All rights reserved.
The information in this document is subject to change without notice. The statements, configurations, technical data, and
recommendations in this document are believed to be accurate and reliable, but are presented without express or implied
warranty. Users must take full responsibility for their applications of any products specified in this document. The
information in this document is proprietary to Nortel Networks NA Inc.
Trademarks
NORTEL NETWORKS, Norstar and Meridian are trademarks of Nortel Networks.
Microsoft, MS, MS-DOS, Windows, and Windows NT are registered trademarks of Microsoft Corporation.
All other trademarks and registered trademarks are the property of their respective owners.
Configuring number manipulation for Source Phone Numbers for IP to Norstar calls . . 52
Configuring number manipulation for Source Phone Numbers for Norstar to IP calls . . 55
The Norstar VoIP Gateway is an accessory for Norstar KSU that provides up to four IP telephony
trunks. These IP telephony trunks allow you to establish voice calls to other IP telephony enabled
telephones systems using a data networking connection. Examples of IP telephony enabled
telephone systems are a Business Communications Manager system, Meridian 1 IPT or another
Norstar KSU equipped with a Norstar VoIP Gateway.
Before you begin
This guide provides information about the configuration and operation of the Norstar VoIP
Gateway. It is intended for persons responsible for the configuration of a Norstar VoIP Gateway
system. Prior knowledge of IP networks is required.
Before using this guide, the Norstar VoIP Gateway must be connected to the Norstar KSU.
This guide assumes:
•You have planned the telephony and data requirements for your Norstar system.
•The Norstar KSU is installed and initialized, and the hardware is working. External lines and
internal telephones and telephony equipment are connected to the KSU or Expansion
Modules.
•Configuration of lines is complete.
•Operators have a working knowledge of the Windows operating system and of graphical user
interfaces.
•Operators who manage the data portion of the system are familiar with network management
and applications.
13
Symbols used in this guide
This guide uses these symbols to draw your attention to important information:
Caution: Caution Symbol
Alerts you to conditions where you can damage the equipment.
Danger: Electrical Shock Hazard Symbol
Alerts you to conditions where you can get an electrical shock.
Warning: Warning Symbol
Alerts you to conditions where you can cause the system to fail or work improperly.
Norstar VoIP Gateway Configuration Guide
Page 14
14 Text conventions
Note: Note/Tip symbol
Alerts you to important information.
Tip: Note/Tip symbol
Alerts you to additional information that can help you perform a task.
Text conventions
This guide uses these following text conventions:
angle brackets (< >)Represent the text you enter based on the description inside the
brackets. Do not type the brackets when entering the command.
Example: If the command syntax is
ping
<ip_address>
, you enter: ping 192.32.10.12
bold Courier text
italic textRepresents terms, book titles and variables in command syntax
bold textRepresents fields names, field entries, and screen names in the
plain Courier
text
dollar sign ($)
Acronyms
This guide uses the following acronyms:
Represent command names, options and text that you need to enter.
Example: Use the
Example: Enter
dinfo command.
show ip {alerts|routes}.
descriptions. If a variable is two or more words, the words are
connected by an underscore.
Example: The command syntax
show at
valid_route
<valid_route>
,
is one variable and you substitute one value for it.
configuration application.
Represents command syntax and system output, such as prompts and
system messages.
Example:
The $ symbol indicates hexadecimal notation.
Set Trap Monitor Filters
ATMAsynchronous Transfer Mode
CDPCoordinated Dialing Plan
CLIDCalling Line Identification
CIRCommitted Information Rate
CNGComfort Noise Generation
ECMError Correction Mode
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Related publications 15
ICMPInternet Control Message Protocol
IEEE802 ESSInstitute of Electrical and Electronics Engineers, Inc., standard 802
Electronic Switching System Identification code
IPInternet Protocol
ISDNIntegrated Services Digital Network
LANLocal Area Network
LATALocal Area and Transport Area
MCRMaximum Cell Rate
MOSMean Opinion Score
PCMPulse Code Modulation
PINGPacket InterNet Groper
PPPPoint-to-Point Protocol
PRIPrimary Rate Interface
PSTNPublic Switched Telephone Network
QoSQuality of Service
RASRegistration, Admissions and Status
RTPReal-time Transfer Protocol
SNMPSimple Network Management Protocol
TCPTransmission Control Protocol
UDPUser Datagram Protocol
UDPUniversal Dialing Plan
VoIPVoice over Internet Protocol
VADVoice Activity Detection
VLANVirtual LAN
WANWide Area Network
Related publications
Documents referenced in the Norstar VoIP Gateway Configuration Guide include:
•Norstar Installer Guide
•Norstar System Coordinator Guide
•Norstar VoIP Gateway Installation Guide
Norstar VoIP Gateway Configuration Guide
Page 16
16 How to get help
How to get help
USA and Canada
Authorized Distributors - ITAS Technical Support
Telephone:
1-800-4NORTEL (1-800-466-7835)
If you already have a PIN Code, you can enter Express Routing Code (ERC) 196#.
If you do not yet have a PIN Code, or for general questions and first line support, you can enter
ERC 338#.
Website:
http://www.nortelnetworks.com/support
Presales Support (CSAN)
Telephone:
1-800-4NORTEL (1-800-466-7835)
Use Express Routing Code (ERC) 1063#
EMEA (Europe, Middle East, Africa)
Technical Support - CTAS
Telephone:
00800 800 89009
Fax:
44-191-555-7980
email:
emeahelp@nortelnetworks.com
CALA (Caribbean & Latin America)
Technical Support - CTAS
Telephone:
1-954-858-7777
email:
csrmgmt@nortelnetworks.com
APAC (Asia Pacific)
Technical Support - CTAS
Telephone:
+61 388664627
Fax:
+61 388664644
email:
asia_support@nortelnetworks.com
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Chapter 1
About Norstar VoIP Gateway
The VoIP Gateway provides excellent voice quality and optimized packet voice streaming over IP
networks. The product enables voice, fax and data traffic to be sent over the same IP network.
The VoIP Gateway incorporates up to four ports for connection to analog trunk ports on a Norstar
KSU or to a fax. These ports supports up to four simultaneous VoIP calls.
Additionally, the VoIP Gateway is equipped with a 10/100 Base-T Ethernet port for connection to
the LAN.
With the VoIP Gateway you can:
•network your Norstar system to another Norstar system or other Nortel Networks Enterprise
communications systems using the IP network
•use your IP network to replace PSTN or other costly private network trunking between
locations
•use the capacity of the enterprise data network for voice and avoid costly access and long
distance charges.
17
The VoIP Gateway is a very compact device, designed to be installed on a desk-top, on the wall, or
in a 19-inch rack.
The VoIP Gateway supports the H.323 ITU protocol, enabling the deployment of "voice over
packet" solutions in environments where each enterprise location is provided with a simple Media
Gateway. This provides the enterprise with the ability to transmit the voice and telephony signals
over a packet network.
The layout diagram, Figure 2 on page 18, illustrates a typical VoIP Gateway application.
Figure 1 Norstar VoIP Gateway front panel
1 2 3 4
C h a n n e l s
Data
Control
LAN
Ready
VoIP Gateway
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18
Figure 2 Typical VoIP Gateway application
VoIP Gateway
Router
Branch Office
Modular ICS
l
y
d
tro
a
t
a
n
N
a
e
o
A
1 2 3 4
C h a n n e l s
R
D
L
C
VoIP Gateway
Branch Office
Compact ICS
Internet
Head Office
Modular ICS
l
y
d
tro
ta
a
n
N
a
e
o
1 2 3 4
C h a n n e l s
R
D
LA
C
VoIP Gateway
VoIP Gateway
Router
VoIP Gateway
1 2 3 4
C h a n n e l s
R
D
LA
C
VoIP Gateway
l
y
o
d
tr
ta
a
n
N
a
e
o
Router
VoIP Gateway key features
•high quality voice, data and fax over IP networks
•supports up to 4 analog telephone loop start ports
•connected to the IP network via a 10/100 Base-T Ethernet interface
•codecs include: G.711, G.723.1, G.729A
•T.38 Fax with superior performance (can handle a round trip delay up to 9 sec.)
•compliant with H.323 (Version 4)
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19
•Emergency Line, connected to the unused pins on port #4, with a relay to an analog line, even
if the VoIP Gateway is powered off.
•LEDs on the front and rear panels that provide information on the operating status of the VoIP
Gateway and the network interface
•Restart button on the front panel that restarts the VoIP Gateway
•compact, rugged enclosure only one-half of a 19-inch rack unit, 1 U high (1.75" or 44.5 mm)
•mounting option of installing two VoIP Gateway in a single 19-inch rack shelf, one U high
(1.75" or 44.5 mm).
Supported H.323 features
The VoIP Gateway implements the RadVision™ H.323 version 4.0 protocol stack. In this version,
the VoIP Gateway features the following:
Gatekeeper
•Works without a Gatekeeper using the internal phone table.
•Registers to a known Gatekeeper.
•Supports Gatekeeper registration with prefixes
•Functions as an H.323 gateway or can imitate an H.323 terminal with up to four aliases.
•Uses routed-mode calls.
•Uses direct-mode calls.
•Uses redundant Gatekeepers if a redundant Gatekeeper is defined.
•Can fallback to internal routing table if there is no communication with the Gatekeeper.
•Supports the "TimeToLive" parameter. The VoIP Gateway sends Registration requests up to
"TimeToLive" expiration.
•Supports IRR messages for KeepAlive.
•Supports the mapping of destination (Alias) numbers in the ACF message by the Gatekeeper.
•Supports RAI (Resource Available Indication) messages, informing gatekeeper that the
gateway resources are below a threshold.
Call setup
•Can use the Normal Connect procedure.
•Can use the Fast Connect procedure with or without immediately opening a H.245 channel.
•Can use Tunneling.
•Can negotiate a codec from a list of given codecs for Normal or Fast Connect procedures.
•Can open a H.245 channel when using Fast Connect.
Norstar VoIP Gateway Configuration Guide
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20
Other:
•Supports using a Country Code (0xB5) and Manufacturers Code (0x28) in H.323 messages
•Supports H.323 Annex D, T.38 real time FAX.
•Supports H.450 Call Hold, Call Transfer and Call Forwarding supplementary services
(H.450.1, H.450.2, H.450.3 and H.450.4).
•Supports the following codecs: G.711 A-law, G.711 µ-law, G.723.1 (6.3 kbps, 5.3 kbps), G.729.
•Supports DTMF and HookFlash signal out of band using the H.245 channel (using the
"Alphanumeric" field).
•Supports DTMF out of band using H.225/Q.931 keypad facility messages.
•Supports of one or two stage dialing for network to VoIP Gateway calls.
•Supports reopening of logical channel and implementation of empty terminal capability set.
•Supports configurable H.323 Port Range.
•Supports H.225/Q.931 Progress Indicator parameter for Fast Connect, enabling playing of
local ringback tone or to cut through the voice channel to listen to remote call progress tones/
messages.
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Chapter 2
IP Telephony overview
This section provides a brief overview of the “Key IP telephony concepts”. It also provides a
“Prerequisites checklist” to help you set up your IP Telephony network.
Key IP telephony concepts
In traditional telephony, the voice path between two telephones is circuit switched. This means that
the analog or digital connection between the two telephones is dedicated to the call. The voice
quality is usually excellent, since there is no other signal to interfere.
In IP telephony, the VoIP Gateway encodes the speech of the call into small data packets called
frames. The system sends the frames across the IP network to the other VoIP Gateway, where the
frames are decoded and sent to the receiving telephone. If some of the frames get lost while in
transit, or are delayed too long, the receiving telephone experiences poor voice quality. On a
properly-configured network, voice quality should be consistent for all IP calls.
21
The following sections describe some of the components for IP telephony:
•“VoIP trunks” on page 21
•“Gatekeepers” on page 22
•“Codecs” on page 22
•“Jitter Buffer” on page 23
•“QoS routing” on page 24
VoIP trunks
VoIP trunks allow voice signals to travel across IP networks. The VoIP Gateway converts the voice
signal into IP packets, which are then transmitted through the IP network to a gateway on the
remote system. The device at the other end reassembles the packets into a voice signal. Business
Communications Manager and Meridian 1 IPT are devices that can use the H.323 protocol trunks
which the VoIP Gateway supports.
VoIP trunks and analog/digital telephones
While analog and digital telephones cannot be connected to the VoIP Gateway system with an IP
connection, they can make and receive calls to and from other systems through VoIP trunks. Calls
received through the VoIP trunks to system telephones are received through the LAN or WAN and
are translated within the VoIP Gateway to voice channels.
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22 Key IP telephony concepts
Gatekeepers
A gatekeeper tracks IP addresses of specified devices, and provides authorization for making and
accepting calls for these devices. A gatekeeper is not required as part of the network to which your
VoIP Gateway is attached, but Gatekeepers can be useful on networks with a large number of
devices.
Note: The VoIP Gateway does not contain a gatekeeper application. If you want to put a
gatekeeper on your network, it must be put on a separate gatekeeper server. The VoIP
Gateway is compatible with RadVision and CSE 1000 gatekeepers.
Codecs
The algorithm used to compress and decompress voice is embedded in a software entity called a
codec (COde-DECode).
Two popular Codecs are G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per
second (kbps) while G.729 samples at a far lower rate of 8 kbps. For actual bandwidth
requirements, refer to “Determining the bandwidth requirements” on page 153, where you will
note that the actual kbps requirements are slightly higher than the label suggests.
Voice quality is better when using a G.711 CODEC, but more network bandwidth is used to
exchange the voice frames between the telephones.
If you experience poor voice quality, and suspect it is due to heavy network traffic, you can get
better voice quality by configuring the IP telephone to use a G.729 CODEC.
The VoIP Gateway supports these codecs:
•G.729
•G.723
•G.711-uLaw
•G.711-aLaw
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Key IP telephony concepts 23
Jitter Buffer
Voice frames are transmitted at a fixed rate, because the time interval between frames is constant.
If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many
cases, however, some frames can arrive slightly faster or slower than the other frames. This is
called jitter, and degrades the perceived voice quality. To minimize this problem, the VoIP
Gateway uses a jitter buffer for arriving frames.
The Norstar VoIP Gateway uses a dynamic jitter buffer that can be configured using two
parameters:
•Minimum delay (0 msec to 150 msec).
This defines the starting jitter capacity of the buffer. For instance, at 0 msec, there is no
buffering at the start. At the default level of 70 msec, the VoIP Gateway will always buffer
incoming packets by at least 70 msec worth of voice frames.
•Optimization Factor (0 to 12).
This defines how the jitter buffer tracks to changing network conditions. When set at its
maximum value of 12, the dynamic buffer will aggressively track changes in delay (based on
packet loss statistics) to increase the size of the buffer and then not decay back down. This
results in the best packet error performance, but at the cost of extra delay. At the minimum
value of 0, the buffer tracks delays only to compensate for clock drift and quickly will decay
back to the minimum level. This optimizes the delay performance but at the expense of a
higher error rate.
The default settings of 70 msec Minimum delay and 7 Optimization Factor should provide a good
compromise between delay and error rate. The jitter buffer "holds" incoming packets for 70 msec
before making them available to the codec for decoding into voice. The codec actually "takes"
frames from the buffer at regular intervals in order to produce continuous speech. As long as
delays in the network do not change (jitter) by more than 70 msec from one packet to the next,
there will always be a sample in the buffer for the codec to use. If there is more than 70 msec of
delay at any time during the call, the packet is too late. The codec will try to access a frame and
will not be able to find one. The codec must produce a voice sample even if a frame is not
available. It will actually create a voice sample to use that minimizes the effect of the loss. This
loss is then flagged as the buffer being too small. The dynamic algorithm then causes the size of
the buffer to increase for the next voice session. The size of the buffer may decrease again if the
gateway notices that the buffer is not filling up as much as expected. At no time will the buffer
shrink to less than the minimum size configured in the Minimum delay parameter.
This delaying of packets can provide somewhat of a communications challenge, as speech is
delayed by the number of frames in the buffer. For one-sided conversations, there are no issues.
However, for two-sided conversations, where one party tries to interrupt the other speaking party,
it can be annoying. In this second situation, by the time the voice of the interrupter reaches the
interruptee, the interruptee has spoken (2*jitter size) frames past the intended point of interruption.
Norstar VoIP Gateway Configuration Guide
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24 Prerequisites checklist
QoS routing
To minimize voice jitter over low bandwidth connections, the VoIP Gateway can assign specific
DiffServ Marking in the IPv4 header of the IP telephony data packets.
The DiffServ Code point (DSCP) is contained in the second byte of the IPv4 header. DSCP is used
by the router to determine how the packets will be separated for Per Hop Behavior (PHB). The
DSCP is contained within the DiffServ field, which was known as the ToS field in older versions.
Prerequisites checklist
Before you set up VoIP trunks on the VoIP Gateway, complete the following checklists to ensure
that the system is correctly set up. Some questions do not apply to all installations.
This guide contains a number of appendices that explain various aspects of the system directly
related to IP telephony functions.
This section includes the following checklists:
•“Network diagram” on page 24
•“Network devices” on page 25
•“Network assessment” on page 25
Network diagram
To aid in installation, a Network Diagram is needed to provide a basic understanding of how the
network is configured. Before you install IP functionality, you must have a network diagram that
captures all of the information described in the following table.
Table 1 Network diagram prerequisites
PrerequisitesYes
1.a Has a network diagram been developed?
1.b Does the network diagram contain any routers, switches or bridges with corresponding
IP addresses and bandwidth values for WAN or LAN links?
1.c Does the network diagram contain IP Addresses, netmasks, and network locations of all Norstar VoIP
Gateways?
1.d Does the network diagram contain IP Addresses and netmasks of any other IP Telephony gateways
that you need to connect to?
1.e Does the network diagram contain the IP address for any Gatekeeper that may be used?
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Network assessment 25
Network devices
The following table contains questions about devices on the network such as firewalls, NAT
devices, and DHCP servers.
•If the network uses public IP addresses, complete 2.c.
•If the network uses private IP addresses, complete 2.d. to 2.e.
Table 2 Network device checklist
PrerequisitesYesNo
2.a Is the network using DHCP?
2.b Is the network using private IP addresses?
2.c Do you have a public IP addresses for the Norstar VoIP Gateway?
2.d Does the system have a firewall/NAT device?
2.e A hub-based core will not have suitable performance for IP Telephony. Does the network
use a non-hub solution at its core?
Network assessment
The following table of questions are meant to ensure that the network is capable of handling IP
telephony, and that existing network services are not adversely affected.
Table 3 Network assessment
PrerequisitesYesNo
3.a Has a network assessment been completed?
3.b Has the number of switch/hub ports available and used in the LAN infrastructure been
calculated?
3.c Does the switch use VLANs? If so, get the VLAN port number and ID.
3.d Have the used and available IP addresses for each LAN segment been calculated?
3.e Has DHCP usage and location been recorded?
3.f Has the speed and configuration of the LAN been calculated?
3.g Has the estimated latency values between network locations been calculated?
3.h Have the Bandwidth/CIR utilization values for all WAN links been calculated?
3.i Has the quality of service availability on the network been calculated?
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26 Network assessment
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Chapter 3
Configuring the VoIP Gateway
The VoIP Gateway has a web interface you use for gateway configuration, including downloading
of configuration files and for run-time monitoring. You can access the web interface from any
standard web browser, such as Microsoft™ Internet Explorer or Netscape™ Navigator.
Specifically, you can employ this facility to set up the gateway configuration parameters needed to
configure the VoIP Gateway. You also have the option to reset the gateway to apply the new set of
parameters.
Computer requirements
To use the web interface, you need the following:
•a computer capable of running your web browser
•a network connection to the VoIP Gateway
•one of the following compatible web browsers
•Microsoft™ Internet Explorer™ (version 5.0 and higher)
•Netscape™ Navigator™ (version 7.0 and higher)
27
Accessing the web interface
To access the web interface:
1Open your web browser.
2In the URL field, enter the IP address of the VoIP Gateway.
When you enter the IP address make sure you include http:// at the start of the IP address (for
example: http://10.1.10.10.
The Enter Network Password screen appears.
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28 Accessing the web interface
Figure 3 Web browser login screen
3Enter the User Name and Password.
If you have not changed the user name and password, the default User Name is Admin and the
default password is Admin.
Note: Nortel Networks recommends that you change the User Name and Password from
their default values. For information about how to change the User Name and Password,
refer to “Changing the VoIP Gateway password” on page 133.
4Click the OK button.
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Configuring the Protocol Definition parameters 29
Configuring the Protocol Definition parameters
To configure the Protocol Definition parameters:
1Access the web interface.
2Click the Protocol Management button.
3Click the Protocol Definition tab.
The Protocol Definition screen appears.
Figure 4 Protocol Definition screen
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30 Configuring the Protocol Definition parameters
4Configure the Protocol Definition parameters according to the following table.
Table 4 Protocol Definition parameters
ParameterDescription
General
Connection ModeSelect Fast Start if you want the VoIP Gateway to use the Fast Start connection mode.
Select Normal if you want the VoIP Gateway to use the Normal connection mode.
The default value is Fast Start.
The Fast Start connection mode allows a media path to be established using H.225,
without having to start the full H.245 protocol session. In some situations, you need to use
fast start in order to control call progress tones.
Enable Annex D/T.38 Fax
relay
Enable DTMF over H.245 Select Yes to enable DTMF over H.245.
H.323-IDEnter the VoIP Gateway H.323-ID you want to use for registration to the Gatekeeper.
Source Number
Encoding Type
Destination Number
Encoding Type
Select No to disable Annex D/T.38 Fax relay.
Select Yes to enable Annex D/T.38 Fax relay.
When you enable this feature, the VoIP Gateway can send and receive Fax messages
using the H.323 Annex D T.38 procedure.
When you enable this feature, the VoIP Gateway sends out of band DTMF signaling using
H.245. Out of band signaling is recommended for use with H.323 protocol and the Norstar
KSU.
You can enter a string up to 19 characters long.
When you are using a Gatekeeper, the VoIP Gateway will send a registration message to
the Gatekeeper with this H323-ID string.
Select the source number encoding type. This defines the encoding type of the calling
phone number in H.225 setup messages.
You can select E.164, H.323-ID, E.164 & H.323-ID, TableValues or TableValues &
H.323-ID.
Select TableValues if you want the VoIP Gateway to use the values configured in the
Tel -> IP Source Number Manipulation table.
Select H323-ID if you want the VoIP Gateway to add the H323-ID to the source
information.
Select E.164 if you want the VoIP Gateway to use E.164 source encoding and not use the
encoding type defined in the table.
The default value is E.164.
Note: If you select an option that includes “H.323-ID”, you must enter a string in the
H.323-ID box.
Select the destination number encoding type. This defines the encoding type of the called
phone number in H.225 setup messages.
You can select E.164, H.323-ID, E.164 and H.323-ID, or TableValues.
Select TableValues if you want the VoIP Gateway to use the values configured in the
Tel -> IP Dest Number manipulation table.
Select H.323-ID if you want the VoIP Gateway to add the H323-ID to the destination
information.
Select E.164 if you want the VoIP Gateway to use E.164 destination encoding and not use
the encoding type defined in the table.
The default value is E.164.
Note: If you select an option that includes “H.323-ID”, you must enter a string in the
H.323-ID box.
Q.931 MultiplexingSelect No to disable H.323 Q.931 multiplexing.
Select Yes to enable H.323 Q.931 multiplexing.
The default value is No.
When you enable Q.931 multiplexing, the VoIP Gateway uses the same socket for all
H.225 messages sent to the same destination.
Gatekeeper
Working with Gatekeeper Select Yes if you are using a Gatekeeper to determine what IP address should be used
Gatekeeper IP addressEnter the IP address of the Gatekeeper you are using.
Gatekeeper IDEnter the string used to identify the Gatekeeper.
Gatekeeper Redundancy Select No if you are using a single Gatekeeper.
First Redundant
Gatekeeper
Second Redundant
Gatekeeper
Fallback to internal
routing
for the telephone number dialed.
Select No if you are not using a Gatekeeper to resolve telephone number to IP address
translation. If you are not using a Gatekeeper, you must configure the Telephone to IP
Routing table on the VoIP Gateway.
Enter the IP address in dotted format notation, for example 192.10.1.255.
Select Yes if you are using two or three Gatekeepers.
If you enable Gatekeeper Redundancy, the VoIP Gateway can work with multiple
Gatekeepers. If there is no response from the current Gatekeeper, the VoIP Gateway tries
to communicate with the other Gatekeepers. When a new Gatekeeper is found, the VoIP
Gateway continues working with it until the next failure.
To use Gatekeeper Redundancy, you must enter an IP address in the First Redundant Gatekeeper box. If you are using three Gatekeepers, you also need to enter an IP
address in the Second Redundant Gatekeeper box.
Enter the IP address of the first redundant Gatekeeper you are using.
Enter the IP address in dotted format notation, for example 192.10.1.255.
Note: This parameter is available only if you select Yes in the Gatekeeper Redundancy
box.
Enter the IP address of the second redundant Gatekeeper you are using.
Enter the IP address in dotted format notation, for example 192.10.1.255.
Note: This parameter is available only if you select Yes in the Gatekeeper Redundancy
box.
Select No if you do not want to use the internal Telephone to IP Routing table when the
Gatekeeper is not available.
Select Yes if you want to use the internal Telephone to IP Routing table when the
Gatekeeper is not available.
When the VoIP Gateway falls back to the internal Telephone to IP Routing table, the VoIP
Gateway continues scanning for the Gatekeeper. When the VoIP Gateway finds an active
Gatekeeper, it switches from internal routing back to Gatekeeper routing.
Register as TerminalSelect No if you want the VoIP Gateway to register and act as a standard gateway.
Enable Mapping of Alias
Number
Gatekeeper Timers
Registration TimeEnter the time in seconds between registrations to the Gatekeeper.
RAS Response TimeoutEnter the time in seconds that the VoIP Gateway waits for a RAS response from the
Number of RAS
retransmission
Time between GK retries Enter the time in seconds before the VoIP Gateway tries to contact the list of Gatekeepers
Select the registration type you want to use. This defines the encoding type of the VoIP
Gateways phone numbers that is used when the VoIP Gateway registers these numbers
with a Gatekeeper.
You can select E.164, H.323-ID, E.164 & H.323-ID, TableValues, or TableValues & H.323-ID.
Select TableValues if you want the VoIP Gateway to use the values configured in the
Gateway Registration Prefixes Table.
Select H323-ID if you want the VoIP Gateway to add the H323-ID to the destination
information.
Select E.164 if you want the VoIP Gateway to use E.164 encoding and not use the
encoding type defined in the table.
The default value is E.164.
Note: If you have entered a string in the H.323 ID box, this string is added to the gateway
registration when you select H.323-ID, E.164 & H.323-ID, or TableValues & H.323-ID.
Select Yes if you want the VoIP Gateway to register and act as an H.323 terminal with
multiple aliases (up to 4). Also, in all gateway messages, the terminal type value is set to
terminal.
The default value is No.
Select Yes to allow the Gatekeeper to change the gateway destination number using the
Alias parameter in the ACF message.
Select No if you do not want the Gatekeeper to change the gateway destination number.
The default Registration Time is 60 seconds.
Note: This setting must match the configuration settings on your Gatekeeper.
Gatekeeper. When this time expires, the VoIP Gateway retransmits the RAS message.
The default time is 2 seconds.
Note: This setting must match the configuration settings on your Gatekeeper.
Enter the number of times that the VoIP Gateway retransmits the RAS message to the
Gatekeeper, before the VoIP Gateway decides that the Gatekeeper is not responding.
If you have enabled Gatekeeper Redundancy, the VoIP Gateway will then try the next
Gatekeeper.
If none of the Gatekeepers are responding and you have enabled Fallback to internal
routing, the VoIP Gateway will start using the internal Telephone to IP Routing table.
The default number of retransmissions is 2.
Note: This setting must match the configuration settings on your Gatekeeper.
again.
The default time is 60 seconds.
Note: This setting must match the configuration settings on your Gatekeeper.
1st CoderSelect the first preferred codec for the VoIP Gateway.
This codec is the highest priority codec and is used by the VoIP Gateway whenever
possible.
Select the size of the Voice Packet used with this codec in milliseconds.
Selecting the size of the packet determines how many codec payloads are combined into
one RTP (Voice) packet.
2nd CoderSelect the second preferred codec for the VoIP Gateway.
If the far end gateway cannot use the codec assigned as the 1st Coder, the VoIP Gateway
attempts to use this codec.
Select the size of the Voice Packet used with this codec in milliseconds.
Selecting the size of the packet determines how many codec payloads are combined into
one RTP (Voice) packet.
3rd CoderSelect the third preferred codec for the VoIP Gateway.
If the far end gateway cannot use the codecs assigned as the 1st Coder or 2nd Coder, the
VoIP Gateway attempts to use this codec.
Select the size of the Voice Packet used with this codec in milliseconds.
Selecting the size of the packet determines how many codec payloads are combined into
one RTP (Voice) packet.
4th CoderSelect the fourth preferred codec for the VoIP Gateway.
If the far end gateway cannot use the codecs assigned as the 1st Coder, 2nd Coder or
3rd Coder, the VoIP Gateway attempts to use this codec.
Select the size of the Voice Packet used with this codec in milliseconds.
Selecting the size of the packet determines how many codec payloads are combined into
one RTP (Voice) packet.
5th CoderSelect the fifth preferred codec for the VoIP Gateway.
If the far end gateway cannot use the codecs assigned as the 1st Coder, 2nd Coder,
3rd Coder or 4th Coder, the VoIP Gateway attempts to use this codec.
If the far end gateway cannot use any of these five codecs, a VoIP connection cannot be
made.
Select the size of the Voice Packet used with this codec in milliseconds.
Selecting the size of the packet determines how many codec payloads are combined into
one RTP (Voice) packet.
DTMF and Dialing Parameters
Enable Automatic Dialing Select Yes to enable Automatic Dialing.
Select No to disable Automatic Dialing.
When you enable Automatic Dialing, a telephone number is automatically dialed as soon
you select one of the VoIP Gateway lines. To use the Automatic Dialing feature, you must
configure the Automatic Dialing Table. For information about how to configure the
Automatic Dialing Table, refer to “Configuring the Automatic Dialing phone numbers” on
page 61.
When you disable Automatic Dialing, you hear dial tone when you select one of the VoIP
Gateway lines. You can then dial the telephone number of the person you want to call.
Recommendations:
We recommend that you select No to disable Automatic Dialing.
Interdigits TimeoutEnter the time in seconds that the VoIP Gateway waits between digits dialed by the user.
Use ‘#’, “*’ digits for
dialing
Dial Tone DurationEnter the time in seconds that the dial tone is played.
Fast Start Parameters
Open H.245 channelSelect No if you do not want the VoIP Gateway to open an H.245 channel when making a
Play Ringback Tone to IP Select Dont Play if you do not want the Ringback tone played to IP side of the call.
Enter the maximum number of digits that can be dialed.
You can enter a value from 2 to 19.
The default value is 4.
Note: Dialing ends when the maximum number of digits are dialed, the Interdigits Timeout
expires, or the '#' key is dialed.
When the Interdigits Timeout expires, the VoIP Gateway will attempt to dial the digits
already received.
You can enter a value of 0 to 5 seconds.
The default value is 4 seconds.
Select No if you do not want to allow the "*" and "#" to be used for telephone numbers
dialed by a user or entered for the endpoint telephone numbers.
Select Yes if you want to allow the "*" and "#" to be used for telephone numbers dialed by
a user or entered for the endpoint telephone number.
The default value is No.
Note: When this feature is activated, the # can not be used to end the dialed number.
Note: The # and * can always be used as first digit of a dialed number, even if you select
No for this parameter.
The default time is 16 seconds.
Note: Dial Tone Duration is not applicable when Automatic Dialing is enabled.
Fast Start connection.
Select Yes if you want the VoIP Gateway to open an H.245 channel immediately after the
Fast Start connection is established. The opening of an H.245 channel may be needed for
relaying DTMF digits over H.245 channel during a call.
The default value is No.
Recommendations:
For quicker call setup when you are using Fast Start, set this parameter to Yes.
Select Play if you want the Ringback tone played to the IP side of the call. When you
select Play, the Progress Indicator is set to 8 in the H.225 Alert message (PI=8).
The default value is Play.
Note: If you are using Auto Answer lines on the Norstar KSU and digit delivery to
automatically ring a destination Norstar set, select Dont Play for this parameter.
Recommendations:
If the destination lines on the Norstar KSU are configured as manual answer, set this
parameter to Play.
If the destination lines on the Norstar KSU are configured as auto answer, set this
parameter to Dont Play.
When you select Dont Play the Norstar KSU provides ringback tone.
Play Ringback Tone to Tel Select Dont Play if you do not want to play the Ringback Tone to the Telephone side of
Progress Indicator to
ISDN
Progress Indicator to IPSelect 0 if you do not want to set the Progress Indicator (PI) in the H.225 Alert messages
Send Media Information
in Connect
the call.
Select Play if you want to play the Ringback Tone to the Telephone side of the call.
Select Play according to PI if want to play the Ringback Tone to the Telephone side of
the call when indicated by the Progress Indicator. When you select this option, the
Ringback Tone is played when the Progress Indicator is set to PI=1 or PI=8 in the
received H.225 Alert message. (default)
The default value is Dont Play.
Recommendations:
If there are any Business Communications Manager or Meridian 1 IPT gateways on your
VoIP network, set this parameter to Dont Play.
Select 0 if you do not want to set the Progress Indicator (PI) in the H.225 Alert messages
sent to an ISDN endpoint.
Select 1 if you want to set the Progress Indicator to PI = 1 in H.225 Alert messages sent.
Select 8 if you want to set the Progress Indicator to PI = 8 in H.225 Alert messages sent.
Select Not configured if you want to set the Progress Indicator in the H.225 Alert
messages according to "Play Ringback Tone to IP" parameter.
The default value is Not Configured.
Recommendations:
If there are any Business Communications Manager or Meridian 1 IPT gateways on your
VoIP network, set this parameter to Not configured.
sent to an IP telephony endpoint.
Select 1 if you want to set the Progress Indicator to PI = 1 in H.225 Alert messages sent.
Select 8 if you want to set the Progress Indicator to PI = 8 in H.225 Alert messages sent.
Select Not configured if you want to set the Progress Indicator in the H.225 Alert
messages according to "Play Ringback Tone to IP" parameter.
The default value is Not Configured.
Recommendations:
If there are any Business Communications Manager or Meridian 1 IPT gateways on your
VoIP network, set this parameter to Not configured.
Select No to send the Fast Start Structure response in the Alert message. This provides
the media path used when the destination endpoint is providing the Ringback Tones.
Select Yes to send the Fast Start Structure response after the call is answered and the
connect message is sent.
After receiving the Fast Start Setup message, the gateway should reply with an H.225
message that includes media information structure. The gateway can send the message
with Alert or Connect messages. Sending this information in Alert message enables the
remote side to open the voice channel before receiving the Connect message.
The default value is No.
Note: This option is only relevant if Fast Start is used.
Recommendations:
If there are any Business Communications Manager or Meridian 1 IPT gateways on your
VoIP network, set this parameter to No.
Open Media On Connect Select No to open the voice channel after sending an Alert message.
Select Yes to open the voice channel after the call is answered and a Connect message is
sent.
The default value is No.
Note: This option is only relevant if you use Fast Start or Tunneling.
Recommendations:
If there are any Business Communications Manager or Meridian 1 IPT gateways on your
VoIP network, set this parameter to No.
Number Manipulation and Routing Modes
Add Trunk Group ID as
Prefix
IP to Tel Routing ModeSelect Route calls before manipulation if you want to route the calls before the number
Tel to IP Routing ModeSelect Route calls before manipulation if you want to route the calls before the number
H450 Supplementary Services
Select No if you do not want the VoIP Gateway to add the Trunk Group ID
Select Yes if you want to add the Hunt Group ID as a prefix to the destination phone
number for incoming calls. This option can be used to define various routing rules.
To use this feature you must configure the Hunt Group IDs.
Recommendations:
We recommend you set this parameter to No.
manipulation rules are applied.
Select Route calls after manipulation if you want to route the calls after the number
manipulation rules are applied.
The option is not applicable if you use a Gatekeeper.
The default is Route calls before manipulation.
Recommendations:
We recommend you set this parameter to Route calls before manipulation.
manipulation rules are applied.
Select Route calls after manipulation if you want to route the calls after the number
manipulation rules are applied.
The option is not applicable if you use a Gatekeeper.
The default is Route calls before manipulation.
Recommendations:
We recommend you set this parameter to Route calls before manipulation.
Enable HoldSelect No to disable Hold service.
Select Yes to enable Hold service (H.450.4).
If the Hold service is enabled, a user can activate Hold (or unhold) using the HookFlash.
On receiving a Hold request, the remote party will be put on hold and hear the hold tone.
The default is No.
Note: To use the H.450 Supplementary Services, the gateways at both ends must support
these services.
Recommendations:
The Business Communications Manager and Meridian 1 IPT gateways do not support
H.450 Supplementary Services. If there are any Business Communications Manager or
Meridian 1 IPT gateways on your VoIP network, set this parameter to No.
Enable TransferSelect No to disable Call Transfer service.
Select Yes to enable Call Transfer service (H.450.2).
If the Transfer service is enabled, the user can activate Transfer using HookFlash
signaling. If this service is enabled, the remote party will perform the call transfer.
The default is No.
Note: To use the H.450 Supplementary Services, the gateways at both ends must support
these services.
Recommendations:
The Business Communications Manager and Meridian 1 IPT gateways do not support
H.450 Supplementary Services. If there are any Business Communications Manager or
Meridian 1 IPT gateways on your VoIP network, set this parameter to No.
Enable ForwardSelect No to disable the Call Forward service.
Select Yes to enable Call Forward service (H.450.3).
A Call Forward table must be defined to use the Call Forward service.
The default is No.
Note: To use the H.450 Supplementary Services, the gateways at both ends must support
these services.
Recommendations:
The Business Communications Manager and Meridian 1 IPT gateways do not support
H.450 Supplementary Services. If there are any Business Communications Manager or
Meridian 1 IPT gateways on your VoIP network, set this parameter to No.
Misc parameters
IP SecuritySelect No if you want the VoIP Gateway to accept all H.323 calls that are sent to the VoIP
Gateway.(default)
Select Yes if you want the VoIP Gateway to accept H.323 calls only from IP addresses
defined in Prefix routing table. The VoIP Gateway rejects all calls from unknown IP
addresses.
When you are using a Gatekeeper, Nortel Networks recommends you set this option to
Yes. This allows the Gatekeeper to decide if H.323 calls are accepted or rejected.
Channel Select ModeSelect By Phone Number if you want the VoIP Gateway to select the channel according
Enable Caller IDSelect Yes for this option.
Caller ID TypeSelect Bellcore for this option.
Polarity ReversalSelect No if you do not want the VoIP Gateway to change the line polarity.
Enable Current
Disconnect
Enable Hook-FlashSelect No to disable Hook-Flash.
Send DTMF over H225/
Q.931
to the destination telephone number received in H.225 setup message. This telephone
number is the number you enter in the Phone Number box on the Channels - Hunt Group
screen.
If you are using Hunt Groups, select one of the following four options.
•Select Cyclic Ascending if you want the VoIP Gateway to always select the next
higher channel number in the Hunt Group. When the VoIP Gateway reaches the
highest channel number in the Hunt Group, it will select the lowest channel number in
the Hunt Group and then start ascending again.
•Select Ascending Always if you want the VoIP Gateway to always start at the lowest
channel number in the Hunt Group and if that channel is not available, select the next
higher channel.
•Select Cyclic Descending if you want the VoIP Gateway to always select the next
lower channel number in the Hunt Group. When the VoIP Gateway reaches the lowest
channel number in the Hunt Group, it will select the highest channel number in the
Hunt Group and then start descending again.
•Select Descending Always if you want the VoIP Gateway to always start at the
highest channel number in the Hunt Group and if that channel is not available, select
the next lower channel.
For information about how to enter the Phone Number and Hunt Group ID, refer to
“Configuring the Channels to Hunt Group parameters” on page 58.
Note: To prevent conflicts, configure the Norstar Hunt Group to start its hunt from the
opposite end of the channel numbers. For example, if the VoIP Gateway starts the hunt
from the lowest channel number, ensure the Norstar starts its hunt from the highest
channel number.
When you select Yes, the Calling number and Display text is sent to the Norstar KSU
between first and second rings.
Bellcore is the Caller ID Type that is compatible with the Norstar KSU.
Select Yes if you want the VoIP Gateway to change the line polarity on call answer and
change it back on call release.
The default value is No.
Recommendations:
For use with the Norstar KSU, you must set this parameter to No.
Select Yes to generate a Current disconnect pulse of 900msec after a call is released
from IP.
Recommendations:
For use with the Norstar KSU, you must set this parameter to Yes.
Select Yes if you want the VoIP Gateway to send an H.245 User Input Indication message
when a hookflash is detected.
The default is No.
Select No for this option.
The Norstar VoIP Gateway uses H.245 to send DTMF out of band (refer to the Enable
DTMF over H.245 option).
Number of Max CallsEnter the maximum number of calls that the VoIP Gateway can have active at the same
H323 Base PortEnter the starting TCP/UDP transport port for H.225/H.245 messages. The VoIP Gateway
H225 Listen PortEnter the port number on which the VoIP Gateway expects to receive H.225/Setup
H225 Dial PortEnter the port number on which the VoIP Gateway sends H.225/Setup messages.
Enable SysLog CDRSelect No if you do not want to send Call Detail Recording (CDR) information to the
Select No if you do not want the VoIP Gateway to release the call when it receives a
Setup Ack message.
Select Yes if you want the VoIP Gateway to release the call when it receives a Setup Ack
message.
Use this option to enable receiving Setup Ack messages, when the Enable DTMF overlap
Dialing option is disabled.
The default is No.
Select No if you want the VoIP Gateway to send an empty SETUP (or Gatekeeper ARQ)
message. An empty SETUP message does not include the called party number.
Select Yes if you want the VoIP Gateway to include the called party number in the SETUP
message.
The default is Yes.
Recommendations:
Select Yes for this parameter.
Enter the telephone number that the VoIP Gateway dials if a call setup is received with no
telephone number.
The number you enter must be one of the End Points Phone Numbers you have defined.
time.
If you enter 0, there is no restriction on the maximum number of calls.
The default value is 0.
will use dynamic ports (except RTP ports) starting from this port to this port +500.
If you enter 0, the default ports are used.
The default port range is 32000 to 65000.
Note: You normally do not have to change the base port number unless you are using the
VoIP Gateway behind NAT. Nortel Networks does not recommend using the VoIP
Gateway behind NAT.
messages.
The default port number is 1720.
Note: You normally do not have to change the base port number unless you are using the
VoIP Gateway behind NAT. Nortel Networks does not recommend using the VoIP
Gateway behind NAT.
The default port number is 1720.
Note: You normally do not have to change the base port number unless you are using the
VoIP Gateway behind NAT. Nortel Networks does not recommend using the VoIP
Gateway behind NAT.
SysLog server.
Select Yes if you want to sent CDR information to the Syslog server at end of each Call.
The default is No.
5Click the Submit button to save your changes.
6To apply the changes, you must reset the VoIP Gateway. To reset the VoIP Gateway:
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40 Configuring the Protocol Definition parameters
aClick the Reset button.
bClick the Restart button.
A confirmation dialog box appears.
cClick the OK button.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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Configuring the Registration Prefixes
The Gatekeeper Registration Prefix Table allows you to associate dialing plan information with
specific digit prefixes.
Gatekeepers can route calls based on two pieces of information: destination digits and numbering
plan. The ability to route by numbering plan is useful when Gatekeepers must route both private
and public calls. For example, company A has an internal private network with private access
codes. These codes are 7 digit numbers comprising a location code and an internal DN. A code of
395-3491 also looks like a valid local public number. The numbering plan information allows the
Gatekeeper to route the apparently ambiguous destination digits that could be dialed on both a
private and public network. For example, a digit string of 395-3491 with a numbering plan of
E.164 Public / National is routed to the destination IP address of a gateway that interfaces with the
PSTN. The same digit string with a numbering plan of Private / Level1 Regional is routed to the IP
address of a gateway on the private network.
The Gatekeeper Registration Prefix Table data is sent when the gateway requests permission to
register with the Gatekeeper. This registration request is triggered by a gateway reset.
If there are no entries in the Registration Prefix Table, then the VoIP Gateway registers with the
Gatekeeper using the numbers defined on the Channels - Hunt Group screen and the NPI / TON
selected in the Gateway Registration Type parameter on the Protocol Definition screen.
Configuring the Registration Prefixes 41
Using a Radvision ECS 3.0 Gatekeeper
The Norstar VoIP Gateway works with the Radvision ECS 3.0 Gatekeeper. The Radvision
Gatekeeper uses both the destination digits and the numbering plan information (NPI/TON) to
resolve a called number into an IP address for the destination. If you are using a Radvision
Gatekeeper, we recommend that you configure the Registration Prefix Table to guarantee
maximum flexibility.
Using a Nortel Networks CSE1000 Gatekeeper
The Norstar VoIP Gateway works with the Nortel Networks CSE1000 Gatekeeper. The CSE1000
Gatekeeper is manually administered to set up its dialing plan and does not require the use of the
Gatekeeper Registration Prefixes Table.
The CSE1000 Gatekeeper is setup based on the H323-ID. The H323-ID is a unique text string used
to represent the gateway. When using the CSE1000 Gatekeeper, select H323-ID as the Gateway
Registration Type.
To configure the Gatekeeper Registration Prefixes parameters:
1Access the web interface.
2Click the Protocol Management button.
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42 Configuring the Registration Prefixes
3Click the Registration Prefixes tab.
The Registration Prefixes Table screen appears.
Figure 5 Registration Prefixes Table screen
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Configuring the Registration Prefixes 43
4Configure the Registration Prefixes according to the following table.
Table 5 Registration Prefixes Table
ParameterDescription
GK Registration Prefixes Enter the Gatekeeper Registration prefix. These prefixes are used by the VoIP Gateway to
register to a Gatekeeper.
When entering a prefix, you can create an entry that represents multiple prefixes using
the following notation:
•[n-m] represents a range of numbers. For example, enter [250-279] to specify all
prefixes from 250 to 279.
Type of NumberSelect the H.225/Q.931 Number Type assigned to this entry.
You can select Unknown, Private or E.164 Public.
The default is Unknown.
Select the H.225/Q.931 Number Plan assigned to this entry.
•If you selected Unknown as the Number Type, you can select Unknown.
•If you selected Private as the Number Type, you can select Unknown, Level 0 Regional (Local), Level 1 Regional, Level 2 Regional or PISN Specific.
•If you selected E.164 Private as the Number Type, you can select Unknown, International, National, Network Specific, Subscriber or Abbreviated.
The default is Unknown.
If you are using a RadVision Gatekeeper, you must configure the Type of Number fields.
If you are using a CSE1k Gatekeeper, you are not required to configure the Type of
Number fields.
Note: The Type of Number information is used only for registration of this number where
the VoIP Gateway is the destination endpoint. The Type of Number encoding used when
dialing from this VoIP Gateway is set in the number manipulation tables for Telephone to
IP calls.
See Appendix A, “Numbering and Dial Plans,” on page 147, for more information about
the Type of Number fields.
5Click the Submit button to save your changes.
6To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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44 Configuring the Number Manipulation tables
Configuring the Number Manipulation tables
The VoIP Gateway provides Number Manipulation tables for incoming and outgoing calls. These
tables are used to modify the destination and source telephone numbers so that the calls can be
routed properly.
The VoIP Gateway has four number Manipulation tables.
•“Configuring number manipulation for Destination Phone Numbers for Norstar to IP calls” on
page 46
•“Configuring number manipulation for Digit Delivery” on page 49
•“Configuring number manipulation for Source Phone Numbers for IP to Norstar calls” on
page 52
•“Configuring number manipulation for Source Phone Numbers for Norstar to IP calls” on
page 55
Note: Number manipulation can occur either before or after a routing decision is made.
To control when number manipulation is done, set the IP to Tel Routing Mode and the Tel to IP Routing Mode parameters. For information about these parameters, refer the
description of these parameters on Table 4 on page 30.
Number Manipulation overview
The Norstar VoIP Gateway contains a number of very powerful and useful utilities for the
manipulation of source and destination telephone numbers used on a call by call basis.
Understanding how the manipulation works and what circumstances would need digit
manipulation is key to setting up the overall network dialing plan.
Basic Concepts
The VoIP Gateway uses four manipulation tables:
•IP -> Tel Source Numbers
•Tel -> IP Source Numbers
•Tel -> IP Destination Numbers
•Digit Delivery (IP -> Tel Destination Numbers)
In addition, similar tables are used for routing decisions in the VoIP Gateway:
•Tel -> IP Routing table
•IP -> Hunt Group Routing table.
These routing tables do not use any manipulation features but simply allow calls to be directed
properly, as determined by the dialed digits.
Each of these will be explained further in sections to follow. Each manipulation table utilizes the
same basic set of rules.
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Configuring the Number Manipulation tables 45
Digit parsing
In order to determine whether or not a particular call requires digit manipulation or routing, the
number (source or destination) is parsed to see if it matches the digit rule in the table (Dialed
number, Source Number, Prefix). Digit parsing options include:
•Digits: Numerals from 0 – 9.
•Wild card: * means any number of digits of any type
•Don’t Care: X means any single digit of any type
•Terminator: # indicates the end of the digit string (no digits follow)
•Range: [x - y] indicates the range of numbers from x to y inclusive
•Range: [x,y,z] indicates the value may be any of the indicate list
For example:
•[5551200 – 5551300] represents all of the numbers from 5551200 to 5551300 inclusive.
•[3,4,5]xxx# would be a 4 digit number beginning in either 3, 4 or 5.
•123[100 -200]* would be numbers that start with 123100 to 123200 inclusive.
Number Classification – NPI / TON
In the tables that provide manipulation from Tel -> IP, you may specify the NPI / TON to be used
in the H.323 message coding. This classification is important in some gatekeepers (Radvision) in
order to provide address resolution. The NPI / TON used in your setup message must be the same
as that used in registration by the destination gateway.
NPI / TON is a term originally used with Q.931 which is the basis for H.323. Choices for
Numbering Plan Indicator (NPI) are:
•Unknown
•Private
•Public (e164)
Public is chosen if the number is part of the PSTN numbering plan that is used for national dialing.
Private is used to represent a numbering plan that is used in a private voice network.
For more information on Numbering and Dial Plans, refer to Appendix A, “Numbering and Dial
Plans,” on page 147.
Stripping and adding digits
The tables provide the ability to strip leading digits from the digit string (number of stripped digits)
and to add new leading digits (prefix to add).
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46 Configuring the Number Manipulation tables
Configuring number manipulation for Destination Phone Numbers
for Norstar to IP calls
This number manipulation table is used to manipulate the digits that are dialed into the VoIP
Gateway prior to setting up the call. Numbers that are dialed by the user are first checked in this
table to see if there is a match. If there is a match, the values entered into the table are used to
manipulate the digits. This is programmed in "Protocol Management / Protocol Definition" by
changing the setting for "Tel to IP Routing Mode". This manipulation can be done either before or
after a routing decision is made for the call. In general, a routing decision will be made before
manipulation.
Uses for manipulation can be as follows:
•To strip dialing plan digits from the number. For example, a user might dial 6 in front of each
number in order to indicate a private network number. If this leading digit is not removed by
Norstar routing, it may be removed here before the call is setup. This would allow the
destination number to be understood by a gatekeeper or by the destination gateway (for
routing to the correct output hunt group).
•Assignment of NPI / TON. The VoIP Gateway can use a single global setting for NPI / TON
classification or it can use the setting in this table on a call by call basis. Control for this is
done using “Protocol Management / Protocol Definition / Destination Number Encoding
Type”
To configure the Destination Phone Number Manipulation Table for TEL->IP calls:
1Access the web interface.
2Click the Protocol Management button.
3Point to the Manipulation Tables tab and click the Tel -> IP Dest Numbers link.
The
Destination Phone Number Manipulation Table for TEL->IP calls screen appears.
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Configuring the Number Manipulation tables 47
Figure 6 Destination Phone Number Manipulation Table for TEL->IP calls screen
4To configure the first ten entries, click the Table Index drop list and click 1-10.
To configure the last ten entries, click the Table Index drop list and click 11-20.
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48 Configuring the Number Manipulation tables
5Configure the Phone Number Manipulation Table according to the following table.
Table 6 Destination Phone Number Manipulation Table for TEL->IP calls (Norstar to IP)
ParameterDescription
Dialed NumberEnter the number you want to match for this table entry.
When entering a number, you can create an entry that represents multiple numbers using
the following notation:
•[n-m] represents a range of numbers
•[n,m] represents multiple numbers
•x represents any single digit
•* represents any digit or multiple digits
•# represents the end of a number
•For example:
[5551200-5551300] represents all of the numbers from 5551200 to 5551300
[2221,2231,2241] represents the numbers 2221, 2231 and 2241
54324* represents any number that starts with 54324
54324xx represents a 7 digit number that starts with 54324
123[100-200]* represents numbers that starts with 123100 to numbers that start with
123200.
When the dialed number matches the number you enter here, the rules in this number
manipulation table entry are applied.
The VoIP Gateway matches the rules starting at the top of the table. For this reason, enter
more specific rules above more generic rules. For example, if you enter 551* in entry 1
and 55* in entry 2, the VoIP Gateway applies rule 1 to numbers that starts with 551 and
applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559.
However if you enter 55* in entry 1 and 551* in entry 2, the VoIP Gateway applies rule 1 to
all numbers that start with 55 including numbers that start with 551.
Num of stripped digitsEnter the number of digits that you want to remove from the destination phone number.
For example, if you enter 3 and the destination phone number is 5551234, the new
destination phone number is 1234.
Prefix to addEnter the number you want to add to the front of the stripped destination phone number.
For example, if you enter 9 and the stripped destination phone number is 1234, the new
destination phone number is 91234.
Type of NumberSelect the H.225/Q.931 Number Type assigned to this entry.
You can select Unknown, Private or E.164 Public.
The default is Unknown.
Select the H.225/Q.931 Number Plan assigned to this entry.
•If you selected Unknown as the Number Type, you can select Unknown.
•If you selected Private as the Number Type, you can select Unknown, Level 0 Regional (Local), Level 1 Regional, Level 2 Regional or PISN Specific.
•If you selected E.164 Private as the Number Type, you can select Unknown, International, National, Network Specific, Subscriber or Abbreviated.
The default is Unknown.
If you are using a RadVision Gatekeeper, you must configure the Type of Number fields.
If you are using a CSE1k Gatekeeper, you are not required to configure the Type of
Number fields.
See Appendix A, “Numbering and Dial Plans,” on page 147, for more information about
the Type of Number fields.
6Click the Submit button to save your changes.
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7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
Configuring number manipulation for Digit Delivery
This number manipulation table defines the rules for changing the destination number received for
an incoming IP to Norstar call.
Digit delivery is a method used to automatically dial through to a target DN on the destination
Norstar system. You can configure the Norstar system to automatically answer certain lines. In this
configuration, the Norstar system provides a dial tone and waits for the caller to dial further digits,
as if they were a set on that system. These extra digits may be for a local telephone set or a local
PSTN call. If the destination digits received by the VoIP Gateway match a row of this table, the
digits will first be manipulated according to the rule in that row. They will then be dialed out after
the Norstar system answers the line. If a call does not match a row of this table, it is still connected
according to the IP to Hunt Group Routing Table. However, the Norstar system will not dial out
digits when the line is answered.
Note: The Digit Delivery Table feature uses auto answer trunks to implement dialing
through to the destination target line. In order to use the Digit Delivery Table feature,
DTMF tone receivers must be available on the target Norstar system. DTMF receivers are
provided with a Services Card, Combo Card or E&M trunk card.
To configure the number manipulation for Digit Delivery:
1Access the web interface.
2Click the Protocol Management button.
3Point to the Manipulation Tables tab and click the Digit Delivery Table link.
The
Digit Delivery Table screen appears.
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Figure 7 Digit Delivery Table screen
4To configure the first ten entries, click the Table Index drop list and click 1-10.
To configure the last ten entries, click the Table Index drop list and click 11-20.
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5Configure the Digit Delivery Table according to the following table.
Table 7 Digit Delivery Table
ParameterDescription
Called NumberEnter the number you want to match for this table entry.
When entering a number, you can create an entry that represents multiple numbers using
the following notation:
•[n-m] represents a range of numbers
•[n,m] represents multiple numbers
•x represents any single digit
•* represents any digit or multiple digits
•# represents the end of a number
•For example:
[5551200-5551300] represents all of the numbers from 5551200 to 5551300
[2221,2231,2241] represents the numbers 2221, 2231 and 2241
54324* represents any number that starts with 54324
54324xx represents a 7 digit number that starts with 54324
123[100-200]* represents numbers that starts with 123100 to numbers that start with
123200.
When the dialed number matches the number you enter here, the rules in this number
manipulation table entry are applied.
The VoIP Gateway matches the rules starting at the top of the table. For this reason, enter
more specific rules above more generic rules. For example, if you enter 551* in entry 1
and 55* in entry 2, the VoIP Gateway applies rule 1 to numbers that starts with 551 and
applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559.
However if you enter 55* in entry 1 and 551* in entry 2, the VoIP Gateway applies rule 1 to
all numbers that start with 55 including numbers that start with 551.
Num of stripped digitsEnter the number of digits that you want to remove from the destination phone number.
For example, if you enter 3 and the destination phone number is 5551234, the new
destination phone number is 1234.
Prefix to addEnter the number you want to add to the front of the stripped source phone number.
For example, if you enter 9 and the stripped source phone number is 1234, the new
source phone number is 91234.
The prefix can include digit and the following:
•D = wait for dial tone
You should always use D as the first prefix digit so that the VoIP Gateway will wait for a
Norstar dial tone before dialing out the digits. Dial tone detection can only be used as
the first prefix digit.
•P = pause
You can use the pause to wait for any secondary dial tone. This might be required for
dialing out through the Norstar to the PSTN. The pause waits for 1.5 seconds. Longer
periods may be achieved by using more than one P character. For example, PP will
wait for 3.0 seconds.
•other digits such as destination codes to access outside lines
Note: The wait for dial tone character is case sensitive. You must use a capital D to
indicate wait for dial tone.
6Click the Submit button to save your changes.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
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52 Configuring the Number Manipulation tables
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
Configuring number manipulation for Source Phone Numbers for IP
to Norstar calls
This number manipulation table defines the rules for changing the source number received in IP to
Norstar calls. A common use of this table is to alter the Calling Line Identification information that
is presented on the destination Norstar telephone.
To configure the number manipulation for Source Phone Numbers for IP to Norstar calls:
1Access the web interface.
2Click the Protocol Management button.
3Point to the Manipulation Tables tab and click the IP -> Tel Source Numbers link.
Source Phone Number Manipulation Table for IP->TEL calls screen appears.
The
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Configuring the Number Manipulation tables 53
Figure 8 Source Phone Number Manipulation Table for IP->TEL calls screen
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54 Configuring the Number Manipulation tables
4Configure the Phone Number Manipulation Table according to the following table.
Table 8 Source Phone Number Manipulation Table for IP->TEL calls (IP to Norstar)
ParameterDescription
PrefixEnter the number you want to match for this table entry.
When entering a number, you can create an entry that represents multiple numbers using
the following notation:
•[n-m] represents a range of numbers
•[n,m] represents multiple numbers
•x represents any single digit
•* represents any digit or multiple digits
•# represents the end of a number
•For example:
[5551200-5551300] represents all of the numbers from 5551200 to 5551300
[2221,2231,2241] represents the numbers 2221, 2231 and 2241
54324* represents any number that starts with 54324
54324xx represents a 7 digit number that starts with 54324
123[100-200]* represents numbers that starts with 123100 to numbers that start with
123200.
When the dialed number matches the number you enter here, the rules in this number
manipulation table entry are applied.
The VoIP Gateway matches the rules starting at the top of the table. For this reason, enter
more specific rules above more generic rules. For example, if you enter 551* in entry 1
and 55* in entry 2, the VoIP Gateway applies rule 1 to numbers that starts with 551 and
applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559.
However if you enter 55* in entry 1 and 551* in entry 2, the VoIP Gateway applies rule 1 to
all numbers that start with 55 including numbers that start with 551.
Num of stripped digitsEnter the number of digits that you want to remove from the source phone number.
For example, if you enter 3 and the source phone number is 5551234, the new source
phone number is 1234.
Prefix to addEnter the number you want to add to the front of the stripped destination phone number.
For example, if you enter 9 and the stripped destination phone number is 1234, the new
destination phone number is 91234.
5Click the Submit button to save your changes.
6To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
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Configuring the Number Manipulation tables 55
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
Configuring number manipulation for Source Phone Numbers for
Norstar to IP calls
This number manipulation table is used to manipulate or assign NPI / TON values to the source
number used for the call setup. Generally speaking, this table is not used since the source phone
number is a statically set parameter for the VoIP Gateway.
Source phone numbers are entered into the VoIP Gateway in the Channels – Hunt Group table. In
that table, a phone number can be entered for each port of the VoIP Gateway. This number may be
different from port to port, which is appropriate for a system that uses a key system style of line
appearances, or could be the same, which is appropriate for a system that uses the PBX system of
line pools.
The source number in an H.323 message is used to identify the caller for use in calling line ID
presentation. As such, manipulation of the source digits would normally only need to be done in
order to change the digits so that they will be in a more appropriate form for calling line ID.
For example, the VoIP Gateway may be setup so that it reflects the telephone number used in the
private dialing plan. Source manipulation can then be used to transform that private number into a
public dialing plan number that represents the normal PSTN number that would be used to dial the
location.
The source manipulation table may also be used to assign a specific NPI / TON classification to
the calling number. Setting an NPI / TON can be done globally or values used in this table can be
applied. Control for this is done using “Protocol Management / Protocol Definition / Source
Number Encoding Type”
To configure the
1Access the web interface.
2Click the Protocol Management button.
3Point to the Manipulation Tables tab and click the Tel -> IP Source Numbers link.
The Source Phone Number Manipulation Table for TEL->IP calls screen appears.
Source Phone Number Manipulation Table for TEL->IP calls:
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56 Configuring the Number Manipulation tables
Figure 9 Source Phone Number Manipulation Table for TEL->IP callsscreen
4To configure the first ten entries, click the Table Index drop list and click 1-10.
To configure the last ten entries, click the Table Index drop list and click 11-20.
5Configure the Phone Number Manipulation Table according to the following table.
Table 9 Source Phone Number Manipulation Table for TEL->IP calls (Norstar to IP)
ParameterDescription
PrefixEnter the number you want to match for this table entry.
When entering a number, you can create an entry that represents multiple numbers using
the following notation:
•[n-m] represents a range of numbers
•[n,m] represents multiple numbers
•x represents any single digit
•* represents any digit or multiple digits
•# represents the end of a number
•For example:
[5551200-5551300] represents all of the numbers from 5551200 to 5551300
[2221,2231,2241] represents the numbers 2221, 2231 and 2241
54324* represents any number that starts with 54324
54324xx represents a 7 digit number that starts with 54324
123[100-200]* represents numbers that starts with 123100 to numbers that start with
123200.
When the dialed number matches the number you enter here, the rules in this number
manipulation table entry are applied.
The VoIP Gateway matches the rules starting at the top of the table. For this reason, enter
more specific rules above more generic rules. For example, if you enter 551* in entry 1
and 55* in entry 2, the VoIP Gateway applies rule 1 to numbers that starts with 551 and
applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559.
However if you enter 55* in entry 1 and 551* in entry 2, the VoIP Gateway applies rule 1 to
all numbers that start with 55 including numbers that start with 551.
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Table 9 Source Phone Number Manipulation Table for TEL->IP calls (Norstar to IP) (Continued)
ParameterDescription
Num of stripped digitsEnter the number of digits that you want to remove from the source phone number.
For example, if you enter 3 and the source phone number is 5551234, the new source
phone number is 1234.
Type of NumberSelect the H.225/Q.931 Number Type assigned to this entry.
You can select Unknown, Private or E.164 Public.
The default is Unknown.
Select the H.225/Q.931 Number Plan assigned to this entry.
•If you selected Unknown as the Number Type, you can select Unknown.
•If you selected Private as the Number Type, you can select Unknown, Level 0 Regional (Local), Level 1 Regional, Level 2 Regional or PISN Specific.
•If you selected E.164 Private as the Number Type, you can select Unknown, International, National, Network Specific, Subscriber or Abbreviated.
The default is Unknown.
If you are using a RadVision Gatekeeper, you must configure the Type of Number fields.
If you are using a CSE1k Gatekeeper, you are not required to configure the Type of
Number fields.
See Appendix A, “Numbering and Dial Plans,” on page 147, for more information about
the Type of Number fields.
6Click the Submit button to save your changes.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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58 Configuring the Channels to Hunt Group parameters
Configuring the Channels to Hunt Group parameters
From the Channels - Hunt Group screen you can enable and assign telephone numbers to the VoIP
Gateway ports.
To configure the Channels to Hunt Group parameters:
1Access the web interface.
2Click the Protocol Management button.
3Click the Channels - Hunt Group link.
The Channels - Hunt Group screen appears.
Figure 10 End Point’s Phone Number table
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4Configure the End Point’s Phone Numbers according to the following table.
You must enter a number in the Phone Number boxes for each port that you want to use.
Table 10 End Point’s Phone Numbers
ParameterDescription
Configuring the Channels to Hunt Group parameters 59
Channel(s)The numbers in the Channel(s) boxes represent the ports on the back of the VoIP
Phone NumberIn each of the Phone Number boxes, enter the telephone number that is assigned to that
Hunt Group IdIn each of the Hunt Group Id boxes, enter the Hunt Group ID assigned to the channel. The
Gateway.
To enable a VoIP Gateway channel, you must enter the port number on this screen.
channel.
You must enter a telephone number for a channel to enable that channel.
If you want to disable a port, leave the Phone Number box for that channel blank.
The Phone Number you enter is the “source” number in the protocol. Also, if you are not
using the Digit Manipulation tables, the Phone Number you enter on this screen is
presented as the Calling Line Identification information on the destination set.
Recommendations:
If the analog loop start lines are configured as one dedicated line per Norstar set, enter a
unique phone number for each channel.
If the analog loop start lines are configured in a line pool, enter the same phone number
for all of the channels. The phone number you enter would typically be the main number
required to reach a receptionist.
If you assign unique phone numbers to channels configured in a line pool, the calling line
identification information would become confusing, since many Norstar sets may have
access to the same channel.
same Hunt Group ID can be used for more than one channel.
The Hunt Group ID is an optional field that is used to define a group of channels that are
used for routing IP to telephone calls. If an incoming call is assigned to a Hunt Group, the
call is routed to the channel or channels that correspond to the Hunt Group ID enter here.
In most situations, the application will need all 4 channels assigned to one hunt group.
Important: If you enter a Hunt Group Id, you must configure the IP to Hunt Group Routing
Table. If you do not configure the IP to Hunt Group Routing Table, calls will not complete.
For information about how to configure this table, refer to “Configuring the IP to Hunt
Group Routing” on page 67.
Note: Only the entries for the first four channels are supported on the VoIP Gateway.
5Click the Submit button to save your changes.
6To apply the changes, you must reset the VoIP Gateway. To reset the VoIP Gateway:
aClick the Reset button.
bClick the Restart button.
A confirmation dialog box appears.
cClick the OK button.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
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60 Configuring the Channels to Hunt Group parameters
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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Configuring the Automatic Dialing phone numbers 61
Configuring the Automatic Dialing phone numbers
The Automatic Dialing Table defines the telephone numbers that are automatically dialed when
this port is used. This table is used only if Automatic Dialing feature is enabled.
To configure the Automatic Dialing Table:
1Access the web interface.
2Click the Protocol Management button.
3Click the Automatic Dialing tab.
The Automatic Dialing Table screen appears.
Figure 11 Automatic Dialing Table
4In the Destination Phone Number box for a port, enter the telephone number to dial.
When this VoIP Gateway port is selected when making a call, the number in the Destination
Phone Number box is automatically dialed.
If you do not want to use this port for Automatic Dialing, leave the Destination Phone Number box blank.
5Repeat step 4 for each port you want to use for Automatic Dialing.
6Click the Submit button to save your changes.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
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62 Configuring the Automatic Dialing phone numbers
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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Configuring the Caller ID Display
The Caller ID Display Table contains the Caller ID information that is sent when a call is made
using the VoIP Gateway. The person receiving the call can use this information for caller
identification. The information on this table is sent in the H.225 Setup message sent to the remote
party.
To configure the Caller ID Display Table:
1Access the web interface.
2Click the Protocol Management button.
3Click the Caller ID tab.
The Caller Display Info screen appears.
Figure 12 Caller Display Info
Configuring the Caller ID Display 63
4In the Caller ID/Name box, enter the Caller ID string that sent when a call is made using this
VoIP Gateway port.
The caller ID string can contain up to 18 characters, but Nortel Networks recommends that
you do not exceed 16 characters to ensure proper display on Norstar telephones.
5Repeat step 4 for each VoIP Gateway port.
6Click the Submit button to save your changes.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
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64 Configuring the Caller ID Display
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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Configuring the Telephone to IP Routing 65
Configuring the Telephone to IP Routing
The IP Routing Table associates a called telephone number prefix with a destination IP address.
When a call is routed through the VoIP Gateway, the number dialed is compared to the list of
prefixes on the IP Routing Table. Calls that match one of the prefixes are sent to the corresponding
IP address. If the number dialed does not match one of the prefixes, the call is not made.
The Telephone to IP Routing table is also used for the IP Security feature. When the IP Security
feature is enabled, the VoIP Gateway accepts calls only if they match one of the prefixes entered in
the Telephone to IP Routing Table.
If you are using a Gatekeeper, the Gatekeeper matches the number dialed to the correct IP address.
When using a Gatekeeper, you do not need to configure the Telephone to IP Routing Table.
However, if you want to use fallback routing when communication with the Gatekeeper is lost,
you will need to configure the IP Routing Table.
To configure the IP Routing Table:
1Access the web interface.
2Click the Protocol Management button.
3Click the Tel to IP Routing tab.
The Tel to IP Routing & IP Security screen appears.
Figure 13 Phone to IP Routing Table
4In the Routing Index drop box, select the range of entries that you want to edit.
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66 Configuring the Telephone to IP Routing
5In the Destination Phone Prefix box, enter a prefix for this IP Routing Table entry.
The prefix can be 1 to 40 digits long. Any telephone number routed to the VoIP Gateway that
starts with these digits is sent to the IP Telephony gateway that has the IP Address entered in
the box to the right of this field.
Note: When entering a number, you can create an entry that represents multiple
numbers using the following notation:
•[n-m] represents a range of numbers
•[n,m] represents multiple numbers
•x represents any single digit
•* represents any digit or multiple digits
•# represents the end of a number
•For example:
[5551200-5551300] represents all of the numbers from 5551200 to 5551300
[2221,2231,2241] represents the numbers 2221, 2231 and 2241
54324* represents any number that starts with 54324
54324xx represents a 7 digit number that starts with 54324
123[100-200]* represents numbers that starts with 123100 to numbers that start
with 123200.
When the dialed number matches the number you enter here, the rules in this number
manipulation table entry are applied.
The VoIP Gateway matches the rules starting at the top of the table. For this reason,
enter more specific rules above more generic rules. For example, if you enter 551* in
entry 1 and 55* in entry 2, the VoIP Gateway applies rule 1 to numbers that starts with
551 and applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557,
558 and 559. However if you enter 55* in entry 1 and 551* in entry 2, the VoIP
Gateway applies rule 1 to all numbers that start with 55 including numbers that start
with 551.
6In the IP Address box, enter the IP address of the IP Telephony gateway to which calls that
match this entry are sent.
The IP Telephony gateway to which the calls are sent must be compatible with the VoIP
Gateway. Examples of compatible IP telephony gateways are: another VoIP Gateway,
Business Communications Manager and Meridian 1 IPT.
7Repeat steps 4 and 6, for each IP Routing Table entry.
Note: You can have maximum of 50 entries in the IP Routing Table.
8Click the Submit button to save your changes.
9To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
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Configuring the IP to Hunt Group Routing 67
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
Configuring the IP to Hunt Group Routing
The IP to Hunt Group Routing Table is used to route incoming IP calls to a group of channels. If an
incoming IP call matches one of the prefixes entered, the call is assigned to a Hunt Group. The call
is then sent to the VoIP Gateway channels assigned to answer that Hunt Group.
To use Hunt Groups you must also do the following.
•You must configure the Channel Select Mode parameter to determine the method in which
new calls are assigned to channels within the hunt groups. For information about how to
enable this option, refer to “Configuring the Protocol Definition parameters” on page 29.
•You must assign a Hunt Group ID to the VoIP Gateway channels on the End Point Phone
Number screen. For information about how to assign a Hunt Group ID to a channel, refer to
“Configuring the Channels to Hunt Group parameters” on page 58.
To configure the IP to Hunt Group Routing Table:
1Access the web interface.
2Click the Protocol Management button.
The H323 Gateway Parameters screen appears.
3Click the IP to Hunt GR Routing tab.
The IP to Hunt Group Routing Table screen appears.
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68 Configuring the IP to Hunt Group Routing
Figure 14 IP to Hunt Group Routing Table
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Configuring the IP to Hunt Group Routing 69
4In the Prefix box, enter a prefix for this Hunt Group Routing Table entry.
The prefix can be 1 to 7 digits long. Any telephone number routed to the VoIP Gateway that
starts with these digits is assigned to the Hunt Group entered in the box to the right of this
field.
Note: When entering a number, you can create an entry that represents multiple
numbers using the following notation:
•[n-m] represents a range of numbers
•[n,m] represents multiple numbers
•x represents any single digit
•* represents any digit or multiple digits
•# represents the end of a number
•For example:
[5551200-5551300] represents all of the numbers from 5551200 to 5551300
[2221,2231,2241] represents the numbers 2221, 2231 and 2241
54324* represents any number that starts with 54324
54324xx represents a 7 digit number that starts with 54324
123[100-200]* represents numbers that starts with 123100 to numbers that start
with 123200.
When the dialed number matches the number you enter here, the rules in this number
manipulation table entry are applied.
The VoIP Gateway matches the rules starting at the top of the table. For this reason,
enter more specific rules above more generic rules. For example, if you enter 551* in
entry 1 and 55* in entry 2, the VoIP Gateway applies rule 1 to numbers that starts with
551 and applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557,
558 and 559. However if you enter 55* in entry 1 and 551* in entry 2, the VoIP
Gateway applies rule 1 to all numbers that start with 55 including numbers that start
with 551.
5In the Hunt Group Id box, enter the Hunt Group ID number to which calls that match the
prefix are assigned.
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70 Configuring the IP to Hunt Group Routing
6Repeat steps 4 and 5, for each IP to Hunt Group Routing Table entry.
Recommendations:
If the Norstar is using a PBX configuration (i.e.: physical lines are auto answer and
sets have target line appearances), we recommend that all lines be associated with the
same hunt group.
One option is to configure a range of valid extensions for a Hunt Group ID. For
example: 967-0100 to 967-0200 are a block of numbers assigned to target lines that
terminate on various Norstar sets. All four channels should be configured as part of
the same Hunt Group where the valid DN range is between 967-0100 and 967-0200.
If the Norstar is using a square configuration (i.e.: physical lines appear and ring on
Norstar sets), we recommend that each line that has its own DN should have a
dedicated Hunt Group that allows only the specific dialed DN access to that channel.
For example: Dedicated lines are assigned to four people: line 1 is 967-0100, Line 2 is
967-0101, line 3 is 967-0102, and line 4 is 967-0103. In this example, four hunt
groups should be defined. Each Hunt Group has a single dedicated DN associated with
the channel that corresponds to the Norstar line.
7Click the Submit button to save your changes.
8To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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Configuring Call Forwarding
The VoIP Gateway allows you to forward incoming calls based on the VoIP Gateway port to which
the call is routed.
This feature uses H.450 protocol and should only be considered for networks that only use Norstar
VoIP Gateways. H.450 protocol is not supported on other Nortel Networks endpoints such as
Business Communications Manager and Meridian1 IPT.
To configure Call Forwarding:
1Access the web interface.
2Click the Protocol Management button.
3Click the Call Forward tab.
The Call Forward Table screen appears.
Figure 15 Call Forward Table
Configuring Call Forwarding 71
4Configure the Call Forward parameters for each port according to the table below.
Table 11 Call Forward parameters
ParameterDescription
Forward TypeSelect Immediate if you want to forward any incoming call to the Phone number specified.
Forward to Phone
Number
Time for No Reply
Forward
This option is equivalent to the Norstar feature Call Forward All Calls (CFAC).
Select Busy if you want to forward incoming calls when the VoIP Gateway port is busy.
This option is equivalent to the Norstar feature Call Forward on Busy (CFB).
Select No reply if you want to forward incoming calls that are not answered with the time
specified in the Time for No Reply Forward box.
This option is equivalent to the Norstar feature Call Forward No Answer (CFNA).
Enter the telephone number to which the call is forwarded.
If you have set the Forward Type for this port to no reply, enter the number of seconds the
VoIP Gateway waits before forwarding the call to the phone number specified.
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72 Configuring the Network Settings
5Click the Submit button to save your changes.
6To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
Configuring the Network Settings
From network settings page you can define the:
•IP settings including the gateway IP address and subnet mask
•Logging settings, such as IP address of SysLog Server
•SNMP settings
•RTP settings, including RTP base port and RTP Differentiated Services or IP TOS and
Precedence QOS parameters.
•Ethernet status
To change the network settings:
1Access the web interface.
2Click the Advanced Configuration button.
3Click the Network Settings tab.
The Network Settings screen appears.
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Figure 16 Network Settings
Configuring the Network Settings 73
4Configure the Network Settings according to the following table.
Table 12 Network Settings Parameters
ParameterDescription
IP AddressEnter the IP address of the VoIP Gateway.
Enter the IP address in dotted format notation, for example 192.10.1.255.
The IP address for the VoIP Gateway must be a static IP address.
If you do not know the IP address for the VoIP Gateway, contact your network
administrator.
Subnet MaskEnter the subnet mask of the VoIP Gateway.
Enter the subnet mask in dotted format notation, for example 255.255.255.0
If you do not know the subnet mask for the VoIP Gateway, contact your network
administrator.
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74 Configuring the Network Settings
Table 12 Network Settings Parameters (Continued)
ParameterDescription
Default Gateway Address If a Default Gateway is used on your network, enter the IP address of the Default Gateway
DNS Primary Server IPIf a DNS server is used on your network, enter the IP address of the primary DNS server.
DNS Secondary Server IPIf a secondary DNS server is used on your network, enter the IP address of the
NAT IP AddressIf the VoIP Gateway is behind a router with NAT enabled, enter the NAT IP address for the
Enable DHCPSelect Enable to enable DHCP support on the VoIP Gateway. When enabled the VoIP
SysLog Server IP
address
Enable SysLogSelect Yes to send the logs and error message generated by the VoIP Gateway to the
SNMP Manager IPEnter the IP address for the computer you are using as the SNMP Manager.
used by the VoIP Gateway.
Enter the IP address in dotted format notation, for example 192.10.1.255.
If you do not know the IP address for the Default Gateway, contact your network
administrator.
Enter the IP address in dotted format notation, for example 192.10.1.255.
A DNS server allows you to use domain names, such as www.NortelNetworks.com,
instead of IP addresses when you are accessing other locations on the Internet.
If you do not know the IP address of the primary DNS server you use on your network,
contact your network administrator.
secondary DNS server.
Enter the IP address in dotted format notation, for example 192.10.1.255.
The secondary DNS server is used if the Primary DNS server is not available or does not
have an entry for the domain name you entered.
If you do not know the IP address of the secondary DNS server you use on your network,
contact your network administrator.
VoIP Gateway.
Nortel Networks does not recommend using the VoIP Gateway behind a router that has
NAT enabled.
Gateway will attempt to get its IP address from the DHCP server.
Select Disable to disable DHCP support on the VoIP Gateway.
The default value is Disable.
Note: The VoIP Gateway requires a static IP address. If you enable DHCP, you must
ensure that a reserved IP address is assigned to the VoIP Gateway.
If you are using a SysLog Server, enter the IP address of the computer you are using to
run the SysLog Server.
The SysLog Server is an application designed to collect the logs and error messages
generated by the VoIP Gateway.
Enter the IP address in dotted format notation, for example 192.10.1.255.
SysLog Server. If you select Yes, you must enter an IP address in the SysLog Server IP
address box.
Select No to send the logs and error messages generated by the VoIP Gateway to the
RS-232 serial port instead of the SysLog Server.
The default value is No.Note: Due to the increase in network traffic caused by the SysLog messages, Nortel
Networks recommends that you do not Enable SysLog for normal operation. You should
enable SysLog only under the instructions of a trained Nortel Networks service
professional. Syslogs are only used by Nortel Networks support personnel to assist in
troubleshooting problems.
The SNMP Manager a device that is used for receiving SNMP Traps.
Enter the IP address in dotted format notation, for example 192.10.1.255.
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Configuring the Network Settings 75
Table 12 Network Settings Parameters (Continued)
ParameterDescription
Enable SNMPSelect Yes if you want to send SNMP traps to the SNMP Manager.
Select No if you do not want to send SNMP traps.
The default value is Yes.
RTP Base UDP PortEnter the base UDP port you want to use for RTP (voice), RTCP and T.38 channels. This
RTP IP Diff ServEnter the Diff Serv Code Point (DSCP) value that is assigned to the RTP packets.
RTP IP TOSEnter the value that is assigned to IP Type Of Service (TOS) field in the IP header for all
RTP IP PrecedenceEnter the value that is assigned to the IP Precedence field in the IP header for all RTP
Port 1 Duplex ModeShows the Duplex mode the Ethernet port is using.
Port 1 SpeedShows the speed, in MBits per second, that the Ethernet port is using.
is the lower boundary of UDP ports used by the VoIP Gateway.
The upper boundary is the Base UDP Port + 40.
The range of possible UDP ports is 1000 to 64000.
The default base UDP port is 4000. You should not change this parameter from its default
value.
The DSCP value is used by DiffServ compatible routers to prioritize how packets are sent.
By prioritizing packets, the DiffServ routers can minimize the transmission delays for time
sensitive packets such as VoIP packets.
DSCP values can be from 0 to 63.
The default value is 46, which enables priority handling by QoS routers.
If enter a Diff Serv value, this value is used instead of the RTP IP TOS and RTP IP
Precedence values.
RTP packets sent by the VoIP Gateway.
You can enter a value from 0 to 15.
The default value is 0.
packets sent by the VoIP Gateway.
You can enter a value from 0 to 7.
The default value is 0.
This is a read-only parameter.
This is a read-only parameter.
5Click the Submit button to save your changes.
6To apply the changes, you must reset the VoIP Gateway. To reset the VoIP Gateway:
aClick the Reset button.
bClick the Restart button.
A confirmation dialog box appears.
cClick the OK button.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
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76 Configuring the Channel Settings
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
Configuring the Channel Settings
The Channels Settings screen allows you to set the VoIP Gateway channel parameters, such as
Input and Output voice gain, Jitter buffer characteristics, Modem, Fax and DTMF transport modes.
These parameters apply to all VoIP Gateway ports.
Note: Nortel Networks recommends that you do not change any settings on the Channel
Settings screen except when instructed to do so by Nortel Networks service personnel.
Note: Set the Loss Plan setting for the Norstar analog loop start lines to MediumCO. You
set the Norstar Loss Plan settings using **CONFIG programming.
To configure the Channel Settings:
1Access the web interface.
2Click the Advanced Configuration button.
3Click the Channel Settings tab.
The Channels Settings screen appears
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Figure 17 Channel Settings
Configuring the Channel Settings 77
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78 Configuring the Channel Settings
4Configure the Channel Settings according to the following table.
Table 13 Channel Settings
ParameterDescription
Voice VolumeEnter the Voice gain control in dB. This parameter sets the level for the transmitted signal.
You can enter a value from -31 to 31 dB.
The default value is -6 dB.
Input Gain Enter the PCM input gain in dB. This parameter sets the level for the received signal.
You can enter a value from -31 to 31 dB.
The default value is 0 dB.
Silence SuppressionSelect On to turn on Silence Suppression.
Select Off to turn off Silence Suppression.
The default value is Off.
Silence Suppression is a method conserving bandwidth on VoIP calls by not sending
packets when silence is detected.
Echo CancellerSelect On to turn on Echo Cancellation.
Select Off to turn off Echo Cancellation.
The default value is On.
DTMF Transport TypeSelect DTMF Mute to erase the DTMF tones from voice stream and not relay the DTMF
MF Transport TypeSelect Mute MF to erase the MF tones from the voice stream and not relay the MF to the
DTMF Volume Enter the DTMF gain control value in dBms.
CAS Transport TypeSelect the CAS Transport Type you want the VoIP Gateway to use.
Fax Transport ModeSelect the Fax Transport Mode that the VoIP Gateway uses.
Caller ID Transport TypeSelect the Caller ID Transport Type you want the VoIP Gateway to use.
Caller ID TypeSelect Bellcore for this option.
to the remote end. DTMF will still be sent in out of band signaling if you select Yes for the
Enable DTMF over H.245 parameter on the Protocol Definition screen.
Select Proprietary DTMF Relay to erase the DTMF digits from voice stream and relay
the DTMF to the remote end.
Select DTMF Transparent to leave the DTMF tones in the voice stream.
Select RFC2833 Relay DTMF to erase the DTMF digits from voice stream and relay the
DTMF to the remote end using RFC 2833 standard.
The default value is DTMF Mute.
remote end.
Select Relay MF to erase the MF tones from the voice stream and relay the MF to the
remote end.
Select Transparent MF to leave the MF tones in the voice stream.
The default is Relay MF.
You can enter a value from -31 to 0 dBm.
The default value is -11 dBm.
You can select CAS Events Only or CAS RFC2833 Relay.
You can select Transparent, Relay or Bypass.
The default value is Relay.
You can select Disable, Bypass or Mute.
Bellcore is the Caller ID Type that is compatible with the Norstar KSU.
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Table 13 Channel Settings (Continued)
ParameterDescription
Configuring the Channel Settings 79
V21 Modem Transport
Ty pe
V22 Modem Transport
Ty pe
V23 Modem Transport
Ty pe
V32 Modem Transport
Ty pe
V34 Modem Transport
Ty pe
Fax Relay Redundancy
Depth
Fax Relay Enhanced
Redundancy Depth
Fax Relay ECM EnableSelect Disable if you do not want to use Error Correction Mode (ECM) mode during Fax
Fax Relay Max Rate Select the maximum rate, in bps, at which fax relay messages are transmitted.
Fax/Modem Bypass
Coder Type
Fax/Modem Bypass
Packing Factor
CNG Detector ModeSelect the Comfort Noise Generation (CNG) Detector Mode you want the VoIP Gateway
Select the V.21 Modem Transport Type that the VoIP Gateway uses.
You can select Transparent, Relay or Bypass.
The default value is Transparent.
Select the V.22 Modem Transport Type that the VoIP Gateway uses.
You can select Transparent, Relay or Bypass.
The default value is Bypass.
Select the V.23 Modem Transport Type that the VoIP Gateway uses.
You can select Transparent, Relay or Bypass.
The default value is Bypass.
Select the V.32 Modem Transport Type that the VoIP Gateway uses.
You can select Transparent, Relay or Bypass.
The default value is Bypass.
Note: This option applies to V.32 and V.32bis modems.
Select the V.34 Modem Transport Type that the VoIP Gateway uses.
You can select Transparent, Relay or Bypass.
The default value is Bypass.
Note: This option applies to V.34 and V.90 modems.
Enter the number of times that each fax relay payload is retransmitted to network.
You can enter a value from 0 to 2.
The default value is 0.
Note: This option is applicable only when T. 38 Protection Mode is set to Redundant
Packets.
Enter the number of times that control packets are retransmitted when using the T.38
standard.
You can enter a value from 0 to 4.
The default value is 0.
relay.
Select Enable if you want to use ECM mode during Fax Relay.
The default value is Enable.
You can select 2400, 4800, 7200, 9600, 12000, or 14400.
The default rate is 14400 bps.
Select the Codec you want the VoIP Gateway to use when performing fax/modem bypass.
Usually, high bit rate coders such as G.711 should be used.
The default codec is G.711 ALaw.
Enter the number of codec payloads are used to generate a Fax/Modem Bypass packet.
You can enter a value of 1, 2 or 3 codec payloads.
The default value is 1 codec payload.
to use.
You can select Disable, Relay or Event Only
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80 Configuring the Channel Settings
Table 13 Channel Settings (Continued)
ParameterDescription
Dynamic Jitter Buffer
Minimum Delay
Dynamic Jitter Buffer
Optimization Factor
RTP Redundancy DepthEnter 0 to disable the generation of redundant packets.
Packing FactorEnter the number of codec payloads are used to generate an RTP packet.
Basic RTP Packet
Interval
RTP Directional ControlSelect the RTP Directional Control you want the VoIP Gateway to use.
RFC 2833 Payload TypeEnter the RFC 2833 Payload Type.
RFC 2198 Payload TypeEnter the RFC 2198 Payload Type.
Fax Bypass Payload Type Enter the Fax Bypass Payload Type.
Enable RFC 3389 CN
Payload Type
Enable Answer DetectorSelect Enable to enable Answer Detector.
Answer Detector Activity
Delay
Answer Detector Silence
Time
Answer Detector
Redirection
Answer Detector
Sensitivity
Enable AGCSelect Enable to enable Automatic Gain Control (AGC).
AGC Gain SlopeEnter the AGC Gain Slope.
AGC RedirectionEnter the AGC Redirection value.
AGC Target EnergyEnter the AGC Target Energy value.
Enable Energy DetectorSelect Enable to enable Energy Detector.
Energy Detector Quality
Factor
Enter the minimum delay for the Dynamic Jitter Buffer.
You can enter a value from 0 to 150 milliseconds.
The default delay is 70 milliseconds.
Note: For more information about the Jitter Buffer, refer to “Jitter Buffer” on page 23.
Enter the Dynamic Jitter Buffer frame error to delay optimization factor.
You can enter a value from 0 to 12.
The default factor is 7.
Note: For more information about the Jitter Buffer, refer to “Jitter Buffer” on page 23.
Enter 1 to enable the generation of RFC 2198 redundancy packets.
The default value is 0.
You can enter a value of 1, 2 or 3 codec payloads.
The default value is 1 codec payload.
Select the RTP Packet Interval you want the VoIP Gateway to use for the codecs.
Select the Enable RFC 3389 CN Payload Type.
Select Disable to disable Answer Detector.
Enter the Answer Detector Activity Delay.
Enter the Answer Detector Silence Time.
Select Enable to enable Answer Detector Redirection.
Select Disable to disable Answer Detector Redirection.
Enter the Answer Detector Sensitivity.
Select Disable to disable AGC.
Select Disable to disable Energy Detector.
Enter the Energy Detector Quality Factor.
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Table 13 Channel Settings (Continued)
ParameterDescription
Configuring the Channel Settings 81
Energy Detector
Threshold
Enable Pattern DetectorSelect Enable to enable Pattern Detector.
Enter the Energy Detector Threshold.
Select Disable to disable Pattern Detector.
5Click the Submit button to save your changes.
6To apply the changes, you must reset the VoIP Gateway. To reset the VoIP Gateway:
aClick the Reset button.
bClick the Restart button.
A confirmation dialog box appears.
cClick the OK button.
7To save the changes so they are available after a power fail, you must save the changes to
non-volatile memory. To save the changes to non-volatile memory:
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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82 Configuring the VoIP Gateway time and date
Configuring the VoIP Gateway time and date
To configure the time and date on the VoIP Gateway:
1Access the web interface.
2Click the Advanced Configuration button.
3Click the Regional Settings tab.
The Regional Settings screen appears.
Figure 18 Regional Settings screen
4In the YYYY box, enter the current year (for example, 2003).
5In the MM box, enter the number for the current month (for example, enter 5 for May).
6In the DD box, enter the current date (for example, 25).
7In the HH box, enter the current hour using 24 hour clock, (for example, enter 14 for 2 p.m.).
8In the MM box, enter the current minutes.
9In the SS box, enter the current seconds.
10 Click the Set Time & Date button.
11 To save the changes so they are available after a power fail, you must save the changes to
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Note: The time you enter is the time at the location where the VoIP Gateway is
installed. Be sure to allow for the time difference if the VoIP Gateway is located in a
different time zone.
non-volatile memory. To save the changes to non-volatile memory:
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Configuring the VoIP Gateway time and date 83
aClick the Save Configuration button.
The Save Configuration to the system screen appears.
bClick the Save Configuration button that is in middle of the screen.
A confirmation message appears when the save is complete.
Note: The Save Configuration option is useful for trying out a new configuration.
Simply make the change and try out the new configuration. If you are happy with the
new configuration, save the changes to non-volatile memory. If you are not happy
with the new configuration, do not save the changes to non-volatile memory and
remove power from the VoIP Gateway. The VoIP Gateway will return to the previous
configuration.
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84 Configuring the VoIP Gateway time and date
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Chapter 4
Configuring the VoIP Gateway using the INI file
You can configure the VoIP Gateway using the web interface or by loading the configuration .INI
file. The configuration INI file is downloaded using the web interface.
To create an INI file, Nortel Networks recommends that you:
1Upload the INI file from the VoIP Gateway.
2Make the necessary changes.
3Download the resulting file to the VoIP Gateway.
This method preserves the programming that already exists, including the special default values
that have been entered when the unit was manufactured.
Retrieving the INI file from the VoIP Gateway
Before you can edit the INI file, you need to get the INI file from the VoIP Gateway.
85
1Access the web interface.
2Click the Advanced Configuration button.
3Click the Configuration File tab.
The Configuration File screen appears
Figure 19 Configuration File screen
4Click the Get INI FILE button.
5Navigate to the folder where you want to save the INI.
6Click the Save button.
The VoIP Gateway starts loading INI file into the folder you selected.
A message appears when the file transfer is completed.
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86 Loading the INI file on the VoIP Gateway
You can now edit the INI file using a standard text editor. The file that you create must have the
Windows extension “.ini”.
Loading the INI file on the VoIP Gateway
After you have finished editing the INI file, you need to load that file onto the VoIP Gateway. To
load the INI file:
1Access the web interface.
2Click the Advanced Configuration button.
3Click the Configuration File tab.
The Configuration File screen appears
Figure 20 Configuration File screen
4Click the Browse button.
5Navigate to the folder that contains the INI file you want to load.
6Click the file and click the Open button.
7Click the Send File button.
8To apply the changes, you must reset the VoIP Gateway. To reset the VoIP Gateway:
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The filename and path of the file appears in the box beside the Browse button.
aClick the Reset button.
bClick the Restart button.
A confirmation dialog box appears.
cClick the OK button.
Page 87
Contents of the INI file
The INI file contains the following information:
•“Basic, Logging and Web parameters”
•“Channel parameters”
•“H.323 Parameters”
All INI file data is downloaded at startup and stored in non-volatile memory. The provisioning
procedure should be used again only to modify VoIP Gateway parameters.
If a parameter is not specified in the INI file, values associated with that parameter are reset to a
default value. These values may not be the same as the values that were configured for the VoIP
Gateway at the time of manufacture. All parameters that are defined in the INI file contained in the
CD ROM or obtained by uploading the INI file from the VoIP Gateway should also be defined in
the user modified INI file in order for the VoIP Gateway to function correctly. You may alter these
parameters, but the parameter should be specifically defined. The easiest way to ensure this is to
upload the INI file from the VoIP Gateway to the computer, edit this file and then download the
resulting file.
Contents of the INI file 87
The Default Channel parameters are applied to all VoIP Gateway channels.
The Channel Parameters define the DTMF, Fax and Modem transfer modes. Refer to “DTMF, Fax
and Modem transport modes” on page 203.
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88 Contents of the INI file
Basic, Logging and Web parameters
Table 14 describes the parameters that are found in the Basic, Logging and Web parameters section
of the INI file.
Table 14 Basic and Logging parameters
INI
File Field Name
MGControlProtocolType4 = for H.323 gateway
DSPVersionTemplateNumber0 = Firmware DSP version supports PCM/ADPCM, G723 and G729 Coders
EnableLanWatchDog0 - Disable LAN WatchDog (default)
SysLogServerIPIP address in dotted format notation, e.g., ‘192.10.1.255’
Valid Range and Description
(default)
1 = Firmware DSP version supports PCM/ADPCM, and NetCoder coders
2 = Same as "0" but with voice and energy detectors
3 = Same as "1" but with voice and energy detectors
DSP templates 2 or 3 should be selected for Disconnect supervision feature,
enabled by: "EnableSilenceDisconnect = 1".
1 = 10 Base-T full-duplex
2 = 100 Base-T half-duplex
3 = 100 Base-T full-duplex
4 = auto-negotiate (Default)
Auto-negotiate falls back to half-duplex mode (HD) when the opposite port is not
in auto-negotiate, but the speed (10 Base-T, 100 Base -T) in this mode is always
configured correctly.
1 – Enable
After the gateway is powered up it will try first to communicate with BootP server.
If BootP server is not responding and "DHCPEnable =1" the gateway will send
DHCP request to configure its IP address and other network parameters from
the enterprise DHCP server.
Note: BootPRetries parameter becomes active after the VoIP Gateway is reset
INI
file is loaded. To change the parameters, first modify the
and
then reset the gateway.
1 = Perform diagnostics
1 = Enable gateway’s watch dog (default)
1 - Enable LAN WatchDog
When enabled, the VoIP Gateway is restarted if a LAN failure is detected.
INI
file, and
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Table 14 Basic and Logging parameters (Continued)
INI
File Field Name
Valid Range and Description
Contents of the INI file 89
EnableSyslog0 = Disable SysLog (default)
DisableRS2320 – RS-232 serial port is enabled (default)
LoggerFormat0 – name + msg
DisableWebTask0 – Enable Web management (default)
ResetWebPassword Allows resetting the default of Web password to:
Disable WebConfig 0 = Enable changing parameters from Web (default)
SNMPManagerIPIP address of SNMP Manager. The SNMP manager is used for receiving SNMP
DisableSNMP0 – SNMP is enabled (default)
HTTPportHTTP port used for Web management (default = 80).
1 = Enable SysLog
If SysLog is disabled all Logs and error messages are sent to RS-232 serial port
if “DisableRS232 = 0”
1 – RS-232 serial port is disabled
To enable sending of all log and error messages to the RS-232 serial port,
define:
“EnableSyslog = 0” and “DisableRS232 = 0”
1 – time + msg
2 – name + time + msg
3 – SysLog prefix + msg (default)
1 – Disable Web management
Username: “Admin”
Password: “Admin”
1 = Operate Web server in “read only” mode
Tr ap s .
For example: SNMPManagerIP = 10.2.1.10
1 – SNMP is disabled
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90 Contents of the INI file
Channel parameters
Table 15 describes the parameters that are found in the Channel Parameters section of the INI file.
Note: Nortel Networks recommends that you do not change any of the Channel
parameters except when instructed to do so by Nortel Networks service personnel.
Table 15 Channel parameters
INI
File Field NameValid Range and Description
DJBufMinDelay 0 to 150 msec (default = 70)
DJBufOptFactor0 to 12 (default = 7)
BaseUDPPortRange 1000-64000 (default 4000)
ECHybridLoss0 = 6 dB (default)
FaxModemBypassM1, 2 (default = 1)
FaxModemRelayVolume -18 to -3, corresponding to -18 dBm to -3 dBm in 1 dB steps. (Default = -12 dBm)
FaxRelayECMEnable0 = Disable using ECM mode during Fax Relay
FaxRelayEnhanced
RedundancyDepth
FaxRelayRedundancyDepth0 to 2 (default =0)
FaxRelayMaxRateLimits the maximum rate at which fax messages are transmitted.
FaxTransportModeSets the Fax transport
UseT38orFRF110 = Use proprietary FRF.11 syntax to send/receive fax relay.
The lower boundary of the UDP ports used by the VoIP Gateway for RTP, RTCP
and T.38 channels.
The upper boundary is the BaseUDPPort + 10*(number of gateway’s channels).
1 = 9 dB
2 = 0 dB
3 = 3 dB
Sets the four wire to two wire worst case Hybrid loss, the ratio between the
signal level sent to the hybrid and the echo level returning from the hybrid.
Number of 20 msec payloads to be used for generating one RTP fax/modem
bypass packet.
Fax gain control.
1 = Enable using ECM mode during Fax Relay. (default)
0 to 4 (default =0)
Number of repetitions applied to control packets when using T.38 standard.
Number of repetitions to be applied to each fax relay payload when transmitting
to network (applicable only when T38ProtectionMode = 0).
1 = Use Forward Error Correction (FEC) algorithm to protect T.38 fax relay
stream (isn’t implemented)
DTMF gain control.
1 = erase digit from voice stream, relay to remote. (Default)
2 = digits remains in voice stream.
3 = erase digit from voice stream, relay to remote according to RFC 2833
standard
7 = digits will be sent using RFC 2833 standard, but received RFC 2833 digits
will be muted from audio
1 = erase MFs from voice transport, relay to remote. (Default)
2 = MFs are not erased from voice, not relayed to remote.
Range: 96 to 99, 106 to 127; Default = 96
The 100, 102 to 105 range is allocated for proprietary usage.
Cisco™ are using payload type 101 for RFC 2833.
The same payload type should be used for receive and for transmit.
input gain.
1 = Enable generation of RFC 2198 redundancy packets.
(Default = 1 dB). Voice gain control
1 = silence compression enabled
Norstar VoIP Gateway Configuration Guide
Page 92
92 Contents of the INI file
Table 15 Channel parameters (Continued)
INI
File Field NameValid Range and Description
ECE0 = Echo Canceler disabled
IPDiffServ0 - 63 value for setting the Diff Services Code Point (DSCP).
IPPrecedence0 to 7 (default 0)
IPTOS0 to 15 (default 0)
FlashHookPeriod300 to 1500 (default 700) Flash Hook time in msec. The parameter is used for
MinimumFlashHookTime25-300, (default = 250)
DTMFDetectionPoint0 = DTMF event is reported when button is pressed
DTMFDigitLengthTime in msec for generating DTMF to PSTN side.
DTMFInterDigitIntervalTime in msec between generated DTMFs to PSTN side
CallerIDType0 = Bellcore GR-30-CORE Type 1(default)
If defined it will override the IP TOS and IP Precedence settings.
Applies only to RTP packets.
Sets the value of the IP precedence field in the IP header for all RTP packets
Sets the value of the IP Type Of Service field in the IP header for all RTP packets
Flash Hook detection and for Flash Hook generation.
Minimum threshold in msec + 50msec for detection of HookFlash.
1 = DTMF event is reported on button release (default)
The parameter is used for out of band dialing (H.245 user input message or
H.225 keypad facility)
Default = 100 msec
Default = 100 msec
1 = ETSI Type 1
Both Bellcore and ETSI Type 1 Caller ID signals are generated/detected
between first and second rings.
1 = PCMLoopback, loopback the incoming PCM to the outgoing PCM.
2 = ToneInjection, generates a 1000 Hz tone to outgoing PCM.
3 = NoLoopback, (default).
P0606298 02
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Contents of the INI file 93
H.323 Parameters
Table 16 describes the parameters that are found in the H.323 Parameters section of the INI file.
Table 16 H.323 Parameters
INI
File Field NameValid Range and Description
GatewayVersionVersion of VoIP Gateway, for example “GatewayVersion = 4.0 GA”.
StaticNatIPStatic NAT IP address.
Global gateway IP address. Should be defined if static Network Address
Translation device (NAT) is used between the gateway and the Internet.
IsGatekeeperUsed0 = no Gatekeeper used (internal phones table used)
GatekeeperIpIP address of the Gatekeeper
IsRedundantGKUsed0 = No redundancy
GWRegistrTypeThe Gateway Registration Type defines the encoding type of the VoIP Gateways
1 = Gatekeeper used (default)
Used if IsGatekeeperUsed = 1
GatekeeperIP parameter can contain up to three IP addresses.
In example below, there are 2 redundant gatekeepers in addition the first
gatekeeper.
GatekeeperIP = 10.2.37.10
GatekeeperIP = 10.2.37.11
GatekeeperIP = 10.2.37.12
To enable the use of redundant gatekeeper option, select
“IsRedundantGKUsed = 1”.
1 = Allow gatekeeper redundancy, according to GatekeeperIP addresses.
If redundancy is used, VoIP Gateway tries to communicate with redundant
Gatekeepers, if there is no response from the current Gatekeeper. If a new
Gatekeeper is found the VoIP Gateway stays working with it, until the next failure.
phone numbers that are used when the VoIP Gateway registers these numbers
with a Gatekeeper.
Select 0 (E.164) if you want the VoIP Gateway to use e164 encoding and not use
the encoding type defined in the table
Select 1 (H.323-ID) if you want the VoIP Gateway to add the H323-ID to the
destination information.
Select 3 (Table Value registration) if you want the VoIP Gateway to use the
values configured in the Gateway Registration Table.
If GWRegistrType=1, 2 or 4, you must configure the H323IDString.
Norstar VoIP Gateway Configuration Guide
Page 94
94 Contents of the INI file
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
RegistrationPrefixThe registration prefix is used to register a VoIP Gateway to a Gatekeeper. The
H323IDStringDefinition of Gateway H.323-ID for registration to Gatekeeper. In the case that a
SourceEncodeModeThe Source Number Encoding Type defines the encoding type of the calling
DestEncodeModeThe Destination Number Encoding Type defines the encoding type of the called
RegistrationTimeTime in seconds between registrations to the Gatekeeper (default 60)
parameter can be repeated up to 10 times.
Registration Prefix can contain single value or contain a number range, such as:
RegistrationPrefix = 123,9,0
RegistrationPrefix = 10-30,9,1
Plan,Type values:
0,0 - Unknown, Unknown
9,0 - Private Unknown
9,1 - Private Level 2 Regional
9,2 - Private Level 1 Regional
9,3 - Private PISN Specific
9,4 - Private Level 0 Regional (local)
1,1 - Public International
1,2 - Public National
1,3 - Public Network Specific
1,4 - Public Subscriber
1,6 - Public Abbreviated
H323IDString exists in configuration and IsGatekeeperUsed=1, the VoIP
Gateway will send a registration to the Gatekeeper with this H323-ID string.
This string can be up to 19 characters.
phone number in H.225 setup messages.
0 - E.164 (default)
1 - H.323-ID
2 - E.164 and H.323-ID
3 - Table Values
4 - Table Values and H.323-ID registration
Select 0 (E.164) if you want the VoIP Gateway to use e164 source encoding and
not use the encoding type defined in the table
Select 1 (H.323-ID) if you want the VoIP Gateway to add the H323-ID to the
source information.
Select 3 (Table Values) if you want the VoIP Gateway to use the values
configured in the Tel -> IP Source Number manipulation table.
If SourceEncodeMode=1, 2 or 4, you must configure the H323IDString.
phone number in H.225 setup messages.
0 - E.164 (default)
Select 0 (E.164) if you want the VoIP Gateway to use e164 destination encoding
and will not use the encoding type defined in the table
Select 1 (H.323-ID) if you want the VoIP Gateway to add the H323-ID to the
destination information.
Select 3 (Table Values) if you want the VoIP Gateway to use the values
configured in the Tel -> IP Dest Number manipulation table.
If DestEncodeMode=1, 2 or 4, you must configure the H323IDString.
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Page 95
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
Contents of the INI file 95
ResponseTimeOutTime in seconds, the VoIP Gateway waits for RAS response from the
MaxRetriesNumber of retransmissions of RAS message to the Gatekeeper, before the VoIP
TimeBetweenGKsLoopsTime in seconds before the VoIP Gateway tries again to go through the
IsGKFallbackUsed0 = Fallback is not used (default)
CanMapAliasesIf enabled, the Gatekeeper can change the gateway destination number, using
IsAutoDiscovery0 = no auto discovery of Gatekeeper (default)
IsTerminal0 = Standard Gateway behavior (default)
IsTunnelingUsed0 = no tunneling (default)
EnableRAI0 - Disable RAI messages
RAIHighThreshold0 - 100,
RAILowThreshold0 - 100,
RAILoopTimeThe VoIP Gateway checks for resource availability each RAILoopTime period
IsFastConnectUsed0 = normal connect (default)
Gatekeeper, before message retransmission (default 2).
Gateway decides that the Gatekeeper is not responding (default 2).
If “IsRedundantGKUsed = 1”, the VoIP Gateway tries to use one of the
redundant Gatekeepers from its list.
Gatekeeper list (default 60).
1 = If the Gatekeeper(s) does not respond, use internal Phone to IP table for call
routing.
The VoIP Gateway leaves the fallback mode as soon as it again finds the
Gatekeeper.
the Alias parameter in the ACF message.
Gatekeeper Auto Discovery requires multicast and is not supported
1 = Gateway imitates an H.323 terminal with up to 4 aliases.
In all Gateway messages the “terminal type” value is set to “terminal”
1 = tunneling
1 - If the VoIP Gateway resources are below the threshold, an almost out of
resources RAI message will be sent to the Gatekeeper.
Usually RAI will be applicable if the VoIP Gateway is configured to use Hunt
groups.
"Out of resources" gateway threshold, in percentage [default = 90%]
When number of VoIP Gateway busy endpoints [%} is above the High Threshold,
the VoIP Gateway will send a RAI message with "almostOutOfResources =
TRUE "
The Gateway Resources [%] are calculated as the number of busy endpoints
divided by total available gateway endpoints.
"Out of resources" gateway threshold, in percentage [default = 90%]
When the number of VoIP Gateway busy endpoints [%] is below the Low
Threshold, the VoIP Gateway will send a RAI message with
"almostOutOfResources = FALSE "
[sec] (default 10sec).
1 = fast connect
Norstar VoIP Gateway Configuration Guide
Page 96
96 Contents of the INI file
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
CoderNameDefines the coder used. Enter in the format: Coder name,ptime.
The Coder name can be one of the following:
g711Ulaw64k- G.711 u-law
Note1: If not specified, the ptime gets a default value.
This parameter can appear up to 5 times. If several coders are used,
IsMSAlgorithmOn should be set to 1 for Normal connect procedure, otherwise,
only the first coder is used.
ChannelListList of phone numbers for the VoIP Gateway channels
Channel2PhonePhone number of channel.
TrunkGroup_xa-b,c or just a,c
EnableHold0 - Disable Hold
a, b, c
a = first channel
b = number of channels starting from “a”
c = the phone number of the first channel
example: ChannelList = 0,4,101
Defines phone numbers 101 to 104 for up to 4 VoIP Gateway channels.
INI
file can include up to ten “ChannelList = “ entries
The
The “ChannelList = “ can be used instead or in addition to Channel2Phone
parameter.
Its format: Channel2Phone = “<channel>, <number>”
<channel> is 0...23.
Example: “Channel2Phone = 0, 1002”
Appears 4 times.
x - Hunt group ID, starting with 1
a - Starting port (from 1)
b - Ending port (up to 4)
c - phone number allocated for the first port
For example:
TrunkGroup_1 = 1-3,100
TrunkGroup_2 = 4,200
The Trunk Group can be used instead of ChannelList and Channel2Phone
parameters.
1 - Enable Hold service (H.450.4)
If enabled the Hold (or unhold) will be activated at the VoIP Gateway by a
HookFlash. On receiving a Hold request, the remote party will be put on hold,
and will play hold tone to the telephone port.
P0606298 02
Page 97
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
Contents of the INI file 97
EnableTransfer0 - Disable Call Transfer (default)
EnableForward0 - Disable call forward (default)
ForwardInfoXa,b,c
IsFaxUsed0 = No
DetFaxOnAnswerTone0 = Receiving Gateway initiates T.38 on receiving HDLC preamble signal from
ChannelToFax#Type of H.323 (annex D) FAX channel
PrefixMapping phone number to IP address, using phone number prefix
PSTNPrefixa,b
IsDialNeeded0 = no dial needed (automatic dialing)
1 - Enable call transfer service (H.450.2)
A transfer is initiated at the VoIP Gateway by using a HookFlash. If this service is
enabled, the remote party will perform the call transfer.
1 - Enable call forward service (H.450.3).
If you enable Call Forward, a "Call forward table" must be used to define call
forward modes (see the ForwardInfoX parameter).
x - Gateway port number (1 to 24)
a - Call forward mode
0 - not in use
1 - On busy
2 - Immediate forward
3 - No reply for time "c"
b - Forwarded number
c - Timeout [sec] for no reply
Incoming IP to Telephone call will be forwarded to "b" number on one of the
conditions: busy, immediate or no reply.
1 = Fax is send/received using H.323 Annex D T.38 procedure.
Fax (default)
1 = Receiving GW initiates T.38 on receiving of CED answer tone from Fax.
0 = Voice only
1 = Fax only (can be used only with Fast Start or Tunneling)
2 = Voice + Fax (default)
Example: ChannelToFax3 = 1 (use port ‘3’ only for Fax)
‘IsFaxUsed’ parameter should be set to ‘1’ in order to use the H.323 Fax
according to Annex D T.38.
When channel is set to 2 (Voice + Fax), at beginning of a call, the logical channel
is opened as Voice channel, and after Fax detection, the channel is reopened as
Fax T.38 channel.
This parameter can appear up to 4 times.
Example: Prefix = 20,10.2.10.2
Any dialed number that starts with “20” is routed to IP address “10.2.10.2”.
This parameter is needed when a Gatekeeper is not used.
Can appear up to 20 times. Maximal prefix size is 7 digits
a - destination number prefix (for IP to telephone calls)
b - Hunt group ID
Outgoing IP to Telephone calls starting with "a" will be routed to Hunt group "b".
1 = dial needed (default)
If 0 = TargetOfChannel parameters define the automatic dialed number.
If "DialisNeeded =1" the VoIP Gateway seizes the line (after detecting the ringing
signal), plays a dial tone, collects DTMF digits and sends Setup to IP
destination.
Norstar VoIP Gateway Configuration Guide
Page 98
98 Contents of the INI file
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
TargetOfChannel###Automatic dialed phone number.
IsUseFreeChannel0 = Select the channel according to destination phone number received in H.225
IsTwoStageDial0 = One stage dialing
IsWaitForDialTone0 = don’t wait for dial tone
FXOWaitForDialTimeDialing timeout [in msec], after seizing the line and before start dialing. (Default
EnableSilenceDisconnectIf enabled, the VoIP Gateway will disconnect the call if silence is detected for
FarEndDisconnectPeriodDuration of Silence period [in seconds] for call disconnection. (Default = 120)
MaxDigits2 to 19 (default 4). Maximum number of digits that can be dialed.
TimeBetweenDigits0 to 5 (default 4) Inter-digit timeout in seconds, used to terminate dialed
IsSETUPIncludeNum0 = Send an empty SETUP (or GK ARQ) message, without called party number.
The automatic dialed number, used if OFF HOOK or ringing signal is detected.
Applicable when IsDialNeeded = 0.
Its format: “TargetOfChannel<channel> = <number>”.
Example: “TargetOfChannel1 = 123”
The parameter, if used, should be defined per port (channel)
setup message, (default)
1 = Select the next available channel
Used for IPÆ VoIP Gateway calls
The next available channel is selected, out of the gateway channels defined in
‘ChannelList’ and Channel2Phone’ parameters.
When using one stage dialing, (‘IsTwoStageDial =0’), ‘IsUseFreeChannel’ should
be equal to ‘1’.
For one stage dialing the VoIP Gateway selects the next free channel, and dials
into the line the destination phone number received in Setup message.
1 = Two Stage Dialing (default)
Used for IP Æ VoIP Gateway calls
For ‘Two Stage Dialing the VoIP Gateway seizes the PSTN/PBX line, without
performing any dial, the remote User is connected over IP to PSTN/PBX, and all
further signaling (dialing and call progress tones) is done directly with the PBX
without gateway intervention.
For ‘One Stage Dialing’ VoIP Gateway seizes the next available channel
(‘IsUseFreeChannel’ should be ‘1’), and dials the destination phone number
received in Setup message. Use the ‘IsWaitForDialTone’ parameter to specify
whether the dialing should come after detection of dial tone, or immediately after
seizing of the line.
1 = Wait for dial tone (default)
Used for ‘One Stage Dialing’.
If IsWaitForDialTone = 0, VoIP Gateway dials phone number immediately after
seizing the PSTN/PBX line, without ‘listening’ to dial tone.
If IsWaitForDialTone = 1, VoIP Gateway dials phone number only after it detects
a dial tone (it can take 3-5 sec to detect a dial tone).
The correct dial tone parameters should be configured in call progress tone file.
=1000)
Applicable for single stage dialing.
120seconds (default)
Applicable for VoIP Gateways that are using DSP template = 3 or 4.
Dialing ends when maximum number of digits dialed or timeout between digits
expired (TimeBetweenDigits parameter), or ‘#’ is dialed.
numbers.
1 = Do not send an empty setup message (default)
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Page 99
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
Contents of the INI file 99
DefaultNumberPhone number
IsH323IDUsed0 = The VoIP Gateway encodes the calling and called phone number into e.164
EnableQ931Cause0 - H.225 Release reason will be sent in the Release Complete H.323 message
IsSetupAckUsed0 = On receiving of Setup Ack message, VoIP Gateway releases the Call
OpenH245onFS0 = Don’t open H.245 channel of Fast Start (default)
IsDTMFUsed0 = not used (default)
IsHookFlashUsed0 = not used (default)
Is931MsgUsed0 = not used (default)
IsSpecialDigits0 = not used (Default)
VoIP Gateway dials this number if the SETUP message is received with no
phone number.
The DefaultNumber must be one of the phone numbers of the end points.
field (default)
1 = The VoIP Gateway encodes the destination phone number in e.164 and
source phone number in H.323-ID fields.
2 = The VoIP Gateway encodes the source phone number in e.164 and
destination phone number in H.323-ID fields.
3 = The VoIP Gateway encodes the calling and called phone number into
H.323-ID field.
(default)
1 - Q.931 cause will be send in Release Complete message
(default)
1 = On receiving of Setup Ack message, the call is not released.
When working in non-overlap mode (‘IsOverlapUsed = 0’), use this parameter to
enable receiving of Setup Ack messages.
This parameter is used for specific, non-standard applications. Usually when
working in non-overlap mode, the Setup Ack messages are not sent.
1 = H.245 channel is opened immediately after Fast Start connection is
established.
Opening of H.245 channel may be needed for relaying DTMF digits over H.245
channel during a call.
1 = DTMF digits are sent over H.245 channel using Facility message
Incoming facility messages of out-of-band DTMF is played into FXS or FXO
channels.
For using out of band DTMF, the User must disable inband DTMF by setting:
‘DTMFTransportType = 0’ (erase digit and don’t relay).
Note that this feature is not supported when using Fast Start, unless other party
opens H.245 channel.
1 = Send H.245 User Input Indication message, when a hookflash is detected
The received facility message with a HookFlash signal generates HookFlash at
the channel
1 = DTMF digits and HookFlash signal is sent in H.225/Q.931 info message
(keypad facility)
Note that incoming Q.931 DTMF digits and HookFlash are always ignored.
To use H.225/Q.931 messages, set the following parameters:
‘IsDTMFUsed = 0’ and ‘IsOverlapUsed = 0’
1 = Allows user to dial “*” and “#”, and allows endpoint telephone number to
contain such digits.
Notes:
a. When this feature is activated, “#” can not be used to end the dialed number.
b. # and * can be used as first digit of dialed number, disregard the value of
"IsSpecialDigits" parameter.
Norstar VoIP Gateway Configuration Guide
Page 100
100 Contents of the INI file
Table 16 H.323 Parameters (Continued)
INI
File Field NameValid Range and Description
IsFSOpenMediaOn
Connect
IsFSMediaInfoSendOn
Connect
IsFSRemoteRBTone0 = Originator of Fast Start call plays local ringback tone, until it receives
PlayRBTone2Tel0 - Don't play for fast start
PlayRBTone2IP0 - Don't play Ringback to IP (default)
ProgressIndicator2IP0 - No PI will be sent in H.225 messages
CallerDisplayInfo#Caller DisplayInfo table is used to send Caller Identification information per VoIP
EnableCallerID0 = Don’t send CallerID signal to VoIP Gateway port (default)
H225ListenPortH.225 TCP listen port (default 1720)
0 = Voice channel is open after sending an Alert message (default)
1 = Voice channel is opened after sending a Connect message
After receiving the Setup message, the VoIP Gateway can open a voice channel
immediately, or it can wait until the call is answered and the Connect message is
sent.
Note that this is only relevant if Fast Start or Tunneling are used.
0 = Fast Start Structure response is sent in Alert message (default)
1 = Fast Start Structure response is sent in Connect message
After receiving Fast Start Setup message, the VoIP Gateway should reply with
H.225 message that includes media information structure. It can be sent with
Alert or Connect messages. Sending this information in Alert message enables
the remote side to open the voice channel before receiving the Connect
message.
Note that this is only relevant if Fast Start is used.
Connect message (default)
1 = If the received Alert message contains media information structure, the VoIP
Gateway opens voice channel, without playing any local tone to the channel. It is
the remote gateway’s responsibility to play the ringback tone over IP.
If the Alert message does not contain a media information structure, the voice
channel is not opened and a local ringback tone is generated, until a Connect
message is received.
Note that this is only relevant if Fast Start is used.
1 - Play
2 - Play according to Progress Indicator (PI). The gateway will not play local
ringback tone if PI=1 or PI=8 was received in H.225 Alert message. (default)
1 - Play Ringback tone to IP, and set PI=8 in H.225 Alert message
1 - PI = 1 will be sent to IP in H.225 Alert
8 - PI = 8 will be sent to IP in H.225 Alert
If not configured at all (default), the PI sent in the H.225 messages will be set
according to "Play Ringback tone to IP" parameter.
Gateway port to remote IP terminal.
This parameter can appear up to 4 times, with up to 18 characters per string.
To ensure proper display on the Norstar telephones, use a maximum of 16
characters per string.
1 = Calling number and Display text is sent to the VoIP Gateway port, between
the first and second rings.
In the VoIP Gateway, if "EnableCallerID=1", the Caller ID signal will be detected
and sent to IP in a H.323 Setup message (as "Display" element).
The VoIP Gateway expects to receive H.225/Setup message on this port
P0606298 02
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