portal on each system that establishes the VoIP trunk.  
An IP gateway protocol used by both the main office and the SRG to create VoIP  
trunking connections.  
IP  
Internet Protocol  
IP specifies the format of packets, also called datagrams, and the addressing scheme  
in the TCP/IP protocol suite. Where IP defines the packet and addressing scheme,  
Transport Control Protocol (TCP) establishes a virtual connection between a  
destination and a source.  
IP telephones  
local mode  
Telephones that can connect directly with a TCP/IP network. Also known as internet  
telephones.  
The operating mode of redirected SRG IP telephones when connectivity with the  
main office is unavailable.  
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Glossary  
main office, main  
office call server  
The system that provides telephony services to redirected SRG IP telephones in  
normal mode.  
NCS  
Network Connection Server  
The NCS is an H.323 gatekeeper. It provides standard H.323 gatekeeper functionality,  
as well as support for branch office and virtual office features.  
normal mode  
QoS  
The operating mode of the SRG when connectivity with the main office is  
established.  
Quality of Service  
In IP telephony, QoS refers to the quality of the voice communication channel. There  
are inherent difficulties associated with transmitting voice over IP and quality of  
service is a significant challenge for service providers. QoS specifications allow  
service providers and their customers to establish and monitor acceptable levels of  
service.  
SIP  
An IP gateway protocol used by both the main office and the SRG to create VoIP  
trunking connections.  
steering codes  
TPS  
Steering codes are similar to line pool access codes and destination codes. Steering  
codes determine where a call gets routed.  
(Internet Telephone) Terminal Proxy Server  
A TPS controls the connection between IP telephones.  
Uniform Dialing Plan  
UDP  
Each location within the network is assigned a Location Code. On a private network,  
this code precedes the directory number of the telephone being dialed. Depending on  
routing configuration, this number may be part of the destination code, or it may  
automatically be dialed out when the appropriate destination code is dialed before the  
directory number. Under the Uniform Dialing Plan (UDP), the SRG must include this  
code in the BUID.  
UDP  
User Datagram Protocol  
A member of the TCP/IP protocol suite that transports data as a connectionless  
protocol, using packet switching. Generally, ports on the SRG support UDP.  
VoIP trunk  
WAN  
Voice over IP trunk  
A pathway between two systems that allows voice to be transmitted over an IP  
connection.  
Wide Area Network  
A computer network that spans a relatively large geographical area. The largest WAN  
in existence is the Internet.  
ZDP  
Zone Digit Prefix  
The main office appends this number to an SRG local-PSTN call dialed from an SRG  
IP telephone. The number differentiates the call from a main office local-PSTN call  
dialed by a main office telephone.  
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99  
Index  
A
G
gatekeeper  
H
B
hardware  
I
IP telephones  
C
call routing  
IP telephony  
CDP  
K
D
L
dialing plan  
line pools  
E
lines  
F
local mode  
fallback  
features  
M
firewall  
main office  
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100  
MCDN  
N
networking  
Normal mode  
normal mode  
T
O
TAT  
outside access  
telephones  
overview  
R
received numbers  
troubleshooting  
S
U
UDP  
SRG  
V
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101  
VoIP trunks  
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NN40140-500  
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