Starplus Triad Xts Digital Key Telephone System System Programming And Operation Manual

							ProgrammingD-11
Appendix D - Voice Over the Internet Protocol
Setting Up the CO Line Groups (FLASH 40)
The CO Line Group must be programmed for the Discovery VoIP CO lines. In addition, a TIE 
circuit must be established to provide system-to-system connectivity.
1. While in programming mode, dial [FLASH 40].
2. Dial the appropriate range that equals the CO time slots used (e.g., 001-008 or 006-013), 
then press HOLD.
3. Press Button 8, enter an available Line Group (01-23) based on your existing system 
programming, then press HOLD.
4. Press Button 20 (Page B), then press Button 1.
5. Dial [5], then press HOLD.
Networking Tables (FLASH 16)
Networking Tables identify the system number (01-16), the CO or line Group (00-23), the 
range of station numbers (from xxxx to xxxx), and the IP address for each system. The system 
number is a database identifier. The CO or line group identifies the line group the system 
should access when the associated extension number is dialed. The extension range defines 
the extensions that can be reached at the associated remote system. The IP address provides 
the IP information of the remote system that is associated to the extension range, CO or line 
group, and the system number.
When a specific extension is dialed, the system looks up the extension in the networking 
table to determine the line group to use. The line group is then accessed and the Discovery 
VoIP card establishes the IP connection to the remote system using the associated IP address.
1. Enter programming, then dial [FLASH 16].
2. Enter the system to be modified (01 to 16).
The Discovery VOIP card uses 8 CO time slots.
Unused COs on the 
Discovery VoIP card do not have to be placed in Line Group 00 (e.g., a 
4-port VoIP card takes 8 CO time slots, but only 4 CO lines are used) since the software 
disables unavailable VoIP ports automatically.
T1/ISDN SIGNAL TYPE 0-7
TIE
Make sure that your 4-digit dialing plans on the remote systems do not conflict with your local 
dialing plan. If there is a conflict, users will be required to access the VoIP line group first and then 
dial the remote extension. Otherwise, dialing the 4-digit extension directly will connect the caller 
to a local system extension and not the remote system extension. For example, if System A uses 
extensions 1000-1099 and System B uses 1000-1099, a user on either system could not dial the 
remote system directly because local system extensions take precedence over remote system 
extensions. In this case, a user would have to access the 
Discovery VoIP line group and dial the 
remote system extension to bypass the local system’s dialing plan.
The 
Discovery VoIP CO’s default into Line Group 1. The line group may need to be changed for 
proper operation on your particular system. Refer to the FLASH 40 programming section for 
more information on changing line groups. 
						

							D-12Programming
Appendix D - Voice Over the Internet Protocol
3. Press the appropriate button listed below to modify the table.
Conditions
»  Feature access codes must not conflict with station numbering.
»  CO line numbers are fixed and cannot be changed other than the current FLASH 42 
reassignment.
»  The VMID digits need to be programmed separately.
»  The SMDR will output up to 4-digit numbers in the station field. If less than 4 digits are 
selected in the numbering plan, leading spaces will be added in place of numbers. Call 
Accounting devices need to be configured to accept 4 digits.
»  When systems are tied together, each system has to have access to 911 through local 
lines.
Button 1 - Line group programming (00 to 23).
Button 2 - Station from (start) range – must be 4 digits.
Button 3 - Station to (end) range – must be 4 digits.
Button 4 - IP address to dial when any station within that range is dialed. For 
example, if station 3001 is dialed, the telephone switch connects 
the call using the IP address: 102. 38. 56. 1 (refer to 
Ta b l e D- 3  )
Button 22 - Page to next Networking table.
Button 23 - Page to previous Network ing table.
Button 24 - Enter a new Networking table.
Table D-3: System Networking Tables
System CO GroupStation Range
   From ToIP Address
01 1 1000 1200 172.016.016.001
02 2 2000 2400 209.054.002.001
03 2 3000 3500 102.038.056.001
04
05
06
07
08
15
16 
						

							OperationD-13
Appendix D - Voice Over the Internet Protocol
Operation
The Discovery VoIP card operates like all other Central Office cards. There are up to 8 lines per 
card, and each line functions as any other CO line functions. It allows bi-directional 
communication and monitors for disconnect while using minimal bandwidth. It also allows 
four-digit dialing and other features.
When using Discovery VoIP lines:

If a button is programmed, a busy lamp field (BLF) will illuminate (on initiating station 
only) when a CO line is in use.

Calling can be done using speakerphone, handset, or headset; bi-directional 
communication is the default.

After a call is connected, it has the ability to use End to End signaling, which allows DTMF 
to be sent over the connection. This is used for functions such as remote VM systems.

Each line monitors for disconnect signaling, simulating Loop Supervision. When a line is 
off hook for 5 seconds or more, and no action is taken, an error tone sounds.
Network Connection
When the network cable is removed, or the network goes down, all COs will go out of service. 
All calls will be denied and an error tone will be heard if the line is accessed.
Dialing
Use Networking Tables (FLASH 16) to program stations to automatically use the appropriate 
line group and IP connection for a dialed number.
Line Access
Off Net Forward
A line is allowed to be transferred to a station that is Off Net Forwarded to a remote location.
DTMF Only
The VoIP card only supports tone dialing (DTMF).
911 Support
The Discovery VoIP card does not support 911 calls. Therefore, the system cannot solely use 
Discovery VoIP CO cards.
Station Access
Discovery VoIP CO access for each station is allowed or denied access to these trunks through 
standard CO Line access. 
						

							D-14 IP Telephony Standards and Protocols
Appendix D - Voice Over the Internet Protocol
IP Telephony Standards and Protocols
The Discovery VoIP card uses the H.323 Revision standard for call processing. This is an 
updated version of the H.323 standard. It provides a set of standards defining real-time 
multimedia communications and conferencing over packet-based networks. These 
standards define how components that are built in compliance with H.323 set up calls, 
exchange compressed audio and video, participate in multi-unit conferences, and operate 
with non-H.323 endpoints.
The IP telephony standards/protocols shown in Ta b le  D - 4 are adhered to by the Discovery 
VoIP card.
Table D-4: IP Telephony Standards/Protocols
Standard Description
H.323 Revision 2 Supports H.323 terminals and is the basis for all IP telephony.
H.225 Media Packetization Provides media packetization and synchronization for video/
audio telephony on non-guaranteed quality of service LANs.
H.245 DTMF Signaling
Close Logic Signaling
Round Trip SignalArbitration of GSM compression -- provides the new audio 
capabilities and supports the signaling entities required for call 
control functions for multimedia communications, while it 
specifies the in-band signaling protocol necessary to establish a 
call, determine capabilities, and issue the commands necessary 
to open/close the media channels.
Supports signaling of a pair of associated unidirectional channels 
which allows for the establishment of a T.120 data channel.
Allows determination of the round trip delay between two 
communicating channels.
G.165 Echo Cancellation When echo is present, a preprogrammed button is available to 
increase or decrease the latency of that call. Upon termination of 
each call, the unit restores the default setting.
G.711 Pulse Code Modulation of 
Voice FrequenciesTransmits and receives A-law and U-law PCM voice at digital bit rates 
of 48, 56, and 64 Kbps. It is used for digital telephone sets on digital 
PBX and ISDN channels. Support for this algorithm is required for 
ITU-T compliant videoconferencing. (slowest protocol)
G.723.1 5.3 Kbps compression for 
dual rate speech coders 
for multimedia 
communicationsRuns at 6.3 or 5.4 Kbps a compression and uses linear predictive 
coding directory for an open architecture to ensure connectivity 
with other switches. Moreover, this compression helps provide 
smoothness. (Default)
G.729 Speech encoding 8 Kbps Encodes/decodes speech at the rate of 8 Kbps using conjugate-
structure, algebraic-code excited linear predictive methods.
Q.931 Messages:
- Progress
- Setup
- AcknowledgmentCreates a unique global identifier that allows all messages 
associated with a call to be interoperable between the 
registration, administration and status protocol used in IP 
networks utilizing the Q.931 signaling protocol used in 
circuit-switched telephony networks. 
Supports messages that are used for call signaling, including all 
mandatory and conditionally mandatory messages, some 
optional messages and information elements, and the facility 
message defined in Q.931 and Q.932. 
						

							Vo I P  Gl o s s a r yD-15
Appendix D - Voice Over the Internet Protocol
Vo I P  G l o s s a r y
This table describes the Internet Telephony terms used in this section:
Term Definition
Asynchronous
TransmissionA method of data transmission which allows characters to be sent at irregular intervals by preceding each 
character with a start bit and following it with a stop bit
bps Data bits per second, also known as Baud Rate
DTMF Dual Tone Multi-Frequency
Gateway Bridges H.323 conferences to other network communications protocol and multimedia formats
Header Protocol control information located at the beginning of a protocol data unit. This portion of a message 
contains information that will guide the message to the correct destination by including sender’s and 
receiver’s addresses, routing instructions, etc.
Hub A physical connection for multiple LAN devices. Commonly, 10/100 Base-T Ethernet support through a RJ-45 
connection is provided.
IANA Internet Assigned Number Authority – the one agency that issues all IP addresses
ISP Internet Service Provider. A vendor who provides access to the Internet and World Wide Web.
Jitter Network-provided variations in latency for different packets, which is particularly disruptive to audio 
communications.
LAN  Local Area Network
Latency A term used to indicate waiting time or time delay in delivering packets over a network.
Octet Three digits between decimal points in an IP address. There are eight binary bits for numbers from 0-255, 
making it an octet.
PSTN Public Switching Telephone Network
QoS Quality of Service
Router A unit that routes Packeted information interfacing two separate networks
TTL Time to Live. Used with the IP protocol, it is the time after which the packet can be deleted from the network. 
This is typically measured in milliseconds.
VoIP Voice Over Internet Protocol
VPN  Virtual Private Network. The definition is very broad. Typically, this means some form of “virtually private” 
network created over a public network (i.e., the Internet) using encryption technologies to create a secure 
connection between two or more sites.
WAN Wide Area Network. A computer or voice network bigger than a metropolitan area. Sometimes used to define 
a network that spans a metropolitan area which could also be called a MAN (Metropolitan Area Network). 
						

							D-16VoIP Glossary
Appendix D - Voice Over the Internet Protocol 
						

							E
Customer Database
Programming
This appendix provides information about database programming. Use the detailed
procedures contained in other chapters and appendices for actual programming via
executive display telephone. Use the Customer Database Worksheets in this appendix
to help keep track of the system programming changes made for each individual
system. 
						

							











 
 
 
 
 
 
						

							IntroductionE-3
Appendix E - Customer Database Programming
Introduction
The XTS system is programmed to meet each customer’s individual needs. All programming 
is done at any 24-Button Executive Telephone as the programming station or through an 
ASCII terminal or PC. The digital display model is required for programming.
When the programming mode is entered, the digital telephone being used no longer 
operates as a telephone but as a programming station with all of the buttons redefined. The 
keys on the dial pad are used to enter data fields (Program Codes) associated with system, 
station, and CO line features as well as specific data that requires a numeric entry. Flexible 
buttons toggle on or off features, or enable entry of specific data fields. LEDs and the LCD 
display provide a visual indication of entered data and their value.
Programming is also performed using an ASCII terminal, or a computer capable of emulating 
an ASCII terminal. This form of programming is done locally (on-site) by connecting the 
terminal directly to the RS-232C connector on the Main Processing Board (MPB) or is 
performed remotely (off-site) through the use of the 19.2K baud modem. The method and 
steps to program the system via a PC are identical to those used when programming from a 
digital key set. A button to keyboard mapping is provided (refer to  Figure E-2 on page E-4) to 
help minimize familiarization and training time.
The system must be initialized to load default data into memory at the time of installation. If 
this pre-programming is acceptable to the customer, initialization is all that is needed. Refer 
to Ta b le F - 1  o n  p a g e F - 3 for a listing of all the default values. 
When features are programmed, tones are provided to determine if a correct or incorrect 
entry has been made. A solid one second tone indicates the data was accepted. An 
interrupted tone means an error was made. When this occurs, re-enter the data and 
information.
Until new data is entered and accepted, the system continues to operate under default or 
previously entered values.
The system database is updated on a real-time basis as new data is entered, by pressing the 
Hold button. The system continues to operate with the current database and is updated with 
any newly entered or changed data without interruption to telephone operation or call 
processing in progress. However, if for example a station’s attributes are changed while that 
station is off-hook on an active call, the newly entered data does not take effect until the 
station goes on-hook or becomes idle. 
						

							E-4Introduction
Appendix E - Customer Database Programming
When using a PC to program the system, the following chart presents the data terminal 
characters that are equivalent to the key set buttons.
Figure E-1: Data Terminal Program Codes Cross Reference
Figure E-2: Programming Button Mapping
Some features must have more than one data field programmed for that feature 
to work. This information is stated in the instructions.


	





	




	





	





	























































 




























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